From: Kuninori Morimoto <morimoto.kuninori@renesas.com>
To: Mark Brown <broonie@opensource.wolfsonmicro.com>
Cc: Linux-ALSA <alsa-devel@alsa-project.org>
Subject: [PATCH 2/3] ASoC: ak4642: Add set_fmt function for snd_soc_dai_ops
Date: Mon, 15 Mar 2010 18:10:50 +0900 [thread overview]
Message-ID: <usk7r4hwa.wl%morimoto.kuninori@renesas.com> (raw)
In-Reply-To: <uvdcn4hx0.wl%morimoto.kuninori@renesas.com>
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
next prev parent reply other threads:[~2010-03-23 7:27 UTC|newest]
Thread overview: 8+ messages / expand[flat|nested] mbox.gz Atom feed top
2010-03-23 7:27 [PATCH 0/3] ASoC: ak4642: sampling rate update Kuninori Morimoto
2010-03-15 9:10 ` Kuninori Morimoto [this message]
2010-03-23 15:36 ` [PATCH 2/3] ASoC: ak4642: Add set_fmt function for snd_soc_dai_ops Liam Girdwood
2010-03-23 7:27 ` [PATCH 1/3] ASoC: ak4642: Add pll select support Kuninori Morimoto
2010-03-23 10:10 ` Liam Girdwood
2010-03-23 7:27 ` [PATCH 3/3] ASoC: ak4642: Add enhanced sampling rate Kuninori Morimoto
2010-03-23 10:10 ` Liam Girdwood
2010-03-23 15:38 ` [PATCH 0/3] ASoC: ak4642: sampling rate update Mark Brown
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