public void StartCall(string strRemoteJID)
{
/// Start up our mic and speaker, start up our mixer
///
if (m_bCallActive == true)
return;
if (addresses.Length <= 0)
throw new Exception("No IP addresses on System");
int nPort = GetNextPort();
IPEndPoint ep = new IPEndPoint(addresses[0], nPort);
source = new AudioStreamSource();
/// may need a lock here to make sure we have this session added to our list before the xmpp response gets back, though this should be many times faster than network traffic
MediaSession = new JingleMediaSession(strRemoteJID, ep, App.XMPPClient);
MediaSession.UseStun = true;
MediaSession.AudioRTPStream.UseInternalTimersForPacketPushPull = false;
MediaSession.ClearAllPayloads();
MediaSession.AddKnownAudioPayload(KnownAudioPayload.G722_40); // Only g711, speex can't be encoded real time on the xoom (oops, this is the windows phone, we'll have to try that later)
MediaSession.SendInitiateSession();
}
public void StartCall(string strRemoteJID)
{
/// Start up our mic and speaker, start up our mixer
///
if (m_bCallActive == true)
{
Console.WriteLine("Cal active return");
return;
}
if (addresses.Length <= 0)
{
throw new Exception("No IP addresses on System");
}
int nPort = GetNextPort();
IPEndPoint ep = new IPEndPoint(addresses[0], nPort);
source = new AudioStreamSource();
/// may need a lock here to make sure we have this session added to our list before the xmpp response gets back, though this should be many times faster than network traffic
MediaSession = new JingleMediaSession(strRemoteJID, ep, App.XMPPClient);
MediaSession.UseStun = true;
MediaSession.AudioRTPStream.UseInternalTimersForPacketPushPull = false;
MediaSession.ClearAllPayloads();
MediaSession.AddKnownAudioPayload(KnownAudioPayload.G722_40); // Only g711, speex can't be encoded real time on the xoom (oops, this is the windows phone, we'll have to try that later)
Console.WriteLine(MediaSession.SendInitiateSession());
}
public void StartCall()
{
//stream init
IsCallActive = true;
stream.Start(remote, 50, 50);
source = new AudioStreamSource();
//stream start recv
SpeakerThread = new Thread(new ThreadStart(SpeakerThreadFunction));
SpeakerThread.Name = "Speaker Thread";
SpeakerThread.Start();
MicrophoneThread = new Thread(new ThreadStart(MicrophoneThreadFunction));
MicrophoneThread.Name = "Microphone Thread";
MicrophoneThread.Start();
}
public void StartCall()
{
//stream init
IsCallActive = true;
stream.Start(remote, 50, 50);
source = new AudioStreamSource();
Log("Stream Initialised");
//stream start recv
SpeakerThread = new Thread(new ThreadStart(SpeakerThreadFunction));
SpeakerThread.Name = "Speaker Thread";
SpeakerThread.Start();
MicrophoneThread = new Thread(new ThreadStart(MicrophoneThreadFunction));
MicrophoneThread.Name = "Microphone Thread";
MicrophoneThread.Start();
}
public void SpeakerThreadFunction()
{
source = new AudioStreamSource();
TimeSpan tsPTime = TimeSpan.FromMilliseconds(stream.PTimeReceive);
int nSamplesPerPacket = stream.AudioCodec.AudioFormat.CalculateNumberOfSamplesForDuration(tsPTime);
int nBytesPerPacket = nSamplesPerPacket * stream.AudioCodec.AudioFormat.BytesPerSample;
byte[] bDummySample = new byte[nBytesPerPacket];
source.PacketSize = nBytesPerPacket;
stream.IncomingRTPPacketBuffer.InitialPacketQueueMinimumSize = 4;
stream.IncomingRTPPacketBuffer.PacketSizeShiftMax = 10;
int nMsTook = 0;
Deployment.Current.Dispatcher.BeginInvoke(new EventHandler(SafeStartMediaElement), null, null);
// while (true) { }
/// Get first packet... have to wait for our rtp buffer to fill
byte[] bData = stream.WaitNextPacketSample(true, stream.PTimeReceive * 5, out nMsTook);
if ((bData != null) && (bData.Length > 0))
{
source.Write(bData);
}
DateTime dtNextPacketExpected = DateTime.Now + tsPTime;
System.Diagnostics.Stopwatch WaitPacketWatch = new System.Diagnostics.Stopwatch();
int nDeficit = 0;
while (IsCallActive == true)
{
bData = stream.WaitNextPacketSample(true, stream.PTimeReceive, out nMsTook);
if ((bData != null) && (bData.Length > 0))
{
source.Write(bData);
}
TimeSpan tsRemaining = dtNextPacketExpected - DateTime.Now;
int nMsRemaining = (int)tsRemaining.TotalMilliseconds;
if (nMsRemaining > 0)
{
nMsRemaining += nDeficit;
if (nMsRemaining > 0)
System.Threading.Thread.Sleep(nMsRemaining);
else
{
nDeficit = nMsRemaining;
}
}
else
nDeficit += nMsRemaining;
dtNextPacketExpected += tsPTime;
}
Deployment.Current.Dispatcher.BeginInvoke(new EventHandler(SafeStopMediaElement), null, null);
}
public void SpeakerThreadFunction()
{
source = new AudioStreamSource();
TimeSpan tsPTime = TimeSpan.FromMilliseconds(stream.PTimeReceive);
int nSamplesPerPacket = stream.AudioCodec.AudioFormat.CalculateNumberOfSamplesForDuration(tsPTime);
int nBytesPerPacket = nSamplesPerPacket * stream.AudioCodec.AudioFormat.BytesPerSample;
byte[] bDummySample = new byte[nBytesPerPacket];
source.PacketSize = nBytesPerPacket;
stream.IncomingRTPPacketBuffer.InitialPacketQueueMinimumSize = 4;
stream.IncomingRTPPacketBuffer.PacketSizeShiftMax = 10;
int nMsTook = 0;
Deployment.Current.Dispatcher.BeginInvoke(new EventHandler(SafeStartMediaElement), null, null);
// while (true) { }
/// Get first packet... have to wait for our rtp buffer to fill
byte[] bData = stream.WaitNextPacketSample(true, stream.PTimeReceive * 5, out nMsTook);
if ((bData != null) && (bData.Length > 0))
{
source.Write(bData);
}
DateTime dtNextPacketExpected = DateTime.Now + tsPTime;
System.Diagnostics.Stopwatch WaitPacketWatch = new System.Diagnostics.Stopwatch();
int nDeficit = 0;
while (IsCallActive == true)
{
bData = stream.WaitNextPacketSample(true, stream.PTimeReceive, out nMsTook);
if ((bData != null) && (bData.Length > 0))
{
source.Write(bData);
}
TimeSpan tsRemaining = dtNextPacketExpected - DateTime.Now;
int nMsRemaining = (int)tsRemaining.TotalMilliseconds;
if (nMsRemaining > 0)
{
nMsRemaining += nDeficit;
if (nMsRemaining > 0)
{
System.Threading.Thread.Sleep(nMsRemaining);
}
else
{
nDeficit = nMsRemaining;
}
}
else
{
nDeficit += nMsRemaining;
}
dtNextPacketExpected += tsPTime;
}
Deployment.Current.Dispatcher.BeginInvoke(new EventHandler(SafeStopMediaElement), null, null);
}