[PATCH 2/3] ASoC: ak4642: Add set_fmt function for snd_soc_dai_ops

* [PATCH 2/3] ASoC: ak4642: Add set_fmt function for snd_soc_dai_ops
  2010-03-23  7:27 [PATCH 0/3] ASoC: ak4642: sampling rate update Kuninori Morimoto
@ 2010-03-15  9:10 ` Kuninori Morimoto
  2010-03-23 15:36   ` Liam Girdwood
  2010-03-23  7:27 ` [PATCH 1/3] ASoC: ak4642: Add pll select support Kuninori Morimoto
                   ` (2 subsequent siblings)
  3 siblings, 1 reply; 8+ messages in thread
From: Kuninori Morimoto @ 2010-03-15  9:10 UTC (permalink / raw)
  To: Mark Brown; +Cc: Linux-ALSA

Signed-off-by: Kuninori Morimoto <morimoto.kuninori@renesas.com>
---
 sound/soc/codecs/ak4642.c |   69 +++++++++++++++++++++++++++++---------------
 sound/soc/sh/fsi-ak4642.c |    4 ++
 2 files changed, 49 insertions(+), 24 deletions(-)

diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index d5bd4ca..3452bd7 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -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:
diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c
index c0207dc..be01854 100644
--- a/sound/soc/sh/fsi-ak4642.c
+++ b/sound/soc/sh/fsi-ak4642.c
@@ -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;
-- 
1.6.3.3

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