Merge branch 'topic/asoc' into for-linus
Conflicts: sound/soc/codecs/ad1938.c
Bu işleme şunda yer alıyor:
Takashi Iwai
@@ -81,12 +81,39 @@
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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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#define PLL1 (1 << 5)
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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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/* MD_CTL2 */
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#define FS0 (1 << 0)
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#define FS1 (1 << 1)
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#define FS2 (1 << 2)
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#define FS3 (1 << 5)
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#define FS_MASK (FS0 | FS1 | FS2 | FS3)
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struct snd_soc_codec_device soc_codec_dev_ak4642;
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/* codec private data */
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struct ak4642_priv {
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struct snd_soc_codec codec;
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unsigned int sysclk;
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};
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static struct snd_soc_codec *ak4642_codec;
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@@ -177,17 +204,12 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
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*
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* PLL, Master Mode
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* Audio I/F Format :MSB justified (ADC & DAC)
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* Sampling Frequency: 44.1kHz
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* Digital Volume: −8dB
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* Digital Volume: -8dB
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* Bass Boost Level : Middle
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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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ak4642_write(codec, 0x0e, 0x19);
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ak4642_write(codec, 0x09, 0x91);
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@@ -195,15 +217,14 @@ 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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*
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* PLL Master Mode
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* Audio I/F Format:MSB justified (ADC & DAC)
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* Sampling Frequency:44.1kHz
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* Pre MIC AMP:+20dB
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* MIC Power On
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* ALC setting:Refer to Table 35
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@@ -212,7 +233,6 @@ static int ak4642_dai_startup(struct snd_pcm_substream *substream,
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* This operation came from example code of
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* "ASAHI KASEI AK4642" (japanese) manual p94.
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*/
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ak4642_write(codec, 0x05, 0x27);
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ak4642_write(codec, 0x02, 0x05);
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ak4642_write(codec, 0x06, 0x3c);
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ak4642_write(codec, 0x08, 0xe1);
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@@ -233,8 +253,8 @@ 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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@@ -250,9 +270,111 @@ static int ak4642_dai_set_sysclk(struct snd_soc_dai *codec_dai,
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int clk_id, unsigned int freq, int dir)
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{
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struct snd_soc_codec *codec = codec_dai->codec;
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struct ak4642_priv *ak4642 = codec->private_data;
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u8 pll;
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switch (freq) {
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case 11289600:
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pll = PLL2;
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break;
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case 12288000:
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pll = PLL2 | PLL0;
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break;
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case 12000000:
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pll = PLL2 | PLL1;
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break;
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case 24000000:
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pll = PLL2 | PLL1 | PLL0;
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break;
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case 13500000:
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pll = PLL3 | PLL2;
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break;
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case 27000000:
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pll = PLL3 | PLL2 | PLL0;
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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, MD_CTL1, PLL_MASK, pll);
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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 int ak4642_dai_hw_params(struct snd_pcm_substream *substream,
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struct snd_pcm_hw_params *params,
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struct snd_soc_dai *dai)
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{
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struct snd_soc_codec *codec = dai->codec;
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u8 rate;
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switch (params_rate(params)) {
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case 7350:
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rate = FS2;
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break;
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case 8000:
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rate = 0;
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break;
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case 11025:
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rate = FS2 | FS0;
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break;
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case 12000:
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rate = FS0;
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break;
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case 14700:
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rate = FS2 | FS1;
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break;
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case 16000:
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rate = FS1;
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break;
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case 22050:
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rate = FS2 | FS1 | FS0;
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break;
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case 24000:
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rate = FS1 | FS0;
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break;
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case 29400:
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rate = FS3 | FS2 | FS1;
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break;
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case 32000:
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rate = FS3 | FS1;
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break;
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case 44100:
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rate = FS3 | FS2 | FS1 | FS0;
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break;
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case 48000:
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rate = FS3 | FS1 | FS0;
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break;
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default:
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return -EINVAL;
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break;
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}
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snd_soc_update_bits(codec, MD_CTL2, FS_MASK, rate);
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ak4642->sysclk = freq;
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return 0;
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}
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@@ -260,6 +382,8 @@ 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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.hw_params = ak4642_dai_hw_params,
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};
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struct snd_soc_dai ak4642_dai = {
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@@ -277,6 +401,7 @@ struct snd_soc_dai ak4642_dai = {
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.rates = SNDRV_PCM_RATE_8000_48000,
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.formats = SNDRV_PCM_FMTBIT_S16_LE },
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.ops = &ak4642_dai_ops,
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.symmetric_rates = 1,
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};
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EXPORT_SYMBOL_GPL(ak4642_dai);
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@@ -307,7 +432,7 @@ static int ak4642_init(struct ak4642_priv *ak4642)
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INIT_LIST_HEAD(&codec->dapm_widgets);
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INIT_LIST_HEAD(&codec->dapm_paths);
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codec->private_data = ak4642;
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snd_soc_codec_set_drvdata(codec, ak4642);
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codec->name = "AK4642";
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codec->owner = THIS_MODULE;
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codec->read = ak4642_read_reg_cache;
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@@ -338,26 +463,6 @@ 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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* Input Master Clock Select at PLL Mode: 11.2896MHz
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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, 0x04, 0x4a);
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ak4642_write(codec, 0x05, 0x27);
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ak4642_write(codec, 0x00, 0x40);
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ak4642_write(codec, 0x01, 0x0b);
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return ret;
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reg_cache_err:
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