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RE: [PATCH v1 2/2] ASoC: dt-bindings: Add Everest ES8389 audio CODEC
by Zhang Yi 03 Mar '25

03 Mar '25

> > + everest,adc-slot: > > + $ref: /schemas/types.yaml#/definitions/uint8 > > + description: | > > + This property is used to set the slots of recording data when multiple > > + codecs are connected in PTDM mode. > > + please set this property to default if you are setting STDM mode. > > + minimum: 0x00 > > + maximum: 0x07 > > + default: 0x00 > > + > > + everest,dac-slot: > > + $ref: /schemas/types.yaml#/definitions/uint8 > > + description: | > > + This property is used to set the slots of playing data when multiple > > + codecs are connected in TDM mode. > > + please do not set this property if you use single codec. > > + minimum: 0x00 > > + maximum: 0x07 > > + default: 0x00 > > Can't the dai-tdm-* properties work for you? We will register multiple codecs inside a single dai_link and differentiate these codecs by of_node. so we can't use dai-tdm-slot* to define the slot for each codec > > + > > + prefix_name: > > + $ref: /schemas/types.yaml#/definitions/string > > + description: device name prefix > > Why do you need this? When using multiple codecs, we utilize prefix_name to distinguish the dapm and kcontro of each individual codec.

1 0

Turn on LED function
by Kailang 03 Mar '25

03 Mar '25

Hi Takashi, Support Mic Mute LED for Lenovo ThinkCentre M series. BR, Kailang

2 1

Fine-grained control bar
by GitHub issues - opened 02 Mar '25

02 Mar '25

alsa-project/alsa-utils issue #295 was opened from egmontkob: The control bars of `alsamixer` are quite coarse. They jump by whole character cells. Often a notable amount of increase or decrease (pressing the Up or Down arrow) isn't reflected visually. I'd like to ask for an opion for more fine-grained looking control bars. If the terminal's / locale's claimed charset, as reported by `nl_langinfo(CODESET)`, is "UTF-8" (as it should be pretty much everywhere nowadays), you could use the Unicode characters [U+2581 to U+2588](https://www.unicode.org/charts/PDF/U2580.pdf) for this purpose. They jump by eighths of a cell's hight. Their foreground color would be the desired bar color (red/white/green) and their background would be the default (or black for `-B`). For consistency, this mode should probably replace the current checkerboard (ACS_CKBOARD) characters by the full block U+2588 and only set its foreground, but maybe this doesn't matter. There are several terminal emulators out there (at least VTE-based ones such as GNOME Terminal and many others; Konsole; and presumably much more) that don't take these glyphs from the font but rather draw them manually to perfectly align them at cell boundaries. In these terminals the bar would look perfectly. In other terminals there might be some ugly gaps at cell boundaries (subject to the selected font), but some people might still prefer this tiny visual glitch over the current coarseness of the bar (again: this would be an option, not a forced new behavior). Thanks for considering this feature request! Issue URL : https://github.com/alsa-project/alsa-utils/issues/295 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

amixer.1: Some remarks and a patch with editorial changes for this man page
by Bjarni Ingi Gislason 01 Mar '25

01 Mar '25

Package: alsa-utils Version: 1.2.13-1 Severity: minor Tags: patch * What led up to the situation? Checking for defects with a new version test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -rCHECKSTYLE=10 -ww -z < "man page" [Use "groff -e ' $' -e '\\~$' <file>" to find obvious trailing spaces.] ["test-groff" is a script in the repository for "groff"; is not shipped] (local copy and "troff" slightly changed by me). [The fate of "test-nroff" was decided in groff bug #55941.] * What was the outcome of this action? an.tmac:<stdin>:1: style: .TH missing fourth argument; consider package/project name and version (e.g., "groff 1.23.0") troff:<stdin>:124: warning: trailing space in the line troff:<stdin>:133: warning: trailing space in the line troff:<stdin>:138: warning: trailing space in the line * What outcome did you expect instead? No output (no warnings). -.- General remarks and further material, if a diff-file exist, are in the attachments. -- System Information: Debian Release: trixie/sid APT prefers testing APT policy: (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 6.12.12-amd64 (SMP w/2 CPU threads; PREEMPT) Locale: LANG=is_IS.iso88591, LC_CTYPE=is_IS.iso88591 (charmap=ISO-8859-1), LANGUAGE not set Shell: /bin/sh linked to /usr/bin/dash Init: sysvinit (via /sbin/init) Versions of packages alsa-utils depends on: ii kmod 33+20240816-2 ii libasound2t64 1.2.13-1+b1 ii libatopology2t64 1.2.13-1+b1 ii libc6 2.40-7 ii libfftw3-single3 3.3.10-2+b1 ii libncursesw6 6.5+20250216-1 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5+20250216-1 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

1 0

alsaucm.1: Some remarks and a patch with editorial changes for this man page
by Bjarni Ingi Gislason 01 Mar '25

01 Mar '25

Package: alsa-utils Version: 1.2.13-1 Severity: minor Tags: patch * What led up to the situation? Checking for defects with a new version test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -rCHECKSTYLE=10 -ww -z < "man page" [Use "groff -e ' $' -e '\\~$' <file>" to find obvious trailing spaces.] ["test-groff" is a script in the repository for "groff"; is not shipped] (local copy and "troff" slightly changed by me). [The fate of "test-nroff" was decided in groff bug #55941.] * What was the outcome of this action? an.tmac:<stdin>:30: style: .TH missing fourth argument; consider package/project name and version (e.g., "groff 1.23.0") * What outcome did you expect instead? No output (no warnings). -.- General remarks and further material, if a diff-file exist, are in the attachments. -- System Information: Debian Release: trixie/sid APT prefers testing APT policy: (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 6.12.12-amd64 (SMP w/2 CPU threads; PREEMPT) Locale: LANG=is_IS.iso88591, LC_CTYPE=is_IS.iso88591 (charmap=ISO-8859-1), LANGUAGE not set Shell: /bin/sh linked to /usr/bin/dash Init: sysvinit (via /sbin/init) Versions of packages alsa-utils depends on: ii kmod 33+20240816-2 ii libasound2t64 1.2.13-1+b1 ii libatopology2t64 1.2.13-1+b1 ii libc6 2.40-7 ii libfftw3-single3 3.3.10-2+b1 ii libncursesw6 6.5+20250216-1 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5+20250216-1 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

1 0

unclear example how to use remap plugin
by GitHub issues - opened 01 Mar '25

01 Mar '25

alsa-project/alsa-lib issue #441 was opened from dibalavs: Hello. I combiled 3 sound cards into one vrurtual 5.1 card for PCM, and now trying to make volume control also to use single virtual device, to set up volume for all chanel of virtual 5.1 card. I am trying to use remap control plugin mentioned at [source](https://github.com/alsa-project/alsa-lib/blob/master/src/control/co… But I unable to understand what does it mean: ``` ctl.name { type remap # Remap controls child STR # Child name # or child { # Child definition type STR ... } remap { # the ID strings are parsed in the amixer style like 'name="Headphone Playback Switch",index=2' SRC_ID1_STR DST_ID1_STR SRC_ID2_STR DST_ID2_STR ... } map { # join two stereo controls to one CREATE_ID1_STR { SRC_ID1_STR { vindex.0 0 # source channel 0 to merged channel 0 vindex.1 1 } SRC_ID2_STR { vindex.2 0 vindex.3 1 # source channel 1 to merged channel 3 } } # split stereo to mono CREATE_ID2_STR { SRC_ID3_STR { vindex.0 0 # stereo to mono (first channel) } } CREATE_ID3_STR { SRC_ID4_STR { vindex.0 1 # stereo to mono (second channel) } } } } ``` I do not understand what should I write as SRC and DST?: ``` # the ID strings are parsed in the amixer style like 'name="Headphone Playback Switch",index=2' SRC_ID1_STR DST_ID1_STR SRC_ID2_STR DST_ID2_STR ``` Can you give real example without placeholders like: "name", "SRC", "DST", "STR" ? Issue URL : https://github.com/alsa-project/alsa-lib/issues/441 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

alsatplg.1: Some remarks and a patch with editorial changes for this man page
by Bjarni Ingi Gislason 01 Mar '25

01 Mar '25

Package: alsa-utils Version: 1.2.13-1 Severity: minor Tags: patch * What led up to the situation? Checking for defects with a new version test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -rCHECKSTYLE=10 -ww -z < "man page" [Use "groff -e ' $' -e '\\~$' <file>" to find obvious trailing spaces.] ["test-groff" is a script in the repository for "groff"; is not shipped] (local copy and "troff" slightly changed by me). [The fate of "test-nroff" was decided in groff bug #55941.] * What was the outcome of this action? an.tmac:<stdin>:30: style: .TH missing fourth argument; consider package/project name and version (e.g., "groff 1.23.0") an.tmac:<stdin>:108: style: 1 leading space(s) on input line troff:<stdin>:108: warning: trailing space in the line troff:<stdin>:112: warning: trailing space in the line * What outcome did you expect instead? No output (no warnings). -.- General remarks and further material, if a diff-file exist, are in the attachments. -- System Information: Debian Release: trixie/sid APT prefers testing APT policy: (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 6.12.12-amd64 (SMP w/2 CPU threads; PREEMPT) Locale: LANG=is_IS.iso88591, LC_CTYPE=is_IS.iso88591 (charmap=ISO-8859-1), LANGUAGE not set Shell: /bin/sh linked to /usr/bin/dash Init: sysvinit (via /sbin/init) Versions of packages alsa-utils depends on: ii kmod 33+20240816-2 ii libasound2t64 1.2.13-1+b1 ii libatopology2t64 1.2.13-1+b1 ii libc6 2.40-7 ii libfftw3-single3 3.3.10-2+b1 ii libncursesw6 6.5+20250216-1 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5+20250216-1 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

1 0

Build with gettext-0.24 intermittently fails
by GitHub issues - edited 28 Feb '25

28 Feb '25

alsa-project/alsa-utils issue #294 was edited from heitbaum: After Autoconf the errors are as follows. The Makefile is not honouring / passing the builddir through. This is a build being done in a subdirectory. ``` Making all in po make[2]: Entering directory '/build/build.LibreELEC-Generic.x86_64-13.0-devel/build/alsa-utils-1.2.13/.x86_64-libreelec-linux-gnu/po' make[2]: *** No rule to make target '/config.status', needed by 'Makefile'. Stop. make[2]: Leaving directory '/build/build.LibreELEC-Generic.x86_64-13.0-devel/build/alsa-utils-1.2.13/.x86_64-libreelec-linux-gnu/po' make[1]: *** [Makefile:471: all-recursive] Error 1 make[1]: Leaving directory '/build/build.LibreELEC-Generic.x86_64-13.0-devel/build/alsa-utils-1.2.13/.x86_64-libreelec-linux-gnu' ``` - This is occurring just like https://github.com/zapping-vbi/zvbi/issues/57 - now testing to see if the change to `AM_GNU_GETTEXT_VERSION([0.24])` fixes the issue Issue URL : https://github.com/alsa-project/alsa-utils/issues/294 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Build with gettext-0.24 intermittently fails
by GitHub issues - opened 28 Feb '25

28 Feb '25

alsa-project/alsa-utils issue #294 was opened from heitbaum: After Autoconf the errors are as follows. The Makefile is not honouring / passing the builddir through. This is a build being done in a subdirectory. ``` Making all in po make[2]: Entering directory '/build/build.LibreELEC-Generic.x86_64-13.0-devel/build/alsa-utils-1.2.13/.x86_64-libreelec-linux-gnu/po' make[2]: *** No rule to make target '/config.status', needed by 'Makefile'. Stop. make[2]: Leaving directory '/build/build.LibreELEC-Generic.x86_64-13.0-devel/build/alsa-utils-1.2.13/.x86_64-libreelec-linux-gnu/po' make[1]: *** [Makefile:471: all-recursive] Error 1 make[1]: Leaving directory '/build/build.LibreELEC-Generic.x86_64-13.0-devel/build/alsa-utils-1.2.13/.x86_64-libreelec-linux-gnu' ``` Issue URL : https://github.com/alsa-project/alsa-utils/issues/294 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

How to clone from https://git.alsa-project.org/?
by GitHub issues - opened 28 Feb '25

28 Feb '25

alsa-project/alsa-utils issue #293 was opened from K0-RR: `git clone https://git.alsa-project.org/?p=alsa.git` gives me fatal: repository 'https://git.alsa-project.org/?p=alsa.git/' not found Issue URL : https://github.com/alsa-project/alsa-utils/issues/293 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v3] ASoC: tas2781: Support dsp firmware Alpha and Beta seaies
by Shenghao Ding 27 Feb '25

27 Feb '25

For calibration, basic version does not contain any calibration addresses, it depends on calibration tool to convery the addresses to the driver. Since Alpha and Beta firmware, all the calibration addresses are saved into the firmware. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v3: - Merge the 4 patches into one patch v2: - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: | https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/2025 | 02160812.P7szOygN-lkp(a)intel.com/__;!!G3vK!WriwhtbQGlg6p00bcWutlhb-y9vS | pDrSu5_g9oOzzGOoABdirEARTntFzyX6Ss3UPRDxi_E-IA$ - | Reported-by: kernel test robot <lkp(a)intel.com> | Closes: | https://urldefense.com/v3/__https://lore.kernel.org/oe-kbuild-all/2025 | 02160632.u0zxG7yD-lkp(a)intel.com/__;!!G3vK!VtEdfur3fZklvPVfHFFfFP60Ub2E | JYvmjSUZ65lJ6BsQPJDl1fjYRDqh3c6Q5y6sA70ssXDG5g$ v1: - update the year from 2024 to 2025 in the files. - Add dspbin_type to distinguish different dspbin types, such as TASDEV_BASIC, TASDEV_ALPHA, and TASDEV_BETA. - Add struct bulk_reg_val alp_cali_bckp to store the value of pilot tone threshold only for TASDEV_ALPHA & TASDEV_BETA firmware. - Add the declaration of (*fw_parse_fct_param_address)() to parse the addresses of calibration params. - Add the minimum version of TASDEV_ALPHA and TASDEV_BETA firmware. - Add the definition of struct fct_param_address. - update the year from 2024 to 2025. - release the memory to save calibration addresses in tasdevice_dsp_remove(). - Add TASDEV_ALPHA and TASDEV_BETA firmware checking code. - Add fct_param_address_parser for Alpha and Beta firmware. - Add an extra branch to handle calibration address for TASDEV_ALPHA and TASDEV_BETA firmware. --- include/sound/tas2781-dsp.h | 26 +++- include/sound/tas2781.h | 13 +- sound/pci/hda/tas2781_spi_fwlib.c | 6 +- sound/soc/codecs/tas2781-comlib.c | 3 +- sound/soc/codecs/tas2781-fmwlib.c | 149 +++++++++++++++++++++- sound/soc/codecs/tas2781-i2c.c | 203 +++++++++++++++++++++++------- 6 files changed, 340 insertions(+), 60 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 3cda9da14f6d..77a240daaae0 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -30,8 +30,10 @@ #define PRE_DEVICE_C 0x12 #define PRE_DEVICE_D 0x16 -#define PPC3_VERSION 0x4100 -#define PPC3_VERSION_TAS2781 0x14600 +#define PPC3_VERSION_BASE 0x4100 +#define PPC3_VERSION_TAS2781_BASIC_MIN 0x14600 +#define PPC3_VERSION_TAS2781_ALPHA_MIN 0x4a00 +#define PPC3_VERSION_TAS2781_BETA_MIN 0x19400 #define TASDEVICE_DEVICE_SUM 8 #define TASDEVICE_CONFIG_SUM 64 @@ -106,6 +108,23 @@ struct tasdevice_calibration { struct tasdevice_data dev_data; }; +struct fct_param_address { + unsigned char thr[3]; + unsigned char thr2[3]; + unsigned char plt_flg[3]; + unsigned char sin_gn[3]; + unsigned char sin_gn2[3]; + /* real-time spk impedance */ + unsigned char r0_reg[3]; + /* check spk connection */ + unsigned char tf_reg[3]; + /* check spk resonant frequency */ + unsigned char a1_reg[3]; + unsigned char a2_reg[3]; + unsigned short size; + char *data; +}; + struct tasdevice_fw { struct tasdevice_dspfw_hdr fw_hdr; unsigned short nr_programs; @@ -114,6 +133,7 @@ struct tasdevice_fw { struct tasdevice_config *configs; unsigned short nr_calibrations; struct tasdevice_calibration *calibrations; + struct fct_param_address fct_par_addr; struct device *dev; }; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 72d2060904f6..5cfbb9e0a626 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -112,6 +112,12 @@ enum audio_device { TAS2781, }; +enum dspbin_type { + TASDEV_BASIC, + TASDEV_ALPHA, + TASDEV_BETA, +}; + enum device_catlog_id { LENOVO = 0, OTHERS @@ -126,6 +132,7 @@ struct bulk_reg_val { struct tasdevice { struct bulk_reg_val *cali_data_backup; + struct bulk_reg_val alp_cali_bckp; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -171,6 +178,7 @@ struct tasdevice_priv { unsigned char dev_name[32]; const char *name_prefix; unsigned char ndev; + unsigned int dspbin_typ; unsigned int magic_num; unsigned int chip_id; unsigned int sysclk; @@ -196,6 +204,9 @@ struct tasdevice_priv { int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset); + int (*fw_parse_fct_param_address)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, struct tasdev_blk *block); diff --git a/sound/pci/hda/tas2781_spi_fwlib.c b/sound/pci/hda/tas2781_spi_fwlib.c index 131d9a77d140..c90641119509 100644 --- a/sound/pci/hda/tas2781_spi_fwlib.c +++ b/sound/pci/hda/tas2781_spi_fwlib.c @@ -370,10 +370,10 @@ static unsigned char map_dev_idx(struct tasdevice_fw *tas_fmw, int i, n = ARRAY_SIZE(non_ppc3_mapping_table); unsigned char dev_idx = 0; - if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781) { + if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781_BASIC_MIN) { p = (struct blktyp_devidx_map *)ppc3_tas2781_mapping_table; n = ARRAY_SIZE(ppc3_tas2781_mapping_table); - } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION) { + } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION_BASE) { p = (struct blktyp_devidx_map *)ppc3_mapping_table; n = ARRAY_SIZE(ppc3_mapping_table); } @@ -1605,7 +1605,7 @@ static int dspfw_default_callback(struct tasdevice_priv *tas_priv, int rc = 0; if (drv_ver == 0x100) { - if (ppcver >= PPC3_VERSION) { + if (ppcver >= PPC3_VERSION_BASE) { tas_priv->fw_parse_variable_header = fw_parse_variable_header_kernel; tas_priv->fw_parse_program_data = diff --git a/sound/soc/codecs/tas2781-comlib.c b/sound/soc/codecs/tas2781-comlib.c index 1e0b3aa95749..bba5abe4736f 100644 --- a/sound/soc/codecs/tas2781-comlib.c +++ b/sound/soc/codecs/tas2781-comlib.c @@ -2,7 +2,7 @@ // // TAS2563/TAS2781 Common functions for HDA and ASoC Audio drivers // -// Copyright 2023 - 2024 Texas Instruments, Inc. +// Copyright 2023 - 2025 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> @@ -429,6 +429,7 @@ void tasdevice_dsp_remove(void *context) if (tas_fmw->configs) tasdev_dsp_cfg_remove(tas_fmw->configs, tas_fmw->nr_configurations); + kfree(tas_fmw->fct_par_addr.data); kfree(tas_fmw); tas_dev->fmw = NULL; } diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 61d9c220b6a4..2fae7be99454 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2,7 +2,7 @@ // // tas2781-fmwlib.c -- TASDEVICE firmware support // -// Copyright 2023 - 2024 Texas Instruments, Inc. +// Copyright 2023 - 2025 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> @@ -389,10 +389,10 @@ static unsigned char map_dev_idx(struct tasdevice_fw *tas_fmw, int i, n = ARRAY_SIZE(non_ppc3_mapping_table); unsigned char dev_idx = 0; - if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781) { + if (fw_fixed_hdr->ppcver >= PPC3_VERSION_TAS2781_BASIC_MIN) { p = (struct blktyp_devidx_map *)ppc3_tas2781_mapping_table; n = ARRAY_SIZE(ppc3_tas2781_mapping_table); - } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION) { + } else if (fw_fixed_hdr->ppcver >= PPC3_VERSION_BASE) { p = (struct blktyp_devidx_map *)ppc3_mapping_table; n = ARRAY_SIZE(ppc3_mapping_table); } @@ -559,6 +559,118 @@ static int fw_parse_configuration_data_kernel( return offset; } +static void hex_parse_u8(unsigned char *item, const unsigned char *val, + int len) +{ + for (int i = 0; i < len; i++) + item[i] = val[i]; +} + +static void hex_parse_u24(unsigned int *item, const unsigned char *val) +{ + *item = TASDEVICE_REG(val[0], val[1], val[2]); +} + +static void fct_param_address_parser(struct cali_reg *r, + struct tasdevice_fw *tas_fmw) +{ + struct fct_param_address *p = &(tas_fmw->fct_par_addr); + int i; + + for (i = 0; i < 20; i++) { + const unsigned char *dat = &p->data[24 * i]; + + if (dat[23] != 1) + break; + + if (strstr(dat, "umg_SsmKEGCye") != NULL) { + hex_parse_u24(&r->pow_reg, &dat[20]); + continue; + } + if (strstr(dat, "iks_E0") != NULL) { + hex_parse_u24(&r->r0_reg, &dat[20]); + continue; + } + if (strstr(dat, "yep_LsqM0") != NULL) { + hex_parse_u24(&r->invr0_reg, &dat[20]); + continue; + } + if (strstr(dat, "oyz_U0_ujx") != NULL) { + hex_parse_u24(&r->r0_low_reg, &dat[20]); + continue; + } + if (strstr(dat, "iks_GC_GMgq") != NULL) { + hex_parse_u24(&r->tlimit_reg, &dat[20]); + continue; + } + if (strstr(dat, "gou_Yao") != NULL) { + hex_parse_u8(p->thr, &dat[20], 3); + continue; + } + if (strstr(dat, "kgd_Wsc_Qsbp") != NULL) { + hex_parse_u8(p->plt_flg, &dat[20], 3); + continue; + } + if (strstr(dat, "yec_CqseSsqs") != NULL) { + hex_parse_u8(p->sin_gn, &dat[20], 3); + continue; + } + if (strstr(dat, "iks_SogkGgog2") != NULL) { + hex_parse_u8(p->sin_gn2, &dat[20], 3); + continue; + } + if (strstr(dat, "yec_Sae_Y") != NULL) { + hex_parse_u8(p->thr2, &dat[20], 3); + continue; + } + if (strstr(dat, "Re_Int") != NULL) { + hex_parse_u8(p->r0_reg, &dat[20], 3); + continue; + } + /* Check whether the spk connection is open */ + if (strstr(dat, "SigFlag") != NULL) { + hex_parse_u8(p->tf_reg, &dat[20], 3); + continue; + } + if (strstr(dat, "a1_Int") != NULL) { + hex_parse_u8(p->a1_reg, &dat[20], 3); + continue; + } + if (strstr(dat, "a2_Int") != NULL) { + hex_parse_u8(p->a2_reg, &dat[20], 3); + continue; + } + } +} + +static int fw_parse_fct_param_address(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset) +{ + struct fct_param_address *p = &(tas_fmw->fct_par_addr); + struct calidata *cali_data = &tas_priv->cali_data; + struct cali_reg *r = &cali_data->cali_reg_array; + const unsigned char *data = fmw->data; + + if (offset + 520 > fmw->size) { + dev_err(tas_priv->dev, "%s: File Size error\n", __func__); + offset = -1; + goto out; + } + + /* skip reserved part */ + offset += 40; + p->size = 480; + p->data = kmemdup(&data[offset], 480, GFP_KERNEL); + if (!p->data) { + offset = -1; + goto out; + } + offset += 480; + fct_param_address_parser(r, tas_fmw); +out: + return offset; +} + static int fw_parse_variable_header_kernel( struct tasdevice_priv *tas_priv, const struct firmware *fmw, int offset) @@ -1686,13 +1798,29 @@ static int tasdevice_load_block(struct tasdevice_priv *tas_priv, return rc; } +static void dspbin_type_check(struct tasdevice_priv *tas_priv, + unsigned int ppcver) +{ + if (ppcver >= PPC3_VERSION_TAS2781_ALPHA_MIN) { + if (ppcver >= PPC3_VERSION_TAS2781_BETA_MIN) + tas_priv->dspbin_typ = TASDEV_BETA; + else if (ppcver >= PPC3_VERSION_TAS2781_BASIC_MIN) + tas_priv->dspbin_typ = TASDEV_BASIC; + else + tas_priv->dspbin_typ = TASDEV_ALPHA; + } + if (tas_priv->dspbin_typ != TASDEV_BASIC) + tas_priv->fw_parse_fct_param_address = + fw_parse_fct_param_address; +} + static int dspfw_default_callback(struct tasdevice_priv *tas_priv, unsigned int drv_ver, unsigned int ppcver) { int rc = 0; if (drv_ver == 0x100) { - if (ppcver >= PPC3_VERSION) { + if (ppcver >= PPC3_VERSION_BASE) { tas_priv->fw_parse_variable_header = fw_parse_variable_header_kernel; tas_priv->fw_parse_program_data = @@ -1701,6 +1829,7 @@ static int dspfw_default_callback(struct tasdevice_priv *tas_priv, fw_parse_configuration_data_kernel; tas_priv->tasdevice_load_block = tasdevice_load_block_kernel; + dspbin_type_check(tas_priv, ppcver); } else { switch (ppcver) { case 0x00: @@ -1716,7 +1845,7 @@ static int dspfw_default_callback(struct tasdevice_priv *tas_priv, default: dev_err(tas_priv->dev, "%s: PPCVer must be 0x0 or 0x%02x", - __func__, PPC3_VERSION); + __func__, PPC3_VERSION_BASE); dev_err(tas_priv->dev, " Current:0x%02x\n", ppcver); rc = -EINVAL; @@ -2023,9 +2152,17 @@ static int tasdevice_dspfw_ready(const struct firmware *fmw, } offset = tas_priv->fw_parse_configuration_data(tas_priv, tas_fmw, fmw, offset); - if (offset < 0) + if (offset < 0) { ret = offset; + goto out; + } + if (tas_priv->fw_parse_fct_param_address) { + offset = tas_priv->fw_parse_fct_param_address(tas_priv, + tas_fmw, fmw, offset); + if (offset < 0) + ret = offset; + } out: return ret; } diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 90c5b2e74d12..8eb9fee27a42 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -349,13 +349,31 @@ static int calib_data_get(struct tasdevice_priv *tas_priv, int reg, return rc; } +static int partial_cali_data_update(int *reg, int j) +{ + switch (tas2781_cali_start_reg[j].reg) { + case 0: + return reg[0]; + case TAS2781_PRM_PLT_FLAG_REG: + return reg[1]; + case TAS2781_PRM_SINEGAIN_REG: + return reg[2]; + case TAS2781_PRM_SINEGAIN2_REG: + return reg[3]; + default: + return 0; + } +} + static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, int *reg, unsigned char *dat) { struct tasdevice *tasdev = tas_priv->tasdevice; struct bulk_reg_val *p = tasdev[i].cali_data_backup; + struct bulk_reg_val *t = &tasdev[i].alp_cali_bckp; const int sum = ARRAY_SIZE(tas2781_cali_start_reg); - int j; + unsigned char val[4]; + int j, r; if (p == NULL) return; @@ -370,30 +388,23 @@ static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, tasdevice_dev_read(tas_priv, i, p[j].reg, (int *)&p[j].val[0]); } else { - switch (tas2781_cali_start_reg[j].reg) { - case 0: { - if (!reg[0]) - continue; - p[j].reg = reg[0]; + if (!tas_priv->dspbin_typ) { + r = partial_cali_data_update(reg, j); + if (r) + p[j].reg = r; } - break; - case TAS2781_PRM_PLT_FLAG_REG: - p[j].reg = reg[1]; - break; - case TAS2781_PRM_SINEGAIN_REG: - p[j].reg = reg[2]; - break; - case TAS2781_PRM_SINEGAIN2_REG: - p[j].reg = reg[3]; - break; - } - tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, - p[j].val, 4); + + if (p[j].reg) + tasdevice_dev_bulk_read(tas_priv, i, p[j].reg, + p[j].val, 4); } } + if (tas_priv->dspbin_typ == TASDEV_ALPHA) + tasdevice_dev_bulk_read(tas_priv, i, t->reg, t->val, 4); + /* Update the setting for calibration */ - for (j = 0; j < sum - 2; j++) { + for (j = 0; j < sum - 4; j++) { if (p[j].val_len == 1) { if (p[j].is_locked) tasdevice_dev_write(tas_priv, i, @@ -401,17 +412,33 @@ static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, TAS2781_TEST_PAGE_UNLOCK); tasdevice_dev_write(tas_priv, i, p[j].reg, tas2781_cali_start_reg[j].val[0]); - } else { - if (!p[j].reg) - continue; - tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, - (unsigned char *) - tas2781_cali_start_reg[j].val, 4); } } - tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, &dat[1], 4); - tasdevice_dev_bulk_write(tas_priv, i, p[j + 1].reg, &dat[5], 4); + if (tas_priv->dspbin_typ == TASDEV_ALPHA) { + val[0] = 0x00; + val[1] = 0x00; + val[2] = 0x21; + val[3] = 0x8e; + } else { + val[0] = tas2781_cali_start_reg[j].val[0]; + val[1] = tas2781_cali_start_reg[j].val[1]; + val[2] = tas2781_cali_start_reg[j].val[2]; + val[3] = tas2781_cali_start_reg[j].val[3]; + } + tasdevice_dev_bulk_write(tas_priv, i, p[j].reg, val, 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 1].reg, + (unsigned char *)tas2781_cali_start_reg[j + 1].val, 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 2].reg, &dat[1], 4); + tasdevice_dev_bulk_write(tas_priv, i, p[j + 3].reg, &dat[5], 4); + if (tas_priv->dspbin_typ == TASDEV_ALPHA) { + val[0] = 0x00; + val[1] = 0x00; + val[2] = 0x2a; + val[3] = 0x0b; + + tasdevice_dev_bulk_read(tas_priv, i, t->reg, val, 4); + } } static int tas2781_calib_start_put(struct snd_kcontrol *kcontrol, @@ -460,6 +487,7 @@ static void tas2781_calib_stop_put(struct tasdevice_priv *tas_priv) for (i = 0; i < tas_priv->ndev; i++) { struct tasdevice *tasdev = tas_priv->tasdevice; struct bulk_reg_val *p = tasdev[i].cali_data_backup; + struct bulk_reg_val *t = &tasdev[i].alp_cali_bckp; if (p == NULL) continue; @@ -479,6 +507,10 @@ static void tas2781_calib_stop_put(struct tasdevice_priv *tas_priv) p[j].val, 4); } } + + if (tas_priv->dspbin_typ == TASDEV_ALPHA) + tasdevice_dev_bulk_write(tas_priv, i, t->reg, + t->val, 4); } } @@ -590,16 +622,20 @@ static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, i += 2; priv->is_user_space_calidata = true; - p->r0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->r0_low_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->invr0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->pow_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; - p->tlimit_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); - i += 3; + if (priv->dspbin_typ == TASDEV_BASIC) { + p->r0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->r0_low_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->invr0_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->pow_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + p->tlimit_reg = TASDEVICE_REG(src[i], src[i + 1], src[i + 2]); + i += 3; + } else { + i += 15; + } memcpy(dst, &src[i], cali_data->total_sz); return 1; @@ -646,12 +682,19 @@ static int tasdev_tf_data_get(struct snd_kcontrol *kcontrol, struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; - unsigned int reg; + unsigned int reg = TAS2781_RUNTIME_RE_REG_TF; + + if (tas_priv->chip_id == TAS2781) { + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); - if (tas_priv->chip_id == TAS2781) reg = TAS2781_RUNTIME_RE_REG_TF; - else + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->tf_reg[0], p->tf_reg[1], + p->tf_reg[2]); + } else { reg = TAS2563_RUNTIME_RE_REG_TF; + } guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; @@ -666,12 +709,19 @@ static int tasdev_re_data_get(struct snd_kcontrol *kcontrol, struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; - unsigned int reg; + unsigned int reg = TAS2781_RUNTIME_RE_REG; - if (tas_priv->chip_id == TAS2781) - reg = TAS2781_RUNTIME_RE_REG; - else + if (tas_priv->chip_id == TAS2781) { + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); + + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->r0_reg[0], p->r0_reg[1], + p->r0_reg[2]); + } else { reg = TAS2563_RUNTIME_RE_REG; + } + guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; return calib_data_get(tas_priv, reg, &dst[1]); @@ -705,11 +755,16 @@ static int tasdev_XMA1_data_get(struct snd_kcontrol *kcontrol, { struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A1_REG; + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->a1_reg[0], p->a1_reg[1], p->a1_reg[2]); + guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; return calib_data_get(tas_priv, reg, &dst[1]); @@ -720,11 +775,16 @@ static int tasdev_XMA2_data_get(struct snd_kcontrol *kcontrol, { struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); + struct tasdevice_fw *tas_fmw = tas_priv->fmw; + struct fct_param_address *p = &(tas_fmw->fct_par_addr); struct soc_bytes_ext *bytes_ext = (struct soc_bytes_ext *) kcontrol->private_value; unsigned char *dst = ucontrol->value.bytes.data; unsigned int reg = TASDEVICE_XM_A2_REG; + if (tas_priv->dspbin_typ) + reg = TASDEVICE_REG(p->a2_reg[0], p->a2_reg[1], p->a2_reg[2]); + guard(mutex)(&tas_priv->codec_lock); dst[0] = bytes_ext->max; return calib_data_get(tas_priv, reg, &dst[1]); @@ -1172,10 +1232,50 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) nr_controls < mix_index ? nr_controls : mix_index); } +static void cali_reg_update(struct bulk_reg_val *p, + struct fct_param_address *t) +{ + const int sum = ARRAY_SIZE(tas2781_cali_start_reg); + int reg = 0; + int j; + + for (j = 0; j < sum; j++) { + switch (tas2781_cali_start_reg[j].reg) { + case 0: + reg = TASDEVICE_REG(t->thr[0], t->thr[1], t->thr[2]); + break; + case TAS2781_PRM_PLT_FLAG_REG: + reg = TASDEVICE_REG(t->plt_flg[0], t->plt_flg[1], + t->plt_flg[2]); + break; + case TAS2781_PRM_SINEGAIN_REG: + reg = TASDEVICE_REG(t->sin_gn[0], t->sin_gn[1], t->sin_gn[2]); + break; + case TAS2781_PRM_SINEGAIN2_REG: + reg = TASDEVICE_REG(t->sin_gn[0], t->sin_gn[1], t->sin_gn[2]); + break; + default: + reg = 0; + break; + } + if (reg) + p[j].reg = reg; + } +} + +static void alpa_cali_update(struct bulk_reg_val *p, + struct fct_param_address *t) +{ + p->is_locked = false; + p->reg = TASDEVICE_REG(t->thr2[0], t->thr2[1], t->thr2[2]); + p->val_len = 4; +} + static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) { struct calidata *cali_data = &priv->cali_data; struct tasdevice *tasdev = priv->tasdevice; + struct tasdevice_fw *fmw = priv->fmw; struct soc_bytes_ext *ext_cali_data; struct snd_kcontrol_new *cali_ctrls; unsigned int nctrls; @@ -1191,14 +1291,25 @@ static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) } if (priv->chip_id == TAS2781) { + struct fct_param_address *t = &(fmw->fct_par_addr); + cali_ctrls = (struct snd_kcontrol_new *)tas2781_cali_controls; nctrls = ARRAY_SIZE(tas2781_cali_controls); for (i = 0; i < priv->ndev; i++) { - tasdev[i].cali_data_backup = + struct bulk_reg_val *p; + + p = tasdev[i].cali_data_backup = kmemdup(tas2781_cali_start_reg, sizeof(tas2781_cali_start_reg), GFP_KERNEL); if (!tasdev[i].cali_data_backup) return -ENOMEM; + if (priv->dspbin_typ) { + cali_reg_update(p, t); + if (priv->dspbin_typ == TASDEV_ALPHA) { + p = &tasdev[i].alp_cali_bckp; + alpa_cali_update(p, t); + } + } } } else { cali_ctrls = (struct snd_kcontrol_new *)tas2563_cali_controls; -- 2.34.1

3 4

[PATCH] soundwire: amd: change the log level for command response log
by Vijendar Mukunda 27 Feb '25

27 Feb '25

Change log level for command response log to dev_dbg_ratelimited when command is ignored. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index dcf85f94950a..a12c68b93b1c 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -346,7 +346,7 @@ static enum sdw_command_response amd_sdw_fill_msg_resp(struct amd_sdw_manager *a msg->dev_num); return SDW_CMD_FAIL; } - dev_err_ratelimited(amd_manager->dev, "command is ignored for Slave %d\n", + dev_dbg_ratelimited(amd_manager->dev, "command is ignored for Slave %d\n", msg->dev_num); return SDW_CMD_IGNORED; } -- 2.34.1

3 2

ALSA USB quirk mixer driver question
by Lena 27 Feb '25

27 Feb '25

Hi all! I'm currently trying to develop an usb quirk mixer driver for a desktop USB audio interface. The interface exposes the following: - interface 0-2 are USB UAC compliant, bound to by snd-usb-audio, and working flawlessly - interface 3 is vendor specific, and we can ignore it (used for firmware updates) - interface 4 exposes a non-standard mixer with a HID interface exposing 2 interrupt endpoints, and is used by the vendor windows app. I have studied the protocol with usbpcap and can successfully talk to the interface on the userspace, but i had a question regarding how to proceed in the kernel. This is because, after creating a new entry in `sound/usb/mixer_quirks.c` with all the needed controls, userspace can interact with them, etc. I can't figure out how to tell ALSA to bind to the correct interface for the mixer (and kick out usbhid). Looking at the other quirk mixer drivers it doesn't look (?) like any other quirk mixer driver is binding to a *different* interface, not even `sound/usb/mixer_scarlett.c`. Generally, in a standalone USB driver you'd have tracking/teardown handled easily with a callback, but i'm having trouble with the dependencies here. (simply using `usb_sndintpipe()` or `usb_interrupt_msg()` triggers kernel oopses, so that's probably not it) What i think i have to do is to use `usb_get_intf()`, then i have to somehow claim it (does `usb_driver_claim_interface()` work here?). But `usb_get_intf()` requires me to then release the interface with `usb_put_intf()`, and it doesn't look like there's anywhere for me to do that. (`snd_usb_mixer_elem_free()`? doesn't look like the order is correct) `iface_ref_find()` would sorta do what i need, but i cannot use it here. Could someone please point me in the right direction? I'm very stuck at the moment. Thanks, Lena

2 2

[PATCH v1] ASoC: tas2781: Clean up for some define
by Baojun Xu 26 Feb '25

26 Feb '25

Do some clean up for some define in header file. Like change lower case in value, up case in define, space add in recommends, change date of files and add author. Signed-off-by: Baojun Xu <baojun.xu(a)ti.com> --- include/sound/tas2781.h | 21 +++++++++++---------- sound/soc/codecs/tas2781-fmwlib.c | 7 ++++--- 2 files changed, 15 insertions(+), 13 deletions(-) diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 72d2060904f6..a2c8eb111e5f 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -11,6 +11,7 @@ // // Author: Shenghao Ding <shenghao-ding(a)ti.com> // Author: Kevin Lu <kevin-lu(a)ti.com> +// Author: Baojun Xu <baojun.xu(a)ti.com> // #ifndef __TAS2781_H__ @@ -31,7 +32,7 @@ SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) -/*PAGE Control Register (available in page0 of each book) */ +/* PAGE Control Register (available in page0 of each book) */ #define TASDEVICE_PAGE_SELECT 0x00 #define TASDEVICE_BOOKCTL_PAGE 0x00 #define TASDEVICE_BOOKCTL_REG 127 @@ -42,12 +43,12 @@ #define TASDEVICE_REG(book, page, reg) (((book * 256 * 128) + \ (page * 128)) + reg) -/*Software Reset */ -#define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x01) +/* Software Reset */ +#define TASDEVICE_REG_SWRESET TASDEVICE_REG(0x0, 0x0, 0x01) #define TASDEVICE_REG_SWRESET_RESET BIT(0) -/*I2C Checksum */ -#define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) +/* I2C Checksum */ +#define TASDEVICE_CHECKSUM_REG TASDEVICE_REG(0x0, 0x0, 0x7e) /* XM_340 */ #define TASDEVICE_XM_A1_REG TASDEVICE_REG(0x64, 0x63, 0x3c) @@ -55,8 +56,8 @@ #define TASDEVICE_XM_A2_REG TASDEVICE_REG(0x64, 0x63, 0x38) /* Volume control */ -#define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0C) -#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) +#define TAS2563_DVC_LVL TASDEVICE_REG(0x00, 0x02, 0x0c) +#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1a) #define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) #define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) @@ -95,8 +96,8 @@ #define TAS2781_PRM_SINEGAIN_REG TASDEVICE_REG(0x00, 0x14, 0x40) #define TAS2781_PRM_SINEGAIN2_REG TASDEVICE_REG(0x00, 0x14, 0x44) -#define TAS2781_TEST_UNLOCK_REG TASDEVICE_REG(0x00, 0xFD, 0x0D) -#define TAS2781_TEST_PAGE_UNLOCK 0x0D +#define TAS2781_TEST_UNLOCK_REG TASDEVICE_REG(0x00, 0xfd, 0x0d) +#define TAS2781_TEST_PAGE_UNLOCK 0x0d #define TAS2781_RUNTIME_LATCH_RE_REG TASDEVICE_REG(0x00, 0x00, 0x49) #define TAS2781_RUNTIME_RE_REG_TF TASDEVICE_REG(0x64, 0x62, 0x48) diff --git a/sound/soc/codecs/tas2781-fmwlib.c b/sound/soc/codecs/tas2781-fmwlib.c index 61d9c220b6a4..bc2f68bd9846 100644 --- a/sound/soc/codecs/tas2781-fmwlib.c +++ b/sound/soc/codecs/tas2781-fmwlib.c @@ -2,9 +2,10 @@ // // tas2781-fmwlib.c -- TASDEVICE firmware support // -// Copyright 2023 - 2024 Texas Instruments, Inc. +// Copyright 2023 - 2025 Texas Instruments, Inc. // // Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Baojun Xu <baojun.xu(a)ti.com> #include <linux/crc8.h> #include <linux/firmware.h> @@ -1496,7 +1497,7 @@ static int tasdev_block_chksum(struct tasdevice_priv *tas_priv, unsigned int nr_value; int ret; - ret = tasdevice_dev_read(tas_priv, chn, TASDEVICE_I2CChecksum, + ret = tasdevice_dev_read(tas_priv, chn, TASDEVICE_CHECKSUM_REG, &nr_value); if (ret < 0) { dev_err(tas_priv->dev, "%s: Chn %d\n", __func__, chn); @@ -1540,7 +1541,7 @@ static int tasdev_load_blk(struct tasdevice_priv *tas_priv, while (block->nr_retry > 0) { if (block->is_pchksum_present) { ret = tasdevice_dev_write(tas_priv, chn, - TASDEVICE_I2CChecksum, 0); + TASDEVICE_CHECKSUM_REG, 0); if (ret < 0) break; } -- 2.34.1

1 0

ucm2: Add support for MSI PRO B650-A w/ Realtek ALC4080
by GitHub pull_request - opened 26 Feb '25

26 Feb '25

alsa-project/alsa-ucm-conf pull request #514 was opened from gbrener: Gets the front panel audio to work. [alsa-info.txt](https://github.com/user-attachments/files/18993322/alsa-info… Thanks in advance. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/514 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/514.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: MediaTek: mt8390-evk: Add support for SOF
by GitHub pull_request - opened 26 Feb '25

26 Feb '25

alsa-project/alsa-ucm-conf pull request #513 was opened from nfraprado: Add support for MT8390 EVK with the SOF enabled. The only differences compared to the UCM with SOF disabled is the device number for playback and capture, which uses the PCMs for the SOF instead, and the number of channels for capture, which is stereo rather than mono. Define those parameters as variables for each case, with and without SOF, and move the common intialization to an init.conf, so the UCM can be shared by both. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/513 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/513.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Potential incorrect argument to snd_seq_port_info_set_ump_group in seqmid.c
by GitHub issues - opened 26 Feb '25

26 Feb '25

alsa-project/alsa-lib issue #440 was opened from reuk: https://github.com/alsa-project/alsa-lib/blob/df8f1cc1ec9d9ee15be5e2c23ad25… I think this call in seqmid.c is incorrect. According to the docs, the second argument to `snd_seq_port_info_set_ump_group` should be a 1-based group number (probably 0 for the UMP endpoint). I'm testing on Fedora 41 with alsa-lib-devel 1.2.13-3.fc41, and I'm seeing the following printed to the console when I call `snd_seq_create_ump_endpoint`: ``` ALSA lib seqmid.c:592:(snd_seq_create_ump_endpoint) Failed to create MIDI 2.0 port ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/440 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[GIT PULL] ASoC fixes for v6.14-rc4
by Mark Brown 26 Feb '25

26 Feb '25

The following changes since commit 571b69f2f9b1ec7cf7d0e9b79e52115a87a869c4: ASoC: imx-audmix: remove cpu_mclk which is from cpu dai device (2025-02-13 11:34:32 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.14-rc4 for you to fetch changes up to 9da0ed4a85027063441fa1c73967cafc38f0677c: ASoC: Intel: don't check number of sdw links when set (2025-02-25 17:47:04 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.14 More driver specific fixes, the firmware change is part of fixing the race conditions in the Cirrus driver. ---------------------------------------------------------------- Bard Liao (2): ASoC: SOF: Intel: don't check number of sdw links when set dmic_fixup ASoC: Intel: sof_sdw: warn both sdw and pch dmic are used Chancel Liu (1): ASoC: fsl: Rename stream name of SAI DAI driver Hector Martin (3): ASoC: tas2770: Fix volume scale ASoC: tas2764: Fix power control mask ASoC: tas2764: Set the SDOUT polarity correctly Mark Brown (1): ASoC: Intel: don't check number of sdw links when set Nicolas Frattaroli (2): ASoC: es8328: fix route from DAC to output ASoC: dapm-graph: set fill colour of turned on nodes Richard Fitzgerald (2): firmware: cs_dsp: Remove async regmap writes ASoC: cs35l56: Prevent races when soft-resetting using SPI control drivers/firmware/cirrus/cs_dsp.c | 24 +++--------- include/sound/cs35l56.h | 31 +++++++++++++++ sound/pci/hda/cs35l56_hda_spi.c | 3 ++ sound/soc/codecs/cs35l56-shared.c | 80 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs35l56-spi.c | 3 ++ sound/soc/codecs/es8328.c | 15 ++------ sound/soc/codecs/tas2764.c | 10 ++++- sound/soc/codecs/tas2764.h | 8 +++- sound/soc/codecs/tas2770.c | 2 +- sound/soc/fsl/fsl_sai.c | 6 +-- sound/soc/fsl/imx-audmix.c | 4 +- sound/soc/intel/boards/sof_sdw.c | 7 ++++ sound/soc/sof/intel/hda.c | 18 +-------- tools/sound/dapm-graph | 2 +- 14 files changed, 159 insertions(+), 54 deletions(-)

2 1

[GIT PULL] ASoC fixes for v6.14-rc3
by Mark Brown 26 Feb '25

26 Feb '25

The following changes since commit 571b69f2f9b1ec7cf7d0e9b79e52115a87a869c4: ASoC: imx-audmix: remove cpu_mclk which is from cpu dai device (2025-02-13 11:34:32 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.14-rc3 for you to fetch changes up to f5468beeab1b1adfc63c2717b1f29ef3f49a5fab: ASoC: tas2764: Set the SDOUT polarity correctly (2025-02-18 20:08:43 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.14 A few fixes I and James Calligero picked out of the Asahi tree. ---------------------------------------------------------------- Hector Martin (3): ASoC: tas2770: Fix volume scale ASoC: tas2764: Fix power control mask ASoC: tas2764: Set the SDOUT polarity correctly sound/soc/codecs/tas2764.c | 10 +++++++++- sound/soc/codecs/tas2764.h | 8 +++++++- sound/soc/codecs/tas2770.c | 2 +- 3 files changed, 17 insertions(+), 3 deletions(-)

1 0

[PATCH 0/2] ASoC: codecs: add ES8389 codec driver
by Zhang Yi 26 Feb '25

26 Feb '25

The driver is for codec ES8389 of everest-semi. Zhang Yi (2): ASoC: codecs: add support for ES8389 ASoC: dt-bindings: Add Everest ES8389 audio CODEC .../bindings/sound/everest,es8389.txt | 68 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/es8389.c | 977 ++++++++++++++++++ sound/soc/codecs/es8389.h | 139 +++ 5 files changed, 1191 insertions(+) create mode 100755 Documentation/devicetree/bindings/sound/everest,es8389.txt create mode 100755 sound/soc/codecs/es8389.c create mode 100755 sound/soc/codecs/es8389.h -- 2.17.1

3 4

add rt713 & rt1318 LED control
by GitHub pull_request - opened 26 Feb '25

26 Feb '25

alsa-project/alsa-ucm-conf pull request #512 was opened from shumingfan: add rt713 & rt1318 LED control Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/512 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/512.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Request for Build Instructions for macOS and Ubuntu
by GitHub issues - opened 24 Feb '25

24 Feb '25

alsa-project/alsa-utils issue #292 was opened from sallavatsal: Hello, I'm having difficulty building alsa-lib on both macOS and Ubuntu. Specifically: macOS: After running autoreconf, I get warnings and issues generating the configure script. Ubuntu: The available libasound2-dev version seems to be older than 1.2.5, causing configuration errors. Could you please provide official build instructions or guidance for these platforms? Also, I want to become a member of mailing list if there's any. Thank you! Issue URL : https://github.com/alsa-project/alsa-utils/issues/292 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

alsa-info.8: Some remarks and a patch with editorial changes for this man page
by Bjarni Ingi Gislason 21 Feb '25

21 Feb '25

Package: alsa-utils Version: 1.2.13-1 Severity: minor Tags: patch * What led up to the situation? Checking for defects with a new version test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -rCHECKSTYLE=10 -ww -z < "man page" [Use "groff -e ' $' -e '\\~$' <file>" to find obvious trailing spaces.] ["test-groff" is a script in the repository for "groff"; is not shipped] (local copy and "troff" slightly changed by me). [The fate of "test-nroff" was decided in groff bug #55941.] * What was the outcome of this action? an.tmac:<stdin>:1: style: .TH missing fourth argument; consider package/project name and version (e.g., "groff 1.23.0") troff:<stdin>:59: warning: trailing space in the line * What outcome did you expect instead? No output (no warnings). -.- General remarks and further material, if a diff-file exist, are in the attachments. -- System Information: Debian Release: trixie/sid APT prefers testing APT policy: (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 6.12.12-amd64 (SMP w/2 CPU threads; PREEMPT) Locale: LANG=is_IS.iso88591, LC_CTYPE=is_IS.iso88591 (charmap=ISO-8859-1), LANGUAGE not set Shell: /bin/sh linked to /usr/bin/dash Init: sysvinit (via /sbin/init) Versions of packages alsa-utils depends on: ii kmod 33+20240816-2 ii libasound2t64 1.2.13-1+b1 ii libatopology2t64 1.2.13-1+b1 ii libc6 2.40-6 ii libfftw3-single3 3.3.10-2+b1 ii libncursesw6 6.5+20250125-2 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5+20250125-2 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

1 0

FiiO K7 USB DAC no Inputs
by GitHub issues - opened 21 Feb '25

21 Feb '25

alsa-project/alsa-lib issue #439 was opened from Starfoxfs: I have problems with Fiio K7 USB DAC in Linux Mint Cinnamon 22.1 (6.8.0-56-generic) Alsa Version: 1.0.25+dfsg-0ubuntu7 Pipewire: 1.0.5-1ubuntu2 Wireplumber: 0.4.17-1ubuntu4 The USB Dac has this Ports: 2x Line IN (Cinch) 2x Line OUT or Pre OUT(Cinch) 1x Digital IN TOSLINK 1x COAX/RCA IN (Yellow Cinch) The Problem is: Alsa didnt find any of this IN Ports. There are 4 OUTPUTS: Analog Stereo works fine with Alsa Digital IEC958 (thats wrong Device has no Digital Output) Digital Surround 51 (thats wrong Device has no Digital Output) Pro Audio works fine with Alsa Would be nice to have the Inputs. Heres my Alsa-Info Output: [alsa-info.txt](https://github.com/user-attachments/files/18907991/alsa-info… Issue URL : https://github.com/alsa-project/alsa-lib/issues/439 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[GIT PULL] ASoC fixes for v6.14-rc3
by Mark Brown 21 Feb '25

21 Feb '25

The following changes since commit 571b69f2f9b1ec7cf7d0e9b79e52115a87a869c4: ASoC: imx-audmix: remove cpu_mclk which is from cpu dai device (2025-02-13 11:34:32 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.14-rc3 for you to fetch changes up to f5468beeab1b1adfc63c2717b1f29ef3f49a5fab: ASoC: tas2764: Set the SDOUT polarity correctly (2025-02-18 20:08:43 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.14 A few fixes I and James Calligero picked out of the Asahi tree. ---------------------------------------------------------------- Hector Martin (3): ASoC: tas2770: Fix volume scale ASoC: tas2764: Fix power control mask ASoC: tas2764: Set the SDOUT polarity correctly sound/soc/codecs/tas2764.c | 10 +++++++++- sound/soc/codecs/tas2764.h | 8 +++++++- sound/soc/codecs/tas2770.c | 2 +- 3 files changed, 17 insertions(+), 3 deletions(-)

2 1

[PATCH] ALSA: usb-audio: Re-add sample rate quirk for Pioneer DJM-900NXS2
by Dmitry Panchenko | d-Systems 20 Feb '25

20 Feb '25

From: Dmitry Panchenko <dmitry(a)d-systems.ee> Re-add the sample-rate quirk for the Pioneer DJM-900NXS2. This device does not not work without setting sample-rate. Signed-off-by: Dmitry Panchenko <dmitry(a)d-systems.ee> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index a97efb7b131e..09210fb4ac60 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1868,6 +1868,7 @@ void snd_usb_set_format_quirk(struct snd_usb_substream *subs, case USB_ID(0x534d, 0x2109): /* MacroSilicon MS2109 */ subs->stream_offset_adj = 2; break; + case USB_ID(0x2b73, 0x000a): /* Pioneer DJM-900NXS2 */ case USB_ID(0x2b73, 0x0013): /* Pioneer DJM-450 */ pioneer_djm_set_format_quirk(subs, 0x0082); break; -- 2.43.0

2 1

[PATCH v2 1/4] ASoC: tas2781: Support dsp firmware Alpha and Beta seaies
by Shenghao Ding 20 Feb '25

20 Feb '25

For calibration, basic version does not contain any calibration addresses, it depends on calibration tool to convery the addresses to the driver. Since Alpha and Beta firmware, all the calibration addresses are saved into the firmware. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v2: - No further update v1: - update the year from 2024 to 2025. - Add dspbin_type to distinguish different dspbin types, such as TASDEV_BASIC, TASDEV_ALPHA, and TASDEV_BETA. - Add struct bulk_reg_val alp_cali_bckp to store the value of pilot tone threshold only for TASDEV_ALPHA & TASDEV_BETA firmware. - Add the declaration of (*fw_parse_fct_param_address)() to parse the addresses of calibration params. - Add the minimum version of TASDEV_ALPHA and TASDEV_BETA firmware. - Add the definition of struct fct_param_address. --- include/sound/tas2781-dsp.h | 26 +++++++++++++++++++++++--- include/sound/tas2781.h | 13 ++++++++++++- 2 files changed, 35 insertions(+), 4 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 3cda9da14f6d..77a240daaae0 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -30,8 +30,10 @@ #define PRE_DEVICE_C 0x12 #define PRE_DEVICE_D 0x16 -#define PPC3_VERSION 0x4100 -#define PPC3_VERSION_TAS2781 0x14600 +#define PPC3_VERSION_BASE 0x4100 +#define PPC3_VERSION_TAS2781_BASIC_MIN 0x14600 +#define PPC3_VERSION_TAS2781_ALPHA_MIN 0x4a00 +#define PPC3_VERSION_TAS2781_BETA_MIN 0x19400 #define TASDEVICE_DEVICE_SUM 8 #define TASDEVICE_CONFIG_SUM 64 @@ -106,6 +108,23 @@ struct tasdevice_calibration { struct tasdevice_data dev_data; }; +struct fct_param_address { + unsigned char thr[3]; + unsigned char thr2[3]; + unsigned char plt_flg[3]; + unsigned char sin_gn[3]; + unsigned char sin_gn2[3]; + /* real-time spk impedance */ + unsigned char r0_reg[3]; + /* check spk connection */ + unsigned char tf_reg[3]; + /* check spk resonant frequency */ + unsigned char a1_reg[3]; + unsigned char a2_reg[3]; + unsigned short size; + char *data; +}; + struct tasdevice_fw { struct tasdevice_dspfw_hdr fw_hdr; unsigned short nr_programs; @@ -114,6 +133,7 @@ struct tasdevice_fw { struct tasdevice_config *configs; unsigned short nr_calibrations; struct tasdevice_calibration *calibrations; + struct fct_param_address fct_par_addr; struct device *dev; }; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 72d2060904f6..5cfbb9e0a626 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -112,6 +112,12 @@ enum audio_device { TAS2781, }; +enum dspbin_type { + TASDEV_BASIC, + TASDEV_ALPHA, + TASDEV_BETA, +}; + enum device_catlog_id { LENOVO = 0, OTHERS @@ -126,6 +132,7 @@ struct bulk_reg_val { struct tasdevice { struct bulk_reg_val *cali_data_backup; + struct bulk_reg_val alp_cali_bckp; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -171,6 +178,7 @@ struct tasdevice_priv { unsigned char dev_name[32]; const char *name_prefix; unsigned char ndev; + unsigned int dspbin_typ; unsigned int magic_num; unsigned int chip_id; unsigned int sysclk; @@ -196,6 +204,9 @@ struct tasdevice_priv { int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset); + int (*fw_parse_fct_param_address)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, struct tasdev_blk *block); -- 2.34.1

2 5

Problems with async handler
by GitHub issues - edited 20 Feb '25

20 Feb '25

alsa-project/alsa-lib issue #438 was edited from m-paq: Hi, I'm currently working on a project and I'm trying to use the async handler through `snd_async_add_pcm_handler`. I don't really understand where the issue comes from and how to debug it but I keep getting an error on this function call in my project. To try to figure out what's wrong I went in the test files and tried to run the test file `pcm.c` and added a print of the error code (`snd_strerror`) Compiling with: `gcc pcm.c -o test_pcm -lasound -lm` Running it with `./test_pcm -m async` But it keeps giving me the following output: ``` Playback device is default Stream parameters are 44100Hz, S16_LE, 1 channels Sine wave rate is 440.0000Hz Using transfer method: async Unable to register async handler Function not implemented ``` from line 434. I am able to run it in write mode with the correct behaviour, so my question is: **Is this an issue with the device or is it an issue somewhere else?** If ever this is not the correct place to ask this question please refer me to a more adequate place. Thanks to anyone that can help me! Issue URL : https://github.com/alsa-project/alsa-lib/issues/438 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Problems with async handler
by GitHub issues - opened 20 Feb '25

20 Feb '25

alsa-project/alsa-lib issue #438 was opened from m-paq: Hi, I'm currently working on a project and I'm trying to use the async handler through `snd_async_add_pcm_handler`. I don't really understand where the issue comes from and how to debug it but I keep getting an error on this function call in my project. To try to figure out what's wrong I went in the test files and tried to run the test file `pcm.c` and added a print of the error code (`snd_strerror`) Compiling with: `gcc pcm.c -o test_pcm -lasound -lm` Running it with `./test_pcm -m async` But it keeps giving me the following output: ``` Playback device is default Stream parameters are 44100Hz, S16_LE, 1 channels Sine wave rate is 440.0000Hz Using transfer method: async Unable to register async handler Function not implemented ``` from line 441. I am able to run it in write mode with the correct behaviour, so my question is: **Is this an issue with the device or is it an issue somewhere else?** If ever this is not the correct place to ask this question please refer me to a more adequate place. Thanks to anyone that can help me! Issue URL : https://github.com/alsa-project/alsa-lib/issues/438 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ALSA: usb-audio: Add Pioneer DJ DJM-900NXS2 support
by Dmitry Panchenko | d-Systems 19 Feb '25

19 Feb '25

Hello, Pioneer DJ DJM-900NXS2 is a widely used DJ mixer with 2 audio USB interfaces. Both have a MIDI controller, 10 playback and 12 capture channels. I've submitted Pioneer DJM-900NXS2 support patches back in 2020, but someone has removed set_format_quirk function for DJM-900NXS2 model. The playback and capture do not work, quirks.c needs to be patched. The patch is attached, please add. Signed-off-by: Dmitry Panchenko <dmitry(a)d-systems.ee> -- Kind regards, Dmitry Panchenko

1 0

Realtek driver assignes wrong pins on Omen 16 with B&O audio
by GitHub issues - edited 19 Feb '25

19 Feb '25

alsa-project/alsa-ucm-conf issue #511 was edited from johnfloe:  # Description: I have a dual boot with Arch Linux and Windows 11 on my HP Omen 16 b1001nd with Bang & Olufsen audio. On Windows, the speakers function normally and the volume is quite loud and full at 100%. But on Arch, the audio is very shallow and the volume is low compared to Windows. I've tried using HDAJackRetask, but the Realtek driver overrides the pin settings. When I blacklist the Realtek module, it switches to Generic and I can enable the pins with hda verbs. But the then auto switching for the headphones doesn't work, so I have to figure out how to enable that.. It does seem a bit better when enabling pin 0x14 and pin 0x1e, but still not the same as in Windows. It would be nice to have a working driver/codec so I don't have to configure all the pins myself. Thanks in advance! # Additional info: * package version(s): latest kernel, linux-firmware and sof-firmware installed * config and/or log files: Related commands output: sudo dmesg | grep snd [ 5.052362] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver [ 5.052469] snd_hda_intel 0000:01:00.1: enabling device (0000 -> 0002) [ 5.052725] snd_hda_intel 0000:01:00.1: Disabling MSI [ 5.052736] snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client [ 5.388591] snd_hda_codec_realtek ehdaudio0D0: ALC245: picked fixup for PCI SSID 103c:0000 [ 5.388944] snd_hda_codec_realtek ehdaudio0D0: autoconfig for ALC245: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 5.388947] snd_hda_codec_realtek ehdaudio0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 5.388949] snd_hda_codec_realtek ehdaudio0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 5.388951] snd_hda_codec_realtek ehdaudio0D0: mono: mono_out=0x0 [ 5.388953] snd_hda_codec_realtek ehdaudio0D0: inputs: [ 5.388954] snd_hda_codec_realtek ehdaudio0D0: Mic=0x19 inxi -A Audio: Device-1: Intel Alder Lake PCH-P High Definition Audio driver: sof-audio-pci-intel-tgl Device-2: NVIDIA GA104 High Definition Audio driver: snd_hda_intel API: ALSA v: k6.13.2-arch1-1 status: kernel-api Server-1: PipeWire v: 1.2.7 status: active Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/511 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Realtek driver assignes wrong pins on Omen 16 with B&O audio
by GitHub issues - edited 19 Feb '25

19 Feb '25

alsa-project/alsa-ucm-conf issue #511 was edited from johnfloe:  # Description: I have a dual boot with Arch Linux and Windows 11 on my HP Omen 16 b1001nd with Bang & Olufsen audio. On Windows, the speakers function normally and the volume is quite loud and full at 100%. But on Arch, the audio is very shallow and the volume is low compared to Windows. I've tried using HDAJackRetask, but the Realtek driver overrides the pin settings. When I blacklist the Realtek module, it switches to Generic and I can enable the pins with hda verbs. But the then auto switching for the headphones doesn't work, so I have to figure out how to enable that.. It does seem a bit better when enabling pin 0x14 and pin 0x1e, but still not the same as in Windows. It would be nice to have a working driver/codec so I don't have to configure all the pins myself. Thanks in advance! # Additional info: * package version(s): latest kernel, linux-firmware and sof-firmware installed * config and/or log files: Related dmesg output: sudo dmesg | grep snd [ 5.052362] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver [ 5.052469] snd_hda_intel 0000:01:00.1: enabling device (0000 -> 0002) [ 5.052725] snd_hda_intel 0000:01:00.1: Disabling MSI [ 5.052736] snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client [ 5.388591] snd_hda_codec_realtek ehdaudio0D0: ALC245: picked fixup for PCI SSID 103c:0000 [ 5.388944] snd_hda_codec_realtek ehdaudio0D0: autoconfig for ALC245: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 5.388947] snd_hda_codec_realtek ehdaudio0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 5.388949] snd_hda_codec_realtek ehdaudio0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 5.388951] snd_hda_codec_realtek ehdaudio0D0: mono: mono_out=0x0 [ 5.388953] snd_hda_codec_realtek ehdaudio0D0: inputs: [ 5.388954] snd_hda_codec_realtek ehdaudio0D0: Mic=0x19 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/511 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Realtek driver assignes wrong pins on Omen 16 with B&O audio
by GitHub issues - edited 19 Feb '25

19 Feb '25

alsa-project/alsa-ucm-conf issue #511 was edited from johnfloe:  # Description: I have a dual boot with Arch Linux and Windows 11 on my HP Omen 16 b1001nd with Bang & Olufsen audio. On Windows, the speakers function normally and the volume is quite loud and full at 100%. But on Arch, the audio is very shallow and the volume is low compared to Windows. I've tried using HDAJackRetask, but the Realtek driver overrides the pin settings. When I blacklist the Realtek module, it switches to Generic and I can enable the pins with hda verbs. But the then auto switching for the headphones doesn't work, so I have to figure out how to enable that.. It does seem a bit better when enabling pin 0x14 and pin 0x1e, but still not the same as in Windows. It would be nice to have a working driver/codec so I don't have to configure all the pins myself. Thanks in advance! # Additional info: * package version(s): latest kernel, linux-firmware and sof-firmware installed * config and/or log files: Related dmesg output: `sudo dmesg | grep snd [ 5.052362] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver [ 5.052469] snd_hda_intel 0000:01:00.1: enabling device (0000 -> 0002) [ 5.052725] snd_hda_intel 0000:01:00.1: Disabling MSI [ 5.052736] snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client [ 5.388591] snd_hda_codec_realtek ehdaudio0D0: ALC245: picked fixup for PCI SSID 103c:0000 [ 5.388944] snd_hda_codec_realtek ehdaudio0D0: autoconfig for ALC245: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 5.388947] snd_hda_codec_realtek ehdaudio0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 5.388949] snd_hda_codec_realtek ehdaudio0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 5.388951] snd_hda_codec_realtek ehdaudio0D0: mono: mono_out=0x0 [ 5.388953] snd_hda_codec_realtek ehdaudio0D0: inputs: [ 5.388954] snd_hda_codec_realtek ehdaudio0D0: Mic=0x19` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/511 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Realtek driver assignes wrong pins on Omen 16 with B&O audio
by GitHub issues - edited 19 Feb '25

19 Feb '25

alsa-project/alsa-ucm-conf issue #511 was edited from johnfloe:  # Description: I have a dual boot with Arch Linux and Windows 11 on my HP Omen 16 b1001nd with Bang & Olufsen audio. On Windows, the speakers function normally and the volume is quite loud and full at 100%. But on Arch, the audio is very shallow and the volume is low compared to Windows. I've tried using HDAJackRetask, but the Realtek driver overrides the pin settings. When I blacklist the Realtek module, it switches to Generic and I can enable the pins with hda verbs. But the then auto switching for the headphones doesn't work, so I have to figure out how to enable that.. It does seem a bit better when enabling pin 0x14 and pin 0x1e, but still not the same as in Windows. It would be nice to have a working driver/codec so I don't have to configure all the pins myself. Thanks in advance! # Additional info: * package version(s): latest kernel, linux-firmware and sof-firmware installed * config and/or log files: Related dmesg output: `sudo dmesg | grep snd\n [ 5.052362] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver [ 5.052469] snd_hda_intel 0000:01:00.1: enabling device (0000 -> 0002) [ 5.052725] snd_hda_intel 0000:01:00.1: Disabling MSI [ 5.052736] snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client [ 5.388591] snd_hda_codec_realtek ehdaudio0D0: ALC245: picked fixup for PCI SSID 103c:0000 [ 5.388944] snd_hda_codec_realtek ehdaudio0D0: autoconfig for ALC245: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 5.388947] snd_hda_codec_realtek ehdaudio0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 5.388949] snd_hda_codec_realtek ehdaudio0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 5.388951] snd_hda_codec_realtek ehdaudio0D0: mono: mono_out=0x0 [ 5.388953] snd_hda_codec_realtek ehdaudio0D0: inputs: [ 5.388954] snd_hda_codec_realtek ehdaudio0D0: Mic=0x19` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/511 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Realtek driver assignes wrong pins on Omen 16 with B&O audio
by GitHub issues - edited 19 Feb '25

19 Feb '25

alsa-project/alsa-ucm-conf issue #511 was edited from johnfloe:  # Description: I have a dual boot with Arch Linux and Windows 11 on my HP Omen 16 b1001nd with Bang & Olufsen audio. On Windows, the speakers function normally and the volume is quite loud and full at 100%. But on Arch, the audio is very shallow and the volume is low compared to Windows. I've tried using HDAJackRetask, but the Realtek driver overrides the pin settings. When I blacklist the Realtek module, it switches to Generic and I can enable the pins with hda verbs. But the then auto switching for the headphones doesn't work, so I have to figure out how to enable that.. It does seem a bit better when enabling pin 0x14 and pin 0x1e, but still not the same as in Windows. It would be nice to have a working driver/codec so I don't have to configure all the pins myself. Thanks in advance! # Additional info: * package version(s): latest kernel, linux-firmware and sof-firmware installed * config and/or log files: Related dmesg output: ` sudo dmesg | grep snd [ 5.052362] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver [ 5.052469] snd_hda_intel 0000:01:00.1: enabling device (0000 -> 0002) [ 5.052725] snd_hda_intel 0000:01:00.1: Disabling MSI [ 5.052736] snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client [ 5.388591] snd_hda_codec_realtek ehdaudio0D0: ALC245: picked fixup for PCI SSID 103c:0000 [ 5.388944] snd_hda_codec_realtek ehdaudio0D0: autoconfig for ALC245: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 5.388947] snd_hda_codec_realtek ehdaudio0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 5.388949] snd_hda_codec_realtek ehdaudio0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 5.388951] snd_hda_codec_realtek ehdaudio0D0: mono: mono_out=0x0 [ 5.388953] snd_hda_codec_realtek ehdaudio0D0: inputs: [ 5.388954] snd_hda_codec_realtek ehdaudio0D0: Mic=0x19` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/511 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Realtek driver assignes wrong pins on Omen 16 with B&O audio
by GitHub issues - opened 19 Feb '25

19 Feb '25

alsa-project/alsa-ucm-conf issue #511 was opened from johnfloe:  # Description: I have a dual boot with Arch Linux and Windows 11 on my HP Omen 16 b1001nd with Bang & Olufsen audio. On Windows, the speakers function normally and the volume is quite loud and full at 100%. But on Arch, the audio is very shallow and the volume is low compared to Windows. I've tried using HDAJackRetask, but the Realtek driver overrides the pin settings. When I blacklist the Realtek module, it switches to Generic and I can enable the pins with hda verbs. But the then auto switching for the headphones doesn't work, so I have to figure out how to enable that.. It does seem a bit better when enabling pin 0x14 and pin 0x1e, but still not the same as in Windows. It would be nice to have a working driver/codec so I don't have to configure all the pins myself. Thanks in advance! # Additional info: * package version(s): latest kernel, linux-firmware and sof-firmware installed * config and/or log files: Related dmesg output: sudo dmesg | grep snd [ 5.052362] snd_hda_intel 0000:00:1f.3: Digital mics found on Skylake+ platform, using SOF driver [ 5.052469] snd_hda_intel 0000:01:00.1: enabling device (0000 -> 0002) [ 5.052725] snd_hda_intel 0000:01:00.1: Disabling MSI [ 5.052736] snd_hda_intel 0000:01:00.1: Handle vga_switcheroo audio client [ 5.388591] snd_hda_codec_realtek ehdaudio0D0: ALC245: picked fixup for PCI SSID 103c:0000 [ 5.388944] snd_hda_codec_realtek ehdaudio0D0: autoconfig for ALC245: line_outs=1 (0x17/0x0/0x0/0x0/0x0) type:speaker [ 5.388947] snd_hda_codec_realtek ehdaudio0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 5.388949] snd_hda_codec_realtek ehdaudio0D0: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) [ 5.388951] snd_hda_codec_realtek ehdaudio0D0: mono: mono_out=0x0 [ 5.388953] snd_hda_codec_realtek ehdaudio0D0: inputs: [ 5.388954] snd_hda_codec_realtek ehdaudio0D0: Mic=0x19 Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/511 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH RESEND 0/7] Initial Support for CS40L26
by Fred Treven 19 Feb '25

19 Feb '25

Introduce driver for Cirrus Logic Device CS40L26: A boosted haptics driver with integrated DSP and waveform memory with advanced closed loop algorithms and LRA protection. The core CS40L26 driver is in MFD and touches the Input Force Feedback subsystem for haptics and the ASoC subsystem for audio to haptics streaming. This patchset includes changes to the CS DSP firmware driver which fixes two bugs and introduces support for multiple coefficient files. Fred Treven (7): firmware: cs_dsp: Fix error checking in wseq_write() firmware: cs_dsp: Check for valid num_regs in cs_dsp_wseq_multi_write() firmware: cs_dsp: Add ability to load multiple coefficient files dt-bindings: mfd: cirrus,cs40l26: Support for CS40L26 mfd: cs40l26: Add support for CS40L26 core driver ASoC: cs40l26: Support I2S streaming to CS40L26 Input: cs40l26 - Add support for CS40L26 haptic driver .../bindings/mfd/cirrus,cs40l26.yaml | 81 + MAINTAINERS | 4 +- drivers/firmware/cirrus/cs_dsp.c | 70 +- drivers/input/misc/Kconfig | 10 + drivers/input/misc/Makefile | 1 + drivers/input/misc/cs40l26-vibra.c | 669 ++++++++ drivers/mfd/Kconfig | 29 + drivers/mfd/Makefile | 4 + drivers/mfd/cs40l26-core.c | 1412 +++++++++++++++++ drivers/mfd/cs40l26-i2c.c | 63 + drivers/mfd/cs40l26-spi.c | 63 + include/linux/firmware/cirrus/cs_dsp.h | 14 + include/linux/mfd/cs40l26.h | 341 ++++ sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs40l26-codec.c | 523 ++++++ 16 files changed, 3281 insertions(+), 17 deletions(-) create mode 100644 Documentation/devicetree/bindings/mfd/cirrus,cs40l26.yaml create mode 100644 drivers/input/misc/cs40l26-vibra.c create mode 100644 drivers/mfd/cs40l26-core.c create mode 100644 drivers/mfd/cs40l26-i2c.c create mode 100644 drivers/mfd/cs40l26-spi.c create mode 100644 include/linux/mfd/cs40l26.h create mode 100644 sound/soc/codecs/cs40l26-codec.c -- 2.34.1

6 17

[PATCH] ASoC: amd: ps: fix inconsistent indenting warning in check_and_handle_sdw_dma_irq()
by Vijendar Mukunda 18 Feb '25

18 Feb '25

Fix below inconsistent indenting smatch warning. smatch warnings: sound/soc/amd/ps/pci-ps.c:68 check_and_handle_sdw_dma_irq() warn: inconsistent indenting Fixes: 4516be370ced ("ASoC: amd: ps: refactor soundwire dma interrupt handling") Reported-by: kernel test robot <lkp(a)intel.com> Closes: https://lore.kernel.org/oe-kbuild-all/202502132134.BlkNw1Iq-lkp@intel.com/ Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/ps/pci-ps.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 2ff8e67c19bd..221c65ff03c9 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -129,7 +129,7 @@ static short int check_and_handle_sdw_dma_irq(struct acp63_dev_data *adata, u32 if (ext_intr_stat1 & ACP63_P1_AUDIO1_RX_THRESHOLD) { writel(ACP63_P1_AUDIO1_RX_THRESHOLD, adata->acp63_base + ACP_EXTERNAL_INTR_STAT1); - adata->acp63_sdw1_dma_intr_stat[ACP63_SDW1_AUDIO1_RX] = 1; + adata->acp63_sdw1_dma_intr_stat[ACP63_SDW1_AUDIO1_RX] = 1; sdw_dma_irq_flag = 1; } if (ext_intr_stat1 & ACP63_P1_AUDIO1_TX_THRESHOLD) { -- 2.34.1

2 1

ALC4080 5.1 don't work - rear speakers recognized as mic
by GitHub issues - edited 18 Feb '25

18 Feb '25

alsa-project/alsa-ucm-conf issue #510 was edited from atazx: kubuntu 24.10 MSI X870 Mag Tomahawk motherboard, with integrated ALC4080 (usb id `0db0:cd0e`) I tried to install the latest version of alsa-ucm-conf (1.2.13) On this card to have 5.1 audio you have to connect the rear speakers to the front panel, in what would normally be the microphone input, in fact when I connect the speakers they are recognized as a microphone. (only with ucm-conf 1.2.13) front speaker always works rear speakers only work after rebooting from windows center speakers do not work is there a way to fix this problem? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/510 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ALC4080 5.1 don't work - rear speakers recognized as mic
by GitHub issues - opened 18 Feb '25

18 Feb '25

alsa-project/alsa-ucm-conf issue #510 was opened from atazx: kubuntu 24.10 MSI X870 Mag Tomahawk motherboard, with integrated ALC4080 (usb id 0db0:cd0e) I tried to install the latest version of alsa-ucm-conf (1.2.13) On this card to have 5.1 audio you have to connect the rear speakers to the front panel, in what would normally be the microphone input, in fact when I connect the speakers they are recognized as a microphone. (only with ucm-conf 1.2.13) front speaker always works rear speakers only work after rebooting from windows center speakers do not work is there a way to fix this problem? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/510 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v1 1/3] ASoC: tas2781: Support dsp firmware Alpha and Beta seaies
by Shenghao Ding 17 Feb '25

17 Feb '25

For calibration, basic version does not contain any calibration addresses, it depends on calibration tool to convery the addresses to the driver. Since Alpha and Beta firmware, all the calibration addresses are saved into the firmware. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - update the year from 2024 to 2025. - Add dspbin_type to distinguish different dspbin types, such as TASDEV_BASIC, TASDEV_ALPHA, and TASDEV_BETA. - Add struct bulk_reg_val alp_cali_bckp to store the value of pilot tone threshold only for TASDEV_ALPHA & TASDEV_BETA firmware. - Add the declaration of (*fw_parse_fct_param_address)() to parse the addresses of calibration params. - Add the minimum version of TASDEV_ALPHA and TASDEV_BETA firmware. - Add the definition of struct fct_param_address. --- include/sound/tas2781-dsp.h | 26 +++++++++++++++++++++++--- include/sound/tas2781.h | 13 ++++++++++++- 2 files changed, 35 insertions(+), 4 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 3cda9da14f6d..77a240daaae0 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -30,8 +30,10 @@ #define PRE_DEVICE_C 0x12 #define PRE_DEVICE_D 0x16 -#define PPC3_VERSION 0x4100 -#define PPC3_VERSION_TAS2781 0x14600 +#define PPC3_VERSION_BASE 0x4100 +#define PPC3_VERSION_TAS2781_BASIC_MIN 0x14600 +#define PPC3_VERSION_TAS2781_ALPHA_MIN 0x4a00 +#define PPC3_VERSION_TAS2781_BETA_MIN 0x19400 #define TASDEVICE_DEVICE_SUM 8 #define TASDEVICE_CONFIG_SUM 64 @@ -106,6 +108,23 @@ struct tasdevice_calibration { struct tasdevice_data dev_data; }; +struct fct_param_address { + unsigned char thr[3]; + unsigned char thr2[3]; + unsigned char plt_flg[3]; + unsigned char sin_gn[3]; + unsigned char sin_gn2[3]; + /* real-time spk impedance */ + unsigned char r0_reg[3]; + /* check spk connection */ + unsigned char tf_reg[3]; + /* check spk resonant frequency */ + unsigned char a1_reg[3]; + unsigned char a2_reg[3]; + unsigned short size; + char *data; +}; + struct tasdevice_fw { struct tasdevice_dspfw_hdr fw_hdr; unsigned short nr_programs; @@ -114,6 +133,7 @@ struct tasdevice_fw { struct tasdevice_config *configs; unsigned short nr_calibrations; struct tasdevice_calibration *calibrations; + struct fct_param_address fct_par_addr; struct device *dev; }; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 0f1e585635bb..f2eda8ac66e8 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -112,6 +112,12 @@ enum audio_device { TAS2781, }; +enum dspbin_type { + TASDEV_BASIC, + TASDEV_ALPHA, + TASDEV_BETA, +}; + enum device_catlog_id { LENOVO = 0, OTHERS @@ -126,6 +132,7 @@ struct bulk_reg_val { struct tasdevice { struct bulk_reg_val *cali_data_backup; + struct bulk_reg_val alp_cali_bckp; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -170,6 +177,7 @@ struct tasdevice_priv { unsigned char dev_name[32]; const char *name_prefix; unsigned char ndev; + unsigned int dspbin_typ; unsigned int magic_num; unsigned int chip_id; unsigned int sysclk; @@ -195,6 +203,9 @@ struct tasdevice_priv { int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset); + int (*fw_parse_fct_param_address)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, struct tasdev_blk *block); -- 2.34.1

2 3

[PATCH alsa-lib] seq: Fix bogus return of snd_seq_client_info_get_ump_conversion()
by Takashi Iwai 17 Feb '25

17 Feb '25

snd_seq_client_info_get_ump_conversion() should have returned the proper bit of group_filter bit field, but it just did return midi_version field -- a stupid copy & paste error. Let's fix it. Fixes: 2aefb5c41cc0 ("seq: Add UMP support") Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- src/seq/seq.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/src/seq/seq.c b/src/seq/seq.c index d3af93f58579..38cb1afd872f 100644 --- a/src/seq/seq.c +++ b/src/seq/seq.c @@ -1870,7 +1870,7 @@ int snd_seq_client_info_get_ump_groupless_enabled(const snd_seq_client_info_t *i int snd_seq_client_info_get_ump_conversion(const snd_seq_client_info_t *info) { assert(info); - return info->midi_version; + return !(info->group_filter & SNDRV_SEQ_FILTER_NO_CONVERT); } /** -- 2.43.0

1 0

Ask snd_seq_set_client_midi_version
by Correo Alternativo 17 Feb '25

17 Feb '25

Hi, I'm trying out MIDI 2.0 and found a problem, when using snd_seq_set_client_midi_version With SND_SEQ_CLIENT_LEGACY_MIDI I get the following sequence of events when connecting a midi device (using the method snd_seq_event_input or snd_seq_ump_event_input) (timidity command:timidity --interface=A --buffer-fragments=2,8 --output-mode=s1l --sampling-freq=44100): SND_SEQ_EVENT_CLIENT_START SND_SEQ_EVENT_PORT_START SND_SEQ_EVENT_PORT_START SND_SEQ_EVENT_PORT_START SND_SEQ_EVENT_PORT_START But with SND_SEQ_CLIENT_UMP_MIDI_1_0 and SND_SEQ_CLIENT_UMP_MIDI_2_0 I don't get the SND_SEQ_EVENT_PORT_START events, I only get the following: SND_SEQ_EVENT_CLIENT_START SND_SEQ_EVENT_SYSTEM SND_SEQ_EVENT_SYSTEM SND_SEQ_EVENT_SYSTEM I don't know if I'm asking this in the right place, what mistake am I making? Just by modifying that it stops working.

3 3

USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 16 Feb '25

16 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> ### Device interface The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are AES3 (S/PDIF) via coax - channels 10 and 11 are AES3 (S/PDIF) via XLR - channels 12 to 19 are ADAT Optical Interface (8 channels) or S/PDIF (output only, 12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring The device exposes all 20 channels in all three playback modes. No mixer controls are available with the current driver. ### Physical connector layout top to bottom, left to right, viewed from device front #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK, ADAT Optical Interface - _ADAT OUT_: TOSLINK, ADAT Optical Interface or or S/PDIF - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) - RCA in: AES3 or S/PDIF (automatic detection) - RCA out: configurable for AES3 (_Professional_) or S/PDIF (_Consumer_) - XLR in: AES3 or S/PDIF (automatic detection) - XLR out: AES3 only ### Channel configurations The _HiFi_ use case only exposes the most common channel configurations to limit combinatorial complexity: - Analog line outputs in stereo as well as 4.0 and 5.1 surround, - analog microphone and instrument inputs in mono only, - ADAT Optical outputs and inputs in direct 8‐channel mapping, and - all other outputs and inputs in stereo. [alsa-info.txt](https://github.com/user-attachments/files/18814815/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 16 Feb '25

16 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> ### Device interface The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are AES3 (S/PDIF) via coax - channels 10 and 11 are AES3 (S/PDIF) via XLR - channels 12 to 19 are ADAT Optical Interface (8 channels) or S/PDIF (output only, 12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring The device exposes all 20 channels in all three playback modes. No mixer controls are available with the current driver. ### Physical connector layout top to bottom, left to right, viewed from device front #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK, ADAT Optical Interface - _ADAT OUT_: TOSLINK, ADAT Optical Interface or or S/PDIF - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) - RCA in: AES3 or S/PDIF (automatic detection) - RCA out: configurable for AES3 (_Professional_) or S/PDIF (_Consumer_) - XLR in: AES3 or S/PDIF (automatic detection) - XLR out: AES3 only ### Channel configurations The _HiFi_ use case only exposes the most common channel configurations to limit combinatorial complexity: - Analog line outputs in stereo as well as 4.0 and 5.1 surround, - analog microphone and instrument inputs in mono only, - ADAT Optical outputs and inputs in direct 8‐channel mapping, and - all other outputs and inputs in stereo. [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 16 Feb '25

16 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> ### Device interface The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are AES3 (S/PDIF) via coax - channels 10 and 11 are AES3 (S/PDIF) via XLR - channels 12 to 19 are ADAT Optical Interface (8 channels) or S/PDIF (output only, 12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring The device exposes all 20 channels in all three playback modes. No hardware controls are available with the current driver. ### Physical connector layout top to bottom, left to right, viewed from device front #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK, ADAT Optical Interface - _ADAT OUT_: TOSLINK, ADAT Optical Interface or or S/PDIF - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) - RCA in: AES3 or S/PDIF (automatic detection) - RCA out: configurable for AES3 (_Professional_) or S/PDIF (_Consumer_) - XLR in: AES3 or S/PDIF (automatic detection) - XLR out: AES3 only ### Channel configurations The _HiFi_ use case only exposes the most common channel configurations to limit combinatorial complexity: - Analog line outputs in stereo as well as 4.0 and 5.1 surround, - analog microphone and instrument inputs in mono only, - ADAT Optical outputs and inputs in direct 8‐channel mapping, and - all other outputs and inputs in stereo. [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 16 Feb '25

16 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are AES3 (S/PDIF) via coax - channels 10 and 11 are AES3 (S/PDIF) via XLR - channels 12 to 19 are ADAT Optical Interface (8 channels) or S/PDIF (output only, 12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring The device exposes all 20 channels in all three playback modes. No hardware controls are available with the current driver. ### Physical connector layout top to bottom, left to right, viewed from device front #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK, ADAT Optical Interface - _ADAT OUT_: TOSLINK, ADAT Optical Interface or or S/PDIF - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) - RCA in: AES3 or S/PDIF (automatic detection) - RCA out: configurable for AES3 (_Professional_) or S/PDIF (_Consumer_) - XLR in: AES3 or S/PDIF (automatic detection) - XLR out: AES3 only ### Channel configurations The _HiFi_ use case only exposes the most common channel configurations to limit combinatorial complexity: - Analog line outputs in stereo as well as 4.0 and 5.1 surround, - analog microphone and instrument inputs in mono only, - ADAT Optical outputs and inputs in direct 8‐channel mapping, and - all other outputs and inputs in stereo. [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 16 Feb '25

16 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are AES3 (S/PDIF) via coax - channels 10 and 11 are AES3 (S/PDIF) via XLR - channels 12 to 19 are ADAT Optical Interface (8 channels) or S/PDIF (output only, 12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES3 L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring ### Physical connector layout top to bottom, left to right, viewed from device front #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK, ADAT Optical Interface - _ADAT OUT_: TOSLINK, ADAT Optical Interface or or S/PDIF - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) - RCA in: AES3 or S/PDIF (automatic) - RCA out: configurable for AES3 (_Professional_) or S/PDIF (_Consumer_) - XLR in: AES3 or S/PDIF (automatic) - XLR out: AES3 only [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 14 Feb '25

14 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are S/PDIF via coax - channels 10 and 11 are AES/EBU or S/PDIF via XLR - channels 12 to 19 are ADAT Optical Interface (8 channels) or S/PDIF (12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES/EBU L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES/EBU L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring ### Physical connector layout top to bottom, left to right, viewed from device front #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK - _ADAT OUT_: TOSLINK - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[GIT PULL] ASoC fixes for v6.14-rc2
by Mark Brown 14 Feb '25

14 Feb '25

The following changes since commit 2014c95afecee3e76ca4a56956a936e23283f05b: Linux 6.14-rc1 (2025-02-02 15:39:26 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.14-rc2 for you to fetch changes up to 571b69f2f9b1ec7cf7d0e9b79e52115a87a869c4: ASoC: imx-audmix: remove cpu_mclk which is from cpu dai device (2025-02-13 11:34:32 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.14 More fixes and deviec quirks, most of them driver specific including a few SOF robustness fixes. Nothing super remarkable individually. ---------------------------------------------------------------- Bard Liao (2): ASoC: Intel: soc-acpi-intel-tgl-match: declare adr as ull ASoC: Intel: soc-acpi-intel-mtl-match: declare adr as ull Cristian Ciocaltea (4): ASoC: SOF: amd: Add post_fw_run_delay ACP quirk ASoC: SOF: amd: Drop unused includes from Vangogh driver ASoC: SOF: amd: Handle IPC replies before FW_BOOT_COMPLETE ASoC: SOF: amd: Add branch prediction hint in ACP IRQ handler John Keeping (1): ASoC: rockchip: i2s-tdm: fix shift config for SND_SOC_DAIFMT_DSP_[AB] Kuninori Morimoto (4): ASoC: simple-card-utils.c: add missing dlc->of_node ASoC: rsnd: indicate unsupported clock rate ASoC: rsnd: don't indicate warning on rsnd_kctrl_accept_runtime() ASoC: rsnd: adjust convert rate limitation Mark Brown (5): ASoC: Intel: soc-acpi-intel-tgl/mtl-match: declare adr ASoC: rsnd: adjust convert rate in 1% ASoC: SOF: Correct sps->stream and cstream nullity ASoC: Intel: sof_sdw: Add support for new devices ASoC: Intel: soc-acpi-intel-ptl-match typo fixups Nikita Zhandarovich (1): ASoC: fsl_micfil: Enable default case in micfil_set_quality() Peter Ujfalusi (5): ASoC: SOF: stream-ipc: Check for cstream nullity in sof_ipc_msg_data() ASoC: SOF: pcm: Clear the susbstream pointer to NULL on close ASoC: SOF: ipc4-topology: Harden loops for looking up ALH copiers ASoC: Intel: soc-acpi-intel-ptl-match: revise typo of rt712_vb + rt1320 support ASoC: Intel: soc-acpi-intel-ptl-match: revise typo of rt713_vb_l2_rt1320_l13 Richard Fitzgerald (2): ASoC: Intel: sof_sdw: Add lookup of quirk using PCI subsystem ID ASoC: Intel: sof_sdw: Add quirk for Asus Zenbook S14 Shenghao Ding (1): ASoC: tas2781: drop a redundant code Shengjiu Wang (1): ASoC: imx-audmix: remove cpu_mclk which is from cpu dai device Stefan Binding (2): ASoC: cs35l41: Fallback to using HID for system_name if no SUB is available ASoC: cs35l41: Fix acpi_device_hid() not found Terry Cheong (1): ASoC: SOF: Intel: hda: add softdep pre to snd-hda-codec-hdmi module Uday M Bhat (1): ASoC: Intel: sof_sdw: Add support for Fatcat board with BT offload enabled in PTL platform Vitaly Rodionov (1): ASoC: arizona/madera: use fsleep() in up/down DAPM event delays. sound/soc/codecs/arizona.c | 14 +-- sound/soc/codecs/cs35l41.c | 30 ++++-- sound/soc/codecs/madera.c | 10 +- sound/soc/codecs/tas2781-i2c.c | 4 +- sound/soc/codecs/wm5110.c | 8 +- sound/soc/fsl/fsl_micfil.c | 2 + sound/soc/fsl/imx-audmix.c | 31 ------ sound/soc/generic/simple-card-utils.c | 1 + sound/soc/intel/boards/sof_sdw.c | 41 ++++++-- sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 2 +- sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 12 +-- sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 16 +-- sound/soc/renesas/rcar/core.c | 14 --- sound/soc/renesas/rcar/rsnd.h | 1 - sound/soc/renesas/rcar/src.c | 116 +++++++++++++++++----- sound/soc/renesas/rcar/ssi.c | 3 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 4 +- sound/soc/sof/amd/acp-ipc.c | 25 +++-- sound/soc/sof/amd/acp.c | 1 + sound/soc/sof/amd/acp.h | 1 + sound/soc/sof/amd/pci-vangogh.c | 2 - sound/soc/sof/amd/vangogh.c | 22 +++- sound/soc/sof/intel/hda-codec.c | 1 + sound/soc/sof/ipc4-topology.c | 12 ++- sound/soc/sof/pcm.c | 2 + sound/soc/sof/stream-ipc.c | 6 +- 26 files changed, 245 insertions(+), 136 deletions(-)

2 1

USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 14 Feb '25

14 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are S/PDIF via coax - channels 10 and 11 are AES/EBU or S/PDIF via XLR - channels 12 to 19 are ADAT (8 channels) or S/PDIF (12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES/EBU L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES/EBU L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring ### Physical connector layout #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK - _ADAT OUT_: TOSLINK - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - edited 14 Feb '25

14 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was edited from illdefined: This adds support for the RME Fireface UCX II interface. User manual: <https://rme-audio.de/downloads/fface_ucx2_e.pdf> The interface provides 20 playback and capture channels each: - channels 0 to 7 are analog inputs and outputs - channels 8 and 9 are S/PDIF via coax - channels 10 and 11 are AES/EBU or S/PDIF via XLR - channels 12 to 19 are ADAT (8 channels) or S/PDIF (12 and 13) via TOSLINK The interface supports three _Class Compliant_ playback modes: - _8 Ch. + Phones_: - channels 0+1 are mirrored to 6+7 (headphones L+R), 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES/EBU L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _8 Ch. Playback_: - channels 0+1 are mirrored to 8+9 (S/PDIF L+R) and 12+13 (ADAT 1/2) - channels 2+3 are mirrored to 10+11 (AES/EBU L+R) and 14+15 (ADAT 3/4) - channels 4+5 are mirrored to 16+17 (ADAT 5/6) - channels 6+7 are mirrored to 18+19 (ADAT 7/8) - _20 Ch. Playback_: no channel mirroring ### Physical connector layout #### Front - _1 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _2 MIC / LINE_: XLR / TRS ¼" combo, balanced mono input, 48 V phantom power - _3 LINE / INSTR._: TRS ¼", balanced mono input - _4 LINE / INSTR._: TRS ¼", balanced mono input - _🎧︎_ (headphones symbol): TRS ¼", unbalanced stereo output #### Back - _LINE INPUTS (BALANCED)_ [_1_, _2_; _3_, _4_; _5_, _6_]: TRS ¼", balanced mono inputs - _LINE OUTPUTS (BALANCED)_ [_5_, _6_; _7_, _8_]: TRS ¼", balanced mono outputs - _ADAT IN_: TOSLINK - _ADAT OUT_: TOSLINK - _AES/EBU & SPDIF_: 9‐pin D-Sub with breakout cable (2 × RCA and 2 × XLR) [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

USB-Audio: Add support for RME Fireface UCX II
by GitHub pull_request - opened 14 Feb '25

14 Feb '25

alsa-project/alsa-ucm-conf pull request #509 was opened from illdefined: [alsa-info.txt](https://github.com/user-attachments/files/18790978/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/509.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH V4 0/6] soundwire: amd: code improvements and new platform support
by Vijendar Mukunda 13 Feb '25

13 Feb '25

This patch series consists of code improvements and new platform support for ACP7.0 & ACP7.1 platforms. Changes since v3: - Remove extra white line in amd_sdw_manager_start() function. Changes since v2: - Rebase the patch series after dropping the patch which adds conditional check for SoundWire manager device resume when wake event is asserted. Changes since v1: - Drop white line insertion in the code. - Add a conditional check for SoundWire manager device resume when wake event is asserted. - Add new patch to set ACP_PME_EN during runtime suspend sequence - Split the existing implementation of host wake interrupt mask enablement and setting device power state in to two separate patches and refactor set device power state logic. - Drop PME status clear logic. Vijendar Mukunda (6): soundwire: amd: change the soundwire wake enable/disable sequence soundwire: amd: add debug log for soundwire wake event soundwire: amd: add support for ACP7.0 & ACP7.1 platforms soundwire: amd: set device power state during suspend/resume sequence soundwire: amd: set ACP_PME_EN during runtime suspend sequence soundwire: amd: add soundwire host wake interrupt enable/disable sequence drivers/soundwire/amd_manager.c | 149 +++++++++++++++++++++++++++++- drivers/soundwire/amd_manager.h | 26 ++++++ include/linux/soundwire/sdw_amd.h | 2 + 3 files changed, 172 insertions(+), 5 deletions(-) -- 2.34.1

3 8

[PATCH] ASoC: amd: ps: use switch statements for acp pci revision id check
by Vijendar Mukunda 13 Feb '25

13 Feb '25

Use switch statements for acp pci revision id check in SoundWire dma irq handling. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/ps/pci-ps.c | 20 ++++++++++++++------ 1 file changed, 14 insertions(+), 6 deletions(-) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index 220dca8cba85..2ff8e67c19bd 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -111,16 +111,21 @@ static short int check_and_handle_sdw_dma_irq(struct acp63_dev_data *adata, u32 stream_id = ACP63_SDW0_AUDIO2_RX; break; } - if (adata->acp_rev >= ACP70_PCI_REV) - adata->acp70_sdw0_dma_intr_stat[stream_id] = 1; - else + switch (adata->acp_rev) { + case ACP63_PCI_REV: adata->acp63_sdw0_dma_intr_stat[stream_id] = 1; - + break; + case ACP70_PCI_REV: + case ACP71_PCI_REV: + adata->acp70_sdw0_dma_intr_stat[stream_id] = 1; + break; + } sdw_dma_irq_flag = 1; } } } - if (adata->acp_rev == ACP63_PCI_REV) { + switch (adata->acp_rev) { + case ACP63_PCI_REV: if (ext_intr_stat1 & ACP63_P1_AUDIO1_RX_THRESHOLD) { writel(ACP63_P1_AUDIO1_RX_THRESHOLD, adata->acp63_base + ACP_EXTERNAL_INTR_STAT1); @@ -133,7 +138,9 @@ static short int check_and_handle_sdw_dma_irq(struct acp63_dev_data *adata, u32 adata->acp63_sdw1_dma_intr_stat[ACP63_SDW1_AUDIO1_TX] = 1; sdw_dma_irq_flag = 1; } - } else { + break; + case ACP70_PCI_REV: + case ACP71_PCI_REV: if (ext_intr_stat1 & ACP70_P1_SDW_DMA_IRQ_MASK) { for (index = ACP70_P1_AUDIO2_RX_THRESHOLD; index <= ACP70_P1_AUDIO0_TX_THRESHOLD; index++) { @@ -166,6 +173,7 @@ static short int check_and_handle_sdw_dma_irq(struct acp63_dev_data *adata, u32 } } } + break; } return sdw_dma_irq_flag; } -- 2.34.1

2 1

Re: [PATCH v14 29/53] usb: host: xhci-plat: Set XHCI max interrupters if property is present
by Dennis Babcock 12 Feb '25

12 Feb '25

Hi Dennis Louis Babcock jr

1 0

Add HIFI Config support for QCM6490-IDP and QCS6490-RB3Gen2 Qualcomm boards
by GitHub pull_request - opened 12 Feb '25

12 Feb '25

alsa-project/alsa-ucm-conf pull request #508 was opened from mohsRafi: Add HIFI Config support for QCM6490-IDP and QCS6490-RB3Gen2 Qualcomm boards Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/508 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/508.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Add HIFI config support for QCM6490-IDP, QCS6490-RB3Gen2 Qualcomm boards
by GitHub pull_request - opened 12 Feb '25

12 Feb '25

alsa-project/alsa-ucm-conf pull request #507 was opened from mohsRafi: Add HIFI config support for QCM6490-IDP, QCS6490-RB3Gen2 Qualcomm boards Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/507 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/507.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

RE: [Bug 219743] [REGRESSION, BISECTED] ASUS GA402XY Headset mic stopped working after 6.12.2
by Kailang 12 Feb '25

12 Feb '25

Fixup headset MIC no function.

1 0

[PATCH V2 00/23] ASoC: amd: acp70: add soundwire and acp pdm support
by Vijendar Mukunda 11 Feb '25

11 Feb '25

This patch series includes below changes - Add Audio IO support for ACP7.0 and ACP7.1 platforms for SoundWire IO and ACP PDM controller combination. - Add common driver modules for ACP7.0 and ACP7.1 platforms (acp pci driver, Soundwire dma driver, pdm platform driver). - Add SoundWire generic machine driver changes for legacy stack (No DSP enabled) for ACP7.0 & ACP7.1 platforms. - Add SoundWire machines for ACP7.0 & ACP7.1 platforms. Changes since v1: - Change the code sequence for handling SoundWire wake interrupts. - Add new patch to restore host wake interrupt mask - Add new patch to enable ACP PME during ACP init sequence. - Change ACP PCI driver PM ops code sequence. Vijendar Mukunda (23): ASoC: amd: add register header file for ACP7.0 platform ASoC: amd: acp70: add acp pci driver for ACP7.0 and ACP7.1 platforms ASoC: amd: acp70: add acp init and de-init functions ASoC: amd: acp70: add logic for scanning acp child devices ASoC: amd: acp70: create platform devices for acp child nodes ASoC: amd: acp70: enable driver build for ACP7.0 platform ASoC: amd: acp70: add acp pdm platform driver ASoC: amd: acp70: add acp pdm driver dma ops and dai ops ASoC: amd: acp70: add acp soundwire dma driver ASoC: amd: update ACP7.0 KConfig option description ASoC: amd: acp70: add soundwire dma driver dma ops ASoC: amd: acp70: add acp ip interrupt handler ASoC: amd: acp70: add acp pdm driver pm ops ASoC: amd: acp70: add pm ops support for soundwire dma driver ASoC: amd: acp70: add acp driver pm ops support ASoC: amd: acp70: enable wake capability for acp pci driver ASoC: amd: acp70: add soundwire wake interrupt handling ASoC: amd: acp70: enable soundwire host wake irq mask ASoC: amd: acp70: enable acp pme enable during acp init sequence ASoC: amd: acp70: create a device node for soundwire machine driver ASoC: amd: acp: add machine driver changes for ACP7.0 and ACP7.1 platforms ASoC: amd: acp: add RT711, RT714 & RT1316 support for ACP7.0 platform ASoC: amd: acp: amd-acp70-acpi-match: Add rt722 support include/sound/acp70_chip_offset_byte.h | 484 +++++++++++++ sound/soc/amd/Kconfig | 12 + sound/soc/amd/Makefile | 1 + sound/soc/amd/acp/Makefile | 2 +- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 16 + sound/soc/amd/acp/acp-sdw-mach-common.c | 34 + sound/soc/amd/acp/amd-acp70-acpi-match.c | 132 ++++ sound/soc/amd/acp/soc_amd_sdw_common.h | 11 + sound/soc/amd/acp70/Makefile | 9 + sound/soc/amd/acp70/acp70-pdm-dma.c | 463 +++++++++++++ sound/soc/amd/acp70/acp70-sdw-dma.c | 586 ++++++++++++++++ sound/soc/amd/acp70/acp70.h | 288 ++++++++ sound/soc/amd/acp70/pci-acp70.c | 833 +++++++++++++++++++++++ sound/soc/amd/mach-config.h | 1 + 14 files changed, 2871 insertions(+), 1 deletion(-) create mode 100644 include/sound/acp70_chip_offset_byte.h create mode 100644 sound/soc/amd/acp/amd-acp70-acpi-match.c create mode 100644 sound/soc/amd/acp70/Makefile create mode 100644 sound/soc/amd/acp70/acp70-pdm-dma.c create mode 100644 sound/soc/amd/acp70/acp70-sdw-dma.c create mode 100644 sound/soc/amd/acp70/acp70.h create mode 100644 sound/soc/amd/acp70/pci-acp70.c -- 2.34.1

4 35

[PATCH V3 00/25] ASoC: amd: Add support for ACP7.0 & ACP7.1 platforms
by Vijendar Mukunda 10 Feb '25

10 Feb '25

This patch series includes the below changes - Refactor existing ACP6.3 platform ACP PCI driver, SoundWire DMA driver code. - Add Audio IO support for ACP7.0 and ACP7.1 platforms for SoundWire IO and ACP PDM controller combination. - Add SoundWire generic machine driver changes for legacy stack (No DSP enabled) for ACP7.0 & ACP7.1 platforms. - Add SoundWire machines for ACP7.0 & ACP7.1 platforms. Changes since v2: - As per Mark's comment, to avoid code duplication drop separate folder(acp70) for ACP7.0 & ACP7.1 platforms and refactor existing ACP6.3 platform ACP PCI driver, SoundWire DMA driver code and add support for ACP7.0 & ACP7.1 platforms. - Add ACP PCI driver hw_ops for ACP6.3, ACP7.0 & ACP7.1 platforms. - Implement platform specific code in Soundwire DMA driver by adding acp pci revision id checks for ACP6.3/ACP7.0/ACP7.1 platforms. - Refactor existing ACP PCI driver Interrupt handler code. - Add new Soundwire machine for RT1320 Speaker + RT722 multi function codec combination. - Add machine driver quirks for Dell SKU's. Changes since v1: - Change the code sequence for handling SoundWire wake interrupts. - Add new patch to restore host wake interrupt mask - Add new patch to enable ACP PME during ACP init sequence. - Change ACP PCI driver PM ops code sequence. Vijendar Mukunda (25): ASoC: amd: ps: rename structure names, variable and other macros ASoC: amd: ps: use macro for ACP6.3 pci revision id ASoC: amd: ps: add acp pci driver hw_ops for acp6.3 platform ASoC: amd: ps: add callback functions for acp pci driver pm ops ASoC: amd: ps: add callback to read acp pin configuration ASoC: amd: ps: add soundwire dma irq thread callback ASoC: amd: ps: refactor soundwire dma interrupt handling ASoC: amd: ps: store acp revision id in SoundWire dma driver private data ASoC: amd: ps: refactor soundwire dma driver code ASoC: amd: ps: refactor soundwire dma interrupts enable/disable sequence ASoC: amd: ps: rename acp_restore_sdw_dma_config() function ASoC: amd: ps: add pci driver hw_ops for ACP7.0 & ACP7.1 variants ASoC: amd: ps: add pm ops related hw_ops for ACP7.0 & ACP7.1 platforms ASoC: amd: ps: add ACP7.0 & ACP7.1 specific soundwire dma driver changes ASoC: amd: ps: implement function to restore dma config for ACP7.0 platform ASoC: amd: ps: add soundwire dma interrupts handling for ACP7.0 platform ASoC: amd: ps: add soundwire wake interrupt handling ASoC: amd: ps: update module description ASoC: amd: ps: update file description and copyright year ASoC: amd: update Pink Sardine platform Kconfig description ASoC: amd: acp: add machine driver changes for ACP7.0 and ACP7.1 platforms ASoC: amd: acp: add RT711, RT714 & RT1316 support for ACP7.0 platform ASoC: amd: acp: amd-acp70-acpi-match: Add rt722 support ASoC: amd: acp: amd-acp70-acpi-match: Add RT1320 & RT722 combination soundwire machine ASoC: amd: amd_sdw: Add quirks for Dell SKU's sound/soc/amd/Kconfig | 10 +- sound/soc/amd/acp/Kconfig | 1 + sound/soc/amd/acp/Makefile | 2 +- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 45 +++ sound/soc/amd/acp/acp-sdw-mach-common.c | 34 ++ sound/soc/amd/acp/amd-acp70-acpi-match.c | 160 ++++++++ sound/soc/amd/acp/soc_amd_sdw_common.h | 12 + sound/soc/amd/mach-config.h | 1 + sound/soc/amd/ps/Makefile | 2 +- sound/soc/amd/ps/acp63.h | 246 ++++++++++-- sound/soc/amd/ps/pci-ps.c | 422 +++++++++----------- sound/soc/amd/ps/ps-common.c | 475 +++++++++++++++++++++++ sound/soc/amd/ps/ps-pdm-dma.c | 6 +- sound/soc/amd/ps/ps-sdw-dma.c | 375 ++++++++++++++---- 14 files changed, 1440 insertions(+), 351 deletions(-) create mode 100644 sound/soc/amd/acp/amd-acp70-acpi-match.c create mode 100644 sound/soc/amd/ps/ps-common.c -- 2.34.1

3 28

[PATCH] ALSA: hda/conexant: Add quirk for HP ProBook 450 G4 mute LED
by John Veness 10 Feb '25

10 Feb '25

Allows the LED on the dedicated mute button on the HP ProBook 450 G4 laptop to change colour correctly. Signed-off-by: John Veness <john-linux(a)pelago.org.uk> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 538c37a78..84ab357b8 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -1080,6 +1080,7 @@ static const struct hda_quirk cxt5066_fixups[] = { SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x103c, 0x822e, "HP ProBook 440 G4", CXT_FIXUP_MUTE_LED_GPIO), + SND_PCI_QUIRK(0x103c, 0x8231, "HP ProBook 450 G4", CXT_FIXUP_MUTE_LED_GPIO), SND_PCI_QUIRK(0x103c, 0x828c, "HP EliteBook 840 G4", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8299, "HP 800 G3 SFF", CXT_FIXUP_HP_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x829a, "HP 800 G3 DM", CXT_FIXUP_HP_MIC_NO_PRESENCE), -- 2.48.1

1 0

alsaloop doesn't pass sound with libsamplerate resampling on raspberry pi 5
by GitHub issues - edited 10 Feb '25

10 Feb '25

alsa-project/alsa-utils issue #291 was edited from willangley: Hi alsa-utils team! I'm using `alsaloop`s on a Raspberry Pi 5 as part of a slightly ridiculous signal chain: * Yamaha P-525 keyboard * USB C-to-A cable * Raspberry Pi 5 * USB A-to-B cable * ESI U24XL * SPDIF in/out * Focusrite Clarett+ 4Pre * ... * MacBook Pro [Why? The analog Aux In/Aux Out on the keyboard have an always on loopback connection, which is undesirable when playing to a video meeting. The USB port has user controllable loopback.] Two alsaloop processes run on the Pi, one moving sound from the computer to the piano, one moving sound from the piano to the computer. Both run under `supervisord`. I've got them running at acceptable latency — RTL Utility says 40ms from my MacBook Pro. But I haven't been able to get any of the high quality resampling options working -- everything other than add/drop stops passing sound within a few seconds. I'm surprised; this Pi should be powerful enough to handle this. Any ideas how to troubleshoot it? I didn't see errors in logs at the default log level. Thank you! Will Issue URL : https://github.com/alsa-project/alsa-utils/issues/291 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

alsaloop doesn't pass sound with libsamplerate resampling on raspberry pi 5
by GitHub issues - edited 10 Feb '25

10 Feb '25

alsa-project/alsa-utils issue #291 was edited from willangley: Hi alsa-utils team! I'm using `alsaloop`s on a Raspberry Pi 5 as part of a slightly ridiculous signal chain: * Yamaha P-525 keyboard * USB C-to-A cable * Raspberry Pi 5 * USB A-to-B cable * ESI U24XL * SPDIF in/out * Focusrite Clarett+ 4Pre * ... * MacBook Pro [Why? The analog Aux In/Aux Out on the keyboard have an always on loopback connection, which is undesirable when playing to a video meeting. The USB port has user controllable loopback.] Two alsaloop processes run on the Pi, one moving sound from the computer to the piano, one moving sound from the piano to the computer. Both run under `supervisord`. I've got them running at acceptable latency — RTL Utility says 40ms from my MacBook Pro. But I haven't been able to get any of the high quality resampling options working -- everything other than add/drop stops passing sound within a few seconds. I'm surprised; this Pi should be powerful enough to handle this. Any ideas how to troubleshoot it? Thank you! Will Issue URL : https://github.com/alsa-project/alsa-utils/issues/291 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

alsaloop doesn't pass sound with libsamplerate on raspberry pi 5
by GitHub issues - opened 10 Feb '25

10 Feb '25

alsa-project/alsa-utils issue #291 was opened from willangley: Hi alsa-utils team! I'm using `alsaloop`s on a Raspberry Pi 5 as part of a slightly ridiculous signal chain: * Yamaha P-525 keyboard * USB C-to-A cable * Raspberry Pi 5 * USB A-to-B cable * ESI U24XL * SPDIF in/out * Focusrite Clarett+ 4Pre * ... * MacBook Pro [Why? The analog Aux In/Aux Out on the keyboard have an always on loopback connection, which is undesirable when playing to a video meeting. The USB port has user controllable loopback.] Two alsaloop processes run on the Pi, one moving sound from the computer to the piano, one moving sound from the piano to the computer. Both run under `supervisord`. I've got them running at acceptable latency — RTL Utility says 40ms from my MacBook Pro. But I haven't been able to get any of the high quality resampling options working -- everything other than add/drop stops passing sound within a few seconds. I'm surprised; this Pi should be powerful enough to handle this. Any ideas how to troubleshoot it? Thank you! Will Issue URL : https://github.com/alsa-project/alsa-utils/issues/291 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Soundblaster G8 USB Audio not working correctly
by GitHub issues - opened 07 Feb '25

07 Feb '25

alsa-project/alsa-lib issue #437 was opened from Starfoxfs: Soundblaster G8 works partly under Linux Mint 22.1 Cinnamon. Audio playback works in Analog stereo or Duplex mode with headphones and these are displayed as Speakers. Digital SPDIF Out works and also the switch button between headphones and Digital out to external Speakers (only with pipewire not with Pulseaudio). What doesn't work is the Direct Mode and all the Crystalizer surround settings as well as output in virtual 5.1 or 7.1 sound. Microphone doesn´t work, if it plugged in i have no sound at all. [alsa-info.txt](https://github.com/user-attachments/files/18710112/alsa-info… Issue URL : https://github.com/alsa-project/alsa-lib/issues/437 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

rt722: change output volume of speaker to 0dB
by GitHub pull_request - opened 07 Feb '25

07 Feb '25

alsa-project/alsa-ucm-conf pull request #506 was opened from shumingfan: This patch also changes hardware gain to software gain when using volume bar. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/506 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/506.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

soundwire : LnK SoundWire Protocol Analyzer on Linux
by Muni Sekhar 06 Feb '25

06 Feb '25

Hi all, I hope this message finds you well. I am planning to use the LnK SoundWire Protocol Analyzer on a Linux platform. However, I am uncertain whether the current Linux kernel includes the necessary SoundWire drivers to support this . I would appreciate any guidance on how to use the LnK SoundWire Protocol Analyzer with the Linux kernel, as well as information on the specific device drivers required. Could you please provide details on: The compatibility of the LnK SoundWire Protocol Analyzer with the Linux kernel. The steps to set up and use the analyzer on a Linux platform. The necessary device drivers and any additional configurations needed. Thank you for your assistance. -- Thanks, Sekhar

1 0

[PATCH v1] ASoC: tas2781: drop a redundant code
by Shenghao Ding 06 Feb '25

06 Feb '25

Report from internal ticket, priv->cali_data.data devm_kzalloc twice, drop the first one, it is the unnecessary one. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - update the year from 2024 to 2025. - drop a redundant code. --- sound/soc/codecs/tas2781-i2c.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index a730ab6ad4e3..90c5b2e74d12 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -1260,8 +1260,6 @@ static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) (cali_data->cali_dat_sz_per_dev + 1) + 1 + 15 + 1; priv->cali_data.total_sz = priv->ndev * (cali_data->cali_dat_sz_per_dev + 1); - priv->cali_data.data = devm_kzalloc(priv->dev, - ext_cali_data->max, GFP_KERNEL); cali_ctrls[i].name = cali_name; cali_ctrls[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; cali_ctrls[i].info = snd_soc_bytes_info_ext; -- 2.34.1

2 1

[PATCH v1] MAINTAINERS: remove Kevin Lu from TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS
by Shenghao Ding 04 Feb '25

04 Feb '25

Due to internal re-org, Kevin Lu has no longer maintained audio driver. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- MAINTAINERS | 1 - 1 file changed, 1 deletion(-) diff --git a/MAINTAINERS b/MAINTAINERS index 896a307fa065..bc0bf402818d 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -23370,7 +23370,6 @@ F: sound/soc/ti/ TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: linux-sound(a)vger.kernel.org S: Maintained -- 2.34.1

1 0

[PATCH alsa-utils] amixer: implement handling of TLV bytes kcontrols
by Jan Dakinevich 03 Feb '25

03 Feb '25

SND_SOC_BYTES_EXT kcontrols are deprecated and SND_SOC_BYTES_TLV should be used instead. However, there is no support of these kcontrols in amixer. This commit introduces a mechanism to show and write the content of SND_SOC_BYTES_TLV-based kcontrols. Example: $ amixer cset name="Test" 0x10,0x20,0x30,0x40,0x50,0x60,0x70,0x80,0x90,0xa0 $ amixer cget name="Test" numid=42,iface=MIXER,name='Test' ; type=BYTES,access=-----RW-,values=10 : values=0x10,0x20,0x30,0x40,0x50,0x60,0x70,0x80,0x90,0xa0 Signed-off-by: Jan Dakinevich <jan.dakinevich(a)salutedevices.com> --- amixer/amixer.c | 180 +++++++++++++++++++++++++++++++++++++----------- 1 file changed, 141 insertions(+), 39 deletions(-) diff --git a/amixer/amixer.c b/amixer/amixer.c index 07e9819e5cdb..09f774fe2885 100644 --- a/amixer/amixer.c +++ b/amixer/amixer.c @@ -609,6 +609,8 @@ static int show_control(const char *space, snd_hctl_elem_t *elem, { int err; unsigned int item, idx, count, *tlv; + uint8_t *tlv_bytes = NULL; + unsigned int tlv_bytes_count = 0; snd_ctl_elem_type_t type; snd_ctl_elem_id_t *id; snd_ctl_elem_info_t *info; @@ -628,6 +630,18 @@ static int show_control(const char *space, snd_hctl_elem_t *elem, } count = snd_ctl_elem_info_get_count(info); type = snd_ctl_elem_info_get_type(info); + if (type == SND_CTL_ELEM_TYPE_BYTES && + !snd_ctl_elem_info_is_readable(info) && + !snd_ctl_elem_info_is_writable(info)) { + if (count <= 2 * sizeof(unsigned int)) + return -EINVAL; + tlv_bytes_count = count; + count -= 2 * sizeof(unsigned int); + + tlv_bytes = malloc(tlv_bytes_count); + if (!tlv_bytes) + return -ENOMEM; + } printf("%s; type=%s,access=%s,values=%u", space, control_type(info), control_access(info), count); switch (type) { case SND_CTL_ELEM_TYPE_INTEGER: @@ -661,16 +675,30 @@ static int show_control(const char *space, snd_hctl_elem_t *elem, break; } if (level & LEVEL_BASIC) { - if (!snd_ctl_elem_info_is_readable(info)) + if (snd_ctl_elem_info_is_readable(info)) { + err = snd_hctl_elem_read(elem, control); + if (err < 0) { + error("Control %s element read error: %s\n", card, snd_strerror(err)); + return err; + } + } else if (tlv_bytes) { + err = snd_hctl_elem_tlv_read(elem, (void *)tlv_bytes, tlv_bytes_count); + if (err < 0) { + error("Control %s element TLV read error: %s\n", card, snd_strerror(err)); + free(tlv_bytes); + return err; + } + } else goto __skip_read; - if ((err = snd_hctl_elem_read(elem, control)) < 0) { - error("Control %s element read error: %s\n", card, snd_strerror(err)); - return err; - } + printf("%s: values=", space); for (idx = 0; idx < count; idx++) { if (idx > 0) printf(","); + if (tlv_bytes) { + printf("0x%02x", tlv_bytes[idx + 2 * sizeof(unsigned int)]); + continue; + } switch (type) { case SND_CTL_ELEM_TYPE_BOOLEAN: printf("%s", snd_ctl_elem_value_get_boolean(control, idx) ? "on" : "off"); @@ -699,17 +727,11 @@ static int show_control(const char *space, snd_hctl_elem_t *elem, } } printf("\n"); + if (tlv_bytes) + goto __skip_tlv; __skip_read: if (!snd_ctl_elem_info_is_tlv_readable(info)) goto __skip_tlv; - /* skip ASoC ext bytes controls that may have huge binary TLV data */ - if (type == SND_CTL_ELEM_TYPE_BYTES && - !snd_ctl_elem_info_is_readable(info) && - !snd_ctl_elem_info_is_writable(info)) { - printf("%s; ASoC TLV Byte control, skipping bytes dump\n", space); - goto __skip_tlv; - } - tlv = malloc(4096); if ((err = snd_hctl_elem_tlv_read(elem, tlv, 4096)) < 0) { error("Control %s element TLV read error: %s\n", card, snd_strerror(err)); @@ -720,6 +742,7 @@ static int show_control(const char *space, snd_hctl_elem_t *elem, free(tlv); } __skip_tlv: + free(tlv_bytes); return 0; } @@ -1133,16 +1156,110 @@ static int parse_simple_id(const char *str, snd_mixer_selem_id_t *sid) return 0; } +static int parse_bytes(const char *str, void *ptr, unsigned int len) +{ + char *p; + unsigned long tmp; + unsigned int idx = 0; + unsigned char *data = ptr; + + for (;;) { + if (!*str) + break; + if (idx == len) + return -EINVAL; + + tmp = strtoul(str, &p, 0); + if (p - str < 1) + return -EINVAL; + if (*p == ',') + str = p + 1; + else if (*p == '\0') + str = p; + else + return -EINVAL; + + if (tmp > 255) + return -EINVAL; + + data[idx] = tmp; + idx++; + } + + if (!idx) + return -EINVAL; + + return idx; +} + +static int cset_tlv(int argc, char *argv[], snd_ctl_t *handle, + snd_ctl_elem_info_t *info, snd_ctl_elem_id_t *id) +{ + int err; + unsigned int count; + unsigned int *tlv = NULL; + + count = snd_ctl_elem_info_get_count(info); + if (count <= 2 * sizeof(unsigned int)) { + err = -EINVAL; + goto out; + } + tlv = malloc(count); + if (!tlv) { + err = -ENOMEM; + goto out; + } + err = parse_bytes(argv[1], tlv + 2, + count - 2 * sizeof(unsigned int)); + if (err < 0) + goto out; + tlv[0] = 0; + tlv[1] = err; + + err = snd_ctl_elem_tlv_write(handle, id, tlv); + +out: + free(tlv); + return err; +} + +static int cset_elem(int argc, char *argv[], snd_ctl_t *handle, + snd_ctl_elem_info_t *info, snd_ctl_elem_id_t *id) +{ + int err; + snd_ctl_elem_value_t *control; + snd_ctl_elem_value_alloca(&control); + + snd_ctl_elem_value_set_id(control, id); + if ((err = snd_ctl_elem_read(handle, control)) < 0) { + if (!ignore_error) + error("Cannot read the given element from control %s\n", card); + return err; + } + err = snd_ctl_ascii_value_parse(handle, control, info, argv[1]); + if (err < 0) { + if (!ignore_error) + error("Control %s parse error: %s\n", card, snd_strerror(err)); + return err; + } + if ((err = snd_ctl_elem_write(handle, control)) < 0) { + if (!ignore_error) + error("Control %s element write error: %s\n", card, snd_strerror(err)); + return err; + } + + return 0; +} + static int cset(int argc, char *argv[], int roflag, int keep_handle) { int err; static snd_ctl_t *handle = NULL; snd_ctl_elem_info_t *info; snd_ctl_elem_id_t *id; - snd_ctl_elem_value_t *control; + snd_ctl_elem_type_t type; snd_ctl_elem_info_alloca(&info); snd_ctl_elem_id_alloca(&id); - snd_ctl_elem_value_alloca(&control); if (argc < 1) { fprintf(stderr, "Specify a full control identifier: [[iface=<iface>,][name='name',][index=<index>,][device=<device>,][subdevice=<subdevice>]]|[numid=<numid>]\n"); @@ -1175,30 +1292,15 @@ static int cset(int argc, char *argv[], int roflag, int keep_handle) } snd_ctl_elem_info_get_id(info, id); /* FIXME: Remove it when hctl find works ok !!! */ if (!roflag) { - snd_ctl_elem_value_set_id(control, id); - if ((err = snd_ctl_elem_read(handle, control)) < 0) { - if (ignore_error) - return 0; - error("Cannot read the given element from control %s\n", card); - if (! keep_handle) { - snd_ctl_close(handle); - handle = NULL; - } - return err; - } - err = snd_ctl_ascii_value_parse(handle, control, info, argv[1]); - if (err < 0) { - if (!ignore_error) - error("Control %s parse error: %s\n", card, snd_strerror(err)); - if (!keep_handle) { - snd_ctl_close(handle); - handle = NULL; - } - return ignore_error ? 0 : err; - } - if ((err = snd_ctl_elem_write(handle, control)) < 0) { - if (!ignore_error) - error("Control %s element write error: %s\n", card, snd_strerror(err)); + type = snd_ctl_elem_info_get_type(info); + if (type == SND_CTL_ELEM_TYPE_BYTES && + !snd_ctl_elem_info_is_readable(info) && + !snd_ctl_elem_info_is_writable(info)) { + err = cset_tlv(argc, argv, handle, info, id); + } else + err = cset_elem(argc, argv, handle, info, id); + + if (err) { if (!keep_handle) { snd_ctl_close(handle); handle = NULL; -- 2.34.1

1 0

Compilation failure with Clang-21 due to macro conflicts
by GitHub issues - opened 02 Feb '25

02 Feb '25

alsa-project/alsa-lib issue #436 was opened from ms178: **Description:** Attempting to compile alsa-lib (both the `git` version and version 1.2.13) with Clang-21 (commit `bfa7edcc6652bdb37d53e0cec64926aab3f280eb`) results in multiple compilation errors. These errors stem from conflicts between macros defined in `midifile.h` and member names within structures defined in `include/alsa/ump_msg.h`. **Environment:** * **alsa-lib version:** `git` (and 1.2.13) * **Compiler:** Clang-21 (`bfa7edcc6652bdb37d53e0cec64926aab3f280eb`) * **Operating System:** CachyOS * **Build System:** Autotools **Steps to Reproduce:** 1. Obtain the PKGBUILD and the other files from https://github.com/ms178/archpkgbuilds/blob/main/packages/alsa-lib-main/PKG… 2. Uncomment the patch that I apply to workaround the issue. 3. Configure the build to use Clang-21 (`bfa7edcc6652bdb37d53e0cec64926aab3f280eb`). 4. Attempt to compile the library. **Expected Behavior:** The compilation should complete successfully without errors. **Actual Behavior:** The compilation fails with multiple errors similar to the following: ``` In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h:31: ../include/alsa/ump_msg.h:170:27: error: expected member name or ';' after declaration specifiers 170 | snd_ump_msg_midi1_note_t note_on; /**< MIDI1 note-on message */ | ~~~~~~~~~~~~~~~~~~~~~~~~ ^ ./midifile.h:55:27: note: expanded from macro 'note_on' 55 | #define note_on 0x90 | ^ In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h:31: ../include/alsa/ump_msg.h:170:26: error: expected ';' at end of declaration list 170 | snd_ump_msg_midi1_note_t note_on; /**< MIDI1 note-on message */ | ^ | ; ../include/alsa/ump_msg.h:171:27: error: expected member name or ';' after declaration specifiers 171 | snd_ump_msg_midi1_note_t note_off; /**< MIDI1 note-off message */ | ~~~~~~~~~~~~~~~~~~~~~~~~ ^ ./midifile.h:54:27: note: expanded from macro 'note_off' 54 | #define note_off 0x80 | ^ In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h:31: ../include/alsa/ump_msg.h:171:26: error: expected ';' at end of declaration list 171 | snd_ump_msg_midi1_note_t note_off; /**< MIDI1 note-off message */ | ^ | ; ../include/alsa/ump_msg.h:173:26: error: expected member name or ';' after declaration specifiers 173 | snd_ump_msg_midi1_cc CCLD rawmidi _t control_change; /**< MIDI1 control-change message */ | ~~~~~~~~~~~~~~~~~~~~~~ ^ ./midifile.h:57:28: note: expanded from macro 'control_change' 57 | #define control_change 0xb0 | ^ In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h CCLD oldapi :31: ../include/alsa/ump_msg.h:173:24: error: expected ';' at end of declaration list 173 | snd_ump_msg_midi1_cc_t control_change; /**< MIDI1 control-change message */ | ^ | ; ../include/alsa/ump_msg.h:432:27: error: expected member name or ';' after declaration specifiers 432 | snd_ump_msg_midi2_note_t note_on; /**< MIDI2 note-on message */ | ~~~~~~~~~~~~~~~~~~~~~~~~ ^ ./midifile.h:55:27: note: expanded from macro 'note_on' 55 | #define note_on 0x90 | ^ CCLD pcm_min In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h:31: ../include/alsa/ump_msg.h:432:26: error: expected ';' at end of declaration list 432 | snd_ump_msg_midi2_note_t note_on; /**< MIDI2 note-on message */ | ^ | ; ../include/alsa/ump_msg.h:433:27: error: expected member name or ';' after declaration specifiers 433 | snd_ump_msg_midi2_note_t note_off; /**< MIDI2 note-off message */ | ~~~~~~~~~~~~~~~~~~~~~~~~ ^ ./midifile.h:54:27: note: expanded from macro 'note_off' 54 | #define note_off 0x80 | ^ In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h:31: ../include/alsa/ump_msg.h:433:26: error: expected ';' at end of declaration list 433 | snd_ump_msg_midi2_note_t note_off; /**< MIDI2 note-off message */ | ^ | ; ../include/alsa/ump_msg.h:438:26: error: expected member name or ';' after declaration specifiers 438 | snd_ump_msg_midi2_cc_t control_change; /**< MIDI2 control-change message */ | ~~~~~~~~~~~~~~~~~~~~~~ ^ ./midifile.h:57:28: note: expanded from macro 'control_change' 57 | #define control_change 0xb0 | ^ In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h:31: ../include/alsa/ump_msg.h:438:24: error: expected ';' at end of declaration list 438 | snd_ump_msg_midi2_cc_t control_change; /**< MIDI2 control-change message */ | ^ | ; ../include/alsa/ump_msg.h:711:27: error: expected member name or ';' after declaration specifiers 711 | snd_ump_msg_set_tempo_t set_tempo; /**< Set Tempo */ | ~~~~~~~~~~~~~~~~~~~~~~~ ^ ./midifile.c:57:20: note: expanded from macro 'set_tempo' 57 | #define set_tempo 0x51 | ^ In file included from playmidi1.c:51: In file included from ./../include/asoundlib.h:63: In file included from ../include/alsa/seq_event.h:31: ../include/alsa/ump_msg.h:711:25: error: expected ';' at end of declaration list 711 | snd_ump_msg_set_tempo_t set_tempo; /**< Set Tempo */ | ^ | ; CCLD control CCLD chmap 14 errors generated. ``` **Root Cause analysis by AI:** The macros `note_on`, `note_off`, `control_change`, and `set_tempo` defined in `midifile.h` are conflicting with member names used in structures within `include/alsa/ump_msg.h`. This leads to incorrect code expansion and subsequent compilation errors. **Workaround:** The following patch applied to `include/asoundlib-head.h` allows the compilation to succeed: ```diff --- a/include/asoundlib-head.h +++ b/include/asoundlib-head.h @@ -28,6 +28,11 @@ #ifndef __ASOUNDLIB_H #define __ASOUNDLIB_H +#undef note_on +#undef note_off +#undef control_change +#undef set_tempo + #include <unistd.h> #include <stdio.h> #include <stdlib.h> ``` **Suggested Solutions by AI:** While the provided workaround resolves the immediate compilation issue, a more robust solution might involve: 1. **Renaming conflicting members:** Consider renaming the members in `ump_msg.h` (e.g., `midi1_note_on`, `midi2_note_off`, etc.) to avoid clashes with commonly used MIDI event names. 2. **Namespace/prefix:** If renaming is not feasible, consider using a namespace or prefix for the `ump_msg.h` members to prevent conflicts. 3. **Conditional undef:** Instead of unconditionally undefining the macros in `asoundlib-head.h`, a more targeted approach could be to undefine them only within the scope of `ump_msg.h`'s inclusion. 4. **Review macro definitions:** Investigate if the macros in `midifile.h` are strictly necessary or if they can be replaced with constants or enums within a namespace. **Additional Notes:** * I am not a programmer and cannot provide a definitive fix. The provided workaround is based on AI feedback. Issue URL : https://github.com/alsa-project/alsa-lib/issues/436 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

alsaucm fails with unknown If.Condition.Type
by GitHub issues - opened 02 Feb '25

02 Feb '25

alsa-project/alsa-utils issue #290 was opened from abewieland: I get the following when running `listcards`: ``` $ alsaucm listcards ALSA lib ucm_cond.c:367:(if_eval) unknown If.Condition.Type ALSA lib main.c:1554:(snd_use_case_mgr_open) error: failed to import hw:0 use case configuration -22 ALSA lib parser.c:2978:(uc_mgr_scan_master_configs) Unable to open '-hw:0': Invalid argument alsaucm: error failed to get card list: Invalid argument ``` Let me know if I can provide any better debugging data. My system is a Dell laptop with Intel Tiger Lake and Realtek ALC3204, if that's helpful. Issue URL : https://github.com/alsa-project/alsa-utils/issues/290 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

ALC4082 - Asrock B850 Steel Legend
by GitHub issues - opened 01 Feb '25

01 Feb '25

alsa-project/alsa-ucm-conf issue #505 was opened from ct85msi: Hello, ALC4082 isn`t functioning properly, please add the needed support. Here is the output of alsa-info.sh: http://alsa-project.org/db/?f=212e4da6e5c545eeb6d105b17a769202aff651a7 Thanks in advance, Sorin Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/505 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[GIT PULL] ASoC fixes for v6.14-merge-window
by Mark Brown 29 Jan '25

29 Jan '25

The following changes since commit fee89ddd76e45841a2b01d87b481bc02483f4572: ASoC: xilinx: xlnx_spdif: Simpify using devm_clk_get_enabled() (2025-01-16 15:20:41 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.14-merge-window for you to fetch changes up to 6c1bb4031729871fa203983bd77bed1ee3c61347: ASoC: codec: es8316: "DAC Soft Ramp Rate" is just a 2 bit control (2025-01-27 16:22:37 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.14 A bunch of fixes that came in during the merge window, plus a few new device IDs. The i.MX changes are a little large since they add some new quirk data as well as device IDs, and the audio graph card change for picking the correct endpoint for links is large due to updating a number of call sites. ---------------------------------------------------------------- Alexander Boehm (1): ASoC: SOF: imx8m: add SAI2,5,6,7 Arnd Bergmann (1): ASoC: fsl_asrc_m2m: select CONFIG_DMA_SHARED_BUFFER Claudiu Beznea (1): ASoC: da7213: Initialize the mutex Daniel Baluta (2): ASoC: SOF: imx: Add mach entry to select cs42888 topology ASoC: amd: acp: Fix possible deadlock Detlev Casanova (1): ASoC: rockchip: i2s_tdm: Re-add the set_sysclk callback Fabio Estevam (1): ASoC: dt-bindings: ti,pcm1681: Fix the binding title Geert Uytterhoeven (2): ASoC: renesas: SND_SIU_MIGOR should depend on DMADEVICES firmware: cs_dsp: FW_CS_DSP_KUNIT_TEST should not select REGMAP Hans de Goede (1): ASoC: Intel: bytcr_rt5640: Add DMI quirk for Vexia Edu Atla 10 tablet 5V Iuliana Prodan (3): dt-bindings: arm: imx: Add board revisions for i.MX8MP, i.MX8QM and i.MX8QXP ASoC: SOF: imx8: Add entries for new 8QM and 8QXP revisions ASoC: SOF: imx8m: Add entry for new 8M Plus revision Ivaylo Dimitrov (1): ASoC: audio-graph-card2: use correct endpoint when getting link parameters Luoxi Li (1): ASoC: use to_platform_device() instead of container_of() Mario Limonciello (1): ASoC: acp: Support microphone from Lenovo Go S Mark Brown (2): ASoC: fsl: two fixes for asrc memory to memory Add SOF support for new board revisions Ondrej Jirman (1): ASoC: codec: es8316: "DAC Soft Ramp Rate" is just a 2 bit control Shengjiu Wang (2): ASoC: fsl_asrc_m2m: only handle pairs for m2m in the suspend ASoC: fsl_asrc_m2m: return error value in asrc_m2m_device_run() Zhang Yi (1): ASoC: codecs: ES8326: Improved PSRR Documentation/devicetree/bindings/arm/fsl.yaml | 3 ++ .../devicetree/bindings/sound/ti,pcm1681.yaml | 2 +- drivers/firmware/cirrus/Kconfig | 6 +-- sound/soc/amd/acp/acp-i2s.c | 1 + sound/soc/amd/yc/acp6x-mach.c | 28 ++++++++++ sound/soc/codecs/da7213.c | 2 + sound/soc/codecs/es8316.c | 2 +- sound/soc/codecs/es8326.c | 4 +- sound/soc/codecs/rt5514.c | 3 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/fsl_asrc_m2m.c | 16 +++--- sound/soc/generic/audio-graph-card2.c | 62 ++++++++++------------ sound/soc/intel/boards/bytcr_rt5640.c | 17 +++++- sound/soc/renesas/Kconfig | 2 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 31 ++++++++++- sound/soc/sof/imx/imx8.c | 24 ++++++++- sound/soc/sof/imx/imx8m.c | 52 +++++++++++++++++- sound/soc/sof/imx/imx8ulp.c | 3 +- sound/soc/sof/intel/bdw.c | 3 +- sound/soc/sof/intel/byt.c | 3 +- sound/soc/sof/mediatek/mt8186/mt8186.c | 2 +- sound/soc/sof/mediatek/mt8195/mt8195.c | 6 +-- 22 files changed, 205 insertions(+), 68 deletions(-)

2 1

[PATCH V3 0/6] soundwire: amd: code improvements and new platform support
by Vijendar Mukunda 28 Jan '25

28 Jan '25

This patch series consists of code improvements and new platform support for ACP7.0 & ACP7.1 platforms. Changes since v2: - Rebase the patch series after dropping the patch which adds conditional check for SoundWire manager device resume when wake event is asserted. Changes since v1: - Drop white line insertion in the code. - Add a conditional check for SoundWire manager device resume when wake event is asserted. - Add new patch to set ACP_PME_EN during runtime suspend sequence - Split the existing implementation of host wake interrupt mask enablement and setting device power state in to two separate patches and refactor set device power state logic. - Drop PME status clear logic. Vijendar Mukunda (6): soundwire: amd: change the soundwire wake enable/disable sequence soundwire: amd: add debug log for soundwire wake event soundwire: amd: add support for ACP7.0 & ACP7.1 platforms soundwire: amd: set device power state during suspend/resume sequence soundwire: amd: set ACP_PME_EN during runtime suspend sequence soundwire: amd: add soundwire host wake interrupt enable/disable sequence drivers/soundwire/amd_manager.c | 150 +++++++++++++++++++++++++++++- drivers/soundwire/amd_manager.h | 26 ++++++ include/linux/soundwire/sdw_amd.h | 2 + 3 files changed, 173 insertions(+), 5 deletions(-) -- 2.34.1

1 6

[PATCH 00/34] address all -Wunused-const warnings
by Arnd Bergmann 27 Jan '25

27 Jan '25

From: Arnd Bergmann <arnd(a)arndb.de> Compilers traditionally warn for unused 'static' variables, but not if they are constant. The reason here is a custom for C++ programmers to define named constants as 'static const' variables in header files instead of using macros or enums. In W=1 builds, we get warnings only static const variables in C files, but not in headers, which is a good compromise, but this still produces warning output in at least 30 files. These warnings are almost all harmless, but also trivial to fix, and there is no good reason to warn only about the non-const variables being unused. I've gone through all the files that I found using randconfig and allmodconfig builds and created patches to avoid these warnings, with the goal of retaining a clean build once the option is enabled by default. Unfortunately, there is one fairly large patch ("drivers: remove incorrect of_match_ptr/ACPI_PTR annotations") that touches 34 individual drivers that all need the same one-line change. If necessary, I can split it up by driver or by subsystem, but at least for reviewing I would keep it as one piece for the moment. Please merge the individual patches through subsystem trees. I expect that some of these will have to go through multiple revisions before they are picked up, so anything that gets applied early saves me from resending. Arnd Arnd Bergmann (31): powerpc/fsl-soc: hide unused const variable ubsan: fix unused variable warning in test module platform: goldfish: remove ACPI_PTR() annotations i2c: pxa: hide unused icr_bits[] variable 3c515: remove unused 'mtu' variable tracing: hide unused ftrace_event_id_fops Input: synaptics: hide unused smbus_pnp_ids[] array power: rt9455: hide unused rt9455_boost_voltage_values efi: sysfb: don't build when EFI is disabled clk: ti: dpll: fix incorrect #ifdef checks apm-emulation: hide an unused variable sisfb: hide unused variables dma/congiguous: avoid warning about unused size_bytes leds: apu: remove duplicate DMI lookup data iio: ad5755: hook up of_device_id lookup to platform driver greybus: arche-ctrl: move device table to its right location lib: checksum: hide unused expected_csum_ipv6_magic[] sunrpc: suppress warnings for unused procfs functions comedi: ni_atmio: avoid warning for unused device_ids[] table iwlegacy: don't warn for unused variables with DEBUG_FS=n drm/komeda: don't warn for unused debugfs files firmware: qcom_scm: mark qcom_scm_qseecom_allowlist as __maybe_unused crypto: ccp - drop platform ifdef checks usb: gadget: omap_udc: remove unused variable isdn: kcapi: don't build unused procfs code cpufreq: intel_pstate: hide unused intel_pstate_cpu_oob_ids[] net: xgbe: remove extraneous #ifdef checks Input: imagis - remove incorrect ifdef checks sata: mv: drop unnecessary #ifdef checks ASoC: remove incorrect of_match_ptr/ACPI_PTR annotations spi: remove incorrect of_match_ptr annotations drivers: remove incorrect of_match_ptr/ACPI_PTR annotations kbuild: always enable -Wunused-const-variable Krzysztof Kozlowski (1): Input: stmpe-ts - mark OF related data as maybe unused arch/powerpc/sysdev/fsl_msi.c | 2 + drivers/ata/sata_mv.c | 64 +++++++++---------- drivers/char/apm-emulation.c | 5 +- drivers/char/ipmi/ipmb_dev_int.c | 2 +- drivers/char/tpm/tpm_ftpm_tee.c | 2 +- drivers/clk/ti/dpll.c | 10 ++- drivers/comedi/drivers/ni_atmio.c | 2 +- drivers/cpufreq/intel_pstate.c | 2 + drivers/crypto/ccp/sp-platform.c | 14 +--- drivers/dma/img-mdc-dma.c | 2 +- drivers/firmware/efi/Makefile | 3 +- drivers/firmware/efi/sysfb_efi.c | 2 - drivers/firmware/qcom/qcom_scm.c | 2 +- drivers/fpga/versal-fpga.c | 2 +- .../gpu/drm/arm/display/komeda/komeda_dev.c | 8 --- drivers/hid/hid-google-hammer.c | 6 +- drivers/i2c/busses/i2c-pxa.c | 2 +- drivers/i2c/muxes/i2c-mux-ltc4306.c | 2 +- drivers/i2c/muxes/i2c-mux-reg.c | 2 +- drivers/iio/dac/ad5755.c | 1 + drivers/input/mouse/synaptics.c | 2 + drivers/input/touchscreen/imagis.c | 4 +- drivers/input/touchscreen/stmpe-ts.c | 2 +- drivers/input/touchscreen/wdt87xx_i2c.c | 2 +- drivers/isdn/capi/Makefile | 3 +- drivers/isdn/capi/kcapi.c | 7 +- drivers/leds/leds-apu.c | 3 +- drivers/mux/adg792a.c | 2 +- drivers/net/ethernet/3com/3c515.c | 3 - drivers/net/ethernet/amd/xgbe/xgbe-platform.c | 8 --- drivers/net/ethernet/apm/xgene-v2/main.c | 2 +- drivers/net/ethernet/hisilicon/hns_mdio.c | 2 +- drivers/net/wireless/intel/iwlegacy/4965-rs.c | 15 +---- drivers/net/wireless/intel/iwlegacy/common.h | 2 - drivers/platform/goldfish/goldfish_pipe.c | 2 +- drivers/power/supply/rt9455_charger.c | 2 + drivers/regulator/pbias-regulator.c | 2 +- drivers/regulator/twl-regulator.c | 2 +- drivers/regulator/twl6030-regulator.c | 2 +- drivers/rtc/rtc-fsl-ftm-alarm.c | 2 +- drivers/scsi/hisi_sas/hisi_sas_v1_hw.c | 2 +- drivers/scsi/hisi_sas/hisi_sas_v2_hw.c | 2 +- drivers/spi/spi-armada-3700.c | 2 +- drivers/spi/spi-img-spfi.c | 2 +- drivers/spi/spi-meson-spicc.c | 2 +- drivers/spi/spi-meson-spifc.c | 2 +- drivers/spi/spi-orion.c | 2 +- drivers/spi/spi-pic32-sqi.c | 2 +- drivers/spi/spi-pic32.c | 2 +- drivers/spi/spi-rockchip.c | 2 +- drivers/spi/spi-s3c64xx.c | 2 +- drivers/spi/spi-st-ssc4.c | 2 +- drivers/staging/greybus/arche-apb-ctrl.c | 1 + drivers/staging/greybus/arche-platform.c | 9 +-- drivers/staging/pi433/pi433_if.c | 2 +- drivers/tty/serial/amba-pl011.c | 6 +- drivers/tty/serial/ma35d1_serial.c | 2 +- drivers/usb/gadget/udc/omap_udc.c | 10 +-- drivers/video/fbdev/sis/init301.c | 3 +- kernel/dma/contiguous.c | 2 +- kernel/trace/trace_events.c | 4 ++ lib/checksum_kunit.c | 2 + lib/test_ubsan.c | 2 +- net/sunrpc/cache.c | 10 +-- scripts/Makefile.extrawarn | 1 - sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/codecs/rt5514-spi.c | 2 +- sound/soc/qcom/lpass-sc7280.c | 2 +- sound/soc/samsung/aries_wm8994.c | 2 +- 69 files changed, 121 insertions(+), 169 deletions(-) -- 2.39.2 Cc: Michael Ellerman <mpe(a)ellerman.id.au> Cc: Christophe Leroy <christophe.leroy(a)csgroup.eu> Cc: Damien Le Moal <dlemoal(a)kernel.org> Cc: Jiri Kosina <jikos(a)kernel.org> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Corey Minyard <minyard(a)acm.org> Cc: Peter Huewe <peterhuewe(a)gmx.de> Cc: Jarkko Sakkinen <jarkko(a)kernel.org> Cc: Tero Kristo <kristo(a)kernel.org> Cc: Stephen Boyd <sboyd(a)kernel.org> Cc: Ian Abbott <abbotti(a)mev.co.uk> Cc: H Hartley Sweeten <hsweeten(a)visionengravers.com> Cc: Srinivas Pandruvada <srinivas.pandruvada(a)linux.intel.com> Cc: Len Brown <lenb(a)kernel.org> Cc: "Rafael J. Wysocki" <rafael(a)kernel.org> Cc: John Allen <john.allen(a)amd.com> Cc: Herbert Xu <herbert(a)gondor.apana.org.au> Cc: Vinod Koul <vkoul(a)kernel.org> Cc: Ard Biesheuvel <ardb(a)kernel.org> Cc: Bjorn Andersson <andersson(a)kernel.org> Cc: Moritz Fischer <mdf(a)kernel.org> Cc: Liviu Dudau <liviu.dudau(a)arm.com> Cc: Benjamin Tissoires <benjamin.tissoires(a)redhat.com> Cc: Andi Shyti <andi.shyti(a)kernel.org> Cc: Michael Hennerich <michael.hennerich(a)analog.com> Cc: Peter Rosin <peda(a)axentia.se> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Cc: Jonathan Cameron <jic23(a)kernel.org> Cc: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> Cc: Markuss Broks <markuss.broks(a)gmail.com> Cc: Alexandre Torgue <alexandre.torgue(a)foss.st.com> Cc: Lee Jones <lee(a)kernel.org> Cc: Jakub Kicinski <kuba(a)kernel.org> Cc: Shyam Sundar S K <Shyam-sundar.S-k(a)amd.com> Cc: Iyappan Subramanian <iyappan(a)os.amperecomputing.com> Cc: Yisen Zhuang <yisen.zhuang(a)huawei.com> Cc: Stanislaw Gruszka <stf_xl(a)wp.pl> Cc: Kalle Valo <kvalo(a)kernel.org> Cc: Sebastian Reichel <sre(a)kernel.org> Cc: Tony Lindgren <tony(a)atomide.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Alexandre Belloni <alexandre.belloni(a)bootlin.com> Cc: Xiang Chen <chenxiang66(a)hisilicon.com> Cc: "Martin K. Petersen" <martin.petersen(a)oracle.com> Cc: Neil Armstrong <neil.armstrong(a)linaro.org> Cc: Heiko Stuebner <heiko(a)sntech.de> Cc: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> Cc: Vaibhav Hiremath <hvaibhav.linux(a)gmail.com> Cc: Alex Elder <elder(a)kernel.org> Cc: Jiri Slaby <jirislaby(a)kernel.org> Cc: Jacky Huang <ychuang3(a)nuvoton.com> Cc: Helge Deller <deller(a)gmx.de> Cc: Christoph Hellwig <hch(a)lst.de> Cc: Robin Murphy <robin.murphy(a)arm.com> Cc: Steven Rostedt <rostedt(a)goodmis.org> Cc: Masami Hiramatsu <mhiramat(a)kernel.org> Cc: Andrew Morton <akpm(a)linux-foundation.org> Cc: Kees Cook <keescook(a)chromium.org> Cc: Trond Myklebust <trond.myklebust(a)hammerspace.com> Cc: Anna Schumaker <anna(a)kernel.org> Cc: Masahiro Yamada <masahiroy(a)kernel.org> Cc: Nathan Chancellor <nathan(a)kernel.org> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: linuxppc-dev(a)lists.ozlabs.org Cc: linux-kernel(a)vger.kernel.org Cc: linux-ide(a)vger.kernel.org Cc: openipmi-developer(a)lists.sourceforge.net Cc: linux-integrity(a)vger.kernel.org Cc: linux-omap(a)vger.kernel.org Cc: linux-clk(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: linux-crypto(a)vger.kernel.org Cc: dmaengine(a)vger.kernel.org Cc: linux-efi(a)vger.kernel.org Cc: linux-arm-msm(a)vger.kernel.org Cc: linux-fpga(a)vger.kernel.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux-input(a)vger.kernel.org Cc: linux-i2c(a)vger.kernel.org Cc: linux-iio(a)vger.kernel.org Cc: linux-stm32(a)st-md-mailman.stormreply.com Cc: linux-arm-kernel(a)lists.infradead.org Cc: netdev(a)vger.kernel.org Cc: linux-leds(a)vger.kernel.org Cc: linux-wireless(a)vger.kernel.org Cc: linux-rtc(a)vger.kernel.org Cc: linux-scsi(a)vger.kernel.org Cc: linux-spi(a)vger.kernel.org Cc: linux-amlogic(a)lists.infradead.org Cc: linux-rockchip(a)lists.infradead.org Cc: linux-samsung-soc(a)vger.kernel.org Cc: greybus-dev(a)lists.linaro.org Cc: linux-staging(a)lists.linux.dev Cc: linux-serial(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org Cc: linux-fbdev(a)vger.kernel.org Cc: iommu(a)lists.linux.dev Cc: linux-trace-kernel(a)vger.kernel.org Cc: kasan-dev(a)googlegroups.com Cc: linux-hardening(a)vger.kernel.org Cc: linux-nfs(a)vger.kernel.org Cc: linux-kbuild(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org Cc: linux-sound(a)vger.kernel.org

7 7

USB-Audio: ALC4082 - Add support for ASUS ROG Crosshair X870E Hero (U…
by GitHub pull_request - opened 27 Jan '25

27 Jan '25

alsa-project/alsa-ucm-conf pull request #503 was opened from Fabcien: …SB 0b05:1b7c) https://rog.asus.com/motherboards/rog-crosshair/rog-crosshair-x870e-hero/sp… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/503 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/503.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

No optical S/PDIF output on Debian 12 Gnome with ASUS ROG STRIX X670E-A
by GitHub issues - opened 24 Jan '25

24 Jan '25

alsa-project/alsa-ucm-conf issue #500 was opened from BGnoinski: I am currently running into an issue with my Gnome desktop where audio from applications like Firefox is not working. I have run `alsa-info.sh` and the contents [have been uploaded here](http://alsa-project.org/db/?f=d18e76790182eaa8b63e03f435397660b5879b0… I know that debian is running an old version of the `alsa-ucm-conf` package (1.2.8) so I have updated the ucm and ucm2 confs with: ``` #!/bin/bash curl -L -o alsa-ucm-conf.tar.gz https://github.com/alsa-project/alsa-ucm-conf/archive/refs/heads/master.tar… tar xvzf alsa-ucm-conf.tar.gz -C /usr/share/alsa --strip-components=1 --wildcards "*/ucm" "*/ucm2" ``` I can see that my boards identifier(not sure if that's the correct terminology) exists in the ucm2 USB-Audio conf ``` grep -Hrn '0b05:1a52' /usr/share/alsa/ /usr/share/alsa/ucm2/USB-Audio/USB-Audio.conf:58: # 0b05:1a52 ASUS ROG Strix X670E-F & Z790-E Gaming Wifi ``` Even though the comment doesn't have my board specifically in it, looking at the uploaded package I see: ``` Bus 003 Device 003: ID 0b05:1a52 ASUSTek Computer, Inc. USB Audio ``` So my understanding is that should be fine. Even when I look at the [motherboard specs](https://rog.asus.com/us/motherboards/rog-strix/rog-strix-x670e-a-gam… I see that it says it's `ROG SupremeFX 7.1 Surround Sound High Definition Audio CODEC ALC4080` so I don't know if this is different from the Realtek ALC4080 Maybe? And the odd part is if I run `aplay -D plughw:Audio,3 /usr/share/sounds/alsa/Front_Right.wav ` I get the following output: ``` Playing WAVE '/usr/share/sounds/alsa/Front_Right.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono ``` And I do in fact hear audio coming from my speakers. Looking in the sound settings it appears as if I am able to select the correct output, and when audio _should_ be playing the little blue line does go up. ![Image](https://github.com/user-attachments/assets/9e865f37-8153-4bdf-946d-3a428b561321) So it seems like I am just missing something obvious to tie all of the components together to get desktop applications working with the correct output. Any help would be greatly appreciated. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/500 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: Intel: avs: Add UCM files for HDA configuration
by GitHub pull_request - opened 23 Jan '25

23 Jan '25

alsa-project/alsa-ucm-conf pull request #499 was opened from amadeuszslawinski-intel: Add configs for devices using HDA codec. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/499 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/499.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Qualcomm: Add SM8750 MTP HiFi config
by GitHub pull_request - opened 22 Jan '25

22 Jan '25

alsa-project/alsa-ucm-conf pull request #498 was opened from krzk: Add UCM2 configs for the Qualcomm SM8750-MTP Board to handle: - on-board Speakers - Headphones speakers - Top and Bottom on-board Microphones Not yet implemented/tested are Headphone Microphones over USB-C port. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/498 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/498.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[RESEND PATCH v1] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 22 Jan '25

22 Jan '25

Due to internal re-org, Kevin has no longer maintained audio driver. With the job responsibility change, drop entries for the audio converter and add entries for both haptics drivers and middle-power audio amplifier drivers. Add audio converter, set the Status as Supported. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- MAINTAINERS | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 58c3381725fb..41a4bd0cd0a7 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -23206,25 +23206,42 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: linux-sound(a)vger.kernel.org -S: Maintained +S: Supported +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml F: Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml +F: drivers/input/misc/drv2*.c +F: include/sound/tas2*.h +F: include/sound/tas5086.h +F: include/sound/tpa6130a2-plat.h +F: sound/pci/hda/tas2781_hda_i2c.c +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tas5*.* +F: sound/soc/codecs/tpa6130a2.* + +TEXAS INSTRUMENTS AUDIO Converter (ASoC) and Touch Screen DRIVERS +L: linux-sound(a)vger.kernel.org +S: Maintained F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm6240.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: include/sound/tas2*.h +F: drivers/input/touchscreen/tsc2*.* F: include/sound/tlv320*.h -F: include/sound/tpa6130a2-plat.h -F: sound/pci/hda/tas2781_hda_i2c.c F: sound/soc/codecs/pcm1681.c F: sound/soc/codecs/pcm1789*.* F: sound/soc/codecs/pcm179x*.* @@ -23234,9 +23251,8 @@ F: sound/soc/codecs/pcm3060*.* F: sound/soc/codecs/pcm3168a*.* F: sound/soc/codecs/pcm5102a.c F: sound/soc/codecs/pcm512x*.* -F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/pcm6240.* F: sound/soc/codecs/tlv320*.* -F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> base-commit: c4b9570cfb63501638db720f3bee9f6dfd044b82 -- 2.34.1

1 0

[PATCH V2 3/7] soundwire: amd: add conditional for check for device resume
by Vijendar Mukunda 21 Jan '25

21 Jan '25

Add a conditional check to resume SoundWire manager device, when the SoundWire manager instance is in the suspended state. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 60be5805715e..6831b3729db5 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -850,7 +850,10 @@ static void amd_sdw_update_slave_status(u32 status_change_0to7, u32 status_chang static void amd_sdw_process_wake_event(struct amd_sdw_manager *amd_manager) { dev_dbg(amd_manager->dev, "SoundWire Wake event reported\n"); - pm_request_resume(amd_manager->dev); + if (pm_runtime_suspended(amd_manager->dev)) { + dev_dbg(amd_manager->dev, "Device is in suspend state\n"); + pm_request_resume(amd_manager->dev); + } writel(0x00, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); writel(0x00, amd_manager->mmio + ACP_SW_STATE_CHANGE_STATUS_8TO11); } -- 2.34.1

2 1

[PATCH] ALSA: hda/realtek: Enable Mute LED on HP Laptop 14s-fq1xxx
by Sebastian Wiese-Wagner 20 Jan '25

20 Jan '25

This HP Laptop uses ALC236 codec with COEF 0x07 controlling the mute LED. Enable existing quirk for this device. Signed-off-by: Sebastian Wiese-Wagner <seb(a)fastmail.to> --- 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 46623cd615a4..d3c9ed963588 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10415,6 +10415,7 @@ static const struct hda_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8870, "HP ZBook Fury 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x8873, "HP ZBook Studio 15.6 Inch G8 Mobile Workstation PC", ALC285_FIXUP_HP_GPIO_AMP_INIT), SND_PCI_QUIRK(0x103c, 0x887a, "HP Laptop 15s-eq2xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), + SND_PCI_QUIRK(0x103c, 0x887c, "HP Laptop 14s-fq1xxx", ALC236_FIXUP_HP_MUTE_LED_COEFBIT2), SND_PCI_QUIRK(0x103c, 0x888a, "HP ENVY x360 Convertible 15-eu0xxx", ALC245_FIXUP_HP_X360_MUTE_LEDS), SND_PCI_QUIRK(0x103c, 0x888d, "HP ZBook Power 15.6 inch G8 Mobile Workstation PC", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8895, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_SPEAKERS_MICMUTE_LED), -- 2.48.1

1 0

[PATCH] ASoC: codecs: ES8326: Improved PSRR
by Zhang Yi 20 Jan '25

20 Jan '25

Modified configuration to improve PSSR when ES8326 is working Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index b06eead7e0f6..066d92b54312 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -911,7 +911,7 @@ static void es8326_jack_detect_handler(struct work_struct *work) regmap_write(es8326->regmap, ES8326_INT_SOURCE, (ES8326_INT_SRC_PIN9 | ES8326_INT_SRC_BUTTON)); regmap_write(es8326->regmap, ES8326_SYS_BIAS, 0x1f); - regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF, 0x0f, 0x08); + regmap_update_bits(es8326->regmap, ES8326_HP_DRIVER_REF, 0x0f, 0x0d); queue_delayed_work(system_wq, &es8326->jack_detect_work, msecs_to_jiffies(400)); es8326->hp = 1; @@ -1023,7 +1023,7 @@ static void es8326_init(struct snd_soc_component *component) struct es8326_priv *es8326 = snd_soc_component_get_drvdata(component); regmap_write(es8326->regmap, ES8326_RESET, 0x1f); - regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x0E); + regmap_write(es8326->regmap, ES8326_VMIDSEL, 0x3E); regmap_write(es8326->regmap, ES8326_ANA_LP, 0xf0); usleep_range(10000, 15000); regmap_write(es8326->regmap, ES8326_HPJACK_TIMER, 0xd9); -- 2.17.1

1 0

[GIT PULL] ASoC updates for v6.14
by Mark Brown 20 Jan '25

20 Jan '25

The following changes since commit 9d89551994a430b50c4fffcb1e617a057fa76e20: Linux 6.13-rc6 (2025-01-05 14:13:40 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v6.14 for you to fetch changes up to fee89ddd76e45841a2b01d87b481bc02483f4572: ASoC: xilinx: xlnx_spdif: Simpify using devm_clk_get_enabled() (2025-01-16 15:20:41 +0000) ---------------------------------------------------------------- ASoC: Updates for v6.14 This was quite a quiet release for what I imagine are holiday related reasons, the diffstat is dominated by some Cirrus Logic Kunit tests. There's the usual mix of small improvements and fixes, plus a few new drivers and features. The diffstat includes some DRM changes due to work on HDMI audio. - Allow clocking on each DAI in an audio graph card to be configured separately. - Improved power management for Renesas RZ-SSI. - KUnit testing for the Cirrus DSP framework. - Memory to meory operation support for Freescale/NXP platforms. - Support for pause operations in SOF. - Support for Allwinner suinv F1C100s, Awinc AW88083, Realtek ALC5682I-VE ---------------------------------------------------------------- Alexey Klimov (1): ASoC: qcom: sdm845: add handling of secondary MI2S clock Amadeusz Sławiński (4): ASoC: Intel: avs: Fix init-config parsing ASoC: Intel: avs: Print IPC error messages in lower layer ASoC: Intel: avs: Add MODULE_FIRMWARE to inform about FW ASoC: Intel: avs: Update ASRC definition Andrew Davis (21): ASoC: ad193x: Remove use of i2c_match_id() ASoC: adau1761: Remove use of i2c_match_id() ASoC: adau1781: Remove use of i2c_match_id() ASoC: adau1977: Remove use of i2c_match_id() ASoC: alc5623: Remove use of i2c_match_id() ASoC: alc5632: Remove use of i2c_match_id() ASoC: max98088: Remove use of i2c_match_id() ASoC: max98090: Remove use of i2c_match_id() ASoC: max98095: Remove use of i2c_match_id() ASoC: pcm186x: Remove use of i2c_match_id() ASoc: pcm6240: Remove use of i2c_match_id() ASoC: ssm2602: Remove use of i2c_match_id() ASoC: tas2562: Remove use of i2c_match_id() ASoC: tas2781: Remove use of i2c_match_id() ASoC: tas5720: Remove use of i2c_match_id() ASoC: tlv320adc3xxx: Remove use of i2c_match_id() ASoC: tlv320aic31xx: Remove use of i2c_match_id() ASoC: tlv320aic3x: Remove use of i2c_match_id() ASoC: tpa6130a2: Remove use of i2c_match_id() ASoC: wm8904: Remove use of i2c_match_id() ASoC: wm8985: Remove use of i2c_match_id() Andy Shevchenko (1): ASoc: mediatek: mt8365: Don't use "proxy" headers Arnd Bergmann (1): firmware: cs_dsp: avoid large local variables Bard Liao (7): ASoC: SOF: Intel: hda-dai: Ensure DAI widget is valid during params ASoC: Intel: sof_sdw: correct mach_params->dmic_num ASoC: Intel: sof_sdw: reduce log level for not using internal dmic ASoC: Intel: sof_sdw: improve the log of DAI link numbers ASoC: Intel: soc-acpi-intel-ptl-match: add rt712_vb + rt1320 support ASoC: Intel: soc-acpi-intel-lnl-match: add rt713_vb_l2_rt1320_l13 support ASoC: Intel: soc-acpi-intel-ptl-match: add rt713_vb_l2_rt1320_l13 support Bence Csókás (1): ASoC: sun4i-codec: Use new devm clk and reset APIs Cezary Rojewski (9): ASoC: Intel: avs: Do not readq() u32 registers ASoC: Intel: avs: Fix the minimum firmware version numbers ASoC: Intel: avs: Fix theoretical infinite loop ASoC: Intel: avs: Update hda component teardown sequences ASoC: Intel: avs: Clearly state assumptions of hw_params() ASoC: Intel: avs: Improve logging of firmware loading ASoC: Intel: avs: Adjust DSP status register names ASoC: Intel: avs: Adjust IPC traces ASoC: Intel: avs: Add missing includes Chancel Liu (4): ASoC: fsl_utils: Add function to constrain rates ASoC: fsl_micfil: Switch to common sample rate constraint function ASoC: fsl_xcvr: Add sample rate constraint ASoC: fsl_sai: Add sample rate constraint Charles Keepax (5): ASoC: SDCA: Add missing header includes ASoC: SDCA: Clean up error messages ASoC: SDCA: Add bounds check for function address ASoC: SDCA: Add missing function type names ASoC: SDCA: Split function type patching and function naming Christophe JAILLET (1): ASoC: cs42l51: Constify struct i2c_device_id Claudiu Beznea (16): ASoC: renesas: rz-ssi: Terminate all the DMA transactions ASoC: renesas: rz-ssi: Use only the proper amount of dividers ASoC: renesas: rz-ssi: Fix typo on SSI_RATES macro comment ASoC: renesas: rz-ssi: Remove pdev member of struct rz_ssi_priv ASoC: renesas: rz-ssi: Remove the rz_ssi_get_dai() function ASoC: renesas: rz-ssi: Remove the first argument of rz_ssi_stream_is_play() ASoC: renesas: rz-ssi: Use readl_poll_timeout_atomic() ASoC: renesas: rz-ssi: Use temporary variable for struct device ASoC: renesas: rz-ssi: Use goto label names that specify their actions ASoC: renesas: rz-ssi: Rely on the ASoC subsystem to runtime resume/suspend the SSI ASoC: renesas: rz-ssi: Enable runtime PM autosuspend support ASoC: renesas: rz-ssi: Add runtime PM support ASoC: renesas: rz-ssi: Issue software reset in hw_params API ASoC: renesas: rz-ssi: Add suspend to RAM support ASoC: dt-bindings: renesas,rz-ssi: Remove DMA description ASoC: dt-bindings: renesas,rz-ssi: Document the Renesas RZ/G3S SoC Colin Ian King (1): ASoC: soc-core: remove redundant assignment to variable ret Csókás Bence (1): ASoC: dt-bindings: Add Allwinner suniv F1C100s Audio Codec Dan Carpenter (1): ASoC: renesas: rz-ssi: Add a check for negative sample_space Derek Fang (1): ASoC: rt5682: Support the ALC5682I-VE codec Dheeraj Reddy Jonnalagadda (1): ASoC: rockchip: i2s-tdm: Fix a useless call issue Dr. David Alan Gilbert (1): ASoC: rt715: Remove unused hda_to_sdw George Lander (1): ASoC: sun4i-spdif: Add clock multiplier settings Krzysztof Kozlowski (3): ASoC: codecs: wcd9335: Add define for number of DAIs ASoC: dt-bindings: qcom,wcd9335: Drop number of DAIs from the header ASoC: dt-bindings: Correct indentation and style in DTS example Kuninori Morimoto (18): ASoC: simple-card-utils: use __free(device_node) for device node ASoC: simple-card-utils: check port reg first on graph_get_dai_id() ASoC: simple-card-utils: use for_each_of_graph_port() on graph_get_dai_id() ASoC: soc-core: tidyup ret handling for card->disable_route_checks ASoC: soc-pcm: don't use soc_pcm_ret() on .prepare callback ASoC: simple-card-utils: Unify clock direction by clk_direction ASoC: dt-bindings: renesas,rsnd: remove post-init-providers property ASoC: soc-card: remove card check ASoC: intel: remove disable_route_checks ASoC: remove disable_route_checks ASoC: audio-graph-card2: use __free(device_node) for device node ASoC: audio-graph-card: use __free(device_node) for device node ASoC: simple-card: use __free(device_node) for device node ASoC: soc-core: return 0 if np was NULL on snd_soc_daifmt_parse_clock_provider_raw() ASoC: audio-graph-card2: use of_graph_get_port_by_id() at graph_get_next_multi_ep() ASoC: soc-core: Enable to use extra format on each DAI ASoC: audio-graph-card2: Use extra format on each DAI ASoC: soc-dapm: remove !card check from snd_soc_dapm_set_bias_level() Marcus Cooper (2): ASoC: sun4i-spdif: Always set the valid data to be the MSB ASoC: sun4i-spdif: Add working 24bit audio support Mark Brown (23): ASoC: fsl_xcvr: Add suspend and resume support ASoC: Correct *-objs usages Add support for codec of F1C100s ASoC: Splitting cs35l56 SoundWire DAI into separate ASoC: Merge up origin to resolve interaction with manline symbol changes ASoC: Merge up v6.12-rc2 ASoC: Intel: boards: updates for 6.14 ASoC: sun4i-spdif: Add 24bit support Add function to constrain rates ASoC: simple-card-utils: tidyup for Multi connection Add audio support for the Renesas RZ/G3S SoC ASoC: SOF: core/Intel: Handle pause supported token firmware: cirrus: Add KUnit tests for cs_dsp ASoC: fsl: add memory to memory function for ASRC ASoC: Intel: Add matches for Cirrus Logic CDB35L56 Minor debugging improvements ASoC: codecs: Add aw88083 amplifier driver ASoC: Merge up v6.13-rc6 ASoC: Intel: avs: Fixes and cleanups ASoC: remove disable_route_checks ASoC: fsl: Support MQS on i.MX943 ASoC: extra format on each DAI ASoC: fsl: Support micfil on i.MX943 Martin Blumenstingl (2): ASoC: soc-dai: add snd_soc_dai_prepare() and use it internally ASoC: dapm: add support for preparing streams Mesih Kilinc (2): ASoC: sun4i-codec: Add DMA Max Burst field ASoC: sun4i-codec: Add support for Allwinner suniv F1C100s Michal Simek (1): ASoC: xilinx: xlnx_spdif: Simpify using devm_clk_get_enabled() Neil Armstrong (1): ASoC: dt-bindings: convert rt5682.txt to dt-schema Nícolas F. R. A. Prado (2): ASoC: dt-bindings: mediatek,mt8188-mt6359: Add compatible for mt8390 evk ASoC: dt-bindings: mediatek,mt8188-mt6359: Allow DL_SRC/UL_SRC dai-links Peter Ujfalusi (4): ASoC: SOF: Add support for pause supported tokens from topology ASoC: SOF: Intel: hda-pcm: Follow the pause_supported flag to drop PAUSE support ASoC: SOF: ipc4-topology: Use macro to set the EXT_PARAM_SIZE in widget setup ASoC: SOF: sof-priv: Remove unused SOF_DAI_STREAM() and SOF_FORMATS Richard Fitzgerald (21): ASoC: cs35l56: Split SoundWire DAI into separate playback and capture ASoC: sdw_utils: cs_amp: Assign non-overlapping TDM masks for each codec on a bus firmware: cs_dsp: Add mock regmap for KUnit testing firmware: cs_dsp: Add mock DSP memory map for KUnit testing firmware: cs_dsp: Add mock wmfw file generator for KUnit testing firmware: cs_dsp: Add mock bin file generator for KUnit testing firmware: cs_dsp: Add KUnit testing of bin file download firmware: cs_dsp: Add KUnit testing of wmfw download firmware: cs_dsp: Add KUnit testing of control parsing firmware: cs_dsp: Add KUnit testing of control cache firmware: cs_dsp: Add KUnit testing of control read/write firmware: cs_dsp: Add KUnit testing of bin error cases firmware: cs_dsp: Add KUnit testing of wmfw error cases firmware: cs_dsp: Add KUnit testing of client callbacks ASoC: Intel: tgl-match: Add entries for CS35L56 on CDB35L56-EIGHT-C ASoC: Intel: mtl-match: Add CDB35L56-EIGHT-C with aggregated speakers ASoC: Intel: mtl-match: Add CDB35L56-EIGHT-C 8x CS35L56 without CS42L43 firmware: cs_dsp: Fix kerneldoc typos in cs_dsp_mock_bin.c firmware: cs_dsp: Avoid using a u32 as a __be32 in cs_dsp_mock_mem_maps.c firmware: cs_dsp: Fix endianness conversion in cs_dsp_mock_wmfw.c firmware: cs_dsp: Delete redundant assignments in cs_dsp_test_bin.c Rob Herring (Arm) (2): ASoC: Use of_property_present() for non-boolean properties ASoC: simple-card: Drop unnecessary "dai-tdm-slot-width-map" property presence check Shengjiu Wang (13): ASoC: fsl_xcvr: Use regmap for PHY and PLL registers ASoC: fsl_xcvr: Add suspend and resume support ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device ASoC: fsl_asrc_m2m: force cast for snd_pcm_format_t type ASoC: fsl_mqs: Add i.MX943 platform support ASoC: dt-bindings: fsl,mqs: Add compatible string for i.MX943 platform ASoC: fsl_micfil: Add i.MX943 platform support ASoC: dt-bindings: fsl,micfil: Add compatible string for i.MX943 platform Simon Trimmer (4): ASoC: Intel: sof_sdw: Correct quirk for Lenovo Yoga Slim 7 ASoC: Intel: sof_sdw: Add a dev_dbg message for the SOC_SDW_CODEC_MIC quirk ASoC: Intel: soc-acpi: arl: Correct naming of a cs35l56 address struct ASoC: Intel: soc-acpi: arl: Add match entries for new cs42l43 laptops Stephen Gordon (2): ASoC: simple_card: Improve debugging messages ASoC: simple_card: Show if link is unidirectional Takashi Iwai (5): ASoC: cs40l50: Use *-y for Makefile ASoC: mediatek: mt8365: Use *-y for Makefile ASoC: SDCA: Use *-y for Makefile ASoC: cs42l84: Use *-y for Makefile ASoC: wcd937x: Use *-y for Makefile Tang Bin (2): ASoC: mediatek: mt8192-afe-pcm: Simplify probe() with local dev variable ASoC: tas2781: Fix redundant logical jump Thorsten Blum (4): ASoC: SOF: Intel: Use str_yes_no() helper in atom_dump() ASoC: madera: Use str_enabled_disabled() helper function ASoC: SOF: Intel: Use str_yes_no() to improve bdw_dump() ASoC: codecs: Use ARRAY_SIZE() to calculate PEB2466_TLV_SIZE Uwe Kleine-König (1): ASoC: Drop explicit initialization of struct i2c_device_id::driver_data to 0 Vijendar Mukunda (2): ASoC: amd: ps: update mach params subsystem_rev variable ASoC: amd: ps: add ZSC control register programming sequence Weidong Wang (2): ASoC: dt-bindings: Add schema for "awinic,aw88083" ASoC: codecs: Add aw88083 amplifier driver Wolfram Sang (1): ASoC: cs42l43: don't include '<linux/find.h>' directly Yu-Chun Lin (1): ASoC: amd: Add ACPI dependency to fix build error .../devicetree/bindings/sound/adi,ssm2518.yaml | 20 +- .../bindings/sound/allwinner,sun4i-a10-codec.yaml | 31 + .../devicetree/bindings/sound/awinic,aw88395.yaml | 2 + .../devicetree/bindings/sound/everest,es71x4.yaml | 10 +- .../devicetree/bindings/sound/everest,es7241.yaml | 19 +- .../devicetree/bindings/sound/fsl,easrc.yaml | 32 +- .../devicetree/bindings/sound/fsl,micfil.yaml | 1 + .../devicetree/bindings/sound/fsl,mqs.yaml | 2 + .../devicetree/bindings/sound/fsl,xcvr.yaml | 34 +- .../bindings/sound/intel,keembay-i2s.yaml | 32 +- .../bindings/sound/mediatek,mt8188-mt6359.yaml | 16 +- .../bindings/sound/neofidelity,ntp8918.yaml | 26 +- .../devicetree/bindings/sound/realtek,rt5682.yaml | 156 + .../devicetree/bindings/sound/renesas,rsnd.yaml | 6 - .../devicetree/bindings/sound/renesas,rz-ssi.yaml | 19 +- Documentation/devicetree/bindings/sound/rt5682.txt | 98 - .../devicetree/bindings/sound/ti,pcm6240.yaml | 32 +- .../devicetree/bindings/sound/ti,tas2562.yaml | 30 +- .../devicetree/bindings/sound/ti,tas2770.yaml | 34 +- .../devicetree/bindings/sound/ti,tas2781.yaml | 38 +- .../devicetree/bindings/sound/ti,tas27xx.yaml | 34 +- .../devicetree/bindings/sound/ti,tas57xx.yaml | 36 +- MAINTAINERS | 4 +- drivers/firmware/cirrus/Kconfig | 20 + drivers/firmware/cirrus/Makefile | 2 + drivers/firmware/cirrus/test/Makefile | 23 + drivers/firmware/cirrus/test/cs_dsp_mock_bin.c | 199 ++ .../firmware/cirrus/test/cs_dsp_mock_mem_maps.c | 752 +++++ drivers/firmware/cirrus/test/cs_dsp_mock_regmap.c | 367 +++ drivers/firmware/cirrus/test/cs_dsp_mock_utils.c | 13 + drivers/firmware/cirrus/test/cs_dsp_mock_wmfw.c | 473 +++ drivers/firmware/cirrus/test/cs_dsp_test_bin.c | 2556 +++++++++++++++ .../firmware/cirrus/test/cs_dsp_test_bin_error.c | 600 ++++ .../firmware/cirrus/test/cs_dsp_test_callbacks.c | 688 ++++ .../cirrus/test/cs_dsp_test_control_cache.c | 3282 ++++++++++++++++++++ .../cirrus/test/cs_dsp_test_control_parse.c | 1851 +++++++++++ .../firmware/cirrus/test/cs_dsp_test_control_rw.c | 2669 ++++++++++++++++ drivers/firmware/cirrus/test/cs_dsp_test_wmfw.c | 2211 +++++++++++++ .../firmware/cirrus/test/cs_dsp_test_wmfw_error.c | 1347 ++++++++ drivers/firmware/cirrus/test/cs_dsp_tests.c | 14 + include/dt-bindings/sound/qcom,wcd9335.h | 1 - include/linux/firmware/cirrus/cs_dsp_test_utils.h | 160 + include/sound/sdca.h | 7 +- include/sound/sdca_function.h | 3 + include/sound/simple_card_utils.h | 8 +- include/sound/soc-dai.h | 3 + include/sound/soc.h | 12 +- include/sound/soc_sdw_utils.h | 2 + include/uapi/sound/compress_params.h | 23 +- include/uapi/sound/sof/tokens.h | 2 + sound/soc/amd/Kconfig | 2 +- sound/soc/amd/ps/pci-ps.c | 16 +- sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/Makefile | 8 +- sound/soc/codecs/ad193x-i2c.c | 3 +- sound/soc/codecs/adau1761-i2c.c | 5 +- sound/soc/codecs/adau1781-i2c.c | 5 +- sound/soc/codecs/adau1977-i2c.c | 5 +- sound/soc/codecs/alc5623.c | 10 +- sound/soc/codecs/alc5632.c | 6 +- sound/soc/codecs/aw88081.c | 333 +- sound/soc/codecs/aw88081.h | 43 + sound/soc/codecs/cs35l56.c | 8 +- sound/soc/codecs/cs42l43.c | 2 +- sound/soc/codecs/cs42l51-i2c.c | 6 +- sound/soc/codecs/cs42l84.c | 2 +- sound/soc/codecs/es8323.c | 2 +- sound/soc/codecs/madera.c | 7 +- sound/soc/codecs/max98088.c | 4 +- sound/soc/codecs/max98090.c | 18 +- sound/soc/codecs/max98095.c | 4 +- sound/soc/codecs/ntp8835.c | 2 +- sound/soc/codecs/ntp8918.c | 2 +- sound/soc/codecs/pcm186x-i2c.c | 3 +- sound/soc/codecs/pcm6240.c | 3 +- sound/soc/codecs/peb2466.c | 3 +- sound/soc/codecs/rt5682-i2c.c | 6 + sound/soc/codecs/rt5682.c | 12 +- sound/soc/codecs/rt5682.h | 2 + sound/soc/codecs/rt715-sdw.c | 41 - sound/soc/codecs/rt715.h | 3 - sound/soc/codecs/sma1307.c | 4 +- sound/soc/codecs/ssm2602-i2c.c | 5 +- sound/soc/codecs/tas2562.c | 4 +- sound/soc/codecs/tas2781-i2c.c | 71 +- sound/soc/codecs/tas5720.c | 10 +- sound/soc/codecs/tlv320adc3xxx.c | 4 +- sound/soc/codecs/tlv320aic31xx.c | 6 +- sound/soc/codecs/tlv320aic3x-i2c.c | 3 +- sound/soc/codecs/tpa6130a2.c | 4 +- sound/soc/codecs/uda1342.c | 2 +- sound/soc/codecs/wcd9335.c | 2 + sound/soc/codecs/wm8904.c | 13 +- sound/soc/codecs/wm8985.c | 4 +- sound/soc/fsl/Kconfig | 2 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl-asoc-card.c | 2 +- sound/soc/fsl/fsl_asrc.c | 179 +- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 + sound/soc/fsl/fsl_asrc_m2m.c | 727 +++++ sound/soc/fsl/fsl_easrc.c | 261 +- sound/soc/fsl/fsl_easrc.h | 4 + sound/soc/fsl/fsl_micfil.c | 131 +- sound/soc/fsl/fsl_micfil.h | 2 +- sound/soc/fsl/fsl_mqs.c | 28 + sound/soc/fsl/fsl_sai.c | 7 +- sound/soc/fsl/fsl_sai.h | 3 + sound/soc/fsl/fsl_utils.c | 45 + sound/soc/fsl/fsl_utils.h | 5 + sound/soc/fsl/fsl_xcvr.c | 404 ++- sound/soc/fsl/fsl_xcvr.h | 13 + sound/soc/fsl/imx-audmux.c | 2 +- sound/soc/fsl/imx-card.c | 2 +- sound/soc/fsl/imx-rpmsg.c | 2 +- sound/soc/generic/audio-graph-card.c | 48 +- sound/soc/generic/audio-graph-card2.c | 262 +- sound/soc/generic/simple-card-utils.c | 79 +- sound/soc/generic/simple-card.c | 58 +- sound/soc/intel/avs/apl.c | 2 +- sound/soc/intel/avs/core.c | 24 +- sound/soc/intel/avs/debugfs.c | 1 + sound/soc/intel/avs/ipc.c | 25 +- sound/soc/intel/avs/loader.c | 36 +- sound/soc/intel/avs/messages.c | 22 +- sound/soc/intel/avs/messages.h | 3 +- sound/soc/intel/avs/pcm.c | 5 +- sound/soc/intel/avs/registers.h | 2 +- sound/soc/intel/avs/topology.c | 4 +- sound/soc/intel/avs/trace.h | 38 +- sound/soc/intel/boards/skl_hda_dsp_generic.c | 2 - sound/soc/intel/boards/sof_sdw.c | 33 +- sound/soc/intel/common/soc-acpi-intel-arl-match.c | 45 +- sound/soc/intel/common/soc-acpi-intel-lnl-match.c | 70 + sound/soc/intel/common/soc-acpi-intel-mtl-match.c | 289 +- sound/soc/intel/common/soc-acpi-intel-ptl-match.c | 148 + sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 194 ++ sound/soc/intel/keembay/kmb_platform.c | 2 +- sound/soc/mediatek/common/mtk-soundcard-driver.c | 4 +- sound/soc/mediatek/mt8192/mt8192-afe-pcm.c | 19 +- sound/soc/mediatek/mt8365/Makefile | 2 +- sound/soc/mediatek/mt8365/mt8365-mt6357.c | 11 +- sound/soc/qcom/common.c | 6 +- sound/soc/qcom/sc7180.c | 2 +- sound/soc/qcom/sdm845.c | 5 +- sound/soc/renesas/rz-ssi.c | 228 +- sound/soc/rockchip/rockchip_i2s_tdm.c | 28 - sound/soc/sdca/Makefile | 2 +- sound/soc/sdca/sdca_device.c | 2 + sound/soc/sdca/sdca_functions.c | 132 +- sound/soc/sdw_utils/soc_sdw_cs_amp.c | 46 + sound/soc/sdw_utils/soc_sdw_utils.c | 13 +- sound/soc/soc-card.c | 4 +- sound/soc/soc-core.c | 58 +- sound/soc/soc-dai.c | 27 +- sound/soc/soc-dapm.c | 14 +- sound/soc/soc-pcm.c | 32 +- sound/soc/soc-topology.c | 10 +- sound/soc/sof/intel/atom.c | 16 +- sound/soc/sof/intel/bdw.c | 16 +- sound/soc/sof/intel/hda-dai.c | 12 + sound/soc/sof/intel/hda-pcm.c | 15 + sound/soc/sof/intel/hda.c | 5 + sound/soc/sof/ipc4-topology.c | 2 +- sound/soc/sof/sof-audio.h | 1 + sound/soc/sof/sof-priv.h | 8 - sound/soc/sof/topology.c | 4 + sound/soc/sunxi/sun4i-codec.c | 409 ++- sound/soc/sunxi/sun4i-spdif.c | 24 +- sound/soc/xilinx/xlnx_spdif.c | 38 +- 170 files changed, 21739 insertions(+), 1441 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/realtek,rt5682.yaml delete mode 100644 Documentation/devicetree/bindings/sound/rt5682.txt create mode 100644 drivers/firmware/cirrus/test/Makefile create mode 100644 drivers/firmware/cirrus/test/cs_dsp_mock_bin.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_mock_mem_maps.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_mock_regmap.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_mock_utils.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_mock_wmfw.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_bin.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_bin_error.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_callbacks.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_control_cache.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_control_parse.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_control_rw.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_wmfw.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_test_wmfw_error.c create mode 100644 drivers/firmware/cirrus/test/cs_dsp_tests.c create mode 100644 include/linux/firmware/cirrus/cs_dsp_test_utils.h create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c

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[PATCH V2 7/7] soundwire: amd: add soundwire host wake interrupt enable/disable sequence
by Vijendar Mukunda 20 Jan '25

20 Jan '25

For wake event, SoundWire host wake interrupt will be asserted based on below pre-conditions for ACP7.0 & ACP7.1 platforms. - ACP device should be in D0 state. - SoundWire manager instance should be in D3 state. - SoundWire manager device state should be set to D3. - ACP_PME_EN should be set to 1. Implement code changes to enable/disable SoundWire host wake interrupt mask during suspend and resume as per design flow for ACP7.0 & ACP7.1 platforms. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 58 +++++++++++++++++++++++++++++++++ drivers/soundwire/amd_manager.h | 2 ++ 2 files changed, 60 insertions(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 03c9e15dce5b..161d963c80e6 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -166,6 +166,34 @@ static int amd_sdw_set_device_state(struct amd_sdw_manager *amd_manager, u32 tar return 0; } +static int amd_sdw_host_wake_enable(struct amd_sdw_manager *amd_manager, bool enable) +{ + u32 intr_cntl1; + u32 sdw_host_wake_irq_mask; + + if (!amd_manager->wake_en_mask) + return 0; + + switch (amd_manager->instance) { + case ACP_SDW0: + sdw_host_wake_irq_mask = AMD_SDW0_HOST_WAKE_INTR_MASK; + break; + case ACP_SDW1: + sdw_host_wake_irq_mask = AMD_SDW1_HOST_WAKE_INTR_MASK; + break; + default: + return -EINVAL; + } + + intr_cntl1 = readl(amd_manager->acp_mmio + ACP_EXTERNAL_INTR_CNTL(ACP_SDW1)); + if (enable) + intr_cntl1 |= sdw_host_wake_irq_mask; + else + intr_cntl1 &= ~sdw_host_wake_irq_mask; + writel(intr_cntl1, amd_manager->acp_mmio + ACP_EXTERNAL_INTR_CNTL(ACP_SDW1)); + return 0; +} + static void amd_sdw_ctl_word_prep(u32 *lower_word, u32 *upper_word, struct sdw_msg *msg, int cmd_offset) { @@ -1186,11 +1214,21 @@ static int __maybe_unused amd_suspend(struct device *dev) if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { amd_sdw_wake_enable(amd_manager, false); + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_host_wake_enable(amd_manager, false); + if (ret) + return ret; + } ret = amd_sdw_clock_stop(amd_manager); if (ret) return ret; } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { amd_sdw_wake_enable(amd_manager, false); + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_host_wake_enable(amd_manager, false); + if (ret) + return ret; + } /* * As per hardware programming sequence on AMD platforms, * clock stop should be invoked first before powering-off @@ -1224,11 +1262,21 @@ static int __maybe_unused amd_suspend_runtime(struct device *dev) } if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { amd_sdw_wake_enable(amd_manager, true); + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_host_wake_enable(amd_manager, true); + if (ret) + return ret; + } ret = amd_sdw_clock_stop(amd_manager); if (ret) return ret; } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { amd_sdw_wake_enable(amd_manager, true); + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_host_wake_enable(amd_manager, true); + if (ret) + return ret; + } ret = amd_sdw_clock_stop(amd_manager); if (ret) return ret; @@ -1269,8 +1317,18 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) ret = amd_sdw_clock_stop_exit(amd_manager); if (ret) return ret; + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_host_wake_enable(amd_manager, false); + if (ret) + return ret; + } } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { writel(0x00, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_host_wake_enable(amd_manager, false); + if (ret) + return ret; + } val = readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); if (val) { val |= AMD_SDW_CLK_RESUME_REQ; diff --git a/drivers/soundwire/amd_manager.h b/drivers/soundwire/amd_manager.h index 1d5e94371f81..6cc916b0c820 100644 --- a/drivers/soundwire/amd_manager.h +++ b/drivers/soundwire/amd_manager.h @@ -194,6 +194,8 @@ #define AMD_SDW_CLK_RESUME_DONE 3 #define AMD_SDW_WAKE_STAT_MASK BIT(16) #define AMD_SDW_WAKE_INTR_MASK BIT(16) +#define AMD_SDW0_HOST_WAKE_INTR_MASK BIT(22) +#define AMD_SDW1_HOST_WAKE_INTR_MASK BIT(23) #define AMD_SDW_DEVICE_STATE 0x1430 #define AMD_SDW0_DEVICE_STATE_MASK GENMASK(1, 0) #define AMD_SDW1_DEVICE_STATE_MASK GENMASK(3, 2) -- 2.34.1

1 0

[PATCH V2 6/7] soundwire: amd: set ACP_PME_EN during runtime suspend sequence
by Vijendar Mukunda 20 Jan '25

20 Jan '25

Set ACP_PME_EN to 1 during runtime suspend sequence as per design flow for ACP7.0 & ACP7.1 platforms. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 9 +++++++++ drivers/soundwire/amd_manager.h | 1 + 2 files changed, 10 insertions(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 42d553878e91..03c9e15dce5b 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -1215,6 +1215,7 @@ static int __maybe_unused amd_suspend_runtime(struct device *dev) struct amd_sdw_manager *amd_manager = dev_get_drvdata(dev); struct sdw_bus *bus = &amd_manager->bus; int ret; + u32 val; if (bus->prop.hw_disabled) { dev_dbg(bus->dev, "SoundWire manager %d is disabled,\n", @@ -1239,6 +1240,14 @@ static int __maybe_unused amd_suspend_runtime(struct device *dev) ret = amd_sdw_set_device_state(amd_manager, AMD_SDW_DEVICE_STATE_D3); if (ret) return ret; + if (amd_manager->wake_en_mask) { + val = readl(amd_manager->acp_mmio + ACP_PME_EN); + if (!val) { + writel(1, amd_manager->acp_mmio + ACP_PME_EN); + val = readl(amd_manager->acp_mmio + ACP_PME_EN); + dev_dbg(amd_manager->dev, "ACP_PME_EN:0x%x\n", val); + } + } } return 0; } diff --git a/drivers/soundwire/amd_manager.h b/drivers/soundwire/amd_manager.h index 8430f279d88e..1d5e94371f81 100644 --- a/drivers/soundwire/amd_manager.h +++ b/drivers/soundwire/amd_manager.h @@ -199,6 +199,7 @@ #define AMD_SDW1_DEVICE_STATE_MASK GENMASK(3, 2) #define AMD_SDW_DEVICE_STATE_D0 0 #define AMD_SDW_DEVICE_STATE_D3 3 +#define ACP_PME_EN 0x0001400 static u32 amd_sdw_freq_tbl[AMD_SDW_MAX_FREQ_NUM] = { AMD_SDW_DEFAULT_CLK_FREQ, -- 2.34.1

1 0

[PATCH V2 5/7] soundwire: amd: set device power state during suspend/resume sequence
by Vijendar Mukunda 20 Jan '25

20 Jan '25

Set SoundWire manager device power state during suspend and resume sequence for ACP7.0 & ACP7.1 platforms. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 58 ++++++++++++++++++++++++++++++--- drivers/soundwire/amd_manager.h | 5 +++ 2 files changed, 58 insertions(+), 5 deletions(-) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index d611498d5488..42d553878e91 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -143,6 +143,29 @@ static void amd_sdw_wake_enable(struct amd_sdw_manager *amd_manager, bool enable writel(wake_ctrl, amd_manager->mmio + ACP_SW_STATE_CHANGE_STATUS_MASK_8TO11); } +static int amd_sdw_set_device_state(struct amd_sdw_manager *amd_manager, u32 target_device_state) +{ + u32 sdw_dev_state; + + sdw_dev_state = readl(amd_manager->acp_mmio + AMD_SDW_DEVICE_STATE); + switch (amd_manager->instance) { + case ACP_SDW0: + u32p_replace_bits(&sdw_dev_state, target_device_state, + AMD_SDW0_DEVICE_STATE_MASK); + break; + case ACP_SDW1: + u32p_replace_bits(&sdw_dev_state, target_device_state, + AMD_SDW1_DEVICE_STATE_MASK); + break; + default: + return -EINVAL; + } + writel(sdw_dev_state, amd_manager->acp_mmio + AMD_SDW_DEVICE_STATE); + sdw_dev_state = readl(amd_manager->acp_mmio + AMD_SDW_DEVICE_STATE); + dev_dbg(amd_manager->dev, "AMD_SDW_DEVICE_STATE:0x%x\n", sdw_dev_state); + return 0; +} + static void amd_sdw_ctl_word_prep(u32 *lower_word, u32 *upper_word, struct sdw_msg *msg, int cmd_offset) { @@ -1163,7 +1186,9 @@ static int __maybe_unused amd_suspend(struct device *dev) if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { amd_sdw_wake_enable(amd_manager, false); - return amd_sdw_clock_stop(amd_manager); + ret = amd_sdw_clock_stop(amd_manager); + if (ret) + return ret; } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { amd_sdw_wake_enable(amd_manager, false); /* @@ -1173,7 +1198,14 @@ static int __maybe_unused amd_suspend(struct device *dev) ret = amd_sdw_clock_stop(amd_manager); if (ret) return ret; - return amd_deinit_sdw_manager(amd_manager); + ret = amd_deinit_sdw_manager(amd_manager); + if (ret) + return ret; + } + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_set_device_state(amd_manager, AMD_SDW_DEVICE_STATE_D3); + if (ret) + return ret; } return 0; } @@ -1191,13 +1223,22 @@ static int __maybe_unused amd_suspend_runtime(struct device *dev) } if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { amd_sdw_wake_enable(amd_manager, true); - return amd_sdw_clock_stop(amd_manager); + ret = amd_sdw_clock_stop(amd_manager); + if (ret) + return ret; } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { amd_sdw_wake_enable(amd_manager, true); ret = amd_sdw_clock_stop(amd_manager); if (ret) return ret; - return amd_deinit_sdw_manager(amd_manager); + ret = amd_deinit_sdw_manager(amd_manager); + if (ret) + return ret; + } + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_set_device_state(amd_manager, AMD_SDW_DEVICE_STATE_D3); + if (ret) + return ret; } return 0; } @@ -1216,7 +1257,9 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) } if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { - return amd_sdw_clock_stop_exit(amd_manager); + ret = amd_sdw_clock_stop_exit(amd_manager); + if (ret) + return ret; } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { writel(0x00, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); val = readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); @@ -1239,6 +1282,11 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) return ret; amd_sdw_set_frameshape(amd_manager); } + if (amd_manager->acp_rev >= ACP70_PCI_REV_ID) { + ret = amd_sdw_set_device_state(amd_manager, AMD_SDW_DEVICE_STATE_D0); + if (ret) + return ret; + } return 0; } diff --git a/drivers/soundwire/amd_manager.h b/drivers/soundwire/amd_manager.h index 30244a31c21c..8430f279d88e 100644 --- a/drivers/soundwire/amd_manager.h +++ b/drivers/soundwire/amd_manager.h @@ -194,6 +194,11 @@ #define AMD_SDW_CLK_RESUME_DONE 3 #define AMD_SDW_WAKE_STAT_MASK BIT(16) #define AMD_SDW_WAKE_INTR_MASK BIT(16) +#define AMD_SDW_DEVICE_STATE 0x1430 +#define AMD_SDW0_DEVICE_STATE_MASK GENMASK(1, 0) +#define AMD_SDW1_DEVICE_STATE_MASK GENMASK(3, 2) +#define AMD_SDW_DEVICE_STATE_D0 0 +#define AMD_SDW_DEVICE_STATE_D3 3 static u32 amd_sdw_freq_tbl[AMD_SDW_MAX_FREQ_NUM] = { AMD_SDW_DEFAULT_CLK_FREQ, -- 2.34.1

1 0

[PATCH V2 4/7] soundwire: amd: add support for ACP7.0 & ACP7.1 platforms
by Vijendar Mukunda 20 Jan '25

20 Jan '25

Add SoundWire support for ACP7.0 and ACP7.1 platforms. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 22 ++++++++++++++++++++++ drivers/soundwire/amd_manager.h | 18 ++++++++++++++++++ include/linux/soundwire/sdw_amd.h | 2 ++ 3 files changed, 42 insertions(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 6831b3729db5..d611498d5488 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -446,6 +446,10 @@ static int amd_sdw_port_params(struct sdw_bus *bus, struct sdw_port_params *p_pa return -EINVAL; } break; + case ACP70_PCI_REV_ID: + case ACP71_PCI_REV_ID: + frame_fmt_reg = acp70_sdw_dp_reg[p_params->num].frame_fmt_reg; + break; default: return -EINVAL; } @@ -494,6 +498,14 @@ static int amd_sdw_transport_params(struct sdw_bus *bus, return -EINVAL; } break; + case ACP70_PCI_REV_ID: + case ACP71_PCI_REV_ID: + frame_fmt_reg = acp70_sdw_dp_reg[params->port_num].frame_fmt_reg; + sample_int_reg = acp70_sdw_dp_reg[params->port_num].sample_int_reg; + hctrl_dp0_reg = acp70_sdw_dp_reg[params->port_num].hctrl_dp0_reg; + offset_reg = acp70_sdw_dp_reg[params->port_num].offset_reg; + lane_ctrl_ch_en_reg = acp70_sdw_dp_reg[params->port_num].lane_ctrl_ch_en_reg; + break; default: return -EINVAL; } @@ -549,6 +561,10 @@ static int amd_sdw_port_enable(struct sdw_bus *bus, return -EINVAL; } break; + case ACP70_PCI_REV_ID: + case ACP71_PCI_REV_ID: + lane_ctrl_ch_en_reg = acp70_sdw_dp_reg[enable_ch->port_num].lane_ctrl_ch_en_reg; + break; default: return -EINVAL; } @@ -898,6 +914,7 @@ int amd_sdw_manager_start(struct amd_sdw_manager *amd_manager) ret = amd_enable_sdw_manager(amd_manager); if (ret) return ret; + amd_sdw_set_frameshape(amd_manager); } /* Enable runtime PM */ @@ -969,6 +986,11 @@ static int amd_sdw_manager_probe(struct platform_device *pdev) return -EINVAL; } break; + case ACP70_PCI_REV_ID: + case ACP71_PCI_REV_ID: + amd_manager->num_dout_ports = AMD_ACP70_SDW_MAX_TX_PORTS; + amd_manager->num_din_ports = AMD_ACP70_SDW_MAX_RX_PORTS; + break; default: return -EINVAL; } diff --git a/drivers/soundwire/amd_manager.h b/drivers/soundwire/amd_manager.h index cc2170e4521e..30244a31c21c 100644 --- a/drivers/soundwire/amd_manager.h +++ b/drivers/soundwire/amd_manager.h @@ -159,8 +159,11 @@ #define AMD_ACP63_SDW0_MAX_RX_PORTS 3 #define AMD_ACP63_SDW1_MAX_TX_PORTS 1 #define AMD_ACP63_SDW1_MAX_RX_PORTS 1 +#define AMD_ACP70_SDW_MAX_TX_PORTS 3 +#define AMD_ACP70_SDW_MAX_RX_PORTS 3 #define AMD_ACP63_SDW0_MAX_DAI 6 #define AMD_ACP63_SDW1_MAX_DAI 2 +#define AMD_ACP70_SDW_MAX_DAI 6 #define AMD_SDW_SLAVE_0_ATTACHED 5 #define AMD_SDW_SSP_COUNTER_VAL 3 @@ -244,6 +247,21 @@ static struct sdw_manager_dp_reg acp63_sdw1_dp_reg[AMD_ACP63_SDW1_MAX_DAI] = { ACP_SW_AUDIO1_RX_OFFSET, ACP_SW_AUDIO1_RX_CHANNEL_ENABLE_DP0} }; +static struct sdw_manager_dp_reg acp70_sdw_dp_reg[AMD_ACP70_SDW_MAX_DAI] = { + {ACP_SW_AUDIO0_TX_FRAME_FORMAT, ACP_SW_AUDIO0_TX_SAMPLEINTERVAL, ACP_SW_AUDIO0_TX_HCTRL_DP0, + ACP_SW_AUDIO0_TX_OFFSET_DP0, ACP_SW_AUDIO0_TX_CHANNEL_ENABLE_DP0}, + {ACP_SW_AUDIO1_TX_FRAME_FORMAT, ACP_SW_AUDIO1_TX_SAMPLEINTERVAL, ACP_SW_AUDIO1_TX_HCTRL, + ACP_SW_AUDIO1_TX_OFFSET, ACP_SW_AUDIO1_TX_CHANNEL_ENABLE_DP0}, + {ACP_SW_AUDIO2_TX_FRAME_FORMAT, ACP_SW_AUDIO2_TX_SAMPLEINTERVAL, ACP_SW_AUDIO2_TX_HCTRL, + ACP_SW_AUDIO2_TX_OFFSET, ACP_SW_AUDIO2_TX_CHANNEL_ENABLE_DP0}, + {ACP_SW_AUDIO0_RX_FRAME_FORMAT, ACP_SW_AUDIO0_RX_SAMPLEINTERVAL, ACP_SW_AUDIO0_RX_HCTRL_DP0, + ACP_SW_AUDIO0_RX_OFFSET_DP0, ACP_SW_AUDIO0_RX_CHANNEL_ENABLE_DP0}, + {ACP_SW_AUDIO1_RX_FRAME_FORMAT, ACP_SW_AUDIO1_RX_SAMPLEINTERVAL, ACP_SW_AUDIO1_RX_HCTRL, + ACP_SW_AUDIO1_RX_OFFSET, ACP_SW_AUDIO1_RX_CHANNEL_ENABLE_DP0}, + {ACP_SW_AUDIO2_RX_FRAME_FORMAT, ACP_SW_AUDIO2_RX_SAMPLEINTERVAL, ACP_SW_AUDIO2_RX_HCTRL, + ACP_SW_AUDIO2_RX_OFFSET, ACP_SW_AUDIO2_RX_CHANNEL_ENABLE_DP0}, +}; + static u32 sdw_manager_reg_mask_array[AMD_SDW_MAX_MANAGER_COUNT] = { AMD_SDW0_EXT_INTR_MASK, AMD_SDW1_EXT_INTR_MASK diff --git a/include/linux/soundwire/sdw_amd.h b/include/linux/soundwire/sdw_amd.h index 799f8578137b..6b839987f14c 100644 --- a/include/linux/soundwire/sdw_amd.h +++ b/include/linux/soundwire/sdw_amd.h @@ -28,6 +28,8 @@ #define ACP_SDW1 1 #define AMD_SDW_MAX_MANAGER_COUNT 2 #define ACP63_PCI_REV_ID 0x63 +#define ACP70_PCI_REV_ID 0x70 +#define ACP71_PCI_REV_ID 0x71 struct acp_sdw_pdata { u16 instance; -- 2.34.1

1 0

[PATCH V2 2/7] soundwire: amd: add debug log for soundwire wake event
by Vijendar Mukunda 20 Jan '25

20 Jan '25

Add debug log in amd_sdw_process_wake_event() function. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 187006e68ad6..60be5805715e 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -849,6 +849,7 @@ static void amd_sdw_update_slave_status(u32 status_change_0to7, u32 status_chang static void amd_sdw_process_wake_event(struct amd_sdw_manager *amd_manager) { + dev_dbg(amd_manager->dev, "SoundWire Wake event reported\n"); pm_request_resume(amd_manager->dev); writel(0x00, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); writel(0x00, amd_manager->mmio + ACP_SW_STATE_CHANGE_STATUS_8TO11); -- 2.34.1

1 0

[PATCH V2 1/7] soundwire: amd: change the soundwire wake enable/disable sequence
by Vijendar Mukunda 20 Jan '25

20 Jan '25

During runtime suspend scenario, SoundWire wake should be enabled and during system level suspend scenario SoundWire wake should be disabled. Implement the SoundWire wake enable/disable sequence as per design flow for SoundWire poweroff mode. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 71b3cf2df89d..187006e68ad6 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -1139,6 +1139,7 @@ static int __maybe_unused amd_suspend(struct device *dev) amd_sdw_wake_enable(amd_manager, false); return amd_sdw_clock_stop(amd_manager); } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { + amd_sdw_wake_enable(amd_manager, false); /* * As per hardware programming sequence on AMD platforms, * clock stop should be invoked first before powering-off @@ -1166,6 +1167,7 @@ static int __maybe_unused amd_suspend_runtime(struct device *dev) amd_sdw_wake_enable(amd_manager, true); return amd_sdw_clock_stop(amd_manager); } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { + amd_sdw_wake_enable(amd_manager, true); ret = amd_sdw_clock_stop(amd_manager); if (ret) return ret; -- 2.34.1

1 0

[PATCH v1 1/3] ASoC: tas2781: Support dsp firmware Alpha and Beta seaies
by Shenghao Ding 20 Jan '25

20 Jan '25

For calibration, basic version does not contain any calibration addresses, it depends on calibration tool to convery the addresses to the driver. Since Alpha and Beta firmware, all the calibration addresses are saved into the firmware. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- v1: - update the year from 2024 to 2025. - Add dspbin_type to distinguish different dspbin types, such as TASDEV_BASIC, TASDEV_ALPHA, and TASDEV_BETA. - Add struct bulk_reg_val alp_cali_bckp to store the value of pilot tone threshold only for TASDEV_ALPHA & TASDEV_BETA firmware. - Add the declaration of (*fw_parse_fct_param_address)() to parse the addresses of calibration params. - Add the minimum version of TASDEV_ALPHA and TASDEV_BETA firmware. - Add the definition of struct fct_param_address. --- include/sound/tas2781-dsp.h | 26 +++++++++++++++++++++++--- include/sound/tas2781.h | 13 ++++++++++++- 2 files changed, 35 insertions(+), 4 deletions(-) diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h index 3cda9da14f6d..77a240daaae0 100644 --- a/include/sound/tas2781-dsp.h +++ b/include/sound/tas2781-dsp.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2781 driver implements a flexible and configurable @@ -30,8 +30,10 @@ #define PRE_DEVICE_C 0x12 #define PRE_DEVICE_D 0x16 -#define PPC3_VERSION 0x4100 -#define PPC3_VERSION_TAS2781 0x14600 +#define PPC3_VERSION_BASE 0x4100 +#define PPC3_VERSION_TAS2781_BASIC_MIN 0x14600 +#define PPC3_VERSION_TAS2781_ALPHA_MIN 0x4a00 +#define PPC3_VERSION_TAS2781_BETA_MIN 0x19400 #define TASDEVICE_DEVICE_SUM 8 #define TASDEVICE_CONFIG_SUM 64 @@ -106,6 +108,23 @@ struct tasdevice_calibration { struct tasdevice_data dev_data; }; +struct fct_param_address { + unsigned char thr[3]; + unsigned char thr2[3]; + unsigned char plt_flg[3]; + unsigned char sin_gn[3]; + unsigned char sin_gn2[3]; + /* real-time spk impedance */ + unsigned char r0_reg[3]; + /* check spk connection */ + unsigned char tf_reg[3]; + /* check spk resonant frequency */ + unsigned char a1_reg[3]; + unsigned char a2_reg[3]; + unsigned short size; + char *data; +}; + struct tasdevice_fw { struct tasdevice_dspfw_hdr fw_hdr; unsigned short nr_programs; @@ -114,6 +133,7 @@ struct tasdevice_fw { struct tasdevice_config *configs; unsigned short nr_calibrations; struct tasdevice_calibration *calibrations; + struct fct_param_address fct_par_addr; struct device *dev; }; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 72d2060904f6..5cfbb9e0a626 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -2,7 +2,7 @@ // // ALSA SoC Texas Instruments TAS2563/TAS2781 Audio Smart Amplifier // -// Copyright (C) 2022 - 2024 Texas Instruments Incorporated +// Copyright (C) 2022 - 2025 Texas Instruments Incorporated // https://www.ti.com // // The TAS2563/TAS2781 driver implements a flexible and configurable @@ -112,6 +112,12 @@ enum audio_device { TAS2781, }; +enum dspbin_type { + TASDEV_BASIC, + TASDEV_ALPHA, + TASDEV_BETA, +}; + enum device_catlog_id { LENOVO = 0, OTHERS @@ -126,6 +132,7 @@ struct bulk_reg_val { struct tasdevice { struct bulk_reg_val *cali_data_backup; + struct bulk_reg_val alp_cali_bckp; struct tasdevice_fw *cali_data_fmw; unsigned int dev_addr; unsigned int err_code; @@ -171,6 +178,7 @@ struct tasdevice_priv { unsigned char dev_name[32]; const char *name_prefix; unsigned char ndev; + unsigned int dspbin_typ; unsigned int magic_num; unsigned int chip_id; unsigned int sysclk; @@ -196,6 +204,9 @@ struct tasdevice_priv { int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, struct tasdevice_fw *tas_fmw, const struct firmware *fmw, int offset); + int (*fw_parse_fct_param_address)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, struct tasdev_blk *block); -- 2.34.1

1 2

Is function delay of the pulse plugin correct?
by Pavel Hofman 19 Jan '25

19 Jan '25

Hi, Please correct me if I am wrong: "avail" means (buffer - buffer_fill), while "delay" means (buffer_fill + device/downstream delay/latency). Some drivers/devices do not track the device delay, and then delay = buffer_fill. While troubleshooting some playback code, I hit that snd_pcm_delay on the plugin pulse (device pulse) was always returning 0 frames, while snd_pcm_avail was returning dynamic (correct) values. IIUC while pulse_pcm.c uses generic code for avail, it overrides the generic snd_pcm_ioplug_delay https://github.com/alsa-project/alsa-lib/blob/35d2efefa95d28e0cd963a06e051f… with specific pulse_delay https://github.com/alsa-project/alsa-plugins/blob/52574cb5ccbb8b546df2759e4… . IIUC this method returns only delay obtained from Pulseaudio, i.e. no fill of the pulse plugin buffer is included. And IIUC in this specific case pulseaudio keeps returning zero for the pa_stream_get_latency call, making snd_pcm_delay return zero too. Therefore the equation (buffer size = avail + delay - driver/device delay) does not seem to hold for the pulse plugin. The new pipewire plugin does not seem to override the generic snd_pcm_ioplug_delay method https://github.com/PipeWire/pipewire-alsa/blob/master/alsa-plugins/pcm_pipe… . Please, is that behavior correct? With regards, Pavel.

1 0

[tiwai-sound:for-next 50/52] sound/usb/fcp.c:889:15: error: incompatible function pointer types assigning to 'long (*)(struct snd_hwdep *, char *, long, loff_t *)' (aka 'long (*)(struct snd_hwdep *, char *, long, long long *)') from 'ssize_t (struct snd_hwdep *, char *, ssize_t, loff...
by kernel test robot 19 Jan '25

19 Jan '25

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: dd85bbb7fc482b5cdb78f8b5e8f9ef7a3919d6ef commit: 46757a3e7d50dac923888e7fbe68377736f13c70 [50/52] ALSA: FCP: Add Focusrite Control Protocol driver config: hexagon-allmodconfig (https://download.01.org/0day-ci/archive/20250119/202501192122.ngosycnT-lkp@…) compiler: clang version 20.0.0git (https://github.com/llvm/llvm-project c23f2417dc5f6dc371afb07af5627ec2a9d373a0) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20250119/202501192122.ngosycnT-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/202501192122.ngosycnT-lkp@intel.com/ All errors (new ones prefixed by >>): >> sound/usb/fcp.c:889:15: error: incompatible function pointer types assigning to 'long (*)(struct snd_hwdep *, char *, long, loff_t *)' (aka 'long (*)(struct snd_hwdep *, char *, long, long long *)') from 'ssize_t (struct snd_hwdep *, char *, ssize_t, loff_t *)' (aka 'int (struct snd_hwdep *, char *, int, long long *)') [-Wincompatible-function-pointer-types] 889 | hw->ops.read = fcp_hwdep_read; | ^ ~~~~~~~~~~~~~~ 1 error generated. vim +889 sound/usb/fcp.c 874 875 static int fcp_hwdep_init(struct usb_mixer_interface *mixer) 876 { 877 struct snd_hwdep *hw; 878 int err; 879 880 err = snd_hwdep_new(mixer->chip->card, "Focusrite Control", 0, &hw); 881 if (err < 0) 882 return err; 883 884 hw->private_data = mixer; 885 hw->exclusive = 1; 886 hw->ops.open = fcp_hwdep_open; 887 hw->ops.ioctl = fcp_hwdep_ioctl; 888 hw->ops.ioctl_compat = fcp_hwdep_ioctl; > 889 hw->ops.read = fcp_hwdep_read; 890 hw->ops.poll = fcp_hwdep_poll; 891 hw->ops.release = fcp_hwdep_release; 892 893 return 0; 894 } 895 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:for-next 49/52] sound/pci/hda/tas2781_hda_spi.c:110:6: warning: variable 'ret' is used uninitialized whenever 'if' condition is false
by kernel test robot 19 Jan '25

19 Jan '25

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: dd85bbb7fc482b5cdb78f8b5e8f9ef7a3919d6ef commit: bb5f86ea50ffb292f42eb1ebdb99991d5c5ac3ba [49/52] ALSA: hda/tas2781: Add tas2781 hda SPI driver config: arm64-allmodconfig (https://download.01.org/0day-ci/archive/20250119/202501192006.Hm9GmKiV-lkp@…) compiler: clang version 18.1.8 (https://github.com/llvm/llvm-project 3b5b5c1ec4a3095ab096dd780e84d7ab81f3d7ff) reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20250119/202501192006.Hm9GmKiV-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/202501192006.Hm9GmKiV-lkp@intel.com/ All warnings (new ones prefixed by >>): >> sound/pci/hda/tas2781_hda_spi.c:110:6: warning: variable 'ret' is used uninitialized whenever 'if' condition is false [-Wsometimes-uninitialized] 110 | if (tas_priv->cur_book != TASDEVICE_BOOK_ID(reg)) { | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/tas2781_hda_spi.c:119:9: note: uninitialized use occurs here 119 | return ret; | ^~~ sound/pci/hda/tas2781_hda_spi.c:110:2: note: remove the 'if' if its condition is always true 110 | if (tas_priv->cur_book != TASDEVICE_BOOK_ID(reg)) { | ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ sound/pci/hda/tas2781_hda_spi.c:108:9: note: initialize the variable 'ret' to silence this warning 108 | int ret; | ^ | = 0 1 warning generated. vim +110 sound/pci/hda/tas2781_hda_spi.c 104 105 static int tasdevice_spi_switch_book(struct tasdevice_priv *tas_priv, int reg) 106 { 107 struct regmap *map = tas_priv->regmap; 108 int ret; 109 > 110 if (tas_priv->cur_book != TASDEVICE_BOOK_ID(reg)) { 111 ret = regmap_write(map, TASDEVICE_BOOKCTL_REG, 112 TASDEVICE_BOOK_ID(reg)); 113 if (ret < 0) { 114 dev_err(tas_priv->dev, "Switch Book E=%d\n", ret); 115 return ret; 116 } 117 tas_priv->cur_book = TASDEVICE_BOOK_ID(reg); 118 } 119 return ret; 120 } 121 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

[tiwai-sound:for-next 50/51] sound/usb/fcp.c:889:22: error: assignment to 'long int (*)(struct snd_hwdep *, char *, long int, loff_t *)' {aka 'long int (*)(struct snd_hwdep *, char *, long int, long long int *)'} from incompatible pointer type 'ssize_t (*)(struct snd_hwdep *, char ...
by kernel test robot 18 Jan '25

18 Jan '25

tree: https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git for-next head: 0ce204d3af3beca1825018e9ca128635ccc8aa85 commit: 46757a3e7d50dac923888e7fbe68377736f13c70 [50/51] ALSA: FCP: Add Focusrite Control Protocol driver config: i386-buildonly-randconfig-004-20250119 (https://download.01.org/0day-ci/archive/20250119/202501190536.A2CiOP0a-lkp@…) compiler: gcc-12 (Debian 12.2.0-14) 12.2.0 reproduce (this is a W=1 build): (https://download.01.org/0day-ci/archive/20250119/202501190536.A2CiOP0a-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/202501190536.A2CiOP0a-lkp@intel.com/ All errors (new ones prefixed by >>): sound/usb/fcp.c: In function 'fcp_hwdep_init': >> sound/usb/fcp.c:889:22: error: assignment to 'long int (*)(struct snd_hwdep *, char *, long int, loff_t *)' {aka 'long int (*)(struct snd_hwdep *, char *, long int, long long int *)'} from incompatible pointer type 'ssize_t (*)(struct snd_hwdep *, char *, ssize_t, loff_t *)' {aka 'int (*)(struct snd_hwdep *, char *, int, long long int *)'} [-Werror=incompatible-pointer-types] 889 | hw->ops.read = fcp_hwdep_read; | ^ cc1: some warnings being treated as errors vim +889 sound/usb/fcp.c 874 875 static int fcp_hwdep_init(struct usb_mixer_interface *mixer) 876 { 877 struct snd_hwdep *hw; 878 int err; 879 880 err = snd_hwdep_new(mixer->chip->card, "Focusrite Control", 0, &hw); 881 if (err < 0) 882 return err; 883 884 hw->private_data = mixer; 885 hw->exclusive = 1; 886 hw->ops.open = fcp_hwdep_open; 887 hw->ops.ioctl = fcp_hwdep_ioctl; 888 hw->ops.ioctl_compat = fcp_hwdep_ioctl; > 889 hw->ops.read = fcp_hwdep_read; 890 hw->ops.poll = fcp_hwdep_poll; 891 hw->ops.release = fcp_hwdep_release; 892 893 return 0; 894 } 895 -- 0-DAY CI Kernel Test Service https://github.com/intel/lkp-tests/wiki

1 0

No audio on Acer Swift SF315-52
by GitHub issues - opened 18 Jan '25

18 Jan '25

alsa-project/alsa-ucm-conf issue #497 was opened from pavethem: [Alsa info](http://alsa-project.org/db/?f=8e6dc59510d7aaa42899f81a8f29a740032e9ce… Ever since kernel 6.12 dropped for Fedora, my laptop only shows a dummy output. This is probably due to the kernel removing legacy sound drivers and replacing them with avs, which doesn’t seem work for the kaby lake architecture present in this model. ([phoronix](https://www.phoronix.com/news/Linux-6.12-Sound-Drivers)). I am unsure whether or not I am filing this bug in the right place. Since I am able to get the microphone to work by copying a file: `cp /usr/share/alsa/ucm2/conf.d/avs_dmic/Acer-Lars-1.0.conf /usr/share/alsa/ucm2/conf.d/avs_dmic/avs_dmic.conf` I am guessing this is a case of a missing ucm file for this particular hardware. If this is incorrect, I would appreciate if someone could point me in the right direction and tell me if this is rather a bug with alsa-firmware or the kernel itself. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/497 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[GIT PULL] ASoC fixes for v6.13-rc7
by Mark Brown 18 Jan '25

18 Jan '25

The following changes since commit 139fa599cea0fd9d38e00246ea9f79af6c59acbd: ASoC: rsnd: check rsnd_adg_clk_enable() return value (2025-01-09 13:17:02 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.13-rc7 for you to fetch changes up to 75a7f7d6e7e10b9052d287be37f965103e1abae1: ASoC: Intel: sof_sdw: Fix DMI match entries for a (2025-01-15 19:50:01 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.13 A few final driver specific fixes - a couple of x86 ID field changes, plus bug fixes for simple-card-utils and nau8824. ---------------------------------------------------------------- Laurentiu Mihalcea (1): ASoC: simple-card-utils: fix priv->dai_props indexing Mark Brown (1): ASoC: Intel: sof_sdw: Fix DMI match entries for a Maxim Kochetkov (1): ASoC: codecs: nau8824: fix max volume for Speaker output Simon Trimmer (2): ASoC: Intel: sof_sdw: Fix DMI match for Lenovo 83LC ASoC: Intel: sof_sdw: Fix DMI match for Lenovo 83JX, 83MC and 83NM include/sound/simple_card_utils.h | 7 +++++++ sound/soc/codecs/nau8824.c | 8 ++++---- sound/soc/generic/simple-card-utils.c | 10 +++++----- sound/soc/intel/boards/sof_sdw.c | 23 +++++++++++++++++++---- 4 files changed, 35 insertions(+), 13 deletions(-)

2 1

RE: [REGRESSION] Distorted sound on Acer Aspire A115-31 laptop
by Kailang 17 Jan '25

17 Jan '25

Attached. > -----Original Message----- > From: Takashi Iwai <tiwai(a)suse.de> > Sent: Thursday, January 16, 2025 10:27 PM > To: Evgeny Kapun <abacabadabacaba(a)gmail.com> > Cc: Kailang <kailang(a)realtek.com>; Takashi Iwai <tiwai(a)suse.de>; Linux > Sound Mailing List <linux-sound(a)vger.kernel.org>; Linux Kernel Mailing List > <linux-kernel(a)vger.kernel.org>; Linux Regressions Mailing List > <regressions(a)lists.linux.dev>; Linux Stable Mailing List > <stable(a)vger.kernel.org> > Subject: Re: [REGRESSION] Distorted sound on Acer Aspire A115-31 laptop > > > External mail. > > > > On Sat, 11 Jan 2025 16:00:33 +0100, > Evgeny Kapun wrote: > > > > On 12/24/24 04:54, Kailang wrote: > > > Please test attach patch. > > > > This patch, when applied to kernel version 6.12.8, appears to fix the > > issue. There are no distortions, and the left and the right channel > > can be controlled independently. > > Good to hear. > > Kailang, care to submit a proper patch for merging? > > > thanks, > > Takashi

2 1

Topology: NHLT: Intel: Fix DMA slots config in SSP blob
by GitHub pull_request - opened 16 Jan '25

16 Jan '25

alsa-project/alsa-utils pull request #289 was opened from singalsu: This fixes a mistake in function ssp_calculate_intern_v30(). The rx_dir[0] and tx_dir[0] value was hard-coded to 0x3 for typical stereo slots configuration. It should be instead retrieved from topology parameters for the SSP. Request URL : https://github.com/alsa-project/alsa-utils/pull/289 Patch URL : https://github.com/alsa-project/alsa-utils/pull/289.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

no internal sound with snd_hda_intel, ALC256/ Lenovo ThinkCentre M90a Pro Gen4
by Ronny Sattler 15 Jan '25

15 Jan '25

Hello, I have a problem with the snd-hda-intel. There is no sound. Only when I plug in headphones do I hear the sound on the headphones. The speakers are onboard (All in One - PC) and always remain mute. I suspect it is because the amplifier is not activated or a PIN assignment is incorrect. Please let me know if there is anything else I can do for you on the PC. Attached are previous log files and general information. Thank you very much for your work :-) Best regards Ronny

1 0

[RFC PATCH] soc: audio-graph-card2: use correct endpoint when getting link parameters
by Ivaylo Dimitrov 15 Jan '25

15 Jan '25

We may have multiple links between ports, with each link having different parameters. Currently, no matter the topology, it is always port endpoint 0 that is used when setting parameters. On a complex sound system, like the one found on Motorola droid4, hifi and voice DAIs require differents formats (i2s vs dsp_a) and curently it is impossible to use DT to set that. Implementing the change leads to partially dropping of at least 0dedbde5062d (ASoC: cpcap: Implement set_tdm_slot for voice call support), as core does most of what is needed to configure voice DAI. We (on Maemo Leste ) use the patch (along with few others) to have voice calls working properly on d4 through UCM. The patch is for linux 6.6, I want to know whether the approach would be accepted before sending a proper patch for current master. the original commit message follows: When link parameters are parsed, it is always endpoint@0 that is used and parameters set to other endpoints are ignored. Fix that by using endpoint that is set in DT when parsing link parameters. Signed-off-by: Ivaylo Dimitrov <ivo.g.dimitrov.75(a)gmail.com> --- sound/soc/generic/audio-graph-card2.c | 59 +++++++++++++-------------- 1 file changed, 28 insertions(+), 31 deletions(-) diff --git a/sound/soc/generic/audio-graph-card2.c b/sound/soc/generic/audio-graph-card2.c index b1c675c6b6db..163a20c8ffee 100644 --- a/sound/soc/generic/audio-graph-card2.c +++ b/sound/soc/generic/audio-graph-card2.c @@ -508,17 +508,16 @@ static int __graph_parse_node(struct asoc_simple_priv *priv, static int graph_parse_node(struct asoc_simple_priv *priv, enum graph_type gtype, - struct device_node *port, + struct device_node *ep, struct link_info *li, int is_cpu) { - struct device_node *ep; int ret = 0; + struct device_node *port = of_get_parent(ep); + bool is_multi = graph_lnk_is_multi(port); - if (graph_lnk_is_multi(port)) { + if (is_multi) { int idx; - of_node_get(port); - for (idx = 0;; idx++) { ep = graph_get_next_multi_ep(&port); if (!ep) @@ -532,9 +531,8 @@ static int graph_parse_node(struct asoc_simple_priv *priv, } } else { /* Single CPU / Codec */ - ep = port_to_endpoint(port); + of_node_put(port); ret = __graph_parse_node(priv, gtype, ep, li, is_cpu, 0); - of_node_put(ep); } return ret; @@ -591,22 +589,20 @@ static void graph_parse_daifmt(struct device_node *node, } static void graph_link_init(struct asoc_simple_priv *priv, - struct device_node *port, + struct device_node *ep, struct link_info *li, int is_cpu_node) { struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); - struct device_node *ep; + struct device_node *port = of_get_parent(ep); + bool is_multi = graph_lnk_is_multi(port); struct device_node *ports; unsigned int daifmt = 0, daiclk = 0; unsigned int bit_frame = 0; - if (graph_lnk_is_multi(port)) { - of_node_get(port); + if (is_multi) { ep = graph_get_next_multi_ep(&port); port = of_get_parent(ep); - } else { - ep = port_to_endpoint(port); } ports = of_get_parent(port); @@ -642,6 +638,9 @@ static void graph_link_init(struct asoc_simple_priv *priv, dai_link->ops = &graph_ops; if (priv->ops) dai_link->ops = priv->ops; + + of_node_put(port); + of_node_put(ports); } int audio_graph2_link_normal(struct asoc_simple_priv *priv, @@ -650,7 +649,7 @@ int audio_graph2_link_normal(struct asoc_simple_priv *priv, { struct device_node *cpu_port = lnk; struct device_node *cpu_ep = port_to_endpoint(cpu_port); - struct device_node *codec_port = of_graph_get_remote_port(cpu_ep); + struct device_node *codec_ep = of_graph_get_remote_endpoint(cpu_ep); int ret; /* @@ -658,20 +657,20 @@ int audio_graph2_link_normal(struct asoc_simple_priv *priv, * see * __graph_parse_node() :: DAI Naming */ - ret = graph_parse_node(priv, GRAPH_NORMAL, codec_port, li, 0); + ret = graph_parse_node(priv, GRAPH_NORMAL, codec_ep, li, 0); if (ret < 0) goto err; /* * call CPU, and set DAI Name */ - ret = graph_parse_node(priv, GRAPH_NORMAL, cpu_port, li, 1); + ret = graph_parse_node(priv, GRAPH_NORMAL, cpu_ep, li, 1); if (ret < 0) goto err; - graph_link_init(priv, cpu_port, li, 1); + graph_link_init(priv, cpu_ep, li, 1); err: - of_node_put(codec_port); + of_node_put(codec_ep); of_node_put(cpu_ep); return ret; @@ -684,7 +683,6 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, { struct device_node *ep = port_to_endpoint(lnk); struct device_node *rep = of_graph_get_remote_endpoint(ep); - struct device_node *rport = of_graph_get_remote_port(ep); struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct simple_dai_props *dai_props = simple_priv_to_props(priv, li->link); int is_cpu = asoc_graph_is_ports0(lnk); @@ -718,7 +716,7 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, dai_link->dynamic = 1; dai_link->dpcm_merged_format = 1; - ret = graph_parse_node(priv, GRAPH_DPCM, rport, li, 1); + ret = graph_parse_node(priv, GRAPH_DPCM, rep, li, 1); if (ret) goto err; } else { @@ -751,7 +749,7 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, dai_link->no_pcm = 1; dai_link->be_hw_params_fixup = asoc_simple_be_hw_params_fixup; - ret = graph_parse_node(priv, GRAPH_DPCM, rport, li, 0); + ret = graph_parse_node(priv, GRAPH_DPCM, rep, li, 0); if (ret < 0) goto err; } @@ -761,11 +759,10 @@ int audio_graph2_link_dpcm(struct asoc_simple_priv *priv, snd_soc_dai_link_set_capabilities(dai_link); - graph_link_init(priv, rport, li, is_cpu); + graph_link_init(priv, rep, li, is_cpu); err: of_node_put(ep); of_node_put(rep); - of_node_put(rport); return ret; } @@ -777,7 +774,7 @@ int audio_graph2_link_c2c(struct asoc_simple_priv *priv, { struct snd_soc_dai_link *dai_link = simple_priv_to_link(priv, li->link); struct device_node *port0, *port1, *ports; - struct device_node *codec0_port, *codec1_port; + struct device_node *codec0_ep, *codec1_ep; struct device_node *ep0, *ep1; u32 val = 0; int ret = -EINVAL; @@ -834,31 +831,31 @@ int audio_graph2_link_c2c(struct asoc_simple_priv *priv, ep0 = port_to_endpoint(port0); ep1 = port_to_endpoint(port1); - codec0_port = of_graph_get_remote_port(ep0); - codec1_port = of_graph_get_remote_port(ep1); + codec0_ep = of_graph_get_remote_endpoint(ep0); + codec1_ep = of_graph_get_remote_endpoint(ep1); /* * call Codec first. * see * __graph_parse_node() :: DAI Naming */ - ret = graph_parse_node(priv, GRAPH_C2C, codec1_port, li, 0); + ret = graph_parse_node(priv, GRAPH_C2C, codec1_ep, li, 0); if (ret < 0) goto err2; /* * call CPU, and set DAI Name */ - ret = graph_parse_node(priv, GRAPH_C2C, codec0_port, li, 1); + ret = graph_parse_node(priv, GRAPH_C2C, codec0_ep, li, 1); if (ret < 0) goto err2; - graph_link_init(priv, codec0_port, li, 1); + graph_link_init(priv, codec0_ep, li, 1); err2: of_node_put(ep0); of_node_put(ep1); - of_node_put(codec0_port); - of_node_put(codec1_port); + of_node_put(codec0_ep); + of_node_put(codec1_ep); err1: of_node_put(ports); of_node_put(port0); -- 2.30.2

3 9

[PATCH v2 0/5] Updates for rawmidi / sequencer
by Takashi Iwai 14 Jan '25

14 Jan '25

The kernel 6.14 will receive a few small enhancements for rawmidi and sequencer APIs, and this patch set covers those. v1->v2: sync with the latest rawmidi API change to pass 0 to tied_device as default v1: https://lore.kernel.org/20250114085515.11880-1-tiwai@suse.de Takashi === Takashi Iwai (5): Sync UAPI asound.h with 6.14 kernel Sync UAPI asequencer.h with 6.14 kernel rawmidi: Extensions for tied device and substream inactive flag rawmidi: Make rawmidi flag bits doxygen-style comments seq: Define new events for UMP EP/FB change notifications include/local.h | 1 + include/rawmidi.h | 4 +++- include/seq_event.h | 12 ++++++++++++ include/sound/uapi/asequencer.h | 12 +++++++++++- include/sound/uapi/asound.h | 8 ++++++-- src/rawmidi/rawmidi.c | 16 ++++++++++++++++ 6 files changed, 49 insertions(+), 4 deletions(-) -- 2.43.0

1 5

[PATCH alsa-lib 0/5] Updates for rawmidi / sequencer
by Takashi Iwai 14 Jan '25

14 Jan '25

The kernel 6.14 will receive a few small enhancements for rawmidi and sequencer APIs, and this patch set covers those. Takashi === Takashi Iwai (5): Sync UAPI asound.h with 6.14 kernel Sync UAPI asequencer.h with 6.14 kernel rawmidi: Extensions for tied device and substream inactive flag rawmidi: Make rawmidi flag bits doxygen-style comments seq: Define new events for UMP EP/FB change notifications include/local.h | 1 + include/rawmidi.h | 4 +++- include/seq_event.h | 12 ++++++++++++ include/sound/uapi/asequencer.h | 12 +++++++++++- include/sound/uapi/asound.h | 8 ++++++-- src/rawmidi/rawmidi.c | 22 +++++++++++++++++++++- 6 files changed, 54 insertions(+), 5 deletions(-) -- 2.43.0

1 5

rt722: change output volume of headphone to 0dB
by GitHub pull_request - opened 14 Jan '25

14 Jan '25

alsa-project/alsa-ucm-conf pull request #496 was opened from shumingfan: rt722: change output volume of headphone to 0dB Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/496 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/496.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v2 1/1] ASoC: codecs: nau8824: fix max volume for Speaker output
by Maxim Kochetkov 13 Jan '25

13 Jan '25

There is no audio output if Speaker volume is set above 25. According to datasheet Rev 2.5 maximum allowed value for the Speaker output is 0b11001 (25) 0x6D CLASSD_GAIN_1/ 0x6E CLASSD_GAIN_2: Left/Right Channel Class-D Driver Gain For DAC Left/Right Input (Step size is 1dB.) 00000 = Mute (DEFAULT) 00001 = 0dB 00002 = 1dB ... 11000 = 23dB 11001 = 24dB So adjust this value in accordance with the datasheet. Signed-off-by: Maxim Kochetkov <fido_max(a)inbox.ru> --- sound/soc/codecs/nau8824.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 12540397fd4d..5aaf8c496300 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -368,13 +368,13 @@ static const struct snd_kcontrol_new nau8824_snd_controls[] = { SOC_ENUM("DAC Oversampling Rate", nau8824_dac_oversampl_enum), SOC_SINGLE_TLV("Speaker Right DACR Volume", - NAU8824_REG_CLASSD_GAIN_1, 8, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_1, 8, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Speaker Left DACL Volume", - NAU8824_REG_CLASSD_GAIN_2, 0, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_2, 0, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Speaker Left DACR Volume", - NAU8824_REG_CLASSD_GAIN_1, 0, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_1, 0, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Speaker Right DACL Volume", - NAU8824_REG_CLASSD_GAIN_2, 8, 0x1f, 0, spk_vol_tlv), + NAU8824_REG_CLASSD_GAIN_2, 8, 0x19, 0, spk_vol_tlv), SOC_SINGLE_TLV("Headphone Right DACR Volume", NAU8824_REG_ATT_PORT0, 8, 0x1f, 0, hp_vol_tlv), -- 2.45.2

2 1

speaker-test.1: Some remarks and a patch with editorial changes for this man page
by Bjarni Ingi Gislason 11 Jan '25

11 Jan '25

Package: alsa-utils Version: 1.2.13-1 Severity: minor Tags: patch * What led up to the situation? Checking for defects with a new version test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -rCHECKSTYLE=10 -ww -z < "man page" [Use "groff -e ' $' <file>" to find trailing spaces.] ["test-groff" is a script in the repository for "groff"; is not shipped] (local copy and "troff" slightly changed by me). [The fate of "test-nroff" was decided in groff bug #55941.] * What was the outcome of this action? an.tmac:<stdin>:183: misuse, warning: .BR is for at least 2 arguments, got 1 Use macro '.B' for one argument or split argument. * What outcome did you expect instead? No output (no warnings). -.- General remarks and further material, if a diff-file exist, are in the attachments. -- System Information: Debian Release: trixie/sid APT prefers testing APT policy: (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 6.12.6-amd64 (SMP w/2 CPU threads; PREEMPT) Locale: LANG=is_IS.iso88591, LC_CTYPE=is_IS.iso88591 (charmap=ISO-8859-1), LANGUAGE not set Shell: /bin/sh linked to /usr/bin/dash Init: sysvinit (via /sbin/init) Versions of packages alsa-utils depends on: ii kmod 33+20240816-2 ii libasound2t64 1.2.13-1+b1 ii libatopology2t64 1.2.13-1+b1 ii libc6 2.40-4 ii libfftw3-single3 3.3.10-2+b1 ii libncursesw6 6.5-2+b1 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5-2+b1 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

1 0

[GIT PULL] ASoC fixes for v6.13-rc6
by Mark Brown 10 Jan '25

10 Jan '25

The following changes since commit 32c9c06adb5b157ef259233775a063a43746d699: ASoC: mediatek: disable buffer pre-allocation (2024-12-19 11:15:09 +0000) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.13-rc6 for you to fetch changes up to 139fa599cea0fd9d38e00246ea9f79af6c59acbd: ASoC: rsnd: check rsnd_adg_clk_enable() return value (2025-01-09 13:17:02 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.13 A collection of device specific fixes that came in over the holidays, plus a MAINTAINERS update and some documentation to help users debug problems with some of the Cirrus CODECs found in modern laptops. ---------------------------------------------------------------- Charles Keepax (3): ASoC: wm8994: Add depends on MFD core ASoC: samsung: Add missing selects for MFD_WM8994 ASoC: samsung: Add missing depends on I2C Kuninori Morimoto (2): MAINTAINERS: add missing maintainers for Simple Audio Card ASoC: rsnd: check rsnd_adg_clk_enable() return value Maciej Strozek (1): ASoC: cs42l43: Add codec force suspend/resume ops Marian Postevca (1): ASoC: codecs: es8316: Fix HW rate calculation for 48Mhz MCLK Richard Fitzgerald (2): ALSA: doc: cs35l56: Add information about Cirrus Logic CS35L54/56/57 ALSA: doc: Add codecs/index.rst to top-level index Shenghao Ding (1): ASoC: tas2781: Fix occasional calibration failture Zhang Yi (1): ASoC: codecs: ES8326: Adjust ANA_MICBIAS to reduce pop noise Documentation/sound/codecs/cs35l56.rst | 292 +++++++++++++++++++++++++++++++++ Documentation/sound/codecs/index.rst | 9 + Documentation/sound/index.rst | 1 + MAINTAINERS | 2 + sound/soc/codecs/Kconfig | 1 + sound/soc/codecs/cs42l43.c | 1 + sound/soc/codecs/es8316.c | 10 +- sound/soc/codecs/es8326.c | 23 ++- sound/soc/codecs/tas2781-i2c.c | 2 +- sound/soc/renesas/rcar/adg.c | 28 +++- sound/soc/renesas/rcar/core.c | 4 +- sound/soc/renesas/rcar/rsnd.h | 2 +- sound/soc/samsung/Kconfig | 6 +- 13 files changed, 364 insertions(+), 17 deletions(-) create mode 100644 Documentation/sound/codecs/cs35l56.rst create mode 100644 Documentation/sound/codecs/index.rst

2 1

alsactl info fails with exit code 25
by GitHub issues - opened 09 Jan '25

09 Jan '25

alsa-project/alsa-utils issue #287 was opened from z-s-e: Here is the output of running alsactl info on my machine: ``` $ alsactl info 0 # # Sound card # - card: 0 id: Generic name: HD-Audio Generic longname: HD-Audio Generic at 0xb88c8000 irq 87 driver_name: HDA-Intel mixer_name: ATI R6xx HDMI components: HDA:1002aa01,00aa0100,00100800 controls_count: 15 pcm: - stream: PLAYBACK devices: - device: 3 id: HDMI 0 name: G246HYL subdevices: - subdevice: 0 name: subdevice #0 - device: 7 id: HDMI 1 name: HDMI 1 subdevices: - subdevice: 0 name: subdevice #0 alsactl: rawmidi_device_list:105: snd_ctl_rawmidi_next_device $ echo $? 25 ``` Apparently it fails when trying to enumerate raw midi devices. I don't think this is supposed to happen, right? alsa-info for reference: [alsa-info.txt](https://github.com/user-attachments/files/18362369/alsa-info… Issue URL : https://github.com/alsa-project/alsa-utils/issues/287 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH v1] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 09 Jan '25

09 Jan '25

Due to internal re-org, Kevin has no longer maintained audio driver. With the job responsibility change, drop entries for the audio converter and add entries for both haptics drivers and middle-power audio amplifier drivers. Add audio converter, set the Status as Supported. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- MAINTAINERS | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 30cbc3d44cd5..4c806ad2fb16 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -23172,25 +23172,42 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: linux-sound(a)vger.kernel.org -S: Maintained +S: Supported +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml F: Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml +F: drivers/input/misc/drv2*.c +F: include/sound/tas2*.h +F: include/sound/tas5086.h +F: include/sound/tpa6130a2-plat.h +F: sound/pci/hda/tas2781_hda_i2c.c +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tas5*.* +F: sound/soc/codecs/tpa6130a2.* + +TEXAS INSTRUMENTS AUDIO Converter (ASoC) and Touch Screen DRIVERS +L: linux-sound(a)vger.kernel.org +S: Maintained F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm6240.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: include/sound/tas2*.h +F: drivers/input/touchscreen/tsc2*.* F: include/sound/tlv320*.h -F: include/sound/tpa6130a2-plat.h -F: sound/pci/hda/tas2781_hda_i2c.c F: sound/soc/codecs/pcm1681.c F: sound/soc/codecs/pcm1789*.* F: sound/soc/codecs/pcm179x*.* @@ -23200,9 +23217,8 @@ F: sound/soc/codecs/pcm3060*.* F: sound/soc/codecs/pcm3168a*.* F: sound/soc/codecs/pcm5102a.c F: sound/soc/codecs/pcm512x*.* -F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/pcm6240.* F: sound/soc/codecs/tlv320*.* -F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> -- 2.34.1

1 0

Topology: NHLT: Intel: Fix mono DMIC configure for MTL platform
by GitHub pull_request - opened 08 Jan '25

08 Jan '25

alsa-project/alsa-utils pull request #286 was opened from singalsu: This change fixes the blob generator for mono microphone configuration. As difference to previous platforms the FIFO packer mono/stereo mode set up in OUTCONTROLx IPM_SOURCE_MODE bit-field. Request URL : https://github.com/alsa-project/alsa-utils/pull/286 Patch URL : https://github.com/alsa-project/alsa-utils/pull/286.patch Repository URL: https://github.com/alsa-project/alsa-utils

1 0

aseqdump version 1.2.11, There are small bugs
by GitHub issues - opened 08 Jan '25

08 Jan '25

alsa-project/alsa-utils issue #285 was opened from masahirokakishita: Hello, I found small bugs in aseqdump.c static void dump_ump_midi2_event(const unsigned int *ump) ... case SND_UMP_MSG_PITCHBEND: printf("Channel pressure %2d, value %s", channel, midi2_pitchbend(m->channel_pressure.data)); break; It is not "Channel pressure", but "Pitch Bend". case SND_UMP_MSG_NOTE_ON: printf("Note on %2d, note %d, velocity %s, attr type = %d, data = 0x%x", channel, m->note_off.note, midi2_velocity(m->note_off.velocity), m->note_off.attr_type, m->note_off.attr_data); break; "note->on" is correct. This bug does not affect the display. Issue URL : https://github.com/alsa-project/alsa-utils/issues/285 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH] ASoC: rt5682: Support the ALC5682I-VE codec
by derek.fang＠realtek.com 07 Jan '25

07 Jan '25

From: Derek Fang <derek.fang(a)realtek.com> The ALC5682I-VD and ALC5682I-VE use the same I2C codec driver with different calibration settings. This patch aims to handle their differences. Signed-off-by: Derek Fang <derek.fang(a)realtek.com> diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index ff9e14fad0cd..a8820435d1e0 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -186,6 +186,12 @@ static int rt5682_i2c_probe(struct i2c_client *i2c) return -ENODEV; } + regmap_read(rt5682->regmap, RT5682_INT_DEVICE_ID, &val); + if (val == 0x6956) { + dev_dbg(&i2c->dev, "ALC5682I-VE device\n"); + rt5682->ve_ic = true; + } + mutex_init(&rt5682->calibrate_mutex); rt5682_calibrate(rt5682); diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index aa163ec40862..b4d72fc4a44d 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -395,6 +395,7 @@ bool rt5682_volatile_register(struct device *dev, unsigned int reg) case RT5682_4BTN_IL_CMD_1: case RT5682_AJD1_CTRL: case RT5682_HP_CALIB_CTRL_1: + case RT5682_INT_DEVICE_ID: case RT5682_DEVICE_ID: case RT5682_I2C_MODE: case RT5682_HP_CALIB_CTRL_10: @@ -419,6 +420,7 @@ bool rt5682_readable_register(struct device *dev, unsigned int reg) { switch (reg) { case RT5682_RESET: + case RT5682_INT_DEVICE_ID: case RT5682_VERSION_ID: case RT5682_VENDOR_ID: case RT5682_DEVICE_ID: @@ -3139,7 +3141,10 @@ void rt5682_calibrate(struct rt5682_priv *rt5682) regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0100); regmap_write(rt5682->regmap, RT5682_HP_IMP_SENS_CTRL_19, 0x3800); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x3000); - regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7005); + if (rt5682->ve_ic) + regmap_write(rt5682->regmap, RT5682_CHOP_ADC, 0x7005); + else + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x7005); regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0x686c); regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0d0d); regmap_write(rt5682->regmap, RT5682_HP_CALIB_CTRL_2, 0x0321); @@ -3168,7 +3173,10 @@ void rt5682_calibrate(struct rt5682_priv *rt5682) regmap_write(rt5682->regmap, RT5682_GLB_CLK, 0x0000); regmap_write(rt5682->regmap, RT5682_PWR_DIG_1, 0x0000); regmap_write(rt5682->regmap, RT5682_CHOP_DAC, 0x2000); - regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); + if (rt5682->ve_ic) + regmap_write(rt5682->regmap, RT5682_CHOP_ADC, 0x2005); + else + regmap_write(rt5682->regmap, RT5682_CALIB_ADC_CTRL, 0x2005); regmap_write(rt5682->regmap, RT5682_STO1_ADC_MIXER, 0xc0c4); regmap_write(rt5682->regmap, RT5682_CAL_REC, 0x0c0c); diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index b2d9e87af259..de43a5d99403 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -22,6 +22,7 @@ /* Info */ #define RT5682_RESET 0x0000 +#define RT5682_INT_DEVICE_ID 0x00f9 #define RT5682_VERSION_ID 0x00fd #define RT5682_VENDOR_ID 0x00fe #define RT5682_DEVICE_ID 0x00ff @@ -1446,6 +1447,7 @@ struct rt5682_priv { bool hw_init; bool first_hw_init; bool is_sdw; + bool ve_ic; #ifdef CONFIG_COMMON_CLK struct clk_hw dai_clks_hw[RT5682_DAI_NUM_CLKS]; -- 2.34.1

2 1

[PATCH v1] ASoC: tas2781: Fix occasional calibration failture
by Shenghao Ding 07 Jan '25

07 Jan '25

The root cause is that TAS2781 internal register is not unlocked before writing data into TAS2781_PRM_TEST_57_REG. Fixes: 49e2e353fb0d ("ASoC: tas2781: Add Calibration Kcontrols for Chromebook") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index fb8cd2284fe8..728bf78ae71f 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -78,7 +78,7 @@ static const struct bulk_reg_val tas2781_cali_start_reg[] = { X2781_CL_STT_VAL(TAS2781_PRM_INT_MASK_REG, 0xfe, false), X2781_CL_STT_VAL(TAS2781_PRM_CLK_CFG_REG, 0xdd, false), X2781_CL_STT_VAL(TAS2781_PRM_RSVD_REG, 0x20, false), - X2781_CL_STT_VAL(TAS2781_PRM_TEST_57_REG, 0x14, false), + X2781_CL_STT_VAL(TAS2781_PRM_TEST_57_REG, 0x14, true), X2781_CL_STT_VAL(TAS2781_PRM_TEST_62_REG, 0x45, true), X2781_CL_STT_VAL(TAS2781_PRM_PVDD_UVLO_REG, 0x03, false), X2781_CL_STT_VAL(TAS2781_PRM_CHNL_0_REG, 0xa8, false), -- 2.34.1

2 1

[PATCH RESEND] ASoC: codecs: es8316: Fix HW rate calculation for 48Mhz MCLK
by Marian Postevca 07 Jan '25

07 Jan '25

For 48Mhz MCLK systems the calculation of the HW rate is broken, and will not produce even one sane rate. Since es83xx supports the option to halve MCLK, calculate also rates with MCLK/2. Signed-off-by: Marian Postevca <posteuca(a)mutex.one> --- sound/soc/codecs/es8316.c | 10 +++++++++- 1 file changed, 9 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index 61729e5b50a8..f508df01145b 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -39,7 +39,9 @@ struct es8316_priv { struct snd_soc_jack *jack; int irq; unsigned int sysclk; - unsigned int allowed_rates[ARRAY_SIZE(supported_mclk_lrck_ratios)]; + /* ES83xx supports halving the MCLK so it supports twice as many rates + */ + unsigned int allowed_rates[ARRAY_SIZE(supported_mclk_lrck_ratios) * 2]; struct snd_pcm_hw_constraint_list sysclk_constraints; bool jd_inverted; }; @@ -386,6 +388,12 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, if (freq % ratio == 0) es8316->allowed_rates[count++] = freq / ratio; + + /* We also check if the halved MCLK produces a valid rate + * since the codec supports halving the MCLK. + */ + if ((freq / ratio) % 2 == 0) + es8316->allowed_rates[count++] = freq / ratio / 2; } if (count) { -- 2.45.2

2 1

[PATCH] ASoC: codecs: ES8326: Adjust ANA_MICBIAS to reduce pop noise
by Zhang Yi 07 Jan '25

07 Jan '25

We modify the value of ES8326_ANA_MICBIAS to reduce the pop noise Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 23 +++++++++++++++++++---- 1 file changed, 19 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index a5603b617688..b06eead7e0f6 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -616,7 +616,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) 0x0F, 0x0F); if (es8326->version > ES8326_VERSION_B) { regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30); } } } else { @@ -631,6 +631,8 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) regmap_write(es8326->regmap, ES8326_HPR_OFFSET_INI, offset_r); es8326->calibrated = true; } + regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0x00); + regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x00); if (direction == SNDRV_PCM_STREAM_PLAYBACK) { regmap_update_bits(es8326->regmap, ES8326_DAC_DSM, 0x01, 0x01); usleep_range(1000, 5000); @@ -645,7 +647,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) } else { msleep(300); if (es8326->version > ES8326_VERSION_B) { - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x50); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x70); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x00); } regmap_update_bits(es8326->regmap, ES8326_ADC_MUTE, @@ -676,6 +678,10 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x00); regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x20); regmap_update_bits(es8326->regmap, ES8326_RESET, 0x02, 0x00); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30); + } break; case SND_SOC_BIAS_PREPARE: break; @@ -683,6 +689,12 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x3b); regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x00); regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, ES8326_IO_INPUT); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); + } + regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0xc0); + regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x80); break; case SND_SOC_BIAS_OFF: clk_disable_unprepare(es8326->mclk); @@ -773,7 +785,10 @@ static void es8326_jack_button_handler(struct work_struct *work) case 0x6f: case 0x4b: /* button volume up */ - cur_button = SND_JACK_BTN_1; + if ((iface == 0x6f) && (es8326->version > ES8326_VERSION_B)) + cur_button = SND_JACK_BTN_0; + else + cur_button = SND_JACK_BTN_1; break; case 0x27: /* button volume down */ @@ -1082,7 +1097,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66); es8326_disable_micbias(es8326->component); if (es8326->version > ES8326_VERSION_B) { - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x13); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x10); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); } -- 2.17.1

2 1

headers: make more headers self-contained
by GitHub pull_request - opened 06 Jan '25

06 Jan '25

alsa-project/alsa-lib pull request #435 was opened from timblechmann: Some IDEs parse headers in isolation to populate their code model. This exposes some missing headers, mainly for alsa internal struct definitions, ssize_t (requiring sys/types.h), size_t (requiring string.h), va_args (requiring stdargs.h) or FILE (requiring stdio.h) This patch makes the main headers parsable by qtcreator Request URL : https://github.com/alsa-project/alsa-lib/pull/435 Patch URL : https://github.com/alsa-project/alsa-lib/pull/435.patch Repository URL: https://github.com/alsa-project/alsa-lib

1 0

bug in the asla driver asla-sink-viritio-snd
by GitHub issues - opened 06 Jan '25

06 Jan '25

alsa-project/alsa-lib issue #434 was opened from LoreFiore: <img width="1440" alt="Screenshot 2025-01-05 alle 23 31 08" src="https://github.com/user-attachments/assets/34b279a0-b256-4e7b-82ec-a9df46f9…" /> Issue URL : https://github.com/alsa-project/alsa-lib/issues/434 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ld.lld default version symbol snd_dlopen@@ALSA_1.1.6 must be defined
by GitHub issues - edited 05 Jan '25

05 Jan '25

alsa-project/alsa-lib issue #433 was edited from trcrsired: ``` clang: warning: argument unused during compilation: '--unwindlib=libunwind' [-Wunused-command-line-argument] CCLD libasound.la ld.lld: error: seq/.libs/libseq.a(seq.o at 50494) <inline asm>:1:1: default version symbol snd_dlopen@@ALSA_1.1.6 must be defined .symver __snd_dlopen, snd_dlopen@@ALSA_1.1.6 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:1:1: default version symbol snd_pcm_hw_params_set_format_first@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_format_first, snd_pcm_hw_params_set_format_first@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:2:1: default version symbol snd_pcm_hw_params_set_channels_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_channels_near, snd_pcm_hw_params_set_channels_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:3:1: default version symbol snd_pcm_hw_params_set_rate_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_rate_near, snd_pcm_hw_params_set_rate_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:4:1: default version symbol snd_pcm_hw_params_set_period_time_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_period_time_near, snd_pcm_hw_params_set_period_time_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:5:1: default version symbol snd_pcm_hw_params_set_period_size_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_period_size_near, snd_pcm_hw_params_set_period_size_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:6:1: default version symbol snd_pcm_hw_params_set_periods_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_periods_near, snd_pcm_hw_params_set_periods_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:7:1: default version symbol snd_pcm_hw_params_set_buffer_time_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_buffer_time_near, snd_pcm_hw_params_set_buffer_time_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:8:1: default version symbol snd_pcm_hw_params_set_buffer_size_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_buffer_size_near, snd_pcm_hw_params_set_buffer_size_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:9:1: default version symbol snd_timer_params_set_filter@@ALSA_0.9.0 must be defined .symver __snd_timer_params_set_filter, snd_timer_params_set_filter@@ALSA_0.9.0 ^ ld.lld: error: .libs/conf.o <inline asm>:1:1: default version symbol snd_dlopen@@ALSA_1.1.6 must be defined .symver __snd_dlopen, snd_dlopen@@ALSA_1.1.6 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_hw.o at 720480) <inline asm>:1:1: default version symbol snd_timer_params_set_filter@@ALSA_0.9.0 must be defined .symver __snd_timer_params_set_filter, snd_timer_params_set_filter@@ALSA_0.9.0 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:1:1: default version symbol snd_pcm_hw_params_get_access@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_access, snd_pcm_hw_params_get_access@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:2:1: default version symbol snd_pcm_hw_params_get_format@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_format, snd_pcm_hw_params_get_format@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:3:1: default version symbol snd_pcm_hw_params_get_subformat@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_subformat, snd_pcm_hw_params_get_subformat@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:4:1: default version symbol snd_pcm_hw_params_get_channels@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_channels, snd_pcm_hw_params_get_channels@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:5:1: default version symbol snd_pcm_hw_params_get_rate@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_rate, snd_pcm_hw_params_get_rate@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:6:1: default version symbol snd_pcm_hw_params_get_period_time@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_period_time, snd_pcm_hw_params_get_period_time@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:7:1: default version symbol snd_pcm_hw_params_get_period_size@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_period_size, snd_pcm_hw_params_get_period_size@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:8:1: default version symbol snd_pcm_hw_params_get_buffer_size@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_buffer_size, snd_pcm_hw_params_get_buffer_size@@ALSA_0.9.0rc4 ^ ld.lld: error: too many errors emitted, stopping now (use --error-limit=0 to see all errors) clang: error: linker command failed with exit code 1 (use -v to see invocation) make[3]: *** [Makefile:506: libasound.la] Error 1 make[3]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make[2]: *** [Makefile:567: all-recursive] Error 1 make[2]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make[1]: *** [Makefile:436: all] Error 2 make[1]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make: *** [Makefile:414: all-recursive] Error 1 alsa-lib build failure ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/433 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

default version symbol snd_dlopen@@ALSA_1.1.6 must be defined
by GitHub issues - opened 05 Jan '25

05 Jan '25

alsa-project/alsa-lib issue #433 was opened from trcrsired: ``` clang: warning: argument unused during compilation: '--unwindlib=libunwind' [-Wunused-command-line-argument] CCLD libasound.la ld.lld: error: seq/.libs/libseq.a(seq.o at 50494) <inline asm>:1:1: default version symbol snd_dlopen@@ALSA_1.1.6 must be defined .symver __snd_dlopen, snd_dlopen@@ALSA_1.1.6 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:1:1: default version symbol snd_pcm_hw_params_set_format_first@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_format_first, snd_pcm_hw_params_set_format_first@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:2:1: default version symbol snd_pcm_hw_params_set_channels_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_channels_near, snd_pcm_hw_params_set_channels_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:3:1: default version symbol snd_pcm_hw_params_set_rate_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_rate_near, snd_pcm_hw_params_set_rate_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:4:1: default version symbol snd_pcm_hw_params_set_period_time_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_period_time_near, snd_pcm_hw_params_set_period_time_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:5:1: default version symbol snd_pcm_hw_params_set_period_size_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_period_size_near, snd_pcm_hw_params_set_period_size_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:6:1: default version symbol snd_pcm_hw_params_set_periods_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_periods_near, snd_pcm_hw_params_set_periods_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:7:1: default version symbol snd_pcm_hw_params_set_buffer_time_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_buffer_time_near, snd_pcm_hw_params_set_buffer_time_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:8:1: default version symbol snd_pcm_hw_params_set_buffer_size_near@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_set_buffer_size_near, snd_pcm_hw_params_set_buffer_size_near@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_direct.o at 2925704) <inline asm>:9:1: default version symbol snd_timer_params_set_filter@@ALSA_0.9.0 must be defined .symver __snd_timer_params_set_filter, snd_timer_params_set_filter@@ALSA_0.9.0 ^ ld.lld: error: .libs/conf.o <inline asm>:1:1: default version symbol snd_dlopen@@ALSA_1.1.6 must be defined .symver __snd_dlopen, snd_dlopen@@ALSA_1.1.6 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_hw.o at 720480) <inline asm>:1:1: default version symbol snd_timer_params_set_filter@@ALSA_0.9.0 must be defined .symver __snd_timer_params_set_filter, snd_timer_params_set_filter@@ALSA_0.9.0 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:1:1: default version symbol snd_pcm_hw_params_get_access@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_access, snd_pcm_hw_params_get_access@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:2:1: default version symbol snd_pcm_hw_params_get_format@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_format, snd_pcm_hw_params_get_format@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:3:1: default version symbol snd_pcm_hw_params_get_subformat@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_subformat, snd_pcm_hw_params_get_subformat@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:4:1: default version symbol snd_pcm_hw_params_get_channels@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_channels, snd_pcm_hw_params_get_channels@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:5:1: default version symbol snd_pcm_hw_params_get_rate@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_rate, snd_pcm_hw_params_get_rate@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:6:1: default version symbol snd_pcm_hw_params_get_period_time@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_period_time, snd_pcm_hw_params_get_period_time@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:7:1: default version symbol snd_pcm_hw_params_get_period_size@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_period_size, snd_pcm_hw_params_get_period_size@@ALSA_0.9.0rc4 ^ ld.lld: error: pcm/.libs/libpcm.a(pcm_params.o at 544772) <inline asm>:8:1: default version symbol snd_pcm_hw_params_get_buffer_size@@ALSA_0.9.0rc4 must be defined .symver __snd_pcm_hw_params_get_buffer_size, snd_pcm_hw_params_get_buffer_size@@ALSA_0.9.0rc4 ^ ld.lld: error: too many errors emitted, stopping now (use --error-limit=0 to see all errors) clang: error: linker command failed with exit code 1 (use -v to see invocation) make[3]: *** [Makefile:506: libasound.la] Error 1 make[3]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make[2]: *** [Makefile:567: all-recursive] Error 1 make[2]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make[1]: *** [Makefile:436: all] Error 2 make[1]: Leaving directory '/home/csrg/toolchains_build/toolchainbuildscripts/wine/.wineartifacts/aarch64-linux-gnu/alsa-lib/src' make: *** [Makefile:414: all-recursive] Error 1 alsa-lib build failure ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/433 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

ucm2: USB-Audio: add Presonus Revelator IO 44 (USB194f:0424)
by GitHub pull_request - edited 04 Jan '25

04 Jan '25

alsa-project/alsa-ucm-conf pull request #495 was edited from markulovi: [alsa-info.txt](https://github.com/user-attachments/files/18306150/alsa-info… This adds support for [Presonus Revelator io44](https://eu.presonus.com/products/revelator-io44). This device has 3 virtual stereo channels out and 4 virtual stereo channels in as follows: ![out](https://github.com/user-attachments/assets/f4cacf14-e75e-4d33-ad64-cca4ca7ee18e) ![in](https://github.com/user-attachments/assets/60515091-51a5-4299-97c7-a12d3c091b8a) Microphone inputs have been separated into two mono channels for convenience. As far as I can tell, the [Universal Control](https://www.presonus.com/pages/support-documents-downloads?product… software is required to configure levels and effects of these channels, but the settings can be saved to the device and operation without the software is possible outside Windows. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/495 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/495.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: USB-Audio: add Presonus Revelator IO 44 (USB194f:0424)
by GitHub pull_request - opened 04 Jan '25

04 Jan '25

alsa-project/alsa-ucm-conf pull request #495 was opened from markulovi: [alsa-info.txt](https://github.com/user-attachments/files/18306150/alsa-info… This adds support for [Presonus Revelator io44](https://eu.presonus.com/products/revelator-io44). This device has 3 virtual stereo channels out and 4 virtual stereo channels in as follows: ![out](https://github.com/user-attachments/assets/f4cacf14-e75e-4d33-ad64-cca4ca7ee18e) ![in](https://github.com/user-attachments/assets/60515091-51a5-4299-97c7-a12d3c091b8a) Microphone inputs have been separated to two mono channels for convenience. As far as I can tell, the [Universal Control](https://www.presonus.com/pages/support-documents-downloads?product… software is required to configure levels and effects of these channels, but the settings can be saved to the device and operation without the software is possible outside Windows. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/495 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/495.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH 0/5] soundwire: amd: code improvements and new platform
by Vijendar Mukunda 02 Jan '25

02 Jan '25

This patch series consists of code improvements and new platform support for ACP7.0 & ACP7.1 platforms. Vijendar Mukunda (5): soundwire: amd: change the soundwire wake enable/disable sequence soundwire: amd: add debug log for soundwire wake event soundwire: amd: add support for ACP7.0 & ACP7.1 platforms soundwire: amd: add soundwire host wake interrupt enable/disable sequence soundwire: amd: clear soundwire pme wake status drivers/soundwire/amd_manager.c | 102 ++++++++++++++++++++++++++++++ drivers/soundwire/amd_manager.h | 27 ++++++++ include/linux/soundwire/sdw_amd.h | 2 + 3 files changed, 131 insertions(+) -- 2.34.1

2 7

Missunderstanding or a BUG in reported frequencies for audio cards present in system.
by GitHub issues - edited 02 Jan '25

02 Jan '25

alsa-project/alsa-lib issue #432 was edited from CapSel: I use ArchLinux with kernel 6.12.7 with "Zen" patches. HW is ASUS Vivobook 14X pro. So the problem I'm having is that I can't set pipewire to 96000 sampling rate. I tried all options in pipewire. Nothing helped. But a long time ago it worked fine - I can't remember when (at most week after Baldur's Gate 3 release ;). What I see in system: when I do `cat /proc/asound/card0/pcm3p/info` (no other pcm* files) I can see `name: HDMI 0` - it's important. with `cat /proc/asound/card0/codec#0` on top there is: ``` Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100700 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x1]: PCM ``` for the other (default) audio card: `cat /proc/asound/card1/pcm0p/info` (there is pcm0c file too but I did not check it) gives: `name: ALC294 Analog` `cat /proc/asound/card1/codec#0` on top: ``` Codec: Realtek ALC294 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0294 Subsystem Id: 0x10431ab2 Revision Id: 0x100004 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM ``` so as you can see: - hdmi supports up to 48000 - analog supports up to 192000 I first noticed there is something wrong with alsacap program that reports: ``` *** Scanning for playback devices *** Card 0, ID `Generic', name `HD-Audio Generic' Device 3, ID `HDMI 0', name `HDMI 0', 1 subdevices (1 available) 2..8 channels, sampling rate 32000..192000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' Card 1, ID `Generic_1', name `HD-Audio Generic' Device 0, ID `ALC294 Analog', name `ALC294 Analog', 1 subdevices (1 available) 2 channels, sampling rate 44100..48000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' ``` As you can see: - hdmi supports up to 192000 - analog supports up to 48000 so it's... flipped? here is a output from `cat /proc/asound/pcm`: ``` 00-03: HDMI 0 : HDMI 0 : playback 1 01-00: ALC294 Analog : ALC294 Analog : playback 1 : capture 1 ``` To confirm this I downloaded flac file with 96000 sample rate and tried to play it with ffmpeg after pipewire,pipewire-pulse,wireplumber services are stopped. I confirmed that these programs are disabled by checking if `ps ax | grep wire` before and after, also I monitored logs from these services. It reported error for device "hw:1,0" because it could not set sampling rate to 96000. It did not fail on "hw:0,3" but of course I did not hear anything since external monitor is disconnected. Replay command: `ffmpeg -i yuri_korzunov-through-moon_ring-96kHz-24bit.flac -f alsa hw:1,0` the file is some random file found in internet that tested with `ffprobe` shows that it has sample rate of 96000. I don't want to attach it since I forgot where I found it ;) I don't have `asound.conf` files in `/etc/` or `~/.config`. After connecting external monitor alsacap reports max 48000 for both cards. FFmpeg reports sampling rate problem with both audio cards: ``` [alsa @ 0x586316542b00] sample rate 96000 not available, nearest is 48000 ``` I had this monitor connected before and alsacap reported "flipped" results - not like now that both cards are limited to 48000. What else can I do to discover the problem? Issue URL : https://github.com/alsa-project/alsa-lib/issues/432 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Missunderstanding or a BUG in reported frequencies for audio cards present in system.
by GitHub issues - edited 02 Jan '25

02 Jan '25

alsa-project/alsa-lib issue #432 was edited from CapSel: I use ArchLinux with kernel 6.12.7 with "Zen" patches. HW is ASUS Vivobook 14X pro. So the problem I'm having is that I can't set pipewire to 96000 sampling rate. I tried all options in pipewire. Nothing helped. But a long time ago it worked fine - I can't remember when (at most week after Baldur's Gate 3 release ;). What I see in system: when I do `cat /proc/asound/card0/pcm3p/info` (no other pcm* files) I can see `name: HDMI 0` - it's important. with `cat /proc/asound/card0/codec#0` on top there is: ``` Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100700 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x1]: PCM ``` for the other (default) audio card: `cat /proc/asound/card1/pcm0p/info` (there is pcm0c file too but I did not check it) gives: `name: ALC294 Analog` `cat /proc/asound/card1/codec#0` on top: ``` Codec: Realtek ALC294 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0294 Subsystem Id: 0x10431ab2 Revision Id: 0x100004 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM ``` so as you can see: - hdmi supports up to 48000 - analog supports up to 192000 I first noticed there is something wrong with alsacap program that reports: ``` *** Scanning for playback devices *** Card 0, ID `Generic', name `HD-Audio Generic' Device 3, ID `HDMI 0', name `HDMI 0', 1 subdevices (1 available) 2..8 channels, sampling rate 32000..192000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' Card 1, ID `Generic_1', name `HD-Audio Generic' Device 0, ID `ALC294 Analog', name `ALC294 Analog', 1 subdevices (1 available) 2 channels, sampling rate 44100..48000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' ``` As you can see: - hdmi supports up to 192000 - analog supports up to 48000 so it's... flipped? here is a output from `cat /proc/asound/pcm`: ``` 00-03: HDMI 0 : HDMI 0 : playback 1 01-00: ALC294 Analog : ALC294 Analog : playback 1 : capture 1 ``` To confirm this I downloaded flac file with 96000 sample rate and tried to play it with ffmpeg after pipewire,pipewire-pulse,wireplumber services are stopped. I confirmed that these programs are disabled by checking if `ps ax | grep wire` before and after, also I monitored logs from these services. It reported error for device "hw:1,0" because it could not set sampling rate to 96000. It did not fail on "hw:0,3" but of course I did not hear anything since external monitor is disconnected. Replay command: `ffmpeg -i yuri_korzunov-through-moon_ring-96kHz-24bit.flac -f alsa hw:1,0` the file is some random file found in internet that tested with `ffprobe` shows that it has sample rate of 96000. I don't want to attach it since I forgot where I found it ;) I don't have `asound.conf` files in `/etc/` or `~/.config`. After connecting external monitor alsacap reports max 48000 for both cards. FFmpeg reports sampling rate problem with both audio cards: ``` [alsa @ 0x586316542b00] sample rate 96000 not available, nearest is 48000 ``` What else can I do to discover the problem? Issue URL : https://github.com/alsa-project/alsa-lib/issues/432 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Missunderstanding or a BUG in reported frequencies for audio cards present in system.
by GitHub issues - edited 02 Jan '25

02 Jan '25

alsa-project/alsa-lib issue #432 was edited from CapSel: I use ArchLinux with kernel 6.12.7 with "Zen" patches. HW is ASUS Vivobook 14X pro. So the problem I'm having is that I can't set pipewire to 96000 sampling rate. I tried all options in pipewire. Nothing helped. But a long time ago it worked fine - I can't remember when. What I see in system: when I do `cat /proc/asound/card0/pcm3p/info` (no other pcm* files) I can see `name: HDMI 0` - it's important. with `cat /proc/asound/card0/codec#0` on top there is: ``` Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100700 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x1]: PCM ``` for the other (default) audio card: `cat /proc/asound/card1/pcm0p/info` (there is pcm0c file too but I did not check it) gives: `name: ALC294 Analog` `cat /proc/asound/card1/codec#0` on top: ``` Codec: Realtek ALC294 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0294 Subsystem Id: 0x10431ab2 Revision Id: 0x100004 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM ``` so as you can see: - hdmi supports up to 48000 - analog supports up to 192000 I first noticed there is something wrong with alsacap program that reports: ``` *** Scanning for playback devices *** Card 0, ID `Generic', name `HD-Audio Generic' Device 3, ID `HDMI 0', name `HDMI 0', 1 subdevices (1 available) 2..8 channels, sampling rate 32000..192000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' Card 1, ID `Generic_1', name `HD-Audio Generic' Device 0, ID `ALC294 Analog', name `ALC294 Analog', 1 subdevices (1 available) 2 channels, sampling rate 44100..48000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' ``` As you can see: - hdmi supports up to 192000 - analog supports up to 48000 so it's... flipped? here is a output from `cat /proc/asound/pcm`: ``` 00-03: HDMI 0 : HDMI 0 : playback 1 01-00: ALC294 Analog : ALC294 Analog : playback 1 : capture 1 ``` To confirm this I downloaded flac file with 96000 sample rate and tried to play it with ffmpeg after pipewire,pipewire-pulse,wireplumber services are stopped. I confirmed that these programs are disabled by checking if `ps ax | grep wire` before and after, also I monitored logs from these services. It reported error for device "hw:1,0" because it could not set sampling rate to 96000. It did not fail on "hw:0,3" but of course I did not hear anything since external monitor is disconnected. Replay command: `ffmpeg -i yuri_korzunov-through-moon_ring-96kHz-24bit.flac -f alsa hw:1,0` the file is some random file found in internet that tested with `ffprobe` shows that it has sample rate of 96000. I don't want to attach it since I forgot where I found it ;) I don't have `asound.conf` files in `/etc/` or `~/.config`. After connecting external monitor alsacap reports max 48000 for both cards. FFmpeg reports sampling rate problem with both audio cards: ``` [alsa @ 0x586316542b00] sample rate 96000 not available, nearest is 48000 ``` What else can I do to discover the problem? Issue URL : https://github.com/alsa-project/alsa-lib/issues/432 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Missunderstanding or a BUG in reported frequencies for audio cards present in system.
by GitHub issues - opened 02 Jan '25

02 Jan '25

alsa-project/alsa-lib issue #432 was opened from CapSel: I use ArchLinux with kernel 6.12.7 with "Zen" patches. HW is ASUS Vivobook 14X pro. So the problem I'm having is that I can't set pipewire to 96000 sampling rate. I tried all options in pipewire. Nothing helped. But a long time ago it worked fine - I can't remember when. What I see in system: when I do `cat /proc/asound/card0/pcm3p/info` (no other pcm* files) I can see `name: HDMI 0` - it's important. with `cat /proc/asound/card0/codec#0` on top there is: ``` Codec: ATI R6xx HDMI Address: 0 AFG Function Id: 0x1 (unsol 0) Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100700 No Modem Function Group found Default PCM: rates [0x70]: 32000 44100 48000 bits [0x2]: 16 formats [0x1]: PCM ``` for the other (default) audio card: `cat /proc/asound/card1/pcm0p/info` (there is pcm0c file too but I did not check it) gives: `name: ALC294 Analog` `cat /proc/asound/card1/codec#0` on top: ``` Codec: Realtek ALC294 Address: 0 AFG Function Id: 0x1 (unsol 1) Vendor Id: 0x10ec0294 Subsystem Id: 0x10431ab2 Revision Id: 0x100004 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM ``` so as you can see: - hdmi supports up to 48000 - analog supports up to 192000 I first noticed there is something wrong with alsacap program that reports: ``` *** Scanning for playback devices *** Card 0, ID `Generic', name `HD-Audio Generic' Device 3, ID `HDMI 0', name `HDMI 0', 1 subdevices (1 available) 2..8 channels, sampling rate 32000..192000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' Card 1, ID `Generic_1', name `HD-Audio Generic' Device 0, ID `ALC294 Analog', name `ALC294 Analog', 1 subdevices (1 available) 2 channels, sampling rate 44100..48000 Hz Sample formats: S16_LE, S32_LE Subdevice 0, name `subdevice #0' ``` As you can see: - hdmi supports up to 192000 - analog supports up to 48000 so it's... flipped? here is a output from `cat /proc/asound/pcm`: ``` 00-03: HDMI 0 : HDMI 0 : playback 1 01-00: ALC294 Analog : ALC294 Analog : playback 1 : capture 1 ``` To confirm this I downloaded flac file with 96000 sample rate and tried to play it with ffmpeg after pipewire,pipewire-pulse,wireplumber services are stopped. I confirmed that these programs are disabled by checking if `ps ax | grep wire` before and after, also I monitored logs from these services. It reported error for device "hw:1,0" because it could not set sampling rate to 96000. It did not fail on "hw:0,3" but of course I did not hear anything since external monitor is disconnected. Replay command: `ffmpeg -i yuri_korzunov-through-moon_ring-96kHz-24bit.flac -f alsa hw:1,0` the file is some random file found in internet that tested with `ffprobe` shows that it has sample rate of 96000. I don't want to attach it since I forgot where I found it ;) I don't have `asound.conf` files in `/etc/` or `~/.config`. What else can I do to discover the problem? Issue URL : https://github.com/alsa-project/alsa-lib/issues/432 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Add Roland QUAD-CAPTURE USB 2.0 support
by GitHub pull_request - opened 31 Dec '24

31 Dec '24

alsa-project/alsa-ucm-conf pull request #494 was opened from bjornmp: Add Roland QUAD-CAPTURE USB 2.0 support Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/494 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/494.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

alsa-lib-1.2.13 is breaking dssi-1.1.1 because of missing header
by GitHub issues - opened 30 Dec '24

30 Dec '24

alsa-project/alsa-lib issue #431 was opened from kobboi: What was working with 1.2.12 no longer isn't: ``` /usr/include/alsa/ump_msg.h:1119:65: error: unknown type name ‘size_t’ 1119 | int snd_ump_msg_sysex_expand(const uint32_t *ump, uint8_t *buf, size_t maxlen, | ^~~~~~ /usr/include/alsa/ump_msg.h:13:1: note: ‘size_t’ is defined in header ‘<stddef.h>’; did you forget to ‘#include <stddef.h>’? 12 | #include <stdint.h> +++ |+#include <stddef.h> 13 | /usr/include/alsa/ump_msg.h:1120:30: error: unknown type name ‘size_t’ 1120 | size_t *filled); | ^~~~~~ /usr/include/alsa/ump_msg.h:1120:30: note: ‘size_t’ is defined in header ‘<stddef.h>’; did you forget to ‘#include <stddef.h>’? ``` Can this be addressed? Happy holidays! Issue URL : https://github.com/alsa-project/alsa-lib/issues/431 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v1] MAINTAINERS: update entries in TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) DRIVERS and add entries for haptic driver
by Shenghao Ding 30 Dec '24

30 Dec '24

Due to internal re-org, Kevin has no longer maintained audio driver. With the job responsibility change, drop entries for the audio converter and add entries for both haptics drivers and middle-power audio amplifier drivers. Add audio converter, set the Status as Supported. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- MAINTAINERS | 32 ++++++++++++++++++++++++-------- 1 file changed, 24 insertions(+), 8 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 910305c11e8a..db50e677c9ca 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -23170,25 +23170,42 @@ S: Maintained F: Documentation/devicetree/bindings/sound/davinci-mcasp-audio.yaml F: sound/soc/ti/ -TEXAS INSTRUMENTS AUDIO (ASoC/HDA) DRIVERS +TEXAS INSTRUMENTS LOW/MIDDLE-POWER AUDIO AMPLIFIER (ASoC/HDA) & HAPTICS DRIVERS M: Shenghao Ding <shenghao-ding(a)ti.com> -M: Kevin Lu <kevin-lu(a)ti.com> M: Baojun Xu <baojun.xu(a)ti.com> L: linux-sound(a)vger.kernel.org -S: Maintained +S: Supported +F: Documentation/devicetree/bindings/input/ti,drv260x.yaml +F: Documentation/devicetree/bindings/input/ti,drv266x.yaml F: Documentation/devicetree/bindings/sound/tas2552.txt +F: Documentation/devicetree/bindings/sound/tas5720.txt F: Documentation/devicetree/bindings/sound/ti,tas2562.yaml F: Documentation/devicetree/bindings/sound/ti,tas2770.yaml +F: Documentation/devicetree/bindings/sound/ti,tas2781.yaml F: Documentation/devicetree/bindings/sound/ti,tas27xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5086.txt +F: Documentation/devicetree/bindings/sound/ti,tas57xx.yaml +F: Documentation/devicetree/bindings/sound/ti,tas5805m.yaml F: Documentation/devicetree/bindings/sound/ti,tpa6130a2.yaml +F: drivers/input/misc/drv2*.c +F: include/sound/tas2*.h +F: include/sound/tas5086.h +F: include/sound/tpa6130a2-plat.h +F: sound/pci/hda/tas2781_hda_i2c.c +F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/tas5*.* +F: sound/soc/codecs/tpa6130a2.* + +TEXAS INSTRUMENTS AUDIO Converter (ASoC) and Touch Screen DRIVERS +L: linux-sound(a)vger.kernel.org +S: Maintained F: Documentation/devicetree/bindings/sound/ti,pcm1681.yaml F: Documentation/devicetree/bindings/sound/ti,pcm3168a.yaml +F: Documentation/devicetree/bindings/sound/ti,pcm6240.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320*.yaml F: Documentation/devicetree/bindings/sound/ti,tlv320adcx140.yaml -F: include/sound/tas2*.h +F: drivers/input/touchscreen/tsc2*.* F: include/sound/tlv320*.h -F: include/sound/tpa6130a2-plat.h -F: sound/pci/hda/tas2781_hda_i2c.c F: sound/soc/codecs/pcm1681.c F: sound/soc/codecs/pcm1789*.* F: sound/soc/codecs/pcm179x*.* @@ -23198,9 +23215,8 @@ F: sound/soc/codecs/pcm3060*.* F: sound/soc/codecs/pcm3168a*.* F: sound/soc/codecs/pcm5102a.c F: sound/soc/codecs/pcm512x*.* -F: sound/soc/codecs/tas2*.* +F: sound/soc/codecs/pcm6240.* F: sound/soc/codecs/tlv320*.* -F: sound/soc/codecs/tpa6130a2.* TEXAS INSTRUMENTS DMA DRIVERS M: Peter Ujfalusi <peter.ujfalusi(a)gmail.com> -- 2.34.1

1 0

UCM2 macro breaks on updates for Motu 828
by GitHub issues - opened 30 Dec '24

30 Dec '24

alsa-project/alsa-ucm-conf issue #493 was opened from beatboxa: It appears that every time there is an update, my UCM profile breaks. So that when I boot my machine, I have no sound (unless I manually map things in a pro audio application). This appears to be due to an issue in the SplitPCM macro--specifically, some parsing. Here is the thread describing the creation of the UCM profile I am using, including a patch that I had to apply in order to get things working: #416 And specifically, here is the patch I apply in order to fix it: https://github.com/alsa-project/alsa-ucm-conf/commit/b68aa52acdd2763fedad5e… It involves adding a single hyphen character to the file: ucm2/common/pcm/split.conf Because it is a single character, I have been manually changing the file. It's hard for me to follow that thread--but is this fixed and I am on older versions, or is this still an open item? Or is there a way to automate this so that my audio doesn't break on every update? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/493 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[RESEND PATCH v2] drm/bridge: dw-hdmi-i2s: set insert_pcuv bit if hardware supports it
by Geraldo Nascimento 30 Dec '24

30 Dec '24

Hi Mark, resending this as it failed to apply in my last submission. Added Neil Armstrong to Cc: as hopefully he will be able to better review this. Thanks, Geraldo Nascimento --- Starting with version 2.10a of Synopsys DesignWare HDMI controller the insert_pcuv bit was introduced. On RK3399pro SoM (Radxa Rock Pi N10), for example, if we neglect to set this bit and proceed to enable hdmi_sound and i2s2 on the device tree there will be extreme clipping of sound output, to the point that music sounds like white noise. Problem could also manifest as just mild cracking depending of HDMI audio implementation of sink. Setting insert_pcuv bit (bit 2 of aud_conf2 Audio Sample register) fixes this. Signed-off-by: Geraldo Nascimento <geraldogabriel(a)gmail.com> --- v1->v2: SoC->SoM on description, better commenting, minor style changes, conditional application of fix for L-PCM only --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio-20221017.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-i2s-audio-20221017.c @@ -42,6 +42,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, struct dw_hdmi *hdmi = audio->hdmi; u8 conf0 = 0; u8 conf1 = 0; + u8 conf2 = 0; u8 inputclkfs = 0; /* it cares I2S only */ @@ -101,6 +102,28 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, return -EINVAL; } + /* + * dw-hdmi introduced insert_pcuv bit in + * version 2.10a. + * + * This single bit (bit 2 of HDMI_AUD_CONF2) + * when set to 1 will enable the insertion of the PCUV + * (Parity, Channel Status, User bit and Validity) + * bits on the incoming audio stream. + * + * Support is limited to Linear PCM audio. If + * neglected, the lack of valid PCUV bits + * on L-PCM streams will cause anything from + * mild cracking to full blown extreme + * clipping depending on the HDMI audio + * implementation of the sink. + * + */ + + if (hdmi_read(audio, HDMI_DESIGN_ID) >= 0x21 && + !(hparms->iec.status[0] & IEC958_AES0_NONAUDIO)) + conf2 = HDMI_AUD_CONF2_INSERT_PCUV; + dw_hdmi_set_sample_rate(hdmi, hparms->sample_rate); dw_hdmi_set_channel_status(hdmi, hparms->iec.status); dw_hdmi_set_channel_count(hdmi, hparms->channels); @@ -109,6 +120,7 @@ static int dw_hdmi_i2s_hw_params(struct device *dev, void *data, hdmi_write(audio, inputclkfs, HDMI_AUD_INPUTCLKFS); hdmi_write(audio, conf0, HDMI_AUD_CONF0); hdmi_write(audio, conf1, HDMI_AUD_CONF1); + hdmi_write(audio, conf2, HDMI_AUD_CONF2); return 0; } --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi-20221017.h +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi-20221017.h @@ -931,6 +931,11 @@ enum { HDMI_AUD_CONF1_WIDTH_16 = 0x10, HDMI_AUD_CONF1_WIDTH_24 = 0x18, +/* AUD_CONF2 field values */ + HDMI_AUD_CONF2_HBR = 0x01, + HDMI_AUD_CONF2_NLPCM = 0x02, + HDMI_AUD_CONF2_INSERT_PCUV = 0x04, + /* AUD_CTS3 field values */ HDMI_AUD_CTS3_N_SHIFT_OFFSET = 5, HDMI_AUD_CTS3_N_SHIFT_MASK = 0xe0,

3 6

[PATCH v4 0/9] ASoC: samsung: midas-audio: Add GPIO-based headset jack detection
by Artur Weber 29 Dec '24

29 Dec '24

Many of Samsung's Exynos 4 devices share the same midas-audio driver to handle the codec setup. While most of these devices, including the Midas itself, use the jack detection provided by the WM8994 driver, other devices such as the Samsung Galaxy Tab 3 8.0 (lt01) use two GPIOs and an ADC channel to determine jack insertion, the jack's type, and button presses (for headsets with volume up/down/play buttons). In the downstream kernel, this behavior is implemented in the sec-jack driver[1], and the per-device settings are configured in *-jack.c files in the mach folder (see e.g. the Tab 3's implementation[2]). This patchset implements this mechanism in the midas_wm1811.c driver, and adds new DTS options to allow for its configuration. It also enables jack detection for the Samsung Galaxy Tab 3 8.0. A very similar mechanism was already present in the aries_wm8994.c driver[3]; this implementation heavily borrows from it, though there are a few extra cleanups as well. Signed-off-by: Artur Weber <aweber.kernel(a)gmail.com> [1] https://github.com/gr8nole/android_kernel_samsung_smdk4x12/blob/lineage-14.… [2] https://github.com/gr8nole/android_kernel_samsung_smdk4x12/blob/lineage-14.… [3] https://github.com/torvalds/linux/blob/master/sound/soc/samsung/aries_wm899… --- Changes in v4: - Replaced manual regulator toggles with SND_SOC_DAPM_REGULATOR_SUPPLY, both for existing mic bias regulators and for the new headset mic bias regulator - Link to v3: https://lore.kernel.org/r/20240519-midas-wm1811-gpio-jack-v3-0-0c1736144c0e… Changes in v3: - Re-added pipe (|) character to samsung,headset-button-threshold-microvolt description to prevent it from being parsed as a mapping (fixes syntax error) - Split out "ASoC: dt-bindings: samsung,midas-audio: Add GPIO-based headset jack detection" into two patches - Link to v2: https://lore.kernel.org/r/20240508-midas-wm1811-gpio-jack-v2-0-b4d36cd02c6e… Changes in v2: - Added vendor prefix to threshold properties - Added separate headset mic bias regulator - Changed some cases of dev_err + return with return dev_err_probe - Added an extra patch to replace some previous dev_err + return cases with dev_err_probe - Moved tab3 DTS wm1811 codec config changes to separate commit --- Artur Weber (9): ASoC: dt-bindings: samsung,midas-audio: Add headset mic bias supply ASoC: dt-bindings: samsung,midas-audio: Add GPIO-based headset jack detection ASoC: samsung: midas_wm1811: Use SND_SOC_DAPM_REGULATOR_SUPPLY for bias regulators ASoC: samsung: midas_wm1811: Add headset mic bias supply support ASoC: samsung: midas_wm1811: Add GPIO-based headset jack detection ASoC: samsung: midas_wm1811: Use dev_err_probe where appropriate ARM: dts: samsung: exynos4212-tab3: Fix headset mic, add jack detection ARM: dts: samsung: exynos4212-tab3: Add MCLK2 clock to WM1811 codec config ARM: dts: samsung: exynos4212-tab3: Drop interrupt from WM1811 codec .../bindings/sound/samsung,midas-audio.yaml | 33 ++ arch/arm/boot/dts/samsung/exynos4212-tab3.dtsi | 31 +- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/midas_wm1811.c | 348 +++++++++++++++++---- 4 files changed, 343 insertions(+), 71 deletions(-) --- base-commit: a38297e3fb012ddfa7ce0321a7e5a8daeb1872b6 change-id: 20240502-midas-wm1811-gpio-jack-b10226b17ecc Best regards, -- Artur Weber <aweber.kernel(a)gmail.com>

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USB-Audio: ALC4080 - add ASUS ROG Crosshair X870E Hero (USB 0b05:1b7c)
by GitHub pull_request - opened 26 Dec '24

26 Dec '24

alsa-project/alsa-ucm-conf pull request #492 was opened from jetm: https://rog.asus.com/motherboards/rog-crosshair/rog-crosshair-x870e-hero/ Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/492 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/492.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: Bug: slab-out-of-bounds in snd_seq_oss_synth_sysex
by Kun Hu 26 Dec '24

26 Dec '24

>> BUG: KASAN: slab-out-of-bounds in snd_seq_oss_synth_sysex+0x5d1/0x6c0 sound/core/seq/oss/seq_oss_synth.c:516 > > We further analyzed the issue at line 516 in ./sound/core/seq/oss/seq_oss_synth.c. > The slab-out-of-bounds crash occurs in line 509, when sysex->len = 128. Specifically, the write operation to dest[0] accesses memory beyond the bounds of sysex->buf (128 byte). > To resolve this issue, we suggest adding 6 lines of code to validate the legality of the address write to sysex->buf before entering the loop: > > if (sysex->len >= MAX_SYSEX_BUFLEN) { > sysex->len = 0; > sysex->skip = 1; > return -EINVAL; /* Exit early if sysex->len is out of bounds */ > } If you need any more information, please don't hesitate to let me know. ————— Best regards, Kun Hu > 2024年12月25日 13:37，Kun Hu <huk23(a)m.fudan.edu.cn> 写道： > > Hello, > >> BUG: KASAN: slab-out-of-bounds in snd_seq_oss_synth_sysex+0x5d1/0x6c0 sound/core/seq/oss/seq_oss_synth.c:516 > > We further analyzed the issue at line 516 in ./sound/core/seq/oss/seq_oss_synth.c. > The slab-out-of-bounds crash occurs in line 509, when sysex->len = 128. Specifically, the write operation to dest[0] accesses memory beyond the bounds of sysex->buf (128 byte). > To resolve this issue, we suggest adding 6 lines of code to validate the legality of the address write to sysex->buf before entering the loop: > > if (sysex->len >= MAX_SYSEX_BUFLEN) { > sysex->len = 0; > sysex->skip = 1; > return -EINVAL; /* Exit early if sysex->len is out of bounds */ > } > > If you fix this issue, please add the following tag to the commit: > Reported-by: Kun Hu <huk23(a)m.fudan.edu.cn> > > ————— > Thanks, > Kun Hu > >> 2024年12月24日 19:16，Kun Hu <huk23(a)m.fudan.edu.cn> 写道： >> >> Hello, >> >> When using fuzzer tool to fuzz the latest Linux kernel, the following crash >> was triggered. >> >> HEAD commit: 78d4f34e2115b517bcbfe7ec0d018bbbb6f9b0b8 >> git tree: upstream >> Console output:https://drive.google.com/file/d/17oCyKDW_kNhSW5Bbvm23vnpD1eo0MHFi/vi… >> Kernel config: https://drive.google.com/file/d/1RhT5dFTs6Vx1U71PbpenN7TPtnPoa3NI/view?usp=… >> C reproducer: https://drive.google.com/file/d/177HJht6a7-6F3YLudKb_d4kiPGd1VA_i/view?usp=… >> Syzlang reproducer: https://drive.google.com/file/d/1AuP5UGGc47rEXXPuvjmCKgJ3d0U1P84j/view?usp=… >> >> >> If you fix this issue, please add the following tag to the commit: >> Reported-by: Kun Hu <huk23(a)m.fudan.edu.cn> >> >> ================================================================== >> BUG: KASAN: slab-out-of-bounds in snd_seq_oss_synth_sysex+0x5d1/0x6c0 sound/core/seq/oss/seq_oss_synth.c:516 >> Write of size 1 at addr ff1100000588e288 by task syz-executor411/824 >> >> CPU: 2 UID: 0 PID: 824 Comm: syz-executor411 Not tainted 6.13.0-rc3 #5 >> Hardware name: QEMU Standard PC (i440FX + PIIX, 1996), BIOS 1.13.0-1ubuntu1.1 04/01/2014 >> Call Trace: >> <TASK> >> __dump_stack lib/dump_stack.c:94 [inline] >> dump_stack_lvl+0x116/0x1b0 lib/dump_stack.c:120 >> print_address_description mm/kasan/report.c:378 [inline] >> print_report+0xcf/0x5f0 mm/kasan/report.c:489 >> kasan_report+0x93/0xc0 mm/kasan/report.c:602 >> snd_seq_oss_synth_sysex+0x5d1/0x6c0 sound/core/seq/oss/seq_oss_synth.c:516 >> snd_seq_oss_process_event+0x46a/0x2620 sound/core/seq/oss/seq_oss_event.c:61 >> insert_queue sound/core/seq/oss/seq_oss_rw.c:167 [inline] >> snd_seq_oss_write+0x261/0x7f0 sound/core/seq/oss/seq_oss_rw.c:135 >> odev_write+0x53/0xa0 sound/core/seq/oss/seq_oss.c:168 >> vfs_write fs/read_write.c:677 [inline] >> vfs_write+0x2e3/0x10f0 fs/read_write.c:659 >> ksys_write+0x122/0x240 fs/read_write.c:731 >> do_syscall_x64 arch/x86/entry/common.c:52 [inline] >> do_syscall_64+0xc3/0x1d0 arch/x86/entry/common.c:83 >> entry_SYSCALL_64_after_hwframe+0x77/0x7f >> RIP: 0033:0x7f69b379994d >> Code: c3 e8 97 2a 00 00 0f 1f 80 00 00 00 00 f3 0f 1e fa 48 89 f8 48 89 f7 48 89 d6 48 89 ca 4d 89 c2 4d 89 c8 4c 8b 4c 24 08 0f 05 <48> 3d 01 f0 ff ff 73 01 c3 48 c7 c1 b8 ff ff ff f7 d8 64 89 01 48 >> RSP: 002b:00007f69b3718d78 EFLAGS: 00000246 ORIG_RAX: 0000000000000001 >> RAX: ffffffffffffffda RBX: 00007f69b382d2d0 RCX: 00007f69b379994d >> RDX: 0000000000000008 RSI: 0000000020000140 RDI: 0000000000000003 >> RBP: 00007f69b382d2d8 R08: 0000000000000000 R09: 0000000000000000 >> R10: 0000000000000000 R11: 0000000000000246 R12: 00007f69b382d2dc >> R13: 0000000020000140 R14: 00007f69b37fa008 R15: 000000000000000d >> </TASK> >> >> Allocated by task 823: >> kasan_save_stack+0x24/0x50 mm/kasan/common.c:47 >> kasan_save_track+0x14/0x30 mm/kasan/common.c:68 >> poison_kmalloc_redzone mm/kasan/common.c:377 [inline] >> __kasan_kmalloc+0xaa/0xb0 mm/kasan/common.c:394 >> kmalloc_noprof include/linux/slab.h:901 [inline] >> kzalloc_noprof include/linux/slab.h:1037 [inline] >> snd_seq_oss_synth_sysex+0x1d9/0x6c0 sound/core/seq/oss/seq_oss_synth.c:502 >> snd_seq_oss_process_event+0x46a/0x2620 sound/core/seq/oss/seq_oss_event.c:61 >> insert_queue sound/core/seq/oss/seq_oss_rw.c:167 [inline] >> snd_seq_oss_write+0x261/0x7f0 sound/core/seq/oss/seq_oss_rw.c:135 >> odev_write+0x53/0xa0 sound/core/seq/oss/seq_oss.c:168 >> vfs_write fs/read_write.c:677 [inline] >> vfs_write+0x2e3/0x10f0 fs/read_write.c:659 >> ksys_write+0x122/0x240 fs/read_write.c:731 >> do_syscall_x64 arch/x86/entry/common.c:52 [inline] >> do_syscall_64+0xc3/0x1d0 arch/x86/entry/common.c:83 >> entry_SYSCALL_64_after_hwframe+0x77/0x7f >> >> The buggy address belongs to the object at ff1100000588e200 >> which belongs to the cache kmalloc-192 of size 192 >> The buggy address is located 0 bytes to the right of >> allocated 136-byte region [ff1100000588e200, ff1100000588e288) >> >> The buggy address belongs to the physical page: >> page: refcount:1 mapcount:0 mapping:0000000000000000 index:0x0 pfn:0x588e >> anon flags: 0x100000000000000(node=0|zone=1) >> page_type: f5(slab) >> raw: 0100000000000000 ff1100000103c3c0 ffd4000000162340 dead000000000003 >> raw: 0000000000000000 0000000080100010 00000001f5000000 0000000000000000 >> page dumped because: kasan: bad access detected >> >> Memory state around the buggy address: >> ff1100000588e180: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc >> ff1100000588e200: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 >>> ff1100000588e280: 00 fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc >> ^ >> ff1100000588e300: fa fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb >> ff1100000588e380: fb fb fb fb fb fb fb fb fc fc fc fc fc fc fc fc >> ================================================================== >> >> --------------- >> thanks, >> Kun Hu > >

1 0

[PATCH 00/21] ASoC: amd: acp70: add soundwire and acp pdm support
by Vijendar Mukunda 25 Dec '24

25 Dec '24

This patch series includes below changes - Add Audio IO support for ACP7.0 and ACP7.1 platforms for SoundWire IO and ACP PDM controller combination. - Add common driver modules for ACP7.0 and ACP7.1 platforms (acp pci driver, Soundwire dma driver, pdm platform driver). - Add SoundWire generic machine driver changes for legacy stack (No DSP enabled) for ACP7.0 & ACP7.1 platforms. - Add SoundWire machines for ACP7.0 & ACP7.1 platforms. Vijendar Mukunda (21): ASoC: amd: add register header file for ACP7.0 platform ASoC: amd: acp70: add acp pci driver for ACP7.0 and ACP7.1 platforms ASoC: amd: acp70: add acp init and de-init functions ASoC: amd: acp70: add logic for scanning acp child devices ASoC: amd: acp70: create platform devices for acp child nodes ASoC: amd: acp70: enable driver build for ACP7.0 platform ASoC: amd: acp70: add acp pdm platform driver ASoC: amd: acp70: add acp pdm driver dma ops and dai ops ASoC: amd: acp70: add acp soundwire dma driver ASoC: amd: update ACP7.0 KConfig option description ASoC: amd: acp70: add soundwire dma driver dma ops ASoC: amd: acp70: add acp ip interrupt handler ASoC: amd: acp70: add acp pdm driver pm ops ASoC: amd: acp70: add pm ops support for soundwire dma driver ASoC: amd: acp70: add acp driver pm ops support ASoC: amd: acp70: enable wake capability for acp pci driver ASoC: amd: acp70: add soundwire host wake interrupt handling ASoC: amd: acp70: create a device node for soundwire machine driver ASoC: amd: acp: add machine driver changes for ACP7.0 and ACP7.1 platforms ASoC: amd: acp: add RT711, RT714 & RT1316 support for ACP7.0 platform ASoC: amd: acp: amd-acp70-acpi-match: Add rt722 support include/sound/acp70_chip_offset_byte.h | 484 ++++++++++++++ sound/soc/amd/Kconfig | 12 + sound/soc/amd/Makefile | 1 + sound/soc/amd/acp/Makefile | 2 +- sound/soc/amd/acp/acp-sdw-legacy-mach.c | 16 + sound/soc/amd/acp/acp-sdw-mach-common.c | 34 + sound/soc/amd/acp/amd-acp70-acpi-match.c | 132 ++++ sound/soc/amd/acp/soc_amd_sdw_common.h | 11 + sound/soc/amd/acp70/Makefile | 9 + sound/soc/amd/acp70/acp70-pdm-dma.c | 463 +++++++++++++ sound/soc/amd/acp70/acp70-sdw-dma.c | 586 ++++++++++++++++ sound/soc/amd/acp70/acp70.h | 283 ++++++++ sound/soc/amd/acp70/pci-acp70.c | 807 +++++++++++++++++++++++ sound/soc/amd/mach-config.h | 1 + 14 files changed, 2840 insertions(+), 1 deletion(-) create mode 100644 include/sound/acp70_chip_offset_byte.h create mode 100644 sound/soc/amd/acp/amd-acp70-acpi-match.c create mode 100644 sound/soc/amd/acp70/Makefile create mode 100644 sound/soc/amd/acp70/acp70-pdm-dma.c create mode 100644 sound/soc/amd/acp70/acp70-sdw-dma.c create mode 100644 sound/soc/amd/acp70/acp70.h create mode 100644 sound/soc/amd/acp70/pci-acp70.c -- 2.34.1

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Acer 315-24P Microphone indicator is not working
by GitHub issues - edited 23 Dec '24

23 Dec '24

alsa-project/alsa-utils issue #284 was edited from River000000: ![изображение](https://github.com/user-attachments/assets/dc8b0b9a-f877-4edc-a9eb-d1c4d1d2fcb7) ![изображение](https://github.com/user-attachments/assets/5f8f1c69-294b-4833-8bdf-9c2f8033c379) alsa-info: [alsa.txt](https://github.com/user-attachments/files/18227950/alsa.txt) Issue URL : https://github.com/alsa-project/alsa-utils/issues/284 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Acer 315-24P Microphone indicator is not working
by GitHub issues - edited 23 Dec '24

23 Dec '24

alsa-project/alsa-utils issue #284 was edited from River000000: ![изображение](https://github.com/user-attachments/assets/dc8b0b9a-f877-4edc-a9eb-d1c4d1d2fcb7) ![изображение](https://github.com/user-attachments/assets/5f8f1c69-294b-4833-8bdf-9c2f8033c379) [alsa.txt](https://github.com/user-attachments/files/18227950/alsa.txt) Issue URL : https://github.com/alsa-project/alsa-utils/issues/284 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

Acer 315-24P Microphone indicator is not working
by GitHub issues - opened 23 Dec '24

23 Dec '24

alsa-project/alsa-utils issue #284 was opened from River000000: [alsa.txt](https://github.com/user-attachments/files/18227950/alsa.txt) ![изображение](https://github.com/user-attachments/assets/dc8b0b9a-f877-4edc-a9eb-d1c4d1d2fcb7) ![изображение](https://github.com/user-attachments/assets/5f8f1c69-294b-4833-8bdf-9c2f8033c379) Issue URL : https://github.com/alsa-project/alsa-utils/issues/284 Repository URL: https://github.com/alsa-project/alsa-utils

1 0

[PATCH RESEND] soundwire: amd: clear wake enable register for power off mode
by Vijendar Mukunda 23 Dec '24

23 Dec '24

As per design for power off mode, clear the wake enable register during resume sequence. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 5a4bfaef65fb..96a3aa6da711 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -1190,6 +1190,7 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { return amd_sdw_clock_stop_exit(amd_manager); } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { + writel(0x00, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); val = readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); if (val) { val |= AMD_SDW_CLK_RESUME_REQ; -- 2.34.1

3 3

[PATCH] soundwire: amd: clear wake enable register for power off mode
by Vijendar Mukunda 23 Dec '24

23 Dec '24

As per design for power off mode, clear the wake enable register during resume sequence. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 5a4bfaef65fb..96a3aa6da711 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -1190,6 +1190,7 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { return amd_sdw_clock_stop_exit(amd_manager); } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { + writel(0x00, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); val = readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); if (val) { val |= AMD_SDW_CLK_RESUME_REQ; -- 2.34.1

3 2

[PATCH V2] soundwire: amd: clear wake enable register for power off mode
by Vijendar Mukunda 23 Dec '24

23 Dec '24

As per design for AMD SoundWire manager power off mode, clear the wake enable register during resume sequence. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- Changes Since v1: - Replaced 0x00 with 0 in register write - Updte Commit message drivers/soundwire/amd_manager.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 5a4bfaef65fb..ca271a728a65 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -1190,6 +1190,7 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { return amd_sdw_clock_stop_exit(amd_manager); } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { + writel(0, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); val = readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); if (val) { val |= AMD_SDW_CLK_RESUME_REQ; -- 2.34.1

1 0

[PATCH RESEND] soundwire: amd: clear wake enable register for
by Vijendar Mukunda 23 Dec '24

23 Dec '24

As per design for power off mode, clear the wake enable register during resume sequence. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- drivers/soundwire/amd_manager.c | 1 + 1 file changed, 1 insertion(+) diff --git a/drivers/soundwire/amd_manager.c b/drivers/soundwire/amd_manager.c index 5a4bfaef65fb..96a3aa6da711 100644 --- a/drivers/soundwire/amd_manager.c +++ b/drivers/soundwire/amd_manager.c @@ -1190,6 +1190,7 @@ static int __maybe_unused amd_resume_runtime(struct device *dev) if (amd_manager->power_mode_mask & AMD_SDW_CLK_STOP_MODE) { return amd_sdw_clock_stop_exit(amd_manager); } else if (amd_manager->power_mode_mask & AMD_SDW_POWER_OFF_MODE) { + writel(0x00, amd_manager->acp_mmio + ACP_SW_WAKE_EN(amd_manager->instance)); val = readl(amd_manager->mmio + ACP_SW_CLK_RESUME_CTRL); if (val) { val |= AMD_SDW_CLK_RESUME_REQ; -- 2.34.1

3 3

make check fails to compile playmidi1.c: "error: expected identifier or '(' before numeric constant"
by GitHub issues - opened 22 Dec '24

22 Dec '24

alsa-project/alsa-lib issue #430 was opened from 746B61: Getting the latest version from branch master (352cbc5e) and running `./gitcompile` followed by `make check` fails with: ``` Making check in test make[1]: Entering directory '/tmp/alsa-lib-master/test' Making check in . make[2]: Entering directory '/tmp/alsa-lib-master/test' make control pcm pcm_min latency seq seq-ump-example playmidi1 timer rawmidi midiloop umpinfo oldapi queue_timer namehint client_event_filter chmap audio_time user-ctl-element-set pcm-multi-thread make[3]: Entering directory '/tmp/alsa-lib-master/test' CC control.o CCLD control CC pcm.o CCLD pcm CC pcm_min.o CCLD pcm_min CC latency.o CCLD latency CC seq.o In file included from seq.c:12: seq-sender.c: In function 'event_sender': seq-sender.c:150:62: warning: variable 'pcm_flag' set but not used [-Wunused-but-set-variable] 150 | int client, port, queue, max, err, v1, v2, time = 0, pcm_flag = 0; | ^~~~~~~~ CCLD seq CC seq-ump-example.o CCLD seq-ump-example CC playmidi1.o In file included from playmidi1.c:48: midifile.h:55:33: error: expected identifier or '(' before numeric constant 55 | #define note_on 0x90 | ^~~~ ../include/alsa/ump_msg.h:170:41: note: in expansion of macro 'note_on' 170 | snd_ump_msg_midi1_note_t note_on; /**< MIDI1 note-on message */ | ^~~~~~~ midifile.h:54:33: error: expected identifier or '(' before numeric constant 54 | #define note_off 0x80 | ^~~~ ../include/alsa/ump_msg.h:171:41: note: in expansion of macro 'note_off' 171 | snd_ump_msg_midi1_note_t note_off; /**< MIDI1 note-off message */ | ^~~~~~~~ midifile.h:57:33: error: expected identifier or '(' before numeric constant 57 | #define control_change 0xb0 | ^~~~ ../include/alsa/ump_msg.h:173:41: note: in expansion of macro 'control_change' 173 | snd_ump_msg_midi1_cc_t control_change; /**< MIDI1 control-change message */ | ^~~~~~~~~~~~~~ midifile.h:55:33: error: expected identifier or '(' before numeric constant 55 | #define note_on 0x90 | ^~~~ ../include/alsa/ump_msg.h:432:41: note: in expansion of macro 'note_on' 432 | snd_ump_msg_midi2_note_t note_on; /**< MIDI2 note-on message */ | ^~~~~~~ midifile.h:54:33: error: expected identifier or '(' before numeric constant 54 | #define note_off 0x80 | ^~~~ ../include/alsa/ump_msg.h:433:41: note: in expansion of macro 'note_off' 433 | snd_ump_msg_midi2_note_t note_off; /**< MIDI2 note-off message */ | ^~~~~~~~ midifile.h:57:33: error: expected identifier or '(' before numeric constant 57 | #define control_change 0xb0 | ^~~~ ../include/alsa/ump_msg.h:438:41: note: in expansion of macro 'control_change' 438 | snd_ump_msg_midi2_cc_t control_change; /**< MIDI2 control-change message */ | ^~~~~~~~~~~~~~ In file included from playmidi1.c:49: midifile.c:57:33: error: expected identifier or '(' before numeric constant 57 | #define set_tempo 0x51 | ^~~~ ../include/alsa/ump_msg.h:711:41: note: in expansion of macro 'set_tempo' 711 | snd_ump_msg_set_tempo_t set_tempo; /**< Set Tempo */ | ^~~~~~~~~ make[3]: *** [Makefile:622: playmidi1.o] Error 1 make[3]: Leaving directory '/tmp/alsa-lib-master/test' make[2]: *** [Makefile:819: check-am] Error 2 make[2]: Leaving directory '/tmp/alsa-lib-master/test' make[1]: *** [Makefile:669: check-recursive] Error 1 make[1]: Leaving directory '/tmp/alsa-lib-master/test' make: *** [Makefile:414: check-recursive] Error 1 ``` The issue was introduced with commit 07cee0ba. playmidi1.c first [includes](https://github.com/alsa-project/alsa-lib/blob/352cbc5eb94a271a9c3… midifile.h from the test directory, which [has](https://github.com/alsa-project/alsa-lib/blob/352cbc5eb94a271a9c3c0ff5… `#define note_on 0x90`. After commit 07cee0ba, ump_msg.h gets included later, which [has](https://github.com/alsa-project/alsa-lib/blob/352cbc5eb94a271a9c3c0ff5… ` snd_ump_msg_midi1_note_t note_on; /**< MIDI1 note-on message */`. The preprocessor replaces `note_on` with `0x90` and the compilation fails with the error above. Issue URL : https://github.com/alsa-project/alsa-lib/issues/430 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

sun4i-codec: add routing for headphones and internal speaker
by GitHub pull_request - opened 21 Dec '24

21 Dec '24

alsa-project/alsa-ucm-conf pull request #491 was opened from tokyovigilante: The sun4i-codec kernel driver covers a large number of Allwinner SoCs. The H616 codec has a single line-out route, which is used in concert with a Toshiba mux chip to send audio to either an internal speaker or headphone jack on a number of Anbernic handheld gaming devices. Add a UCM configuration to allow enabling/disabling the speaker amp depending on whether headphones are in use or not. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/491 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/491.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

API memory allocator
by GitHub issues - opened 20 Dec '24

20 Dec '24

alsa-project/alsa-lib issue #429 was opened from sylware: If I recall properly some data returned from the alsa lib API must be free using the C runtime free(). Would it be appropriate to move away from the C allocator to an API install-able memory allocator (like vulkan3D)? Backward compatibility would be achieved since the "default memory allocator" would be the C one, if the C runtime is around. Unless that part of the API is already deprecated (and I should not have used it). Issue URL : https://github.com/alsa-project/alsa-lib/issues/429 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH 00/35] Reorganize kerneldoc parameter names
by Julia Lawall 20 Dec '24

20 Dec '24

Reorganize kerneldoc parameter names to match the parameter order in the function header. The misordered cases were identified using the following Coccinelle semantic patch: // <smpl> @initialize:ocaml@ @@ let parse_doc l = let pieces = List.map String.trim (String.split_on_char '*' l) in let l = String.concat " " pieces in match String.split_on_char ':' l with x::xs -> x | _ -> "" let params ps = List.rev (List.fold_left (fun prev (pm,_) -> let ty = String.trim(Pretty_print_c.string_of_fullType pm.Ast_c.p_type) in if ty = "void" && pm.Ast_c.p_namei = None then prev else let name = match pm.Ast_c.p_namei with Some name -> name | None -> failwith "function parameter has no name" in (String.trim (Pretty_print_c.string_of_name name),ty)::prev) [] ps) @r@ comments c; identifier fn; position p; parameter list ps; type T; @@ T@c fn@p(ps) { ... } @script:ocaml@ p << r.p; c << r.c; (_,ps) << r.ps; @@ let isdoc c ps = List.length ps > 1 && (let c = String.trim c in String.length c > 3 && String.sub c 0 3 = "/**" && String.get c 3 != '*') in let subset l1 l2 = List.for_all (fun x -> List.mem x l2) l1 in let (cb,cm,ca) = List.hd c in match List.rev cb with c::_ when isdoc c ps -> let pieces = String.split_on_char '@' c in (match pieces with _::tl -> let d_names = List.map parse_doc tl in (* check parameters *) let p_names = List.map fst (params ps) in if d_names <> [] && not(d_names = p_names) then begin if List.sort compare d_names = List.sort compare p_names then Coccilib.print_main "out of order" p else if subset d_names p_names then Coccilib.print_main "doc is missing a parameter" p else if subset d_names p_names then Coccilib.print_main "doc has an extra parameter" p end | _ -> ()) | _ -> () // </smpl> --- arch/arm/mach-omap2/prm2xxx_3xxx.c | 1 - arch/powerpc/platforms/ps3/interrupt.c | 2 +- arch/powerpc/platforms/ps3/repository.c | 2 +- drivers/base/firmware_loader/main.c | 2 +- drivers/comedi/drivers/comedi_8254.c | 2 +- drivers/gpu/drm/amd/amdgpu/amdgpu_vm.c | 2 +- drivers/gpu/drm/amd/display/dc/core/dc.c | 2 +- drivers/gpu/drm/drm_gem_framebuffer_helper.c | 3 +-- drivers/gpu/drm/drm_gpuvm.c | 2 +- drivers/gpu/drm/radeon/radeon_ib.c | 2 +- drivers/iommu/iommu.c | 2 +- drivers/leds/leds-gpio-register.c | 2 +- drivers/mfd/atmel-smc.c | 4 ++-- drivers/misc/mei/bus.c | 2 +- drivers/mtd/ubi/eba.c | 2 +- drivers/net/ethernet/broadcom/bnxt/bnxt_hwrm.c | 2 +- drivers/net/ethernet/intel/e1000/e1000_hw.c | 2 +- drivers/net/ethernet/intel/i40e/i40e_common.c | 7 +++---- drivers/net/ethernet/intel/ice/ice_common.c | 2 +- drivers/net/ethernet/intel/ixgbe/ixgbe_common.c | 2 +- drivers/nvdimm/dimm_devs.c | 2 +- drivers/pci/hotplug/pci_hotplug_core.c | 2 +- drivers/pinctrl/pinmux.c | 2 +- drivers/slimbus/messaging.c | 2 +- drivers/soc/qcom/qmi_interface.c | 2 +- drivers/soundwire/stream.c | 2 +- drivers/usb/gadget/config.c | 4 ++-- fs/char_dev.c | 2 +- fs/dcache.c | 4 ++-- fs/seq_file.c | 2 +- kernel/audit.c | 2 +- kernel/resource.c | 2 +- kernel/sysctl.c | 1 - kernel/trace/ring_buffer.c | 2 +- lib/lru_cache.c | 2 +- lib/maple_tree.c | 2 +- mm/mmu_notifier.c | 2 +- net/dccp/feat.c | 2 +- net/mac80211/mesh_hwmp.c | 6 +++--- net/mac80211/mesh_pathtbl.c | 10 +++++----- net/socket.c | 2 +- net/sunrpc/xprt.c | 2 +- net/tipc/link.c | 14 +++++++------- net/tipc/msg.c | 2 +- sound/pci/hda/hda_codec.c | 2 +- 45 files changed, 60 insertions(+), 64 deletions(-)

4 5

[alsa-ucm-conf] sof-soundwire: Use correct speaker data port and PCM for cs35l56 bridge
by Maciej Strozek 20 Dec '24

20 Dec '24

Signed-off-by: Maciej Strozek <mstrozek(a)opensource.cirrus.com> --- ucm2/sof-soundwire/cs35l56-bridge.conf | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/ucm2/sof-soundwire/cs35l56-bridge.conf b/ucm2/sof-soundwire/cs35l56-bridge.conf index 176f880..2e8c1c6 100644 --- a/ucm2/sof-soundwire/cs35l56-bridge.conf +++ b/ucm2/sof-soundwire/cs35l56-bridge.conf @@ -23,8 +23,8 @@ SectionDevice."Speaker" { EnableSequence [ cset "name='cs42l43 Speaker L Input 1' 'ASPRX1'" cset "name='cs42l43 Speaker R Input 1' 'ASPRX2'" - cset "name='cs42l43 ASPTX1 Input' 'DP5RX1'" - cset "name='cs42l43 ASPTX2 Input' 'DP5RX2'" + cset "name='cs42l43 ASPTX1 Input' 'DP6RX1'" + cset "name='cs42l43 ASPTX2 Input' 'DP6RX2'" cset "name='AMPL ASP1 TX1 Source' 'DSP1TX1'" cset "name='AMPL ASP1 TX2 Source' 'None'" @@ -54,7 +54,7 @@ SectionDevice."Speaker" { Value { PlaybackPriority 100 - PlaybackPCM "hw:${CardId},0" + PlaybackPCM "hw:${CardId},2" PlaybackMixer "default:${CardId}" PlaybackMixerElem "Speaker" } -- 2.39.5

2 1

hda: how to implement component master_unbind?
by Kai Vehmanen 20 Dec '24

20 Dec '24

Hi Takashi et al, I've been having multiple discussions with our i915 team w.r.t. audio driver behaviour when the aggregate component is unbound, triggered by i915 unbind. This came up again in the recent regression with devres allocations and I now dug into the topic again. In short, I'm puzzled how to really implement this. ALSA (or ASoC) don't really have support for individual components of a card disappearing in a hotplug fashion. At card level, we do have such support (USB, firewire and recent work for PCI hotplug). But not individual components of a card getting unplugged. In current code we have this: static void hdac_component_master_unbind(struct device *dev) { » struct drm_audio_component *acomp = hdac_get_acomp(dev); » if (acomp->audio_ops && acomp->audio_ops->master_unbind) » » acomp->audio_ops->master_unbind(dev, acomp); » module_put(acomp->ops->owner); » component_unbind_all(dev, acomp); » WARN_ON(acomp->ops || acomp->dev); } ... when e.g. i915 driver is unbound (but could be any driver using the component framework, not jus Intel hw), i915 calls component_del() and HDA gets call to the above. After this, acomp ops are null are audio no longer has ability to talk to i915 driver (which makes sense given it's unbound). If audio was runtime suspended, this kind of works (but relies on some good luck). Upon HDA controller resume, we note acomp ops are NULL and HDMI/DP operations (like snd_hdac_display_power()) silently become no-ops. PCM streaming will go to /dev/null, but this is ok'ish since no monitor can be connected anyways. If audio was active, we start to get warnings or worse. Audio expects hw display codec to be powered and programmed, but suddenly it mey lose state. If the audio controller is actually part of the display hardware (e.g. discrete GPUs), then controller registers can stop functioning (they should be still available, but given the main diplay driver is unbound, odds of suprising behaviour are high). So this is less than ideal. I've been looking at options: 1) prevent/block the unbind if audio device is busy The component framework does not allow individual components to return -EBUSY, so there's no way to let others know that unbind is not possible now. This would force anyone testing DRM driver unbind to first unbind the audio driver and stop any active audio use-cases if needed. 2) unbind the ALSA card I've experimented doing a device_unregister() from the above callback, but this has not really worked (onion peeling exercise of new probelms). The code is shared by multiple controllers, so getting a handle to an snd_card object is not straighforward (drvdata is differnet for SOF and snd-hda-intel for instance). But with some new work, maybe we could call snd_card_disconnect() to detach the card and inform user-space. 3) continue as-is and try to fix bugs If audio is active, maybe we could force a acomp->put_power() to ensure a clean unregister of the display part. But this leaves a big chunk of issues especially with HDA controllers that require the display to be powered on, to function. .. Any ideas, and/or has there been prior work? It seems Takashi it's been mostly you who has been active working on the acomp integration recently. I also included Chris, Daniel and Jani who've had patches to hdac_component.c. Br, Kai

3 5

[GIT PULL] ASoC fixes for v6.13-rc3
by Mark Brown 20 Dec '24

20 Dec '24

The following changes since commit 78d4f34e2115b517bcbfe7ec0d018bbbb6f9b0b8: Linux 6.13-rc3 (2024-12-15 15:58:23 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.13-rc3 for you to fetch changes up to 32c9c06adb5b157ef259233775a063a43746d699: ASoC: mediatek: disable buffer pre-allocation (2024-12-19 11:15:09 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.13 A mix of quirks and small fixes, nothing too major anywhere. ---------------------------------------------------------------- Chen-Yu Tsai (2): ASoC: dt-bindings: realtek,rt5645: Fix CPVDD voltage comment ASoC: mediatek: disable buffer pre-allocation Mark Brown (1): ASoC: Intel: sof_sdw: Update DMI matches for Lenovo Peter Ujfalusi (1): ASoC: SOF: Intel: hda-dai: Do not release the link DMA on STOP Richard Fitzgerald (2): ASoC: Intel: sof_sdw: Fix DMI match for Lenovo 21Q6 and 21Q7 ASoC: Intel: sof_sdw: Fix DMI match for Lenovo 21QA and 21QB Shuming Fan (1): ASoC: rt722: add delay time to wait for the calibration procedure Venkata Prasad Potturu (1): ASoC: amd: ps: Fix for enabling DMIC on acp63 platform via _DSD entry .../devicetree/bindings/sound/realtek,rt5645.yaml | 2 +- sound/soc/amd/ps/pci-ps.c | 17 ++++++++++++++- sound/soc/codecs/rt722-sdca.c | 7 +++++- sound/soc/intel/boards/sof_sdw.c | 23 +++++++++++++++++--- .../soc/mediatek/common/mtk-afe-platform-driver.c | 4 ++-- sound/soc/sof/intel/hda-dai.c | 25 ++++++++++++++++------ sound/soc/sof/intel/hda.h | 2 -- 7 files changed, 64 insertions(+), 16 deletions(-)

2 1

[PATCH] ASoC: codecs: ES8326: Reduce pop noise
by Zhang Yi 20 Dec '24

20 Dec '24

We modify the value of ES8326_ANA_MICBIAS to reduce the pop noise Signed-off-by: Zhang Yi <zhangyi(a)everest-semi.com> --- sound/soc/codecs/es8326.c | 22 +++++++++++++++++++--- 1 file changed, 19 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/es8326.c b/sound/soc/codecs/es8326.c index a5603b617688..27736c6933c5 100755 --- a/sound/soc/codecs/es8326.c +++ b/sound/soc/codecs/es8326.c @@ -616,7 +616,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) 0x0F, 0x0F); if (es8326->version > ES8326_VERSION_B) { regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30); } } } else { @@ -631,6 +631,8 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) regmap_write(es8326->regmap, ES8326_HPR_OFFSET_INI, offset_r); es8326->calibrated = true; } + regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0x00); + regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x00); if (direction == SNDRV_PCM_STREAM_PLAYBACK) { regmap_update_bits(es8326->regmap, ES8326_DAC_DSM, 0x01, 0x01); usleep_range(1000, 5000); @@ -645,7 +647,7 @@ static int es8326_mute(struct snd_soc_dai *dai, int mute, int direction) } else { msleep(300); if (es8326->version > ES8326_VERSION_B) { - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x50); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x70); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x00); } regmap_update_bits(es8326->regmap, ES8326_ADC_MUTE, @@ -676,6 +678,10 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x00); regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x20); regmap_update_bits(es8326->regmap, ES8326_RESET, 0x02, 0x00); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x30); + } break; case SND_SOC_BIAS_PREPARE: break; @@ -683,6 +689,12 @@ static int es8326_set_bias_level(struct snd_soc_component *codec, regmap_write(es8326->regmap, ES8326_ANA_PDN, 0x3b); regmap_update_bits(es8326->regmap, ES8326_CLK_CTL, 0x20, 0x00); regmap_write(es8326->regmap, ES8326_SDINOUT1_IO, ES8326_IO_INPUT); + if (es8326->version > ES8326_VERSION_B) { + regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x70, 0x10); + } + regmap_update_bits(es8326->regmap, ES8326_CLK_INV, 0xc0, 0xc0); + regmap_update_bits(es8326->regmap, ES8326_CLK_MUX, 0x80, 0x80); break; case SND_SOC_BIAS_OFF: clk_disable_unprepare(es8326->mclk); @@ -771,6 +783,10 @@ static void es8326_jack_button_handler(struct work_struct *work) cur_button = SND_JACK_BTN_0; break; case 0x6f: + if (es8326->version > ES8326_VERSION_B) { + cur_button = SND_JACK_BTN_0; + break; + } case 0x4b: /* button volume up */ cur_button = SND_JACK_BTN_1; @@ -1082,7 +1098,7 @@ static void es8326_init(struct snd_soc_component *component) regmap_write(es8326->regmap, ES8326_ADC2_SRC, 0x66); es8326_disable_micbias(es8326->component); if (es8326->version > ES8326_VERSION_B) { - regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x13); + regmap_update_bits(es8326->regmap, ES8326_ANA_MICBIAS, 0x73, 0x10); regmap_update_bits(es8326->regmap, ES8326_VMIDSEL, 0x40, 0x40); } -- 2.17.1

1 0

USB-Audio: Add support for TASCAM Model 12
by GitHub pull_request - opened 20 Dec '24

20 Dec '24

alsa-project/alsa-ucm-conf pull request #490 was opened from Unveiled53: Adds support for the TASCAM Model 12 USB multi-track mixer. Couple of things to note: - This profile assumes the "USB AUDIO" setting on the mixer itself is set to "MULTI INPUT". - Under the "MULTI INPUT" mode, this mixer has 12 logical input channels (Last two are a stereo mix) and 10 logical output channels. - Most of the accessible I/O is combo XLR/TRS jacks, so I opted to have the mono capture devices (With the exception of physical inputs 8 and 10 which are TRS only) named as "Mic" with the rest as "Line". Otherwise, the profile is tested and working with the latest USB-Audio profiles. [alsa-info.txt](https://github.com/user-attachments/files/18207924/alsa-info… Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/490 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/490.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

CS42L43 headset mic enable/disable not applied when jack insert state changes
by GitHub issues - opened 19 Dec '24

19 Dec '24

alsa-project/alsa-ucm-conf issue #489 was opened from rfvirgil: Originally the configuration for sof-soundwire/cs42l43*.conf had the headset mic and internal mics on the same ALSA capture device 4. The Headset Mic enable and disable sequences ran as expected when the headset jack was inserted and removed. But SOF expects that device 4 is the internal mics, and headset mic should be on device 1. So we made this change to move the headset mic to capture device 1: https://github.com/alsa-project/alsa-ucm-conf/commit/30989bd0c2aa3f9f4b6f5e… But now the Headset Mic enable and disable sequences do not run when the headset jack state changes. Seen on multiple systems with Pipewire + alsa-ucm + CS42L43 on sof-soundwire. Example from a system that booted with jack unplugged. The jack was then plugged in. The CS42L43 has reported that the headset mic is present: ``` Simple mixer control 'Headset Mic',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] ``` But the enable sequence has not been applied (DP2TX1 and DP2TX2 should have been set to Decimator 1): ``` Simple mixer control 'cs42l43 DP2TX1 Input',0 Capabilities: enum Items: 'None' 'Tone Generator 1' 'Tone Generator 2' 'Decimator 1' 'Decimator 2' 'Decimator 3' 'Decimator 4' 'ASPRX1' 'ASPRX2' 'ASPRX3' 'ASPRX4' 'ASPRX5' 'ASPRX6' 'DP5RX1' 'DP5RX2' 'DP6RX1' 'DP6RX2' 'DP7RX1' 'DP7RX2' 'ASRC INT1' 'ASRC INT2' 'ASRC INT3' 'ASRC INT4' 'ASRC DEC1' 'ASRC DEC2' 'ASRC DEC3' 'ASRC DEC4' 'ISRC1 INT1' 'ISRC1 INT2' 'ISRC1 DEC1' 'ISRC1 DEC2' 'ISRC2 INT1' 'ISRC2 INT2' 'ISRC2 DEC1' 'ISRC2 DEC2' 'EQ1' 'EQ2' Item0: 'None' Simple mixer control 'cs42l43 DP2TX2 Input',0 Capabilities: enum Items: 'None' 'Tone Generator 1' 'Tone Generator 2' 'Decimator 1' 'Decimator 2' 'Decimator 3' 'Decimator 4' 'ASPRX1' 'ASPRX2' 'ASPRX3' 'ASPRX4' 'ASPRX5' 'ASPRX6' 'DP5RX1' 'DP5RX2' 'DP6RX1' 'DP6RX2' 'DP7RX1' 'DP7RX2' 'ASRC INT1' 'ASRC INT2' 'ASRC INT3' 'ASRC INT4' 'ASRC DEC1' 'ASRC DEC2' 'ASRC DEC3' 'ASRC DEC4' 'ISRC1 INT1' 'ISRC1 INT2' 'ISRC1 DEC1' 'ISRC1 DEC2' 'ISRC2 INT1' 'ISRC2 INT2' 'ISRC2 DEC1' 'ISRC2 DEC2' 'EQ1' 'EQ2' Item0: 'None' ``` Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/489 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

snd_pcm_writei() takes to much time to write data
by GitHub issues - edited 19 Dec '24

19 Dec '24

alsa-project/alsa-lib issue #428 was edited from nnamera: I am using the ALSA open-source library to read and write audio data for my project. To write audio data, I am calling the snd_pcm_writei function. Initially, it works as expected, but after writing 2-3 frames, the function starts taking significantly more time (around 1 second) to complete. Normally, snd_pcm_writei should execute within 1-3 ms, but this delay is affecting my application's overall performance, causing audio chopping issues. Here are the details of my setup: Frame size: ~8000 bytes (writing 8000 bytes per call) Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 Slave: Soft volume PCM Control: amba-playback min_dB: -20 max_dB: 20 resolution: 256 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 Slave: Hardware PCM card 0 'es8389' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAV[ 22.206911] ambarella-i2c e400a000.i2c: No ACK from address 0x6c, 0:0! ED format [ 22.214533] os04c10: probe of 2-0001 failed with error -5 : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 appl_ptr : 0 hw_ptr : 0 I would appreciate any guidance or suggestions on why this delay is happening and how I can resolve it. Let me know if you need any additional details or logs. Issue URL : https://github.com/alsa-project/alsa-lib/issues/428 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

snd_pcm_readi() takes to much time to write data
by GitHub issues - opened 19 Dec '24

19 Dec '24

alsa-project/alsa-lib issue #428 was opened from nnamera: I am using the ALSA open-source library to read and write audio data for my project. To write audio data, I am calling the snd_pcm_writei function. Initially, it works as expected, but after writing 2-3 frames, the function starts taking significantly more time (around 1 second) to complete. Normally, snd_pcm_writei should execute within 1-3 ms, but this delay is affecting my application's overall performance, causing audio chopping issues. Here are the details of my setup: Frame size: ~8000 bytes (writing 8000 bytes per call) Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 Slave: Soft volume PCM Control: amba-playback min_dB: -20 max_dB: 20 resolution: 256 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 Slave: Hardware PCM card 0 'es8389' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAV[ 22.206911] ambarella-i2c e400a000.i2c: No ACK from address 0x6c, 0:0! ED format [ 22.214533] os04c10: probe of 2-0001 failed with error -5 : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tstamp_mode : NONE tstamp_type : MONOTONIC period_step : 1 avail_min : 1000 period_event : 0 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 9007199254740992000 appl_ptr : 0 hw_ptr : 0 I would appreciate any guidance or suggestions on why this delay is happening and how I can resolve it. Let me know if you need any additional details or logs. Issue URL : https://github.com/alsa-project/alsa-lib/issues/428 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v2] ASoC: amd: yc: Fix the wrong return value
by Venkata Prasad Potturu 17 Dec '24

17 Dec '24

With the current implementation, when ACP driver fails to read ACPI _WOV entry then the DMI overrides code won't invoke, may cause regressions for some BIOS versions. Add a condition check to jump to check the DMI entries incase of ACP driver fail to read ACPI _WOV method. Fixes: 4095cf872084 (ASoC: amd: yc: Fix for enabling DMIC on acp6x via _DSD entry) Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- Changes since v1: - removed redundant "&& wov_en" check sound/soc/amd/yc/acp6x-mach.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index e38c5885dadf..ecf57a6cb7c3 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -578,14 +578,19 @@ static int acp6x_probe(struct platform_device *pdev) handle = ACPI_HANDLE(pdev->dev.parent); ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status); - if (!ACPI_FAILURE(ret)) + if (!ACPI_FAILURE(ret)) { wov_en = dmic_status; + if (!wov_en) + return -ENODEV; + } else { + /* Incase of ACPI method read failure then jump to check_dmi_entry */ + goto check_dmi_entry; + } - if (is_dmic_enable && wov_en) + if (is_dmic_enable) platform_set_drvdata(pdev, &acp6x_card); - else - return 0; +check_dmi_entry: /* check for any DMI overrides */ dmi_id = dmi_first_match(yc_acp_quirk_table); if (dmi_id) -- 2.25.1

5 5

UCM2: Intel: sof-hda-dsp: Enable Dmic0 pre-processing and SDW speaker post-processing
by GitHub pull_request - edited 16 Dec '24

16 Dec '24

alsa-project/alsa-ucm-conf pull request #481 was edited from singalsu: This patch adds enable of the microphone capture pre-processing features as BootSequence inital values if the controls for them are available. If the user later switches these off, they will remain off. The DRC is dynamic range control that boosts the capture level, and the TDFB (time-domain fixed beamformer) improves capture signa-to-noise ratio for set beam angle direction. The used 0 degrees angle is defined as angle towards notebook computer user. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/481 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/481.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] ASoC: amd: ps: Fix for enabling DMIC on acp63 platform via _DSD entry
by Venkata Prasad Potturu 16 Dec '24

16 Dec '24

Add condition check to register ACP PDM sound card by reading _WOV acpi entry. Fixes: 0386d765f27a ("ASoC: amd: ps: refactor acp device configuration read logic") Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/ps/pci-ps.c | 17 ++++++++++++++++- 1 file changed, 16 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index a7583844f5b4..983d7fa13938 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -374,11 +374,18 @@ static int get_acp63_device_config(struct pci_dev *pci, struct acp63_dev_data *a { struct acpi_device *pdm_dev; const union acpi_object *obj; + acpi_handle handle; + acpi_integer dmic_status; u32 config; bool is_dmic_dev = false; bool is_sdw_dev = false; + bool wov_en, dmic_en; int ret; + /* IF WOV entry not found, enable dmic based on acp-audio-device-type entry*/ + wov_en = true; + dmic_en = false; + config = readl(acp_data->acp63_base + ACP_PIN_CONFIG); switch (config) { case ACP_CONFIG_4: @@ -411,10 +418,18 @@ static int get_acp63_device_config(struct pci_dev *pci, struct acp63_dev_data *a if (!acpi_dev_get_property(pdm_dev, "acp-audio-device-type", ACPI_TYPE_INTEGER, &obj) && obj->integer.value == ACP_DMIC_DEV) - is_dmic_dev = true; + dmic_en = true; } + + handle = ACPI_HANDLE(&pci->dev); + ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status); + if (!ACPI_FAILURE(ret)) + wov_en = dmic_status; } + if (dmic_en && wov_en) + is_dmic_dev = true; + if (acp_data->is_sdw_config) { ret = acp_scan_sdw_devices(&pci->dev, ACP63_SDW_ADDR); if (!ret && acp_data->info.link_mask) -- 2.25.1

3 2

[PATCH v7 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 13 Dec '24

13 Dec '24

This function is base on the accelerator implementation for compress API: 04177158cf98 ("ALSA: compress_offload: introduce accel operation mode") Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v7: - add 'select SND_COMPRESS_OFFLOAD' for fixing compile issue with imx_v6_v7_defconfig changes in v6: - rebase to latest tree, change to use 'MODULE_IMPORT_NS("DMA_BUF")' - Add Acked-by: Vinod Koul for patch 1/6 changes in v5: - Drop Jaroslav Kysela's patch as it has been merged. - Add Jaroslav Kysela's Acked-by tag, received in v3. changes in v4: - remove the RFC tag, no comments receive in v3 - Add Jaroslav Kysela's patch in this patch set. because it may be better for reviewing in a full patch set. - Fix the list_for_each_entry_reverse to list_for_each_entry_safe_reverse - Fix some coding style issues in Jaroslav Kysela's patch changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_params.h | 23 +- sound/soc/fsl/Kconfig | 2 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 179 ++++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 +++ sound/soc/fsl/fsl_asrc_m2m.c | 727 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 261 +++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1261 insertions(+), 9 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

3 9

[PATCH v6 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 13 Dec '24

13 Dec '24

This function is base on the accelerator implementation for compress API: 04177158cf98 ("ALSA: compress_offload: introduce accel operation mode") Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v6: - rebase to latest tree, change to use 'MODULE_IMPORT_NS("DMA_BUF")' - Add Acked-by: Vinod Koul for patch 1/6 changes in v5: - Drop Jaroslav Kysela's patch as it has been merged. - Add Jaroslav Kysela's Acked-by tag, received in v3. changes in v4: - remove the RFC tag, no comments receive in v3 - Add Jaroslav Kysela's patch in this patch set. because it may be better for reviewing in a full patch set. - Fix the list_for_each_entry_reverse to list_for_each_entry_safe_reverse - Fix some coding style issues in Jaroslav Kysela's patch changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_params.h | 23 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 179 ++++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 +++ sound/soc/fsl/fsl_asrc_m2m.c | 727 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 261 +++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1260 insertions(+), 9 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

2 7

Re: [syzbot] [tipc?] kernel BUG in __pskb_pull_tail
by syzbot 13 Dec '24

13 Dec '24

syzbot has bisected this issue to: commit de4f5fed3f231a8ff4790bf52975f847b95b85ea Author: Jens Axboe <axboe(a)kernel.dk> Date: Wed Mar 29 14:52:15 2023 +0000 iov_iter: add iter_iovec() helper bisection log: https://syzkaller.appspot.com/x/bisect.txt?x=17424730580000 start commit: 96b6fcc0ee41 Merge branch 'net-dsa-cleanup-eee-part-1' git tree: net-next final oops: https://syzkaller.appspot.com/x/report.txt?x=14c24730580000 console output: https://syzkaller.appspot.com/x/log.txt?x=10c24730580000 kernel config: https://syzkaller.appspot.com/x/.config?x=1362a5aee630ff34 dashboard link: https://syzkaller.appspot.com/bug?extid=4f66250f6663c0c1d67e syz repro: https://syzkaller.appspot.com/x/repro.syz?x=166944f8580000 C reproducer: https://syzkaller.appspot.com/x/repro.c?x=1287ecdf980000 Reported-by: syzbot+4f66250f6663c0c1d67e(a)syzkaller.appspotmail.com Fixes: de4f5fed3f23 ("iov_iter: add iter_iovec() helper") For information about bisection process see: https://goo.gl/tpsmEJ#bisection

2 1

[GIT PULL] ASoC fixes for v6.12-rc2
by Mark Brown 12 Dec '24

12 Dec '24

The following changes since commit fac04efc5c793dccbd07e2d59af9f90b7fc0dca4: Linux 6.13-rc2 (2024-12-08 14:03:39 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.12-rc2 for you to fetch changes up to 255cc582e6e16191a20d54bcdbca6c91d3e90c5e: ASoC: Intel: sof_sdw: Add space for a terminator into DAIs array (2024-12-12 11:08:49 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.13 A small pile of driver specific fixes, all quite small and not particularly major. ---------------------------------------------------------------- Charles Keepax (1): ASoC: Intel: sof_sdw: Add space for a terminator into DAIs array Shenghao Ding (1): ASoC: tas2781: Fix calibration issue in stress test Shengjiu Wang (2): ASoC: fsl_xcvr: change IFACE_PCM to IFACE_MIXER ASoC: fsl_spdif: change IFACE_PCM to IFACE_MIXER Stephen Gordon (1): ASoC: audio-graph-card: Call of_node_put() on correct node Venkata Prasad Potturu (1): ASoC: amd: yc: Fix the wrong return value sound/soc/amd/yc/acp6x-mach.c | 13 +++++++++---- sound/soc/codecs/tas2781-i2c.c | 2 +- sound/soc/fsl/fsl_spdif.c | 2 +- sound/soc/fsl/fsl_xcvr.c | 2 +- sound/soc/generic/audio-graph-card2.c | 2 +- sound/soc/intel/boards/sof_sdw.c | 8 ++++++-- 6 files changed, 19 insertions(+), 10 deletions(-)

2 1

Re: [PATCH v6 4/6] ASoC: fsl_asrc_m2m: Add memory to memory function
by Shengjiu Wang 12 Dec '24

12 Dec '24

On Wed, Dec 11, 2024 at 11:28 PM Mark Brown <broonie(a)kernel.org> wrote: > > On Wed, Dec 11, 2024 at 11:08:47AM +0800, Shengjiu Wang wrote: > > Implement the ASRC memory to memory function using > > the compress framework, user can use this function with > > compress ioctl interface. > > > > This feature can be shared by ASRC and EASRC drivers > > This breaks an imx_v6_v7_defconfig build: > > arm-linux-gnueabihf-ld: sound/soc/fsl/fsl_asrc_m2m.o: in function `fsl_asrc_m2m_ > init': > fsl_asrc_m2m.c:(.text+0x7ac): undefined reference to `snd_compress_new' > > Looks like we're missing some select statements. Thanks for noticing this issue. Sorry about that, I will fix it. Best regards Shengjiu Wang

1 0

git clone failure
by GitHub issues - opened 12 Dec '24

12 Dec '24

alsa-project/tinycompress issue #27 was opened from antonio-fr: Working on embedded project, there's a recipe for a system that uses tinycompress, and points to the alsa-project git server. Unfortunately, tinycompress can't be downloaded, leading the building system unable to make the system image. ``` $ git clone git://git.alsa-project.org/tinycompress.git Cloning into 'tinycompress'... fatal: Could not read from remote repository. Please make sure you have the correct access rights and the repository exists. ``` It worked yesterday, And I can see the project [there in the web interface](https://git.alsa-project.org/?p=tinycompress.git;a=tree). But right now, no luck. Issue URL : https://github.com/alsa-project/tinycompress/issues/27 Repository URL: https://github.com/alsa-project/tinycompress

1 0

Re: [PATCH v2 30/37] ALSA: seq: Automatic conversion of UMP events
by Mike Lothian 11 Dec '24

11 Dec '24

Hi I'm not sure when this first appeared but I see the following when compiling/linking with Clang 19.1.5 vmlinux.o: warning: objtool: cc_ev_to_ump_midi2+0x5a5: return with modified stack frame This is the only commit I can see that has cc_ev_to_ump_midi2 in it Cheers Mike

2 1

[PATCH v1] ASoC: tas2781: Fix calibration issue in stress test
by Shenghao Ding 11 Dec '24

11 Dec '24

One specific test condition: the default registers of p[j].reg ~ p[j+3].reg are 0, TASDEVICE_REG(0x00, 0x14, 0x38)(PLT_FLAG_REG), TASDEVICE_REG(0x00, 0x14, 0x40)(SINEGAIN_REG), and TASDEVICE_REG(0x00, 0x14, 0x44)(SINEGAIN2_REG). After first calibration, they are freshed to TASDEVICE_REG(0x00, 0x1a, 0x20), TASDEVICE_REG(0x00, 0x16, 0x58)(PLT_FLAG_REG), TASDEVICE_REG(0x00, 0x14, 0x44)(SINEGAIN_REG), and TASDEVICE_REG(0x00, 0x16, 0x64)(SINEGAIN2_REG) via "Calibration Start" kcontrol. In second calibration, the p[j].reg ~ p[j+3].reg have already become tas2781_cali_start_reg. However, p[j+2].reg, TASDEVICE_REG(0x00, 0x14, 0x44)(SINEGAIN_REG), will be freshed to TASDEVICE_REG(0x00, 0x16, 0x64), which is the third register in the input params of the kcontrol. This is why only first calibration can work, the second-time, third-time or more-time calibration always failed without reboot. Of course, if no p[j].reg is in the list of tas2781_cali_start_reg, this stress test can work well. Fixes: 49e2e353fb0d ("ASoC: tas2781: Add Calibration Kcontrols for Chromebook") Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- sound/soc/codecs/tas2781-i2c.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index be2ca5eb6c93..fb8cd2284fe8 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -370,7 +370,7 @@ static void sngl_calib_start(struct tasdevice_priv *tas_priv, int i, tasdevice_dev_read(tas_priv, i, p[j].reg, (int *)&p[j].val[0]); } else { - switch (p[j].reg) { + switch (tas2781_cali_start_reg[j].reg) { case 0: { if (!reg[0]) continue; -- 2.34.1

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rt722: add headset mic led control
by GitHub pull_request - edited 11 Dec '24

11 Dec '24

alsa-project/alsa-ucm-conf pull request #487 was edited from shumingfan: rt722: add headset mic led control Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/487 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/487.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH v4 00/24] Add audio support for the Renesas RZ/G3S SoC
by Claudiu 10 Dec '24

10 Dec '24

From: Claudiu Beznea <claudiu.beznea.uj(a)bp.renesas.com> Hi, Series enables the audio support for the Renesas RZ/G3S SoC along with runtime PM and suspend to RAM. Patches: - 01-03/24 - update versaclock3 clock generator driver to support the 5L35023 hardware variant; versaclock3 provides clocks for the audio devices (SSIF, DA7212 codec) - 04-19/24 - add SSIF support for the RZ/G3S SoC; fixes and cleanups were also included - 20-24/24 - add device tree support Merge strategy, if any: - clock patches (01-03/24) can go the clock tree - audio patches (04-19/24) can go through the audio tree - device tree patches (20-24/24) can go through the Renesas tree Thank you, Claudiu Beznea Changes in v4: - added patch 18/24 "ASoC: dt-bindings: renesas,rz-ssi: Remove DMA description" - collected tags - addressed review comments - dropped already integrated patches - the changelog for each patch is detailed within the individual patches Changes in v3: - use renesas instead of sh in audio patches title - use proper fixes tag for patch "ASoC: renesas: rz-ssi: Terminate all the DMA transactions" - collected tags Changes in v2: - fixed typos pointed out in the review process - dropped da7213 patches already applied - dropped patch "ASoC: sh: rz-ssi: Use a proper bitmask for clear bits" as requested in the review process Claudiu Beznea (24): clk: versaclock3: Prepare for the addition of 5L35023 device dt-bindings: clock: versaclock3: Document 5L35023 Versa3 clock generator clk: versaclock3: Add support for the 5L35023 variant ASoC: renesas: rz-ssi: Terminate all the DMA transactions ASoC: renesas: rz-ssi: Use only the proper amount of dividers ASoC: renesas: rz-ssi: Fix typo on SSI_RATES macro comment ASoC: renesas: rz-ssi: Remove pdev member of struct rz_ssi_priv ASoC: renesas: rz-ssi: Remove the rz_ssi_get_dai() function ASoC: renesas: rz-ssi: Remove the first argument of rz_ssi_stream_is_play() ASoC: renesas: rz-ssi: Use readl_poll_timeout_atomic() ASoC: renesas: rz-ssi: Use temporary variable for struct device ASoC: renesas: rz-ssi: Use goto label names that specify their actions ASoC: renesas: rz-ssi: Rely on the ASoC subsystem to runtime resume/suspend the SSI ASoC: renesas: rz-ssi: Enable runtime PM autosuspend support ASoC: renesas: rz-ssi: Add runtime PM support ASoC: renesas: rz-ssi: Issue software reset in hw_params API ASoC: renesas: rz-ssi: Add suspend to RAM support ASoC: dt-bindings: renesas,rz-ssi: Remove DMA description ASoC: dt-bindings: renesas,rz-ssi: Document the Renesas RZ/G3S SoC arm64: dts: renesas: r9a08g045: Add SSI nodes arm64: dts: renesas: rzg3s-smarc-som: Add versa3 clock generator node arm64: dts: renesas: Add da7212 audio codec node arm64: dts: renesas: rzg3s-smarc: Enable SSI3 arm64: dts: renesas: rzg3s-smarc: Add sound card .../bindings/clock/renesas,5p35023.yaml | 1 + .../bindings/sound/renesas,rz-ssi.yaml | 19 +- arch/arm64/boot/dts/renesas/r9a08g045.dtsi | 94 ++++++++ .../boot/dts/renesas/rzg3s-smarc-som.dtsi | 47 +++- arch/arm64/boot/dts/renesas/rzg3s-smarc.dtsi | 63 +++++ drivers/clk/clk-versaclock3.c | 67 ++++-- sound/soc/renesas/rz-ssi.c | 226 +++++++++++------- 7 files changed, 388 insertions(+), 129 deletions(-) -- 2.39.2

2 27

[PATCH] ASoC: amd: yc: Fix the wrong return value
by Venkata Prasad Potturu 10 Dec '24

10 Dec '24

With the current implementation, when ACP driver fails to read ACPI _WOV entry then the DMI overrides code won't invoke, may cause regressions for some BIOS versions. Add a condition check to jump to check the DMI entries incase of ACP driver fail to read ACPI _WOV method. Fixes: 4095cf872084 (ASoC: amd: yc: Fix for enabling DMIC on acp6x via _DSD entry) Signed-off-by: Venkata Prasad Potturu <venkataprasad.potturu(a)amd.com> --- sound/soc/amd/yc/acp6x-mach.c | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/sound/soc/amd/yc/acp6x-mach.c b/sound/soc/amd/yc/acp6x-mach.c index e38c5885dadf..13af1d485eb0 100644 --- a/sound/soc/amd/yc/acp6x-mach.c +++ b/sound/soc/amd/yc/acp6x-mach.c @@ -578,14 +578,19 @@ static int acp6x_probe(struct platform_device *pdev) handle = ACPI_HANDLE(pdev->dev.parent); ret = acpi_evaluate_integer(handle, "_WOV", NULL, &dmic_status); - if (!ACPI_FAILURE(ret)) + if (!ACPI_FAILURE(ret)) { wov_en = dmic_status; + if (!wov_en) + return -ENODEV; + } else { + /* Incase of ACPI method read failure then jump to check_dmi_entry */ + goto check_dmi_entry; + } if (is_dmic_enable && wov_en) platform_set_drvdata(pdev, &acp6x_card); - else - return 0; +check_dmi_entry: /* check for any DMI overrides */ dmi_id = dmi_first_match(yc_acp_quirk_table); if (dmi_id) -- 2.25.1

4 5

Support for ASUS ROG STRIX X870E-E GAMING WIFI
by GitHub pull_request - opened 10 Dec '24

10 Dec '24

alsa-project/alsa-ucm-conf pull request #486 was opened from aidvu: Add support for [ASUS ROG STRIX X870E-E GAMING WIFI](https://rog.asus.com/motherboards/rog-strix/rog-strix-x870e-e-gaming-… ). ```Bus 003 Device 004: ID 0b05:1b9b ASUSTek Computer, Inc. USB Audio``` Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/486 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/486.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

ucm2: sof-soundwire: Correct FixedBootSequence for dmic info
by GitHub pull_request - opened 10 Dec '24

10 Dec '24

alsa-project/alsa-ucm-conf pull request #485 was opened from ujfalusi: Replace the curly brackets with square brackets. The curly ones break the FixedBootSequence parsing by resetting the sequence. If the /var/lib/alsa/card0.conf.d is not present then UCM will fail because all sequences before the dmic is lost, including the card-init: # alsactl init alsa-lib main.c:619:(execute_cfgsave) unable to open file '/var/lib/alsa/card0.conf.d/42-sof-hdmi.conf': No such file or directory alsa-lib main.c:2456:(set_fixedboot_user) Unable to execute force boot sequence The full sequence supposed to be: card-init ctl-remap cs42l43 dmic hdmi Fixes: 9fa70add7638 ("sof-hda-dsp,sof-soundwire: add CaptureMicInfoFile fields for dmics") Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/485 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/485.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

[PATCH] dt-bindings: sound:atmel-at91sam9g20ek: convert bindings to json-schema
by Balakrishnan Sambath 10 Dec '24

10 Dec '24

Convert atmel-at91sam9g20ek-wm8731-audio DT binding to yaml based json-schema.Change file name to match json-scheme naming. Signed-off-by: Balakrishnan Sambath <balakrishnan.s(a)microchip.com> --- .../bindings/sound/atmel,at91sam9g20ek-wm8731.yaml | 60 ++++++++++++++++++++++ .../sound/atmel-at91sam9g20ek-wm8731-audio.txt | 26 ---------- 2 files changed, 60 insertions(+), 26 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/atmel,at91sam9g20ek-wm8731.yaml b/Documentation/devicetree/bindings/sound/atmel,at91sam9g20ek-wm8731.yaml new file mode 100644 index 000000000000..f6330707fe1b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/atmel,at91sam9g20ek-wm8731.yaml @@ -0,0 +1,60 @@ +# SPDX-License-Identifier: (GPL-2.0-only OR BSD-2-Clause) +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/atmel,at91sam9g20ek-wm8731.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: Atmel at91sam9g20ek wm8731 audio complex + +maintainers: + - Balakrishnan Sambath <balakrishnan.s(a)microchip.com> + +properties: + compatible: + const: atmel,at91sam9g20ek-wm8731-audio + + atmel,model: + $ref: /schemas/types.yaml#/definitions/string + description: The user-visible name of this sound complex. + atmel,audio-routing: + $ref: /schemas/types.yaml#/definitions/non-unique-string-array + description: A list of the connections between audio components. + minItems: 2 + items: + enum: + # Board Connectors + - "Ext Spk" + - "Int MIC" + + # CODEC Pins + - LHPOUT + - MICIN + atmel,ssc-controller: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of the SSC controller + atmel,audio-codec: + $ref: /schemas/types.yaml#/definitions/phandle + description: The phandle of WM8731 audio codec + +required: + - compatible + - atmel,model + - atmel,audio-routing + - atmel,ssc-controller + - atmel,audio-codec + +additionalProperties: false + +examples: + - | + sound { + compatible = "atmel,at91sam9g20ek-wm8731-audio"; + pinctrl-names = "default"; + pinctrl-0 = <&pinctrl_pck0_as_mck>; + atmel,model = "wm8731 @ AT91SAMG20EK"; + atmel,audio-routing = + "Ext Spk", "LHPOUT", + "Int MIC", "MICIN"; + atmel,ssc-controller = <&ssc0>; + atmel,audio-codec = <&wm8731>; + }; diff --git a/Documentation/devicetree/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt b/Documentation/devicetree/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt deleted file mode 100644 index 9c5a9947b64d..000000000000 --- a/Documentation/devicetree/bindings/sound/atmel-at91sam9g20ek-wm8731-audio.txt +++ /dev/null @@ -1,26 +0,0 @@ -* Atmel at91sam9g20ek wm8731 audio complex - -Required properties: - - compatible: "atmel,at91sam9g20ek-wm8731-audio" - - atmel,model: The user-visible name of this sound complex. - - atmel,audio-routing: A list of the connections between audio components. - - atmel,ssc-controller: The phandle of the SSC controller - - atmel,audio-codec: The phandle of the WM8731 audio codec -Optional properties: - - pinctrl-names, pinctrl-0: Please refer to pinctrl-bindings.txt - -Example: -sound { - compatible = "atmel,at91sam9g20ek-wm8731-audio"; - pinctrl-names = "default"; - pinctrl-0 = <&pinctrl_pck0_as_mck>; - - atmel,model = "wm8731 @ AT91SAMG20EK"; - - atmel,audio-routing = - "Ext Spk", "LHPOUT", - "Int MIC", "MICIN"; - - atmel,ssc-controller = <&ssc0>; - atmel,audio-codec = <&wm8731>; -}; --- base-commit: 0dd3ee31125508cd67f7e7172247f05b7fd1753a change-id: 20240214-at91sam9g20ek-wm8731-yaml-404ddddbb91a Best regards, -- Balakrishnan Sambath <balakrishnan.s(a)microchip.com>

5 5

[PATCH 00/17] ASoC: remove snd_soc_dai_set_tristate()
by Kuninori Morimoto 10 Dec '24

10 Dec '24

Hi Mark snd_soc_dai_set_tristate() has added in v2.6.27 (2008/Oct) by this commit 8c6529dbf881303920a415c2d14a500218661949 ("ALSA: asoc: core - add Digital Audio Interface (DAI) control functions.") But it has never been used during this 16 years. Let's remove it. Kuninori Morimoto (17): ASoC: pxa: remove snd_soc_dai_set_tristate() ASoC: codecs: cs35l32: remove snd_soc_dai_set_tristate() ASoC: codecs: cs35l33: remove snd_soc_dai_set_tristate() ASoC: codecs: cs35l34: remove snd_soc_dai_set_tristate() ASoC: codecs: cs42l73: remove snd_soc_dai_set_tristate() ASoC: codecs: cs53l30: remove snd_soc_dai_set_tristate() ASoC: codecs: wm8961: remove snd_soc_dai_set_tristate() ASoC: codecs: wm8994: remove snd_soc_dai_set_tristate() ASoC: codecs: wm8995: remove snd_soc_dai_set_tristate() ASoC: codecs: wm9713: remove snd_soc_dai_set_tristate() ASoC: codecs: adau1372: remove snd_soc_dai_set_tristate() ASoC: codecs: adau1977: remove snd_soc_dai_set_tristate() ASoC: codecs: adau7118: remove snd_soc_dai_set_tristate() ASoC: codecs: madera: remove snd_soc_dai_set_tristate() ASoC: codecs: arizona: remove snd_soc_dai_set_tristate() ASoC: codecs: twl4030: remove snd_soc_dai_set_tristate() ASoC: remove snd_soc_dai_set_tristate() include/sound/soc-dai.h | 3 --- sound/soc/codecs/adau1372.c | 14 -------------- sound/soc/codecs/adau1977.c | 15 --------------- sound/soc/codecs/adau7118.c | 19 ------------------- sound/soc/codecs/arizona.c | 17 ----------------- sound/soc/codecs/cs35l32.c | 9 --------- sound/soc/codecs/cs35l33.c | 21 --------------------- sound/soc/codecs/cs35l34.c | 15 --------------- sound/soc/codecs/cs42l73.c | 10 ---------- sound/soc/codecs/cs53l30.c | 10 ---------- sound/soc/codecs/madera.c | 22 ---------------------- sound/soc/codecs/twl4030.c | 28 ---------------------------- sound/soc/codecs/wm8961.c | 14 -------------- sound/soc/codecs/wm8994.c | 28 ---------------------------- sound/soc/codecs/wm8995.c | 33 --------------------------------- sound/soc/codecs/wm9713.c | 17 ----------------- sound/soc/pxa/pxa-ssp.c | 21 --------------------- sound/soc/soc-dai.c | 19 ------------------- 18 files changed, 315 deletions(-) -- 2.43.0

3 20

Re: [vdr] Identifying MPEG-4 HE-AAC (LATM, LAOS) audio formats
by schorpp 09 Dec '24

09 Dec '24

Am 06.12.24 um 21:32 schrieb Marko Mäkelä: > A couple of weeks ago, Finland took the penultimate step to get rid of > DVB-T. Since years, we had DVB-T and DVB-T2 running in parallel. The > DVB-T transponders should be shut down in 2025. same here but h.265 video and aac audio. > > The step was to introduce a second free-to-air DVB-T2 bouquet to carry > some channels that were previously only available with DVB-T and MPEG-2. > The VDR channel search failed to find this transponder; I ran a patched > version of https://github.com/mighty-p/t2scan/pull/15/ to generate a new > channels.conf. (Is this a bug of VDR, or a feature? I remember that when > I originally set up VDR, I had to get a channels.conf from tscan or > t2scan.) have You tried the latest git version of scan in dvb-utils @linuxtx.org repo? > > Unlike our first free-to-air DVB-T2 bouquet, which uses MPEG-4 H.264 > video with AC3 audio, the second one uses H.264 video and HE-AAC audio, > also known as LATM or LOAS. This had to be explicitly enabled for my > output device: https://github.com/reufer/rpihddevice/issues/20 > > The HE-AAC support in rpihddevice is experimental, as it has been in ffmpeg until today and rpihddevive is based on it? > maybe because the > sampling rate is hard-coded as 48 kHz and the number of channels as 2. > Unlike AC3, it does not look like these parameters would be available in > any audio frame header. As far I understood from a NorDig.org > specification, the sampling rate would always be 48 kHz, but the number > of channels could be identified by something in the container, > specifically, Table 12.28 and Table 12.29 at page 164 of > https://nordig.org/wp-content/uploads/2016/03/NorDig-Unified- > Requirements-ver.-3.2.1.pdf Have You tried the latest git ffprobe version, does it detect the aac flavor of Your dvb-t2 DVB channels correctly? Have You tried latest git mediainfo? > > What would have to be changed in VDR or rpihddevice or both to correctly > switch between 2-channel and 5.1-channel HE-AAC audio? This is mostly an > academic question and exercise, because I only have stereo audio > hardware and don't know if any programming is actually available with > multichannel audio. Not at all academic. Since AVRs do not implement Fraunhofer codecs we need a HDNI PCM Multichannel driver for the raspberry pi to playback 5.1 channel aac without transcoding it to ac3 which can be done with mplayer on the fly. I've made a HDMI PCM Multichannel output to AVR driver patch for the Allwinner SoC alsa audio driver, look for it on the alsa-dev list archives. > > Best regards, > > Marko y tom

2 1

[PATCH treewide 00/11] ASoC: Clean up {hp,mic}-det-gpio handling
by Geert Uytterhoeven 09 Dec '24

09 Dec '24

Hi all, The "gpio" suffix for GPIO consumers was deprecated a while ago, in favor of the "gpios" suffix. However, there are still several users of the "hp-det-gpio" and "mic-det-gpio" properties, in DT bindings, drivers, and DT source files. Drivers that use gpiolib handle both the deprecated and new properties transparently, but that is not the case when using of_property_*() calls directly. - The first patch fixes a bug, as the Freescale Generic ASoC Sound driver handles only the deprecated properties, while there is already a user of the new variants, - The second patch updates the Audio Graph and Simple Audio Card DT bindings, - Patches 3-9 converts various DTS files to use the new properties, - The last 2 patches convert Freescale sound device nodes to use the new properties. All patches can be applied independently, except for the last two, which depend on the first patch to avoid regressions. This series was compile-tested only. Thanks for your comments! Geert Uytterhoeven (11): ASoC: fsl-asoc-card: Add missing handling of {hp,mic}-dt-gpios ASoC: dt-bindings: Deprecate {hp,mic}-det-gpio ARM: dts: marvell: mmp2-olpc-xo-1-75: Switch to {hp,mic}-det-gpios arm64: dts: freescale: imx: Switch to simple-audio-card,hp-det-gpios arm64: dts: renesas: beacon-renesom: Switch to mic-det-gpios arm64: dts: rockchip: Switch to simple-audio-card,hp-det-gpios arm64: dts: rockchip: Switch to hp-det-gpios arm64: dts: uniphier: Switch to hp-det-gpios mips: dts: ingenic: Switch to simple-audio-card,hp-det-gpios ARM: dts: nxp: imx: Switch to {hp,mic}-det-gpios arm64: dts: freescale: imx: Switch to hp-det-gpios .../devicetree/bindings/sound/audio-graph.yaml | 6 ++++++ .../devicetree/bindings/sound/simple-card.yaml | 12 ++++++++++++ arch/arm/boot/dts/marvell/mmp2-olpc-xo-1-75.dts | 4 ++-- arch/arm/boot/dts/nxp/imx/imx6qdl-sabresd.dtsi | 4 ++-- arch/arm/boot/dts/nxp/imx/imx6sl-evk.dts | 2 +- arch/arm/boot/dts/nxp/imx/imx6sll-evk.dts | 2 +- arch/arm/boot/dts/nxp/imx/imx6sx-sdb.dtsi | 2 +- arch/arm/boot/dts/nxp/imx/imx6ul-14x14-evk.dtsi | 2 +- arch/arm/boot/dts/nxp/imx/imx7d-sdb.dts | 2 +- .../boot/dts/freescale/imx8mq-librem5-devkit.dts | 2 +- arch/arm64/boot/dts/freescale/imx8mq-librem5.dtsi | 2 +- arch/arm64/boot/dts/freescale/imx8qm-mek.dts | 2 +- arch/arm64/boot/dts/freescale/imx8qxp-mek.dts | 2 +- arch/arm64/boot/dts/freescale/imx95-19x19-evk.dts | 2 +- .../boot/dts/renesas/beacon-renesom-baseboard.dtsi | 2 +- .../boot/dts/rockchip/rk3326-anbernic-rg351m.dtsi | 2 +- .../arm64/boot/dts/rockchip/rk3326-gameforce-chi.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3326-odroid-go.dtsi | 2 +- arch/arm64/boot/dts/rockchip/rk3368-lba3368.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3399-eaidk-610.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3399-pinebook-pro.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3399-roc-pc-plus.dts | 2 +- .../boot/dts/rockchip/rk3399-rock-pi-4a-plus.dts | 2 +- .../boot/dts/rockchip/rk3399-rock-pi-4b-plus.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3399-rock-pi-4c.dts | 2 +- .../boot/dts/rockchip/rk3399-sapphire-excavator.dts | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg-arc.dtsi | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg353p.dts | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg353ps.dts | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg353v.dts | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg353vs.dts | 2 +- .../boot/dts/rockchip/rk3566-anbernic-rg503.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3566-pinetab2.dtsi | 2 +- arch/arm64/boot/dts/rockchip/rk3566-powkiddy-x55.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3568-odroid-m1.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-armsom-sige7.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-evb1-v10.dts | 2 +- .../dts/rockchip/rk3588-friendlyelec-cm3588-nas.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-nanopc-t6.dtsi | 2 +- arch/arm64/boot/dts/rockchip/rk3588-ok3588-c.dts | 2 +- .../boot/dts/rockchip/rk3588-orangepi-5-plus.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-quartzpro64.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-rock-5-itx.dts | 2 +- arch/arm64/boot/dts/rockchip/rk3588-rock-5b.dts | 2 +- .../boot/dts/rockchip/rk3588s-gameforce-ace.dts | 2 +- .../boot/dts/socionext/uniphier-ld11-global.dts | 2 +- .../boot/dts/socionext/uniphier-ld20-global.dts | 2 +- arch/mips/boot/dts/ingenic/gcw0.dts | 2 +- arch/mips/boot/dts/ingenic/rs90.dts | 2 +- sound/soc/fsl/fsl-asoc-card.c | 8 +++++--- 50 files changed, 72 insertions(+), 52 deletions(-) -- 2.34.1 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

8 30

Audio Crackling with some Steam games
by orbea 06 Dec '24

06 Dec '24

Hi, OS: Gentoo Kernel: 6.12.1 alsa-lib: 1.2.12 Product Name: X670E PG Lightning GPU: [AMD/ATI] Navi 10 [Radeon RX 5600 OEM/5600 XT / 5700/5700 XT] (rev c1) When using HDMI audio I am having an issue where some Steam games as run using Proton will have audio crackling or static after a short period of time after starting the game. The games that I know do this are 'Tales of Symphonia', 'Ys VIII: Lacrimosa of DANA' and 'Metaphor: Refantazio'. While many other games do not have this problem. I am not sure if this is a problem specific to the Steam ecosystem, the games, alsa, the kernel or something else? At least in the case of 'Ys VIII' I can only reproduce the problem when running the game in Steam where the same game files when run using the system wine or self compiled proton there are no audio issues. In the Proton logs it shows many underruns as soon as the issue starts: ALSA lib pcm.c:8545:(snd_pcm_recover) underrun occurred The problem also appears when using a simple /etc/asound.conf that contains only: defaults.pcm.card 0 defaults.pcm.device 9 defaults.ctl.card 0 I have also tried asking about this issue on the Proton Github issue tracker and the FAudio issue tracker. https://github.com/FNA-XNA/FAudio/issues/367 https://github.com/ValveSoftware/Proton/issues/3820#issuecomment-2477846139 https://github.com/ValveSoftware/Proton/issues/822#issuecomment-2326999328 In the FAudio issue it was suggested it was related to protonmediaconverter where the system wine and proton use ffmpeg, but I am not sure how to confirm this. I attached the alsa-info.txt. Is there anything I can do to further diagnose, resolve or workaround this? Any further info I neglected to provide that can help this? Thanks!

1 0

[PATCH v2 0/3] ASoC: mediatek:mt8186: fix both the speaker and hdmi
by Jiaxin Yu 06 Dec '24

06 Dec '24

The speaker and hdmi of mt8186 platform are shared the same port of I2S, when connect the external display, use build-in speakers to play audio, they both playback at the same time. So we want to manage the playback device through DAPM events. Change since v1: Support audio function for it6505 bridge to hdmi-codec Jiaxin Yu (3): ASoC: hdmi-codec: Add event handler for hdmi TX ASoC: mediatek: mt8186: correct the HDMI widgets drm/bridge: it6505: Add audio support drivers/gpu/drm/bridge/ite-it6505.c | 81 +++++++++++++++++-- include/sound/hdmi-codec.h | 6 ++ sound/soc/codecs/hdmi-codec.c | 37 ++++++++- .../mt8186/mt8186-mt6366-da7219-max98357.c | 2 +- .../mt8186/mt8186-mt6366-rt1019-rt5682s.c | 2 +- 5 files changed, 116 insertions(+), 12 deletions(-) -- 2.18.0

3 16

[PATCH v5 RESEND 0/6] ASoC: fsl: add memory to memory function for ASRC
by Shengjiu Wang 06 Dec '24

06 Dec '24

This function is base on the accelerator implementation for compress API: https://patchwork.kernel.org/project/alsa-devel/patch/20241002093904.180979… Audio signal processing also has the requirement for memory to memory similar as Video. This asrc memory to memory (memory ->asrc->memory) case is a non real time use case. User fills the input buffer to the asrc module, after conversion, then asrc sends back the output buffer to user. So it is not a traditional ALSA playback and capture case. Because we had implemented the "memory -> asrc ->i2s device-> codec" use case in ALSA. Now the "memory->asrc->memory" needs to reuse the code in asrc driver, so the patch 1 and patch 2 is for refining the code to make it can be shared by the "memory->asrc->memory" driver. Other change is to add memory to memory support for two kinds of i.MX ASRC modules. changes in v5: - Drop Jaroslav Kysela's patch as it has been merged. - Add Jaroslav Kysela's Acked-by tag, received in v3. changes in v4: - remove the RFC tag, no comments receive in v3 - Add Jaroslav Kysela's patch in this patch set. because it may be better for reviewing in a full patch set. - Fix the list_for_each_entry_reverse to list_for_each_entry_safe_reverse - Fix some coding style issues in Jaroslav Kysela's patch changes in v3: - use Jaroslav's suggestion for header file compress_params.h (PATCH 01) - remove the ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE definition - remove ASRC_RATIO_MOD in this version because it uses .set_metadata() Will wait Jaroslav's update or other better method in the future. - Address some comments from Pierre. changes in v2: - Remove the changes in compress API - drop the SNDRV_COMPRESS_SRC_RATIO_MOD - drop the SND_AUDIOCODEC_SRC and struct snd_dec_src - define private metadata key value ASRC_OUTPUT_FORMAT/ASRC_OUTPUT_RATE/ASRC_RATIO_MOD Shengjiu Wang (6): ALSA: compress: Add output rate and output format support ASoC: fsl_asrc: define functions for memory to memory usage ASoC: fsl_easrc: define functions for memory to memory usage ASoC: fsl_asrc_m2m: Add memory to memory function ASoC: fsl_asrc: register m2m platform device ASoC: fsl_easrc: register m2m platform device include/uapi/sound/compress_params.h | 23 +- sound/soc/fsl/Kconfig | 1 + sound/soc/fsl/Makefile | 2 +- sound/soc/fsl/fsl_asrc.c | 179 ++++++- sound/soc/fsl/fsl_asrc.h | 2 + sound/soc/fsl/fsl_asrc_common.h | 70 +++ sound/soc/fsl/fsl_asrc_m2m.c | 727 +++++++++++++++++++++++++++ sound/soc/fsl/fsl_easrc.c | 261 +++++++++- sound/soc/fsl/fsl_easrc.h | 4 + 9 files changed, 1260 insertions(+), 9 deletions(-) create mode 100644 sound/soc/fsl/fsl_asrc_m2m.c -- 2.34.1

2 7

[GIT PULL] ASoC fixes for v6.13-rc1
by Mark Brown 05 Dec '24

05 Dec '24

The following changes since commit 40384c840ea1944d7c5a392e8975ed088ecf0b37: Linux 6.13-rc1 (2024-12-01 14:28:56 -0800) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v6.13-rc1 for you to fetch changes up to ec16a3cdf37e507013062f9c4a2067eacdd12b62: ASoC: mediatek: mt8188-mt6359: Remove hardcoded dmic codec (2024-12-05 13:39:58 +0000) ---------------------------------------------------------------- ASoC: Fixes for v6.13 A few small fixes for v6.13, all system specific - the biggest thing is the fix for jack handling over suspend on some Intel laptops. ---------------------------------------------------------------- Dan Carpenter (1): ASoC: SOF: ipc3-topology: fix resource leaks in sof_ipc3_widget_setup_comp_dai() Marek Maslanka (1): ASoC: Intel: avs: da7219: Remove suspend_pre() and resume_post() Nícolas F. R. A. Prado (1): ASoC: mediatek: mt8188-mt6359: Remove hardcoded dmic codec sound/soc/intel/avs/boards/da7219.c | 17 ----------------- sound/soc/mediatek/mt8188/mt8188-mt6359.c | 4 +--- sound/soc/sof/ipc3-topology.c | 7 ++++--- 3 files changed, 5 insertions(+), 23 deletions(-)

2 1

With SOF enabled in sof-hda-generic.tplg device (no DMIC) the control 'Capture Source' is not controlled by UCM
by GitHub issues - edited 05 Dec '24

05 Dec '24

alsa-project/alsa-ucm-conf issue #484 was edited from singalsu: The configuration is not usual. If there is no DMIC in the system usually the legacy mode is selected (as with `options snd_intel_dspcfg dsp_driver=1`). In such PCs SOF can be enabled with `options snd_intel_dspcfg dsp_driver=3`. When a headset is plugged/unplugged the playback correctly switches between speaker and headphone/headset earbuds. But the microphone capture remains muted. I noticed with legacy mode where internal microphones work that control name='Capture Source' changes to 1 when headset is removed: ``` singalsu@asteri:~$ amixer -c0 cget name='Capture Source' numid=6,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Headset Mic' ; Item #1 'Internal Mic' : values=1 singalsu@asteri:~$ amixer -c0 cget name='Capture Source' numid=6,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Headset Mic' ; Item #1 'Internal Mic' : values=0 ``` This is not happening when SOF is enabled. But I could make internal microphones capture work with `amixer -c0 cset name='Capture Source' 1`. Is there a way to fix this in ucm2? I'm not absolutely sure about this issue in Fedora 41 but this happens with my ucm2 development code with plain git master branch (commit 0782bf435b8332833e7b3fc91501d78473dd6edf). Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/484 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

With SOF enabled in sof-hda-generic.tplg device (no DMIC) the control 'Capture Source' is not controlled by UCM
by GitHub issues - opened 05 Dec '24

05 Dec '24

alsa-project/alsa-ucm-conf issue #484 was opened from singalsu: The configuration is not usual. If there is no DMIC in the system usually the legacy mode is selected (as with `options snd_intel_dspcfg dsp_driver=1`). In such PCs SOF can be enabled with `options snd_intel_dspcfg dsp_driver=3`. When a headset is plugged/unplugged the playback correctly switches between speaker and headphone/headset earbuds. But the microphone capture remains muted. I noticed with legacy mode where internal microphones work that control name='Capture Source' changes to 1 when headset is removed: ``` singalsu@asteri:~$ amixer -c0 cget name='Capture Source' numid=6,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Headset Mic' ; Item #1 'Internal Mic' : values=1 singalsu@asteri:~$ amixer -c0 cget name='Capture Source' numid=6,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Headset Mic' ; Item #1 'Internal Mic' : values=0 ``` This is not happening when SOF is enabled. But I could make internal microphones capture work with `amixer -c0 cset name='Capture Source' 1`. Is there a way to fix this in ucm2? I'm not absolutely sure about this issue in Fedora 41 but this happens with my modified ucm2 development code with git master branch. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/484 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

scsi: aic7xxx: Fix build 'aicasm' warning
by wangdicheng 05 Dec '24

05 Dec '24

1 0

aplay doesn't play audio sometimes "unable to open slave"
by GitHub issues - opened 05 Dec '24

05 Dec '24

alsa-project/alsa-lib issue #426 was opened from haoyi-codes: I have initially thought it was a bug that was related to chromium as stated in #425. However, now I realize it has to be something else that is wrong. `aplay -D default /dev/zero` ``` ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave aplay: main:850: audio open error: No such file or directory ``` `aplay -D plughw:0 /dev/zero` ``` aplay: main:850: audio open error: No such file or directory ``` `aplay -l` ``` **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 3: Generic Digital [Generic Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 3: Generic Digital [Generic Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Audio [SA9123 USB Audio], device 0: USB Audio [USB Audio] Subdevices: 1/1 Subdevice #0: subdevice #0 ``` `cat /proc/asound/cards` ``` 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0x81080000 irq 17 1 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0x81220000 irq 165 2 [Audio ]: USB-Audio - SA9123 USB Audio SYC SA9123 USB Audio at usb-0000:00:14.0-3.2, full speed 3 [Edition ]: USB-Audio - Logi 4K Stream Edition Logi 4K Stream Edition at usb-0000:00:14.0-3.3, super speed ``` `ls -la /dev/snd/*` ``` crw-rw----+ 1 root audio 116, 0 Dec 4 23:18 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 32 Dec 4 23:18 /dev/snd/controlC1 crw-rw----+ 1 root audio 116, 64 Dec 4 23:18 /dev/snd/controlC2 crw-rw----+ 1 root audio 116, 96 Dec 4 23:18 /dev/snd/controlC3 crw-rw----+ 1 root audio 116, 19 Dec 4 23:18 /dev/snd/pcmC0D3p crw-rw----+ 1 root audio 116, 51 Dec 4 23:18 /dev/snd/pcmC1D3p crw-rw----+ 1 root audio 116, 80 Dec 4 23:18 /dev/snd/pcmC2D0p crw-rw----+ 1 root audio 116, 120 Dec 4 23:18 /dev/snd/pcmC3D0c crw-rw----+ 1 root audio 116, 1 Dec 4 23:18 /dev/snd/seq crw-rw----+ 1 root audio 116, 33 Dec 4 23:18 /dev/snd/timer /dev/snd/by-id: total 0 drwxr-xr-x 2 root root 80 Dec 4 23:18 . drwxr-xr-x 4 root root 280 Dec 4 23:17 .. lrwxrwxrwx 1 root root 12 Dec 4 23:18 usb-046d_Logi_4K_Stream_Edition_51C07EAF-03 -> ../controlC3 lrwxrwxrwx 1 root root 12 Dec 4 23:18 usb-SYC_SA9123_USB_Audio-01 -> ../controlC2 /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 160 Dec 4 23:18 . drwxr-xr-x 4 root root 280 Dec 4 23:17 .. lrwxrwxrwx 1 root root 12 Dec 4 23:18 pci-0000:00:14.0-usb-0:3.2:1.1 -> ../controlC2 lrwxrwxrwx 1 root root 12 Dec 4 23:18 pci-0000:00:14.0-usb-0:3.3:1.3 -> ../controlC3 lrwxrwxrwx 1 root root 12 Dec 4 23:18 pci-0000:00:14.0-usbv2-0:3.2:1.1 -> ../controlC2 lrwxrwxrwx 1 root root 12 Dec 4 23:18 pci-0000:00:14.0-usbv3-0:3.3:1.3 -> ../controlC3 lrwxrwxrwx 1 root root 12 Dec 4 23:18 pci-0000:01:00.1 -> ../controlC0 lrwxrwxrwx 1 root root 12 Dec 4 23:18 pci-0000:05:00.1 -> ../controlC1 ``` For reference I am still able to play audio on my system as pulseaudio is running but for applications that use alsa directly it does not work. ``` Server String: /run/user/1011/pulse/native Library Protocol Version: 35 Server Protocol Version: 35 Is Local: yes Client Index: 59 Tile Size: 65472 User Name: user Host Name: system Server Name: PulseAudio (on PipeWire 1.2.7) Server Version: 15.0.0 Default Sample Specification: float32le 2ch 48000Hz Default Channel Map: front-left,front-right Default Sink: alsa_output.usb-SYC_SA9123_USB_Audio-01.iec958-stereo Default Source: alsa_output.usb-SYC_SA9123_USB_Audio-01.iec958-stereo.monitor Cookie: XXXX:XXXX ``` Issue URL : https://github.com/alsa-project/alsa-lib/issues/426 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

pulseaudio bails out on 1.2.13 (SIGSEGV), with MOTU Ultralite mk5
by GitHub issues - opened 04 Dec '24

04 Dec '24

alsa-project/alsa-ucm-conf issue #483 was opened from karolyi: Hey, I have this sound card and with the latest update (1.2.13), pulseaudio started to SIGSEGV and I have no sound. Upon going to the pulseaudio IRC channel and talking to the devs, we've managed to point out that it's the UCM configuration that makes pulseaudio crash. Reverting to 1.2.12 made the crash cease. It seems that the newly added configuration for the Ultralite makes pulseaudio crash. According to the below gdb traces, it seems the configuration drives pulseaudio into an infinite loop. Beginning of the trace: ``` Debuginfod has been enabled. To make this setting permanent, add 'set debuginfod enabled on' to .gdbinit. [Thread debugging using libthread_db enabled] Using host libthread_db library "/usr/lib/libthread_db.so.1". Program received signal SIGSEGV, Segmentation fault. 0x00007ffff7ae98df in _int_malloc (av=av@entry=0x7ffff7c2dac0 <main_arena>, bytes=bytes@entry=2080) at malloc.c:4059 4059 size_t tc_idx = csize2tidx (nb); Thread 1 (Thread 0x7ffff73298c0 (LWP 40542) "pulseaudio"): #0 0x00007ffff7ae98df in _int_malloc (av=av@entry=0x7ffff7c2dac0 <main_arena>, bytes=bytes@entry=2080) at malloc.c:4059 #1 0x00007ffff7aec069 in __libc_calloc (n=<optimized out>, elem_size=<optimized out>) at malloc.c:3754 #2 0x00007ffff7e5e366 in pa_xmalloc0 (size=2080) at ../src/pulse/xmalloc.c:74 #3 0x00007ffff7e7d3d4 in pa_idxset_new (hash_func=0x7ffff7e7d371 <pa_idxset_trivial_hash_func>, compare_func=0x7ffff7e7d37f <pa_idxset_trivial_compare_func>) at ../src/pulsecore/idxset.c:86 #4 0x00007ffff7e7f5a9 in pa_idxset_copy (s=0x5555556f5890, copy_func=0x0) at ../src/pulsecore/idxset.c:623 #5 0x00007ffff6b93ff0 in iterate_device_subsets (devices=0x5555556f5060, state=0x7fffffffca38) at ../src/modules/alsa/alsa-ucm.c:1399 #6 0x00007ffff6b94108 in iterate_maximal_device_subsets (devices=0x5555556f5060, state=0x7fffffffca38) at ../src/modules/alsa/alsa-ucm.c:1412 #7 0x00007ffff6b9418a in iterate_maximal_device_subsets (devices=0x5555556f5060, state=0x7fffffffca38) at ../src/modules/alsa/alsa-ucm.c:1422 #8 0x00007ffff6b9418a in iterate_maximal_device_subsets (devices=0x5555556f5060, state=0x7fffffffca38) at ../src/modules/alsa/alsa-ucm.c:1422 ``` Inbetween there's just a lot of recursive calls. End of the trace: ``` #130916 0x00007ffff6b9418a in iterate_maximal_device_subsets (devices=0x5555556f5060, state=0x7fffffffca38) at ../src/modules/alsa/alsa-ucm.c:1422 #130917 0x00007ffff6b9418a in iterate_maximal_device_subsets (devices=0x5555556f5060, state=0x7fffffffca38) at ../src/modules/alsa/alsa-ucm.c:1422 #130918 0x00007ffff6b9418a in iterate_maximal_device_subsets (devices=0x5555556f5060, state=0x7fffffffca38) at ../src/modules/alsa/alsa-ucm.c:1422 #130919 0x00007ffff6b9418a in iterate_maximal_device_subsets (devices=0x5555556f5060, state=0x7fffffffca38) at ../src/modules/alsa/alsa-ucm.c:1422 #130920 0x00007ffff6b96635 in ucm_create_verb_profiles (ucm=0x55555567bc60, ps=0x5555556ca560, verb=0x5555556c8970, verb_name=0x5555556c6d60 "HiFi", verb_desc=0x5555556a3120 "Default") at ../src/modules/alsa/alsa-ucm.c:2042 #130921 0x00007ffff6b973bf in pa_alsa_ucm_add_profile_set (ucm=0x55555567bc60, default_channel_map=0x555555570de0) at ../src/modules/alsa/alsa-ucm.c:2305 #130922 0x00007ffff6c6bb43 in module_alsa_card_LTX_pa__init (m=0x55555566d9d0) at ../src/modules/alsa/module-alsa-card.c:966 #130923 0x00007ffff7f26919 in pa_module_load (module=0x7fffffffccf0, c=0x555555570d20, name=0x7ffff6c7747d "module-alsa-card", argument=0x555555587f10 "device_id=\"2\" name=\"usb-MOTU_UltraLite-mk5_UL5LFF34E6-00\" card_name=\"alsa_card.usb-MOTU_UltraLite-mk5_UL5LFF34E6-00\" namereg_fail=false tsched=yes fixed_latency_range=no ignore_dB=no deferred_volume=y"...) at ../src/pulsecore/module.c:208 #130924 0x00007ffff6c74260 in verify_access (u=0x55555566ae40, d=0x55555566e430) at ../src/modules/module-udev-detect.c:342 #130925 0x00007ffff6c7488f in card_changed (u=0x55555566ae40, dev=0x55555568ae50) at ../src/modules/module-udev-detect.c:437 #130926 0x00007ffff6c74c6f in process_device (u=0x55555566ae40, dev=0x55555568ae50) at ../src/modules/module-udev-detect.c:479 #130927 0x00007ffff6c74cf8 in process_path (u=0x55555566ae40, path=0x5555556879b0 "/sys/devices/pci0000:00/0000:00:01.2/0000:02:00.0/0000:03:08.0/0000:07:00.1/usb1/1-1/1-1.1/1-1.1:1.0/sound/card2") at ../src/modules/module-udev-detect.c:496 #130928 0x00007ffff6c75ed0 in module_udev_detect_LTX_pa__init (m=0x555555666220) at ../src/modules/module-udev-detect.c:812 #130929 0x00007ffff7f26919 in pa_module_load (module=0x7fffffffcf28, c=0x555555570d20, name=0x555555668c20 "module-udev-detect", argument=0x0) at ../src/pulsecore/module.c:208 #130930 0x00007ffff7f073ba in pa_cli_command_load (c=0x555555570d20, t=0x555555668c40, buf=0x5555555717b0, fail=0x55555556f6b5) at ../src/pulsecore/cli-command.c:440 #130931 0x00007ffff7f0fac4 in pa_cli_command_execute_line_stateful (c=0x555555570d20, s=0x7fffffffd100 "load-module module-udev-detect", buf=0x5555555717b0, fail=0x55555556f6b5, ifstate=0x7fffffffd0f8) at ../src/pulsecore/cli-command.c:2207 #130932 0x00007ffff7f0fda2 in pa_cli_command_execute_file_stream (c=0x555555570d20, f=0x55555556fce0, buf=0x5555555717b0, fail=0x55555556f6b5) at ../src/pulsecore/cli-command.c:2247 #130933 0x00007ffff7f10031 in pa_cli_command_execute_file (c=0x555555570d20, fn=0x555555570c70 "/home/karolyi/Work/opensource/pulseaudio/build/src/daemon/default.pa", buf=0x5555555717b0, fail=0x55555556f6b5) at ../src/pulsecore/cli-command.c:2278 #130934 0x00007ffff7f0f7d8 in pa_cli_command_execute_line_stateful (c=0x555555570d20, s=0x555555570560 ".include /home/karolyi/Work/opensource/pulseaudio/build/src/daemon/default.pa", buf=0x5555555717b0, fail=0x55555556f6b5, ifstate=0x7fffffffdb1c) at ../src/pulsecore/cli-command.c:2170 #130935 0x00007ffff7f10239 in pa_cli_command_execute (c=0x555555570d20, s=0x555555577bc0 ".include /home/karolyi/Work/opensource/pulseaudio/build/src/daemon/default.pa\n", buf=0x5555555717b0, fail=0x55555556f6b5) at ../src/pulsecore/cli-command.c:2304 #130936 0x0000555555564162 in main (argc=7, argv=0x7fffffffde98) at ../src/daemon/main.c:1267 Continuing. Program terminated with signal SIGSEGV, Segmentation fault. The program no longer exists. ``` Requested debug output from the pulseaudio project: https://termbin.com/n04r Please investigate. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/483 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Should SOF ucm2/blobs/sof user_configs and product_configs be without .conf suffix?
by GitHub issues - edited 04 Dec '24

04 Dec '24

alsa-project/alsa-ucm-conf issue #482 was edited from singalsu: It seems updates have dropped .conf from looked up SOF blobs customization files. Is the change intentional? Then the example in ucm2/blobs/sof/product_configs/AAEON/UPX-TGL01.conf should be renamed to be without .conf to work and to be applied correctly. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/482 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Should SOF ucm2/blobs/sof user_configs and product_configs be without .conf suffix?
by GitHub issues - opened 04 Dec '24

04 Dec '24

alsa-project/alsa-ucm-conf issue #482 was opened from singalsu: It seems updates have dropped .conf from looked up SOF blobs customization files. Is the change intentional? Then the example in ucm2/blobs/sof/product_configs/AAEON/UPX-TGL01.conf should be renamed to be without .conf to work and be applied correctly. Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/482 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

axfer-transfer.1: Some remarks and editorial changes for this man page
by Bjarni Ingi Gislason 04 Dec '24

04 Dec '24

Package: alsa-utils Version: 1.2.12-1 Severity: minor Tags: patch * What led up to the situation? Checking for defects with test-[g|n]roff -mandoc -t -K utf8 -rF0 -rHY=0 -ww -b -z < "man page" [Use "groff -e ' $' <file>" to find 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: backtrace: file '<stdin>':1006 troff:<stdin>:1006: warning: trailing space in the line * What outcome did you expect instead? No output (no warnings). -.- General remarks and further material, if a diff-file exist, are in the attachments. -- System Information: Debian Release: trixie/sid APT prefers testing APT policy: (500, 'testing') Architecture: amd64 (x86_64) Kernel: Linux 6.11.10-amd64 (SMP w/2 CPU threads; PREEMPT) Locale: LANG=is_IS.iso88591, LC_CTYPE=is_IS.iso88591 (charmap=ISO-8859-1), LANGUAGE not set Shell: /bin/sh linked to /usr/bin/dash Init: sysvinit (via /sbin/init) Versions of packages alsa-utils depends on: ii kmod 33+20240816-2 ii libasound2t64 1.2.12-1+b1 ii libatopology2t64 1.2.12-1+b1 ii libc6 2.40-3 ii libfftw3-single3 3.3.10-1+b3 ii libncursesw6 6.5-2+b1 ii libsamplerate0 0.2.2-4+b2 ii libtinfo6 6.5-2+b1 alsa-utils recommends no packages. Versions of packages alsa-utils suggests: pn dialog <none> -- no debconf information

1 0

[PATCH 1/2] ASoC: amd: ps: update mach params subsystem_rev variable
by Vijendar Mukunda 03 Dec '24

03 Dec '24

Update mach_params subsystem_rev with acp_rev variable for ACP6.3 platform. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/ps/pci-ps.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/amd/ps/pci-ps.c b/sound/soc/amd/ps/pci-ps.c index aef73ec6f7ef..4ed50cebadb3 100644 --- a/sound/soc/amd/ps/pci-ps.c +++ b/sound/soc/amd/ps/pci-ps.c @@ -312,6 +312,7 @@ static struct snd_soc_acpi_mach *acp63_sdw_machine_select(struct device *dev) if (mach && mach->link_mask) { mach->mach_params.links = mach->links; mach->mach_params.link_mask = mach->link_mask; + mach->mach_params.subsystem_rev = acp_data->acp_rev; return mach; } } -- 2.34.1

2 2

UCM2: Intel: sof-hda-dsp: Enable Dmic0 DRC and TDFB pre-processing
by GitHub pull_request - opened 03 Dec '24

03 Dec '24

alsa-project/alsa-ucm-conf pull request #481 was opened from singalsu: This patch adds enable of the microphone capture pre-processing features as BootSequence inital values if the controls for them are available. If the user later switches these off, they will remain off. The DRC is dynamic range control that boosts the capture level, and the TDFB (time-domain fixed beamformer) improves capture signa-to-noise ratio for set beam angle direction. The used 0 degrees angle is defined as angle towards notebook computer user. Request URL : https://github.com/alsa-project/alsa-ucm-conf/pull/481 Patch URL : https://github.com/alsa-project/alsa-ucm-conf/pull/481.patch Repository URL: https://github.com/alsa-project/alsa-ucm-conf

1 0

Re: native DSD on PCI linux driver
by Takashi Iwai 03 Dec '24

03 Dec '24

On Tue, 26 Nov 2024 10:41:21 +0100, Filip Van Lunteren wrote: > > > Hello, > > I am currently developing PCI audio output hardware and driver and want to > integrate native DSD playback. I am struggling to find documentation on this. > Is there DSD native support in ALSA? And how do I incorporate it? PCM is > already set up and works. AFAIK, in the driver side, there is not much special; it assumes that DSD samples are passed over the standard PCM stream, and the driver uses special formats SNDRV_PCM_FORMAT_DSD_U8, *_U16 and *_U32 (with _LE and _BE suffix) for 8, 16 and 32bit, instead of the standard SNDRV_PCM_FORMAT_U8, *_S16, etc. For those formats, the silence data is assumed to be 0x69, 0x6969, etc. Takashi

1 1

[PATCH] ASoC: tas2781: Fix redundant logical jump
by Tang Bin 02 Dec '24

02 Dec '24

In these functions, some logical jump of "goto" and variable are redundant, thus remove them. Signed-off-by: Tang Bin <tangbin(a)cmss.chinamobile.com> --- sound/soc/codecs/tas2781-i2c.c | 68 +++++++++++++--------------------- 1 file changed, 25 insertions(+), 43 deletions(-) diff --git a/sound/soc/codecs/tas2781-i2c.c b/sound/soc/codecs/tas2781-i2c.c index 68887799e..27f2f84a4 100644 --- a/sound/soc/codecs/tas2781-i2c.c +++ b/sound/soc/codecs/tas2781-i2c.c @@ -489,14 +489,11 @@ static int tas2563_calib_start_put(struct snd_kcontrol *kcontrol, struct snd_soc_component *comp = snd_soc_kcontrol_component(kcontrol); struct tasdevice_priv *tas_priv = snd_soc_component_get_drvdata(comp); const int sum = ARRAY_SIZE(tas2563_cali_start_reg); - int rc = 1; int i, j; guard(mutex)(&tas_priv->codec_lock); - if (tas_priv->chip_id != TAS2563) { - rc = -1; - goto out; - } + if (tas_priv->chip_id != TAS2563) + return -1; for (i = 0; i < tas_priv->ndev; i++) { struct tasdevice *tasdev = tas_priv->tasdevice; @@ -523,8 +520,8 @@ static int tas2563_calib_start_put(struct snd_kcontrol *kcontrol, q[j].val, 4); } } -out: - return rc; + + return 1; } static void tas2563_calib_stop_put(struct tasdevice_priv *tas_priv) @@ -576,7 +573,7 @@ static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, struct cali_reg *p = &cali_data->cali_reg_array; unsigned char *src = ucontrol->value.bytes.data; unsigned char *dst = cali_data->data; - int rc = 1, i = 0; + int i = 0; int j; guard(mutex)(&priv->codec_lock); @@ -605,7 +602,7 @@ static int tasdev_cali_data_put(struct snd_kcontrol *kcontrol, i += 3; memcpy(dst, &src[i], cali_data->total_sz); - return rc; + return 1; } static int tas2781_latch_reg_get(struct snd_kcontrol *kcontrol, @@ -1115,25 +1112,21 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) char *conf_name, *prog_name; int nr_controls = 4; int mix_index = 0; - int ret; /* Alloc kcontrol via devm_kzalloc, which don't manually * free the kcontrol */ dsp_ctrls = devm_kcalloc(tas_priv->dev, nr_controls, sizeof(dsp_ctrls[0]), GFP_KERNEL); - if (!dsp_ctrls) { - ret = -ENOMEM; - goto out; - } + if (!dsp_ctrls) + return -ENOMEM; /* Create mixer items for selecting the active Program and Config */ prog_name = devm_kstrdup(tas_priv->dev, "Speaker Program Id", GFP_KERNEL); - if (!prog_name) { - ret = -ENOMEM; - goto out; - } + if (!prog_name) + return -ENOMEM; + dsp_ctrls[mix_index].name = prog_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_programs; @@ -1143,10 +1136,9 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) conf_name = devm_kstrdup(tas_priv->dev, "Speaker Config Id", GFP_KERNEL); - if (!conf_name) { - ret = -ENOMEM; - goto out; - } + if (!conf_name) + return -ENOMEM; + dsp_ctrls[mix_index].name = conf_name; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_configurations; @@ -1156,10 +1148,9 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) active_dev_num = devm_kstrdup(tas_priv->dev, "Activate Tasdevice Num", GFP_KERNEL); - if (!active_dev_num) { - ret = -ENOMEM; - goto out; - } + if (!active_dev_num) + return -ENOMEM; + dsp_ctrls[mix_index].name = active_dev_num; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_active_num; @@ -1168,21 +1159,17 @@ static int tasdevice_dsp_create_ctrls(struct tasdevice_priv *tas_priv) mix_index++; chip_id = devm_kstrdup(tas_priv->dev, "Tasdevice Chip Id", GFP_KERNEL); - if (!chip_id) { - ret = -ENOMEM; - goto out; - } + if (!chip_id) + return -ENOMEM; + dsp_ctrls[mix_index].name = chip_id; dsp_ctrls[mix_index].iface = SNDRV_CTL_ELEM_IFACE_MIXER; dsp_ctrls[mix_index].info = tasdevice_info_chip_id; dsp_ctrls[mix_index].get = tasdevice_get_chip_id; mix_index++; - ret = snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, + return snd_soc_add_component_controls(tas_priv->codec, dsp_ctrls, nr_controls < mix_index ? nr_controls : mix_index); - -out: - return ret; } static int tasdevice_create_cali_ctrls(struct tasdevice_priv *priv) @@ -1469,7 +1456,6 @@ static int tasdevice_hw_params(struct snd_pcm_substream *substream, unsigned int slot_width; unsigned int fsrate; int bclk_rate; - int rc = 0; fsrate = params_rate(params); switch (fsrate) { @@ -1479,8 +1465,7 @@ static int tasdevice_hw_params(struct snd_pcm_substream *substream, default: dev_err(tas_priv->dev, "%s: incorrect sample rate = %u\n", __func__, fsrate); - rc = -EINVAL; - goto out; + return -EINVAL; } slot_width = params_width(params); @@ -1493,20 +1478,17 @@ static int tasdevice_hw_params(struct snd_pcm_substream *substream, default: dev_err(tas_priv->dev, "%s: incorrect slot width = %u\n", __func__, slot_width); - rc = -EINVAL; - goto out; + return -EINVAL; } bclk_rate = snd_soc_params_to_bclk(params); if (bclk_rate < 0) { dev_err(tas_priv->dev, "%s: incorrect bclk rate = %d\n", __func__, bclk_rate); - rc = bclk_rate; - goto out; + return bclk_rate; } -out: - return rc; + return 0; } static int tasdevice_set_dai_sysclk(struct snd_soc_dai *codec_dai, -- 2.33.0

3 2

alsa-lib-1.2.13 prevents chromium-131.0.6778.85 from playing audio on gentoo.
by GitHub issues - edited 02 Dec '24

02 Dec '24

alsa-project/alsa-lib issue #425 was edited from haoyi-codes: I have recently compiled chromium against alsa-lib-1.2.13 for two different systems. However, I am unable to play audio. The terminal output shows: System 1: ``` Failed to create secure directory (/run/user/1011//pulse): Permission denied ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave [6429:6429:1130/194540.301739:ERROR:alsa_util.cc(204)] PcmOpen: default,No such file or directory ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave [6429:6429:1130/194540.302803:ERROR:alsa_util.cc(204)] PcmOpen: plug:default,No such file or directory libpng warning: iCCP: known incorrect sRGB profile Warning: disabling flag --expose_wasm due to conflicting flags Warning: disabling flag --expose_wasm due to conflicting flags ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave [6429:6429:1130/194614.580896:ERROR:alsa_util.cc(204)] PcmOpen: default,No such file or directory ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave [6429:6429:1130/194614.582080:ERROR:alsa_util.cc(204)] PcmOpen: plug:default,No such file or directory libpng warning: iCCP: known incorrect sRGB profile ``` System 2: ``` socket(): Operation not permitted ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration [2516:2516:1129/230412.353283:ERROR:alsa_util.cc(204)] PcmOpen: default,No such file or directory ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration [2516:2516:1129/230412.353697:ERROR:alsa_util.cc(204)] PcmOpen: plug:default,No such file or directory ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration libpng warning: iCCP: known incorrect sRGB profile libpng warning: iCCP: known incorrect sRGB profile ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration [2516:2516:1129/230432.095492:ERROR:alsa_util.cc(204)] PcmOpen: default,No such file or directory ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration [2516:2516:1129/230432.095901:ERROR:alsa_util.cc(204)] PcmOpen: plug:default,No such file or directory libpng warning: iCCP: known incorrect sRGB profile ``` After downgrading alsa-lib to 1.2.12 and recompiling chromium I was able to play audio just fine. Further information can also be found at: https://bugs.gentoo.org/945749 Issue URL : https://github.com/alsa-project/alsa-lib/issues/425 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

alsa-lib-1.2.13 prevents chromium-131.0.6778.85 from playing audio on gentoo.
by GitHub issues - opened 02 Dec '24

02 Dec '24

alsa-project/alsa-lib issue #425 was opened from haoyi-codes: I have recently compiled chromium against alsa-lib-1.2.13 for two different systems, however I am unable to play audio. The terminal output shows: System 1: ``` Failed to create secure directory (/run/user/1011//pulse): Permission denied ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave [6429:6429:1130/194540.301739:ERROR:alsa_util.cc(204)] PcmOpen: default,No such file or directory ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave [6429:6429:1130/194540.302803:ERROR:alsa_util.cc(204)] PcmOpen: plug:default,No such file or directory libpng warning: iCCP: known incorrect sRGB profile Warning: disabling flag --expose_wasm due to conflicting flags Warning: disabling flag --expose_wasm due to conflicting flags ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave [6429:6429:1130/194614.580896:ERROR:alsa_util.cc(204)] PcmOpen: default,No such file or directory ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/pcm/pcm_dmix.c:1000:(snd_pcm_dmix_open) unable to open slave [6429:6429:1130/194614.582080:ERROR:alsa_util.cc(204)] PcmOpen: plug:default,No such file or directory libpng warning: iCCP: known incorrect sRGB profile ``` System 2: ``` socket(): Operation not permitted ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration [2516:2516:1129/230412.353283:ERROR:alsa_util.cc(204)] PcmOpen: default,No such file or directory ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration [2516:2516:1129/230412.353697:ERROR:alsa_util.cc(204)] PcmOpen: plug:default,No such file or directory ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration libpng warning: iCCP: known incorrect sRGB profile libpng warning: iCCP: known incorrect sRGB profile ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration [2516:2516:1129/230432.095492:ERROR:alsa_util.cc(204)] PcmOpen: default,No such file or directory ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4029:(snd_config_hooks_call) Cannot open shared library (null) (Error loading shared library /lib/libasound.so.2: Permission denied) ALSA lib /var/tmp/portage/media-libs/alsa-lib-1.2.13/work/alsa-lib-1.2.13/src/conf.c:4650:(snd_config_update_r) hooks failed, removing configuration [2516:2516:1129/230432.095901:ERROR:alsa_util.cc(204)] PcmOpen: plug:default,No such file or directory libpng warning: iCCP: known incorrect sRGB profile ``` After downgrading alsa-lib to 1.2.12 and recompiling chromium I was able to play audio just fine. Further information can also be found at: https://bugs.gentoo.org/945749 Issue URL : https://github.com/alsa-project/alsa-lib/issues/425 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH v4] ASoC: Intel: avs: da7219: Remove suspend_pre() and resume_post()
by Marek Maslanka 02 Dec '24

02 Dec '24

The presence of a plugged jack is not detected after resuming the device if the jack was plugged before the device was suspended. This problem is caused by calling the sound/soc/codecs/da7219-aad.c:da7219_aad_jack_det() function on resume, which forces the jack insertion state to be unplugged. Signed-off-by: Marek Maslanka <mmaslanka(a)google.com> --- Changes in v4: - Remove redundant spaces from the commit message - Link to v3: https://lore.kernel.org/all/20241128155926.1774625-1-mmaslanka@google.com/ Changes in v3: - Fix line breaks in commit message - Link to v2: https://lore.kernel.org/all/20241128151239.1666582-1-mmaslanka@google.com/ Changes in v2: - Rephrase commit title and message as suggested in v1 - Link to v1: https://lore.kernel.org/all/20241128122732.1205732-1-mmaslanka@google.com/ --- --- sound/soc/intel/avs/boards/da7219.c | 17 ----------------- 1 file changed, 17 deletions(-) diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c index 80c0a1a956542..daf53ca490a14 100644 --- a/sound/soc/intel/avs/boards/da7219.c +++ b/sound/soc/intel/avs/boards/da7219.c @@ -211,21 +211,6 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in return 0; } -static int avs_card_suspend_pre(struct snd_soc_card *card) -{ - struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME); - - return snd_soc_component_set_jack(codec_dai->component, NULL, NULL); -} - -static int avs_card_resume_post(struct snd_soc_card *card) -{ - struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME); - struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); - - return snd_soc_component_set_jack(codec_dai->component, jack, NULL); -} - static int avs_da7219_probe(struct platform_device *pdev) { struct snd_soc_dai_link *dai_link; @@ -257,8 +242,6 @@ static int avs_da7219_probe(struct platform_device *pdev) card->name = "avs_da7219"; card->dev = dev; card->owner = THIS_MODULE; - card->suspend_pre = avs_card_suspend_pre; - card->resume_post = avs_card_resume_post; card->dai_link = dai_link; card->num_links = 1; card->controls = card_controls; -- 2.47.0.338.g60cca15819-goog

3 2

[PATCH v3] ASoC: Intel: avs: da7219: Remove suspend_pre() and resume_post()
by Marek Maslanka 02 Dec '24

02 Dec '24

The presence of a plugged jack is not detected after resuming the device if the jack was plugged before the device was suspended. This problem is caused by calling the sound/soc/codecs/da7219-aad.c:da7219_aad_jack_det() function on resume, which forces the jack insertion state to be unplugged. Signed-off-by: Marek Maslanka <mmaslanka(a)google.com> --- Changes in v3: - Fix line breaks in commit message - Link to v2: https://lore.kernel.org/all/20241128151239.1666582-1-mmaslanka@google.com/ Changes in v2: - Rephrase commit title and message as suggested in v1 - Link to v1: https://lore.kernel.org/all/20241128122732.1205732-1-mmaslanka@google.com/ --- --- sound/soc/intel/avs/boards/da7219.c | 17 ----------------- 1 file changed, 17 deletions(-) diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c index 80c0a1a956542..daf53ca490a14 100644 --- a/sound/soc/intel/avs/boards/da7219.c +++ b/sound/soc/intel/avs/boards/da7219.c @@ -211,21 +211,6 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in return 0; } -static int avs_card_suspend_pre(struct snd_soc_card *card) -{ - struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME); - - return snd_soc_component_set_jack(codec_dai->component, NULL, NULL); -} - -static int avs_card_resume_post(struct snd_soc_card *card) -{ - struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME); - struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); - - return snd_soc_component_set_jack(codec_dai->component, jack, NULL); -} - static int avs_da7219_probe(struct platform_device *pdev) { struct snd_soc_dai_link *dai_link; @@ -257,8 +242,6 @@ static int avs_da7219_probe(struct platform_device *pdev) card->name = "avs_da7219"; card->dev = dev; card->owner = THIS_MODULE; - card->suspend_pre = avs_card_suspend_pre; - card->resume_post = avs_card_resume_post; card->dai_link = dai_link; card->num_links = 1; card->controls = card_controls; -- 2.47.0.338.g60cca15819-goog

2 1

[PATCH] ASoC: Intel: avs: Remove component->set_jack() on suspend/resume
by Marek Maslanka 02 Dec '24

02 Dec '24

Removed calls to component->set_jack() during suspend/resume to fix an issue where an inserted jack is not detected after resuming from suspend if the jack was inserted before suspend. This problem is caused by calls to the sound/soc/codecs/da7219-aad.c:da7219_aad_jack_det() function during resume, which forces the jack insertion state to be unplugged. Signed-off-by: Marek Maslanka <mmaslanka(a)google.com> --- sound/soc/intel/avs/boards/da7219.c | 17 ----------------- 1 file changed, 17 deletions(-) diff --git a/sound/soc/intel/avs/boards/da7219.c b/sound/soc/intel/avs/boards/da7219.c index 80c0a1a956542..daf53ca490a14 100644 --- a/sound/soc/intel/avs/boards/da7219.c +++ b/sound/soc/intel/avs/boards/da7219.c @@ -211,21 +211,6 @@ static int avs_create_dai_link(struct device *dev, const char *platform_name, in return 0; } -static int avs_card_suspend_pre(struct snd_soc_card *card) -{ - struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME); - - return snd_soc_component_set_jack(codec_dai->component, NULL, NULL); -} - -static int avs_card_resume_post(struct snd_soc_card *card) -{ - struct snd_soc_dai *codec_dai = snd_soc_card_get_codec_dai(card, DA7219_DAI_NAME); - struct snd_soc_jack *jack = snd_soc_card_get_drvdata(card); - - return snd_soc_component_set_jack(codec_dai->component, jack, NULL); -} - static int avs_da7219_probe(struct platform_device *pdev) { struct snd_soc_dai_link *dai_link; @@ -257,8 +242,6 @@ static int avs_da7219_probe(struct platform_device *pdev) card->name = "avs_da7219"; card->dev = dev; card->owner = THIS_MODULE; - card->suspend_pre = avs_card_suspend_pre; - card->resume_post = avs_card_resume_post; card->dai_link = dai_link; card->num_links = 1; card->controls = card_controls; -- 2.47.0.338.g60cca15819-goog

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No microphone USB audio Behringer UMC204HD ALSA update
by GitHub issues - edited 02 Dec '24

02 Dec '24

alsa-project/alsa-ucm-conf issue #479 was edited from emiltoacs: No input on my Behringer UMC204HD because of an update in the following packages on Arch Linux alsa-card-profiles,alsa-lib,alsa-ucm-conf,alsa-utils,lib32-alsa-lib alsa-card-profiles 1:1.2.6-1 -> 1:1.2.7-1 alsa-lib 1.2.12-1 -> 1.2.13-1 alsa-ucm-conf 1.2.12-1 -> 1.2.13-2 alsa-utils 1.2.12-1 -> 1.2.13-2 lib32-alsa-lib 1.2.12-1 -> 1.2.13-1 The error messages related by pipewire : ``` pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) graph xrun not-triggered (0 suppressed) pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) xrun state:0x7ba2cfaed008 pending:6/6 s:0 a:0 f:0 waiting:0 process:0 status:triggered pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) graph xrun not-triggered (4 suppressed) pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) xrun state:0x7ba2cfaed008 pending:1/1 s:2663559587040 a:2663559606728 f:2663559630434 waiting:19688 process:23706 status:triggered pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) graph xrun not-triggered (0 suppressed) pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-57) xrun state:0x7ba2cfaed008 pending:1/1 s:5553029600968 a:5553029610813 f:5553029623757 waiting:9845 process:12944 status:triggered ``` If I revert package the package to these version it works again : alsa-card-profiles-1:1.2.6-1-x86_64.pkg.tar.zst alsa-lib-1.2.12-1-x86_64.pkg.tar.zst lib32-alsa-lib-1.2.12-1-x86_64.pkg.tar.zst alsa-utils-1.2.12-1-x86_64.pkg.tar.zst alsa-ucm-conf-1.2.12-1-any.pkg.tar.zst In the meantime I add : --ignore alsa-card-profiles,alsa-lib,alsa-ucm-conf,alsa-utils,lib32-alsa-lib to my pacman update. **EDIT** only the alsa-ucm-conf package is faulty somewhere between these two version alsa-ucm-conf 1.2.12-1 -> 1.2.13-2 Workaround: Downgrade to alsa-ucm-conf-1.2.12-1 With this package built by the maintainer @christian-heusel for testing https://pkgbuild.com/\~gromit/alsa-ucm-conf-v1.2.13.r6.g62acef2-1-any.pkg.t… the microphone seens to work but in the logs I have the same error message as above : ``` pipewire[2522]: pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-56) graph xrun not-triggered (0 suppressed) pipewire[2522]: pw.node: (alsa_output.pci-0000_00_1f.3.stereo-fallback-56) xrun state:0x7e7d78f97008 pending:6/6 s:69949980861 a:69950023367 f:69950232612 waiting:42506 process:209245 status:triggered ``` although I manage to receive the signal of my microphone. See https://bbs.archlinux.org/viewtopic.php?pid=2211786 for the details of our discussion with @christian-heusel Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/479 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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Unable to request ucm2 to load BootSequence controls defaults
by GitHub issues - opened 02 Dec '24

02 Dec '24

alsa-project/alsa-ucm-conf issue #480 was opened from singalsu: I'm stuck with my ucm2 patch to set by default control switches for SOF DRC and beamformer "on" when the controls appear for first time. I don't seem to be able to reset the controls for the SOF sound card to defaults. There's a similar feature in https://github.com/alsa-project/alsa-ucm-conf/blob/62acef2efffdcb75a35f90d2… for "Dmic0 Capture Volume". But I haven't been able to see the volume to go to 70% in my tries to reset the configuration. In Fedora 41 I've tried to remove /var/lib/alsa/asound.state and rename /usr/sbin/alsactl, reboot, and rename it back. But no success. The volume control remains in 100%. Similarly my tries to get DRC and TDFB enabled fail. What is the right trick needed to trigger the BootSequence? Issue URL : https://github.com/alsa-project/alsa-ucm-conf/issues/480 Repository URL: https://github.com/alsa-project/alsa-ucm-conf

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[PATCH AUTOSEL 5.10 4/6] ASoC: stm: Prevent potential division by zero in stm32_sai_mclk_round_rate()
by Sasha Levin 02 Dec '24

02 Dec '24

From: Luo Yifan <luoyifan(a)cmss.chinamobile.com> [ Upstream commit 63c1c87993e0e5bb11bced3d8224446a2bc62338 ] This patch checks if div is less than or equal to zero (div <= 0). If div is zero or negative, the function returns -EINVAL, ensuring the division operation (*prate / div) is safe to perform. Signed-off-by: Luo Yifan <luoyifan(a)cmss.chinamobile.com> Link: https://patch.msgid.link/20241106014654.206860-1-luoyifan@cmss.chinamobile.… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/stm/stm32_sai_sub.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/stm/stm32_sai_sub.c b/sound/soc/stm/stm32_sai_sub.c index 3aa1cf2624020..3a7f0102b4c5c 100644 --- a/sound/soc/stm/stm32_sai_sub.c +++ b/sound/soc/stm/stm32_sai_sub.c @@ -380,8 +380,8 @@ static long stm32_sai_mclk_round_rate(struct clk_hw *hw, unsigned long rate, int div; div = stm32_sai_get_clk_div(sai, *prate, rate); - if (div < 0) - return div; + if (div <= 0) + return -EINVAL; mclk->freq = *prate / div; -- 2.43.0

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