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c#wcfaudio-streamingnaudioaudio

Send mixed sound from WCF using callbacks

I would like transfer mixed sound from WCF server to all connected clients. Using WCF service callbacks for this. Sound is mixed using naudio library.

Here is little example of server-side (WCF method):

MixingSampleProvider _mixer = new MixingSampleProvider(sound32.WaveFormat);
SampleToWaveProvider _sampleToWave = new SampleToWaveProvider(_mixer);

// service method
byte[] buffer = new byte[1000];
do{
     _sampleToWave.Read(buffer, 0, 1000);
     client.Callback.SendBuffer(buffer);
} while (_isPlaying)

and client-side:

BufferedWaveProvider _bufferedWave = new BufferedWaveProvider(sound32.WaveFormat);
// DirectSoundOut _output = new DirectSoundOut();
WaveOut _output = new WaveOut();

_output.Init(_bufferedWave);

// callback event method
if (_output.PlaybackState != PlaybackState.Playing)
    _bufferedWave.AddSamples(buffer, 0, 1000);

// now in timer_tick event method
// if(_bufferedWave.BufferedDuration.TotalSeconds > 0.5)
//    _output.Play();
// else
//    _output.Pause();

I'm new in this, so I have a few questions. Is this idea a good one? Is there simpler option to handle this?

[EDIT_1] I created test app with local two methods, which should simulate this and I found, that _bufferedWave.BufferedBytes are not clearing when is buffered sound playing (and it will overflow immediately). Can somebody tell me, why?

[EDIT_1] Changed type of _output field from DirectSoundOut to WaveOut and it's helpful. Second change I did was, that I added DispatcherTimer to handle when is buffered duration greater than 0.5 (according naudio MP3Streaming example).

Now, I'm fighting with buffer time. I can hear sound only for time in _timer_Tick event method:

_bufferedWave.BufferedDuration.TotalSeconds > XX // this XX is time I can hear sound

Any ideas or opinions?

Solution

  • I'm not sure that this will work the way you hope. WCF is TCP based and TCP is not designed for broadcasting audio video or anything that requires speed (like games) due to it's constant checking of packets.

    I have previously used naudio to transmit sound over a network to a listener client but for it to work you will need to use UDP.

    If you want this to work over the internet then you will need to look into UDP hole-punching.

    Also before you transmit your audio you should compress it from 16bit to 8bit and then back to 16bit upon receiving it using something like an ALaw or MuLaw Decoder/Encoder.