Fixing Real Time Audio with PortAudio in Windows 10

Question

I created an application a couple of years ago that allowed me to process audio by downmixing a 6 channel or 8 channel a.k.a 5.1 as 7.1 as matrixed stereo encoded for that purpose I used the portaudio library with great results this is an example of the open stream function and callback to downmix a 7.1 signal

 Pa_OpenStream(&Flujo, &inputParameters, &outParameters, SAMPLE_RATE, 1, paClipOff, ptrFunction, NULL);

Notice the use of framesPerBuffer value of just one (1), this is my callback function

int downmixed8channels(const void *input, void *output, unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo * info, PaStreamCallbackFlags state, void * userData)
{
    (void)userData;
    (void)info;
    (void)state;
    (void)framesBuffer;

    float *ptrInput = (float*)input;
    float *ptrOutput = (float*)ouput;

    /*This is a struct to identify samples*/

    AudioSamples->L = ptrInput[0];
    AudioSamples->R = ptrInput[1];
    AudioSamples->C = ptrInput[2];
    AudioSamples->LFE = ptrInput[3];
    AudioSamples->RL = ptrInput[4];
    AudioSamples->RR = ptrInput[5];
    AudioSamples->SL = ptrInput[6];
    AudioSamples->SR = ptrInput[7];

    Encoder->8channels(AudioSamples->L,
        AudioSamples->R,
        AudioSamples->C,
        AudioSamples->LFE,
        MuestrasdeAudio->SL,
        MuestrasdeAudio->SR,
        MuestrasdeAudio->RL,
        MuestrasdeAudio->RR,);


    ptrOutput[0] = Encoder->gtLT();
    ptrOutput[1] = Encoder->gtRT();


    return paContinue;
}

As you can see the order set by the index in the output and input buffer correspond to a discrete channel in the case of the output 0 = Left channel, 1 = right Channel. This used to work well, until entered Windows 10 2004, since I updated my system to this new version my audio glitch and I get artifacts like those

Those are captures from the sound of the channel test window under the audio device panel of windows. By the images is clear my program is dropping frames, so the first try to solve this was to use a larger buffer than one to hold samples process them and send then, the reason I did not use a buffer size larger than one in the first place was that the program would drop frames.

But before implementing a I did a proof of concept, would not include audio processing at all, of simple passing of data from input to ouput, for that I set the oputput channelCount parameters to 8 just like the input, resulting in something as simple as this.

    for (int i = 0; i < FramesPerBuffer /*1000*/; i++)
    {
        ptrOutput[i] = ptrOutput[i];
    }

but still the program is still dropping samples.

Next I used two callbacks one for writing to a buffer and a second one to read it and send it to output

    (void)info;
    (void)userData;
    (void)state;
    (void)output;
    
    float* ptrInput = (float*)input;

    for (int i = 0; i < FRAME_SIZE; i++)
    {
        buffer_input[i] = ptrInput[i];
    }

    return paContinue;

Callback to store.

    (void)info;
    (void)userData;
    (void)state;
    (void)output;

    float* ptrOutput = (float*)output;

    for (int i = 0; i < FRAME_SIZE; i++)
    {
        AudioSamples->L = (buffer_input[i] );
        AudioSamples->R = (buffer_input[i++]);
        AudioSamples->C = (buffer_input[i++] );
        AudioSamples->LFE = (buffer_input[i++]);
        AudioSamples->SL = (buffer_input[i++] );
        AudioSamples->SR = (buffer_input[i++]);

        Encoder->Encoder(AudioSamples->L, AudioSamples->R, AudioSamples->C, AudioSamples->LFE, 
        AudioSamples->SL, AudioSamples->SR);

        bufferTransformed[w] = (Encoder->getLT() );
        bufferTransformed[w++] = (Encoder->getRT() );
    }

    w = 0;

    for (int i = 0; i < FRAME_REDUCED; i++)
    {
        ptrOutput[i] = buffer_Transformed[i];
    }

return paContinue;

Callback for processing

The processing callback use a reduced frames per buffer since 2 channel is less than eight since it seems in portaudio a frame is composed of a sample for each audio channel.

This also did not work, since the first problem, is how to syncronize the two callback?, after all of this, what recommendation or advice, can you give me to solve this issue,

Notes: the samplerate must be same for both devices, I implemeted logic in the program to prevent this, the bitdepth is also the same I am using paFloat32, .The portaudio is the modified one use by audacity, since I wanted to use their implementation of WASAPI loopback

Thank very much in advance!.

D.P D.P · Accepted Answer · 2020-12-30T01:41:54

At the end of the day it I did not have to change my callbacks functions in any way, what solved it, was changing or increasing the parameter ".suggestedLatency" of the input and output parameters, to 1.0, even the devices defaultLowOutputLatency or defaultHighOutputLatency values where causing to much glitching, I test it until 1.0 was de sweepspot, higher values did not seen to improve.

TL;DR Increased the suggestedLatency until the glitching is gone.

Fixing Real Time Audio with PortAudio in Windows 10

1 Answers