Porting adau1761 codec

Steve,

ADAU1761 has integrated PLL, while AIC3x not. DM814x device PSP is based on linux kernel 2.6.37

The first step from the Audio.ppt is:

1. Find the supported codec from the list <kernel src>/sound/soc/codec/*
codec must be compatible with the Audio interface(TDM/I2S) data format

The AIC3x codec driver is:
<kernel src>/sound/soc/codec/tlv320aic3x.c

What about the ADAU1761 codec driver, I can not find it in the list of available audio codec drivers, which one you are using?

BR
Pavel

Steve,

Steve Hooper said:

when I do a arecord it fails

Could you provide me your steps to do arecord to check them on my side?

Steve Hooper said:

fails with this message 'asoc: can't set codec adau1761-aif1 hw params' tracing where exactly it fails pointed me to
function davinci_pcm_hw_params(), inside this it uses snd_pcm_lib_malloc_pages() to which this call fails.

Any pointers why malloc pages fails ?

What is the software release you are using? Is it DM814x EZSDK 5.05.02.00?

BR
Pavel

Steve,

Steve Hooper said:

$ arecord -f cd
Recording WAVE 'stdin' : Signed 16 bit Little En[ 109.320000] dbg1
dian, Rate 44100[ 109.320000] dbg2
Hz, Stereo
[ 109.320000] dbg3
[ 109.330000] dbg4
[ 109.330000] dbg5
[ 109.330000] dbg6
[ 109.330000] dbg7
[ 109.460000] dbg8
[ 109.460000] dbg9
[ 109.460000] dbg20
[ 109.460000] dbg21
[ 109.460000] dbg22
[ 109.460000] dbg23
[ 109.470000] dbg24
[ 109.470000] dbg25
[ 109.470000] dbg26!!!!
RIFF$�WAVEfmt D��data�
[ 119.470000] ALSA sound/core/pcm_lib.c:1765: capture write error (DMA or IRQ trouble?)
arecord: pcm_read:2031: read error: Input/output error

Can you try the below commands, what is the result?

root@dm814x-evm:~# arecord -l
**** List of CAPTURE Hardware Devices ****
card 0: EVM [TI81XX EVM], device 0: AIC3X tlv320aic3x-hifi-0 []
  Subdevices: 1/1
  Subdevice #0: subdevice #0
root@dm814x-evm:~# arecord -D 'hw:0,0' -r 44100 -d 10 -f cd song.dat
Recording WAVE 'song.dat' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
root@dm814x-evm:~#

hw:0,0 parameter comes from card 0, device 0

BR
Pavel

Steve,

Steve Hooper said:

#define AUDIO_FORMAT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM)

But it in some file it mentioned davinci doesnt support i2s mode. Is it the case I cant use I2S mode ?

See the below e2e thread, it might help you:

BR
Pavel

Steve,

Steve Hooper said:

[ 115.610000] asoc: can't set codec adau1761-aif1 hw params
arecord: set_params:1297: Unable to install hw params:

May be the audio codec (adau1761) can not accept these settings, check its datasheet if these settings are supported. Check also with the snd_pcm_hw_params_test_ <<parameter>>  function:

snd_pcm_hw_params_test_ <<parameter>> - Test the availability of important parameters like number of channels, sample rate etc. For e.g. snd_pcm_hw_params_test_format, snd_pcm_hw_params_test_rate etc.

BR
Pavel

Steve,

Steve Hooper said:

Any pointer to change the params_rate(params) its always 44100 I want it 48K

From ALSA utils (arecord), you can pass the rate with -r:

# arecord -D 'hw:0,0' -r 48000 -f S16_LE -c 2  song.dat

For the user space application, you should set in the rate variable:

int main (int argc, char *argv[])
{
  int err;
  short buf[BUFF_SIZE];
  int rate = 44100; /* Sample rate */
  unsigned int exact_rate; /* Sample rate returned by */

.........

/* Set sample rate. If the exact rate is not supported by the hardware, use nearest possible rate. */
  exact_rate = rate;
  if ((err = snd_pcm_hw_params_set_rate_near (playback_handle, hw_params, &exact_rate, 0)) < 0) {
    fprintf (stderr, "cannot set sample rate (%s)\n", snd_strerror (err));
    exit (1);
  }

  if (rate != exact_rate) {
    fprintf(stderr, "The rate %d Hz is not supported by your hardware.\n ==> Using %d Hz instead.\n", rate, exact_rate);
  }

 

Pavel

Steve,

Steve Hooper said:

Does the davinci mcasp set it to 44100 by default ? I searched in the driver but couldnt find anything .

No, I do not think so. The sample rate can be also set in the root filesystem (/usr/share/alsa/), but by default it is not.

Steve Hooper said:

I tried the following:
# arecord -D 'hw:0,0' -r 48000 -d 10 -f cd song.dat

and printed the Params rate in evm_hw_params() in davinci-evm.c
but its the following:
Recording WAVE '[ 33.240000] Params rate: 44100
song.dat' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo

I have the same at my side:

root@dm814x-evm:~# arecord -D 'hw:0,0' -r 48000 -d 10 -f cd song.dat
Recording WAVE 'song.dat' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo

I think this is caused by the format (-f cd). This "-f cd" is short cut to this whole command:

-f S16_LE -c2 -r44100

Like "-f dat" is short cut to:

-f S16_LE -c2 -r48000

This is what I have on my side:

root@dm814x-evm:~# arecord -D 'hw:0,0' -r 48000 -f S16_LE -d 10 song.dat     
Recording WAVE 'song.dat' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono

root@dm814x-evm:~# arecord -D 'hw:0,0' -r 48000 -f S16_LE -c 2 -d 10 song.dat
Recording WAVE 'song.dat' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo

root@dm814x-evm:~# arecord -D 'hw:0,0' -r 48000 -d 10 -f dat song.dat
Recording WAVE 'song.dat' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo

root@dm814x-evm:~# arecord -D 'hw:0,0' -d 10 -f dat song.dat         
Recording WAVE 'song.dat' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo


Regards,
Pavel