January 2025 - Alsa-devel - mailman.alsa-project.org

[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

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[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

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[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

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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.

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[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

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[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

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[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

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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

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[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(-)

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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

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