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| author | Takashi Iwai <tiwai@suse.de> | 2014-08-04 18:28:21 +0200 |
|---|---|---|
| committer | Takashi Iwai <tiwai@suse.de> | 2014-08-04 18:28:21 +0200 |
| commit | e9e3bdffe38b256f920eacc455d30deba5e97655 (patch) | |
| tree | cfa546f6370c117e012c12605da22ce189542127 /sound/soc/codecs/cs42l52.c | |
| parent | f42bb22243d2ae264d721b055f836059fe35321f (diff) | |
| parent | ae34a78c430c37c06404f032fb04e51315204281 (diff) | |
Merge tag 'asoc-v3.17' of git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound into for-linus
ASoC: Updates for v3.17
This has been a pretty exciting release in terms of the framework, we've
finally got support for multiple CODECs attached to a single DAI link
which has been something there's been interest in as long as I've been
working on ASoC. A big thanks to Benoit and Misael for their work on
this.
Otherwise it's been a fairly standard release for development, including
more componentisation work from Lars-Peter and a good selection of both
CODEC and CPU drivers.
- Support for multiple CODECs attached to a single DAI, enabling
systems with for example multiple DAC/speaker drivers on a single
link, contributed by Benoit Cousson based on work from Misael Lopez
Cruz.
- Support for byte controls larger than 256 bytes based on the use of
TLVs contributed by Omair Mohammed Abdullah.
- More componentisation work from Lars-Peter Clausen.
- The remainder of the conversions of CODEC drivers to params_width()
- Drivers for Cirrus Logic CS4265, Freescale i.MX ASRC blocks, Realtek
RT286 and RT5670, Rockchip RK3xxx I2S controllers and Texas Instruments
TAS2552.
- Lots of updates and fixes, especially to the DaVinci, Intel,
Freescale, Realtek, and rcar drivers.
Diffstat (limited to 'sound/soc/codecs/cs42l52.c')
| -rw-r--r-- | sound/soc/codecs/cs42l52.c | 14 |
1 files changed, 7 insertions, 7 deletions
diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 071fc77f2f06..969167d8b71e 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -399,15 +399,15 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_MASTERB_VOL, 0, 0x34, 0xE4, hl_tlv), SOC_DOUBLE_R_SX_TLV("Headphone Volume", CS42L52_HPA_VOL, - CS42L52_HPB_VOL, 0, 0x34, 0xCC, hpd_tlv), + CS42L52_HPB_VOL, 0, 0x34, 0xC0, hpd_tlv), SOC_ENUM("Headphone Analog Gain", hp_gain_enum), SOC_DOUBLE_R_SX_TLV("Speaker Volume", CS42L52_SPKA_VOL, - CS42L52_SPKB_VOL, 0, 0x1, 0xff, hl_tlv), + CS42L52_SPKB_VOL, 0, 0x40, 0xC0, hl_tlv), SOC_DOUBLE_R_SX_TLV("Bypass Volume", CS42L52_PASSTHRUA_VOL, - CS42L52_PASSTHRUB_VOL, 6, 0x18, 0x90, pga_tlv), + CS42L52_PASSTHRUB_VOL, 0, 0x88, 0x90, pga_tlv), SOC_DOUBLE("Bypass Mute", CS42L52_MISC_CTL, 4, 5, 1, 0), @@ -417,10 +417,10 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { SOC_ENUM("MIC Bias Level", mic_bias_level_enum), SOC_DOUBLE_R_SX_TLV("ADC Volume", CS42L52_ADCA_VOL, - CS42L52_ADCB_VOL, 7, 0x80, 0xA0, ipd_tlv), + CS42L52_ADCB_VOL, 0, 0xA0, 0x78, ipd_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L52_ADCA_MIXER_VOL, CS42L52_ADCB_MIXER_VOL, - 6, 0x7f, 0x19, ipd_tlv), + 0, 0x19, 0x7F, ipd_tlv), SOC_DOUBLE("ADC Switch", CS42L52_ADC_MISC_CTL, 0, 1, 1, 0), @@ -428,11 +428,11 @@ static const struct snd_kcontrol_new cs42l52_snd_controls[] = { CS42L52_ADCB_MIXER_VOL, 7, 1, 1), SOC_DOUBLE_R_SX_TLV("PGA Volume", CS42L52_PGAA_CTL, - CS42L52_PGAB_CTL, 0, 0x28, 0x30, pga_tlv), + CS42L52_PGAB_CTL, 0, 0x28, 0x24, pga_tlv), SOC_DOUBLE_R_SX_TLV("PCM Mixer Volume", CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, - 0, 0x7f, 0x19, mix_tlv), + 0, 0x19, 0x7f, mix_tlv), SOC_DOUBLE_R("PCM Mixer Switch", CS42L52_PCMA_MIXER_VOL, CS42L52_PCMB_MIXER_VOL, 7, 1, 1), |