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authorLinus Torvalds <torvalds@linux-foundation.org>2016-05-19 23:41:32 +0300
committerLinus Torvalds <torvalds@linux-foundation.org>2016-05-19 23:41:32 +0300
commitf4c80d5a16eb4b08a0d9ade154af1ebdc63f5752 (patch)
tree5334acabf48210285333bc80d4a3e326efb36750 /sound/soc/soc-generic-dmaengine-pcm.c
parent7afd16f882887c9adc69cd1794f5e57777723217 (diff)
parent17e1717c11a34f9b0956e33e0c4a4e4ae8c51a57 (diff)
downloadlinux-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.c57
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);
}