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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/codecs/adau1761.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/codecs/adau1761.c')
-rw-r--r--sound/soc/codecs/adau1761.c27
1 files changed, 12 insertions, 15 deletions
diff --git a/sound/soc/codecs/adau1761.c b/sound/soc/codecs/adau1761.c
index a1baeee160f4..2f12477e539e 100644
--- a/sound/soc/codecs/adau1761.c
+++ b/sound/soc/codecs/adau1761.c
@@ -466,7 +466,6 @@ static int adau1761_set_bias_level(struct snd_soc_codec *codec,
break;
}
- codec->dapm.bias_level = level;
return 0;
}
@@ -483,6 +482,7 @@ static enum adau1761_output_mode adau1761_get_lineout_mode(
static int adau1761_setup_digmic_jackdetect(struct snd_soc_codec *codec)
{
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
struct adau1761_platform_data *pdata = codec->dev->platform_data;
struct adau *adau = snd_soc_codec_get_drvdata(codec);
enum adau1761_digmic_jackdet_pin_mode mode;
@@ -515,21 +515,18 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_codec *codec)
if (ret)
return ret;
case ADAU1761_DIGMIC_JACKDET_PIN_MODE_NONE: /* fallthrough */
- ret = snd_soc_dapm_add_routes(&codec->dapm,
- adau1761_no_dmic_routes,
+ ret = snd_soc_dapm_add_routes(dapm, adau1761_no_dmic_routes,
ARRAY_SIZE(adau1761_no_dmic_routes));
if (ret)
return ret;
break;
case ADAU1761_DIGMIC_JACKDET_PIN_MODE_DIGMIC:
- ret = snd_soc_dapm_new_controls(&codec->dapm,
- adau1761_dmic_widgets,
+ ret = snd_soc_dapm_new_controls(dapm, adau1761_dmic_widgets,
ARRAY_SIZE(adau1761_dmic_widgets));
if (ret)
return ret;
- ret = snd_soc_dapm_add_routes(&codec->dapm,
- adau1761_dmic_routes,
+ ret = snd_soc_dapm_add_routes(dapm, adau1761_dmic_routes,
ARRAY_SIZE(adau1761_dmic_routes));
if (ret)
return ret;
@@ -547,6 +544,7 @@ static int adau1761_setup_digmic_jackdetect(struct snd_soc_codec *codec)
static int adau1761_setup_headphone_mode(struct snd_soc_codec *codec)
{
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
struct adau *adau = snd_soc_codec_get_drvdata(codec);
struct adau1761_platform_data *pdata = codec->dev->platform_data;
enum adau1761_output_mode mode;
@@ -577,12 +575,12 @@ static int adau1761_setup_headphone_mode(struct snd_soc_codec *codec)
}
if (mode == ADAU1761_OUTPUT_MODE_HEADPHONE_CAPLESS) {
- ret = snd_soc_dapm_new_controls(&codec->dapm,
+ ret = snd_soc_dapm_new_controls(dapm,
adau1761_capless_dapm_widgets,
ARRAY_SIZE(adau1761_capless_dapm_widgets));
if (ret)
return ret;
- ret = snd_soc_dapm_add_routes(&codec->dapm,
+ ret = snd_soc_dapm_add_routes(dapm,
adau1761_capless_dapm_routes,
ARRAY_SIZE(adau1761_capless_dapm_routes));
} else {
@@ -590,12 +588,12 @@ static int adau1761_setup_headphone_mode(struct snd_soc_codec *codec)
ARRAY_SIZE(adau1761_mono_controls));
if (ret)
return ret;
- ret = snd_soc_dapm_new_controls(&codec->dapm,
+ ret = snd_soc_dapm_new_controls(dapm,
adau1761_mono_dapm_widgets,
ARRAY_SIZE(adau1761_mono_dapm_widgets));
if (ret)
return ret;
- ret = snd_soc_dapm_add_routes(&codec->dapm,
+ ret = snd_soc_dapm_add_routes(dapm,
adau1761_mono_dapm_routes,
ARRAY_SIZE(adau1761_mono_dapm_routes));
}
@@ -640,6 +638,7 @@ static bool adau1761_readable_register(struct device *dev, unsigned int reg)
static int adau1761_codec_probe(struct snd_soc_codec *codec)
{
+ struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec);
struct adau1761_platform_data *pdata = codec->dev->platform_data;
struct adau *adau = snd_soc_codec_get_drvdata(codec);
int ret;
@@ -692,14 +691,12 @@ static int adau1761_codec_probe(struct snd_soc_codec *codec)
return ret;
if (adau->type == ADAU1761) {
- ret = snd_soc_dapm_new_controls(&codec->dapm,
- adau1761_dapm_widgets,
+ ret = snd_soc_dapm_new_controls(dapm, adau1761_dapm_widgets,
ARRAY_SIZE(adau1761_dapm_widgets));
if (ret)
return ret;
- ret = snd_soc_dapm_add_routes(&codec->dapm,
- adau1761_dapm_routes,
+ ret = snd_soc_dapm_add_routes(dapm, adau1761_dapm_routes,
ARRAY_SIZE(adau1761_dapm_routes));
if (ret)
return ret;