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authorLinus Torvalds <torvalds@linux-foundation.org>2015-06-26 03:15:18 +0300
committerLinus Torvalds <torvalds@linux-foundation.org>2015-06-26 03:15:18 +0300
commit4570a37169d4b44d316f40b2ccc681dc93fedc7b (patch)
treecafffb586c60dddfb04b8619fa1ae0e859600de7 /sound/soc/soc-pcm.c
parentf7b08217c755e88a29b5bd53b4a1d10cd8b3c5f8 (diff)
parent60b93030b44a8c2cd015cebe5624fd7552ec67ec (diff)
downloadlinux-4570a37169d4b44d316f40b2ccc681dc93fedc7b.tar.xz
Merge tag 'sound-4.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound updates from Takashi Iwai: "It was a busy development cycle at this time, as you can see a wide range of changes in diffstat. There are no big changes but many refactoring and improvements. Here we go some highlights: ALSA core: - Procfs codes were cleaned up to use seq_file - Procfs can be opt out via Kconfig (only for EXPERT) - Two types of jack API were unified finally; now both kctl and input jack devs are handled via a single function call. HD-audio: - Continued code restructuring for the future ASoC driver; now HDA controller driver is split to a core helper module. - Preliminary codes for Skylake audio support in HDA core. - Proper i915 gfx power well management for SKL & co - Enabled runtime PM as default for Intel HDMI/DP codecs - Newer Tegra chip supports - More quirks for Dell headsets, Alienware (with CA0132), etc. - A couple of DRM ELD helper API functions ASoC: - Support for loading ASoC topology maps from firmware, intended to be used to allow self-describing DSP firmware images to be built which can map controls added by the DSP to userspace without the kernel needing to know about individual DSP firmwares - Lots of refactoring to avoid direct access to snd_soc_codec where it's not needed supporting future refactoring - Big refactoring, cleanup and enhancement for the Wolfson ADSP driver - Cleanup series for TI TAS2552 and R-CAR drivers - Fixes and improvements on RT56xx codecs - Support for TI TAS571x power amplifiers - Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs - Support for x86 systems with RT5650 and Qualcomm Storm - Support for Mediatek AFE (Audio Front End) unit - Other various small fixes to ASoC codec drivers Firewire: - Enhanced to allow non-blocking streams to use timestamp synchronization - Improve support for DM1500 and BeBoBv3 Misc: - Cleanup of old pci API functions over all PCI sound drivers - Fix long-standing regression of the old powermac i2c setup" * tag 'sound-4.2-rc1' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound: (533 commits) ALSA: pcm: Fix pcm_class sysfs output ALSA: hda-beep: Update authors dead email address ASoC: wm_adsp: Move DSP Rate controls into the codec ASoC: wm8995: Fix setting sysclk for WM8995_SYSCLK_MCLK2 case ALSA: hda: provide default bus io ops extended hdac ALSA: hda: add hda link cleanup routine ALSA: hda: add hdac_ext stream creation and cleanup routines ASoC: rsrc-card: remove unused ret ALSA: HDAC: move SND_HDA_PREALLOC_SIZE to core ASoC: mediatek: Add machine driver for rt5650 rt5676 codec ASoC: mediatek: Add machine driver for MAX98090 codec ASoC: mediatek: Add AFE platform driver ASoC: rsnd: remove io from rsnd_mod ASoC: rsnd: move rsnd_mod_is_working() to rsnd_io_is_working() ASoC: rsnd: don't use rsnd_mod_to_io() on snd_kcontrol ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_src_xxx() ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_ssi_xxx() ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_dma_xxx() ASoC: rsnd: don't use rsnd_mod_to_io() on rsnd_get_adinr() ASoC: rsnd: add common interrupt handler for SSI/SRC/DMA ...
Diffstat (limited to 'sound/soc/soc-pcm.c')
-rw-r--r--sound/soc/soc-pcm.c47
1 files changed, 42 insertions, 5 deletions
diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c
index 35fe58f4fa86..256b9c91aa94 100644
--- a/sound/soc/soc-pcm.c
+++ b/sound/soc/soc-pcm.c
@@ -1485,30 +1485,67 @@ unwind:
}
static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime,
- struct snd_soc_pcm_stream *stream)
+ struct snd_soc_pcm_stream *stream,
+ u64 formats)
{
runtime->hw.rate_min = stream->rate_min;
runtime->hw.rate_max = stream->rate_max;
runtime->hw.channels_min = stream->channels_min;
runtime->hw.channels_max = stream->channels_max;
if (runtime->hw.formats)
- runtime->hw.formats &= stream->formats;
+ runtime->hw.formats &= formats & stream->formats;
else
- runtime->hw.formats = stream->formats;
+ runtime->hw.formats = formats & stream->formats;
runtime->hw.rates = stream->rates;
}
+static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream)
+{
+ struct snd_soc_pcm_runtime *fe = substream->private_data;
+ struct snd_soc_dpcm *dpcm;
+ u64 formats = ULLONG_MAX;
+ int stream = substream->stream;
+
+ if (!fe->dai_link->dpcm_merged_format)
+ return formats;
+
+ /*
+ * It returns merged BE codec format
+ * if FE want to use it (= dpcm_merged_format)
+ */
+
+ list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) {
+ struct snd_soc_pcm_runtime *be = dpcm->be;
+ struct snd_soc_dai_driver *codec_dai_drv;
+ struct snd_soc_pcm_stream *codec_stream;
+ int i;
+
+ for (i = 0; i < be->num_codecs; i++) {
+ codec_dai_drv = be->codec_dais[i]->driver;
+ if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+ codec_stream = &codec_dai_drv->playback;
+ else
+ codec_stream = &codec_dai_drv->capture;
+
+ formats &= codec_stream->formats;
+ }
+ }
+
+ return formats;
+}
+
static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream)
{
struct snd_pcm_runtime *runtime = substream->runtime;
struct snd_soc_pcm_runtime *rtd = substream->private_data;
struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver;
+ u64 format = dpcm_runtime_base_format(substream);
if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
- dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback);
+ dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback, format);
else
- dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture);
+ dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture, format);
}
static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd);