Re: [alsa-devel] ALSA Core: Sound Card registration (CPU --> CODEC_1 --> CODEC_2)
We have a question regarding how sound card can be registered for the following scenario:
CPU connecting to Audio CODEC_1 and then CODEC_1 connecting to another Audio CODEC_2 (over some bus like i2c). CODEC_1 can be represented as a
so you mean codec_1 is controlling codec_2 ? or you mean the CPU is going to control both the codecs?
codec_dai and can be registered to ASoC framework (with snd_soc_register_codec()) during system boot up. However, for end-to-end audio to work on this system, we also need CODEC_2 somehow registered to ASoC framework so that audio path on CODEC_2 can configured.
From this i understand that your CPU is going to configure the codec_2
but then why it is connected to codec_1 with i2c?
Could you please let us know if CODEC_2 can also registered as another dai and if yes, how it can be achieved ? Or is there any other way to probe_defer sound card registration till CODEC_2 is enumerated on the bus.
why not register both as codecs to the ALSA? Anyway i think ALSA can support multiple codecs.
In this case, to what component of the sound card dailink structure, the Audio CODEC_2 can be represented ?
as a codec. However this would not work if there is a dependecny in detection i.e. if you have a requirement where codec_1 needs to be detected first and then codec_2 then probably you need a different approach as there is no way to set dependency.
Please share your thoughts on this.
I would explain more about the setup if i were you to get ideas from ALSA experts.
Best Regards, Phani
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Message: 2 Date: Thu, 4 Sep 2014 12:03:52 +0530 From: Vinod Koul vinod.koul@intel.com To: Lars-Peter Clausen lars@metafoo.de Cc: alsa-devel@alsa-project.org, Mark Brown broonie@kernel.org, "Subhransu S. Prusty" subhransu.s.prusty@intel.com, Liam Girdwood lgirdwood@gmail.com Subject: Re: [alsa-devel] [PATCH 1/4] ASoC: Add snd_soc_component_{get, set}_drvdata() Message-ID: 20140904063352.GC1610@intel.com Content-Type: text/plain; charset=us-ascii
On Wed, Sep 03, 2014 at 09:06:49PM +0200, Lars-Peter Clausen wrote:
On 09/03/2014 02:22 PM, Vinod Koul wrote:
On Wed, Aug 20, 2014 at 01:08:46PM +0200, Lars-Peter Clausen wrote:
Add Add snd_soc_component_{get,set}_drvdata() similar to snd_soc_codec_{get,set}_drvdata() and snd_soc_platform_{get,set}_drvdata(). Also update them to use the new functions internally.
Signed-off-by: Lars-Peter Clausen lars@metafoo.de
Reviewed-by: Vinod Koul vinod.koul@intel.com
Is this for the first patch or for the whole series?
Patch only. The HSW driver changes, I think Liam should ack
-- ~Vinod
Message: 3 Date: Thu, 4 Sep 2014 10:52:53 +0300 From: Peter Ujfalusi peter.ujfalusi@ti.com To: Mark Brown broonie@kernel.org, Liam Girdwood lgirdwood@gmail.com Cc: alsa-devel@alsa-project.org, jsarha@ti.com, zonque@gmail.com Subject: [alsa-devel] [PATCH] ASoC: davinci-mcasp: Correct rx format unit configuration Message-ID: 1409817173-29234-1-git-send-email-peter.ujfalusi@ti.com Content-Type: text/plain
In case of capture we should not use rotation. The reverse and mask is enough to get the data align correctly from the bus to MCU: Format data from bus after reverse (XRBUF) S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB| S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB| S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB|
With this patch all supported formats will work for playback and capture.
Reported-by: Jyri Sarha jsarha@ti.com (broken S24_3LE capture) Signed-off-by: Peter Ujfalusi peter.ujfalusi@ti.com
sound/soc/davinci/davinci-mcasp.c | 11 ++++++++++- 1 file changed, 10 insertions(+), 1 deletion(-)
diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 6a6b2ff7d7d7..68347b55f6e1 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -467,8 +467,17 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, { u32 fmt; u32 tx_rotate = (word_length / 4) & 0x7;
u32 rx_rotate = (32 - word_length) / 4; u32 mask = (1ULL << word_length) - 1;
/** For captured data we should not rotate, inversion and masking is* enoguh to get the data to the right position:* Format data from bus after reverse (XRBUF)* S16_LE: |LSB|MSB|xxx|xxx| |xxx|xxx|MSB|LSB|* S24_3LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB|* S24_LE: |LSB|DAT|MSB|xxx| |xxx|MSB|DAT|LSB|* S32_LE: |LSB|DAT|DAT|MSB| |MSB|DAT|DAT|LSB|*/u32 rx_rotate = 0; /* * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv()-- 2.1.0
Message: 4 Date: Thu, 4 Sep 2014 10:59:41 +0300 From: Peter Ujfalusi peter.ujfalusi@ti.com To: Mark Brown broonie@kernel.org, Liam Girdwood lgirdwood@gmail.com Cc: alsa-devel@alsa-project.org, jsarha@ti.com Subject: [alsa-devel] [PATCH] ASoC: tlv320aic31xx: Enable support for S24_LE format Message-ID: 1409817581-29496-1-git-send-email-peter.ujfalusi@ti.com Content-Type: text/plain
S24_LE is the same on the bus as S24_3LE, which means the codec can support it.
Signed-off-by: Peter Ujfalusi peter.ujfalusi@ti.com
sound/soc/codecs/tlv320aic31xx.h | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-)
diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 52ed57c69dfa..fe16c34607bb 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -18,7 +18,8 @@ #define AIC31XX_RATES SNDRV_PCM_RATE_8000_192000
#define AIC31XX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \
| SNDRV_PCM_FMTBIT_S24_3LE |SNDRV_PCM_FMTBIT_S32_LE)
| SNDRV_PCM_FMTBIT_S24_3LE |SNDRV_PCM_FMTBIT_S24_LE \
| SNDRV_PCM_FMTBIT_S32_LE)#define AIC31XX_STEREO_CLASS_D_BIT 0x1
2.1.0
Message: 5 Date: Thu, 4 Sep 2014 17:13:06 +0800 From: Xavier Hsu xavier.hsu@linaro.org To: Charles Keepax ckeepax@opensource.wolfsonmicro.com Cc: Andy Green andy.green@linaro.org, alsa-devel@alsa-project.org, patches@opensource.wolfsonmicro.com, Patch Tracking patches@linaro.org Subject: Re: [alsa-devel] [PATCHv3 5/9] Using the constraint based on wm8971_set_dai_sysclk Message-ID: CAHa2TOYHbA7nEQQeqxB+r3KyA=BxXOmfQ6psuouZgqWomb=pEg@mail.gmail.com Content-Type: text/plain; charset=UTF-8
Hi Charles : Thanks for yours feedback.
According your suggestion, I combine the redundant codes and fix some mistakes. (rates_12288[] and rates_18432[] => rates_48000[] rates_112896[] and rates_169344[] => rates_44100[] 2400 => 24000 ) I also add snd_pcm_hw_constraint_list() to restrict sysclk.
Thanks.
BR, Xavier
2014-09-02 17:28 GMT+08:00 Charles Keepax < ckeepax@opensource.wolfsonmicro.com>:
On Tue, Sep 02, 2014 at 11:27:46AM +0800, Xavier Hsu wrote:
This patch improves WM8971. We use the constraint based on the function of wm8971_set_dai_sysclk().
Any comments about improving the patch are welcome. Thanks.
Comments like this are probably best put after the --- as they don't need to appear in the change log.
Signed-off-by: Xavier Hsu xavier.hsu@linaro.org Signed-off-by: Andy Green andy.green@linaro.org
sound/soc/codecs/wm8971.c | 77
+++++++++++++++++++++++++++++++++++++++++----
1 file changed, 71 insertions(+), 6 deletions(-)
diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 64ed226..20cfdd3 100755 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -41,6 +41,7 @@ static struct workqueue_struct *wm8971_workq; /* codec private data */ struct wm8971_priv { unsigned int sysclk;
struct snd_pcm_hw_constraint_list *sysclk_constraints; int playback_fs; bool deemph;}; @@ -528,6 +529,53 @@ static int get_coeff(int mclk, int rate) return -EINVAL; }
+/* The set of rates we can generate from the above for each SYSCLK */ +static const unsigned int rates_12288[] = {
8000, 12000, 16000, 24000, 32000, 48000, 96000+};
+static struct snd_pcm_hw_constraint_list constraints_12288 = {
.count = ARRAY_SIZE(rates_12288),.list = rates_12288,+};
+static const unsigned int rates_112896[] = {
8000, 11025, 22050, 44100, 88200+};
+static struct snd_pcm_hw_constraint_list constraints_112896 = {
.count = ARRAY_SIZE(rates_112896),.list = rates_112896,+};
+static const unsigned int rates_18432[] = {
8000, 12000, 16000, 24000, 32000, 48000, 96000+};
+static struct snd_pcm_hw_constraint_list constraints_18432 = {
.count = ARRAY_SIZE(rates_18432),.list = rates_18432,+};
This one is identical to the 12288 array, why not combine them and call them the 48k array?
+static const unsigned int rates_169344[] = {
8000, 11025, 22050, 44100, 88200+};
This one is identical to the 112896 array, why not combine them and call them the 44k1 array?
+static struct snd_pcm_hw_constraint_list constraints_169344 = {
.count = ARRAY_SIZE(rates_169344),.list = rates_169344,+};
+static const unsigned int rates_12[] = {
8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000,Typo 0 missing ^^^^
48000, 88235, 96000,+};
+static struct snd_pcm_hw_constraint_list constraints_12 = {
.count = ARRAY_SIZE(rates_12),.list = rates_12,+};
static int wm8971_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { @@ -535,15 +583,32 @@ static int wm8971_set_dai_sysclk(struct
snd_soc_dai *codec_dai,
struct wm8971_priv *wm8971 = snd_soc_codec_get_drvdata(codec); switch (freq) {
case 11289600:case 12000000: case 12288000:case 16934400:
case 24576000:wm8971->sysclk_constraints = &constraints_12288;break;case 11289600:case 22579200:wm8971->sysclk_constraints = &constraints_112896;break; case 18432000:
wm8971->sysclk = freq;return 0;
wm8971->sysclk_constraints = &constraints_18432;break;case 16934400:case 33868800:wm8971->sysclk_constraints = &constraints_169344;break;case 12000000:case 24000000:wm8971->sysclk_constraints = &constraints_12;break;default:return -EINVAL; }
return -EINVAL;You pick out the constraints here but you never actually set them with snd_pcm_hw_constraint_list?
wm8971->sysclk = freq;return 0;}
static int wm8971_set_dai_fmt(struct snd_soc_dai *codec_dai,
Thanks, Charles
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