Frames during Audio playback are consumed very fast

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.