[alsa-devel] [PATCH 1/1] ASoC: TWL4030: Add support Voice DAI
Add Voice DAI to support the PCM voice interface of the twl4030 codec.
The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz (voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX.
If the system master clock is not 26MHz, the voice PCM interface is not available.
The PCM voice interface has two modes - PCM mode1 : This uses the rising edge of the clock signal - PCM mode2 : This uses the falling edge of the clock signal
PCM mode1 and mode2 have a look of DSP_A and DSP_B, so we used DSP_A and DSP_B.
Signed-off-by: Joonyoung Shim jy0922.shim@samsung.com --- sound/soc/codecs/twl4030.c | 135 +++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/twl4030.h | 16 +++++- sound/soc/omap/omap2evm.c | 2 +- sound/soc/omap/omap3beagle.c | 2 +- sound/soc/omap/omap3pandora.c | 4 +- sound/soc/omap/overo.c | 2 +- sound/soc/omap/sdp3430.c | 2 +- 7 files changed, 150 insertions(+), 13 deletions(-)
diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index db7fecf..13461d1 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -1437,6 +1437,107 @@ static int twl4030_set_dai_fmt(struct snd_soc_dai *codec_dai, return 0; }
+static int twl4030_voice_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u8 infreq; + + /* If the system master clock is not 26MHz, the voice PCM interface is + * not avilable. + */ + infreq = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL) + & TWL4030_APLL_INFREQ; + + if (infreq != TWL4030_APLL_INFREQ_26000KHZ) + return -EPERM; + + return 0; +} + +static int twl4030_voice_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + u8 old_mode, mode; + + /* bit rate */ + old_mode = twl4030_read_reg_cache(codec, + TWL4030_REG_CODEC_MODE) & ~TWL4030_CODECPDZ; + mode = old_mode; + + switch (params_rate(params)) { + case 8000: + mode &= ~(TWL4030_SEL_16K); + break; + case 16000: + mode |= TWL4030_SEL_16K; + break; + default: + printk(KERN_ERR "TWL4030 voice hw params: unknown rate %d\n", + params_rate(params)); + return -EINVAL; + } + + if (mode != old_mode) { + /* change rate and set CODECPDZ */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode); + twl4030_codec_enable(codec, 1); + } + + return 0; +} + +static int twl4030_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 old_format, format; + + /* get format */ + old_format = twl4030_read_reg_cache(codec, TWL4030_REG_VOICE_IF); + format = old_format; + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFM: + format &= ~(TWL4030_VIF_SLAVE_EN); + break; + case SND_SOC_DAIFMT_CBS_CFS: + format |= TWL4030_VIF_SLAVE_EN; + break; + default: + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_DSP_A: + format &= ~(TWL4030_VIF_FORMAT); + break; + case SND_SOC_DAIFMT_DSP_B: + format |= TWL4030_VIF_FORMAT; + break; + default: + return -EINVAL; + } + + if (format != old_format) { + /* change format and set CODECPDZ */ + twl4030_codec_enable(codec, 0); + twl4030_write(codec, TWL4030_REG_VOICE_IF, format); + twl4030_codec_enable(codec, 1); + } + + return 0; +} + #define TWL4030_RATES (SNDRV_PCM_RATE_8000_48000) #define TWL4030_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FORMAT_S24_LE)
@@ -1448,8 +1549,14 @@ static struct snd_soc_dai_ops twl4030_dai_ops = { .set_fmt = twl4030_set_dai_fmt, };
-struct snd_soc_dai twl4030_dai = { - .name = "twl4030", +static struct snd_soc_dai_ops twl4030_dai_voice_ops = { + .startup = twl4030_voice_startup, + .hw_params = twl4030_voice_hw_params, + .set_fmt = twl4030_voice_set_dai_fmt, +}; + +struct snd_soc_dai twl4030_dai[] = { +{ .name = "twl4030", .playback = { .stream_name = "Playback", .channels_min = 2, @@ -1463,6 +1570,22 @@ struct snd_soc_dai twl4030_dai = { .rates = TWL4030_RATES, .formats = TWL4030_FORMATS,}, .ops = &twl4030_dai_ops, +}, +{ .name = "twl4030 Voice", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 1, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_16000, + .formats = SNDRV_PCM_FMTBIT_S16_LE,}, + .ops = &twl4030_dai_voice_ops, +}, }; EXPORT_SYMBOL_GPL(twl4030_dai);
@@ -1503,8 +1626,8 @@ static int twl4030_init(struct snd_soc_device *socdev) codec->read = twl4030_read_reg_cache; codec->write = twl4030_write; codec->set_bias_level = twl4030_set_bias_level; - codec->dai = &twl4030_dai; - codec->num_dai = 1; + codec->dai = twl4030_dai; + codec->num_dai = ARRAY_SIZE(twl4030_dai), codec->reg_cache_size = sizeof(twl4030_reg); codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), GFP_KERNEL); @@ -1598,13 +1721,13 @@ EXPORT_SYMBOL_GPL(soc_codec_dev_twl4030);
static int __init twl4030_modinit(void) { - return snd_soc_register_dai(&twl4030_dai); + return snd_soc_register_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); } module_init(twl4030_modinit);
static void __exit twl4030_exit(void) { - snd_soc_unregister_dai(&twl4030_dai); + snd_soc_unregister_dais(&twl4030_dai[0], ARRAY_SIZE(twl4030_dai)); } module_exit(twl4030_exit);
diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index cb63765..305b8ba 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -171,6 +171,17 @@ #define TWL4030_CLK256FS_EN 0x02 #define TWL4030_AIF_EN 0x01
+/* VOICE_IF (0x0F) Fields */ + +#define TWL4030_VIF_SLAVE_EN 0x80 +#define TWL4030_VIF_DIN_EN 0x40 +#define TWL4030_VIF_DOUT_EN 0x20 +#define TWL4030_VIF_SWAP 0x10 +#define TWL4030_VIF_FORMAT 0x08 +#define TWL4030_VIF_TRI_EN 0x04 +#define TWL4030_VIF_SUB_EN 0x02 +#define TWL4030_VIF_EN 0x01 + /* EAR_CTL (0x21) */ #define TWL4030_EAR_GAIN 0x30
@@ -236,7 +247,10 @@ #define TWL4030_SMOOTH_ANAVOL_EN 0x02 #define TWL4030_DIGMIC_LR_SWAP_EN 0x01
-extern struct snd_soc_dai twl4030_dai; +#define TWL4030_DAI_HIFI 0 +#define TWL4030_DAI_VOICE 1 + +extern struct snd_soc_dai twl4030_dai[2]; extern struct snd_soc_codec_device soc_codec_dev_twl4030;
#endif /* End of __TWL4030_AUDIO_H__ */ diff --git a/sound/soc/omap/omap2evm.c b/sound/soc/omap/omap2evm.c index 0c2322d..027e1a4 100644 --- a/sound/soc/omap/omap2evm.c +++ b/sound/soc/omap/omap2evm.c @@ -86,7 +86,7 @@ static struct snd_soc_dai_link omap2evm_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap2evm_ops, };
diff --git a/sound/soc/omap/omap3beagle.c b/sound/soc/omap/omap3beagle.c index fd24a4a..6aa428e 100644 --- a/sound/soc/omap/omap3beagle.c +++ b/sound/soc/omap/omap3beagle.c @@ -83,7 +83,7 @@ static struct snd_soc_dai_link omap3beagle_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3beagle_ops, };
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c index fe282d4..ad219aa 100644 --- a/sound/soc/omap/omap3pandora.c +++ b/sound/soc/omap/omap3pandora.c @@ -228,14 +228,14 @@ static struct snd_soc_dai_link omap3pandora_dai[] = { .name = "PCM1773", .stream_name = "HiFi Out", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3pandora_out_ops, .init = omap3pandora_out_init, }, { .name = "TWL4030", .stream_name = "Line/Mic In", .cpu_dai = &omap_mcbsp_dai[1], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &omap3pandora_in_ops, .init = omap3pandora_in_init, } diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c index a72dc4e..ec4f8fd 100644 --- a/sound/soc/omap/overo.c +++ b/sound/soc/omap/overo.c @@ -83,7 +83,7 @@ static struct snd_soc_dai_link overo_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .ops = &overo_ops, };
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c index 10f1c86..1c79741 100644 --- a/sound/soc/omap/sdp3430.c +++ b/sound/soc/omap/sdp3430.c @@ -197,7 +197,7 @@ static struct snd_soc_dai_link sdp3430_dai = { .name = "TWL4030", .stream_name = "TWL4030", .cpu_dai = &omap_mcbsp_dai[0], - .codec_dai = &twl4030_dai, + .codec_dai = &twl4030_dai[TWL4030_DAI_HIFI], .init = sdp3430_twl4030_init, .ops = &sdp3430_ops, };
On Thu, Apr 09, 2009 at 10:54:37PM +0900, Joonyoung Shim wrote:
The PCM voice interface has two modes
- PCM mode1 : This uses the rising edge of the clock signal
- PCM mode2 : This uses the falling edge of the clock signal
PCM mode1 and mode2 have a look of DSP_A and DSP_B, so we used DSP_A and DSP_B.
That sounds wrong - the difference between modes A and B is an extra cycle on BCLK after LRP, not the polarity of the signal. I'd expect that it should be one or the other of these modes with the option to invert BCLK.
+static int twl4030_voice_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)+{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_device *socdev = rtd->socdev;
- struct snd_soc_codec *codec = socdev->card->codec;
- u8 infreq;
- /* If the system master clock is not 26MHz, the voice PCM interface is
* not avilable.*/- infreq = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL)
& TWL4030_APLL_INFREQ;- if (infreq != TWL4030_APLL_INFREQ_26000KHZ)
return -EPERM;- return 0;
+}
It's probably worth a comment here telling users that they'll need to call set_sysclk() in their init() function rather than hw_params() - otherwise this might get called before the clock is set up.
/* change rate and set CODECPDZ */twl4030_codec_enable(codec, 0);twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);twl4030_codec_enable(codec, 1);
Hrm. This codec_enable() call looks fishy - what's the effect if the other DAI is running when the voice DAI is configured? Though there may be no way round this...
+struct snd_soc_dai twl4030_dai[] = { +{ .name = "twl4030",
The { should ideally be on a line by itself (for both DAIs).
On 4/9/2009 11:06 PM, Mark Brown wrote:
On Thu, Apr 09, 2009 at 10:54:37PM +0900, Joonyoung Shim wrote:
The PCM voice interface has two modes
- PCM mode1 : This uses the rising edge of the clock signal
- PCM mode2 : This uses the falling edge of the clock signal
PCM mode1 and mode2 have a look of DSP_A and DSP_B, so we used DSP_A and DSP_B.
That sounds wrong - the difference between modes A and B is an extra cycle on BCLK after LRP, not the polarity of the signal. I'd expect that it should be one or the other of these modes with the option to invert BCLK.
Oh, i see. I will use "DAI hardware signal inversions" defines.
+static int twl4030_voice_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)+{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_device *socdev = rtd->socdev;
- struct snd_soc_codec *codec = socdev->card->codec;
- u8 infreq;
- /* If the system master clock is not 26MHz, the voice PCM interface is
* not avilable.*/- infreq = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL)
& TWL4030_APLL_INFREQ;- if (infreq != TWL4030_APLL_INFREQ_26000KHZ)
return -EPERM;- return 0;
+}
It's probably worth a comment here telling users that they'll need to call set_sysclk() in their init() function rather than hw_params() - otherwise this might get called before the clock is set up.
It seems better that i remove startup function and add set_sysclk function, and inform supporting only 26MHz system master clock through comment. i think it is not important where set_sysclk() is called, the voice PCM interface needs just 26MHz system master clock.
/* change rate and set CODECPDZ */twl4030_codec_enable(codec, 0);twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);twl4030_codec_enable(codec, 1);Hrm. This codec_enable() call looks fishy - what's the effect if the other DAI is running when the voice DAI is configured? Though there may be no way round this...
Frankly, i don't know whether this codec_enable() is called. The existing the HIFI DAI calls this codec_enable(). If this call is unnecessary i think it seems better to remove it.
+struct snd_soc_dai twl4030_dai[] = { +{ .name = "twl4030",
The { should ideally be on a line by itself (for both DAIs).
Ok, i will fix it.
Thanks for you review.
On Fri, Apr 10, 2009 at 11:03:42AM +0900, Joonyoung Shim wrote:
On 4/9/2009 11:06 PM, Mark Brown wrote:
- if (infreq != TWL4030_APLL_INFREQ_26000KHZ)
return -EPERM;
It's probably worth a comment here telling users that they'll need to call set_sysclk() in their init() function rather than hw_params() - otherwise this might get called before the clock is set up.
It seems better that i remove startup function and add set_sysclk function, and inform supporting only 26MHz system master clock through comment. i think it is not important where set_sysclk() is called, the voice PCM interface needs just 26MHz system master clock.
It does feel right to have some sort of check for a valid sysclk so that there's some software level documentation for the requirement since it's easy to set up an ASoC driver without reading the datasheet for the part so it's helpful if the driver can flag up broken configurations to them.
The reason for worrying about where set_sysclk() is called from is that if you are going to have a check like that the user needs to call set_sysclk() before the check is performed to ensure that the correct configuration has been applied.
On Thursday 09 April 2009 17:06:54 ext Mark Brown wrote:
On Thu, Apr 09, 2009 at 10:54:37PM +0900, Joonyoung Shim wrote:
/* change rate and set CODECPDZ */twl4030_codec_enable(codec, 0);twl4030_write(codec, TWL4030_REG_CODEC_MODE, mode);twl4030_codec_enable(codec, 1);Hrm. This codec_enable() call looks fishy - what's the effect if the other DAI is running when the voice DAI is configured? Though there may be no way round this...
The TRM is not that clear on what changes in the CODEC_MODE register needs a codec reset. Changing the OPT_MODE does need a reset, most probably other fields also..
On Thursday 09 April 2009 16:54:37 ext Joonyoung Shim wrote:
Add Voice DAI to support the PCM voice interface of the twl4030 codec.
The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz (voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX.
If the system master clock is not 26MHz, the voice PCM interface is not available.
The PCM voice interface has two modes
- PCM mode1 : This uses the rising edge of the clock signal
- PCM mode2 : This uses the falling edge of the clock signal
PCM mode1 and mode2 have a look of DSP_A and DSP_B, so we used DSP_A and DSP_B.
Both of the PCM mode1 and mode2 is actually DSP_A, but the FS polarity is inverted and also the data is shifted/sampled on the opposite edge of the bitclock.
+static int twl4030_voice_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)+{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_device *socdev = rtd->socdev;
- struct snd_soc_codec *codec = socdev->card->codec;
- u8 infreq;
- /* If the system master clock is not 26MHz, the voice PCM interface is
* not avilable.*/- infreq = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL)
& TWL4030_APLL_INFREQ;- if (infreq != TWL4030_APLL_INFREQ_26000KHZ)
return -EPERM;
Would it be feasible to add a check here against the CODEC_MODE:OPT_MODE and if the codec is not in Option2 mode, than return with error?
On 4/14/2009 4:07 PM, Peter Ujfalusi wrote:
On Thursday 09 April 2009 16:54:37 ext Joonyoung Shim wrote:
Add Voice DAI to support the PCM voice interface of the twl4030 codec.
The PCM voice interface can be used with 8-kHz(voice narrowband) or 16-kHz (voice wideband) sampling rates, and 16bits, and mono RX and mono TX or stereo TX.
If the system master clock is not 26MHz, the voice PCM interface is not available.
The PCM voice interface has two modes
- PCM mode1 : This uses the rising edge of the clock signal
- PCM mode2 : This uses the falling edge of the clock signal
PCM mode1 and mode2 have a look of DSP_A and DSP_B, so we used DSP_A and DSP_B.
Both of the PCM mode1 and mode2 is actually DSP_A, but the FS polarity is inverted and also the data is shifted/sampled on the opposite edge of the bitclock.
So, I will use that the PCM mode1 is SND_SOC_DAIFMT_IB_NF and PCM mode2 is SND_SOC_DAIFMT_NB_IF. Is this right?
+static int twl4030_voice_startup(struct snd_pcm_substream *substream,
struct snd_soc_dai *dai)+{
- struct snd_soc_pcm_runtime *rtd = substream->private_data;
- struct snd_soc_device *socdev = rtd->socdev;
- struct snd_soc_codec *codec = socdev->card->codec;
- u8 infreq;
- /* If the system master clock is not 26MHz, the voice PCM interface is
* not avilable.*/- infreq = twl4030_read_reg_cache(codec, TWL4030_REG_APLL_CTL)
& TWL4030_APLL_INFREQ;- if (infreq != TWL4030_APLL_INFREQ_26000KHZ)
return -EPERM;Would it be feasible to add a check here against the CODEC_MODE:OPT_MODE and if the codec is not in Option2 mode, than return with error?
Ok, it's good, i will add. Thanks.
On Monday 20 April 2009 08:31:03 ext Joonyoung Shim wrote:
On 4/14/2009 4:07 PM, Peter Ujfalusi wrote:
Both of the PCM mode1 and mode2 is actually DSP_A, but the FS polarity is inverted and also the data is shifted/sampled on the opposite edge of the bitclock.
So, I will use that the PCM mode1 is SND_SOC_DAIFMT_IB_NF and PCM mode2 is SND_SOC_DAIFMT_NB_IF. Is this right?
Yes, I think this is right.
participants (3)
-
Joonyoung Shim -
Mark Brown -
Peter Ujfalusi