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authorLinus Torvalds <torvalds@linux-foundation.org>2015-06-26 03:15:18 +0300
committerLinus Torvalds <torvalds@linux-foundation.org>2015-06-26 03:15:18 +0300
commit4570a37169d4b44d316f40b2ccc681dc93fedc7b (patch)
treecafffb586c60dddfb04b8619fa1ae0e859600de7 /sound/soc/pxa/brownstone.c
parentf7b08217c755e88a29b5bd53b4a1d10cd8b3c5f8 (diff)
parent60b93030b44a8c2cd015cebe5624fd7552ec67ec (diff)
downloadlinux-4570a37169d4b44d316f40b2ccc681dc93fedc7b.tar.xz
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/soc/pxa/brownstone.c')
-rw-r--r--sound/soc/pxa/brownstone.c25
1 files changed, 1 insertions, 24 deletions
diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c
index 79936e3e80e7..2b26318bc200 100644
--- a/sound/soc/pxa/brownstone.c
+++ b/sound/soc/pxa/brownstone.c
@@ -45,29 +45,6 @@ static const struct snd_soc_dapm_route brownstone_audio_map[] = {
{"MICBIAS1", NULL, "Main Mic"},
};
-static int brownstone_wm8994_init(struct snd_soc_pcm_runtime *rtd)
-{
- struct snd_soc_codec *codec = rtd->codec;
- struct snd_soc_dapm_context *dapm = &codec->dapm;
-
- /* set endpoints to not connected */
- snd_soc_dapm_nc_pin(dapm, "HPOUT2P");
- snd_soc_dapm_nc_pin(dapm, "HPOUT2N");
- snd_soc_dapm_nc_pin(dapm, "LINEOUT1N");
- snd_soc_dapm_nc_pin(dapm, "LINEOUT1P");
- snd_soc_dapm_nc_pin(dapm, "LINEOUT2N");
- snd_soc_dapm_nc_pin(dapm, "LINEOUT2P");
- snd_soc_dapm_nc_pin(dapm, "IN1LN");
- snd_soc_dapm_nc_pin(dapm, "IN1LP");
- snd_soc_dapm_nc_pin(dapm, "IN1RP");
- snd_soc_dapm_nc_pin(dapm, "IN2LP:VXRN");
- snd_soc_dapm_nc_pin(dapm, "IN2RN");
- snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP");
- snd_soc_dapm_nc_pin(dapm, "IN2LN");
-
- return 0;
-}
-
static int brownstone_wm8994_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
@@ -115,7 +92,6 @@ static struct snd_soc_dai_link brownstone_wm8994_dai[] = {
.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
SND_SOC_DAIFMT_CBS_CFS,
.ops = &brownstone_ops,
- .init = brownstone_wm8994_init,
},
};
@@ -132,6 +108,7 @@ static struct snd_soc_card brownstone = {
.num_dapm_widgets = ARRAY_SIZE(brownstone_dapm_widgets),
.dapm_routes = brownstone_audio_map,
.num_dapm_routes = ARRAY_SIZE(brownstone_audio_map),
+ .fully_routed = true,
};
static int brownstone_probe(struct platform_device *pdev)