ASoC: ak4642: Add set_fmt function for snd_soc_dai_ops
Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com> Acked-by: Liam Girdwood <lrg@slimlogic.co.uk> Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
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4b6316b4b1
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0643ce8f42
2 changed files with 49 additions and 24 deletions
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@ -80,6 +80,17 @@
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#define AK4642_CACHEREGNUM 0x25
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/* PW_MGMT2 */
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#define HPMTN (1 << 6)
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#define PMHPL (1 << 5)
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#define PMHPR (1 << 4)
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#define MS (1 << 3) /* master/slave select */
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#define MCKO (1 << 1)
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#define PMPLL (1 << 0)
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#define PMHP_MASK (PMHPL | PMHPR)
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#define PMHP PMHP_MASK
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/* MD_CTL1 */
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#define PLL3 (1 << 7)
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#define PLL2 (1 << 6)
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@ -87,6 +98,9 @@
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#define PLL0 (1 << 4)
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#define PLL_MASK (PLL3 | PLL2 | PLL1 | PLL0)
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#define BCKO_MASK (1 << 3)
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#define BCKO_64 BCKO_MASK
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struct snd_soc_codec_device soc_codec_dev_ak4642;
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/* codec private data */
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@ -188,9 +202,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
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*
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* This operation came from example code of
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* "ASAHI KASEI AK4642" (japanese) manual p97.
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*
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* Example code use 0x39, 0x79 value for 0x01 address,
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* But we need MCKO (0x02) bit now
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*/
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ak4642_write(codec, 0x05, 0x27);
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ak4642_write(codec, 0x0f, 0x09);
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@ -200,8 +211,8 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
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ak4642_write(codec, 0x0a, 0x28);
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ak4642_write(codec, 0x0d, 0x28);
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ak4642_write(codec, 0x00, 0x64);
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ak4642_write(codec, 0x01, 0x3b); /* + MCKO bit */
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ak4642_write(codec, 0x01, 0x7b); /* + MCKO bit */
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snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, PMHP);
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snd_soc_update_bits(codec, PW_MGMT2, HPMTN, HPMTN);
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} else {
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/*
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* start stereo input
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@ -238,8 +249,8 @@ static void ak4642_dai_shutdown(struct snd_pcm_substream *substream,
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if (is_play) {
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/* stop headphone output */
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ak4642_write(codec, 0x01, 0x3b);
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ak4642_write(codec, 0x01, 0x0b);
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snd_soc_update_bits(codec, PW_MGMT2, HPMTN, 0);
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snd_soc_update_bits(codec, PW_MGMT2, PMHP_MASK, 0);
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ak4642_write(codec, 0x00, 0x40);
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ak4642_write(codec, 0x0e, 0x11);
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ak4642_write(codec, 0x0f, 0x08);
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@ -284,10 +295,37 @@ static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai,
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return 0;
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}
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static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt)
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{
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struct snd_soc_codec *codec = dai->codec;
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u8 data;
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u8 bcko;
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data = MCKO | PMPLL; /* use MCKO */
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bcko = 0;
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/* set master/slave audio interface */
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switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
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case SND_SOC_DAIFMT_CBM_CFM:
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data |= MS;
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bcko = BCKO_64;
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break;
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case SND_SOC_DAIFMT_CBS_CFS:
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break;
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default:
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return -EINVAL;
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}
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snd_soc_update_bits(codec, PW_MGMT2, MS, data);
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snd_soc_update_bits(codec, MD_CTL1, BCKO_MASK, bcko);
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return 0;
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}
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static struct snd_soc_dai_ops ak4642_dai_ops = {
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.startup = ak4642_dai_startup,
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.shutdown = ak4642_dai_shutdown,
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.set_sysclk = ak4642_dai_set_sysclk,
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.set_fmt = ak4642_dai_set_fmt,
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};
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struct snd_soc_dai ak4642_dai = {
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@ -366,23 +404,6 @@ static int ak4642_init(struct ak4642_priv *ak4642)
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goto reg_cache_err;
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}
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/*
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* clock setting
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*
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* Audio I/F Format: MSB justified (ADC & DAC)
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* BICK frequency at Master Mode: 64fs
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* MCKO: Enable
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* Sampling Frequency: 44.1kHz
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*
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* This operation came from example code of
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* "ASAHI KASEI AK4642" (japanese) manual p89.
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*
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* please fix-me
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*/
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ak4642_write(codec, 0x01, 0x08);
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ak4642_write(codec, 0x05, 0x27);
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ak4642_write(codec, 0x04, 0x0a);
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return ret;
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reg_cache_err:
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@ -26,6 +26,10 @@ static int fsi_ak4642_dai_init(struct snd_soc_codec *codec)
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{
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int ret;
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ret = snd_soc_dai_set_fmt(&ak4642_dai, SND_SOC_DAIFMT_CBM_CFM);
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if (ret < 0)
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return ret;
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ret = snd_soc_dai_set_sysclk(&ak4642_dai, 0, 11289600, 0);
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return ret;
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