/// <summary>
/// Default constructor.
/// </summary>
/// <param name="stream">RTP stream.</param>
/// <exception cref="ArgumentNullException">Is raised when <b>stream</b> is null reference.</exception>
public RTP_ReceiveStreamEventArgs(RTP_ReceiveStream stream)
{
if(stream == null){
throw new ArgumentNullException("stream");
}
m_pStream = stream;
}
/// <summary>
/// Default constructor.
/// </summary>
/// <param name="stream">RTP stream.</param>
/// <exception cref="ArgumentNullException">Is raised when <b>stream</b> is null reference.</exception>
public RTP_ReceiveStreamEventArgs(RTP_ReceiveStream stream)
{
if (stream == null)
{
throw new ArgumentNullException("stream");
}
m_pStream = stream;
}
/// <summary>
/// Cleans up any resource being used.
/// </summary>
public void Dispose()
{
if(m_IsDisposed){
return;
}
Stop();
this.Error = null;
m_pAudioOutDevice = null;
m_pRTP_Stream = null;
m_pAudioCodecs = null;
m_pActiveCodec = null;
}
/// <summary>
/// Default constructor.
/// </summary>
/// <param name="audioOutDevice">Audio-out device used to play out RTP audio.</param>
/// <param name="stream">RTP receive stream which audio to play.</param>
/// <param name="codecs">Audio codecs with RTP payload number. For example: 0-PCMU,8-PCMA.</param>
/// <exception cref="ArgumentNullException">Is raised when <b>audioOutDevice</b>,<b>stream</b> or <b>codecs</b> is null reference.</exception>
public AudioOut_RTP(AudioOutDevice audioOutDevice,RTP_ReceiveStream stream,Dictionary<int,AudioCodec> codecs)
{
if(audioOutDevice == null){
throw new ArgumentNullException("audioOutDevice");
}
if(stream == null){
throw new ArgumentNullException("stream");
}
if(codecs == null){
throw new ArgumentNullException("codecs");
}
m_pAudioOutDevice = audioOutDevice;
m_pRTP_Stream = stream;
m_pAudioCodecs = codecs;
}
/// <summary>
/// Is called when RTP session receives new RTP packet.
/// </summary>
/// <param name="packet">RTP packet.</param>
/// <param name="size">Packet size in bytes.</param>
/// <exception cref="ArgumentNullException">Is raised when <b>packet</b> is null reference.</exception>
internal void OnRtpPacketReceived(RTP_Packet packet,int size)
{
if(packet == null){
throw new ArgumentNullException("packet");
}
SetLastRtpPacket(DateTime.Now);
// Passive source and first RTP packet.
if(m_pStream == null){
m_pStream = new RTP_ReceiveStream(this.Session,this,packet.SeqNo);
SetState(RTP_SourceState.Active);
}
m_pStream.Process(packet,size);
}
/// <summary>
/// Default constructor.
/// </summary>
/// <param name="stream">RTP receive stream.</param>
public wfrm_Receive(RTP_ReceiveStream stream)
{
if(stream == null){
throw new ArgumentNullException("stream");
}
// Force window handle creation.
IntPtr handle = this.Handle;
m_pStream = stream;
m_pStream.PacketReceived += new EventHandler<RTP_PacketEventArgs>(m_pStream_PacketReceived);
m_RtpMap = new Dictionary<int,AudioCodec>();
m_RtpMap.Add(0,new PCMU());
m_RtpMap.Add(8,new PCMA());
InitUI();
}
/// <summary>
/// Is called when RTP session receives new RTP packet.
/// </summary>
/// <param name="packet">RTP packet.</param>
/// <param name="size">Packet size in bytes.</param>
/// <exception cref="ArgumentNullException">Is raised when <b>packet</b> is null reference.</exception>
internal void OnRtpPacketReceived(RTP_Packet packet, int size)
{
if (packet == null)
{
throw new ArgumentNullException("packet");
}
SetLastRtpPacket(DateTime.Now);
// Passive source and first RTP packet.
if (m_pStream == null)
{
m_pStream = new RTP_ReceiveStream(this.Session, this, packet.SeqNo);
SetState(RTP_SourceState.Active);
}
m_pStream.Process(packet, size);
}
/// <summary>
/// Default constructor.
/// </summary>
/// <param name="stream">RTP receive stream.</param>
public wfrm_Receive2(RTP_ReceiveStream stream, Robot1 owner)
{
if(stream == null){
throw new ArgumentNullException("stream");
}
if (owner == null)
{
throw new ArgumentNullException("owner");
}
m_pMainUI = owner;
// Force window handle creation.
////////////////////////////////IntPtr handle = this.Handle;
m_pStream = stream;
m_pStream.PacketReceived += new EventHandler<RTP_PacketEventArgs>(m_pStream_PacketReceived);
m_RtpMap = new Dictionary<int,AudioCodec>();
m_RtpMap.Add(0,new PCMU());
m_RtpMap.Add(8,new PCMA());
}
/// <summary>
/// Raises <b>NewReceiveStream</b> event.
/// </summary>
/// <param name="stream">New receive stream.</param>
internal void OnNewReceiveStream(RTP_ReceiveStream stream)
{
if(this.NewReceiveStream != null){
this.NewReceiveStream(this,new RTP_ReceiveStreamEventArgs(stream));
}
}
/// <summary>
/// Sends RTCP report.
/// </summary>
private void SendRtcp()
{
/* RFC 3550 6.4 Sender and Receiver Reports
RTP receivers provide reception quality feedback using RTCP report
packets which may take one of two forms depending upon whether or not
the receiver is also a sender. The only difference between the
sender report (SR) and receiver report (RR) forms, besides the packet
type code, is that the sender report includes a 20-byte sender
information section for use by active senders. The SR is issued if a
site has sent any data packets during the interval since issuing the
last report or the previous one, otherwise the RR is issued.
Both the SR and RR forms include zero or more reception report
blocks, one for each of the synchronization sources from which this
receiver has received RTP data packets since the last report.
Reports are not issued for contributing sources listed in the CSRC
list. Each reception report block provides statistics about the data
received from the particular source indicated in that block. Since a
maximum of 31 reception report blocks will fit in an SR or RR packet,
additional RR packets SHOULD be stacked after the initial SR or RR
packet as needed to contain the reception reports for all sources
heard during the interval since the last report. If there are too
many sources to fit all the necessary RR packets into one compound
RTCP packet without exceeding the MTU of the network path, then only
the subset that will fit into one MTU SHOULD be included in each
interval. The subsets SHOULD be selected round-robin across multiple
intervals so that all sources are reported.
*/
bool we_sent = false;
try{
m_pRtcpSource.SetLastRtcpPacket(DateTime.Now);
RTCP_CompoundPacket compundPacket = new RTCP_CompoundPacket();
RTCP_Packet_RR rr = null;
// Find active send streams.
List<RTP_SendStream> activeSendStreams = new List<RTP_SendStream>();
foreach(RTP_SendStream stream in this.SendStreams){
if(stream.RtcpCyclesSinceWeSent < 2){
activeSendStreams.Add(stream);
we_sent = true;
}
// Notify stream about RTCP cycle.
stream.RtcpCycle();
}
#region SR(s) / RR
// We are sender.
if(we_sent){
// Create SR for each active send stream.
for(int i=0;i<activeSendStreams.Count;i++){
RTP_SendStream sendStream = activeSendStreams[i];
RTCP_Packet_SR sr = new RTCP_Packet_SR(sendStream.Source.SSRC);
sr.NtpTimestamp = RTP_Utils.DateTimeToNTP64(DateTime.Now);
sr.RtpTimestamp = m_pRtpClock.RtpTimestamp;
sr.SenderPacketCount = (uint)sendStream.RtpPacketsSent;
sr.SenderOctetCount = (uint)sendStream.RtpBytesSent;
compundPacket.Packets.Add(sr);
}
}
// We are receiver.
else{
rr = new RTCP_Packet_RR();
rr.SSRC = m_pRtcpSource.SSRC;
compundPacket.Packets.Add(rr);
// Report blocks added later.
}
#endregion
#region SDES
RTCP_Packet_SDES sdes = new RTCP_Packet_SDES();
// Add default SSRC.
RTCP_Packet_SDES_Chunk sdesChunk = new RTCP_Packet_SDES_Chunk(m_pRtcpSource.SSRC,m_pSession.LocalParticipant.CNAME);
// Add next optional SDES item, if any. (We round-robin optional items)
m_pSession.LocalParticipant.AddNextOptionalSdesItem(sdesChunk);
sdes.Chunks.Add(sdesChunk);
// Add all active send streams SSRC -> CNAME. This enusres that all send streams will be mapped to participant.
foreach(RTP_SendStream stream in activeSendStreams){
sdes.Chunks.Add(new RTCP_Packet_SDES_Chunk(stream.Source.SSRC,m_pSession.LocalParticipant.CNAME));
}
compundPacket.Packets.Add(sdes);
#endregion
#region RR filling
/* RR reporting:
Report up to 31 active senders, if more senders, reoprt next with next interval.
Report oldest not reported first,then ventually all sources will be reported with this algorythm.
*/
RTP_Source[] senders = this.Senders;
DateTime[] acitveSourceRRTimes = new DateTime[senders.Length];
RTP_ReceiveStream[] activeSenders = new RTP_ReceiveStream[senders.Length];
int activeSenderCount = 0;
foreach(RTP_Source sender in senders){
// Remote sender sent RTP data during last RTCP interval.
if(!sender.IsLocal && sender.LastRtpPacket > m_RtcpLastTransmission){
acitveSourceRRTimes[activeSenderCount] = sender.LastRRTime;
activeSenders[activeSenderCount] = ((RTP_Source_Remote)sender).Stream;
activeSenderCount++;
}
}
// Create RR is SR report and no RR created yet.
if(rr == null){
rr = new RTCP_Packet_RR();
rr.SSRC = m_pRtcpSource.SSRC;
compundPacket.Packets.Add(rr);
}
// Sort ASC.
Array.Sort(acitveSourceRRTimes,activeSenders,0,activeSenderCount);
// Add up to 31 oldest not reported sources to report.
for(int i=1;i<31;i++){
if((activeSenderCount - i) < 0){
break;
}
rr.ReportBlocks.Add(activeSenders[activeSenderCount - i].CreateReceiverReport());
}
#endregion
// Send RTPC packet.
SendRtcpPacket(compundPacket);
// Timeout conflicting transport addresses, if not conflicting any more.
lock(m_pConflictingEPs){
string[] keys = new string[m_pConflictingEPs.Count];
m_pConflictingEPs.Keys.CopyTo(keys,0);
foreach(string key in keys){
if(m_pConflictingEPs[key].AddMinutes(3) < DateTime.Now){
m_pConflictingEPs.Remove(key);
}
}
}
// Since we must check timing out sources at least once per RTCP interval, so we
// check this before sending RTCP.
TimeOutSsrc();
}
catch(Exception x){
if(this.IsDisposed){
return;
}
m_pSession.OnError(x);
}
m_RtcpLastTransmission = DateTime.Now;
// Schedule next RTCP sending.
Schedule(ComputeRtcpTransmissionInterval(m_pMembers.Count,m_pSenders.Count,m_Bandwidth * 0.25,we_sent,m_RtcpAvgPacketSize,false));
}