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authorLinus Torvalds <torvalds@linux-foundation.org>2015-06-26 02:15:18 +0200
committerLinus Torvalds <torvalds@linux-foundation.org>2015-06-26 02:15:18 +0200
commit4570a37169d4b44d316f40b2ccc681dc93fedc7b (patch)
treecafffb586c60dddfb04b8619fa1ae0e859600de7 /sound/firewire/bebob/bebob_maudio.c
parentMerge branch 'dmi-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git... (diff)
parentALSA: pcm: Fix pcm_class sysfs output (diff)
downloadlinux-4570a37169d4b44d316f40b2ccc681dc93fedc7b.tar.xz
linux-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 'sound/firewire/bebob/bebob_maudio.c')
-rw-r--r--sound/firewire/bebob/bebob_maudio.c23
1 files changed, 16 insertions, 7 deletions
diff --git a/sound/firewire/bebob/bebob_maudio.c b/sound/firewire/bebob/bebob_maudio.c
index 9ee25a63f684..057495d54ab0 100644
--- a/sound/firewire/bebob/bebob_maudio.c
+++ b/sound/firewire/bebob/bebob_maudio.c
@@ -340,9 +340,12 @@ end:
}
/* Clock source control for special firmware */
-static const char *const special_clk_labels[] = {
- SND_BEBOB_CLOCK_INTERNAL " with Digital Mute", "Digital",
- "Word Clock", SND_BEBOB_CLOCK_INTERNAL};
+static enum snd_bebob_clock_type special_clk_types[] = {
+ SND_BEBOB_CLOCK_TYPE_INTERNAL, /* With digital mute */
+ SND_BEBOB_CLOCK_TYPE_EXTERNAL, /* SPDIF/ADAT */
+ SND_BEBOB_CLOCK_TYPE_EXTERNAL, /* Word Clock */
+ SND_BEBOB_CLOCK_TYPE_INTERNAL,
+};
static int special_clk_get(struct snd_bebob *bebob, unsigned int *id)
{
struct special_params *params = bebob->maudio_special_quirk;
@@ -352,7 +355,13 @@ static int special_clk_get(struct snd_bebob *bebob, unsigned int *id)
static int special_clk_ctl_info(struct snd_kcontrol *kctl,
struct snd_ctl_elem_info *einf)
{
- return snd_ctl_enum_info(einf, 1, ARRAY_SIZE(special_clk_labels),
+ static const char *const special_clk_labels[] = {
+ "Internal with Digital Mute",
+ "Digital",
+ "Word Clock",
+ "Internal"
+ };
+ return snd_ctl_enum_info(einf, 1, ARRAY_SIZE(special_clk_types),
special_clk_labels);
}
static int special_clk_ctl_get(struct snd_kcontrol *kctl,
@@ -371,7 +380,7 @@ static int special_clk_ctl_put(struct snd_kcontrol *kctl,
int err, id;
id = uval->value.enumerated.item[0];
- if (id >= ARRAY_SIZE(special_clk_labels))
+ if (id >= ARRAY_SIZE(special_clk_types))
return -EINVAL;
mutex_lock(&bebob->mutex);
@@ -708,8 +717,8 @@ static struct snd_bebob_rate_spec special_rate_spec = {
.set = &special_set_rate,
};
static struct snd_bebob_clock_spec special_clk_spec = {
- .num = ARRAY_SIZE(special_clk_labels),
- .labels = special_clk_labels,
+ .num = ARRAY_SIZE(special_clk_types),
+ .types = special_clk_types,
.get = &special_clk_get,
};
static struct snd_bebob_meter_spec special_meter_spec = {