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author | Linus Torvalds <torvalds@linux-foundation.org> | 2016-05-19 23:41:32 +0300 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2016-05-19 23:41:32 +0300 |
commit | f4c80d5a16eb4b08a0d9ade154af1ebdc63f5752 (patch) | |
tree | 5334acabf48210285333bc80d4a3e326efb36750 /sound/soc/soc-generic-dmaengine-pcm.c | |
parent | 7afd16f882887c9adc69cd1794f5e57777723217 (diff) | |
parent | 17e1717c11a34f9b0956e33e0c4a4e4ae8c51a57 (diff) | |
download | linux-f4c80d5a16eb4b08a0d9ade154af1ebdc63f5752.tar.xz |
Merge tag 'sound-4.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai:
"This time was again a relatively calm development cycle; most of
updates are about drivers, and no radical changes are seen in any core
code. Here are some highlights:
ALSA core:
- Continued hardening of ALSA hrtimer
- A few leak fixes in timer interface
- Fix poll error handling in PCM and compress
- Add error propagation in compress API
- Removal of dead rtctimer driver
HD-audio:
- Native ELD notify support for i915 HDMI
- Realtek ALC234 & co support
- Code refactoring to standardize chmap support
- Continued development for SKL HDMI core support
Firewire:
- Apply delayed card registration to all drivers
- Improved / stabilized the handling of PCM stream start / stop
- Add tracepoints to dump a part of isochronous packet data
- Fixed incoming/outgoing packet parameter usages
- Add support for M-Audio profire series
USB-audio:
- Fixes for UAC2 clock source
- SS+ support
- Workaround for oft-seen repeated sample rate read errors
ASoC:
- Further slow progress on the topology code
- Substantial updates and improvements for the da7219, es8328,
fsl-ssi, Intel and rcar drivers.
- Compress error handling in WM ADSP driver"
* tag 'sound-4.7-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (177 commits)
ALSA: firewire-lib: change a member of event structure to suppress sparse wanings to bool type
sound: oss: Use setup_timer and mod_timer.
ASoC: hdac_hdmi: Remove the unused 'timeout' variable
ASoC: fsl_ssi: Fix channel slipping on capture (or playback) restart in full duplex.
ASoC: fsl_ssi: Fix channel slipping in Playback at startup
ASoC: fsl_ssi: Fix samples being dropped at Playback startup
ASoC: fsl_ssi: Save a dev reference for dev_err() purpose.
ASoC: fsl_ssi: The IPG/5 limitation concerns the bitclk, not the sysclk.
ASoC: fsl_ssi: Real hardware channels max number is 32
ASoC: pcm5102a: Add support for PCM5102A codec
ASoC: hdac_hdmi: add link management
ASoC: Intel: Skylake: add link management
ALSA: hdac: add link pm and ref counting
ALSA: au88x0: Fix zero clear of stream->resources
ASoC: rt298: Add DMI match for Broxton-P reference platform
ASoC: rt298: fix null deref on acpi driver data
ASoC: dapm: deprecate MICBIAS widget type
ALSA: firewire-lib: drop skip argument from helper functions to queue a packet
ALSA: firewire-lib: add context information to tracepoints
ALSA: firewire-lib: permit to flush queued packets only in process context for better PCM period granularity
...
Diffstat (limited to 'sound/soc/soc-generic-dmaengine-pcm.c')
-rw-r--r-- | sound/soc/soc-generic-dmaengine-pcm.c | 57 |
1 files changed, 34 insertions, 23 deletions
diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 6fd1906af387..6cef3977507a 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -163,31 +163,42 @@ static int dmaengine_pcm_set_runtime_hwparams(struct snd_pcm_substream *substrea } /* - * Prepare formats mask for valid/allowed sample types. If the dma does - * not have support for the given physical word size, it needs to be - * masked out so user space can not use the format which produces - * corrupted audio. - * In case the dma driver does not implement the slave_caps the default - * assumption is that it supports 1, 2 and 4 bytes widths. + * If SND_DMAENGINE_PCM_DAI_FLAG_PACK is set keep + * hw.formats set to 0, meaning no restrictions are in place. + * In this case it's the responsibility of the DAI driver to + * provide the supported format information. */ - for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { - int bits = snd_pcm_format_physical_width(i); - - /* Enable only samples with DMA supported physical widths */ - switch (bits) { - case 8: - case 16: - case 24: - case 32: - case 64: - if (addr_widths & (1 << (bits / 8))) - hw.formats |= (1LL << i); - break; - default: - /* Unsupported types */ - break; + if (!(dma_data->flags & SND_DMAENGINE_PCM_DAI_FLAG_PACK)) + /* + * Prepare formats mask for valid/allowed sample types. If the + * dma does not have support for the given physical word size, + * it needs to be masked out so user space can not use the + * format which produces corrupted audio. + * In case the dma driver does not implement the slave_caps the + * default assumption is that it supports 1, 2 and 4 bytes + * widths. + */ + for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) { + int bits = snd_pcm_format_physical_width(i); + + /* + * Enable only samples with DMA supported physical + * widths + */ + switch (bits) { + case 8: + case 16: + case 24: + case 32: + case 64: + if (addr_widths & (1 << (bits / 8))) + hw.formats |= (1LL << i); + break; + default: + /* Unsupported types */ + break; + } } - } return snd_soc_set_runtime_hwparams(substream, &hw); } |