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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 /sound/firewire/bebob/bebob.h
parentf7b08217c755e88a29b5bd53b4a1d10cd8b3c5f8 (diff)
parent60b93030b44a8c2cd015cebe5624fd7552ec67ec (diff)
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.h')
-rw-r--r--sound/firewire/bebob/bebob.h20
1 files changed, 14 insertions, 6 deletions
diff --git a/sound/firewire/bebob/bebob.h b/sound/firewire/bebob/bebob.h
index dfbcd233178c..d23caca7f369 100644
--- a/sound/firewire/bebob/bebob.h
+++ b/sound/firewire/bebob/bebob.h
@@ -49,10 +49,15 @@ struct snd_bebob_stream_formation {
extern const unsigned int snd_bebob_rate_table[SND_BEBOB_STRM_FMT_ENTRIES];
/* device specific operations */
-#define SND_BEBOB_CLOCK_INTERNAL "Internal"
+enum snd_bebob_clock_type {
+ SND_BEBOB_CLOCK_TYPE_INTERNAL = 0,
+ SND_BEBOB_CLOCK_TYPE_EXTERNAL,
+ SND_BEBOB_CLOCK_TYPE_SYT,
+};
struct snd_bebob_clock_spec {
unsigned int num;
const char *const *labels;
+ enum snd_bebob_clock_type *types;
int (*get)(struct snd_bebob *bebob, unsigned int *id);
};
struct snd_bebob_rate_spec {
@@ -92,8 +97,7 @@ struct snd_bebob {
struct amdtp_stream rx_stream;
struct cmp_connection out_conn;
struct cmp_connection in_conn;
- atomic_t capture_substreams;
- atomic_t playback_substreams;
+ atomic_t substreams_counter;
struct snd_bebob_stream_formation
tx_stream_formations[SND_BEBOB_STRM_FMT_ENTRIES];
@@ -110,6 +114,9 @@ struct snd_bebob {
/* for M-Audio special devices */
void *maudio_special_quirk;
bool deferred_registration;
+
+ /* For BeBoB version quirk. */
+ unsigned int version;
};
static inline int
@@ -159,7 +166,8 @@ enum avc_bridgeco_plug_type {
AVC_BRIDGECO_PLUG_TYPE_MIDI = 0x02,
AVC_BRIDGECO_PLUG_TYPE_SYNC = 0x03,
AVC_BRIDGECO_PLUG_TYPE_ANA = 0x04,
- AVC_BRIDGECO_PLUG_TYPE_DIG = 0x05
+ AVC_BRIDGECO_PLUG_TYPE_DIG = 0x05,
+ AVC_BRIDGECO_PLUG_TYPE_ADDITION = 0x06
};
static inline void
avc_bridgeco_fill_unit_addr(u8 buf[AVC_BRIDGECO_ADDR_BYTES],
@@ -205,8 +213,8 @@ int avc_bridgeco_get_plug_strm_fmt(struct fw_unit *unit,
/* for AMDTP streaming */
int snd_bebob_stream_get_rate(struct snd_bebob *bebob, unsigned int *rate);
int snd_bebob_stream_set_rate(struct snd_bebob *bebob, unsigned int rate);
-int snd_bebob_stream_check_internal_clock(struct snd_bebob *bebob,
- bool *internal);
+int snd_bebob_stream_get_clock_src(struct snd_bebob *bebob,
+ enum snd_bebob_clock_type *src);
int snd_bebob_stream_discover(struct snd_bebob *bebob);
int snd_bebob_stream_init_duplex(struct snd_bebob *bebob);
int snd_bebob_stream_start_duplex(struct snd_bebob *bebob, unsigned int rate);