Created
March 10, 2017 19:53
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| 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?
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Hi, I am very interested in your audio capture module, but I could not understand how to incorporate this into the WebRTC.NET project...
What are the steps to integrate this ? Can you send me a repository where this is already integrated ?
Thanks!