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author | Linus Torvalds <torvalds@linux-foundation.org> | 2015-06-25 17:15:18 -0700 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2015-06-25 17:15:18 -0700 |
commit | 4570a37169d4b44d316f40b2ccc681dc93fedc7b (patch) | |
tree | cafffb586c60dddfb04b8619fa1ae0e859600de7 /sound/firewire/bebob/bebob_pcm.c | |
parent | Merge branch 'dmi-for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/jdelvare/staging (diff) | |
parent | ALSA: pcm: Fix pcm_class sysfs output (diff) | |
download | linux-dev-4570a37169d4b44d316f40b2ccc681dc93fedc7b.tar.xz linux-dev-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_pcm.c')
-rw-r--r-- | sound/firewire/bebob/bebob_pcm.c | 14 |
1 files changed, 7 insertions, 7 deletions
diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c index 4a55561ed4ec..7a2c1f53bc44 100644 --- a/sound/firewire/bebob/bebob_pcm.c +++ b/sound/firewire/bebob/bebob_pcm.c @@ -157,7 +157,7 @@ pcm_open(struct snd_pcm_substream *substream) struct snd_bebob *bebob = substream->private_data; struct snd_bebob_rate_spec *spec = bebob->spec->rate; unsigned int sampling_rate; - bool internal; + enum snd_bebob_clock_type src; int err; err = snd_bebob_stream_lock_try(bebob); @@ -168,7 +168,7 @@ pcm_open(struct snd_pcm_substream *substream) if (err < 0) goto err_locked; - err = snd_bebob_stream_check_internal_clock(bebob, &internal); + err = snd_bebob_stream_get_clock_src(bebob, &src); if (err < 0) goto err_locked; @@ -176,7 +176,7 @@ pcm_open(struct snd_pcm_substream *substream) * When source of clock is internal or any PCM stream are running, * the available sampling rate is limited at current sampling rate. */ - if (!internal || + if (src == SND_BEBOB_CLOCK_TYPE_EXTERNAL || amdtp_stream_pcm_running(&bebob->tx_stream) || amdtp_stream_pcm_running(&bebob->rx_stream)) { err = spec->get(bebob, &sampling_rate); @@ -213,7 +213,7 @@ pcm_capture_hw_params(struct snd_pcm_substream *substream, struct snd_bebob *bebob = substream->private_data; if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) - atomic_inc(&bebob->capture_substreams); + atomic_inc(&bebob->substreams_counter); amdtp_stream_set_pcm_format(&bebob->tx_stream, params_format(hw_params)); return snd_pcm_lib_alloc_vmalloc_buffer(substream, @@ -226,7 +226,7 @@ pcm_playback_hw_params(struct snd_pcm_substream *substream, struct snd_bebob *bebob = substream->private_data; if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) - atomic_inc(&bebob->playback_substreams); + atomic_inc(&bebob->substreams_counter); amdtp_stream_set_pcm_format(&bebob->rx_stream, params_format(hw_params)); return snd_pcm_lib_alloc_vmalloc_buffer(substream, @@ -239,7 +239,7 @@ pcm_capture_hw_free(struct snd_pcm_substream *substream) struct snd_bebob *bebob = substream->private_data; if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) - atomic_dec(&bebob->capture_substreams); + atomic_dec(&bebob->substreams_counter); snd_bebob_stream_stop_duplex(bebob); @@ -251,7 +251,7 @@ pcm_playback_hw_free(struct snd_pcm_substream *substream) struct snd_bebob *bebob = substream->private_data; if (substream->runtime->status->state != SNDRV_PCM_STATE_OPEN) - atomic_dec(&bebob->playback_substreams); + atomic_dec(&bebob->substreams_counter); snd_bebob_stream_stop_duplex(bebob); |