c# - NAudio 缓冲区已满异常

Question

I try to make a softphone. It passes sound, but after 5 seconds i get "Buffer Full" exception.

Here's my sending code:

public class Media
{
    static WaveInEvent s_WaveIn = new WaveInEvent();
    Action<byte[]> waveHandler;
    public void Capture(Action<byte[]> toRtp)
    {
        s_WaveIn = new WaveInEvent();
        s_WaveIn.WaveFormat = new WaveFormat(8000, 1);//44100, 2);
        waveHandler = toRtp;

        s_WaveIn.DataAvailable += new EventHandler<WaveInEventArgs>(SendCaptureSamples);
        s_WaveIn.StartRecording();
    }

    void SendCaptureSamples(object sender, WaveInEventArgs e)
    {
        waveHandler(e.Buffer);
    }

    public void Stop()
    {
        s_WaveIn.StopRecording();
    }
}

ToRtp is

    private void ToRtp(byte[] buffer)
    {
        myRTP.SequenceNumber++;
        Sender.SendResponse(myRTP.MakePacket(alaw.Encode(buffer,0,buffer.Length)), RTPClient, rtpPort);//ToRTPData(buffer, 8000, 1), myUdpClient);
    }

Receiving:

class Client
{
        WaveFormat pcmFormat = new WaveFormat(8000, 16, 1);
        WaveFormat alawFormat = WaveFormat.CreateALawFormat(8000, 1);

        WaveOut waveOut;
        BufferedWaveProvider waveProvider;
        ALawChatCodec alaw = new ALawChatCodec();

    public Client(IHandlerFactory handlerFactory, IPAddress hostAddress, int portNumber)
    {
        waveOut = new WaveOut();
        waveProvider = new BufferedWaveProvider(pcmFormat);
        waveOut.Init(waveProvider);
        waveOut.Play();
    }

    private void HandleIncomingRTPRequest(IAsyncResult ar)
    {
        IPEndPoint temp = new IPEndPoint(IPAddress.Parse(asteriskip), rtpPort);
        byte[] received = RTPClient.EndReceive(ar, ref temp);
        byte[] decoded = alaw.Decode(received, 12, received.Length - 12);
        waveProvider.AddSamples(decoded, 0, decoded.Length);//Exception occures here
    }
}

I read similar questions, everyone suggest to not use WaveInProvider, but in those questions they don't need to stream sound, they just save it. Why i get this exception, and if it's because of WaveInProvider, how can i stream without it?

EDIT. The problem was that I didn't send correct ACK request via SIP, and I send it after I get OK response to initial INVITE request. As a result, Asterisk sent me another OK response, and when I receive OK to an INVITE request I start streaming sound, so there were multiple streaming threads.

Answer 1

The buffer full exception means you are writing data into the buffer faster than you are reading it out. Which is strange if you are playing audio that is being streamed in real-time. Are you sure that whatever is sending the audio isn't sending you more than you expect?

To mitigate occasional buffer full exceptions you can always clear the BufferedWaveProvider if it gets full. But if it's happening regularly it means you are getting incoming audio faster than you can play it.

Answer 2

As an alternative, if you can't figure out the source of the buffer issue, you can set DiscardOnBufferOverflow = true on the BufferedWaveProvider instance. This is somewhat of a hack, as you'll likely be dropping samples, but it's a quick and dirty workaround if you don't need perfect precision.