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author | Linus Torvalds <torvalds@linux-foundation.org> | 2020-01-28 16:26:57 -0800 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2020-01-28 16:26:57 -0800 |
commit | fb95aae6e67c4e319a24b3eea32032d4246a5335 (patch) | |
tree | c310d68211634ef594d180fdd93844fec44de2fe /sound/soc/au1x | |
parent | bd2463ac7d7ec51d432f23bf0e893fb371a908cd (diff) | |
parent | 90fb04f890bcb7384b4d4c216dc2640b0a870df3 (diff) | |
download | linux-fb95aae6e67c4e319a24b3eea32032d4246a5335.tar.bz2 |
Merge tag 'sound-5.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"As the diffstat shows we've had again a lot of works done for this
cycle: the majority of changes are the continued componentization and
code refactoring in ASoC, the tree-wide PCM API updates and cleanups
and SOF updates while a few ASoC driver updates are seen, too.
Here we go, some highlights:
Core:
- Finally y2038 support landed to ALSA ABI; some ioctls have been
extended and lots of tricks were applied
- Applying the new managed PCM buffer API to all drivers; the API
itself was already merged in 5.5
- The already deprecated dimension support in ALSA control API is
dropped completely now
- Verification of ALSA control elements to catch API misuses
ASoC:
- Further code refactorings and moving things to the component level
- Lots of updates and improvements on SOF / Intel drivers; now
including common HDMI driver and SoundWire support
- New driver support for Ingenic JZ4770, Mediatek MT6660, Qualcomm
WCD934x and WSA881x, and Realtek RT700, RT711, RT715, RT1011,
RT1015 and RT1308
HD-audio:
- Improved ring-buffer communications using waitqueue
- Drop the superfluous buffer preallocation on x86
Others:
- Many code cleanups, mostly constifications over the whole tree
- USB-audio: quirks for MOTU, Corsair Virtuoso, Line6 Helix
- FireWire: code refactoring for oxfw and dice drivers"
* tag 'sound-5.6-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (638 commits)
ALSA: usb-audio: add quirks for Line6 Helix devices fw>=2.82
ALSA: hda: Add Clevo W65_67SB the power_save blacklist
ASoC: soc-core: remove null_snd_soc_ops
ASoC: soc-pcm: add soc_rtd_trigger()
ASoC: soc-pcm: add soc_rtd_hw_free()
ASoC: soc-pcm: add soc_rtd_hw_params()
ASoC: soc-pcm: add soc_rtd_prepare()
ASoC: soc-pcm: add soc_rtd_shutdown()
ASoC: soc-pcm: add soc_rtd_startup()
ASoC: rt1015: add rt1015 amplifier driver
ASoC: madera: Correct some kernel doc
ASoC: topology: fix soc_tplg_fe_link_create() - link->dobj initialization order
ASoC: Intel: skl_hda_dsp_common: Fix global-out-of-bounds bug
ASoC: madera: Correct DMIC only input hook ups
ALSA: cs46xx: fix spelling mistake "to" -> "too"
ALSA: hda - Add docking station support for Lenovo Thinkpad T420s
ASoC: Add MediaTek MT6660 Speaker Amp Driver
ASoC: dt-bindings: rt5645: add suppliers
ASoC: max98090: fix deadlock in max98090_dapm_put_enum_double()
ASoC: dapm: add snd_soc_dapm_put_enum_double_locked
...
Diffstat (limited to 'sound/soc/au1x')
-rw-r--r-- | sound/soc/au1x/ac97c.c | 1 | ||||
-rw-r--r-- | sound/soc/au1x/dbdma2.c | 15 | ||||
-rw-r--r-- | sound/soc/au1x/dma.c | 22 | ||||
-rw-r--r-- | sound/soc/au1x/psc-ac97.c | 1 |
4 files changed, 7 insertions, 32 deletions
diff --git a/sound/soc/au1x/ac97c.c b/sound/soc/au1x/ac97c.c index d28302153d74..3b1700e665f5 100644 --- a/sound/soc/au1x/ac97c.c +++ b/sound/soc/au1x/ac97c.c @@ -206,7 +206,6 @@ static int au1xac97c_dai_probe(struct snd_soc_dai *dai) static struct snd_soc_dai_driver au1xac97c_dai_driver = { .name = "alchemy-ac97c", - .bus_control = true, .probe = au1xac97c_dai_probe, .playback = { .rates = AC97_RATES, diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 4553108ec92a..8f855644c6b4 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -197,10 +197,6 @@ static int au1xpsc_pcm_hw_params(struct snd_soc_component *component, struct au1xpsc_audio_dmadata *pcd; int stype, ret; - ret = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); - if (ret < 0) - goto out; - stype = substream->stream; pcd = to_dmadata(substream, component); @@ -232,13 +228,6 @@ out: return ret; } -static int au1xpsc_pcm_hw_free(struct snd_soc_component *component, - struct snd_pcm_substream *substream) -{ - snd_pcm_lib_free_pages(substream); - return 0; -} - static int au1xpsc_pcm_prepare(struct snd_soc_component *component, struct snd_pcm_substream *substream) { @@ -315,7 +304,7 @@ static int au1xpsc_pcm_new(struct snd_soc_component *component, struct snd_card *card = rtd->card->snd_card; struct snd_pcm *pcm = rtd->pcm; - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_DEV, card->dev, AU1XPSC_BUFFER_MIN_BYTES, (4096 * 1024) - 1); return 0; @@ -326,9 +315,7 @@ static struct snd_soc_component_driver au1xpsc_soc_component = { .name = DRV_NAME, .open = au1xpsc_pcm_open, .close = au1xpsc_pcm_close, - .ioctl = snd_soc_pcm_lib_ioctl, .hw_params = au1xpsc_pcm_hw_params, - .hw_free = au1xpsc_pcm_hw_free, .prepare = au1xpsc_pcm_prepare, .trigger = au1xpsc_pcm_trigger, .pointer = au1xpsc_pcm_pointer, diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c index 054dfda89d3e..c9a038a5e2d3 100644 --- a/sound/soc/au1x/dma.c +++ b/sound/soc/au1x/dma.c @@ -231,19 +231,10 @@ static int alchemy_pcm_hw_params(struct snd_soc_component *component, struct snd_pcm_hw_params *hw_params) { struct audio_stream *stream = ss_to_as(substream, component); - int err; - - err = snd_pcm_lib_malloc_pages(substream, - params_buffer_bytes(hw_params)); - if (err < 0) - return err; - err = au1000_setup_dma_link(stream, - params_period_bytes(hw_params), - params_periods(hw_params)); - if (err) - snd_pcm_lib_free_pages(substream); - return err; + return au1000_setup_dma_link(stream, + params_period_bytes(hw_params), + params_periods(hw_params)); } static int alchemy_pcm_hw_free(struct snd_soc_component *component, @@ -251,7 +242,7 @@ static int alchemy_pcm_hw_free(struct snd_soc_component *component, { struct audio_stream *stream = ss_to_as(substream, component); au1000_release_dma_link(stream); - return snd_pcm_lib_free_pages(substream); + return 0; } static int alchemy_pcm_trigger(struct snd_soc_component *component, @@ -292,8 +283,8 @@ static int alchemy_pcm_new(struct snd_soc_component *component, { struct snd_pcm *pcm = rtd->pcm; - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, - NULL, 65536, (4096 * 1024) - 1); + snd_pcm_set_managed_buffer_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, + NULL, 65536, (4096 * 1024) - 1); return 0; } @@ -302,7 +293,6 @@ static struct snd_soc_component_driver alchemy_pcm_soc_component = { .name = DRV_NAME, .open = alchemy_pcm_open, .close = alchemy_pcm_close, - .ioctl = snd_soc_pcm_lib_ioctl, .hw_params = alchemy_pcm_hw_params, .hw_free = alchemy_pcm_hw_free, .trigger = alchemy_pcm_trigger, diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 08bc04e2da2a..0227993c5da8 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -339,7 +339,6 @@ static const struct snd_soc_dai_ops au1xpsc_ac97_dai_ops = { }; static const struct snd_soc_dai_driver au1xpsc_ac97_dai_template = { - .bus_control = true, .probe = au1xpsc_ac97_probe, .playback = { .rates = AC97_RATES, |