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

[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

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[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

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

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