提交 0643ce8f 编写于 作者: K Kuninori Morimoto 提交者: Mark Brown

ASoC: ak4642: Add set_fmt function for snd_soc_dai_ops

Signed-off-by: NKuninori Morimoto <morimoto.kuninori@renesas.com>
Acked-by: NLiam Girdwood <lrg@slimlogic.co.uk>
Signed-off-by: NMark Brown <broonie@opensource.wolfsonmicro.com>
上级 4b6316b4
......@@ -80,6 +80,17 @@
#define AK4642_CACHEREGNUM 0x25
/* PW_MGMT2 */
#define HPMTN (1 << 6)
#define PMHPL (1 << 5)
#define PMHPR (1 << 4)
#define MS (1 << 3) /* master/slave select */
#define MCKO (1 << 1)
#define PMPLL (1 << 0)
#define PMHP_MASK (PMHPL | PMHPR)
#define PMHP PMHP_MASK
/* MD_CTL1 */
#define PLL3 (1 << 7)
#define PLL2 (1 << 6)
......@@ -87,6 +98,9 @@
#define PLL0 (1 << 4)
#define PLL_MASK (PLL3 | PLL2 | PLL1 | PLL0)
#define BCKO_MASK (1 << 3)
#define BCKO_64 BCKO_MASK
struct snd_soc_codec_device soc_codec_dev_ak4642;
/* codec private data */
......@@ -188,9 +202,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
*
* This operation came from example code of
* "ASAHI KASEI AK4642" (japanese) manual p97.
*
* Example code use 0x39, 0x79 value for 0x01 address,
* But we need MCKO (0x02) bit now
*/
ak4642_write(codec, 0x05, 0x27);
ak4642_write(codec, 0x0f, 0x09);
......@@ -200,8 +211,8 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
ak4642_write(codec, 0x0a, 0x28);
ak4642_write(codec, 0x0d, 0x28);
ak4642_write(codec, 0x00, 0x64);
ak4642_write(codec, 0x01, 0x3b); /* + MCKO bit */
ak4642_write(codec, 0x01, 0x7b); /* + MCKO bit */
snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP);
snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN);
} else {
/*
* start stereo input
......@@ -238,8 +249,8 @@ static void ak4642_dai_shutdown(struct snd_pcm_substream *substream,
if (is_play) {
/* stop headphone output */
ak4642_write(codec, 0x01, 0x3b);
ak4642_write(codec, 0x01, 0x0b);
snd_soc_update_bits(codec, PW_MGMT2, HPMTN, 0);
snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, 0);
ak4642_write(codec, 0x00, 0x40);
ak4642_write(codec, 0x0e, 0x11);
ak4642_write(codec, 0x0f, 0x08);
......@@ -284,10 +295,37 @@ static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai,
return 0;
}
static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
{
struct snd_soc_codec *codec = dai->codec;
u8 data;
u8 bcko;
data = MCKO | PMPLL; /* use MCKO */
bcko = 0;
/* set master/slave audio interface */
switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
case SND_SOC_DAIFMT_CBM_CFM:
data |= MS;
bcko = BCKO_64;
break;
case SND_SOC_DAIFMT_CBS_CFS:
break;
default:
return -EINVAL;
}
snd_soc_update_bits(codec, PW_MGMT2, MS, data);
snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko);
return 0;
}
static struct snd_soc_dai_ops ak4642_dai_ops = {
.startup = ak4642_dai_startup,
.shutdown = ak4642_dai_shutdown,
.set_sysclk = ak4642_dai_set_sysclk,
.set_fmt = ak4642_dai_set_fmt,
};
struct snd_soc_dai ak4642_dai = {
......@@ -366,23 +404,6 @@ static int ak4642_init(struct ak4642_priv *ak4642)
goto reg_cache_err;
}
/*
* clock setting
*
* Audio I/F Format: MSB justified (ADC & DAC)
* BICK frequency at Master Mode: 64fs
* MCKO: Enable
* Sampling Frequency: 44.1kHz
*
* This operation came from example code of
* "ASAHI KASEI AK4642" (japanese) manual p89.
*
* please fix-me
*/
ak4642_write(codec, 0x01, 0x08);
ak4642_write(codec, 0x05, 0x27);
ak4642_write(codec, 0x04, 0x0a);
return ret;
reg_cache_err:
......
......@@ -26,6 +26,10 @@ static int fsi_ak4642_dai_init(struct snd_soc_codec *codec)
{
int ret;
ret = snd_soc_dai_set_fmt(&ak4642_dai, SND_SOC_DAIFMT_CBM_CFM);
if (ret < 0)
return ret;
ret = snd_soc_dai_set_sysclk(&ak4642_dai, 0, 11289600, 0);
return ret;
......
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