Jacinto 6 ALSA Audio Capture and Playback Issue

Mythili Ramamoorthi

Hi,

We are using Jacinto 6 Vayu EVM board.Audio Capture and Playback using ALSA libs APIs working fine in Low volume.When the volume is increased to the maximum level some background noise in playback.We are capturing the audio using AUX cable(Mic input ).Kindly suggest the root cause.

The sample code i attached here.The same code working fine in PC but in J6 it makes problem.Is any pin configuration changes we need to do?

#define PERIOD_SIZE 1024
#define BUF_SIZE (PERIOD_SIZE * 2)

#include <stdio.h>
#include <stdlib.h>
#include <alsa/asoundlib.h>

void print_pcm_state(snd_pcm_t *handle, char *name) {
switch (snd_pcm_state(handle)) {
case SND_PCM_STATE_OPEN:
    printf("state open %s\n", name);
    break;

case SND_PCM_STATE_SETUP:
    printf("state setup %s\n", name);
    break;

case SND_PCM_STATE_PREPARED:
    printf("state prepare %s\n", name);
    break;

case SND_PCM_STATE_RUNNING:
    printf("state running %s\n", name);
    break;

case SND_PCM_STATE_XRUN:
    printf("state xrun %s\n", name);
    break;

default:
    printf("state other %s\n", name);
    break;

}
}

int setparams(snd_pcm_t *handle, char *name) {
snd_pcm_hw_params_t *hw_params;
int err;

if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) {
    fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n",
         snd_strerror (err));
    exit (1);
}

if ((err = snd_pcm_hw_params_any (handle, hw_params)) < 0) {
    fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n",
         snd_strerror (err));
    exit (1);
}

if ((err = snd_pcm_hw_params_set_access (handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
    fprintf (stderr, "cannot set access type (%s)\n",
         snd_strerror (err));
    exit (1);
}

if ((err = snd_pcm_hw_params_set_format (handle, hw_params, SND_PCM_FORMAT_S16_LE)) < 0) {
    fprintf (stderr, "cannot set sample format (%s)\n",
         snd_strerror (err));
    exit (1);
}

unsigned int rate = 48000;
if ((err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &rate, 0)) < 0) {
    fprintf (stderr, "cannot set sample rate (%s)\n",
         snd_strerror (err));
    exit (1);
}
printf("Rate for %s is %d\n", name, rate);

if ((err = snd_pcm_hw_params_set_channels (handle, hw_params, 2)) < 0) {
    fprintf (stderr, "cannot set channel count (%s)\n",
         snd_strerror (err));
    exit (1);
}

snd_pcm_uframes_t buffersize = BUF_SIZE;
if ((err = snd_pcm_hw_params_set_buffer_size_near(handle, hw_params, &buffersize)) < 0) {
    printf("Unable to set buffer size %li: %s\n", BUF_SIZE, snd_strerror(err));
    exit (1);;
}

snd_pcm_uframes_t periodsize = PERIOD_SIZE;
fprintf(stderr, "period size now %d\n", periodsize);
if ((err = snd_pcm_hw_params_set_period_size_near(handle, hw_params, &periodsize, 0)) < 0) {
    printf("Unable to set period size %li: %s\n", periodsize, snd_strerror(err));
    exit (1);
}

if ((err = snd_pcm_hw_params (handle, hw_params)) < 0) {
    fprintf (stderr, "cannot set parameters (%s)\n",
         snd_strerror (err));
    exit (1);
}

snd_pcm_uframes_t p_psize;
snd_pcm_hw_params_get_period_size(hw_params, &p_psize, NULL);
fprintf(stderr, "period size %d\n", p_psize);

snd_pcm_hw_params_get_buffer_size(hw_params, &p_psize);
fprintf(stderr, "buffer size %d\n", p_psize);

snd_pcm_hw_params_free (hw_params);

if ((err = snd_pcm_prepare (handle)) < 0) {
    fprintf (stderr, "cannot prepare audio interface for use (%s)\n",
         snd_strerror (err));
    exit (1);
}

return 0;
}

int set_sw_params(snd_pcm_t *handle, char *name) {
snd_pcm_sw_params_t *swparams;
int err;

snd_pcm_sw_params_alloca(&swparams);

err = snd_pcm_sw_params_current(handle, swparams);
if (err < 0) {
    fprintf(stderr, "Broken configuration for this PCM: no configurations available\n");
    exit(1);
}

err = snd_pcm_sw_params_set_start_threshold(handle, swparams, PERIOD_SIZE);
if (err < 0) {
    printf("Unable to set start threshold: %s\n", snd_strerror(err));
    return err;
}
err = snd_pcm_sw_params_set_avail_min(handle, swparams, PERIOD_SIZE);
if (err < 0) {
    printf("Unable to set avail min: %s\n", snd_strerror(err));
    return err;
}

if (snd_pcm_sw_params(handle, swparams) < 0) {
    fprintf(stderr, "unable to install sw params:\n");
    exit(1);
}

return 0;
}

/************** some code from latency.c *****************/

main (int argc, char *argv[])
{
int i;
int err;
int buf[BUF_SIZE];
snd_pcm_t *playback_handle;
snd_pcm_t *capture_handle;
snd_pcm_hw_params_t *hw_params;
FILE *fin;
size_t nread;
snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
if (argc != 3) {
    fprintf(stderr, "Usage: %s in-card out-card\n", argv[0]);
    exit(1);
}

/**** Out card *******/
if ((err = snd_pcm_open (&playback_handle, argv[2], SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
    fprintf (stderr, "cannot open audio device %s (%s)\n",
         argv[2],
         snd_strerror (err));
    exit (1);
}

setparams(playback_handle, "playback");
set_sw_params(playback_handle, "playback");

/*********** In card **********/

if ((err = snd_pcm_open (&capture_handle, argv[1], SND_PCM_STREAM_CAPTURE, 0)) < 0) {
    fprintf (stderr, "cannot open audio device %s (%s)\n",
         argv[1],
         snd_strerror (err));
    exit (1);
}

setparams(capture_handle, "capture");
set_sw_params(capture_handle, "capture");

if ((err = snd_pcm_link(capture_handle, playback_handle)) < 0) {
    printf("Streams link error: %s\n", snd_strerror(err));
    exit(0);
}

if ((err = snd_pcm_prepare (playback_handle)) < 0) {
    fprintf (stderr, "cannot prepare playback audio interface for use (%s)\n",
         snd_strerror (err));
    exit (1);
}

/**************** stuff something into the playback buffer ****************/
if (snd_pcm_format_set_silence(format, buf, 2*BUF_SIZE) < 0) {
    fprintf(stderr, "silence error\n");
    exit(1);
}

int n = 0;
while (n++ < 2) {
    if (snd_pcm_writei (playback_handle, buf, BUF_SIZE) < 0) {
      fprintf(stderr, "write error\n");
      exit(1);
    }
}

/************* COPY ************/
while (1) {
    int nread;
    if ((nread = snd_pcm_readi (capture_handle, buf, BUF_SIZE)) != BUF_SIZE) {
      if (nread < 0) {
   fprintf (stderr, "read from audio interface failed (%s)\n",
       snd_strerror (nread));
      } else {
   fprintf (stderr, "read from audio interface failed after %d frames\n", nread);
      }
      snd_pcm_prepare(capture_handle);
      continue;
    }

    if ((err = snd_pcm_writei (playback_handle, buf, nread)) != nread) {
      if (err < 0) {
   fprintf (stderr, "write to audio interface failed (%s)\n",
       snd_strerror (err));
      } else {
   fprintf (stderr, "write to audio interface failed after %d frames\n", err);
      }
      snd_pcm_prepare(playback_handle);
    }
}

snd_pcm_drain(playback_handle);
snd_pcm_close (playback_handle);
exit (0);
#define PERIOD_SIZE 1024
#define BUF_SIZE (PERIOD_SIZE * 2)

#include <stdio.h>
#include <stdlib.h>
#include <alsa/asoundlib.h>

void print_pcm_state(snd_pcm_t *handle, char *name) {
switch (snd_pcm_state(handle)) {
case SND_PCM_STATE_OPEN:
    printf("state open %s\n", name);
    break;

case SND_PCM_STATE_SETUP:
    printf("state setup %s\n", name);
    break;

case SND_PCM_STATE_PREPARED:
    printf("state prepare %s\n", name);
    break;

case SND_PCM_STATE_RUNNING:
    printf("state running %s\n", name);
    break;

case SND_PCM_STATE_XRUN:
    printf("state xrun %s\n", name);
    break;

default:
    printf("state other %s\n", name);
    break;

}
}

int setparams(snd_pcm_t *handle, char *name) {
snd_pcm_hw_params_t *hw_params;
int err;

if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) {
    fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n",
         snd_strerror (err));
    exit (1);
}

if ((err = snd_pcm_hw_params_any (handle, hw_params)) < 0) {
    fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n",
         snd_strerror (err));
    exit (1);
}

if ((err = snd_pcm_hw_params_set_access (handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
    fprintf (stderr, "cannot set access type (%s)\n",
         snd_strerror (err));
    exit (1);
}

if ((err = snd_pcm_hw_params_set_format (handle, hw_params, SND_PCM_FORMAT_S16_LE)) < 0) {
    fprintf (stderr, "cannot set sample format (%s)\n",
         snd_strerror (err));
    exit (1);
}

unsigned int rate = 48000;
if ((err = snd_pcm_hw_params_set_rate_near (handle, hw_params, &rate, 0)) < 0) {
    fprintf (stderr, "cannot set sample rate (%s)\n",
         snd_strerror (err));
    exit (1);
}
printf("Rate for %s is %d\n", name, rate);

if ((err = snd_pcm_hw_params_set_channels (handle, hw_params, 2)) < 0) {
    fprintf (stderr, "cannot set channel count (%s)\n",
         snd_strerror (err));
    exit (1);
}

snd_pcm_uframes_t buffersize = BUF_SIZE;
if ((err = snd_pcm_hw_params_set_buffer_size_near(handle, hw_params, &buffersize)) < 0) {
    printf("Unable to set buffer size %li: %s\n", BUF_SIZE, snd_strerror(err));
    exit (1);;
}

snd_pcm_uframes_t periodsize = PERIOD_SIZE;
fprintf(stderr, "period size now %d\n", periodsize);
if ((err = snd_pcm_hw_params_set_period_size_near(handle, hw_params, &periodsize, 0)) < 0) {
    printf("Unable to set period size %li: %s\n", periodsize, snd_strerror(err));
    exit (1);
}

if ((err = snd_pcm_hw_params (handle, hw_params)) < 0) {
    fprintf (stderr, "cannot set parameters (%s)\n",
         snd_strerror (err));
    exit (1);
}

snd_pcm_uframes_t p_psize;
snd_pcm_hw_params_get_period_size(hw_params, &p_psize, NULL);
fprintf(stderr, "period size %d\n", p_psize);

snd_pcm_hw_params_get_buffer_size(hw_params, &p_psize);
fprintf(stderr, "buffer size %d\n", p_psize);

snd_pcm_hw_params_free (hw_params);

if ((err = snd_pcm_prepare (handle)) < 0) {
    fprintf (stderr, "cannot prepare audio interface for use (%s)\n",
         snd_strerror (err));
    exit (1);
}

return 0;
}

int set_sw_params(snd_pcm_t *handle, char *name) {
snd_pcm_sw_params_t *swparams;
int err;

snd_pcm_sw_params_alloca(&swparams);

err = snd_pcm_sw_params_current(handle, swparams);
if (err < 0) {
    fprintf(stderr, "Broken configuration for this PCM: no configurations available\n");
    exit(1);
}

err = snd_pcm_sw_params_set_start_threshold(handle, swparams, PERIOD_SIZE);
if (err < 0) {
    printf("Unable to set start threshold: %s\n", snd_strerror(err));
    return err;
}
err = snd_pcm_sw_params_set_avail_min(handle, swparams, PERIOD_SIZE);
if (err < 0) {
    printf("Unable to set avail min: %s\n", snd_strerror(err));
    return err;
}

if (snd_pcm_sw_params(handle, swparams) < 0) {
    fprintf(stderr, "unable to install sw params:\n");
    exit(1);
}

return 0;
}

/************** some code from latency.c *****************/

main (int argc, char *argv[])
{
int i;
int err;
int buf[BUF_SIZE];
snd_pcm_t *playback_handle;
snd_pcm_t *capture_handle;
snd_pcm_hw_params_t *hw_params;
FILE *fin;
size_t nread;
snd_pcm_format_t format = SND_PCM_FORMAT_S16_LE;
if (argc != 3) {
    fprintf(stderr, "Usage: %s in-card out-card\n", argv[0]);
    exit(1);
}

/**** Out card *******/
if ((err = snd_pcm_open (&playback_handle, argv[2], SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
    fprintf (stderr, "cannot open audio device %s (%s)\n",
         argv[2],
         snd_strerror (err));
    exit (1);
}

setparams(playback_handle, "playback");
set_sw_params(playback_handle, "playback");

/*********** In card **********/

if ((err = snd_pcm_open (&capture_handle, argv[1], SND_PCM_STREAM_CAPTURE, 0)) < 0) {
    fprintf (stderr, "cannot open audio device %s (%s)\n",
         argv[1],
         snd_strerror (err));
    exit (1);
}

setparams(capture_handle, "capture");
set_sw_params(capture_handle, "capture");

if ((err = snd_pcm_link(capture_handle, playback_handle)) < 0) {
    printf("Streams link error: %s\n", snd_strerror(err));
    exit(0);
}

if ((err = snd_pcm_prepare (playback_handle)) < 0) {
    fprintf (stderr, "cannot prepare playback audio interface for use (%s)\n",
         snd_strerror (err));
    exit (1);
}

/**************** stuff something into the playback buffer ****************/
if (snd_pcm_format_set_silence(format, buf, 2*BUF_SIZE) < 0) {
    fprintf(stderr, "silence error\n");
    exit(1);
}

int n = 0;
while (n++ < 2) {
    if (snd_pcm_writei (playback_handle, buf, BUF_SIZE) < 0) {
      fprintf(stderr, "write error\n");
      exit(1);
    }
}

/************* COPY ************/
while (1) {
    int nread;
    if ((nread = snd_pcm_readi (capture_handle, buf, BUF_SIZE)) != BUF_SIZE) {
      if (nread < 0) {
   fprintf (stderr, "read from audio interface failed (%s)\n",
       snd_strerror (nread));
      } else {
   fprintf (stderr, "read from audio interface failed after %d frames\n", nread);
      }
      snd_pcm_prepare(capture_handle);
      continue;
    }

    if ((err = snd_pcm_writei (playback_handle, buf, nread)) != nread) {
      if (err < 0) {
   fprintf (stderr, "write to audio interface failed (%s)\n",
       snd_strerror (err));
      } else {
   fprintf (stderr, "write to audio interface failed after %d frames\n", err);
      }
      snd_pcm_prepare(playback_handle);
    }
}

snd_pcm_drain(playback_handle);
snd_pcm_close (playback_handle);
exit (0);
}
}

over 10 years ago

0 Yordan Kovachev over 10 years ago

TI__Guru**** 161600 points

Hi,

The code you posted seems OK.

Which SDK are you using? Could you share your amixer settings?

Have you verified that your hardware connections (board audio output port & contact to headphones) are OK? Can you try playing the recorded file with different headphones?

Best Regards,
Yordan

0 Mythili Ramamoorthi over 10 years ago in reply to Yordan Kovachev

Intellectual 425 points

Hi Yordan,

Thanks much for your response.

The SDK version we are using is ti-glsdk_dra7xx-evm_7_03_00_03 and the amixer setting is

amixer cset name="PCM Playback Volume" 127

I have done the testing in two j6 boards and different headphones but the behavior is same. For Ethernet communication implementation the Audio is captured (MIC input) from PC and transmitted to the board for playback(Headphone out). From this I came to know that the Board Audio output port working fine and got smooth playback.But when the Audio is captured from the Board (MIC input ) and transmitted to another j6 board for playback means making the back ground noise in playback at high volume level.At low volume we got smooth playback without noise.The above is the observed behavior. Kindly suggest me.

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Jacinto 6 ALSA Audio Capture and Playback Issue