WebRTCSample.cs

var bits = 16;
var channels = 1;
var sampleRate = 8000;
var sampleSize = bits / 8 * channels;
var frameSamples = sampleRate / 100; // RTC accepts only 10ms frames so sample rate/1000ms * 10

var duration = 100; // Jitter buffer cache, it should be multiple of 10s
var processDelay = 0;
var renderedTime = 0L;
var syncTime = Stopwatch.StartNew();
using (AudioRTPProcessor processor = new AudioRTPProcessor("", 0, "", ""))
    do
    {
        var data = processor.GetStreamDataAsync(duration).Result;
        unsafe
        {
            fixed (byte* dataPtr = data)
            {
                for (int i = 0; i < data.Length; i += (frameSamples * sampleSize))
                    rtcSession.PushAudioFrame(dataPtr + i, bits, sampleRate, channels, frameSamples);
            }
        }

        renderedTime += duration;
        processDelay = duration - (int)(syncTime.ElapsedMilliseconds - renderedTime);
        if (processDelay < 0) processDelay = 0;

    } while (!processor.AudioDataCompleted && socket.State == WebSocketState.Open && rtcSession.ProcessMessages(processDelay) && continueProcessing);

What is AudioRTPProcessor ?! Is it NuGet Package?