diff options
author | Kuninori Morimoto <kuninori.morimoto.gx@renesas.com> | 2010-10-15 14:23:18 +0900 |
---|---|---|
committer | Mark Brown <broonie@opensource.wolfsonmicro.com> | 2010-10-15 11:54:51 +0100 |
commit | a34712391a66260e442a9ab1eb7edb22a2d0ca3c (patch) | |
tree | cda8ad5d806d0bfee805542459482227c7794621 /sound/soc | |
parent | c14c05c19f2a2ab87b8ebabd245f53945a97695b (diff) |
ASoC: ak4642: make sure name of register/value
This patch replace magic code with defined name,
and remove unnecessary settings which set default value
Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx@renesas.com>
Acked-by: Liam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Diffstat (limited to 'sound/soc')
-rw-r--r-- | sound/soc/codecs/ak4642.c | 64 |
1 files changed, 46 insertions, 18 deletions
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 009068f57375..90c90b7f4a2e 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -72,6 +72,12 @@ #define AK4642_CACHEREGNUM 0x25 +/* PW_MGMT1*/ +#define PMVCM (1 << 6) /* VCOM Power Management */ +#define PMMIN (1 << 5) /* MIN Input Power Management */ +#define PMDAC (1 << 2) /* DAC Power Management */ +#define PMADL (1 << 0) /* MIC Amp Lch and ADC Lch Power Management */ + /* PW_MGMT2 */ #define HPMTN (1 << 6) #define PMHPL (1 << 5) @@ -83,6 +89,23 @@ #define PMHP_MASK (PMHPL | PMHPR) #define PMHP PMHP_MASK +/* PW_MGMT3 */ +#define PMADR (1 << 0) /* MIC L / ADC R Power Management */ + +/* SG_SL1 */ +#define MINS (1 << 6) /* Switch from MIN to Speaker */ +#define DACL (1 << 4) /* Switch from DAC to Stereo or Receiver */ +#define PMMP (1 << 2) /* MPWR pin Power Management */ +#define MGAIN0 (1 << 0) /* MIC amp gain*/ + +/* TIMER */ +#define ZTM(param) ((param & 0x3) << 4) /* ALC Zoro Crossing TimeOut */ +#define WTM(param) (((param & 0x4) << 4) | ((param & 0x3) << 2)) + +/* ALC_CTL1 */ +#define ALC (1 << 5) /* ALC Enable */ +#define LMTH0 (1 << 0) /* ALC Limiter / Recovery Level */ + /* MD_CTL1 */ #define PLL3 (1 << 7) #define PLL2 (1 << 6) @@ -100,6 +123,11 @@ #define FS3 (1 << 5) #define FS_MASK (FS0 | FS1 | FS2 | FS3) +/* MD_CTL3 */ +#define BST1 (1 << 3) + +/* MD_CTL4 */ +#define DACH (1 << 0) /* * Playback Volume (table 39) @@ -216,11 +244,12 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p97. */ - ak4642_write(codec, 0x0f, 0x09); - ak4642_write(codec, 0x0e, 0x19); - ak4642_write(codec, 0x09, 0x91); - ak4642_write(codec, 0x0c, 0x91); - snd_soc_update_bits(codec, 0x00, 0x64, 0x64); + snd_soc_update_bits(codec, MD_CTL4, DACH, DACH); + snd_soc_update_bits(codec, MD_CTL3, BST1, BST1); + ak4642_write(codec, L_IVC, 0x91); /* volume */ + ak4642_write(codec, R_IVC, 0x91); /* volume */ + snd_soc_update_bits(codec, PW_MGMT1, PMVCM | PMMIN | PMDAC, + PMVCM | PMMIN | PMDAC); snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP); snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN); } else { @@ -237,13 +266,12 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream, * This operation came from example code of * "ASAHI KASEI AK4642" (japanese) manual p94. */ - ak4642_write(codec, 0x02, 0x05); - ak4642_write(codec, 0x06, 0x3c); - ak4642_write(codec, 0x08, 0xe1); - ak4642_write(codec, 0x0b, 0x00); - ak4642_write(codec, 0x07, 0x21); - snd_soc_update_bits(codec, 0x00, 0x41, 0x41); - ak4642_write(codec, 0x10, 0x01); + ak4642_write(codec, SG_SL1, PMMP | MGAIN0); + ak4642_write(codec, TIMER, ZTM(0x3) | WTM(0x3)); + ak4642_write(codec, ALC_CTL1, ALC | LMTH0); + snd_soc_update_bits(codec, PW_MGMT1, PMVCM | PMADL, + PMVCM | PMADL); + snd_soc_update_bits(codec, PW_MGMT3, PMADR, PMADR); } return 0; @@ -259,14 +287,14 @@ static void ak4642_dai_shutdown(struct snd_pcm_substream *substream, /* stop headphone output */ snd_soc_update_bits(codec, PW_MGMT2, HPMTN, 0); snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, 0); - snd_soc_update_bits(codec, 0x00, 0x64, 0x40); - ak4642_write(codec, 0x0e, 0x11); - ak4642_write(codec, 0x0f, 0x08); + snd_soc_update_bits(codec, PW_MGMT1, PMMIN | PMDAC, 0); + snd_soc_update_bits(codec, MD_CTL3, BST1, 0); + snd_soc_update_bits(codec, MD_CTL4, DACH, 0); } else { /* stop stereo input */ - snd_soc_update_bits(codec, 0x00, 0x41, 0x40); - ak4642_write(codec, 0x10, 0x00); - ak4642_write(codec, 0x07, 0x01); + snd_soc_update_bits(codec, PW_MGMT1, PMADL, 0); + snd_soc_update_bits(codec, PW_MGMT3, PMADR, 0); + snd_soc_update_bits(codec, ALC_CTL1, ALC, 0); } } |