private void m_pToggleSend_Click(object sender,EventArgs e)
{
if(m_pAudioInRTP == null){
m_pSendStream = m_pSession.CreateSendStream();
m_pAudioInRTP = new AudioIn_RTP(AudioIn.Devices[m_pInDevices.SelectedIndex],20,m_pMainUI.AudioCodecs,m_pSendStream);
m_pAudioInRTP.Start();
m_pTimer = new Timer();
m_pTimer.Interval = 500;
m_pTimer.Tick += delegate(object s1,EventArgs e1){
m_pCodec.Text = m_pAudioInRTP.AudioCodec.Name;
m_pPacketsSent.Text = m_pSendStream.RtpPacketsSent.ToString();
m_pKBSent.Text = Convert.ToString(m_pSendStream.RtpBytesSent / 1000);
};
m_pTimer.Start();
m_pToggleSend.Text = "Stop";
}
else{
m_pTimer.Dispose();
m_pTimer = null;
m_pAudioInRTP.Dispose();
m_pAudioInRTP = null;
m_pSendStream.Close();
m_pSendStream = null;
m_pToggleSend.Text = "Send";
}
}
public void Start()
{
if (IsDisposed)
{
throw new ObjectDisposedException(this.GetType().Name);
}
if (!Started)
{
_hTrigger.Init();
_multiSession = new RTP_MultimediaSession(RTP_Utils.GenerateCNAME());
RTP_Session session = _multiSession.CreateSession(new RTP_Address(IPAddress.Parse(_localIP), _localPort, _localPort + 1), new RTP_Clock(0, VideoRate));
session.Payload = RTP_PayloadTypes.H264;
session.Start();
_sendStream = session.CreateSendStream();
_vSoure = VideoSourceCreator.Instance.GetVideoSource(_videoId);
if (_targets.Count > 0)
{
foreach (string key in _targets.Keys)
{
TargetItem ti = _targets[key];
session.AddTarget(new RTP_Address(ti.IP, ti.Port, ti.Port + 1));
}
startPlay();
}
Started = true;
}
}
/// <summary>
/// Default constructor.
/// </summary>
/// <param name="stream">RTP send stream.</param>
/// <exception cref="ArgumentNullException">Is raised when <b>stream</b> is null reference.</exception>
public RTP_SendStreamInfo(RTP_SendStream stream)
{
if (stream == null)
{
throw new ArgumentNullException("stream");
}
m_pStream = stream;
}
public void SendPPS(RTP_SendStream sender, uint timestamp)
{
if (_pps != null)
{
RTP_Packet packet = new RTP_Packet();
packet.Timestamp = timestamp;
packet.Data = _pps.NaluBytes();
sender.Send(packet);
_ppsStamp = Environment.TickCount;
}
}
private void wfrm_SendMic_FormClosing(object sender, FormClosingEventArgs e)
{
if(m_pTimer != null){
m_pTimer.Dispose();
m_pTimer = null;
}
if(m_pAudioInRTP != null){
m_pAudioInRTP.Dispose();
m_pAudioInRTP = null;
}
if(m_pSendStream != null){
m_pSendStream.Close();
m_pSendStream = null;
}
}
/// <summary>
/// Sends audio to RTP session target(s).
/// </summary>
private void SendAudio()
{
try{
using (FileStream fs = File.OpenRead(m_SendFile)){
RTP_SendStream sendStream = m_pSession.CreateSendStream();
byte[] buffer = new byte[400];
int readedCount = fs.Read(buffer, 0, buffer.Length);
long lastSendTime = DateTime.Now.Ticks;
long packetsSent = 0;
long totalSent = 0;
while (readedCount > 0)
{
if (m_pMainUI.ActiveCodec != null)
{
byte[] encodedData = m_pMainUI.ActiveCodec.Encode(buffer, 0, buffer.Length);
// Send audio frame.
RTP_Packet packet = new RTP_Packet();
packet.Timestamp = m_pSession.RtpClock.RtpTimestamp;
packet.Data = encodedData;
sendStream.Send(packet);
// Read next audio frame.
readedCount = fs.Read(buffer, 0, buffer.Length);
totalSent += encodedData.Length;
packetsSent++;
this.BeginInvoke(new MethodInvoker(delegate(){
m_pCodec.Text = m_pMainUI.ActiveCodec.Name;
m_pPacketsSent.Text = packetsSent.ToString();
m_pKBSent.Text = Convert.ToString(totalSent / 1000);
}));
}
Thread.Sleep(25);
lastSendTime = DateTime.Now.Ticks;
}
sendStream.Close();
}
}
catch (Exception x) {
string dummy = x.Message;
}
}
private void btnCall_Click_1(object sender, RoutedEventArgs e)
{
IsConnected = false;
m_pSession = m_pRtpSession.Sessions[0];
if (m_pAudioInRTP == null)
{
m_pSendStream = m_pSession.CreateSendStream();
}
else
{
m_pAudioInRTP.Dispose();
m_pAudioInRTP = null;
m_pSendStream.Close();
m_pSendStream = null;
}
// _soundSender.Start();
// _soundReceiver.Start();
}
/// <summary>
/// Cleans up any resources being used.
/// </summary>
public void Dispose()
{
if (m_IsDisposed)
{
return;
}
Stop();
m_IsDisposed = true;
this.Error = null;
m_pAudioInDevice = null;
m_pAudioCodecs = null;
m_pRTP_Stream.Session.PayloadChanged -= new EventHandler(m_pRTP_Stream_PayloadChanged);
m_pRTP_Stream = null;
m_pActiveCodec = null;
}
/// <summary>
/// Default constructor.
/// </summary>
/// <param name="audioInDevice">Audio-in device to capture.</param>
/// <param name="audioFrameSize">Audio frame size in milliseconds.</param>
/// <param name="codecs">Audio codecs with RTP payload number. For example: 0-PCMU,8-PCMA.</param>
/// <param name="stream">RTP stream to use for audio sending.</param>
/// <exception cref="ArgumentNullException">Is raised when <b>audioInDevice</b>,<b>codecs</b> or <b>stream</b> is null reference.</exception>
public AudioIn_RTP(AudioInDevice audioInDevice, int audioFrameSize, Dictionary <int, AudioCodec> codecs, RTP_SendStream stream)
{
if (audioInDevice == null)
{
throw new ArgumentNullException("audioInDevice");
}
if (codecs == null)
{
throw new ArgumentNullException("codecs");
}
if (stream == null)
{
throw new ArgumentNullException("stream");
}
m_pAudioInDevice = audioInDevice;
m_AudioFrameSize = audioFrameSize;
m_pAudioCodecs = codecs;
m_pRTP_Stream = stream;
m_pRTP_Stream.Session.PayloadChanged += new EventHandler(m_pRTP_Stream_PayloadChanged);
m_pAudioCodecs.TryGetValue(m_pRTP_Stream.Session.Payload, out m_pActiveCodec);
}
/// <summary>
/// Default constructor.
/// </summary>
/// <param name="stream">RTP send stream.</param>
/// <exception cref="ArgumentNullException">Is raised when <b>stream</b> is null reference.</exception>
public RTP_SendStreamInfo(RTP_SendStream stream)
{
if (stream == null)
{
throw new ArgumentNullException("stream");
}
m_pStream = stream;
}