C#

private void Receive()
{
    try
    {
        /*BufferDescription*/
        BufferDescription playbackBufferDescription = new BufferDescription();
        playbackBufferDescription.BufferBytes = waveFormat.AverageBytesPerSecond / 5;
        playbackBufferDescription.Format = waveFormat;

        bbStop = false;
        IPEndPoint remoteEP = new IPEndPoint(IPAddress.Any, 0);

        byte[] byteData;
        byte[] byteDecodedData;

        /*While the voice stream is being recieved */
        while (!bbStop)
        {
            //Receive data.
            byteData = udpClient.Receive(ref remoteEP);
            byteDecodedData = new byte[byteData.Length * 2];

            //I've used G711 for voice stream compression
            //G711 compresses the data by 50%, so I allocate a buffer of double
            //the size to store the decompressed data.
            //byteDecodedData = new byte[byteData.Length * 2];

            //Decompress data using the proper vocoder.
            ALawDecoder.ALawDecode(byteData, out byteDecodedData);

            //Play the data received to the user.
** HERE ==> playbackBuffer = new SecondaryBuffer(playbackBufferDescription, device);
            playbackBuffer.Write(0, byteDecodedData, LockFlag.None);
            playbackBuffer.Play(0, BufferPlayFlags.Default);

            first = false;
            //byteData = null;
            playbackBuffer = null;
            //byteDecodedData = null;
        }
        playbackBuffer.Dispose(); //Here is the problem! It doesn't work! Confused
    }
    catch (Exception ex)
    {
        MessageBox.Show(ex.Message, "Error on Receive", MessageBoxButtons.OK, MessageBoxIcon.Error);
    }
}

//Play the data received to the user.
using (playbackBuffer = new SecondaryBuffer(playbackBufferDescription, device))
{
    playbackBuffer.Write(0, byteDecodedData, LockFlag.None);
    playbackBuffer.Play(0, BufferPlayFlags.Default);
}

//Play the data received to the user.
using (playbackBuffer = new SecondaryBuffer(playbackBufferDescription, device))
{
    playbackBuffer.Write(0, byteDecodedData, LockFlag.None);
    playbackBuffer.Play(0, BufferPlayFlags.Default);
}

playbackBuffer = new SecondaryBuffer(playbackBufferDescription, device);

if (!isPlaying && timeInBuffer > 5)
{
    buffer.Play();
    isPlaying = true;
}
else
{
    isPlaying = // Check if buffer has finished playing
    timeInBuffer = // Check the amount of time left in the buffer
}

playbackBuffer = new SecondaryBuffer(playbackBufferDescription, device)

DaveyM69 wrote:

'circular buffer'

serima nazarian wrote:

my application already plays the buffer without any problem or any cut in the voice

Edit:<br />
Actually, looking at the code again, you may not have too much problem<br />
with jitter or drift, since you're using a new secondary buffer for every<br />
receive.  There will probably be a problem if the GC gets around to collecting<br />
any secondary buffers that are still playing asynchronously, however.

Mark Salsbery
Microsoft MVP - Visual C++

Pedram Behroozi wrote:

string acl = "thisisawhitespace"; or
string acl = "nonblankspace"; or

<acl>nonblankspace</acl> and 
<acl>thisisawhitespace</acl>

Chithra Anoop wrote:

How can we open the selected file in a separate window