private static byte[] PacketizeSingleNAL(Span <byte> raw_nal, UInt32 rtp_timestamp)
{
// Put the whole NAL into one RTP packet.
// Note some receivers will have maximum buffers and be unable to handle large RTP packets.
// Also with RTP over RTSP there is a limit of 65535 bytes for the RTP packet.
byte[] rtp_packet = new byte[RTPPacketUtil.HeaderLength + raw_nal.Length]; // 12 is header size when there are no CSRCs or extensions
// Create an single RTP fragment
bool rtp_marker = (raw_nal[0] & 0x1F) <= 5;
RTPPacketUtil.WriteHeader(rtp_packet, rtp_version, rtp_padding, rtp_extension, rtp_csrc_count, rtp_marker, rtp_payload_type);
UInt32 empty_sequence_id = 0;
RTPPacketUtil.WriteSequenceNumber(rtp_packet, empty_sequence_id);
RTPPacketUtil.WriteTS(rtp_packet, rtp_timestamp);
UInt32 empty_ssrc = 0;
RTPPacketUtil.WriteSSRC(rtp_packet, empty_ssrc);
// Now append the raw NAL
raw_nal.CopyTo(new Span <byte>(rtp_packet, 12, raw_nal.Length));
return(rtp_packet);
}
private static void PacketizeNAL_FUA(ref List <byte[]> rtp_packets, Span <byte> raw_nal, UInt32 rtp_timestamp)
{
int start_bit = 1;
int end_bit = 0;
// consume first byte of the raw_nal. It is used in the FU header
byte first_byte = raw_nal[0];
raw_nal = raw_nal.Slice(1);
while (raw_nal.Length > 0)
{
int payload_size = Math.Min(RTPPacketUtil.MaxPayloadSize - 2, raw_nal.Length);
if (raw_nal.Length - payload_size == 0)
{
end_bit = 1;
}
//for FU-A the marker is set when this is the last RTP packet
bool rtp_marker = end_bit == 1;
byte[] rtp_packet = new byte[RTPPacketUtil.HeaderLength + 2 + payload_size]; // 2 bytes for FU-A header.
RTPPacketUtil.WriteHeader(rtp_packet, rtp_version, rtp_padding, rtp_extension, rtp_csrc_count, rtp_marker, rtp_payload_type);
UInt32 empty_sequence_id = 0;
RTPPacketUtil.WriteSequenceNumber(rtp_packet, empty_sequence_id);
RTPPacketUtil.WriteTS(rtp_packet, rtp_timestamp);
UInt32 empty_ssrc = 0;
RTPPacketUtil.WriteSSRC(rtp_packet, empty_ssrc);
// Now append the Fragmentation Header (with Start and End marker) and part of the raw_nal
byte f_bit = 0;
byte nri = (byte)((first_byte >> 5) & 0x03); // Part of the 1st byte of the Raw NAL (NAL Reference ID)
byte type = 28; // FU-A Fragmentation
rtp_packet[12] = (byte)((f_bit << 7) + (nri << 5) + type);
rtp_packet[13] = (byte)((start_bit << 7) + (end_bit << 6) + (0 << 5) + (first_byte & 0x1F));
raw_nal.Slice(0, payload_size).CopyTo(new Span <byte>(rtp_packet, 14, payload_size));
raw_nal = raw_nal.Slice(payload_size);
rtp_packets.Add(rtp_packet);
start_bit = 0;
}
}
//TODO: reimplement these using Memory and Span<byte>
public static List <byte[]> PacketizeSamples(Span <byte> smp, ulong tsMs)
{
List <byte[]> rtp_packets = new List <byte[]>();
double diffTs = 0;
while (smp.Length > 0)
{
int dataLen = smp.Length > RTPPacketUtil.MaxPayloadSize ? RTPPacketUtil.MaxPayloadSize : smp.Length;
byte[] packet = new byte[dataLen + RTPPacketUtil.HeaderLength];
Span <byte> packetData = new Span <byte>(packet, RTPPacketUtil.HeaderLength, dataLen);
for (int i = 0; i < dataLen; i += 2)
{
packetData[i] = smp[i + 1];
packetData[i + 1] = smp[i];
}
RTPPacketUtil.WriteHeader(packet, rtp_version, rtp_padding, rtp_extension, rtp_csrc_count, false, rtp_payload_type);
UInt32 empty_sequence_id = 0;
RTPPacketUtil.WriteSequenceNumber(packet, empty_sequence_id);
RTPPacketUtil.WriteTS(packet, (UInt32)((tsMs + diffTs) * 48));
UInt32 empty_ssrc = 0;
RTPPacketUtil.WriteSSRC(packet, empty_ssrc);
rtp_packets.Add(packet);
smp = smp.Slice(dataLen);
diffTs += AudiolenRtpPackeMs;
}
return(rtp_packets);
}
void PushRtspData(StreamKind stream, List <byte[]> rtp_packets, ulong tsMsec)
{
int s = (int)stream;
lock (rtsp_list)
{
// Go through each RTSP connection and output the NAL on the Video Session
foreach (RTSPConnection connection in rtsp_list.ToArray()) // ToArray makes a temp copy of the list.
// This lets us delete items in the foreach
// eg when there is Write Error
{
// Only process Sessions in Play Mode
if (connection.play == false)
{
continue;
}
#if DEBUG && LOG
String connection_type = "";
if (connection[s].client_transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.TCP)
{
connection_type = "TCP";
}
if (connection[s].client_transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP &&
connection[s].client_transport.IsMulticast == false)
{
connection_type = "UDP";
}
if (connection[s].client_transport.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP &&
connection[s].client_transport.IsMulticast == true)
{
connection_type = "Multicast";
}
Console.WriteLine($"[{stream}] Sending video session {connection[s].session_id} {connection_type} Timestamp(ms)={tsMsec} Sequence={connection[s].sequence_number}");
#endif
// There could be more than 1 RTP packet (if the data is fragmented)
Boolean write_error = false;
foreach (byte[] rtp_packet in rtp_packets)
{
// Add the specific data for each transmission
RTPPacketUtil.WriteSequenceNumber(rtp_packet, connection[s].sequence_number);
connection[s].sequence_number++;
// Add the specific SSRC for each transmission
RTPPacketUtil.WriteSSRC(rtp_packet, connection.ssrc);
//if (stream == StreamKind.Video) bin.Write(rtp_packet);
// Send as RTP over RTSP (Interleaved)
if (connection[s].transport_reply.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.TCP)
{
int video_channel = connection[s].transport_reply.Interleaved.First; // second is for RTCP status messages)
object state = new object();
try
{
// send the whole NAL. With RTP over RTSP we do not need to Fragment the NAL (as we do with UDP packets or Multicast)
//session.listener.BeginSendData(video_channel, rtp_packet, new AsyncCallback(session.listener.EndSendData), state);
connection.listener.SendData(video_channel, rtp_packet);
}
catch
{
Console.WriteLine($"[{stream}] Error writing to listener " + connection.listener.RemoteAdress);
write_error = true;
break; // exit out of foreach loop
}
}
// Send as RTP over UDP
if (connection[s].transport_reply.LowerTransport == Rtsp.Messages.RtspTransport.LowerTransportType.UDP && connection[s].transport_reply.IsMulticast == false)
{
try
{
// send the whole NAL. ** We could fragment the RTP packet into smaller chuncks that fit within the MTU
// Send to the IP address of the Client
// Send to the UDP Port the Client gave us in the SETUP command
connection[s].udp_pair.Write_To_Data_Port(rtp_packet, connection.client_hostname, connection[s].client_transport.ClientPort.First);
}
catch (Exception e)
{
Console.WriteLine($"[{stream}] UDP Write Exception " + e.ToString());
Console.WriteLine($"[{stream}] Error writing to listener " + connection.listener.RemoteAdress);
write_error = true;
break; // exit out of foreach loop
}
}
// TODO. Add Multicast
}
if (write_error)
{
Console.WriteLine($"[{stream}] Removing session " + connection[s].session_id + " due to write error");
connection.sessions[s].Close();
rtsp_list.Remove(connection);
connection.Dispose();
}
}
}
}