diff options
| author | Linus Torvalds <torvalds@linux-foundation.org> | 2025-01-24 07:54:34 -0800 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2025-01-24 07:54:34 -0800 |
| commit | 2c8d2a510c15c003749e43ac2b8e1bc79a7a00d6 (patch) | |
| tree | 932d3f6b826a8093dc8dc2bcfd476fa73f9c2dcc /include/sound | |
| parent | 454cb97726fe62a04b187a0d631ec0a69f6b713a (diff) | |
| parent | 6aa96f780204bfdac225eb4c8f51f86c38cc1a26 (diff) | |
Merge tag 'sound-6.14-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This was a relatively calm cycle, and most of changes are rather small
device-specific fixes. Here are highlights:
Core:
- Further enhancements of ALSA rawmidi and sequencer APIs for MIDI
2.0
- compress-offload API extensions for ASRC support
ASoC:
- Allow clocking on each DAI in an audio graph card to be configured
separately
- Improved power management for Renesas RZ-SSI
- KUnit testing for the Cirrus DSP framework
- Memory to meory operation support for Freescale/NXP platforms
- Support for pause operations in SOF
- Support for Allwinner suinv F1C100s, Awinc AW88083, Realtek
ALC5682I-VE
HD- and USB-audio:
- Add support for Focusrite Scarlett 4th Gen 16i16, 18i16, and 18i20
interfaces via new FCP driver
- TAS2781 SPI HD-audio sub-codec support
- Various device-specific quirks as usual"
* tag 'sound-6.14-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (235 commits)
ALSA: hda: tas2781-spi: Fix bogus error handling in tas2781_hda_spi_probe()
ALSA: hda: tas2781-spi: Fix error code in tas2781_read_acpi()
ALSA: hda: tas2781-spi: Delete some dead code
ALSA: usb: fcp: Fix return code from poll ops
ALSA: usb: fcp: Fix incorrect resp->opcode retrieval
ALSA: usb: fcp: Fix meter_levels type to __le32
ALSA: hda/realtek: Enable Mute LED on HP Laptop 14s-fq1xxx
ALSA: hda: tas2781-spi: Fix -Wsometimes-uninitialized in tasdevice_spi_switch_book()
ALSA: ctxfi: Simplify dao_clear_{left,right}_input() functions
ALSA: hda: tas2781-spi: select CRC32 instead of CRC32_SARWATE
ALSA: usb: fcp: Fix hwdep read ops types
ALSA: scarlett2: Add device_setup option to use FCP driver
ALSA: FCP: Add Focusrite Control Protocol driver
ALSA: hda/tas2781: Add tas2781 hda SPI driver
ALSA: hda/realtek - Fixed headphone distorted sound on Acer Aspire A115-31 laptop
ASoC: xilinx: xlnx_spdif: Simpify using devm_clk_get_enabled()
ALSA: hda: Support for Ideapad hotkey mute LEDs
ASoC: Intel: sof_sdw: Fix DMI match for Lenovo 83JX, 83MC and 83NM
ASoC: Intel: sof_sdw: Fix DMI match for Lenovo 83LC
ASoC: dapm: add support for preparing streams
...
Diffstat (limited to 'include/sound')
| -rw-r--r-- | include/sound/hdaudio_ext.h | 45 | ||||
| -rw-r--r-- | include/sound/pcm.h | 7 | ||||
| -rw-r--r-- | include/sound/rawmidi.h | 11 | ||||
| -rw-r--r-- | include/sound/sdca.h | 7 | ||||
| -rw-r--r-- | include/sound/sdca_function.h | 3 | ||||
| -rw-r--r-- | include/sound/simple_card_utils.h | 15 | ||||
| -rw-r--r-- | include/sound/soc-dai.h | 3 | ||||
| -rw-r--r-- | include/sound/soc.h | 12 | ||||
| -rw-r--r-- | include/sound/soc_sdw_utils.h | 2 | ||||
| -rw-r--r-- | include/sound/ump.h | 1 |
10 files changed, 53 insertions, 53 deletions
diff --git a/include/sound/hdaudio_ext.h b/include/sound/hdaudio_ext.h index 957295364a5e..4c7a40e149a5 100644 --- a/include/sound/hdaudio_ext.h +++ b/include/sound/hdaudio_ext.h @@ -2,8 +2,6 @@ #ifndef __SOUND_HDAUDIO_EXT_H #define __SOUND_HDAUDIO_EXT_H -#include <linux/io-64-nonatomic-lo-hi.h> -#include <linux/iopoll.h> #include <sound/hdaudio.h> int snd_hdac_ext_bus_init(struct hdac_bus *bus, struct device *dev, @@ -119,49 +117,6 @@ int snd_hdac_ext_bus_link_put(struct hdac_bus *bus, struct hdac_ext_link *hlink) void snd_hdac_ext_bus_link_power(struct hdac_device *codec, bool enable); -#define snd_hdac_adsp_writeb(chip, reg, value) \ - snd_hdac_reg_writeb(chip, (chip)->dsp_ba + (reg), value) -#define snd_hdac_adsp_readb(chip, reg) \ - snd_hdac_reg_readb(chip, (chip)->dsp_ba + (reg)) -#define snd_hdac_adsp_writew(chip, reg, value) \ - snd_hdac_reg_writew(chip, (chip)->dsp_ba + (reg), value) -#define snd_hdac_adsp_readw(chip, reg) \ - snd_hdac_reg_readw(chip, (chip)->dsp_ba + (reg)) -#define snd_hdac_adsp_writel(chip, reg, value) \ - snd_hdac_reg_writel(chip, (chip)->dsp_ba + (reg), value) -#define snd_hdac_adsp_readl(chip, reg) \ - snd_hdac_reg_readl(chip, (chip)->dsp_ba + (reg)) -#define snd_hdac_adsp_writeq(chip, reg, value) \ - snd_hdac_reg_writeq(chip, (chip)->dsp_ba + (reg), value) -#define snd_hdac_adsp_readq(chip, reg) \ - snd_hdac_reg_readq(chip, (chip)->dsp_ba + (reg)) - -#define snd_hdac_adsp_updateb(chip, reg, mask, val) \ - snd_hdac_adsp_writeb(chip, reg, \ - (snd_hdac_adsp_readb(chip, reg) & ~(mask)) | (val)) -#define snd_hdac_adsp_updatew(chip, reg, mask, val) \ - snd_hdac_adsp_writew(chip, reg, \ - (snd_hdac_adsp_readw(chip, reg) & ~(mask)) | (val)) -#define snd_hdac_adsp_updatel(chip, reg, mask, val) \ - snd_hdac_adsp_writel(chip, reg, \ - (snd_hdac_adsp_readl(chip, reg) & ~(mask)) | (val)) -#define snd_hdac_adsp_updateq(chip, reg, mask, val) \ - snd_hdac_adsp_writeq(chip, reg, \ - (snd_hdac_adsp_readq(chip, reg) & ~(mask)) | (val)) - -#define snd_hdac_adsp_readb_poll(chip, reg, val, cond, delay_us, timeout_us) \ - readb_poll_timeout((chip)->dsp_ba + (reg), val, cond, \ - delay_us, timeout_us) -#define snd_hdac_adsp_readw_poll(chip, reg, val, cond, delay_us, timeout_us) \ - readw_poll_timeout((chip)->dsp_ba + (reg), val, cond, \ - delay_us, timeout_us) -#define snd_hdac_adsp_readl_poll(chip, reg, val, cond, delay_us, timeout_us) \ - readl_poll_timeout((chip)->dsp_ba + (reg), val, cond, \ - delay_us, timeout_us) -#define snd_hdac_adsp_readq_poll(chip, reg, val, cond, delay_us, timeout_us) \ - readq_poll_timeout((chip)->dsp_ba + (reg), val, cond, \ - delay_us, timeout_us) - struct hdac_ext_device; /* ops common to all codec drivers */ diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 67c99ffbf51b..8becb4504887 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -1532,9 +1532,10 @@ static inline u64 pcm_format_to_bits(snd_pcm_format_t pcm_format) dev_dbg((pcm)->card->dev, fmt, ##args) /* helpers for copying between iov_iter and iomem */ -int copy_to_iter_fromio(struct iov_iter *itert, const void __iomem *src, - size_t count); -int copy_from_iter_toio(void __iomem *dst, struct iov_iter *iter, size_t count); +size_t copy_to_iter_fromio(const void __iomem *src, size_t bytes, + struct iov_iter *iter) __must_check; +size_t copy_from_iter_toio(void __iomem *dst, size_t bytes, + struct iov_iter *iter) __must_check; struct snd_pcm_status64 { snd_pcm_state_t state; /* stream state */ diff --git a/include/sound/rawmidi.h b/include/sound/rawmidi.h index f31cabf0158c..6916f7133597 100644 --- a/include/sound/rawmidi.h +++ b/include/sound/rawmidi.h @@ -89,6 +89,7 @@ struct snd_rawmidi_substream { unsigned int framing; /* whether to frame input data */ unsigned int clock_type; /* clock source to use for input framing */ int use_count; /* use counter (for output) */ + bool inactive; /* inactive substream (for UMP legacy) */ size_t bytes; spinlock_t lock; struct snd_rawmidi *rmidi; @@ -118,6 +119,7 @@ struct snd_rawmidi { struct list_head list; unsigned int device; /* device number */ unsigned int info_flags; /* SNDRV_RAWMIDI_INFO_XXXX */ + unsigned int tied_device; char id[64]; char name[80]; @@ -189,4 +191,13 @@ long snd_rawmidi_kernel_read(struct snd_rawmidi_substream *substream, long snd_rawmidi_kernel_write(struct snd_rawmidi_substream *substream, const unsigned char *buf, long count); +/* set up the tied devices */ +static inline void snd_rawmidi_tie_devices(struct snd_rawmidi *r1, + struct snd_rawmidi *r2) +{ + /* tied_device field keeps the device+1 (so that 0 being unknown) */ + r1->tied_device = r2->device + 1; + r2->tied_device = r1->device + 1; +} + #endif /* __SOUND_RAWMIDI_H */ diff --git a/include/sound/sdca.h b/include/sound/sdca.h index 7e138229e8f3..973252d0adac 100644 --- a/include/sound/sdca.h +++ b/include/sound/sdca.h @@ -9,6 +9,9 @@ #ifndef __SDCA_H__ #define __SDCA_H__ +#include <linux/types.h> +#include <linux/kconfig.h> + struct sdw_slave; #define SDCA_MAX_FUNCTION_COUNT 8 @@ -20,9 +23,9 @@ struct sdw_slave; * @name: human-readable string */ struct sdca_function_desc { - u64 adr; - u32 type; const char *name; + u32 type; + u8 adr; }; /** diff --git a/include/sound/sdca_function.h b/include/sound/sdca_function.h index 9dc5bfec07e5..c051c17903e8 100644 --- a/include/sound/sdca_function.h +++ b/include/sound/sdca_function.h @@ -9,6 +9,8 @@ #ifndef __SDCA_FUNCTION_H__ #define __SDCA_FUNCTION_H__ +#include <linux/bits.h> + /* * SDCA Function Types from SDCA specification v1.0a Section 5.1.2 * all Function types not described are reserved @@ -40,6 +42,7 @@ enum sdca_function_type { #define SDCA_FUNCTION_TYPE_RJ_NAME "RJ" #define SDCA_FUNCTION_TYPE_SIMPLE_NAME "SimpleJack" #define SDCA_FUNCTION_TYPE_HID_NAME "HID" +#define SDCA_FUNCTION_TYPE_IMP_DEF_NAME "ImplementationDefined" enum sdca_entity0_controls { SDCA_CTL_ENTITY_0_COMMIT_GROUP_MASK = 0x01, diff --git a/include/sound/simple_card_utils.h b/include/sound/simple_card_utils.h index 3360d9eab068..892f70532363 100644 --- a/include/sound/simple_card_utils.h +++ b/include/sound/simple_card_utils.h @@ -89,6 +89,13 @@ struct simple_util_priv { #define simple_props_to_dai_codec(props, i) ((props)->codec_dai + i) #define simple_props_to_codec_conf(props, i) ((props)->codec_conf + i) +/* has the same effect as simple_priv_to_props(). Preferred over + * simple_priv_to_props() when dealing with PCM runtime data as + * the ID stored in rtd->id may not be a valid array index. + */ +#define runtime_simple_priv_to_props(priv, rtd) \ + ((priv)->dai_props + ((rtd)->dai_link - (priv)->dai_link)) + #define for_each_prop_dlc_cpus(props, i, cpu) \ for ((i) = 0; \ ((i) < (props)->num.cpus) && \ @@ -264,9 +271,13 @@ static inline void simple_util_debug_info(struct simple_util_priv *priv) simple_util_debug_dai(priv, "codec", dai); if (link->name) - dev_dbg(dev, "dai name = %s\n", link->name); + dev_dbg(dev, "link name = %s\n", link->name); if (link->dai_fmt) - dev_dbg(dev, "dai format = %04x\n", link->dai_fmt); + dev_dbg(dev, "link format = %04x\n", link->dai_fmt); + if (link->playback_only) + dev_dbg(dev, "link has playback_only"); + if (link->capture_only) + dev_dbg(dev, "link has capture_only"); if (props->adata.convert_rate) dev_dbg(dev, "convert_rate = %d\n", props->adata.convert_rate); if (props->adata.convert_channels) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index aab57c19f62b..a11501752637 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -193,6 +193,9 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); +int snd_soc_dai_prepare(struct snd_soc_dai *dai, + struct snd_pcm_substream *substream); + /* Digital Audio Interface mute */ int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute, int direction); diff --git a/include/sound/soc.h b/include/sound/soc.h index 4f5d411e3823..fcdb5adfcd5e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -681,6 +681,17 @@ struct snd_soc_dai_link_component { struct device_node *of_node; const char *dai_name; const struct of_phandle_args *dai_args; + + /* + * Extra format = SND_SOC_DAIFMT_Bx_Fx + * + * [Note] it is Bx_Fx base, not CBx_CFx + * + * It will be used with dai_link->dai_fmt + * see + * snd_soc_runtime_set_dai_fmt() + */ + unsigned int ext_fmt; }; /* @@ -1118,7 +1129,6 @@ struct snd_soc_card { unsigned int instantiated:1; unsigned int topology_shortname_created:1; unsigned int fully_routed:1; - unsigned int disable_route_checks:1; unsigned int probed:1; unsigned int component_chaining:1; diff --git a/include/sound/soc_sdw_utils.h b/include/sound/soc_sdw_utils.h index 0e82598e10af..36a4a1e1d8ca 100644 --- a/include/sound/soc_sdw_utils.h +++ b/include/sound/soc_sdw_utils.h @@ -224,6 +224,8 @@ int asoc_sdw_cs_amp_init(struct snd_soc_card *card, struct snd_soc_dai_link *dai_links, struct asoc_sdw_codec_info *info, bool playback); +int asoc_sdw_cs_spk_feedback_rtd_init(struct snd_soc_pcm_runtime *rtd, + struct snd_soc_dai *dai); /* MAXIM codec support */ int asoc_sdw_maxim_init(struct snd_soc_card *card, diff --git a/include/sound/ump.h b/include/sound/ump.h index 532c2c3ea28e..73f97f88e2ed 100644 --- a/include/sound/ump.h +++ b/include/sound/ump.h @@ -83,6 +83,7 @@ struct snd_ump_ops { struct snd_seq_ump_ops { void (*input_receive)(struct snd_ump_endpoint *ump, const u32 *data, int words); + int (*notify_ep_change)(struct snd_ump_endpoint *ump); int (*notify_fb_change)(struct snd_ump_endpoint *ump, struct snd_ump_block *fb); int (*switch_protocol)(struct snd_ump_endpoint *ump); |
