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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/intel/atom/sst-atom-controls.h
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/intel/atom/sst-atom-controls.h')
-rw-r--r--sound/soc/intel/atom/sst-atom-controls.h9
1 files changed, 8 insertions, 1 deletions
diff --git a/sound/soc/intel/atom/sst-atom-controls.h b/sound/soc/intel/atom/sst-atom-controls.h
index daecc58f28af..93de8045d4e1 100644
--- a/sound/soc/intel/atom/sst-atom-controls.h
+++ b/sound/soc/intel/atom/sst-atom-controls.h
@@ -562,6 +562,8 @@ struct sst_ssp_config {
u8 active_slot_map;
u8 start_delay;
u16 fs_width;
+ u8 frame_sync_polarity;
+ u8 data_polarity;
};
struct sst_ssp_cfg {
@@ -695,7 +697,7 @@ struct sst_gain_mixer_control {
u16 module_id;
u16 pipe_id;
u16 task_id;
- char pname[44];
+ char pname[SNDRV_CTL_ELEM_ID_NAME_MAXLEN];
struct snd_soc_dapm_widget *w;
};
@@ -867,4 +869,9 @@ struct sst_enum {
SOC_DAPM_ENUM(SST_MUX_CTL_NAME(xpname, xinstance), \
SST_SSP_MUX_ENUM(xreg, xshift, xtexts))
+int sst_fill_ssp_slot(struct snd_soc_dai *dai, unsigned int tx_mask,
+ unsigned int rx_mask, int slots, int slot_width);
+int sst_fill_ssp_config(struct snd_soc_dai *dai, unsigned int fmt);
+void sst_fill_ssp_defaults(struct snd_soc_dai *dai);
+
#endif