Frames during Audio playback are consumed very fast - linux

Question

My requirement is to play 16 bit, 22KHz linear PCM data to audioout using alsa API. I have added 2 section of code which i have written to configure ALSA and audioout wrapper which writes 2048 frames using write API.
Issue : Getting distorted noise and consuming available frames very quickly
I am configuring ALSA using below api's.
//opening audio handle
if ((err = snd_pcm_open (&rsettings->ppcm_handle, name, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
{
return -1;
}
if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0)
{
return -1;
}
if ((err = snd_pcm_hw_params_any (rsettings->ppcm_handle, hw_params)) < 0)
{
return -1;
}
//setting interleaved mode
if ((err = snd_pcm_hw_params_set_access (rsettings->ppcm_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
{
return -1;
}
//setting bit depth to 2 byte
if ((err = snd_pcm_hw_params_set_format (rsettings->ppcm_handle, hw_params, SND_PCM_FORMAT_S16_LE) ) < 0)
{
return -1;
}
//Setting sample rate to 22050 Hz
if ((err = snd_pcm_hw_params_set_rate(rsettings->ppcm_handle, hw_params, 22050, 0 ) ) < 0 )
{
return -1;
}
//setting number of channels to 1
if ((err = snd_pcm_hw_params_set_channels (rsettings->ppcm_handle, hw_params, 1 ) ) < 0)
{
return -1;
}
snd_pcm_uframes_t frame_size = 1024;
if(( err = snd_pcm_hw_params_set_period_size_near(rsettings->ppcm_handle, hw_params, &frame_size, 0 ) ) < 0 )
{
printf("\n cannot set size");
return -1;
}
// setting number of period to 1
if ((err = snd_pcm_hw_params_set_periods(rsettings->ppcm_handle, hw_params, 1, 1 ) ) < 0)
{
return -1;
}
frame_size = 2048;
//setting buffer size to 2048
if ((err = snd_pcm_hw_params_set_buffer_size_near(rsettings->ppcm_handle, hw_params, &frame_size) ) < 0)
{
return -1;
}
if ((err = snd_pcm_nonblock (rsettings->ppcm_handle, 0 ) ) < 0)
{
return -1;
}
if ( (err = snd_pcm_hw_params (rsettings->ppcm_handle, hw_params)) < 0)
{
return -1;
}
snd_pcm_hw_params_free (hw_params);
if ((err = snd_pcm_prepare (rsettings->ppcm_handle)) < 0)
{
return -1;
}
I have audioout API using POLLING as below
noOfFrames = 2024
while( noOfFrames > 0 )
{
if( 0 > ( poll( descriptors, pollDescriptorCount, 500 ) ) )
{
printf("error polling for data");
}
else
{
printf("\nPoll Unblocked");
if( 0 > ( retVal = snd_pcm_poll_descriptors_revents( rsettings->ppcm_handle, descriptors, pollDescriptorCount, &pollEvent ) ) )
{
printf("\nError getting poll events, return value : %d", retVal);
break;
}
else
{
frames = snd_pcm_avail_update( rsettings->ppcm_handle );
printf("\nAvailable channel frames = %d",frames );
if( frames < 0 )
{
if( frames == -EPIPE )
{
snd_pcm_prepare( rsettings->ppcm_handle );
}
else if( frames == -ESTRPIPE )
{
while (-EAGAIN == snd_pcm_resume( rsettings->ppcm_handle ) )
usleep(2000);
}
else
{
snd_pcm_prepare( rsettings->ppcm_handle);
}
}
else
{
if( frames >= noOfFrames )
{
if ((framesWritten = snd_pcm_writei(rsettings->ppcm_handle, &pBuffer, noOfFrames)) > 0 )
{
printf("\nNo of frames written : %d",framesWritten);
noOfFrames = noOfFrames - framesWritten;
free(descriptors);
}
else if( framesWritten < 0 )
{
if( framesWritten == -EPIPE )
{
snd_pcm_prepare( rsettings->ppcm_handle );
}
else if( framesWritten == -ESTRPIPE )
{
while (-EAGAIN == snd_pcm_resume( rsettings->ppcm_handle ))
usleep(2000);
}
else
{
snd_pcm_prepare( rsettings->ppcm_handle );
}
}
}
else
{
if ( ( framesWritten = snd_pcm_writei(rsettings->ppcm_handle, &pBuffer, frames ) ) > 0 )
{
printf("\nNo of frames written : %d",framesWritten);
noOfFrames = noOfFrames - framesWritten;
free(descriptors);
}
else if( framesWritten < 0 )
{
if( framesWritten == -EPIPE )
{
snd_pcm_prepare( rsettings->ppcm_handle );
}
else if( framesWritten == -ESTRPIPE )
{
while (-EAGAIN == snd_pcm_resume( rsettings->ppcm_handle ))
usleep(2000);
}
else
{
snd_pcm_prepare( rsettings->ppcm_handle );
}
}
}
}
}
}
}
Can some tell where i am going wrong.

Answer 1

Polling to fill Audio Unit buffers with just the requested amount of data will not work. You have to use the Audio Unit callback routines (which the OS calls using a lower latency real-time thread) to fill the audio buffers.
You can poll to pre-fill a lock-free circular fifo with a sufficient amount of audio samples for the Audio Unit callbacks to copy from.