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authorTakashi Iwai <tiwai@suse.de>2010-05-20 12:00:43 +0200
committerTakashi Iwai <tiwai@suse.de>2010-05-20 12:00:43 +0200
commitd71f4cece4bd97d05592836202fc04ff2e7817e3 (patch)
tree6c877c7a938758b1323d9c97d46b9c536e618c69 /sound/soc/codecs/ak4642.c
parent19008bdacb9f7841166ebafe0aef361ee582ffbf (diff)
parentad8332c1302bcb4f80d593fd3eb477be9d7f5604 (diff)
downloadop-kernel-dev-d71f4cece4bd97d05592836202fc04ff2e7817e3.zip
op-kernel-dev-d71f4cece4bd97d05592836202fc04ff2e7817e3.tar.gz
Merge branch 'topic/asoc' into for-linus
Conflicts: sound/soc/codecs/ad1938.c
Diffstat (limited to 'sound/soc/codecs/ak4642.c')
-rw-r--r--sound/soc/codecs/ak4642.c177
1 files changed, 141 insertions, 36 deletions
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c
index 729859c..7528a54 100644
--- a/sound/soc/codecs/ak4642.c
+++ b/sound/soc/codecs/ak4642.c
@@ -81,12 +81,39 @@
#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)
+#define PLL1 (1 << 5)
+#define PLL0 (1 << 4)
+#define PLL_MASK (PLL3 | PLL2 | PLL1 | PLL0)
+
+#define BCKO_MASK (1 << 3)
+#define BCKO_64 BCKO_MASK
+
+/* MD_CTL2 */
+#define FS0 (1 << 0)
+#define FS1 (1 << 1)
+#define FS2 (1 << 2)
+#define FS3 (1 << 5)
+#define FS_MASK (FS0 | FS1 | FS2 | FS3)
+
struct snd_soc_codec_device soc_codec_dev_ak4642;
/* codec private data */
struct ak4642_priv {
struct snd_soc_codec codec;
- unsigned int sysclk;
};
static struct snd_soc_codec *ak4642_codec;
@@ -177,17 +204,12 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
*
* PLL, Master Mode
* Audio I/F Format :MSB justified (ADC & DAC)
- * Sampling Frequency: 44.1kHz
- * Digital Volume: −8dB
+ * Digital Volume: -8dB
* Bass Boost Level : Middle
*
* 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);
ak4642_write(codec, 0x0e, 0x19);
ak4642_write(codec, 0x09, 0x91);
@@ -195,15 +217,14 @@ 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
*
* PLL Master Mode
* Audio I/F Format:MSB justified (ADC & DAC)
- * Sampling Frequency:44.1kHz
* Pre MIC AMP:+20dB
* MIC Power On
* ALC setting:Refer to Table 35
@@ -212,7 +233,6 @@ 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, 0x05, 0x27);
ak4642_write(codec, 0x02, 0x05);
ak4642_write(codec, 0x06, 0x3c);
ak4642_write(codec, 0x08, 0xe1);
@@ -233,8 +253,8 @@ 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);
@@ -250,9 +270,111 @@ static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai,
int clk_id, unsigned int freq, int dir)
{
struct snd_soc_codec *codec = codec_dai->codec;
- struct ak4642_priv *ak4642 = codec->private_data;
+ u8 pll;
+
+ switch (freq) {
+ case 11289600:
+ pll = PLL2;
+ break;
+ case 12288000:
+ pll = PLL2 | PLL0;
+ break;
+ case 12000000:
+ pll = PLL2 | PLL1;
+ break;
+ case 24000000:
+ pll = PLL2 | PLL1 | PLL0;
+ break;
+ case 13500000:
+ pll = PLL3 | PLL2;
+ break;
+ case 27000000:
+ pll = PLL3 | PLL2 | PLL0;
+ break;
+ default:
+ return -EINVAL;
+ }
+ snd_soc_update_bits(codec, MD_CTL1, PLL_MASK, pll);
+
+ 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 int ak4642_dai_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *params,
+ struct snd_soc_dai *dai)
+{
+ struct snd_soc_codec *codec = dai->codec;
+ u8 rate;
+
+ switch (params_rate(params)) {
+ case 7350:
+ rate = FS2;
+ break;
+ case 8000:
+ rate = 0;
+ break;
+ case 11025:
+ rate = FS2 | FS0;
+ break;
+ case 12000:
+ rate = FS0;
+ break;
+ case 14700:
+ rate = FS2 | FS1;
+ break;
+ case 16000:
+ rate = FS1;
+ break;
+ case 22050:
+ rate = FS2 | FS1 | FS0;
+ break;
+ case 24000:
+ rate = FS1 | FS0;
+ break;
+ case 29400:
+ rate = FS3 | FS2 | FS1;
+ break;
+ case 32000:
+ rate = FS3 | FS1;
+ break;
+ case 44100:
+ rate = FS3 | FS2 | FS1 | FS0;
+ break;
+ case 48000:
+ rate = FS3 | FS1 | FS0;
+ break;
+ default:
+ return -EINVAL;
+ break;
+ }
+ snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate);
- ak4642->sysclk = freq;
return 0;
}
@@ -260,6 +382,8 @@ 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,
+ .hw_params = ak4642_dai_hw_params,
};
struct snd_soc_dai ak4642_dai = {
@@ -277,6 +401,7 @@ struct snd_soc_dai ak4642_dai = {
.rates = SNDRV_PCM_RATE_8000_48000,
.formats = SNDRV_PCM_FMTBIT_S16_LE },
.ops = &ak4642_dai_ops,
+ .symmetric_rates = 1,
};
EXPORT_SYMBOL_GPL(ak4642_dai);
@@ -307,7 +432,7 @@ static int ak4642_init(struct ak4642_priv *ak4642)
INIT_LIST_HEAD(&codec->dapm_widgets);
INIT_LIST_HEAD(&codec->dapm_paths);
- codec->private_data = ak4642;
+ snd_soc_codec_set_drvdata(codec, ak4642);
codec->name = "AK4642";
codec->owner = THIS_MODULE;
codec->read = ak4642_read_reg_cache;
@@ -338,26 +463,6 @@ 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
- * Input Master Clock Select at PLL Mode: 11.2896MHz
- * 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, 0x04, 0x4a);
- ak4642_write(codec, 0x05, 0x27);
- ak4642_write(codec, 0x00, 0x40);
- ak4642_write(codec, 0x01, 0x0b);
-
return ret;
reg_cache_err:
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