May 2023 - Alsa-devel - mailman.alsa-project.org

[PATCH v3 1/5] ASoC: tas2781: Add Header file for tas2781 driver
by Shenghao Ding 19 May '23

19 May '23

Create Header file for tas2781 driver. Signed-off-by: Shenghao Ding <13916275206(a)139.com> --- Changes in v3: - correct some enums have capitalized Chn, chn to all Changes to be committed: new file: include/sound/tas2781-dsp.h new file: include/sound/tas2781-tlv.h new file: include/sound/tas2781.h --- include/sound/tas2781-dsp.h | 190 ++++++++++++++++++++++++++++++++++++ include/sound/tas2781-tlv.h | 22 +++++ include/sound/tas2781.h | 176 +++++++++++++++++++++++++++++++++ 3 files changed, 388 insertions(+) create mode 100644 include/sound/tas2781-dsp.h create mode 100644 include/sound/tas2781-tlv.h create mode 100644 include/sound/tas2781.h diff --git a/include/sound/tas2781-dsp.h b/include/sound/tas2781-dsp.h new file mode 100644 index 000000000000..b05f20bc4a1f --- /dev/null +++ b/include/sound/tas2781-dsp.h @@ -0,0 +1,190 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#ifndef __TASDEVICE_DSP_H__ +#define __TASDEVICE_DSP_H__ + +#define MAIN_ALL_DEVICES 0x0d +#define MAIN_DEVICE_A 0x01 +#define MAIN_DEVICE_B 0x08 +#define MAIN_DEVICE_C 0x10 +#define MAIN_DEVICE_D 0x14 +#define COEFF_DEVICE_A 0x03 +#define COEFF_DEVICE_B 0x0a +#define COEFF_DEVICE_C 0x11 +#define COEFF_DEVICE_D 0x15 +#define PRE_DEVICE_A 0x04 +#define PRE_DEVICE_B 0x0b +#define PRE_DEVICE_C 0x12 +#define PRE_DEVICE_D 0x16 + +#define PPC3_VERSION 0x4100 +#define PPC3_VERSION_TAS2781 0x14c00 +#define RCA_CONFIGID_BYPASS_ALL 0 +#define TASDEVICE_DEVICE_SUM 8 +#define TASDEVICE_CONFIG_SUM 64 + +enum channel { + top_left_Chn, + top_right_chn, + bottom_left_Chn, + bottom_right_chn, + max_chn, +}; + +enum tasdevice_dsp_dev_idx { + TASDEVICE_DSP_TAS_2555 = 0, + TASDEVICE_DSP_TAS_2555_STEREO, + TASDEVICE_DSP_TAS_2557_MONO, + TASDEVICE_DSP_TAS_2557_DUAL_MONO, + TASDEVICE_DSP_TAS_2559, + TASDEVICE_DSP_TAS_2563, + TASDEVICE_DSP_TAS_2563_DUAL_MONO = 7, + TASDEVICE_DSP_TAS_2563_QUAD, + TASDEVICE_DSP_TAS_2563_21, + TASDEVICE_DSP_TAS_2781, + TASDEVICE_DSP_TAS_2781_DUAL_MONO, + TASDEVICE_DSP_TAS_2781_21, + TASDEVICE_DSP_TAS_2781_QUAD, + TASDEVICE_DSP_TAS_MAX_DEVICE +}; + +struct tasdevice_fw_fixed_hdr { + unsigned int fwsize; + unsigned int ppcver; + unsigned int drv_ver; +}; + +struct tasdevice_dspfw_hdr { + struct tasdevice_fw_fixed_hdr fixed_hdr; + unsigned short device_family; + unsigned short device; + unsigned char ndev; +}; + +struct tasdev_blk { + int nr_retry; + unsigned int type; + unsigned char is_pchksum_present; + unsigned char pchksum; + unsigned char is_ychksum_present; + unsigned char ychksum; + unsigned int nr_cmds; + unsigned int blk_size; + unsigned int nr_subblocks; + unsigned char *data; +}; + +struct tasdevice_data { + char name[64]; + unsigned int nr_blk; + struct tasdev_blk *dev_blks; +}; + +struct tasdevice_prog { + unsigned int prog_size; + struct tasdevice_data dev_data; +}; + +struct tasdevice_config { + unsigned int cfg_size; + char name[64]; + struct tasdevice_data dev_data; +}; + +struct tasdevice_calibration { + struct tasdevice_data dev_data; +}; + +struct tasdevice_fw { + struct tasdevice_dspfw_hdr fw_hdr; + unsigned short nr_programs; + struct tasdevice_prog *programs; + unsigned short nr_configurations; + struct tasdevice_config *configs; + unsigned short nr_calibrations; + struct tasdevice_calibration *calibrations; + struct device *dev; +}; + +enum tasdevice_dsp_fw_state { + TASDEVICE_DSP_FW_NONE = 0, + TASDEVICE_DSP_FW_PENDING, + TASDEVICE_DSP_FW_FAIL, + TASDEVICE_DSP_FW_ALL_OK, +}; + +enum tasdevice_bin_blk_type { + TASDEVICE_BIN_BLK_COEFF = 1, + TASDEVICE_BIN_BLK_POST_POWER_UP, + TASDEVICE_BIN_BLK_PRE_SHUTDOWN, + TASDEVICE_BIN_BLK_PRE_POWER_UP, + TASDEVICE_BIN_BLK_POST_SHUTDOWN +}; + +struct tasdevice_rca_hdr { + unsigned int img_sz; + unsigned int checksum; + unsigned int binary_version_num; + unsigned int drv_fw_version; + unsigned char plat_type; + unsigned char dev_family; + unsigned char reserve; + unsigned char ndev; + unsigned char devs[TASDEVICE_DEVICE_SUM]; + unsigned int nconfig; + unsigned int config_size[TASDEVICE_CONFIG_SUM]; +}; + +struct tasdev_blk_data { + unsigned char dev_idx; + unsigned char block_type; + unsigned short yram_checksum; + unsigned int block_size; + unsigned int n_subblks; + unsigned char *regdata; +}; + +struct tasdevice_config_info { + unsigned int nblocks; + unsigned int real_nblocks; + unsigned char active_dev; + struct tasdev_blk_data **blk_data; +}; + +struct tasdevice_rca { + struct tasdevice_rca_hdr fw_hdr; + int ncfgs; + struct tasdevice_config_info **cfg_info; + int profile_cfg_id; +}; + +void tasdevice_select_cfg_blk(void *context, int conf_no, + unsigned char block_type); +void tasdevice_config_info_remove(void *context); +void tasdevice_dsp_remove(void *context); +int tasdevice_dsp_parser(void *context); +int tasdevice_rca_parser(void *context, const struct firmware *fmw); +void tasdevice_dsp_remove(void *context); +void tasdevice_calbin_remove(void *context); +int tasdevice_select_tuningprm_cfg(void *context, int prm, + int cfg_no, int rca_conf_no); +int tasdevice_prmg_load(void *context, int prm_no); +int tasdevice_prmg_calibdata_load(void *context, int prm_no); +void tasdevice_tuning_switch(void *context, int state); +int tas2781_load_calibration(void *context, char *file_name, + enum channel i); + +#endif diff --git a/include/sound/tas2781-tlv.h b/include/sound/tas2781-tlv.h new file mode 100644 index 000000000000..f4310dce655a --- /dev/null +++ b/include/sound/tas2781-tlv.h @@ -0,0 +1,22 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#ifndef __TAS2781_TLV_H__ +#define __TAS2781_TLV_H__ + +static const DECLARE_TLV_DB_SCALE(dvc_tlv, -10000, 100, 0); +static const DECLARE_TLV_DB_SCALE(amp_vol_tlv, 1100, 50, 0); + +#endif /* __TAS2781_LIB_H__ */ diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h new file mode 100644 index 000000000000..c98955bda07e --- /dev/null +++ b/include/sound/tas2781.h @@ -0,0 +1,176 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +// +// ALSA SoC Texas Instruments TAS2781 Audio Smart Amplifier +// +// Copyright (C) 2022 - 2023 Texas Instruments Incorporated +// https://www.ti.com +// +// The TAS2781 driver implements a flexible and configurable +// algo coefficient setting for one, two, or even multiple +// TAS2781 chips. +// +// Author: Shenghao Ding <shenghao-ding(a)ti.com> +// Author: Kevin Lu <kevin-lu(a)ti.com> +// + +#ifndef __TAS2781_H__ +#define __TAS2781_H__ + +#include <linux/kernel.h> +#include "tas2781-dsp.h" + +/* version number */ +#define TAS2781_DRV_VER 1 +#define SMARTAMP_MODULE_NAME "tas2781" +#define TAS2781_GLOBAL_ADDR 0x40 +#define TASDEVICE_RATES (SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_88200) +#define TASDEVICE_MAX_CHANNELS 8 + +#define TASDEVICE_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/*PAGE Control Register (available in page0 of each book) */ +#define TASDEVICE_PAGE_SELECT 0x00 +#define TASDEVICE_BOOKCTL_PAGE 0x00 +#define TASDEVICE_BOOKCTL_REG 127 +#define TASDEVICE_BOOK_ID(reg) (reg / (256 * 128)) +#define TASDEVICE_PAGE_ID(reg) ((reg % (256 * 128)) / 128) +#define TASDEVICE_PAGE_REG(reg) ((reg % (256 * 128)) % 128) +#define TASDEVICE_PGRG(reg) (reg % (256 * 128)) +#define TASDEVICE_REG(book, page, reg) (((book * 256 * 128) + \ + (page * 128)) + reg) + +/*Software Reset */ +#define TAS2781_REG_SWRESET TASDEVICE_REG(0x0, 0X0, 0x02) +#define TAS2781_REG_SWRESET_RESET BIT(0) + +/* Enable Global addresses */ +#define TAS2781_MISC_CFG2 TASDEVICE_REG(0x0, 0X0, 0x07) +#define TAS2781_GLOBAL_ADDR_MASK BIT(1) +#define TAS2781_GLOBAL_ADDR_ENABLE BIT(1) + +/*I2C Checksum */ +#define TASDEVICE_I2CChecksum TASDEVICE_REG(0x0, 0x0, 0x7E) + +/* Volume control */ +#define TAS2781_DVC_LVL TASDEVICE_REG(0x0, 0x0, 0x1A) +#define TAS2781_AMP_LEVEL TASDEVICE_REG(0x0, 0x0, 0x03) +#define TAS2781_AMP_LEVEL_MASK GENMASK(5, 1) + +#define TASDEVICE_CMD_SING_W 0x1 +#define TASDEVICE_CMD_BURST 0x2 +#define TASDEVICE_CMD_DELAY 0x3 +#define TASDEVICE_CMD_FIELD_W 0x4 + +enum audio_device { + TAS2781 = 0, +}; + +struct tasdevice { + struct tasdevice_fw *cali_data_fmw; + unsigned int dev_addr; + unsigned int err_code; + unsigned char cur_book; + short cur_prog; + short cur_conf; + bool is_loading; + bool is_loaderr; +}; + +/* + * This item is used to store the generic i2c address of + * all the tas2781 devices for I2C broadcast during the multi-device + * writes, useless in mono case. + */ +struct global_addr { + int ref_cnt; + unsigned int dev_addr; + unsigned char cur_book; +}; + +struct tasdevice_irqinfo { + int irq_gpio; + int irq; +}; + +struct calidata { + unsigned char *data; + unsigned long total_sz; +}; + +struct tasdevice_priv { + struct tasdevice tasdevice[max_chn]; + struct tasdevice_irqinfo irq_info; + struct global_addr glb_addr; + struct tasdevice_rca rcabin; + struct tasdevice_fw *fmw; + struct regmap *regmap; + struct mutex codec_lock; + struct calidata cali_data; + struct gpio_desc *reset; + struct device *dev; + struct tm tm; + + unsigned char crc8_lkp_tbl[CRC8_TABLE_SIZE]; + unsigned char cal_binaryname[max_chn][64]; + unsigned char coef_binaryname[64]; + unsigned char rca_binaryname[64]; + unsigned char dev_name[32]; + unsigned char ndev; + unsigned int magic_num; + unsigned int chip_id; + unsigned int sysclk; + const char *acpi_subsystem_id; + int index; + int cur_prog; + int cur_conf; + int fw_state; + void *client; + void *codec; + bool isacpi; + bool playback_started; + void (*set_global_mode)(struct tasdevice_priv *tas_priv); + int (*fw_parse_variable_header)(struct tasdevice_priv *tas_priv, + const struct firmware *fmw, int offset); + int (*fw_parse_program_data)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*fw_parse_configuration_data)(struct tasdevice_priv *tas_priv, + struct tasdevice_fw *tas_fmw, + const struct firmware *fmw, int offset); + int (*tasdevice_load_block)(struct tasdevice_priv *tas_priv, + struct tasdev_blk *block); +}; + +void tas2781_reset(struct tasdevice_priv *tas_dev); +int tascodec_init(struct tasdevice_priv *tas_priv, void *codec, + void (*cont)(const struct firmware *fw, void *context)); +struct tasdevice_priv *tasdevice_kzalloc(struct i2c_client *i2c); +int tasdevice_init(struct tasdevice_priv *tas_priv); +void tasdevice_remove(struct tasdevice_priv *tas_priv); +int tasdevice_dev_read(struct tasdevice_priv *tas_priv, + enum channel chn, unsigned int reg, unsigned int *value); +int tasdevice_dev_write(struct tasdevice_priv *tas_priv, + enum channel chn, unsigned int reg, unsigned int value); +int tasdevice_dev_bulk_write( + struct tasdevice_priv *tas_priv, enum channel chn, + unsigned int reg, unsigned char *p_data, unsigned int n_length); +int tasdevice_dev_bulk_read(struct tasdevice_priv *tas_priv, + enum channel chn, unsigned int reg, unsigned char *p_data, + unsigned int n_length); +int tasdevice_dev_update_bits( + struct tasdevice_priv *tasdevice, enum channel chn, + unsigned int reg, unsigned int mask, unsigned int value); +int tasdevice_amp_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +int tasdevice_amp_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +int tasdevice_digital_putvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); +int tasdevice_digital_getvol(struct tasdevice_priv *tas_priv, + struct snd_ctl_elem_value *ucontrol, struct soc_mixer_control *mc); + +#endif /* __TAS2781_H__ */ -- 2.34.1

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[PATCH 0/3] ASoC: cs35l56: Bugfixes and efficiency improvement
by Richard Fitzgerald 19 May '23

19 May '23

First two patches are bugfixes. Third patch skips the overhead of rebooting the amp after applying firmware files when we know that it isn't necessary. Simon Trimmer (3): ASoC: cs35l56: Move DSP part string generation so that it is done only once ASoC: cs35l56: sdw_write_no_pm() should be performed under a pm_runtime request ASoC: cs35l56: In secure mode skip SHUTDOWN and RESET around fw download include/sound/cs35l56.h | 1 + sound/soc/codecs/cs35l56.c | 65 +++++++++++++++++++++++++++----------- 2 files changed, 47 insertions(+), 19 deletions(-) -- 2.30.2

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[PATCH AUTOSEL 6.3 01/24] ASoC: jack: allow multiple interrupt per gpio
by Sasha Levin 19 May '23

19 May '23

From: Svyatoslav Ryhel <clamor95(a)gmail.com> [ Upstream commit a2d4051b0bd6dffcd736888ae89a550d6f60b060 ] This feature is required for coupled hp-mic quirk used by some Nvidia Tegra 3 based devices work properly. Signed-off-by: Svyatoslav Ryhel <clamor95(a)gmail.com> Link: https://lore.kernel.org/r/20230221183211.21964-3-clamor95@gmail.com Signed-off-by: Mark Brown <broonie(a)kernel.org> Signed-off-by: Sasha Levin <sashal(a)kernel.org> --- sound/soc/soc-jack.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/soc-jack.c b/sound/soc/soc-jack.c index fcece5ca38c69..f951acb2ce362 100644 --- a/sound/soc/soc-jack.c +++ b/sound/soc/soc-jack.c @@ -367,6 +367,7 @@ int snd_soc_jack_add_gpios(struct snd_soc_jack *jack, int count, ret = request_any_context_irq(gpiod_to_irq(gpios[i].desc), gpio_handler, + IRQF_SHARED | IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING, gpios[i].name, -- 2.39.2

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[PATCH] ASoC: soc-pcm: test if a BE can be prepared
by Pierre-Louis Bossart 19 May '23

19 May '23

From: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> In the BE hw_params configuration, the existing code checks if any of the existing FEs are prepared, running, paused or suspended - and skips the configuration in those cases. This allows multiple calls of hw_params which the ALSA state machine supports. This check is not handled for the prepare stage, which can lead to the same BE being prepared multiple times. This patch adds a check similar to that of the hw_params, with the main difference being that the suspended state is allowed: the ALSA state machine allows a transition from suspended to prepared with hw_params skipped. This problem was detected on Intel IPC4/SoundWire devices, where the BE dailink .prepare stage is used to configure the SoundWire stream with a bank switch. Multiple .prepare calls lead to conflicts with the .trigger operation with IPC4 configurations. This problem was not detected earlier on Intel devices, HDaudio BE dailinks detect that the link is already prepared and skip the configuration, and for IPC3 devices there is no BE trigger. Link: https://github.com/thesofproject/sof/issues/7596 Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- include/sound/soc-dpcm.h | 4 ++++ sound/soc/soc-pcm.c | 20 ++++++++++++++++++++ 2 files changed, 24 insertions(+) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 4d6ac7699833..ebd24753dd00 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -122,6 +122,10 @@ int snd_soc_dpcm_can_be_free_stop(struct snd_soc_pcm_runtime *fe, int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, struct snd_soc_pcm_runtime *be, int stream); +/* can this BE perform prepare */ +int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream); + /* is the current PCM operation for this FE ? */ int snd_soc_dpcm_fe_can_update(struct snd_soc_pcm_runtime *fe, int stream); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index adb69d7820a8..4fb1ac8e1c4a 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2405,6 +2405,9 @@ int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) if (!snd_soc_dpcm_be_can_update(fe, be, stream)) continue; + if (!snd_soc_dpcm_can_be_prepared(fe, be, stream)) + continue; + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_HW_PARAMS) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && (be->dpcm[stream].state != SND_SOC_DPCM_STATE_SUSPEND) && @@ -3042,3 +3045,20 @@ int snd_soc_dpcm_can_be_params(struct snd_soc_pcm_runtime *fe, return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); } EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_params); + +/* + * We can only prepare a BE DAI if any of it's FE are not prepared, + * running or paused for the specified stream direction. + */ +int snd_soc_dpcm_can_be_prepared(struct snd_soc_pcm_runtime *fe, + struct snd_soc_pcm_runtime *be, int stream) +{ + const enum snd_soc_dpcm_state state[] = { + SND_SOC_DPCM_STATE_START, + SND_SOC_DPCM_STATE_PAUSED, + SND_SOC_DPCM_STATE_PREPARE, + }; + + return snd_soc_dpcm_check_state(fe, be, stream, state, ARRAY_SIZE(state)); +} +EXPORT_SYMBOL_GPL(snd_soc_dpcm_can_be_prepared); -- 2.37.2

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[PATCH v2 0/7] Add Chameleon v3 ASoC audio
by Paweł Anikiel 19 May '23

19 May '23

The Google Chameleon v3 is a device made for testing audio and video paths of other devices. This patchset adds support for ASoC audio on this device. It has two audio sources: HDMI audio from the it68051 chip (RX only), and analog audio from the ssm2603 chip (RX and TX). The patchset adds the ASoC platform and codec drivers, as well as some changes to the existing ssm2602 codec driver. v2 changes: * Replace chv3-audio machine driver with simple-audio-card * Use regmap_register_patch() in ssm2602 workaround * Remove ssm2602 mute gpio patch, add a gpio hog instead * Simplify hw_pointer logic in chv3-i2s * Commit splits & other misc changes Paweł Anikiel (7): ASoC: Add Google Chameleon v3 i2s driver ASoC: Add Google Chameleon v3 codec driver ASoC: dt-bindings: Add Google Chameleon v3 i2s device ASoC: dt-bindings: Add Google Chameleon v3 audio codec ARM: dts: chameleonv3: Add mute gpio hog ARM: dts: chameleonv3: Add Chameleon v3 audio ASoC: ssm2602: Add workaround for playback distortions .../bindings/sound/google,chv3-codec.yaml | 31 ++ .../bindings/sound/google,chv3-i2s.yaml | 44 +++ .../boot/dts/socfpga_arria10_chameleonv3.dts | 62 ++++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 8 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/chv3-codec.c | 41 +++ sound/soc/codecs/ssm2602.c | 15 + sound/soc/google/Kconfig | 6 + sound/soc/google/Makefile | 2 + sound/soc/google/chv3-i2s.c | 338 ++++++++++++++++++ 12 files changed, 551 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/google,chv3-codec.yaml create mode 100644 Documentation/devicetree/bindings/sound/google,chv3-i2s.yaml create mode 100644 sound/soc/codecs/chv3-codec.c create mode 100644 sound/soc/google/Kconfig create mode 100644 sound/soc/google/Makefile create mode 100644 sound/soc/google/chv3-i2s.c -- 2.40.1.521.gf1e218fcd8-goog

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[PATCH] SoC: rt5682: Disable jack detection interrupt during suspend
by Matthias Kaehlcke 18 May '23

18 May '23

The rt5682 driver switches its regmap to cache-only when the device suspends and back to regular mode on resume. When the jack detect interrupt fires rt5682_irq() schedules the jack detect work. This can result in invalid reads from the regmap in cache-only mode if the work runs before the device has resumed: [ 56.245502] rt5682 9-001a: ASoC: error at soc_component_read_no_lock on rt5682.9-001a for register: [0x000000f0] -16 Disable the jack detection interrupt during suspend and re-enable it on resume. The driver already schedules the jack detection work on resume, so any state change during suspend is still handled. This is essentially the same as commit f7d00a9be147 ("SoC: rt5682s: Disable jack detection interrupt during suspend") for the rt5682s. Cc: stable(a)kernel.org Signed-off-by: Matthias Kaehlcke <mka(a)chromium.org> --- sound/soc/codecs/rt5682-i2c.c | 4 +++- sound/soc/codecs/rt5682.c | 6 ++++++ sound/soc/codecs/rt5682.h | 1 + 3 files changed, 10 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5682-i2c.c b/sound/soc/codecs/rt5682-i2c.c index 2935c1bb81f3..5bc46b041786 100644 --- a/sound/soc/codecs/rt5682-i2c.c +++ b/sound/soc/codecs/rt5682-i2c.c @@ -267,7 +267,9 @@ static int rt5682_i2c_probe(struct i2c_client *i2c) ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL, rt5682_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING | IRQF_ONESHOT, "rt5682", rt5682); - if (ret) + if (!ret) + rt5682->irq = i2c->irq; + else dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); } diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index f6c798b65c08..5d992543b791 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2959,6 +2959,9 @@ static int rt5682_suspend(struct snd_soc_component *component) if (rt5682->is_sdw) return 0; + if (rt5682->irq) + disable_irq(rt5682->irq); + cancel_delayed_work_sync(&rt5682->jack_detect_work); cancel_delayed_work_sync(&rt5682->jd_check_work); if (rt5682->hs_jack && (rt5682->jack_type & SND_JACK_HEADSET) == SND_JACK_HEADSET) { @@ -3027,6 +3030,9 @@ static int rt5682_resume(struct snd_soc_component *component) mod_delayed_work(system_power_efficient_wq, &rt5682->jack_detect_work, msecs_to_jiffies(0)); + if (rt5682->irq) + enable_irq(rt5682->irq); + return 0; } #else diff --git a/sound/soc/codecs/rt5682.h b/sound/soc/codecs/rt5682.h index d568c6993c33..e8efd8a84a6c 100644 --- a/sound/soc/codecs/rt5682.h +++ b/sound/soc/codecs/rt5682.h @@ -1462,6 +1462,7 @@ struct rt5682_priv { int pll_out[RT5682_PLLS]; int jack_type; + int irq; int irq_work_delay_time; }; -- 2.40.1.606.ga4b1b128d6-goog

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[PATCH v2] ASoC:codecs: lpass: Fix for KASAN use_after_free out of bounds
by Ravulapati Vishnu Vardhan Rao 18 May '23

18 May '23

When we run syzkaller we get below Out of Bounds error. "KASAN: slab-out-of-bounds Read in regcache_flat_read" Below is the backtrace of the issue: BUG: KASAN: slab-out-of-bounds in regcache_flat_read+0x10c/0x110 Read of size 4 at addr ffffff8088fbf714 by task syz-executor.4/14144 CPU: 6 PID: 14144 Comm: syz-executor.4 Tainted: G W Hardware name: Qualcomm Technologies, Inc. sc7280 CRD platform (rev5+) (DT) Call trace: dump_backtrace+0x0/0x4ec show_stack+0x34/0x50 dump_stack_lvl+0xdc/0x11c print_address_description+0x30/0x2d8 kasan_report+0x178/0x1e4 __asan_report_load4_noabort+0x44/0x50 regcache_flat_read+0x10c/0x110 regcache_read+0xf8/0x5a0 _regmap_read+0x45c/0x86c _regmap_update_bits+0x128/0x290 regmap_update_bits_base+0xc0/0x15c snd_soc_component_update_bits+0xa8/0x22c snd_soc_component_write_field+0x68/0xd4 tx_macro_put_dec_enum+0x1d0/0x268 snd_ctl_elem_write+0x288/0x474 By Error checking and checking valid values issue gets rectifies. Signed-off-by: Ravulapati Vishnu Vardhan Rao <quic_visr(a)quicinc.com> --- sound/soc/codecs/lpass-tx-macro.c | 5 +++++ 1 file changed, 5 insertions(+) Changes Since v1: - Check value with enum item value getting from control. - Removing of previously added code which is unwanted as already we get out with -EINVAL if value is out of bound. diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index da6fcf7f0991..de978c3d70b7 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -746,6 +746,8 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol, struct tx_macro *tx = snd_soc_component_get_drvdata(component); val = ucontrol->value.enumerated.item[0]; + if (val >= e->items) + return -EINVAL; switch (e->reg) { case CDC_TX_INP_MUX_ADC_MUX0_CFG0: @@ -772,6 +774,9 @@ static int tx_macro_put_dec_enum(struct snd_kcontrol *kcontrol, case CDC_TX_INP_MUX_ADC_MUX7_CFG0: mic_sel_reg = CDC_TX7_TX_PATH_CFG0; break; + default: + dev_err(component->dev, "Error in configuration!!\n"); + return -EINVAL; } if (val != 0) { -- 2.17.1

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[PATCH] ASoC: mediatek: mt8192-mt6359: Remove " Jack" from Headphone pin name
by AngeloGioacchino Del Regno 18 May '23

18 May '23

Function jack_kctl_name_gen() will remove the redundant " Jack" from the name, if present, and then it will add it back, so that all of the controls are named "(pin-name) Jack". Remove " Jack" from the Headphone pin name to spare some CPU cycles. This commit brings no functional changes. Signed-off-by: AngeloGioacchino Del Regno <angelogioacchino.delregno(a)collabora.com> --- sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c index 5e163e23a207..4e0d5bf12b47 100644 --- a/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c +++ b/sound/soc/mediatek/mt8192/mt8192-mt6359-rt1015-rt5682.c @@ -46,7 +46,7 @@ struct mt8192_mt6359_priv { /* Headset jack detection DAPM pins */ static struct snd_soc_jack_pin mt8192_jack_pins[] = { { - .pin = "Headphone Jack", + .pin = "Headphone", .mask = SND_JACK_HEADPHONE, }, { -- 2.40.1

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[PATCH 01/12] ASoC: codecs: rt1308: do not store status in state container
by Krzysztof Kozlowski 18 May '23

18 May '23

Driver in its update status callback stores Soundwire device status in state container but it never uses it later. Simplify the code a bit. Signed-off-by: Krzysztof Kozlowski <krzysztof.kozlowski(a)linaro.org> --- sound/soc/codecs/rt1308-sdw.c | 5 +---- sound/soc/codecs/rt1308-sdw.h | 1 - 2 files changed, 1 insertion(+), 5 deletions(-) diff --git a/sound/soc/codecs/rt1308-sdw.c b/sound/soc/codecs/rt1308-sdw.c index 1797af824f60..313e97c94532 100644 --- a/sound/soc/codecs/rt1308-sdw.c +++ b/sound/soc/codecs/rt1308-sdw.c @@ -304,9 +304,6 @@ static int rt1308_update_status(struct sdw_slave *slave, { struct rt1308_sdw_priv *rt1308 = dev_get_drvdata(&slave->dev); - /* Update the status */ - rt1308->status = status; - if (status == SDW_SLAVE_UNATTACHED) rt1308->hw_init = false; @@ -314,7 +311,7 @@ static int rt1308_update_status(struct sdw_slave *slave, * Perform initialization only if slave status is present and * hw_init flag is false */ - if (rt1308->hw_init || rt1308->status != SDW_SLAVE_ATTACHED) + if (rt1308->hw_init || status != SDW_SLAVE_ATTACHED) return 0; /* perform I/O transfers required for Slave initialization */ diff --git a/sound/soc/codecs/rt1308-sdw.h b/sound/soc/codecs/rt1308-sdw.h index 04ff18fa18e2..f816c73e247e 100644 --- a/sound/soc/codecs/rt1308-sdw.h +++ b/sound/soc/codecs/rt1308-sdw.h @@ -159,7 +159,6 @@ struct rt1308_sdw_priv { struct snd_soc_component *component; struct regmap *regmap; struct sdw_slave *sdw_slave; - enum sdw_slave_status status; struct sdw_bus_params params; bool hw_init; bool first_hw_init; -- 2.34.1

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[PATCH 0/2] ALSA: emu10k1: various improvements to the DSP-based mixer code
by Oswald Buddenhagen 18 May '23

18 May '23

Oswald Buddenhagen (2): ALSA: emu10k1: enable bit-exact playback, part 3: pitch ALSA: emu10k1: enable bit-exact playback, part 4: send amounts include/sound/emu10k1.h | 2 ++ sound/pci/emu10k1/emupcm.c | 34 ++++++++++++++++++++++++++++++---- 2 files changed, 32 insertions(+), 4 deletions(-) -- 2.40.0.152.g15d061e6df

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