diff options
| author | Linus Torvalds <torvalds@linux-foundation.org> | 2021-04-30 12:48:14 -0700 |
|---|---|---|
| committer | Linus Torvalds <torvalds@linux-foundation.org> | 2021-04-30 12:48:14 -0700 |
| commit | b71428d7ab333a157216a1d73c8c82a178efada9 (patch) | |
| tree | 94d268210d84948d5984f2fbe7d890c4aed1fabe /sound/soc/sof/pcm.c | |
| parent | 95275402f66e88c56144a2d859c13594b651b29b (diff) | |
| parent | 2e6a731296be9d356fdccee9fb6ae345dad96438 (diff) | |
| download | linux-b71428d7ab333a157216a1d73c8c82a178efada9.tar.bz2 | |
Merge tag 'sound-5.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"No surprises in this development cycle, and most of work is about the
fixes and the improvements of the existing code, while a new LED
control layer and a few new drivers have been introduced.
Here are some highlights:
Core:
- A common mute-LED framework was introduced. It is used by HD-audio
for now, more adaption will follow later. The former "Mic Mute-LED
Mode" mixer control has been replaced with the corresponding sysfs
now.
- User-control management was changed to count consumed bytes instead
of capping by number of elements; this will allow more controls in
the normal usage pattern while avoiding the possible memory
exhaustion DoS
ASoC:
- Continued refactoring and cleanups in ASoC core and generic card
drivers
- Wide range of small cppcheck and warning fixes
- New drivers for Freescale i.MX DMA over rpmsg, Mediatek MT6358
accessory detection, and Realtek RT1019, RT1316, RT711 and RT715
USB-audio:
- Continued improvements and fixes of the implicit feedback mode,
including better support for Pioneer and Roland/BOSS devices
HD-audio:
- Default back to non-buffer preallocation on x86
- Cirrus codec improvements, more quirks for Realtek codecs
Others:
- New virtio sound driver
- FireWire Bebob updates"
* tag 'sound-5.13-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (587 commits)
ALSA: hda/conexant: Re-order CX5066 quirk table entries
ALSA: hda/realtek: Remove redundant entry for ALC861 Haier/Uniwill devices
ALSA: hda/realtek: Re-order ALC662 quirk table entries
ALSA: hda/realtek: Re-order remaining ALC269 quirk table entries
ALSA: hda/realtek: Re-order ALC269 Lenovo quirk table entries
ALSA: hda/realtek: Re-order ALC269 Sony quirk table entries
ALSA: hda/realtek: Re-order ALC269 ASUS quirk table entries
ALSA: hda/realtek: Re-order ALC269 Dell quirk table entries
ALSA: hda/realtek: Re-order ALC269 Acer quirk table entries
ALSA: hda/realtek: Re-order ALC269 HP quirk table entries
ALSA: hda/realtek: Re-order ALC882 Clevo quirk table entries
ALSA: hda/realtek: Re-order ALC882 Sony quirk table entries
ALSA: hda/realtek: Re-order ALC882 Acer quirk table entries
ALSA: usb-audio: Remove redundant assignment to len
ALSA: hda/realtek: Add quirk for Intel Clevo PCx0Dx
ALSA: virtio: fix kernel-doc
ALSA: hda/cirrus: Use CS8409 filter to fix abnormal sounds on Bullseye
ALSA: hda/cirrus: Set Initial DMIC volume for Bullseye to -26 dB
ALSA: sb: Fix two use after free in snd_sb_qsound_build
ALSA: emu8000: Fix a use after free in snd_emu8000_create_mixer
...
Diffstat (limited to 'sound/soc/sof/pcm.c')
| -rw-r--r-- | sound/soc/sof/pcm.c | 38 |
1 files changed, 34 insertions, 4 deletions
diff --git a/sound/soc/sof/pcm.c b/sound/soc/sof/pcm.c index 61c3fe17342d..9893b182da43 100644 --- a/sound/soc/sof/pcm.c +++ b/sound/soc/sof/pcm.c @@ -619,6 +619,31 @@ capture: return 0; } +static void ssp_dai_config_pcm_params_match(struct snd_sof_dev *sdev, const char *link_name, + struct snd_pcm_hw_params *params) +{ + struct sof_ipc_dai_config *config; + struct snd_sof_dai *dai; + int i; + + /* + * Search for all matching DAIs as we can have both playback and capture DAI + * associated with the same link. + */ + list_for_each_entry(dai, &sdev->dai_list, list) { + if (!dai->name || strcmp(link_name, dai->name)) + continue; + for (i = 0; i < dai->number_configs; i++) { + config = &dai->dai_config[i]; + if (config->ssp.fsync_rate == params_rate(params)) { + dev_dbg(sdev->dev, "DAI config %d matches pcm hw params\n", i); + dai->current_config = i; + break; + } + } + } +} + /* fixup the BE DAI link to match any values from topology */ int sof_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_params *params) { @@ -631,6 +656,7 @@ int sof_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_pa snd_soc_rtdcom_lookup(rtd, SOF_AUDIO_PCM_DRV_NAME); struct snd_sof_dai *dai = snd_sof_find_dai(component, (char *)rtd->dai_link->name); + struct snd_sof_dev *sdev = snd_soc_component_get_drvdata(component); struct snd_soc_dpcm *dpcm; /* no topology exists for this BE, try a common configuration */ @@ -673,10 +699,13 @@ int sof_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_pa /* read rate and channels from topology */ switch (dai->dai_config->type) { case SOF_DAI_INTEL_SSP: - rate->min = dai->dai_config->ssp.fsync_rate; - rate->max = dai->dai_config->ssp.fsync_rate; - channels->min = dai->dai_config->ssp.tdm_slots; - channels->max = dai->dai_config->ssp.tdm_slots; + /* search for config to pcm params match, if not found use default */ + ssp_dai_config_pcm_params_match(sdev, (char *)rtd->dai_link->name, params); + + rate->min = dai->dai_config[dai->current_config].ssp.fsync_rate; + rate->max = dai->dai_config[dai->current_config].ssp.fsync_rate; + channels->min = dai->dai_config[dai->current_config].ssp.tdm_slots; + channels->max = dai->dai_config[dai->current_config].ssp.tdm_slots; dev_dbg(component->dev, "rate_min: %d rate_max: %d\n", rate->min, rate->max); @@ -746,6 +775,7 @@ int sof_pcm_dai_link_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_hw_pa return 0; } +EXPORT_SYMBOL(sof_pcm_dai_link_fixup); static int sof_pcm_probe(struct snd_soc_component *component) { |