[alsa-devel] [PATCH 0/3] ASoC: ak4642: sampling rate update
Dear Mark
These patches are sampling rate update for ak4642
Kuninori Morimoto (3): ASoC: ak4642: Add pll select support ASoC: ak4642: Add set_fmt function for snd_soc_dai_ops ASoC: ak4642: Add enhanced sampling rate
It are tested on ms7724se/AP4 board.
Best regards -- Kuninori Morimoto
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;
On Mon, 2010-03-15 at 18:10 +0900, Kuninori Morimoto wrote:
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(-)
Acked-by: Liam Girdwood lrg@slimlogic.co.uk
Current ak4642 was not able to select pll. This patch add support it. It still expect PLL base input pin is MCKI. see Table 5 "setting of PLL Mode" of datasheet
Signed-off-by: Kuninori Morimoto morimoto.kuninori@renesas.com --- sound/soc/codecs/ak4642.c | 40 +++++++++++++++++++++++++++++++++------- sound/soc/sh/fsi-ak4642.c | 10 ++++++++++ 2 files changed, 43 insertions(+), 7 deletions(-)
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 3ef16bb..d5bd4ca 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -80,12 +80,18 @@
#define AK4642_CACHEREGNUM 0x25
+/* 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) + 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; @@ -249,9 +255,32 @@ 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);
- ak4642->sysclk = freq; return 0; }
@@ -342,7 +371,6 @@ static int ak4642_init(struct ak4642_priv *ak4642) * * 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 * @@ -352,10 +380,8 @@ static int ak4642_init(struct ak4642_priv *ak4642) * 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); + ak4642_write(codec, 0x04, 0x0a);
return ret;
diff --git a/sound/soc/sh/fsi-ak4642.c b/sound/soc/sh/fsi-ak4642.c index 5263ab1..c0207dc 100644 --- a/sound/soc/sh/fsi-ak4642.c +++ b/sound/soc/sh/fsi-ak4642.c @@ -22,11 +22,21 @@ #include <sound/sh_fsi.h> #include <../sound/soc/codecs/ak4642.h>
+static int fsi_ak4642_dai_init(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dai_set_sysclk(&ak4642_dai, 0, 11289600, 0); + + return ret; +} + static struct snd_soc_dai_link fsi_dai_link = { .name = "AK4642", .stream_name = "AK4642", .cpu_dai = &fsi_soc_dai[0], /* fsi */ .codec_dai = &ak4642_dai, + .init = fsi_ak4642_dai_init, .ops = NULL, };
On Tue, 2010-03-23 at 16:27 +0900, Kuninori Morimoto wrote:
Current ak4642 was not able to select pll. This patch add support it. It still expect PLL base input pin is MCKI. see Table 5 "setting of PLL Mode" of datasheet
Signed-off-by: Kuninori Morimoto morimoto.kuninori@renesas.com
Acked-by: Liam Girdwood lrg@slimlogic.co.uk
Signed-off-by: Kuninori Morimoto morimoto.kuninori@renesas.com --- sound/soc/codecs/ak4642.c | 68 +++++++++++++++++++++++++++++++++++++++++--- 1 files changed, 63 insertions(+), 5 deletions(-)
diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 3452bd7..de1809d 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -101,6 +101,13 @@ #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 */ @@ -196,14 +203,12 @@ 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. */ - ak4642_write(codec, 0x05, 0x27); ak4642_write(codec, 0x0f, 0x09); ak4642_write(codec, 0x0e, 0x19); ak4642_write(codec, 0x09, 0x91); @@ -219,7 +224,6 @@ 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 * Pre MIC AMP:+20dB * MIC Power On * ALC setting:Refer to Table 35 @@ -228,7 +232,6 @@ 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); @@ -321,11 +324,65 @@ static int ak4642_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) 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); + + 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, + .hw_params = ak4642_dai_hw_params, };
struct snd_soc_dai ak4642_dai = { @@ -343,6 +400,7 @@ 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);
On Tue, 2010-03-23 at 16:27 +0900, Kuninori Morimoto wrote:
Signed-off-by: Kuninori Morimoto morimoto.kuninori@renesas.com
sound/soc/codecs/ak4642.c | 68 +++++++++++++++++++++++++++++++++++++++++--- 1 files changed, 63 insertions(+), 5 deletions(-)
Acked-by: Liam Girdwood lrg@slimlogic.co.uk
participants (3)
-
Kuninori Morimoto -
Liam Girdwood -
Mark Brown