diff options
author | Linus Torvalds | 2024-03-14 11:10:43 -0700 |
---|---|---|
committer | Linus Torvalds | 2024-03-14 11:10:43 -0700 |
commit | fe46a7dd189e25604716c03576d05ac8a5209743 (patch) | |
tree | de7572a9f41bb91e570dce1053283e56d1efdd94 /include/sound | |
parent | 705c1da8fa4816fb0159b5602fef1df5946a3ee2 (diff) | |
parent | a39d51ff1f52cd0b6fe7d379ac93bd8b4237d1b7 (diff) |
Merge tag 'sound-6.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This was a relatively calm development cycle. Most of changes are
rather small device-specific fixes and enhancements. The only
significant changes in ALSA core are code refactoring with the recent
cleanup infrastructure, which should bring no functionality changes.
Some highlights below:
Core:
- Lots of cleanups in ALSA core code with automatic kfree cleanup and
locking guard macros
- New ALSA core kunit test
ASoC:
- SoundWire support for AMD ACP 6.3 systems
- Support for reporting version information for AVS firmware
- Support DSPless mode for Intel Soundwire systems
- Support for configuring CS35L56 amplifiers using EFI calibration
data
- Log which component is being operated on as part of power
management trace events.
- Support for Microchip SAM9x7, NXP i.MX95 and Qualcomm WCD939x
HD- and USB-audio:
- More Cirrus HD-audio codec support
- TAS2781 HD-audio codec fixes
- Scarlett2 mixer fixes
Others:
- Enhancement of virtio driver for audio control supports
- Cleanups of legacy PM code with new macros
- Firewire sound updates"
* tag 'sound-6.9-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (307 commits)
ALSA: usb-audio: Stop parsing channels bits when all channels are found.
ALSA: hda/tas2781: remove unnecessary runtime_pm calls
ALSA: hda/realtek - ALC236 fix volume mute & mic mute LED on some HP models
ALSA: aaci: Delete unused variable in aaci_do_suspend
ALSA: scarlett2: Fix Scarlett 4th Gen input gain range again
ALSA: scarlett2: Fix Scarlett 4th Gen input gain range
ALSA: scarlett2: Fix Scarlett 4th Gen autogain status values
ALSA: scarlett2: Fix Scarlett 4th Gen 4i4 low-voltage detection
ALSA: hda/tas2781: restore power state after system_resume
ALSA: hda/tas2781: do not call pm_runtime_force_* in system_resume/suspend
ALSA: hda/tas2781: do not reset cur_* values in runtime_suspend
ALSA: hda/tas2781: add lock to system_suspend
ALSA: hda/tas2781: use dev_dbg in system_resume
ALSA: hda/realtek: fix ALC285 issues on HP Envy x360 laptops
platform/x86: serial-multi-instantiate: Add support for CS35L54 and CS35L57
ALSA: hda: cs35l56: Add support for CS35L54 and CS35L57
ASoC: cs35l56: Add support for CS35L54 and CS35L57
ASoC: Intel: catpt: Carefully use PCI bitwise constants
ALSA: hda: hda_component: Include sound/hda_codec.h
ALSA: hda: hda_component: Add missing #include guards
...
Diffstat (limited to 'include/sound')
-rw-r--r-- | include/sound/ak4531_codec.h | 3 | ||||
-rw-r--r-- | include/sound/cs-amp-lib.h | 66 | ||||
-rw-r--r-- | include/sound/cs35l56.h | 11 | ||||
-rw-r--r-- | include/sound/cs42l42.h | 5 | ||||
-rw-r--r-- | include/sound/emux_synth.h | 2 | ||||
-rw-r--r-- | include/sound/hda-mlink.h | 2 | ||||
-rw-r--r-- | include/sound/hda_register.h | 2 | ||||
-rw-r--r-- | include/sound/pcm.h | 12 | ||||
-rw-r--r-- | include/sound/sb.h | 3 | ||||
-rw-r--r-- | include/sound/soc.h | 4 | ||||
-rw-r--r-- | include/sound/sof/dai-amd.h | 7 | ||||
-rw-r--r-- | include/sound/sof/dai.h | 2 | ||||
-rw-r--r-- | include/sound/tas2781.h | 1 |
13 files changed, 112 insertions, 8 deletions
diff --git a/include/sound/ak4531_codec.h b/include/sound/ak4531_codec.h index 9a4429970d92..64402347d7a2 100644 --- a/include/sound/ak4531_codec.h +++ b/include/sound/ak4531_codec.h @@ -65,6 +65,9 @@ int snd_ak4531_mixer(struct snd_card *card, struct snd_ak4531 *_ak4531, #ifdef CONFIG_PM void snd_ak4531_suspend(struct snd_ak4531 *ak4531); void snd_ak4531_resume(struct snd_ak4531 *ak4531); +#else +static inline void snd_ak4531_suspend(struct snd_ak4531 *ak4531) {} +static inline void snd_ak4531_resume(struct snd_ak4531 *ak4531) {} #endif #endif /* __SOUND_AK4531_CODEC_H */ diff --git a/include/sound/cs-amp-lib.h b/include/sound/cs-amp-lib.h new file mode 100644 index 000000000000..f481148735e1 --- /dev/null +++ b/include/sound/cs-amp-lib.h @@ -0,0 +1,66 @@ +/* SPDX-License-Identifier: GPL-2.0-only */ +/* + * Copyright (C) 2024 Cirrus Logic, Inc. and + * Cirrus Logic International Semiconductor Ltd. + */ + +#ifndef CS_AMP_LIB_H +#define CS_AMP_LIB_H + +#include <linux/efi.h> +#include <linux/types.h> + +struct cs_dsp; + +struct cirrus_amp_cal_data { + u32 calTarget[2]; + u32 calTime[2]; + s8 calAmbient; + u8 calStatus; + u16 calR; +} __packed; + +struct cirrus_amp_efi_data { + u32 size; + u32 count; + struct cirrus_amp_cal_data data[]; +} __packed; + +/** + * struct cirrus_amp_cal_controls - definition of firmware calibration controls + * @alg_id: ID of algorithm containing the controls. + * @mem_region: DSP memory region containing the controls. + * @ambient: Name of control for calAmbient value. + * @calr: Name of control for calR value. + * @status: Name of control for calStatus value. + * @checksum: Name of control for checksum value. + */ +struct cirrus_amp_cal_controls { + unsigned int alg_id; + int mem_region; + const char *ambient; + const char *calr; + const char *status; + const char *checksum; +}; + +int cs_amp_write_cal_coeffs(struct cs_dsp *dsp, + const struct cirrus_amp_cal_controls *controls, + const struct cirrus_amp_cal_data *data); +int cs_amp_get_efi_calibration_data(struct device *dev, u64 target_uid, int amp_index, + struct cirrus_amp_cal_data *out_data); + +struct cs_amp_test_hooks { + efi_status_t (*get_efi_variable)(efi_char16_t *name, + efi_guid_t *guid, + unsigned long *size, + void *buf); + + int (*write_cal_coeff)(struct cs_dsp *dsp, + const struct cirrus_amp_cal_controls *controls, + const char *ctl_name, u32 val); +}; + +extern const struct cs_amp_test_hooks * const cs_amp_test_hooks; + +#endif /* CS_AMP_LIB_H */ diff --git a/include/sound/cs35l56.h b/include/sound/cs35l56.h index b24716ab2750..e0629699b563 100644 --- a/include/sound/cs35l56.h +++ b/include/sound/cs35l56.h @@ -12,6 +12,7 @@ #include <linux/firmware/cirrus/cs_dsp.h> #include <linux/regulator/consumer.h> #include <linux/regmap.h> +#include <sound/cs-amp-lib.h> #define CS35L56_DEVID 0x0000000 #define CS35L56_REVID 0x0000004 @@ -23,6 +24,9 @@ #define CS35L56_BLOCK_ENABLES2 0x000201C #define CS35L56_REFCLK_INPUT 0x0002C04 #define CS35L56_GLOBAL_SAMPLE_RATE 0x0002C0C +#define CS35L56_OTP_MEM_53 0x00300D4 +#define CS35L56_OTP_MEM_54 0x00300D8 +#define CS35L56_OTP_MEM_55 0x00300DC #define CS35L56_ASP1_ENABLES1 0x0004800 #define CS35L56_ASP1_CONTROL1 0x0004804 #define CS35L56_ASP1_CONTROL2 0x0004808 @@ -257,11 +261,15 @@ struct cs35l56_base { struct regmap *regmap; int irq; struct mutex irq_lock; + u8 type; u8 rev; bool init_done; bool fw_patched; bool secured; bool can_hibernate; + bool cal_data_valid; + s8 cal_index; + struct cirrus_amp_cal_data cal_data; struct gpio_desc *reset_gpio; }; @@ -269,6 +277,8 @@ extern struct regmap_config cs35l56_regmap_i2c; extern struct regmap_config cs35l56_regmap_spi; extern struct regmap_config cs35l56_regmap_sdw; +extern const struct cirrus_amp_cal_controls cs35l56_calibration_controls; + extern const char * const cs35l56_tx_input_texts[CS35L56_NUM_INPUT_SRC]; extern const unsigned int cs35l56_tx_input_values[CS35L56_NUM_INPUT_SRC]; @@ -286,6 +296,7 @@ int cs35l56_is_fw_reload_needed(struct cs35l56_base *cs35l56_base); int cs35l56_runtime_suspend_common(struct cs35l56_base *cs35l56_base); int cs35l56_runtime_resume_common(struct cs35l56_base *cs35l56_base, bool is_soundwire); void cs35l56_init_cs_dsp(struct cs35l56_base *cs35l56_base, struct cs_dsp *cs_dsp); +int cs35l56_get_calibration(struct cs35l56_base *cs35l56_base); int cs35l56_read_prot_status(struct cs35l56_base *cs35l56_base, bool *fw_missing, unsigned int *fw_version); int cs35l56_hw_init(struct cs35l56_base *cs35l56_base); diff --git a/include/sound/cs42l42.h b/include/sound/cs42l42.h index 3994e933db19..1bd8eee54f66 100644 --- a/include/sound/cs42l42.h +++ b/include/sound/cs42l42.h @@ -809,8 +809,7 @@ #define CS42L42_PLL_LOCK_TIMEOUT_US 1250 #define CS42L42_HP_ADC_EN_TIME_US 20000 #define CS42L42_PDN_DONE_POLL_US 1000 -#define CS42L42_PDN_DONE_TIMEOUT_US 200000 -#define CS42L42_PDN_DONE_TIME_MS 100 -#define CS42L42_FILT_DISCHARGE_TIME_MS 46 +#define CS42L42_PDN_DONE_TIMEOUT_US 235000 +#define CS42L42_PDN_DONE_TIME_MS 65 #endif /* __CS42L42_H */ diff --git a/include/sound/emux_synth.h b/include/sound/emux_synth.h index 1cc530434b97..3f7f365ed248 100644 --- a/include/sound/emux_synth.h +++ b/include/sound/emux_synth.h @@ -103,7 +103,7 @@ struct snd_emux { int ports[SNDRV_EMUX_MAX_PORTS]; /* The ports for this device */ struct snd_emux_port *portptrs[SNDRV_EMUX_MAX_PORTS]; int used; /* use counter */ - char *name; /* name of the device (internal) */ + const char *name; /* name of the device (internal) */ struct snd_rawmidi **vmidi; struct timer_list tlist; /* for pending note-offs */ int timer_active; diff --git a/include/sound/hda-mlink.h b/include/sound/hda-mlink.h index 228114aca415..d849d9b24f13 100644 --- a/include/sound/hda-mlink.h +++ b/include/sound/hda-mlink.h @@ -181,4 +181,4 @@ hdac_bus_eml_enable_offload(struct hdac_bus *bus, bool alt, int elid, bool enabl { return 0; } -#endif /* CONFIG_SND_SOC_SOF_HDA */ +#endif /* CONFIG_SND_SOC_SOF_HDA_MLINK */ diff --git a/include/sound/hda_register.h b/include/sound/hda_register.h index 55958711d697..5ff31e6d41c1 100644 --- a/include/sound/hda_register.h +++ b/include/sound/hda_register.h @@ -131,6 +131,8 @@ enum { SDI0, SDI1, SDI2, SDI3, SDO0, SDO1, SDO2, SDO3 }; #define AZX_REG_VS_SDXEFIFOS_XBASE 0x1094 #define AZX_REG_VS_SDXEFIFOS_XINTERVAL 0x20 +#define AZX_REG_VS_LTRP_GB_MASK GENMASK(6, 0) + /* PCI space */ #define AZX_PCIREG_TCSEL 0x44 diff --git a/include/sound/pcm.h b/include/sound/pcm.h index cc175c623dac..210096f124ee 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -659,6 +659,18 @@ void snd_pcm_stream_unlock_irqrestore(struct snd_pcm_substream *substream, flags = _snd_pcm_stream_lock_irqsave_nested(substream); \ } while (0) +/* definitions for guard(); use like guard(pcm_stream_lock) */ +DEFINE_LOCK_GUARD_1(pcm_stream_lock, struct snd_pcm_substream, + snd_pcm_stream_lock(_T->lock), + snd_pcm_stream_unlock(_T->lock)) +DEFINE_LOCK_GUARD_1(pcm_stream_lock_irq, struct snd_pcm_substream, + snd_pcm_stream_lock_irq(_T->lock), + snd_pcm_stream_unlock_irq(_T->lock)) +DEFINE_LOCK_GUARD_1(pcm_stream_lock_irqsave, struct snd_pcm_substream, + snd_pcm_stream_lock_irqsave(_T->lock, _T->flags), + snd_pcm_stream_unlock_irqrestore(_T->lock, _T->flags), + unsigned long flags) + /** * snd_pcm_group_for_each_entry - iterate over the linked substreams * @s: the iterator diff --git a/include/sound/sb.h b/include/sound/sb.h index f540339fb0c7..24970f36c38a 100644 --- a/include/sound/sb.h +++ b/include/sound/sb.h @@ -290,6 +290,9 @@ int snd_sbmixer_new(struct snd_sb *chip); #ifdef CONFIG_PM void snd_sbmixer_suspend(struct snd_sb *chip); void snd_sbmixer_resume(struct snd_sb *chip); +#else +static inline void snd_sbmixer_suspend(struct snd_sb *chip) {} +static inline void snd_sbmixer_resume(struct snd_sb *chip) {} #endif /* sb8_init.c */ diff --git a/include/sound/soc.h b/include/sound/soc.h index 6defc5547ff9..39613b406b1d 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1401,8 +1401,8 @@ void snd_soc_remove_pcm_runtime(struct snd_soc_card *card, void snd_soc_dlc_use_cpu_as_platform(struct snd_soc_dai_link_component *platforms, struct snd_soc_dai_link_component *cpus); struct of_phandle_args *snd_soc_copy_dai_args(struct device *dev, - struct of_phandle_args *args); -struct snd_soc_dai *snd_soc_get_dai_via_args(struct of_phandle_args *dai_args); + const struct of_phandle_args *args); +struct snd_soc_dai *snd_soc_get_dai_via_args(const struct of_phandle_args *dai_args); struct snd_soc_dai *snd_soc_register_dai(struct snd_soc_component *component, struct snd_soc_dai_driver *dai_drv, bool legacy_dai_naming); diff --git a/include/sound/sof/dai-amd.h b/include/sound/sof/dai-amd.h index 9df7ac824efe..59cd014392c1 100644 --- a/include/sound/sof/dai-amd.h +++ b/include/sound/sof/dai-amd.h @@ -26,4 +26,11 @@ struct sof_ipc_dai_acpdmic_params { uint32_t pdm_ch; } __packed; +/* ACP_SDW Configuration Request - SOF_IPC_DAI_AMD_SDW_CONFIG */ +struct sof_ipc_dai_acp_sdw_params { + struct sof_ipc_hdr hdr; + u32 rate; + u32 channels; +} __packed; + #endif diff --git a/include/sound/sof/dai.h b/include/sound/sof/dai.h index 4773a5f616a4..0764a80c17a9 100644 --- a/include/sound/sof/dai.h +++ b/include/sound/sof/dai.h @@ -89,6 +89,7 @@ enum sof_ipc_dai_type { SOF_DAI_AMD_SP_VIRTUAL, /**< AMD ACP SP VIRTUAL */ SOF_DAI_AMD_HS_VIRTUAL, /**< AMD ACP HS VIRTUAL */ SOF_DAI_IMX_MICFIL, /** < i.MX MICFIL PDM */ + SOF_DAI_AMD_SDW, /**< AMD ACP SDW */ }; /* general purpose DAI configuration */ @@ -119,6 +120,7 @@ struct sof_ipc_dai_config { struct sof_ipc_dai_acp_params acphs; struct sof_ipc_dai_mtk_afe_params afe; struct sof_ipc_dai_micfil_params micfil; + struct sof_ipc_dai_acp_sdw_params acp_sdw; }; } __packed; diff --git a/include/sound/tas2781.h b/include/sound/tas2781.h index 9aff384941de..99ca3e401fd1 100644 --- a/include/sound/tas2781.h +++ b/include/sound/tas2781.h @@ -103,7 +103,6 @@ struct tasdevice_priv { struct tm tm; enum device_catlog_id catlog_id; - const char *acpi_subsystem_id; unsigned char cal_binaryname[TASDEVICE_MAX_CHANNELS][64]; unsigned char crc8_lkp_tbl[CRC8_TABLE_SIZE]; unsigned char coef_binaryname[64]; |