March 2024 - Alsa-devel - mailman.alsa-project.org

[PATCH 0/5] ALSA: cirrus: Tidy up of firmware control read/write
by Richard Fitzgerald 03 Apr '24

03 Apr '24

This set of patches factors out some repeated code to clean up firmware control read/write functions, and removes some redundant control notification code. Simon Trimmer (5): firmware: cs_dsp: Add locked wrappers for coeff read and write ASoC: wm_adsp: Use cs_dsp_coeff_lock_and_[read|write]_ctrl() ALSA: hda: hda_cs_dsp_ctl: Use cs_dsp_coeff_lock_and_[read|write]_ctrl() ASoC: wm_adsp: Remove notification of driver write ALSA: hda: hda_cs_dsp_ctl: Remove notification of driver write drivers/firmware/cirrus/cs_dsp.c | 54 ++++++++++++++++++++++++++ include/linux/firmware/cirrus/cs_dsp.h | 4 ++ sound/pci/hda/hda_cs_dsp_ctl.c | 22 +---------- sound/soc/codecs/wm_adsp.c | 32 ++++----------- 4 files changed, 67 insertions(+), 45 deletions(-) -- 2.39.2

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[PATCH v12] ASoc: tas2783: Add tas2783 codec driver
by Shenghao Ding 02 Apr '24

02 Apr '24

The tas2783 is a smart audio amplifier with integrated MIPI SoundWire interface (Version 1.2.1 compliant), I2C, and I2S/TDM interfaces designed for portable applications. An on-chip DSP supports Texas Instruments SmartAmp speaker protection algorithm. The integrated speaker voltage and current sense provides for real-time monitoring of loudspeakers. The ASoC component provides the majority of the functionality of the device, all the audio functions. Signed-off-by: Shenghao Ding <shenghao-ding(a)ti.com> --- Change in v12: - simplify tasdevice_set_sdw_stream - fixed some Linux coding style - fixed the spelling mistakes - Select left/right channel based on unique id - a longer description has been added - remove unused crc8 in KCONFIG - Invert of RE here means reciprocal of RE - correct indentation - correct the wrong pow calibrated register - correct comment on RESET register - fixed usually lines don't end with an open parenthesis - fixed tasdevice_sdw_shutdown - use defines for function, entity, etc. for readable - remove unnecessary codec_lock in component_probe() - remove struct sdw_stream_data - remove unused macro definitions - use BIT(2) and BIT(1) instead of 0x2 and 0x1 - implement pm_runtime - reg_start is u8, reg_start + i only adds i byte, it should be reg_start + i * 4, because the calibration values - fixed total_sz in tas2783_calibration - add a define for TASDEVICE_REG(0, 0, 0x0a) - rework macro style in header file - Add select CRC32 - rewrite the mute and the return value handling - Add comments on bin file - Add link_id into the fw name to distinguish the same sdw id on different sdw id. - remove unnecessary mutex_lock - use set_active() in tasdevice_io_init and move all other parts in the probe. - move pm_runtime_get_noresume to the probe - fix the return value of putvol function - add put_noidle in tasdevice_sdw_remove - remove unused header file(of.h) - correct the style of the multi-line comment - Set the readable registers in tas2783_readable_register for debugfs - Use new PM macros - Use default put and get inline SOC_SINGLE_RANGE_TLV and SOC_SINGLE_SX_TLV - Use respective %pt to log the calibrated time - Remove unnecessary struct tm - Add trailing comma tasdevice_dapm_widgets and tasdevice_audio_map - Rewrite the return value handling in tasdevice_sdw_probe - use dev_num instead of id.unique_id - rework the log - Add code to check the size of calibrated data in UEFI - Add description of Layout of calibrated Data in UEFI - switch back link_id and unique_id to identify the devcie - unique_id offset as the device index of the calibrated data - update calibrated data layout stored in UEFI, add link_id and unique_id into the calibrated data in case of tas2783 sitting in different SDQW buses. - correct the wrong unique_id - set the unique_id as hex style in dsp bin file name - use __func__ in the dev_err message in two identical logs intas2783_apply_calibv1/v2. - In tas2783_apply_calibv2, the very least move initializations to a different line. - No need for separate j variable, reuse i in tasdevice_read_prop. - remove unused "struct sdw_bus_params params", unsigned char dev_name[32] pstream, and chip_id - define dspfw_binaryname as a local variable in tasdevice_comp_probe. - reduce a lot of indentation in tasdevice_mute and spilte tasdevice_mute into tasdevice_enable_echo_ref and tasdevice_playback - remove tas2783_reg_defaults - use REGCACHE_MAPLE instead of REGCACHE_RBTREE - define fw_node_sz to store the node size - remove "struct snd_soc_component *component" - Add some SDCA registers as readable - remove unnecessary dev_dbg log for timestamp - playback still work without calibrated data stored in UEFI, for example bypass mode. Even if in case of dsp mode, algo can still work with default calibrated data. So, not an error, use dev_info for them --- sound/soc/codecs/Kconfig | 16 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas2783-sdw.c | 892 +++++++++++++++++++++++++++++++++ sound/soc/codecs/tas2783.h | 106 ++++ 4 files changed, 1016 insertions(+) create mode 100644 sound/soc/codecs/tas2783-sdw.c create mode 100644 sound/soc/codecs/tas2783.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f78ea2f86fa6..939384837d5e 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -248,6 +248,7 @@ config SND_SOC_ALL_CODECS imply SND_SOC_TAS2781_COMLIB imply SND_SOC_TAS2781_FMWLIB imply SND_SOC_TAS2781_I2C + imply SND_SOC_TAS2783 imply SND_SOC_TAS5086 imply SND_SOC_TAS571X imply SND_SOC_TAS5720 @@ -1880,6 +1881,21 @@ config SND_SOC_TAS2781_I2C algo coefficient setting, for one, two or even multiple TAS2781 chips. +config SND_SOC_TAS2783 + tristate "Texas Instruments TAS2783 speaker amplifier (sdw)" + depends on SOUNDWIRE + depends on EFI + select REGMAP + select REGMAP_SOUNDWIRE + select CRC32 + select CRC32_SARWATE + help + Enable support for Texas Instruments TAS2783 Smart Amplifier + Digital input mono Class-D and DSP-inside audio power amplifiers. + TAS2783 supports the MIPI SoundWire v1.2 compatible interface + for audio and control data. At the end of calibrated data appends + the crc data with crc32_sarwate for error detection. + config SND_SOC_TAS5086 tristate "Texas Instruments TAS5086 speaker amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 7c075539dc47..6f27f810affc 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -287,6 +287,7 @@ snd-soc-tas2770-objs := tas2770.o snd-soc-tas2781-comlib-objs := tas2781-comlib.o snd-soc-tas2781-fmwlib-objs := tas2781-fmwlib.o snd-soc-tas2781-i2c-objs := tas2781-i2c.o +snd-soc-tas2783-objs := tas2783-sdw.o snd-soc-tfa9879-objs := tfa9879.o snd-soc-tfa989x-objs := tfa989x.o snd-soc-tlv320adc3xxx-objs := tlv320adc3xxx.o @@ -672,6 +673,7 @@ obj-$(CONFIG_SND_SOC_TAS2780) += snd-soc-tas2780.o obj-$(CONFIG_SND_SOC_TAS2781_COMLIB) += snd-soc-tas2781-comlib.o obj-$(CONFIG_SND_SOC_TAS2781_FMWLIB) += snd-soc-tas2781-fmwlib.o obj-$(CONFIG_SND_SOC_TAS2781_I2C) += snd-soc-tas2781-i2c.o +obj-$(CONFIG_SND_SOC_TAS2783) += snd-soc-tas2783.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TAS571X) += snd-soc-tas571x.o obj-$(CONFIG_SND_SOC_TAS5720) += snd-soc-tas5720.o diff --git a/sound/soc/codecs/tas2783-sdw.c b/sound/soc/codecs/tas2783-sdw.c new file mode 100644 index 000000000000..429191b79f8d --- /dev/null +++ b/sound/soc/codecs/tas2783-sdw.c @@ -0,0 +1,892 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier +// +// Copyright (C) 2023 - 2024 Texas Instruments Incorporated +// https://www.ti.com +// +// Author: Baojun Xu <baojun.xu(a)ti.com> +// Kevin Lu <kevin-lu(a)ti.com> +// Shenghao Ding <shenghao-ding(a)ti.com> +// + +#include <asm/unaligned.h> +#include <linux/acpi.h> +#include <linux/crc32.h> +#include <linux/efi.h> +#include <linux/err.h> +#include <linux/firmware.h> +#include <linux/init.h> +#include <linux/module.h> +#include <linux/pm.h> +#include <linux/pm_runtime.h> +#include <linux/regmap.h> +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_registers.h> +#include <linux/soundwire/sdw_type.h> +#include <sound/pcm_params.h> +#include <sound/sdw.h> +#include <sound/soc.h> +#include <sound/tlv.h> + +#include <sound/tas2781-tlv.h> +#include "tas2783.h" + +struct calibdatav2_info { + unsigned int number_of_devices; + unsigned int crc32_pos; + unsigned int byt_sz; + unsigned int *cali_data; +}; + +static const unsigned int tas2783_cali_reg[] = { + TAS2783_CALIBRATION_RE, /* Resistance */ + TAS2783_CALIBRATION_RE_LOW, /* Low limitation of RE */ + TAS2783_CALIBRATION_INV_RE, /* Reciprocal of RE */ + TAS2783_CALIBRATION_POW, /* RMS Power */ + TAS2783_CALIBRATION_TLIMIT /* Temperature limitation */ +}; + +static bool tas2783_readable_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + case 0x8000 ... 0x807F: + case SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_MUTE, 0): + case SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU21, + TAS2783_SDCA_CTL_FU_VOLUME, 0): + case SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0): + return true; + default: + return false; + } +} + +static bool tas2783_volatile_register(struct device *dev, + unsigned int reg) +{ + switch (reg) { + /* + * Only software reset register was volatiled. + * Bit is self clearing. + * 0b = Don't reset + * 1b = reset + */ + case 0x8001: + return true; + default: + return false; + } +} + +static const struct regmap_config tasdevice_regmap = { + .reg_bits = 32, + .val_bits = 8, + .readable_reg = tas2783_readable_register, + .volatile_reg = tas2783_volatile_register, + .max_register = 0x44ffffff, + .cache_type = REGCACHE_MAPLE, + .use_single_read = true, + .use_single_write = true, +}; + +static const struct snd_kcontrol_new tas2783_snd_controls[] = { + SOC_SINGLE_RANGE_TLV("Digital Volume", TAS2783_DVC_LVL, + 0, 0, 200, 1, dvc_tlv), + SOC_SINGLE_SX_TLV("Amp Gain Volume", TAS2783_AMP_LEVEL, + 1, 0, 20, amp_vol_tlv), +}; + +/* + * Load the calibration data, including speaker impedance, f0, etc. + * Calibration is done by the manufacturer in the factory. The calibration + * data are used by the algorithm for calculating the speaker temperature, + * speaker membrane excursion and f0 in real time during playback. + * The DSP will work with default data values if calibrated data are + * missing or are invalid. + * Layout of calibrated Data V1 in UEFI(total 256 bytes): + * Calibrated Data of Dev 0(unique_id offset 0) (20 bytes) + * Calibrated Data of Dev 1(unique_id offset 1) (20 bytes) + * Calibrated Data of Dev 2(unique_id offset 2) (20 bytes) + * Calibrated Data of Dev 3(unique_id offset 3) (20 bytes) + * Calibrated Data of Dev 4(unique_id offset 4) (20 bytes) + * Calibrated Data of Dev 5(unique_id offset 5) (20 bytes) + * Calibrated Data of Dev 6(unique_id offset 6) (20 bytes) + * Calibrated Data of Dev 7(unique_id offset 7) (20 bytes) + * TimeStamp of Calibration (4 bytes) + * CRC (4 bytes) + * Reserved (88 bytes) + */ +static void tas2783_apply_calibv1(struct tasdevice_priv *tas_dev, + unsigned int *cali_data) +{ + struct regmap *map = tas_dev->regmap; + u8 *cali_start; + u16 offset; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", + __func__); + return; + } + + offset = tas_dev->sdw_peripheral->id.unique_id - + TAS2783_UNIQUE_ID_BASE; + if (offset >= TAS2783_MAX_DEV_NUM) { + dev_err(tas_dev->dev, "%s: offset(%u) is out of range.\n", + __func__, offset); + return; + } + + /* + * The area saving tas2783 calibrated data is specified by its + * unique_id offset. cali_start is the first address of current + * tas2783's calibrated data. + */ + cali_start = (u8 *)(cali_data + offset * sizeof(tas2783_cali_reg)); + for (int i = 0; i < ARRAY_SIZE(tas2783_cali_reg); i++) { + ret = regmap_bulk_write(map, tas2783_cali_reg[i], + &cali_start[4 * i], 4); + if (ret) { + dev_err(tas_dev->dev, "%s: Cali failed %x:%d\n", + __func__, tas2783_cali_reg[i], ret); + break; + } + } +} + +/* + * New features were added in calibrated Data V2: + * 1. Added some the fields to store the link_id and uniqie_id for + * multi-link solutions + * 2. Support flexible number of devices instead of fixed one in V1. + * Layout of calibrated data V2 in UEFI(total 256 bytes): + * ChipID (2783, 4 bytes) + * Device-Sum (4 bytes) + * TimeStamp of Calibration (4 bytes) + * for (i = 0; i < Device-Sum; i++) { + * Device #i index_info () { + * SDW link id (2bytes) + * SDW unique_id (2bytes) + * } + * Calibrated Data of Device #i (20 bytes) + * } + * CRC (4 bytes) + * Reserved (the rest) + */ +static void tas2783_apply_calibv2(struct tasdevice_priv *tas_dev, + unsigned int *tmp_val) +{ + const unsigned int arr_size = ARRAY_SIZE(tas2783_cali_reg); + const struct calibdatav2_info calib_info = { + .number_of_devices = tmp_val[1], + .crc32_pos = 3 + tmp_val[1] * 6, + .byt_sz = 12 + tmp_val[1] * 24, + .cali_data = &tmp_val[3] + }; + struct regmap *map = tas_dev->regmap; + unsigned int i, j, k, crc; + u8 *cali_start; + u16 dev_indx; + int ret; + + if (!tas_dev->sdw_peripheral) { + dev_err(tas_dev->dev, "%s: peripheral doesn't exist.\n", + __func__); + return; + } + + if (calib_info.number_of_devices > TAS2783_MAX_DEV_NUM || + calib_info.number_of_devices == 0) { + dev_info(tas_dev->dev, + "using default calibration data, wrong data in EFI."); + return; + } + crc = crc32(~0, tmp_val, calib_info.byt_sz) ^ ~0; + if (crc != tmp_val[calib_info.crc32_pos]) { + dev_info(tas_dev->dev, + "using default calibration data, crc not match."); + return; + } + + dev_indx = tas_dev->sdw_peripheral->bus->link_id | + tas_dev->sdw_peripheral->id.unique_id << 16; + + for (i = 0; i < calib_info.number_of_devices; i++) { + k = i * (arr_size + 1); + + if (dev_indx != calib_info.cali_data[k]) + continue; + + /* +1: skip 4 byte dev-indx info */ + cali_start = (u8 *)&calib_info.cali_data[k + 1]; + + for (j = 0; j < arr_size; j++) { + k = 4 * j; + ret = regmap_bulk_write(map, tas2783_cali_reg[j], + &cali_start[k], 4); + if (ret) { + dev_err(tas_dev->dev, + "%s: Cali data writing failed %x:%d\n", + __func__, tas2783_cali_reg[j], ret); + break; + } + } + return; + } +} + +static void tas2783_calibration(struct tasdevice_priv *tas_dev) +{ + efi_guid_t efi_guid = EFI_GUID(0x1f52d2a1, 0xbb3a, 0x457d, 0xbc, + 0x09, 0x43, 0xa3, 0xf4, 0x31, 0x0a, 0x92); + static efi_char16_t efi_name[] = L"CALI_DATA"; + struct calibration_data cali_data; + unsigned int *tmp_val; + unsigned int crc; + efi_status_t status; + + cali_data.total_sz = 0; + + status = efi.get_variable(efi_name, &efi_guid, NULL, + &cali_data.total_sz, NULL); + if (status == EFI_BUFFER_TOO_SMALL + && cali_data.total_sz < TAS2783_MAX_CALIDATA_SIZE) { + status = efi.get_variable(efi_name, &efi_guid, NULL, + &cali_data.total_sz, + cali_data.data); + dev_dbg(tas_dev->dev, "%s: cali get %lx bytes result:%ld\n", + __func__, cali_data.total_sz, status); + } + if (status != 0) { + /* Failed got calibration data from EFI. */ + dev_dbg(tas_dev->dev, "No calibration data in UEFI."); + return; + } + + tmp_val = (unsigned int *)cali_data.data; + + /* Check Calibrated Data V1 */ + crc = crc32(~0, cali_data.data, TAS2783_CALIDATAV1_BYTE_SIZE) ^ ~0; + if (crc == tmp_val[TAS2783_CALIDATAV1_CRC32_INDX]) { + /* Date and time of when calibration was done. */ + tas2783_apply_calibv1(tas_dev, tmp_val); + return; + } + + /* Check Calibrated Data V2 */ + if (tmp_val[0] != 2783) { + dev_err(tas_dev->dev, "No tas2783 calibration data\n"); + return; + } + + tas2783_apply_calibv2(tas_dev, tmp_val); +} + +static void tasdevice_dspfw_ready(const struct firmware *fmw, + void *context) +{ + struct tasdevice_priv *tas_dev = (struct tasdevice_priv *) context; + struct regmap *map = tas_dev->regmap; + struct tas2783_firmware_node *p; + unsigned char *buf; + int fw_node_sz = sizeof(struct tas2783_firmware_node); + int img_sz, ret; + int offset = 0; + + if (!fmw || !fmw->data) { + dev_warn(tas_dev->dev, + "%s: failed to read fw, work in bypass-dsp mode.\n", + __func__); + return; + } + buf = (unsigned char *)fmw->data; + + img_sz = get_unaligned_le32(&buf[offset]); + offset += sizeof(img_sz); + if (img_sz != fmw->size) { + dev_warn(tas_dev->dev, "%s: size not matching, %d %u.", + __func__, (int)fmw->size, img_sz); + return; + } + + while (offset < img_sz) { + p = (struct tas2783_firmware_node *)(buf + offset); + if (p->length > 1) + ret = regmap_bulk_write(map, p->download_addr, + buf + offset + fw_node_sz, p->length); + else + ret = regmap_write(map, p->download_addr, + *(buf + offset + fw_node_sz)); + + if (ret != 0) { + dev_dbg(tas_dev->dev, + "%s: load FW fail: %d, work in bypass.\n", + __func__, ret); + return; + } + offset += fw_node_sz + p->length; + } + + tas2783_calibration(tas_dev); +} + +static const struct snd_soc_dapm_widget tasdevice_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("ASI", "ASI Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("ASI OUT", "ASI Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_OUTPUT("OUT"), + SND_SOC_DAPM_INPUT("DMIC"), +}; + +static const struct snd_soc_dapm_route tasdevice_audio_map[] = { + {"OUT", NULL, "ASI"}, + {"ASI OUT", NULL, "DMIC"}, +}; + +static int tasdevice_set_sdw_stream(struct snd_soc_dai *dai, + void *sdw_stream, int direction) +{ + snd_soc_dai_dma_data_set(dai, direction, sdw_stream); + + return 0; +} + +static void tasdevice_sdw_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + snd_soc_dai_set_dma_data(dai, substream, NULL); +} + +static int tasdevice_sdw_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_config stream_config = {0}; + struct sdw_port_config port_config = {0}; + struct sdw_stream_runtime *sdw_stream; + int ret; + + dev_dbg(dai->dev, "%s: dai_name %s", __func__, dai->name); + sdw_stream = snd_soc_dai_get_dma_data(dai, substream); + + if (!sdw_stream) + return -EINVAL; + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + /* SoundWire specific configuration */ + snd_sdw_params_to_config(substream, params, + &stream_config, &port_config); + + /* port 1 for playback */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + port_config.num = 1; + else + port_config.num = 2; + + ret = sdw_stream_add_slave(tas_priv->sdw_peripheral, + &stream_config, &port_config, 1, sdw_stream); + if (ret) { + dev_err(dai->dev, "%s: Unable to configure port\n", __func__); + return ret; + } + + dev_dbg(dai->dev, "%s: format: %i rate: %i\n", __func__, + params_format(params), params_rate(params)); + + return 0; +} + +static int tasdevice_sdw_pcm_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_priv = + snd_soc_component_get_drvdata(component); + struct sdw_stream_runtime *sdw_stream = + snd_soc_dai_get_dma_data(dai, substream); + + if (!tas_priv->sdw_peripheral) + return -EINVAL; + + sdw_stream_remove_slave(tas_priv->sdw_peripheral, sdw_stream); + + return 0; +} + +/* Capture stream is the echo ref data for voice mode.*/ +static int tasdevice_enable_echo_ref(struct tasdevice_priv *tas_dev, + int mute) +{ + struct regmap *map = tas_dev->regmap; + unsigned int val; + int ret; + + if (mute) + val = TAS2783_REG_AEF_INACTIVE; + else + val = TAS2783_REG_AEF_ACTIVE; + + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_AEF_MASK, TAS2783_REG_AEF_INACTIVE); + if (ret) + dev_err(tas_dev->dev, "%s: Disable AEF failed.\n", __func__); + + return ret; +} + +static int tasdevice_playback(struct tasdevice_priv *tas_dev, int mute) +{ + struct regmap *map = tas_dev->regmap; + int ret; + + if (mute) { + /* FU23 mute (0x40400108) */ + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 1); + if (ret) { + dev_err(tas_dev->dev, "%s: FU23 mute failed.\n", + __func__); + goto out; + } + + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK | + TAS2783_REG_AEF_MASK, + TAS2783_REG_PWR_MODE_ACTIVE | + TAS2783_REG_PWR_MODE_SW_PWD); + if (ret) { + dev_err(tas_dev->dev, "%s: shutdown failed.\n", + __func__); + } + } else { + /* FU23 Unmute, 0x40400108. */ + ret = regmap_write(map, + SDW_SDCA_CTL(TAS2783_FUNC_TYPE_SMART_AMP, + TAS2783_SDCA_ENT_FU23, + TAS2783_SDCA_CTL_FU_MUTE, 0), 0); + if (ret) { + dev_err(tas_dev->dev, + "%s: FU23 Unmute failed.\n", __func__); + goto out; + } + ret = regmap_update_bits(map, TAS2873_REG_PWR_CTRL, + TAS2783_REG_PWR_MODE_MASK, + TAS2783_REG_PWR_MODE_ACTIVE); + if (ret) { + dev_err(tas_dev->dev, + "%s: PWR Unmute failed.\n", __func__); + } + } +out: + return ret; +} + +static int tasdevice_mute(struct snd_soc_dai *dai, int mute, + int direction) +{ + struct snd_soc_component *component = dai->component; + struct tasdevice_priv *tas_dev = + snd_soc_component_get_drvdata(component); + + int ret; + + dev_dbg(tas_dev->dev, "%s: %d.\n", __func__, mute); + + if (direction == SNDRV_PCM_STREAM_CAPTURE) + ret = tasdevice_enable_echo_ref(tas_dev, mute); + else + ret = tasdevice_playback(tas_dev, mute); + + if (ret) + dev_err(tas_dev->dev, "Mute or unmute %d failed %d.\n", + mute, ret); + + return ret; +} + +static const struct snd_soc_dai_ops tasdevice_dai_ops = { + .mute_stream = tasdevice_mute, + .hw_params = tasdevice_sdw_hw_params, + .hw_free = tasdevice_sdw_pcm_hw_free, + .set_stream = tasdevice_set_sdw_stream, + .shutdown = tasdevice_sdw_shutdown, +}; + +static struct snd_soc_dai_driver tasdevice_dai_driver[] = { + { + .name = "tas2783-codec", + .id = 0, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = TAS2783_DEVICE_RATES, + .formats = TAS2783_DEVICE_FORMATS, + }, + .ops = &tasdevice_dai_ops, + .symmetric_rate = 1, + }, +}; + +static int tasdevice_comp_probe(struct snd_soc_component *comp) +{ + struct tasdevice_priv *tas_dev = snd_soc_component_get_drvdata(comp); + unsigned char dspfw_binaryname[TAS2783_DSPFW_FILENAME_LEN]; + acpi_handle handle = ACPI_HANDLE(tas_dev->dev); + const struct firmware *fw_entry = NULL; + const char *sub = NULL; + int ret; + + if (handle) { + sub = acpi_get_subsystem_id(handle); + if (IS_ERR(sub)) + sub = NULL; + } + + /* + * Each tas2783 in the system has its own dspfw. + */ + if (comp->name_prefix) { + /* + * name_prefix.bin stores the dsp firmware including speaker + * protection algorithm, audio acoustic algorithm, speaker + * characters and algorithm params, it must be copied into + * firmware folder. + */ + scnprintf(dspfw_binaryname, TAS2783_DSPFW_FILENAME_LEN, + "%s-tas2783.bin", comp->name_prefix); + } else { + /* Compatible with the previous naming rule */ + if (sub) { + /* + * subsys_id-link_id[0,1,...,N]-unique_id + * [0x8,...,0xf].bin stores the dsp firmware including + * speaker protection algorithm, audio acoustic + * algorithm, speaker characters and algorithm params, + * it must be copied into firmware folder. + */ + scnprintf(dspfw_binaryname, TAS2783_DSPFW_FILENAME_LEN, + "%s-%d-0x%x.bin", sub, + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->id.unique_id); + } else { + /* + * tas2783-link_id[0,1,...,N]-unique_id + * [0x8,...,0xf].bin stores the dsp firmware including + * speaker protection algorithm, audio acoustic + * algorithm, speaker characters and algorithm params, + * it must be copied into firmware folder. + */ + scnprintf(dspfw_binaryname, TAS2783_DSPFW_FILENAME_LEN, + "tas2783-%d-0x%x.bin", + tas_dev->sdw_peripheral->bus->link_id, + tas_dev->sdw_peripheral->id.unique_id); + } + } + + ret = request_firmware(&fw_entry, dspfw_binaryname, tas_dev->dev); + if (ret) { + dev_err(tas_dev->dev, + "%s: request_firmware %x open status: %d.\n", __func__, + tas_dev->sdw_peripheral->id.unique_id, ret); + goto out; + } + + tasdevice_dspfw_ready(fw_entry, tas_dev); + +out: + if (fw_entry) + release_firmware(fw_entry); + return 0; +} + +static const struct snd_soc_component_driver + soc_codec_driver_tasdevice = { + .probe = tasdevice_comp_probe, + .controls = tas2783_snd_controls, + .num_controls = ARRAY_SIZE(tas2783_snd_controls), + .dapm_widgets = tasdevice_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(tasdevice_dapm_widgets), + .dapm_routes = tasdevice_audio_map, + .num_dapm_routes = ARRAY_SIZE(tasdevice_audio_map), + .idle_bias_on = 1, + .endianness = 1, +}; + +static int tasdevice_init(struct tasdevice_priv *tas_dev) +{ + int ret; + + dev_set_drvdata(tas_dev->dev, tas_dev); + + ret = devm_snd_soc_register_component(tas_dev->dev, + &soc_codec_driver_tasdevice, tasdevice_dai_driver, + ARRAY_SIZE(tasdevice_dai_driver)); + if (ret) { + dev_err(tas_dev->dev, "%s: codec register error:%d.\n", + __func__, ret); + return ret; + } + + /* set autosuspend parameters */ + pm_runtime_set_autosuspend_delay(tas_dev->dev, 3000); + pm_runtime_use_autosuspend(tas_dev->dev); + + /* make sure the device does not suspend immediately */ + pm_runtime_mark_last_busy(tas_dev->dev); + pm_runtime_enable(tas_dev->dev); + + return 0; +} + +static int tasdevice_read_prop(struct sdw_slave *slave) +{ + struct sdw_slave_prop *prop = &slave->prop; + struct sdw_dpn_prop *dpn; + unsigned long addr; + int nval, i; + u32 bit; + + prop->scp_int1_mask = SDW_SCP_INT1_BUS_CLASH | SDW_SCP_INT1_PARITY; + prop->quirks = SDW_SLAVE_QUIRKS_INVALID_INITIAL_PARITY; + + prop->paging_support = true; + + /* first we need to allocate memory for set bits in port lists */ + prop->source_ports = BIT(2); /* BITMAP: 00000100 */ + prop->sink_ports = BIT(1); /* BITMAP: 00000010 */ + + nval = hweight32(prop->source_ports); + prop->src_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->src_dpn_prop), GFP_KERNEL); + if (!prop->src_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->src_dpn_prop; + addr = prop->source_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* do this again for sink now */ + nval = hweight32(prop->sink_ports); + prop->sink_dpn_prop = devm_kcalloc(&slave->dev, nval, + sizeof(*prop->sink_dpn_prop), GFP_KERNEL); + if (!prop->sink_dpn_prop) + return -ENOMEM; + + i = 0; + dpn = prop->sink_dpn_prop; + addr = prop->sink_ports; + for_each_set_bit(bit, &addr, 32) { + dpn[i].num = bit; + dpn[i].type = SDW_DPN_FULL; + dpn[i].simple_ch_prep_sm = true; + dpn[i].ch_prep_timeout = 10; + i++; + } + + /* set the timeout values */ + prop->clk_stop_timeout = 20; + + return 0; +} + +static int tasdevice_io_init(struct device *dev, + struct sdw_slave *slave) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + int ret; + + regcache_cache_only(tas_priv->regmap, false); + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, true); + } else { + /* + * PM runtime is only enabled when a Slave reports as Attached + * Update count of parent 'active' children. + */ + pm_runtime_set_active(&slave->dev); + } + + /* sw reset */ + ret = regmap_write(tas_priv->regmap, TAS2873_REG_SWRESET, + TAS2873_REG_SWRESET_RESET); + if (ret) { + dev_err(tas_priv->dev, "Reset failed.\n"); + goto out; + } + + if (tas_priv->first_hw_init) { + regcache_cache_bypass(tas_priv->regmap, false); + regcache_mark_dirty(tas_priv->regmap); + } + + tas_priv->first_hw_init = true; + tas_priv->hw_init = true; + +out: + return ret; +} + +static int tasdevice_update_status(struct sdw_slave *slave, + enum sdw_slave_status status) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(&slave->dev); + + if (status == SDW_SLAVE_UNATTACHED) + tas_priv->hw_init = false; + /* + * Perform initialization only if slave status + * is present and hw_init flag is false + */ + if (tas_priv->hw_init || status != SDW_SLAVE_ATTACHED) + return 0; + + /* perform I/O transfers required for Slave initialization */ + return tasdevice_io_init(&slave->dev, slave); +} + +/* + * slave_ops: callbacks for get_clock_stop_mode, clock_stop and + * port_prep are not defined for now + */ +static const struct sdw_slave_ops tasdevice_sdw_ops = { + .read_prop = tasdevice_read_prop, + .update_status = tasdevice_update_status, +}; + +static int tasdevice_sdw_probe(struct sdw_slave *peripheral, + const struct sdw_device_id *id) +{ + struct device *dev = &peripheral->dev; + struct tasdevice_priv *tas_dev; + + tas_dev = devm_kzalloc(dev, sizeof(*tas_dev), GFP_KERNEL); + if (!tas_dev) + return -ENOMEM; + + tas_dev->dev = dev; + tas_dev->sdw_peripheral = peripheral; + /* + * Mark hw_init to false + * HW init will be performed when device reports present + */ + tas_dev->hw_init = false; + tas_dev->first_hw_init = false; + + dev_set_drvdata(dev, tas_dev); + + tas_dev->regmap = devm_regmap_init_sdw(peripheral, &tasdevice_regmap); + if (IS_ERR(tas_dev->regmap)) + return dev_err_probe(dev, PTR_ERR(tas_dev->regmap), + "Failed devm_regmap_init_sdw."); + + return tasdevice_init(tas_dev); +} + +static int tasdevice_sdw_remove(struct sdw_slave *peripheral) +{ + struct tasdevice_priv *tas_dev = dev_get_drvdata(&peripheral->dev); + + if (tas_dev->first_hw_init) + pm_runtime_disable(tas_dev->dev); + + return 0; +} + +static const struct sdw_device_id tasdevice_sdw_id[] = { + SDW_SLAVE_ENTRY(0x0102, 0x0000, 0), + {} +}; +MODULE_DEVICE_TABLE(sdw, tasdevice_sdw_id); + +static int __maybe_unused tas2783_sdca_dev_suspend(struct device *dev) +{ + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + + if (!tas_priv->hw_init) + return 0; + + regcache_cache_only(tas_priv->regmap, true); + + return 0; +} + +#define TAS2783_PROBE_TIMEOUT_MS 5000 + +static int __maybe_unused tas2783_sdca_dev_resume(struct device *dev) +{ + struct sdw_slave *slave = dev_to_sdw_dev(dev); + struct tasdevice_priv *tas_priv = dev_get_drvdata(dev); + unsigned long time; + + if (!tas_priv->first_hw_init) + return 0; + + if (!slave->unattach_request) + goto regmap_sync; + + time = wait_for_completion_timeout(&slave->initialization_complete, + msecs_to_jiffies(TAS2783_PROBE_TIMEOUT_MS)); + if (!time) { + dev_err(&slave->dev, "Init not complete, timed out\n"); + sdw_show_ping_status(slave->bus, true); + + return -ETIMEDOUT; + } + +regmap_sync: + slave->unattach_request = 0; + regcache_cache_only(tas_priv->regmap, false); + regcache_sync(tas_priv->regmap); + + return 0; +} + +static const struct dev_pm_ops tas2783_sdca_pm = { + SYSTEM_SLEEP_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume) + RUNTIME_PM_OPS(tas2783_sdca_dev_suspend, + tas2783_sdca_dev_resume, NULL) +}; + +static struct sdw_driver tasdevice_sdw_driver = { + .driver = { + .name = "slave-tas2783", + .pm = &tas2783_sdca_pm, + }, + .probe = tasdevice_sdw_probe, + .remove = tasdevice_sdw_remove, + .ops = &tasdevice_sdw_ops, + .id_table = tasdevice_sdw_id, +}; + +module_sdw_driver(tasdevice_sdw_driver); + +MODULE_AUTHOR("Baojun Xu <baojun.xu(a)ti.com>"); +MODULE_AUTHOR("Shenghao Ding <shenghao-ding(a)ti.com>"); +MODULE_DESCRIPTION("ASoC TAS2783 SoundWire Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tas2783.h b/sound/soc/codecs/tas2783.h new file mode 100644 index 000000000000..1b91eae47f29 --- /dev/null +++ b/sound/soc/codecs/tas2783.h @@ -0,0 +1,106 @@ +/* SPDX-License-Identifier: GPL-2.0 + * + * ALSA SoC Texas Instruments TAS2783 Audio Smart Amplifier + * + * Copyright (C) 2023 - 2024 Texas Instruments Incorporated + * https://www.ti.com + * + * Author: Baojun Xu <baojun.xu(a)ti.com> + * Shenghao Ding <shenghao-ding(a)ti.com> + */ + +#ifndef __TAS2783_H__ +#define __TAS2783_H__ + +/* Unique id start */ +#define TAS2783_UNIQUE_ID_BASE 0x08 +#define TAS2783_MAX_DEV_NUM 8 + +#define TAS2783_DEVICE_RATES (SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | \ + SNDRV_PCM_RATE_96000 | \ + SNDRV_PCM_RATE_88200) + +#define TAS2783_DEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* BOOK, PAGE Control Register */ +#define TASDEVICE_REG(book, page, reg) ((book * 256 * 256) + 0x8000 + \ + (page * 128) + reg) + +/*Software Reset */ +#define TAS2873_REG_SWRESET TASDEVICE_REG(0x0, 0x00, 0x01) +#define TAS2873_REG_SWRESET_MASK BIT(0) +#define TAS2873_REG_SWRESET_RESET BIT(0) + +#define TAS2873_REG_PWR_CTRL TASDEVICE_REG(0x0, 0x00, 0x02) +#define TAS2783_REG_AEF_MASK GENMASK(4, 3) +#define TAS2783_REG_AEF_ACTIVE 0 +#define TAS2783_REG_AEF_INACTIVE GENMASK(4, 3) +#define TAS2783_REG_PWR_MODE_MASK GENMASK(2, 0) +#define TAS2783_REG_PWR_MODE_ACTIVE 0 +#define TAS2783_REG_PWR_MODE_SW_PWD BIT(1) + +#define TAS2783_REG_TDM_RX_CFG TASDEVICE_REG(0x0, 0x00, 0x0a) + +/* Volume control */ +#define TAS2783_DVC_LVL TASDEVICE_REG(0x0, 0x00, 0x1a) +#define TAS2783_AMP_LEVEL TASDEVICE_REG(0x0, 0x00, 0x03) +#define TAS2783_AMP_LEVEL_MASK GENMASK(5, 1) + +/* Calibration data */ +#define TAS2783_CALIBRATION_RE TASDEVICE_REG(0x0, 0x17, 0x74) +#define TAS2783_CALIBRATION_RE_LOW TASDEVICE_REG(0x0, 0x18, 0x14) +#define TAS2783_CALIBRATION_INV_RE TASDEVICE_REG(0x0, 0x18, 0x0c) +#define TAS2783_CALIBRATION_POW TASDEVICE_REG(0x0, 0x13, 0x70) +#define TAS2783_CALIBRATION_TLIMIT TASDEVICE_REG(0x0, 0x18, 0x7c) + +/* TAS2783 SDCA Control - function number */ +#define TAS2783_FUNC_TYPE_SMART_AMP 0x01 + +/* TAS2783 SDCA entity */ +#define TAS2783_SDCA_ENT_FU21 0x01 +#define TAS2783_SDCA_ENT_FU23 0x02 + +/* TAS2783 SDCA control */ +#define TAS2783_SDCA_CTL_REQ_POWER_STATE 0x01 +#define TAS2783_SDCA_CTL_FU_MUTE 0x01 +#define TAS2783_SDCA_CTL_FU_VOLUME 0x02 +#define TAS2783_SDCA_CTL_UDMPU_CLUSTER 0x10 + +#define TAS2783_DEVICE_CHANNEL_LEFT 1 +#define TAS2783_DEVICE_CHANNEL_RIGHT 2 +#define TAS2783_DSPFW_FILENAME_LEN 64 + +#define TAS2783_MAX_CALIDATA_SIZE 256 + +#define TAS2783_CALIDATAV1_TIMESTAMP_INDX 40 +#define TAS2783_CALIDATAV1_CRC32_INDX 41 +/* 8 devs * 20 bytes calibrated data/dev + 4 bytes Timestamp */ +#define TAS2783_CALIDATAV1_BYTE_SIZE (8 * 20 + 4) + +#define TAS2783_CALIDATAV2_TIMESTAMP_INDX 2 + +struct tas2783_firmware_node { + unsigned int vendor_id; + unsigned int file_id; + unsigned int version_id; + unsigned int length; + unsigned int download_addr; +}; + +struct calibration_data { + unsigned long total_sz; + unsigned char data[TAS2783_MAX_CALIDATA_SIZE]; +}; + +struct tasdevice_priv { + struct sdw_slave *sdw_peripheral; + struct regmap *regmap; + struct device *dev; + bool hw_init; + bool first_hw_init; +}; + +#endif /*__TAS2783_H__ */ -- 2.34.1

3 2

[PATCH] Add sound quirks for Lenovo Legion slim 7 16ARHA7 models.
by Christian＠bendiksen.me 02 Apr '24

02 Apr '24

From: Christian Bendiksen <christian(a)bendiksen.me> This fixes the sound not working from internal speakers on Lenovo Legion Slim 7 16ARHA7 models. The correct subsystem ID have been added to cs35l41_hda_property.c and patch_realtek.c. Signed-off-by: Christian Bendiksen <christian(a)bendiksen.me> --- sound/pci/hda/cs35l41_hda_property.c | 4 ++++ sound/pci/hda/patch_realtek.c | 2 ++ 2 files changed, 6 insertions(+) diff --git a/sound/pci/hda/cs35l41_hda_property.c b/sound/pci/hda/cs35l41_hda_property.c index 72ec872afb8d..959864994eca 100644 --- a/sound/pci/hda/cs35l41_hda_property.c +++ b/sound/pci/hda/cs35l41_hda_property.c @@ -115,6 +115,8 @@ static const struct cs35l41_config cs35l41_config_table[] = { { "17AA38B5", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, { "17AA38B6", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, { "17AA38B7", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, + { "17AA3877", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, + { "17AA3878", 2, EXTERNAL, { CS35L41_LEFT, CS35L41_RIGHT, 0, 0 }, 0, 1, -1, 0, 0, 0 }, {} }; @@ -503,6 +505,8 @@ static const struct cs35l41_prop_model cs35l41_prop_model_table[] = { { "CSC3551", "17AA38B5", generic_dsd_config }, { "CSC3551", "17AA38B6", generic_dsd_config }, { "CSC3551", "17AA38B7", generic_dsd_config }, + { "CSC3551", "17AA3877", generic_dsd_config }, + { "CSC3551", "17AA3878", generic_dsd_config }, {} }; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a17c36a36aa5..540868d6c602 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -10347,6 +10347,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x38b5, "Legion Slim 7 16IRH8", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38b6, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38b7, "Legion Slim 7 16APH8", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x3877, "Legion Slim 7 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x17aa, 0x3878, "Legion Slim 7 16ARHA7", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x17aa, 0x38ba, "Yoga S780-14.5 Air AMD quad YC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38bb, "Yoga S780-14.5 Air AMD quad AAC", ALC287_FIXUP_TAS2781_I2C), SND_PCI_QUIRK(0x17aa, 0x38be, "Yoga S980-14.5 proX YC Dual", ALC287_FIXUP_TAS2781_I2C), -- 2.44.0

3 4

[PATCH 0/9] enabled -Wformat-truncation for clang
by Arnd Bergmann 02 Apr '24

02 Apr '24

From: Arnd Bergmann <arnd(a)arndb.de> With randconfig build testing, I found only eight files that produce warnings with clang when -Wformat-truncation is enabled. This means we can just turn it on by default rather than only enabling it for "make W=1". Unfortunately, gcc produces a lot more warnings when the option is enabled, so it's not yet possible to turn it on both both compilers. I hope that the patches can get picked up by platform maintainers directly, so the final patch can go in later on. Arnd Arnd Bergmann (9): fbdev: shmobile: fix snprintf truncation enetc: avoid truncating error message qed: avoid truncating work queue length mlx5: avoid truncating error message surface3_power: avoid format string truncation warning Input: IMS: fix printf string overflow scsi: mylex: fix sysfs buffer lengths ALSA: aoa: avoid false-positive format truncation warning kbuild: enable -Wformat-truncation on clang drivers/input/misc/ims-pcu.c | 4 ++-- drivers/net/ethernet/freescale/enetc/enetc.c | 2 +- .../ethernet/mellanox/mlx5/core/esw/bridge.c | 2 +- drivers/net/ethernet/qlogic/qed/qed_main.c | 9 ++++--- drivers/platform/surface/surface3_power.c | 2 +- drivers/scsi/myrb.c | 20 ++++++++-------- drivers/scsi/myrs.c | 24 +++++++++---------- drivers/video/fbdev/sh_mobile_lcdcfb.c | 2 +- scripts/Makefile.extrawarn | 2 ++ sound/aoa/soundbus/i2sbus/core.c | 2 +- 10 files changed, 35 insertions(+), 34 deletions(-) -- 2.39.2 Cc: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> Cc: Claudiu Manoil <claudiu.manoil(a)nxp.com> Cc: Vladimir Oltean <vladimir.oltean(a)nxp.com> Cc: Jakub Kicinski <kuba(a)kernel.org> Cc: Saeed Mahameed <saeedm(a)nvidia.com> Cc: Leon Romanovsky <leon(a)kernel.org> Cc: Ariel Elior <aelior(a)marvell.com> Cc: Manish Chopra <manishc(a)marvell.com> Cc: Hans de Goede <hdegoede(a)redhat.com> Cc: "Ilpo Järvinen" <ilpo.jarvinen(a)linux.intel.com> Cc: Maximilian Luz <luzmaximilian(a)gmail.com> Cc: Hannes Reinecke <hare(a)kernel.org> Cc: "Martin K. Petersen" <martin.petersen(a)oracle.com> Cc: Helge Deller <deller(a)gmx.de> Cc: Masahiro Yamada <masahiroy(a)kernel.org> Cc: Nathan Chancellor <nathan(a)kernel.org> Cc: Nicolas Schier <nicolas(a)fjasle.eu> Cc: Johannes Berg <johannes(a)sipsolutions.net> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Nick Desaulniers <ndesaulniers(a)google.com> Cc: Bill Wendling <morbo(a)google.com> Cc: Justin Stitt <justinstitt(a)google.com> Cc: linux-input(a)vger.kernel.org Cc: linux-kernel(a)vger.kernel.org Cc: netdev(a)vger.kernel.org Cc: linux-rdma(a)vger.kernel.org Cc: platform-driver-x86(a)vger.kernel.org Cc: linux-scsi(a)vger.kernel.org Cc: linux-fbdev(a)vger.kernel.org Cc: dri-devel(a)lists.freedesktop.org Cc: linux-kbuild(a)vger.kernel.org Cc: linuxppc-dev(a)lists.ozlabs.org Cc: alsa-devel(a)alsa-project.org Cc: linux-sound(a)vger.kernel.org Cc: llvm(a)lists.linux.dev

5 5

[PATCH v6 0/2] ASoC: nau8325: Modify driver code and dtschema.
by Seven Lee 01 Apr '24

01 Apr '24

Revise properties description and use standard units in dtschema. The unit conversion driver based on the attribute must also be changed accordingly. Change: V1 -> V2: - Revise the driver description part for the C++ comment. - In the nau8325_clkdet_put function, modify the max variable to hard code. - Removed "Clock Detection" switch control. - modify the "ALC Enable" switch name. - Revise the dtschema for "nuvoton,dac-vref". V2 -> V3: - Properties use standard unit suffixes. - Modify the enum definition. - Driver code should be used dev_dbg(). V3 -> V4: - Properties use standard unit suffixes. - Modify the enum definition. V4 -> V5: - Properties use standard unit suffixes for vref-impedance and dac-vref. - Revise the enum definition. - Modify the code related to properties usage in the NAU8325 codec driver. - Re-arrange header files in alphabetical order. V5 -> V6: - Revise the nau8325_dac_oversampl_enum structure definition. Seven Lee (2): ASoC: dt-bindings: Added schema for "nuvoton,nau8325" ASoC: nau8325: new driver .../bindings/sound/nuvoton,nau8325.yaml | 80 ++ sound/soc/codecs/nau8325.c | 900 ++++++++++++++++++ sound/soc/codecs/nau8325.h | 391 ++++++++ 3 files changed, 1371 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nuvoton,nau8325.yaml create mode 100644 sound/soc/codecs/nau8325.c create mode 100644 sound/soc/codecs/nau8325.h -- 2.25.1

3 4

Segmentation fault on Python 3.12
by GitHub issues - opened 30 Mar '24

30 Mar '24

alsa-project/alsa-python issue #9 was opened from FrancescoCeruti: When using Python 3.12, I'm getting a segmentation fault, by simply importing the `alsasq` module (`from pyalsa import alsaseq`) Not sure if this it's the same as #8, here's the backtrace: ``` (gdb) bt #0 __memset_avx512_unaligned_erms () at ../sysdeps/x86_64/multiarch/memset-vec-unaligned-erms.S:216 #1 0x00007ffff79e6122 in _PyType_AllocNoTrack (nitems=1, type=0x7ffff7f571c0 <ConstantType>) at Objects/typeobject.c:1708 #2 PyType_GenericAlloc (type=0x7ffff7f571c0 <ConstantType>, nitems=1) at Objects/typeobject.c:1722 #3 0x00007ffff79b0a0e in long_subtype_new (obase=<optimized out>, x=<optimized out>, type=0x7ffff7f571c0 <ConstantType>) at Objects/longobject.c:5616 #4 long_new_impl (obase=<optimized out>, x=<optimized out>, type=0x7ffff7f571c0 <ConstantType>) at Objects/longobject.c:5555 #5 long_new (type=0x7ffff7f571c0 <ConstantType>, args=<optimized out>, kwargs=<optimized out>) at Objects/clinic/longobject.c.h:65 #6 0x00007ffff7f49963 in Constant_create (name=name@entry=0x7ffff7f4e47c "SEQ_OPEN_INPUT", value=value@entry=2) at pyalsa/alsaseq.c:387 #7 0x00007ffff7f4abce in PyInit_alsaseq () at pyalsa/alsaseq.c:3337 #8 0x00007ffff7ac1ed1 in _PyImport_LoadDynamicModuleWithSpec (spec=spec@entry=0x7ffff71fe840, fp=fp@entry=0x0) at ./Python/importdl.c:169 #9 0x00007ffff7abee8a in _imp_create_dynamic_impl (module=<optimized out>, file=0x0, spec=0x7ffff71fe840) at Python/import.c:3775 #10 _imp_create_dynamic (module=<optimized out>, args=<optimized out>, nargs=<optimized out>) at Python/clinic/import.c.h:506 #11 0x00007ffff79cac88 in cfunction_vectorcall_FASTCALL (func=0x7ffff71972e0, args=0x7ffff71fe6c8, nargsf=<optimized out>, kwnames=0x0) at Objects/methodobject.c:422 #12 0x00007ffff790c8bf in _PyEval_EvalFrameDefault (tstate=<optimized out>, frame=<optimized out>, throwflag=<optimized out>) at Python/bytecodes.c:3254 #13 0x00007ffff7972487 in _PyObject_VectorcallTstate (kwnames=0x0, nargsf=2, args=0x7fffffffbae0, callable=0x7ffff71a4040, tstate=0x7ffff7ec3988 <_PyRuntime+459656>) at ./Include/internal/pycore_call.h:92 #14 object_vacall (tstate=tstate@entry=0x7ffff7ec3988 <_PyRuntime+459656>, base=base@entry=0x0, callable=0x7ffff71a4040, vargs=vargs@entry=0x7fffffffbb70) at Objects/call.c:850 #15 0x00007ffff79726e7 in PyObject_CallMethodObjArgs (obj=0x0, name=0x7ffff7e5c0b8 <_PyRuntime+35512>) at Objects/call.c:911 #16 0x00007ffff7ac076b in import_find_and_load (abs_name=0x7ffff7028eb0, tstate=0x7ffff7ec3988 <_PyRuntime+459656>) at Python/import.c:2779 #17 PyImport_ImportModuleLevelObject (name=name@entry=0x7ffff7028eb0, globals=globals@entry=0x0, locals=locals@entry=0x0, fromlist=fromlist@entry=0x0, level=0) at Python/import.c:2862 #18 0x00007ffff7a82496 in builtin___import___impl (level=<optimized out>, fromlist=0x0, locals=0x0, globals=0x0, name=0x7ffff7028eb0, module=<optimized out>) at Python/bltinmodule.c:275 #19 builtin___import__ (module=<optimized out>, args=<optimized out>, nargs=<optimized out>, kwnames=<optimized out>) at Python/clinic/bltinmodule.c.h:107 #20 0x00007ffff79ca9fd in cfunction_vectorcall_FASTCALL_KEYWORDS (func=0x7ffff71959e0, args=0x7ffff719ad28, nargsf=<optimized out>, kwnames=0x0) at Objects/methodobject.c:438 #21 0x00007ffff790c8bf in _PyEval_EvalFrameDefault (tstate=<optimized out>, frame=<optimized out>, throwflag=<optimized out>) at Python/bytecodes.c:3254 #22 0x00007ffff7972487 in _PyObject_VectorcallTstate (kwnames=0x0, nargsf=3, args=0x7fffffffbf60, callable=0x7ffff71a4180, tstate=0x7ffff7ec3988 <_PyRuntime+459656>) at ./Include/internal/pycore_call.h:92 #23 object_vacall (tstate=tstate@entry=0x7ffff7ec3988 <_PyRuntime+459656>, base=base@entry=0x0, callable=0x7ffff71a4180, vargs=vargs@entry=0x7fffffffbff0) at Objects/call.c:850 #24 0x00007ffff79726e7 in PyObject_CallMethodObjArgs (obj=0x0, name=name@entry=0x7ffff7e5c190 <_PyRuntime+35728>) at Objects/call.c:911 #25 0x00007ffff7ac0326 in PyImport_ImportModuleLevelObject (name=name@entry=0x7ffff71fd7d0, globals=<optimized out>, locals=locals@entry=0x7ffff71f9880, fromlist=fromlist@entry=0x7ffff71fc2e0, level=0) at Python/import.c:2931 #26 0x00007ffff791562e in import_name (level=0x7ffff7e542c8 <_PyRuntime+3272>, fromlist=0x7ffff71fc2e0, name=0x7ffff71fd7d0, frame=<optimized out>, tstate=0x7ffff71959e0) at Python/ceval.c:2482 #27 _PyEval_EvalFrameDefault (tstate=tstate@entry=0x7ffff7ec3988 <_PyRuntime+459656>, frame=<optimized out>, frame@entry=0x7ffff7f9b020, throwflag=throwflag@entry=0) at Python/bytecodes.c:2135 #28 0x00007ffff7a89247 in _PyEval_EvalFrame (throwflag=0, frame=0x7ffff7f9b020, tstate=0x7ffff7ec3988 <_PyRuntime+459656>) at ./Include/internal/pycore_ceval.h:89 #29 _PyEval_Vector (args=0x0, argcount=0, kwnames=0x0, locals=0x7ffff71f9880, func=0x7ffff71da160, tstate=0x7ffff7ec3988 <_PyRuntime+459656>) at Python/ceval.c:1683 #30 PyEval_EvalCode (co=co@entry=0x7ffff70183f0, globals=globals@entry=0x7ffff71f9880, locals=locals@entry=0x7ffff71f9880) at Python/ceval.c:578 #31 0x00007ffff7ae3506 in run_eval_code_obj (tstate=tstate@entry=0x7ffff7ec3988 <_PyRuntime+459656>, co=co@entry=0x7ffff70183f0, globals=globals@entry=0x7ffff71f9880, locals=locals@entry=0x7ffff71f9880) at Python/pythonrun.c:1722 #32 0x00007ffff7ae3619 in run_mod (mod=<optimized out>, filename=filename@entry=0x7ffff712f450, globals=globals@entry=0x7ffff71f9880, locals=locals@entry=0x7ffff71f9880, flags=flags@entry=0x7fffffffc520, arena=arena@entry=0x7ffff711be30) at Python/pythonrun.c:1743 #33 0x00007ffff7ae650c in pyrun_file (flags=0x7fffffffc520, closeit=1, locals=0x7ffff71f9880, globals=0x7ffff71f9880, start=257, filename=0x7ffff712f450, fp=0x55555555a590) at Python/pythonrun.c:1643 #34 _PyRun_SimpleFileObject (fp=fp@entry=0x55555555a590, filename=filename@entry=0x7ffff712f450, closeit=closeit@entry=1, flags=flags@entry=0x7fffffffc520) at Python/pythonrun.c:433 #35 0x00007ffff7ae6afc in _PyRun_AnyFileObject (fp=0x55555555a590, filename=filename@entry=0x7ffff712f450, closeit=closeit@entry=1, flags=flags@entry=0x7fffffffc520) at Python/pythonrun.c:78 #36 0x00007ffff7b0d72c in pymain_run_file_obj (skip_source_first_line=0, filename=0x7ffff712f450, program_name=0x7ffff7148880) at Modules/main.c:360 #37 pymain_run_file (config=0x7ffff7e66568 <_PyRuntime+77672>) at Modules/main.c:379 #38 pymain_run_python (exitcode=exitcode@entry=0x7fffffffc634) at Modules/main.c:629 #39 0x00007ffff7b0db66 in Py_RunMain () at Modules/main.c:709 #40 0x00007ffff7b0dd1a in pymain_main (args=0x7fffffffc660) at Modules/main.c:739 #41 Py_BytesMain (argc=<optimized out>, argv=<optimized out>) at Modules/main.c:763 #42 0x00007ffff7557cd0 in __libc_start_call_main (main=main@entry=0x555555555040 <main>, argc=argc@entry=2, argv=argv@entry=0x7fffffffc7e8) at ../sysdeps/nptl/libc_start_call_main.h:58 #43 0x00007ffff7557d8a in __libc_start_main_impl (main=0x555555555040 <main>, argc=2, argv=0x7fffffffc7e8, init=<optimized out>, fini=<optimized out>, rtld_fini=<optimized out>, stack_end=0x7fffffffc7d8) at ../csu/libc-start.c:360 #44 0x0000555555555075 in _start () ``` Issue URL : https://github.com/alsa-project/alsa-python/issues/9 Repository URL: https://github.com/alsa-project/alsa-python

1 0

[PATCH] ALSA: hda/realtek: Add quirks for ASUS Laptops using CS35L56
by Simon Trimmer 30 Mar '24

30 Mar '24

These ASUS laptops use the Realtek HDA codec combined with a number of CS35L56 amplifiers. The SSID of the GA403U matches a previous ASUS laptop - we can tell them apart because they use different codecs. Signed-off-by: Simon Trimmer <simont(a)opensource.cirrus.com> --- sound/pci/hda/patch_realtek.c | 53 ++++++++++++++++++++++++++++++++++- 1 file changed, 52 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a17c36a36aa5..7d229f4a8aa6 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6875,11 +6875,38 @@ static void alc287_fixup_legion_16ithg6_speakers(struct hda_codec *cdc, const st comp_generic_fixup(cdc, action, "i2c", "CLSA0101", "-%s:00-cs35l41-hda.%d", 2); } +static void cs35l56_fixup_i2c_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "i2c", "CSC3556", "-%s:00-cs35l56-hda.%d", 2); +} + +static void cs35l56_fixup_i2c_four(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "i2c", "CSC3556", "-%s:00-cs35l56-hda.%d", 4); +} + +static void cs35l56_fixup_spi_two(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + comp_generic_fixup(cdc, action, "spi", "CSC3556", "-%s:00-cs35l56-hda.%d", 2); +} + static void cs35l56_fixup_spi_four(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { comp_generic_fixup(cdc, action, "spi", "CSC3556", "-%s:00-cs35l56-hda.%d", 4); } +static void alc285_fixup_asus_ga403u(struct hda_codec *cdc, const struct hda_fixup *fix, int action) +{ + /* + * The same SSID has been re-used in different hardware, they have + * different codecs and the newer GA403U has a ALC285. + */ + if (cdc->core.vendor_id == 0x10ec0285) + cs35l56_fixup_i2c_two(cdc, fix, action); + else + alc_fixup_inv_dmic(cdc, fix, action); +} + static void tas2781_fixup_i2c(struct hda_codec *cdc, const struct hda_fixup *fix, int action) { @@ -7436,6 +7463,10 @@ enum { ALC256_FIXUP_ACER_SFG16_MICMUTE_LED, ALC256_FIXUP_HEADPHONE_AMP_VOL, ALC245_FIXUP_HP_SPECTRE_X360_EU0XXX, + ALC285_FIXUP_CS35L56_SPI_2, + ALC285_FIXUP_CS35L56_I2C_2, + ALC285_FIXUP_CS35L56_I2C_4, + ALC285_FIXUP_ASUS_GA403U, }; /* A special fixup for Lenovo C940 and Yoga Duet 7; @@ -9643,6 +9674,22 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc245_fixup_hp_spectre_x360_eu0xxx, }, + [ALC285_FIXUP_CS35L56_SPI_2] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l56_fixup_spi_two, + }, + [ALC285_FIXUP_CS35L56_I2C_2] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l56_fixup_i2c_two, + }, + [ALC285_FIXUP_CS35L56_I2C_4] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l56_fixup_i2c_four, + }, + [ALC285_FIXUP_ASUS_GA403U] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc285_fixup_asus_ga403u, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -10096,7 +10143,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1a83, "ASUS UM5302LA", ALC294_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1a8f, "ASUS UX582ZS", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1b11, "ASUS UX431DA", ALC294_FIXUP_ASUS_COEF_1B), - SND_PCI_QUIRK(0x1043, 0x1b13, "Asus U41SV", ALC269_FIXUP_INV_DMIC), + SND_PCI_QUIRK(0x1043, 0x1b13, "ASUS U41SV/GA403U", ALC285_FIXUP_ASUS_GA403U), SND_PCI_QUIRK(0x1043, 0x1b93, "ASUS G614JVR/JIR", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1bbd, "ASUS Z550MA", ALC255_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1043, 0x1c03, "ASUS UM3406HA", ALC287_FIXUP_CS35L41_I2C_2), @@ -10104,6 +10151,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1c33, "ASUS UX5304MA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1c43, "ASUS UX8406MA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1c62, "ASUS GU603", ALC289_FIXUP_ASUS_GA401), + SND_PCI_QUIRK(0x1043, 0x1c63, "ASUS GU605M", ALC285_FIXUP_CS35L56_SPI_2), SND_PCI_QUIRK(0x1043, 0x1c92, "ASUS ROG Strix G15", ALC285_FIXUP_ASUS_G533Z_PINS), SND_PCI_QUIRK(0x1043, 0x1c9f, "ASUS G614JU/JV/JI", ALC285_FIXUP_ASUS_HEADSET_MIC), SND_PCI_QUIRK(0x1043, 0x1caf, "ASUS G634JY/JZ/JI/JG", ALC285_FIXUP_ASUS_SPI_REAR_SPEAKERS), @@ -10115,11 +10163,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1d42, "ASUS Zephyrus G14 2022", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1d4e, "ASUS TM420", ALC256_FIXUP_ASUS_HPE), SND_PCI_QUIRK(0x1043, 0x1da2, "ASUS UP6502ZA/ZD", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x1043, 0x1df3, "ASUS UM5606", ALC285_FIXUP_CS35L56_I2C_4), SND_PCI_QUIRK(0x1043, 0x1e02, "ASUS UX3402ZA", ALC245_FIXUP_CS35L41_SPI_2), SND_PCI_QUIRK(0x1043, 0x1e11, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA502), SND_PCI_QUIRK(0x1043, 0x1e12, "ASUS UM3402", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e51, "ASUS Zephyrus M15", ALC294_FIXUP_ASUS_GU502_PINS), SND_PCI_QUIRK(0x1043, 0x1e5e, "ASUS ROG Strix G513", ALC294_FIXUP_ASUS_G513_PINS), + SND_PCI_QUIRK(0x1043, 0x1e63, "ASUS H7606W", ALC285_FIXUP_CS35L56_I2C_2), + SND_PCI_QUIRK(0x1043, 0x1e83, "ASUS GA605W", ALC285_FIXUP_CS35L56_I2C_2), SND_PCI_QUIRK(0x1043, 0x1e8e, "ASUS Zephyrus G15", ALC289_FIXUP_ASUS_GA401), SND_PCI_QUIRK(0x1043, 0x1ee2, "ASUS UM6702RA/RC", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x1043, 0x1c52, "ASUS Zephyrus G15 2022", ALC289_FIXUP_ASUS_GA401), -- 2.34.1

2 1

[PATCH v18 00/41] Introduce QC USB SND audio offloading support
by Wesley Cheng 30 Mar '24

30 Mar '24

Several Qualcomm based chipsets can support USB audio offloading to a dedicated audio DSP, which can take over issuing transfers to the USB host controller. The intention is to reduce the load on the main processors in the SoC, and allow them to be placed into lower power modes. There are several parts to this design: 1. Adding ASoC binding layer 2. Create a USB backend for Q6DSP 3. Introduce XHCI interrupter support 4. Create vendor ops for the USB SND driver USB | ASoC -------------------------------------------------------------------- | _________________________ | |sm8250 platform card | | |_________________________| | | | | ___V____ ____V____ | |Q6USB | |Q6AFE | | |"codec" | |"cpu" | | |________| |_________| | ^ ^ ^ | | |________| | ___V____ | | |SOC-USB | | ________ ________ | | | |USB SND |<--->|QC offld|<------------>|________| | |(card.c)| | |<---------- | |________| |________|___ | | | ^ ^ | | | ____________V_________ | | | | | |APR/GLINK | __ V_______________V_____ | | | |______________________| |USB SND (endpoint.c) | | | | ^ |_________________________| | | | | ^ | | | ___________V___________ | | | |->|audio DSP | ___________V_____________ | | |_______________________| |XHCI HCD |<- | |_________________________| | Adding ASoC binding layer: soc-usb: Intention is to treat a USB port similar to a headphone jack. The port is always present on the device, but cable/pin status can be enabled/disabled. Expose mechanisms for USB backend ASoC drivers to communicate with USB SND. Create a USB backend for Q6DSP: q6usb: Basic backend driver that will be responsible for maintaining the resources needed to initiate a playback stream using the Q6DSP. Will be the entity that checks to make sure the connected USB audio device supports the requested PCM format. If it does not, the PCM open call will fail, and userpsace ALSA can take action accordingly. Introduce XHCI interrupter support: XHCI HCD supports multiple interrupters, which allows for events to be routed to different event rings. This is determined by "Interrupter Target" field specified in Section "6.4.1.1 Normal TRB" of the XHCI specification. Events in the offloading case will be routed to an event ring that is assigned to the audio DSP. Create vendor ops for the USB SND driver: qc_audio_offload: This particular driver has several components associated with it: - QMI stream request handler - XHCI interrupter and resource management - audio DSP memory management When the audio DSP wants to enable a playback stream, the request is first received by the ASoC platform sound card. Depending on the selected route, ASoC will bring up the individual DAIs in the path. The Q6USB backend DAI will send an AFE port start command (with enabling the USB playback path), and the audio DSP will handle the request accordingly. Part of the AFE USB port start handling will have an exchange of control messages using the QMI protocol. The qc_audio_offload driver will populate the buffer information: - Event ring base address - EP transfer ring base address and pass it along to the audio DSP. All endpoint management will now be handed over to the DSP, and the main processor is not involved in transfers. Overall, implementing this feature will still expose separate sound card and PCM devices for both the platorm card and USB audio device: 0 [SM8250MTPWCD938]: sm8250 - SM8250-MTP-WCD9380-WSA8810-VA-D SM8250-MTP-WCD9380-WSA8810-VA-DMIC 1 [Audio ]: USB-Audio - USB Audio Generic USB Audio at usb-xhci-hcd.1.auto-1.4, high speed This is to ensure that userspace ALSA entities can decide which route to take when executing the audio playback. In the above, if card#1 is selected, then USB audio data will take the legacy path over the USB PCM drivers, etc... This feature was validated using: - tinymix: set/enable the multimedia path to route to USB backend - tinyplay: issue playback on platform card Changelog -------------------------------------------- Changes in v18: - Rebased to usb-next, which merged in part of the series. Removed these patches. - Reworked Kconfigs for the ASoC USB related components from QCOM Q6DSP drivers to keep dependencies in place for SoC USB and USB SND. - Removed the repurposing of the stop ep sync API into existing XHCI operations. This will be solely used by the XHCI sideband for now. Changes in v17: - Fixed an issue where one patch was squashed into another. - Re-added some kconfig checks for helpers exposed in USB SND for the soc usb driver, after running different kconfigs. Changes in v16: - Modified some code layer dependencies so that soc usb can be split as a separate module. - Split the kcontrols from ASoC QCOM common layer into a separate driver - Reworked SOC USB kcontrols for controlling card + pcm offload routing and status so that there are individual controls for card and pcm devices. - Added a kcontrol remove API in SOC USB to remove the controls on the fly. This required to add some kcontrol management to SOC USB. - Removed the disconnect work and workqueue for the QC USB offload as it is not required, since QMI interface driver ensures events are handled in its own WQ. Changes in v15: - Removed some already merged XHCI changes - Separated SOC USB driver from being always compiled into SOC core. Now configurable from kconfig. - Fixed up ASoC kcontrol naming to fit guidelines. - Removed some unnecessary dummy ifdefs. - Moved usb snd offload capable kcontrol to be initialized by the platform offloading driver. Changes in v14: - Cleaned up some USB SND related feedback: - Renamed SNDUSB OFFLD playback available --> USB offload capable card - Fixed locking while checking if stream is in use - Replaced some mutex pairs with guard(mutex) Changes in v13: - Pulled in secondary/primary interrupter rework from Mathias from: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/drivers… - Did some cleanup and commit message updates, and tested on current code base. - Added mutex locking to xhci sideband to help prevent any race conditions, esp. for when accessing shared references. - Addresed concerns from Hillf about gfp_flags and locking used in qc_usb_audio_offload. - Rebased onto usb-next Changes in v12: - Updated copyright year to 2024. Happy new years! - Fixed newline format on mixer offload driver. Changes in v11: - Modified QMI format structures to be const Changes in v10: - Added new mixer for exposing kcontrol for sound card created by USB SND. This allows for applications to know which platform sound card has offload support. Will return the card number. - Broke down and cleaned up some functions/APIs within qc_audio_offload driver. - Exported xhci_initialize_ring_info(), and modified XHCI makefile to allow for the XHCI sideband to exist as a module. - Reworked the jack registration and moved it to the QCOM platform card driver, ie sm8250. - Added an SOC USB API to fetch a standard component tag that can be appended to the platform sound card. Added this tag to sm8250 if any USB path exists within the DT node. - Moved kcontrols that existed in the Q6USB driver, and made it a bit more generic, so that naming can be standardized across solutions. SOC USB is now responsible for creation of these kcontrols. - Added a SOC USB RST document explaining some code flows and implementation details so that other vendors can utilize the framework. - Addressed a case where USB device connection events are lost if usb offload driver (qc_audio_offload) is not probed when everything else has been initialized, ie USB SND, SOC USB and ASoC sound card. Add a rediscover device call during module init, to ensure that connection events will be propagated. - Rebased to usb-next. Changes in v9: - Fixed the dt binding check issue with regards to num-hc-interrupters. Changes in v8: - Cleaned up snd_soc_usb_find_priv_data() based on Mark's feedback. Removed some of the duplicate looping code that was present on previous patches. Also renamed the API. - Integrated Mathias' suggestions on his new sideband changes: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Addressed some of Mathias' fixme tags, such as: - Resetting transfer ring dequeue/enqueue pointers - Issuing stop endpoint command during ep removal - Reset ERDP properly to first segment ring during interrupter removal. (this is currently just being cleared to 0, but should be pointing to a valid segment if controller is still running. Changes in v7: - Fixed dt check error for q6usb bindings - Updated q6usb property from qcom,usb-audio-intr-num --> qcom,usb-audio-intr-idx - Removed separate DWC3 HC interrupters num property, and place limits to XHCI one. - Modified xhci_ring_to_sgtable() to use assigned IOVA/DMA address to fetch pages, as it is not ensured event ring allocated is always done in the vmalloc range. Changes in v6: - Fixed limits and description on several DT bindings (XHCI and Q6USB) - Fixed patch subjects to follow other ALSA/ASoC notations. USB SND - Addressed devices which expose multiple audio (UAC) interfaces. These devices will create a single USB sound card with multiple audio streams, and receive multiple interface probe routines. QC offload was not properly considering cases with multiple probe calls. - Renamed offload module name and kconfig to fit within the SND domain. - Renamed attach/detach endpoint API to keep the hw_params notation. Changes in v5: - Removed some unnescessary files that were included - Fixed some typos mentioned - Addressed dt-binding issues and added hc-interrupters definition to usb-xhci.yaml XHCI: - Moved secondary skip events API to xhci-ring and updated implementation - Utilized existing XHCI APIs, such as inc_deq and xhci_update_erst_dequeue() USB SND - Renamed and reworked the APIs in "sound: usb: Export USB SND APIs for modules" patch to include suggestions to utilize snd_usb_hw_params/free and to avoid generic naming. - Added a resume_cb() op for completion sake. - Addressed some locking concerns with regards to when registering for platform hooks. - Added routine to disconnect all offloaded devices during module unbind. ASoC - Replaced individual PCM parameter arguments in snd_soc_usb_connect() with new snd_soc_usb_device structure to pass along PCM info. - Modified snd_jack set report to notify HEADPHONE event, as we do not support record path. Changes in v4: - Rebased to xhci/for-usb-next - Addressed some dt-bindings comments XHCI: - Pulled in latest changes from Mathias' feature_interrupters branch: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… - Fixed commit text and signage for the XHCI sideband/interrupter related changes - Added some logic to address the FIXME tags mentioned throughout the commits, such as handling multi segment rings and building the SGT, locking concerns, and ep cleanup operations. - Removed some fixme tags for conditions that may not be needed/addressed. - Repurposed the new endpoint stop sync API to be utilized in other places. - Fixed potential compile issue if XHCI sideband config is not defined. ASoC: - Added sound jack control into the Q6USB driver. Allows for userpsace to know when an offload capable device is connected. USB SND: - Avoided exporting _snd_pcm_hw_param_set based on Takashi's recommendation. - Split USB QMI packet header definitions into a separate commit. This is used to properly allow the QMI interface driver to parse and route QMI packets accordingly - Added a "depends on" entry when enabling QC audio offload to avoid compile time issues. Changes in v3: - Changed prefix from RFC to PATCH - Rebased entire series to usb-next - Updated copyright years XHCI: - Rebased changes on top of XHCI changes merged into usb-next, and only added changes that were still under discussion. - Added change to read in the "num-hc-interrupters" device property. ASoC: - qusb6 USB backend - Incorporated suggestions to fetch iommu information with existing APIs - Added two new sound kcontrols to fetch offload status and offload device selection. - offload status - will return the card and pcm device in use tinymix -D 0 get 1 --> 1, 0 (offload in progress on card#1 pcm#0) - device selection - set the card and pcm device to enable offload on. Ex.: tinymix -D 0 set 1 2 0 --> sets offload on card#2 pcm#0 (this should be the USB card) USB SND: - Fixed up some locking related concerns for registering platform ops. - Moved callbacks under the register_mutex, so that - Modified APIs to properly pass more information about the USB SND device, so that the Q6USB backend can build a device list/map, in order to monitor offload status and device selection. Changes in v2: XHCI: - Replaced XHCI and HCD changes with Mathias' XHCI interrupter changes in his tree: https://git.kernel.org/pub/scm/linux/kernel/git/mnyman/xhci.git/log/?h=feat… Adjustments made to Mathias' changes: - Created xhci-intr.h to export/expose interrupter APIs versus exposing xhci.h. Moved dependent structures to this file as well. (so clients can parse out information from "struct xhci_interrupter") - Added some basic locking when requesting interrupters. - Fixed up some sanity checks. - Removed clearing of the ERSTBA during freeing of the interrupter. (pending issue where SMMU fault occurs if DMA addr returned is 64b - TODO) - Clean up pending events in the XHCI secondary interrupter. While testing USB bus suspend, it was seen that on bus resume, the xHCI HC would run into a command timeout. - Added offloading APIs to xHCI to fetch transfer and event ring information. ASoC: - Modified soc-usb to allow for multiple USB port additions. For this to work, the USB offload driver has to have a reference to the USB backend by adding a "usb-soc-be" DT entry to the device saved into XHCI sysdev. - Created separate dt-bindings for defining USB_RX port. - Increased APR timeout to accommodate the situation where the AFE port start command could be delayed due to having to issue a USB bus resume while handling the QMI stream start command. USB SND: - Added a platform ops during usb_audio_suspend(). This allows for the USB offload driver to halt the audio stream when system enters PM suspend. This ensures the audio DSP is not issuing transfers on the USB bus. - Do not override platform ops if they are already populated. - Introduce a shared status variable between the USB offload and USB SND layers, to ensure that only one path is active at a time. If the USB bus is occupied, then userspace is notified that the path is busy. Mathias Nyman (2): xhci: add helper to stop endpoint and wait for completion xhci: sideband: add initial api to register a sideband entity Wesley Cheng (39): usb: host: xhci: Export enable and disable interrupter APIs usb: host: xhci: Repurpose event handler for skipping interrupter events xhci: export XHCI IMOD setting helper for interrupters usb: host: xhci-sideband: Expose a sideband interrupter enable API usb: host: xhci-mem: Cleanup pending secondary event ring events usb: host: xhci-mem: Allow for interrupter clients to choose specific index ASoC: Add SOC USB APIs for adding an USB backend ASoC: dt-bindings: qcom,q6dsp-lpass-ports: Add USB_RX port ASoC: qcom: qdsp6: Introduce USB AFE port to q6dsp ASoC: qdsp6: q6afe: Increase APR timeout ASoC: qcom: qdsp6: Add USB backend ASoC driver for Q6 ALSA: usb-audio: Introduce USB SND platform op callbacks ALSA: usb-audio: Export USB SND APIs for modules ALSA: usb-audio: Save UAC sample size information usb: dwc3: Specify maximum number of XHCI interrupters usb: host: xhci-plat: Set XHCI max interrupters if property is present ALSA: usb-audio: qcom: Add USB QMI definitions ALSA: usb-audio: qcom: Introduce QC USB SND offloading support ALSA: usb-audio: Check for support for requested audio format ASoC: usb: Add PCM format check API for USB backend ASoC: qcom: qdsp6: Ensure PCM format is supported by USB audio device ALSA: usb-audio: Prevent starting of audio stream if in use ALSA: usb-audio: Do not allow USB offload path if PCM device is in use ASoC: dt-bindings: Update example for enabling USB offload on SM8250 ALSA: usb-audio: qcom: Populate PCM and USB chip information ASoC: qcom: qdsp6: Add support to track available USB PCM devices ASoC: Introduce SND kcontrols to select sound card and PCM device ASoC: qcom: qdsp6: Add SOC USB offload select get/put callbacks ASoC: Introduce SND kcontrols to track USB offloading state ASoC: qcom: qdsp6: Add PCM ops to track current state ASoC: usb: Create SOC USB SND jack kcontrol ASoC: qcom: qdsp6: Add headphone jack for offload connection status ASoC: usb: Fetch ASoC sound card information ALSA: usb-audio: Add USB offloading capable kcontrol ALSA: usb-audio: Allow for rediscovery of connected USB SND devices ALSA: usb-audio: qcom: Use card and PCM index from QMI request ASoC: usb: Rediscover USB SND devices on USB port add ASoC: qcom: Populate SoC components string ASoC: doc: Add documentation for SOC USB .../bindings/sound/qcom,sm8250.yaml | 15 + Documentation/sound/soc/index.rst | 1 + Documentation/sound/soc/usb.rst | 611 ++++++ drivers/usb/dwc3/core.c | 12 + drivers/usb/dwc3/core.h | 2 + drivers/usb/dwc3/host.c | 3 + drivers/usb/host/Kconfig | 9 + drivers/usb/host/Makefile | 2 + drivers/usb/host/xhci-mem.c | 37 +- drivers/usb/host/xhci-plat.c | 2 + drivers/usb/host/xhci-ring.c | 50 +- drivers/usb/host/xhci-sideband.c | 439 ++++ drivers/usb/host/xhci.c | 48 +- drivers/usb/host/xhci.h | 18 +- .../sound/qcom,q6dsp-lpass-ports.h | 1 + include/linux/usb/xhci-sideband.h | 70 + include/sound/q6usboffload.h | 20 + include/sound/soc-usb.h | 182 ++ sound/soc/Kconfig | 10 + sound/soc/Makefile | 2 + sound/soc/qcom/Kconfig | 15 + sound/soc/qcom/Makefile | 2 + sound/soc/qcom/qdsp6/Makefile | 1 + sound/soc/qcom/qdsp6/q6afe-dai.c | 60 + sound/soc/qcom/qdsp6/q6afe.c | 193 +- sound/soc/qcom/qdsp6/q6afe.h | 36 +- sound/soc/qcom/qdsp6/q6dsp-lpass-ports.c | 23 + sound/soc/qcom/qdsp6/q6dsp-lpass-ports.h | 1 + sound/soc/qcom/qdsp6/q6routing.c | 9 + sound/soc/qcom/qdsp6/q6usb.c | 408 ++++ sound/soc/qcom/sm8250.c | 15 +- sound/soc/qcom/usb_offload_utils.c | 51 + sound/soc/qcom/usb_offload_utils.h | 26 + sound/soc/soc-usb.c | 660 ++++++ sound/usb/Kconfig | 25 + sound/usb/Makefile | 2 +- sound/usb/card.c | 109 + sound/usb/card.h | 15 + sound/usb/endpoint.c | 1 + sound/usb/format.c | 1 + sound/usb/helper.c | 1 + sound/usb/pcm.c | 104 +- sound/usb/pcm.h | 11 + sound/usb/qcom/Makefile | 6 + sound/usb/qcom/mixer_usb_offload.c | 65 + sound/usb/qcom/mixer_usb_offload.h | 17 + sound/usb/qcom/qc_audio_offload.c | 1914 +++++++++++++++++ sound/usb/qcom/usb_audio_qmi_v01.c | 892 ++++++++ sound/usb/qcom/usb_audio_qmi_v01.h | 162 ++ 49 files changed, 6306 insertions(+), 53 deletions(-) create mode 100644 Documentation/sound/soc/usb.rst create mode 100644 drivers/usb/host/xhci-sideband.c create mode 100644 include/linux/usb/xhci-sideband.h create mode 100644 include/sound/q6usboffload.h create mode 100644 include/sound/soc-usb.h create mode 100644 sound/soc/qcom/qdsp6/q6usb.c create mode 100644 sound/soc/qcom/usb_offload_utils.c create mode 100644 sound/soc/qcom/usb_offload_utils.h create mode 100644 sound/soc/soc-usb.c create mode 100644 sound/usb/qcom/Makefile create mode 100644 sound/usb/qcom/mixer_usb_offload.c create mode 100644 sound/usb/qcom/mixer_usb_offload.h create mode 100644 sound/usb/qcom/qc_audio_offload.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.c create mode 100644 sound/usb/qcom/usb_audio_qmi_v01.h

2 47

Sending 0xFx events causes libasound to segfault
by GitHub issues - edited 29 Mar '24

29 Mar '24

alsa-project/alsa-lib issue #392 was edited from human222: ALSA 1.1.9. This issue breaks DOSBox-X MIDI playback. Discovered through this thread: https://github.com/joncampbell123/dosbox-x/issues/4873 In particular, this comment: https://github.com/joncampbell123/dosbox-x/issues/4873#issuecomment-2026694… Issue URL : https://github.com/alsa-project/alsa-lib/issues/392 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

Sending 0xFx events causes libasound to segfault
by GitHub issues - opened 29 Mar '24

29 Mar '24

alsa-project/alsa-lib issue #392 was opened from human222: ALSA 1.1.9. This issue breaks DOSBox-X MIDI playback. Discovered through this thread: https://github.com/joncampbell123/dosbox-x/issues/4873 Issue URL : https://github.com/alsa-project/alsa-lib/issues/392 Repository URL: https://github.com/alsa-project/alsa-lib

1 0

[PATCH V2] ASoC: amd: acp: fix for acp_init function error handling
by Vijendar Mukunda 29 Mar '24

29 Mar '24

If acp_init() fails, acp pci driver probe should return error. Add acp_init() function return value check logic. Fixes: e61b415515d3 ("ASoC: amd: acp: refactor the acp init and de-init sequence") Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- changes from v1->v2: - handle clean up sequence instead of returning error directly sound/soc/amd/acp/acp-pci.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/soc/amd/acp/acp-pci.c b/sound/soc/amd/acp/acp-pci.c index 440b91d4f261..5f35b90eab8d 100644 --- a/sound/soc/amd/acp/acp-pci.c +++ b/sound/soc/amd/acp/acp-pci.c @@ -115,7 +115,10 @@ static int acp_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id goto unregister_dmic_dev; } - acp_init(chip); + ret = acp_init(chip); + if (ret) + goto unregister_dmic_dev; + res = devm_kcalloc(&pci->dev, num_res, sizeof(struct resource), GFP_KERNEL); if (!res) { ret = -ENOMEM; -- 2.34.1

2 1

alsa-info uploads re periodic sound failures on P16 w/ Mint
by Robert Hoskin 29 Mar '24

29 Mar '24

Greetings! The alsa-info output links below is from a Lenovo P16 Gen2, running Linux Mint 21.3, with an intermittent sound problem. The problem is that: Sound will be available from either built in speakers or headphones, perhaps for several days after a startup, but at some point will fail, with both Headphones (via jack) and internal Speakers disappearing from the "sound" app, and "Dummy Output" appearing in their place. If it's possible to reproduce this failure, I haven't figured out how. It happens when it happens, usually after a few days of uptime. I've found that sound can be restored with "sudo alsa force-reload", so I do have a workaround. *** For before-and-after comparison purposes... *** This report was taken when "Dummy Output" was shown in the Sound app: http://alsa-project.org/db/?f=4ec46710d176deb77fcdd950520eaafc4c9db1c3 *** This report was taken after a "sudo alsa force-reload", which restored both headphones and speakers to the Sound app" http://alsa-project.org/db/?f=3832b192eaa22ea4d51b0e6073f389318b983356 *** In the "System Reports" app, there are no crash reports shown. BUT... In the system logs app, I did find some lines in the Hardware section that may be relevant: 09:41:50 kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ASoC: error at snd_soc_pcm_component_prepare on 0000:00:1f.3: -22 09:41:50 kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ASoC: error at snd_soc_pcm_component_prepare on 0000:00:1f.3: -22 09:41:50 kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: error: set pcm hw_params after resume 09:41:50 kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: HW params ipc failed for stream 1 09:41:50 kernel: sof-audio-pci-intel-tgl 0000:00:1f.3: ipc tx error for 0x60010000 (msg/reply size: 108/20): -22 I really have no idea whether I'm submitting this in the right place or not, whether this is a kernel problem, a driver problem, a config problem, or whatever. I can tell you that this Mint install is new on a new machine, otherwise well-behaved, with subsequent updates that have not changed this sound behavior. Hope this is useful to you, love to have it fixed. Thanks! - Bob -- Robert Hoskin rhoskin(a)fastmail.org

1 1

[PATCH 00/22] virtio: store owner from modules with register_virtio_driver()
by Krzysztof Kozlowski 29 Mar '24

29 Mar '24

Merging ======= All further patches depend on the first virtio patch, therefore please ack and this should go via one tree: virtio? Description =========== Modules registering driver with register_virtio_driver() often forget to set .owner field. Solve the problem by moving this task away from the drivers to the core amba bus code, just like we did for platform_driver in commit 9447057eaff8 ("platform_device: use a macro instead of platform_driver_register"). Best regards, Krzysztof Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- Krzysztof Kozlowski (22): virtio: store owner from modules with register_virtio_driver() um: virt-pci: drop owner assignment virtio_blk: drop owner assignment bluetooth: virtio: drop owner assignment hwrng: virtio: drop owner assignment virtio_console: drop owner assignment crypto: virtio - drop owner assignment firmware: arm_scmi: virtio: drop owner assignment gpio: virtio: drop owner assignment drm/virtio: drop owner assignment iommu: virtio: drop owner assignment misc: nsm: drop owner assignment net: caif: virtio: drop owner assignment net: virtio: drop owner assignment net: 9p: virtio: drop owner assignment net: vmw_vsock: virtio: drop owner assignment wireless: mac80211_hwsim: drop owner assignment nvdimm: virtio_pmem: drop owner assignment rpmsg: virtio: drop owner assignment scsi: virtio: drop owner assignment fuse: virtio: drop owner assignment sound: virtio: drop owner assignment Documentation/driver-api/virtio/writing_virtio_drivers.rst | 1 - arch/um/drivers/virt-pci.c | 1 - drivers/block/virtio_blk.c | 1 - drivers/bluetooth/virtio_bt.c | 1 - drivers/char/hw_random/virtio-rng.c | 1 - drivers/char/virtio_console.c | 2 -- drivers/crypto/virtio/virtio_crypto_core.c | 1 - drivers/firmware/arm_scmi/virtio.c | 1 - drivers/gpio/gpio-virtio.c | 1 - drivers/gpu/drm/virtio/virtgpu_drv.c | 1 - drivers/iommu/virtio-iommu.c | 1 - drivers/misc/nsm.c | 1 - drivers/net/caif/caif_virtio.c | 1 - drivers/net/virtio_net.c | 1 - drivers/net/wireless/virtual/mac80211_hwsim.c | 1 - drivers/nvdimm/virtio_pmem.c | 1 - drivers/rpmsg/virtio_rpmsg_bus.c | 1 - drivers/scsi/virtio_scsi.c | 1 - drivers/virtio/virtio.c | 6 ++++-- fs/fuse/virtio_fs.c | 1 - include/linux/virtio.h | 7 +++++-- net/9p/trans_virtio.c | 1 - net/vmw_vsock/virtio_transport.c | 1 - sound/virtio/virtio_card.c | 1 - 24 files changed, 9 insertions(+), 27 deletions(-) --- base-commit: 7fdcff3312e16ba8d1419f8a18f465c5cc235ecf change-id: 20240327-module-owner-virtio-546763b3ca22 Best regards, -- Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org>

10 38

[PATCH 00/13] ASoC: ti: davinci-i2s: Add features to McBSP driver
by Bastien Curutchet 29 Mar '24

29 Mar '24

This series aims to add some features to McBSP driver. Convert bindings from .txt to .yaml. Add possibility to use an external clock as sample rate generator's input. Add handling of new formats (TDM, S24_LE, BP_FC). Add optional properties in DT: - ti,enable-sync-err : Enable the detection of unexpected frame pulses - ti,disable-free-run : Disable the free-running mode where McBSP drives serial clocks during emulation halt - ti,drive-dx : Outputs a chosen pattern on DX pin during capture streams. This has been tested on a platform designed off of the DAVINCI/OMAP-L138 connected to 3 daisy-chained AD7767. An external clock drives the sample rate generator through the CLKS pin. The hardware I have only allowed me to test acquisition side of McBSP. It is connected to a 6 channels TDM and acts as Bit clock provider and Frame clock consumer. Bastien Curutchet (13): ASoC: dt-bindings: davinci-mcbsp: convert McBSP bindings to yaml schema ASoC: dt-bindings: davinci-mcbsp: Add new properties ASoC: ti: davinci-i2s: Remove the unused clk_input_pin attribute ASoC: ti: davinci-i2s: Replace dev_err with dev_err_probe ASoC: ti: davinci-i2s: Use external clock to drive sample rate generator ASoC: ti: davinci-i2s: Delete unnecessary assignment ASoC: ti: davinci-i2s: Add TDM support ASoC: ti: davinci-i2s: Add handling of BP_FC format ASoC: ti: davinci-i2s: Enable unexpected frame pulses detection ASoC: ti: davinci-i2s: Make free-running mode optional ASoC: ti: davinci-i2s: Add S24_LE to supported formats ASoC: dt-bindings: davinic-mcbsp: Add the 'ti,drive-dx' property ASoC: ti: davinci-i2s: Opitonally drive DX pin during capture streams .../bindings/sound/davinci-mcbsp.txt | 50 --- .../bindings/sound/davinci-mcbsp.yaml | 119 +++++++ include/linux/platform_data/davinci_asp.h | 15 - sound/soc/ti/davinci-i2s.c | 333 ++++++++++++++---- 4 files changed, 376 insertions(+), 141 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/davinci-mcbsp.txt create mode 100644 Documentation/devicetree/bindings/sound/davinci-mcbsp.yaml -- 2.43.2

5 38

[PATCH AUTOSEL 5.15 13/34] ALSA: firewire-lib: handle quirk to calculate payload quadlets as data block counter
by Sasha Levin 29 Mar '24

29 Mar '24

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit 4a486439d2ca85752c46711f373b6ddc107bb35d ] Miglia Harmony Audio (OXFW970) has a quirk to put the number of accumulated quadlets in CIP payload into the dbc field of CIP header. This commit handles the quirk in the packet processing layer. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20240218074128.95210-4-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/amdtp-stream.c | 12 ++++++++---- sound/firewire/amdtp-stream.h | 4 ++++ 2 files changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index f8b644cb9157a..8753125683692 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -771,10 +771,14 @@ static int check_cip_header(struct amdtp_stream *s, const __be32 *buf, } else { unsigned int dbc_interval; - if (*data_blocks > 0 && s->ctx_data.tx.dbc_interval > 0) - dbc_interval = s->ctx_data.tx.dbc_interval; - else - dbc_interval = *data_blocks; + if (!(s->flags & CIP_DBC_IS_PAYLOAD_QUADLETS)) { + if (*data_blocks > 0 && s->ctx_data.tx.dbc_interval > 0) + dbc_interval = s->ctx_data.tx.dbc_interval; + else + dbc_interval = *data_blocks; + } else { + dbc_interval = payload_length / sizeof(__be32); + } lost = dbc != ((*data_block_counter + dbc_interval) & 0xff); } diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 1f957c946c956..cf9ab347277f2 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -37,6 +37,9 @@ * the value of current SYT_INTERVAL; e.g. initial value is not zero. * @CIP_UNAWARE_SYT: For outgoing packet, the value in SYT field of CIP is 0xffff. * For incoming packet, the value in SYT field of CIP is not handled. + * @CIP_DBC_IS_PAYLOAD_QUADLETS: Available for incoming packet, and only effective with + * CIP_DBC_IS_END_EVENT flag. The value of dbc field is the number of accumulated quadlets + * in CIP payload, instead of the number of accumulated data blocks. */ enum cip_flags { CIP_NONBLOCKING = 0x00, @@ -51,6 +54,7 @@ enum cip_flags { CIP_NO_HEADER = 0x100, CIP_UNALIGHED_DBC = 0x200, CIP_UNAWARE_SYT = 0x400, + CIP_DBC_IS_PAYLOAD_QUADLETS = 0x800, }; /** -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 20/52] ALSA: firewire-lib: handle quirk to calculate payload quadlets as data block counter
by Sasha Levin 29 Mar '24

29 Mar '24

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit 4a486439d2ca85752c46711f373b6ddc107bb35d ] Miglia Harmony Audio (OXFW970) has a quirk to put the number of accumulated quadlets in CIP payload into the dbc field of CIP header. This commit handles the quirk in the packet processing layer. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20240218074128.95210-4-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/amdtp-stream.c | 12 ++++++++---- sound/firewire/amdtp-stream.h | 4 ++++ 2 files changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index f8b644cb9157a..8753125683692 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -771,10 +771,14 @@ static int check_cip_header(struct amdtp_stream *s, const __be32 *buf, } else { unsigned int dbc_interval; - if (*data_blocks > 0 && s->ctx_data.tx.dbc_interval > 0) - dbc_interval = s->ctx_data.tx.dbc_interval; - else - dbc_interval = *data_blocks; + if (!(s->flags & CIP_DBC_IS_PAYLOAD_QUADLETS)) { + if (*data_blocks > 0 && s->ctx_data.tx.dbc_interval > 0) + dbc_interval = s->ctx_data.tx.dbc_interval; + else + dbc_interval = *data_blocks; + } else { + dbc_interval = payload_length / sizeof(__be32); + } lost = dbc != ((*data_block_counter + dbc_interval) & 0xff); } diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index 1f957c946c956..cf9ab347277f2 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -37,6 +37,9 @@ * the value of current SYT_INTERVAL; e.g. initial value is not zero. * @CIP_UNAWARE_SYT: For outgoing packet, the value in SYT field of CIP is 0xffff. * For incoming packet, the value in SYT field of CIP is not handled. + * @CIP_DBC_IS_PAYLOAD_QUADLETS: Available for incoming packet, and only effective with + * CIP_DBC_IS_END_EVENT flag. The value of dbc field is the number of accumulated quadlets + * in CIP payload, instead of the number of accumulated data blocks. */ enum cip_flags { CIP_NONBLOCKING = 0x00, @@ -51,6 +54,7 @@ enum cip_flags { CIP_NO_HEADER = 0x100, CIP_UNALIGHED_DBC = 0x200, CIP_UNAWARE_SYT = 0x400, + CIP_DBC_IS_PAYLOAD_QUADLETS = 0x800, }; /** -- 2.43.0

1 0

[PATCH AUTOSEL 6.1 13/52] ASoC: Intel: common: DMI remap for rebranded Intel NUC M15 (LAPRC710) laptops
by Sasha Levin 29 Mar '24

29 Mar '24

From: mosomate <mosomate(a)gmail.com> [ Upstream commit c13e03126a5be90781084437689724254c8226e1 ] Added DMI quirk to handle the rebranded variants of Intel NUC M15 (LAPRC710) laptops. The DMI matching is based on motherboard attributes. Link: https://github.com/thesofproject/linux/issues/4218 Signed-off-by: Máté Mosonyi <mosomate(a)gmail.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240208165545.93811-20-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/dmi-quirks.c | 8 ++++++++ sound/soc/intel/boards/sof_sdw.c | 11 +++++++++++ 2 files changed, 19 insertions(+) diff --git a/drivers/soundwire/dmi-quirks.c b/drivers/soundwire/dmi-quirks.c index 9ebdd0cd0b1cf..91ab97a456fa9 100644 --- a/drivers/soundwire/dmi-quirks.c +++ b/drivers/soundwire/dmi-quirks.c @@ -130,6 +130,14 @@ static const struct dmi_system_id adr_remap_quirk_table[] = { }, .driver_data = (void *)intel_rooks_county, }, + { + /* quirk used for NUC15 LAPRC710 skew */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "LAPRC710"), + }, + .driver_data = (void *)intel_rooks_county, + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 985012f2003e2..d1e6e4208c376 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -224,6 +224,17 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_SDW_PCH_DMIC | RT711_JD2_100K), }, + { + /* NUC15 LAPRC710 skews */ + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "LAPRC710"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + SOF_SDW_PCH_DMIC | + RT711_JD2_100K), + }, /* TigerLake-SDCA devices */ { .callback = sof_sdw_quirk_cb, -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 33/75] ASoC: Intel: avs: Populate board selection with new I2S entries
by Sasha Levin 29 Mar '24

29 Mar '24

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit 5b417fe0cded0b5917683398e6519aae8045cd40 ] Update board selection with tables specifying supported I2S configurations. DMIC/HDAudio board selection require no update as dmic/hdaudio machine boards are generic and not tied to any specific codec. Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://msgid.link/r/20240220115035.770402-11-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/avs/board_selection.c | 85 +++++++++++++++++++++++++++ 1 file changed, 85 insertions(+) diff --git a/sound/soc/intel/avs/board_selection.c b/sound/soc/intel/avs/board_selection.c index 59a13feec57b2..0db5e530a8de2 100644 --- a/sound/soc/intel/avs/board_selection.c +++ b/sound/soc/intel/avs/board_selection.c @@ -227,6 +227,82 @@ static struct snd_soc_acpi_mach avs_gml_i2s_machines[] = { {}, }; +static struct snd_soc_acpi_mach avs_cnl_i2s_machines[] = { + { + .id = "INT34C2", + .drv_name = "avs_rt274", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt274-tplg.bin", + }, + { + .id = "10EC5682", + .drv_name = "avs_rt5682", + .mach_params = { + .i2s_link_mask = AVS_SSP(1), + }, + .tplg_filename = "rt5682-tplg.bin", + }, + {}, +}; + +static struct snd_soc_acpi_mach avs_icl_i2s_machines[] = { + { + .id = "INT343A", + .drv_name = "avs_rt298", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt298-tplg.bin", + }, + { + .id = "INT34C2", + .drv_name = "avs_rt274", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt274-tplg.bin", + }, + {}, +}; + +static struct snd_soc_acpi_mach avs_tgl_i2s_machines[] = { + { + .id = "INT34C2", + .drv_name = "avs_rt274", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt274-tplg.bin", + }, + { + .id = "10EC0298", + .drv_name = "avs_rt298", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt298-tplg.bin", + }, + { + .id = "10EC1308", + .drv_name = "avs_rt1308", + .mach_params = { + .i2s_link_mask = AVS_SSP(1), + }, + .tplg_filename = "rt1308-tplg.bin", + }, + { + .id = "ESSX8336", + .drv_name = "avs_es8336", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "es8336-tplg.bin", + }, + {}, +}; + static struct snd_soc_acpi_mach avs_test_i2s_machines[] = { { .drv_name = "avs_i2s_test", @@ -287,6 +363,15 @@ static const struct avs_acpi_boards i2s_boards[] = { AVS_MACH_ENTRY(HDA_KBL_LP, avs_kbl_i2s_machines), AVS_MACH_ENTRY(HDA_APL, avs_apl_i2s_machines), AVS_MACH_ENTRY(HDA_GML, avs_gml_i2s_machines), + AVS_MACH_ENTRY(HDA_CNL_LP, avs_cnl_i2s_machines), + AVS_MACH_ENTRY(HDA_CNL_H, avs_cnl_i2s_machines), + AVS_MACH_ENTRY(HDA_CML_LP, avs_cnl_i2s_machines), + AVS_MACH_ENTRY(HDA_ICL_LP, avs_icl_i2s_machines), + AVS_MACH_ENTRY(HDA_TGL_LP, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_EHL_0, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_ADL_P, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_RPL_P_0, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_RPL_M, avs_tgl_i2s_machines), {}, }; -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 32/75] ALSA: firewire-lib: handle quirk to calculate payload quadlets as data block counter
by Sasha Levin 29 Mar '24

29 Mar '24

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit 4a486439d2ca85752c46711f373b6ddc107bb35d ] Miglia Harmony Audio (OXFW970) has a quirk to put the number of accumulated quadlets in CIP payload into the dbc field of CIP header. This commit handles the quirk in the packet processing layer. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20240218074128.95210-4-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/amdtp-stream.c | 12 ++++++++---- sound/firewire/amdtp-stream.h | 4 ++++ 2 files changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 7be17bca257f0..c9f153f85ae6b 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -773,10 +773,14 @@ static int check_cip_header(struct amdtp_stream *s, const __be32 *buf, } else { unsigned int dbc_interval; - if (*data_blocks > 0 && s->ctx_data.tx.dbc_interval > 0) - dbc_interval = s->ctx_data.tx.dbc_interval; - else - dbc_interval = *data_blocks; + if (!(s->flags & CIP_DBC_IS_PAYLOAD_QUADLETS)) { + if (*data_blocks > 0 && s->ctx_data.tx.dbc_interval > 0) + dbc_interval = s->ctx_data.tx.dbc_interval; + else + dbc_interval = *data_blocks; + } else { + dbc_interval = payload_length / sizeof(__be32); + } lost = dbc != ((*data_block_counter + dbc_interval) & 0xff); } diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index b7ff44751ab93..a1ed2e80f91a7 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -37,6 +37,9 @@ * the value of current SYT_INTERVAL; e.g. initial value is not zero. * @CIP_UNAWARE_SYT: For outgoing packet, the value in SYT field of CIP is 0xffff. * For incoming packet, the value in SYT field of CIP is not handled. + * @CIP_DBC_IS_PAYLOAD_QUADLETS: Available for incoming packet, and only effective with + * CIP_DBC_IS_END_EVENT flag. The value of dbc field is the number of accumulated quadlets + * in CIP payload, instead of the number of accumulated data blocks. */ enum cip_flags { CIP_NONBLOCKING = 0x00, @@ -51,6 +54,7 @@ enum cip_flags { CIP_NO_HEADER = 0x100, CIP_UNALIGHED_DBC = 0x200, CIP_UNAWARE_SYT = 0x400, + CIP_DBC_IS_PAYLOAD_QUADLETS = 0x800, }; /** -- 2.43.0

1 0

[PATCH AUTOSEL 6.6 17/75] ASoC: Intel: common: DMI remap for rebranded Intel NUC M15 (LAPRC710) laptops
by Sasha Levin 29 Mar '24

29 Mar '24

From: mosomate <mosomate(a)gmail.com> [ Upstream commit c13e03126a5be90781084437689724254c8226e1 ] Added DMI quirk to handle the rebranded variants of Intel NUC M15 (LAPRC710) laptops. The DMI matching is based on motherboard attributes. Link: https://github.com/thesofproject/linux/issues/4218 Signed-off-by: Máté Mosonyi <mosomate(a)gmail.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240208165545.93811-20-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/dmi-quirks.c | 8 ++++++++ sound/soc/intel/boards/sof_sdw.c | 11 +++++++++++ 2 files changed, 19 insertions(+) diff --git a/drivers/soundwire/dmi-quirks.c b/drivers/soundwire/dmi-quirks.c index 9ebdd0cd0b1cf..91ab97a456fa9 100644 --- a/drivers/soundwire/dmi-quirks.c +++ b/drivers/soundwire/dmi-quirks.c @@ -130,6 +130,14 @@ static const struct dmi_system_id adr_remap_quirk_table[] = { }, .driver_data = (void *)intel_rooks_county, }, + { + /* quirk used for NUC15 LAPRC710 skew */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "LAPRC710"), + }, + .driver_data = (void *)intel_rooks_county, + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 9ed572141fe5b..0ea7812125fee 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -243,6 +243,17 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_SDW_PCH_DMIC | RT711_JD2_100K), }, + { + /* NUC15 LAPRC710 skews */ + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "LAPRC710"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + SOF_SDW_PCH_DMIC | + RT711_JD2_100K), + }, /* TigerLake-SDCA devices */ { .callback = sof_sdw_quirk_cb, -- 2.43.0

1 0

[PATCH AUTOSEL 6.8 48/98] ASoC: Intel: avs: Populate board selection with new I2S entries
by Sasha Levin 29 Mar '24

29 Mar '24

From: Cezary Rojewski <cezary.rojewski(a)intel.com> [ Upstream commit 5b417fe0cded0b5917683398e6519aae8045cd40 ] Update board selection with tables specifying supported I2S configurations. DMIC/HDAudio board selection require no update as dmic/hdaudio machine boards are generic and not tied to any specific codec. Reviewed-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> Link: https://msgid.link/r/20240220115035.770402-11-cezary.rojewski@intel.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/avs/board_selection.c | 85 +++++++++++++++++++++++++++ 1 file changed, 85 insertions(+) diff --git a/sound/soc/intel/avs/board_selection.c b/sound/soc/intel/avs/board_selection.c index 8e91eece992db..8360ce557401c 100644 --- a/sound/soc/intel/avs/board_selection.c +++ b/sound/soc/intel/avs/board_selection.c @@ -236,6 +236,82 @@ static struct snd_soc_acpi_mach avs_gml_i2s_machines[] = { {}, }; +static struct snd_soc_acpi_mach avs_cnl_i2s_machines[] = { + { + .id = "INT34C2", + .drv_name = "avs_rt274", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt274-tplg.bin", + }, + { + .id = "10EC5682", + .drv_name = "avs_rt5682", + .mach_params = { + .i2s_link_mask = AVS_SSP(1), + }, + .tplg_filename = "rt5682-tplg.bin", + }, + {}, +}; + +static struct snd_soc_acpi_mach avs_icl_i2s_machines[] = { + { + .id = "INT343A", + .drv_name = "avs_rt298", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt298-tplg.bin", + }, + { + .id = "INT34C2", + .drv_name = "avs_rt274", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt274-tplg.bin", + }, + {}, +}; + +static struct snd_soc_acpi_mach avs_tgl_i2s_machines[] = { + { + .id = "INT34C2", + .drv_name = "avs_rt274", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt274-tplg.bin", + }, + { + .id = "10EC0298", + .drv_name = "avs_rt298", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "rt298-tplg.bin", + }, + { + .id = "10EC1308", + .drv_name = "avs_rt1308", + .mach_params = { + .i2s_link_mask = AVS_SSP(1), + }, + .tplg_filename = "rt1308-tplg.bin", + }, + { + .id = "ESSX8336", + .drv_name = "avs_es8336", + .mach_params = { + .i2s_link_mask = AVS_SSP(0), + }, + .tplg_filename = "es8336-tplg.bin", + }, + {}, +}; + static struct snd_soc_acpi_mach avs_test_i2s_machines[] = { { .drv_name = "avs_i2s_test", @@ -296,6 +372,15 @@ static const struct avs_acpi_boards i2s_boards[] = { AVS_MACH_ENTRY(HDA_KBL_LP, avs_kbl_i2s_machines), AVS_MACH_ENTRY(HDA_APL, avs_apl_i2s_machines), AVS_MACH_ENTRY(HDA_GML, avs_gml_i2s_machines), + AVS_MACH_ENTRY(HDA_CNL_LP, avs_cnl_i2s_machines), + AVS_MACH_ENTRY(HDA_CNL_H, avs_cnl_i2s_machines), + AVS_MACH_ENTRY(HDA_CML_LP, avs_cnl_i2s_machines), + AVS_MACH_ENTRY(HDA_ICL_LP, avs_icl_i2s_machines), + AVS_MACH_ENTRY(HDA_TGL_LP, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_EHL_0, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_ADL_P, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_RPL_P_0, avs_tgl_i2s_machines), + AVS_MACH_ENTRY(HDA_RPL_M, avs_tgl_i2s_machines), {}, }; -- 2.43.0

1 0

[PATCH AUTOSEL 6.8 45/98] ALSA: firewire-lib: handle quirk to calculate payload quadlets as data block counter
by Sasha Levin 29 Mar '24

29 Mar '24

From: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> [ Upstream commit 4a486439d2ca85752c46711f373b6ddc107bb35d ] Miglia Harmony Audio (OXFW970) has a quirk to put the number of accumulated quadlets in CIP payload into the dbc field of CIP header. This commit handles the quirk in the packet processing layer. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> Link: https://lore.kernel.org/r/20240218074128.95210-4-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/firewire/amdtp-stream.c | 12 ++++++++---- sound/firewire/amdtp-stream.h | 4 ++++ 2 files changed, 12 insertions(+), 4 deletions(-) diff --git a/sound/firewire/amdtp-stream.c b/sound/firewire/amdtp-stream.c index 7be17bca257f0..c9f153f85ae6b 100644 --- a/sound/firewire/amdtp-stream.c +++ b/sound/firewire/amdtp-stream.c @@ -773,10 +773,14 @@ static int check_cip_header(struct amdtp_stream *s, const __be32 *buf, } else { unsigned int dbc_interval; - if (*data_blocks > 0 && s->ctx_data.tx.dbc_interval > 0) - dbc_interval = s->ctx_data.tx.dbc_interval; - else - dbc_interval = *data_blocks; + if (!(s->flags & CIP_DBC_IS_PAYLOAD_QUADLETS)) { + if (*data_blocks > 0 && s->ctx_data.tx.dbc_interval > 0) + dbc_interval = s->ctx_data.tx.dbc_interval; + else + dbc_interval = *data_blocks; + } else { + dbc_interval = payload_length / sizeof(__be32); + } lost = dbc != ((*data_block_counter + dbc_interval) & 0xff); } diff --git a/sound/firewire/amdtp-stream.h b/sound/firewire/amdtp-stream.h index b7ff44751ab93..a1ed2e80f91a7 100644 --- a/sound/firewire/amdtp-stream.h +++ b/sound/firewire/amdtp-stream.h @@ -37,6 +37,9 @@ * the value of current SYT_INTERVAL; e.g. initial value is not zero. * @CIP_UNAWARE_SYT: For outgoing packet, the value in SYT field of CIP is 0xffff. * For incoming packet, the value in SYT field of CIP is not handled. + * @CIP_DBC_IS_PAYLOAD_QUADLETS: Available for incoming packet, and only effective with + * CIP_DBC_IS_END_EVENT flag. The value of dbc field is the number of accumulated quadlets + * in CIP payload, instead of the number of accumulated data blocks. */ enum cip_flags { CIP_NONBLOCKING = 0x00, @@ -51,6 +54,7 @@ enum cip_flags { CIP_NO_HEADER = 0x100, CIP_UNALIGHED_DBC = 0x200, CIP_UNAWARE_SYT = 0x400, + CIP_DBC_IS_PAYLOAD_QUADLETS = 0x800, }; /** -- 2.43.0

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[PATCH AUTOSEL 6.8 27/98] ASoC: Intel: common: DMI remap for rebranded Intel NUC M15 (LAPRC710) laptops
by Sasha Levin 29 Mar '24

29 Mar '24

From: mosomate <mosomate(a)gmail.com> [ Upstream commit c13e03126a5be90781084437689724254c8226e1 ] Added DMI quirk to handle the rebranded variants of Intel NUC M15 (LAPRC710) laptops. The DMI matching is based on motherboard attributes. Link: https://github.com/thesofproject/linux/issues/4218 Signed-off-by: Máté Mosonyi <mosomate(a)gmail.com> Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240208165545.93811-20-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- drivers/soundwire/dmi-quirks.c | 8 ++++++++ sound/soc/intel/boards/sof_sdw.c | 11 +++++++++++ 2 files changed, 19 insertions(+) diff --git a/drivers/soundwire/dmi-quirks.c b/drivers/soundwire/dmi-quirks.c index 9ebdd0cd0b1cf..91ab97a456fa9 100644 --- a/drivers/soundwire/dmi-quirks.c +++ b/drivers/soundwire/dmi-quirks.c @@ -130,6 +130,14 @@ static const struct dmi_system_id adr_remap_quirk_table[] = { }, .driver_data = (void *)intel_rooks_county, }, + { + /* quirk used for NUC15 LAPRC710 skew */ + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "LAPRC710"), + }, + .driver_data = (void *)intel_rooks_county, + }, { .matches = { DMI_MATCH(DMI_SYS_VENDOR, "Dell Inc"), diff --git a/sound/soc/intel/boards/sof_sdw.c b/sound/soc/intel/boards/sof_sdw.c index 300391fbc2fc2..cf14bd3053fc9 100644 --- a/sound/soc/intel/boards/sof_sdw.c +++ b/sound/soc/intel/boards/sof_sdw.c @@ -236,6 +236,17 @@ static const struct dmi_system_id sof_sdw_quirk_table[] = { SOF_SDW_PCH_DMIC | RT711_JD2_100K), }, + { + /* NUC15 LAPRC710 skews */ + .callback = sof_sdw_quirk_cb, + .matches = { + DMI_MATCH(DMI_BOARD_VENDOR, "Intel Corporation"), + DMI_MATCH(DMI_BOARD_NAME, "LAPRC710"), + }, + .driver_data = (void *)(SOF_SDW_TGL_HDMI | + SOF_SDW_PCH_DMIC | + RT711_JD2_100K), + }, /* TigerLake-SDCA devices */ { .callback = sof_sdw_quirk_cb, -- 2.43.0

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[PATCH AUTOSEL 6.8 26/98] ASoC: Intel: sof_rt5682: dmi quirk cleanup for mtl boards
by Sasha Levin 29 Mar '24

29 Mar '24

From: Brent Lu <brent.lu(a)intel.com> [ Upstream commit 7a2a8730d51f95b263a1e8b888598dc6395220dc ] Some dmi quirks are duplicated since codec and amplifier type are removed from board quirk. Remove redundant quirks. Reviewed-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Brent Lu <brent.lu(a)intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20240208165545.93811-7-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/intel/boards/sof_rt5682.c | 40 ----------------------------- 1 file changed, 40 deletions(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index cd50f26d1edbe..02705034e5713 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -154,46 +154,6 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { SOF_RT5682_SSP_AMP(2) | SOF_RT5682_NUM_HDMIDEV(4)), }, - { - .callback = sof_rt5682_quirk_cb, - .matches = { - DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Rex"), - DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98360_ALC5682I_I2S"), - }, - .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(2) | - SOF_RT5682_SSP_AMP(0) | - SOF_RT5682_NUM_HDMIDEV(3) | - SOF_BT_OFFLOAD_SSP(1) | - SOF_SSP_BT_OFFLOAD_PRESENT - ), - }, - { - .callback = sof_rt5682_quirk_cb, - .matches = { - DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Rex"), - DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98360_ALC5682I_DISCRETE_I2S_BT"), - }, - .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(2) | - SOF_RT5682_SSP_AMP(0) | - SOF_RT5682_NUM_HDMIDEV(3) | - SOF_BT_OFFLOAD_SSP(1) | - SOF_SSP_BT_OFFLOAD_PRESENT - ), - }, - { - .callback = sof_rt5682_quirk_cb, - .matches = { - DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Rex"), - DMI_MATCH(DMI_OEM_STRING, "AUDIO-ALC1019_ALC5682I_I2S"), - }, - .driver_data = (void *)(SOF_RT5682_MCLK_EN | - SOF_RT5682_SSP_CODEC(2) | - SOF_RT5682_SSP_AMP(0) | - SOF_RT5682_NUM_HDMIDEV(3) - ), - }, { .callback = sof_rt5682_quirk_cb, .matches = { -- 2.43.0

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[PATCH 0/6] ASoC: rt-sdw: fix locking and improve error logs
by Pierre-Louis Bossart 29 Mar '24

29 Mar '24

Fix a set of problematic locking sequences and update error messages, tested on SOF/SoundWire platforms. Pierre-Louis Bossart (6): ASoC: rt5682-sdw: fix locking sequence ASoC: rt711-sdca: fix locking sequence ASoC: rt711-sdw: fix locking sequence ASoC: rt712-sdca-sdw: fix locking sequence ASoC: rt722-sdca-sdw: fix locking sequence ASoC: rt-sdw*: add __func__ to all error logs sound/soc/codecs/rt1316-sdw.c | 8 +++--- sound/soc/codecs/rt1318-sdw.c | 8 +++--- sound/soc/codecs/rt5682-sdw.c | 16 +++++------ sound/soc/codecs/rt700.c | 16 +++++------ sound/soc/codecs/rt711-sdca-sdw.c | 6 ++-- sound/soc/codecs/rt711-sdca.c | 18 ++++++------ sound/soc/codecs/rt711-sdw.c | 8 +++--- sound/soc/codecs/rt711.c | 16 +++++------ sound/soc/codecs/rt712-sdca-dmic.c | 24 +++++++++------- sound/soc/codecs/rt712-sdca-sdw.c | 7 +++-- sound/soc/codecs/rt712-sdca.c | 20 ++++++------- sound/soc/codecs/rt715-sdca-sdw.c | 2 +- sound/soc/codecs/rt715-sdca.c | 46 +++++++++++++++--------------- sound/soc/codecs/rt715-sdw.c | 4 +-- sound/soc/codecs/rt715.c | 24 ++++++++-------- sound/soc/codecs/rt722-sdca-sdw.c | 4 +-- sound/soc/codecs/rt722-sdca.c | 21 +++++++------- 17 files changed, 126 insertions(+), 122 deletions(-) -- 2.40.1

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