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| author | Takashi Iwai <tiwai@suse.de> | 2015-06-08 20:47:53 +0200 |
|---|---|---|
| committer | Takashi Iwai <tiwai@suse.de> | 2015-06-08 20:47:53 +0200 |
| commit | 8ffc57093bb1c270050f4229b5afd38ee8cef2bd (patch) | |
| tree | ba78b568b446b9c119385ef17a0fa5d5d0724be2 /sound/soc/codecs/88pm860x-codec.c | |
| parent | ALSA: hda - add new HDA registers (diff) | |
| parent | Merge remote-tracking branches 'asoc/topic/wm8994', 'asoc/topic/wm8996' and '... (diff) | |
| download | linux-8ffc57093bb1c270050f4229b5afd38ee8cef2bd.tar.xz linux-8ffc57093bb1c270050f4229b5afd38ee8cef2bd.zip | |
Merge tag 'asoc-v4.2' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-next
ASoC: Updates for v4.2
The big thing this release has been Liam's addition of topology support
to the core. We've also seen quite a bit of driver work and the
continuation of Lars' refactoring for component support.
- 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 and cleanup serieses for the Wolfson ADSP and TI
TAS2552 drivers.
- Support for TI TAS571x power amplifiers.
- Support for Qualcomm APQ8016 and ZTE ZX296702 SoCs.
- Support for x86 systems with RT5650 and Qualcomm Storm.
Diffstat (limited to 'sound/soc/codecs/88pm860x-codec.c')
| -rw-r--r-- | sound/soc/codecs/88pm860x-codec.c | 19 |
1 files changed, 9 insertions, 10 deletions
diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index a0f265327fdf..38b3dad9d48a 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1140,7 +1140,7 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_STANDBY: - if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + if (snd_soc_codec_get_bias_level(codec) == SND_SOC_BIAS_OFF) { /* Enable Audio PLL & Audio section */ data = AUDIO_PLL | AUDIO_SECTION_ON; pm860x_reg_write(pm860x->i2c, REG_MISC2, data); @@ -1156,7 +1156,6 @@ static int pm860x_set_bias_level(struct snd_soc_codec *codec, pm860x_set_bits(pm860x->i2c, REG_MISC2, data, 0); break; } - codec->dapm.bias_level = level; return 0; } @@ -1187,16 +1186,16 @@ static struct snd_soc_dai_driver pm860x_dai[] = { .channels_min = 2, .channels_max = 2, .rates = PM860X_RATES, - .formats = SNDRV_PCM_FORMAT_S16_LE | \ - SNDRV_PCM_FORMAT_S18_3LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE, }, .capture = { .stream_name = "PCM Capture", .channels_min = 2, .channels_max = 2, .rates = PM860X_RATES, - .formats = SNDRV_PCM_FORMAT_S16_LE | \ - SNDRV_PCM_FORMAT_S18_3LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE, }, .ops = &pm860x_pcm_dai_ops, }, { @@ -1208,16 +1207,16 @@ static struct snd_soc_dai_driver pm860x_dai[] = { .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FORMAT_S16_LE | \ - SNDRV_PCM_FORMAT_S18_3LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE, }, .capture = { .stream_name = "I2S Capture", .channels_min = 2, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, - .formats = SNDRV_PCM_FORMAT_S16_LE | \ - SNDRV_PCM_FORMAT_S18_3LE, + .formats = SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S18_3LE, }, .ops = &pm860x_i2s_dai_ops, }, |