diff options
author | Linus Torvalds <torvalds@linux-foundation.org> | 2015-06-25 17:15:18 -0700 |
---|---|---|
committer | Linus Torvalds <torvalds@linux-foundation.org> | 2015-06-25 17:15:18 -0700 |
commit | 4570a37169d4b44d316f40b2ccc681dc93fedc7b (patch) | |
tree | cafffb586c60dddfb04b8619fa1ae0e859600de7 /include/sound/soc-dapm.h | |
parent | f7b08217c755e88a29b5bd53b4a1d10cd8b3c5f8 (diff) | |
parent | 60b93030b44a8c2cd015cebe5624fd7552ec67ec (diff) | |
download | linux-stable-4570a37169d4b44d316f40b2ccc681dc93fedc7b.tar.gz linux-stable-4570a37169d4b44d316f40b2ccc681dc93fedc7b.tar.bz2 linux-stable-4570a37169d4b44d316f40b2ccc681dc93fedc7b.zip |
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 'include/sound/soc-dapm.h')
-rw-r--r-- | include/sound/soc-dapm.h | 49 |
1 files changed, 49 insertions, 0 deletions
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 1065095c6973..37d95a898275 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -15,6 +15,8 @@ #include <linux/types.h> #include <sound/control.h> +#include <sound/soc-topology.h> +#include <sound/asoc.h> struct device; @@ -107,6 +109,10 @@ struct device; { .id = snd_soc_dapm_mux, .name = wname, \ SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ .kcontrol_news = wcontrols, .num_kcontrols = 1} +#define SND_SOC_DAPM_DEMUX(wname, wreg, wshift, winvert, wcontrols) \ +{ .id = snd_soc_dapm_demux, .name = wname, \ + SND_SOC_DAPM_INIT_REG_VAL(wreg, wshift, winvert), \ + .kcontrol_news = wcontrols, .num_kcontrols = 1} /* Simplified versions of above macros, assuming wncontrols = ARRAY_SIZE(wcontrols) */ #define SOC_PGA_ARRAY(wname, wreg, wshift, winvert,\ @@ -444,11 +450,15 @@ int snd_soc_dapm_dai_get_connected_widgets(struct snd_soc_dai *dai, int stream, struct snd_soc_dapm_context *snd_soc_dapm_kcontrol_dapm( struct snd_kcontrol *kcontrol); +int snd_soc_dapm_force_bias_level(struct snd_soc_dapm_context *dapm, + enum snd_soc_bias_level level); + /* dapm widget types */ enum snd_soc_dapm_type { snd_soc_dapm_input = 0, /* input pin */ snd_soc_dapm_output, /* output pin */ snd_soc_dapm_mux, /* selects 1 analog signal from many inputs */ + snd_soc_dapm_demux, /* connects the input to one of multiple outputs */ snd_soc_dapm_mixer, /* mixes several analog signals together */ snd_soc_dapm_mixer_named_ctl, /* mixer with named controls */ snd_soc_dapm_pga, /* programmable gain/attenuation (volume) */ @@ -563,6 +573,7 @@ struct snd_soc_dapm_widget { int num_kcontrols; const struct snd_kcontrol_new *kcontrol_news; struct snd_kcontrol **kcontrols; + struct snd_soc_dobj dobj; /* widget input and outputs */ struct list_head sources; @@ -585,6 +596,10 @@ struct snd_soc_dapm_update { int val; }; +struct snd_soc_dapm_wcache { + struct snd_soc_dapm_widget *widget; +}; + /* DAPM context */ struct snd_soc_dapm_context { enum snd_soc_bias_level bias_level; @@ -606,6 +621,9 @@ struct snd_soc_dapm_context { int (*set_bias_level)(struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level); + struct snd_soc_dapm_wcache path_sink_cache; + struct snd_soc_dapm_wcache path_source_cache; + #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dapm; #endif @@ -623,4 +641,35 @@ struct snd_soc_dapm_stats { int neighbour_checks; }; +/** + * snd_soc_dapm_init_bias_level() - Initialize DAPM bias level + * @dapm: The DAPM context to initialize + * @level: The DAPM level to initialize to + * + * This function only sets the driver internal state of the DAPM level and will + * not modify the state of the device. Hence it should not be used during normal + * operation, but only to synchronize the internal state to the device state. + * E.g. during driver probe to set the DAPM level to the one corresponding with + * the power-on reset state of the device. + * + * To change the DAPM state of the device use snd_soc_dapm_set_bias_level(). + */ +static inline void snd_soc_dapm_init_bias_level( + struct snd_soc_dapm_context *dapm, enum snd_soc_bias_level level) +{ + dapm->bias_level = level; +} + +/** + * snd_soc_dapm_get_bias_level() - Get current DAPM bias level + * @dapm: The context for which to get the bias level + * + * Returns: The current bias level of the passed DAPM context. + */ +static inline enum snd_soc_bias_level snd_soc_dapm_get_bias_level( + struct snd_soc_dapm_context *dapm) +{ + return dapm->bias_level; +} + #endif |