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author | Linus Torvalds <torvalds@linux-foundation.org> | 2014-08-21 17:24:40 -0400 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2014-08-21 17:24:40 -0400 |
commit | e9d99a1dec59a9ebf59ddc040ac34a92b1a93c97 (patch) | |
tree | 364e24f8da27267179688871193f0819cd0e6b35 /sound/soc/codecs | |
parent | 29fdd5ba6297f865046e3793481be4979f75d85a (diff) | |
parent | ca2e7224d7e7d424e69616634f90f3f428710085 (diff) |
Merge tag 'sound-3.17-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound
Pull sound fixes from Takashi Iwai:
"A bunch of ASoC fixes with a few HD-audio fixes in this pull request.
All fairly small, boring and device-specific fixes, in addition to
MAINTAINERS update for better reviewing"
* tag 'sound-3.17-rc2' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound:
ALSA: hda/hdmi - apply Valleyview fix-ups to Cherryview display codec
ALSA: hda/hdmi - set depop_delay for haswell plus
ALSA: hda - restore the gpio led after resume
ALSA: hda/realtek - Avoid setting wrong COEF on ALC269 & co
ASoC: pxa-ssp: drop SNDRV_PCM_FMTBIT_S24_LE
ASoC: fsl-esai: Revert .xlate_tdm_slot_mask() support
ASoC: mcasp: Fix implicit BLCK divider setting
ASoC: arizona: Fix TDM slot length handling in arizona_hw_params
ASoC: pcm512x: Correct Digital Playback control names
ASoC: dapm: Fix uninitialized variable in snd_soc_dapm_get_enum_double()
ASoC: Intel: Restore Baytrail ADSP streams only when ADSP was in reset
ASoC: Intel: Wait Baytrail ADSP boot at resume_early stage
ASoC: Intel: Merge Baytrail ADSP suspend_noirq into suspend_late
MAINTAINERS: Add i.MX maintainers and paths to Freescale ASoC entry
ASoC: Intel: Update Baytrail ADSP firmware name
Diffstat (limited to 'sound/soc/codecs')
-rw-r--r-- | sound/soc/codecs/arizona.c | 6 | ||||
-rw-r--r-- | sound/soc/codecs/pcm512x.c | 4 |
2 files changed, 6 insertions, 4 deletions
diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index bd41ee4da078..2c71f16bd661 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c | |||
@@ -1278,6 +1278,8 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, | |||
1278 | else | 1278 | else |
1279 | rates = &arizona_48k_bclk_rates[0]; | 1279 | rates = &arizona_48k_bclk_rates[0]; |
1280 | 1280 | ||
1281 | wl = snd_pcm_format_width(params_format(params)); | ||
1282 | |||
1281 | if (tdm_slots) { | 1283 | if (tdm_slots) { |
1282 | arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n", | 1284 | arizona_aif_dbg(dai, "Configuring for %d %d bit TDM slots\n", |
1283 | tdm_slots, tdm_width); | 1285 | tdm_slots, tdm_width); |
@@ -1285,6 +1287,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, | |||
1285 | channels = tdm_slots; | 1287 | channels = tdm_slots; |
1286 | } else { | 1288 | } else { |
1287 | bclk_target = snd_soc_params_to_bclk(params); | 1289 | bclk_target = snd_soc_params_to_bclk(params); |
1290 | tdm_width = wl; | ||
1288 | } | 1291 | } |
1289 | 1292 | ||
1290 | if (chan_limit && chan_limit < channels) { | 1293 | if (chan_limit && chan_limit < channels) { |
@@ -1319,8 +1322,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, | |||
1319 | arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n", | 1322 | arizona_aif_dbg(dai, "BCLK %dHz LRCLK %dHz\n", |
1320 | rates[bclk], rates[bclk] / lrclk); | 1323 | rates[bclk], rates[bclk] / lrclk); |
1321 | 1324 | ||
1322 | wl = snd_pcm_format_width(params_format(params)); | 1325 | frame = wl << ARIZONA_AIF1TX_WL_SHIFT | tdm_width; |
1323 | frame = wl << ARIZONA_AIF1TX_WL_SHIFT | wl; | ||
1324 | 1326 | ||
1325 | reconfig = arizona_aif_cfg_changed(codec, base, bclk, lrclk, frame); | 1327 | reconfig = arizona_aif_cfg_changed(codec, base, bclk, lrclk, frame); |
1326 | 1328 | ||
diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 163ec3855fd4..0c8aefab404c 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c | |||
@@ -259,13 +259,13 @@ static const struct soc_enum pcm512x_veds = | |||
259 | pcm512x_ramp_step_text); | 259 | pcm512x_ramp_step_text); |
260 | 260 | ||
261 | static const struct snd_kcontrol_new pcm512x_controls[] = { | 261 | static const struct snd_kcontrol_new pcm512x_controls[] = { |
262 | SOC_DOUBLE_R_TLV("Playback Digital Volume", PCM512x_DIGITAL_VOLUME_2, | 262 | SOC_DOUBLE_R_TLV("Digital Playback Volume", PCM512x_DIGITAL_VOLUME_2, |
263 | PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv), | 263 | PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv), |
264 | SOC_DOUBLE_TLV("Playback Volume", PCM512x_ANALOG_GAIN_CTRL, | 264 | SOC_DOUBLE_TLV("Playback Volume", PCM512x_ANALOG_GAIN_CTRL, |
265 | PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv), | 265 | PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv), |
266 | SOC_DOUBLE_TLV("Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST, | 266 | SOC_DOUBLE_TLV("Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST, |
267 | PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv), | 267 | PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv), |
268 | SOC_DOUBLE("Playback Digital Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, | 268 | SOC_DOUBLE("Digital Playback Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, |
269 | PCM512x_RQMR_SHIFT, 1, 1), | 269 | PCM512x_RQMR_SHIFT, 1, 1), |
270 | 270 | ||
271 | SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1), | 271 | SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1), |