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authorLinus Torvalds <torvalds@linux-foundation.org>2015-06-25 17:15:18 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2015-06-25 17:15:18 -0700
commit4570a37169d4b44d316f40b2ccc681dc93fedc7b (patch)
treecafffb586c60dddfb04b8619fa1ae0e859600de7 /include/sound/soc-dapm.h
parentf7b08217c755e88a29b5bd53b4a1d10cd8b3c5f8 (diff)
parent60b93030b44a8c2cd015cebe5624fd7552ec67ec (diff)
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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.h49
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