Wire up transport sequence number / send time callbacks to webrtc via libjingle.

talk/session/media/channel.cc

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 19 matching lines @@
 #include "talk/media/base/constants.h"
 #include "talk/media/base/rtputils.h"
 #include "webrtc/p2p/base/transportchannel.h"
 #include "talk/session/media/channelmanager.h"
 #include "webrtc/base/bind.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/dscp.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/call.h"
+#include "webrtc/common_types.h"
 
 namespace cricket {
 
 using rtc::Bind;
 
 enum {
   MSG_EARLYMEDIATIMEOUT = 1,
   MSG_SCREENCASTWINDOWEVENT,
   MSG_RTPPACKET,
   MSG_RTCPPACKET,
@@ skipping 10 matching lines @@
 static const int kAgcMinus10db = -10;
 
 static void SafeSetError(const std::string& message, std::string* error_desc) {
   if (error_desc) {
     *error_desc = message;
   }
 }
 
 struct PacketMessageData : public rtc::MessageData {
   rtc::Buffer packet;
-  rtc::DiffServCodePoint dscp;
+  rtc::PacketOptions options;
 };
 
 struct ScreencastEventMessageData : public rtc::MessageData {
   ScreencastEventMessageData(uint32_t s, rtc::WindowEvent we)
       : ssrc(s), event(we) {}
   uint32_t ssrc;
   rtc::WindowEvent event;
 };
 
 struct VoiceChannelErrorMessageData : public rtc::MessageData {
@@ skipping 335 matching lines @@
 bool BaseChannel::IsReadyToSend() const {
   // Send outgoing data if we are enabled, have local and remote content,
   // and we have had some form of connectivity.
   return enabled() &&
          IsReceiveContentDirection(remote_content_direction_) &&
          IsSendContentDirection(local_content_direction_) &&
          was_ever_writable();
 }
 
 bool BaseChannel::SendPacket(rtc::Buffer* packet,
-                             rtc::DiffServCodePoint dscp) {
-  return SendPacket(false, packet, dscp);
+                             const rtc::PacketOptions& options) {
+  return SendPacket(false, packet, options);
 }
 
 bool BaseChannel::SendRtcp(rtc::Buffer* packet,
-                           rtc::DiffServCodePoint dscp) {
-  return SendPacket(true, packet, dscp);
+                           const rtc::PacketOptions& options) {
+  return SendPacket(true, packet, options);
 }
 
 int BaseChannel::SetOption(SocketType type, rtc::Socket::Option opt,
                            int value) {
   TransportChannel* channel = NULL;
   switch (type) {
     case ST_RTP:
       channel = transport_channel_;
       socket_options_.push_back(
           std::pair<rtc::Socket::Option, int>(opt, value));
@@ skipping 48 matching lines @@
     media_channel_->OnReadyToSend(false);
   }
 }
 
 bool BaseChannel::PacketIsRtcp(const TransportChannel* channel,
                                const char* data, size_t len) {
   return (channel == rtcp_transport_channel_ ||
           rtcp_mux_filter_.DemuxRtcp(data, static_cast<int>(len)));
 }
 
-bool BaseChannel::SendPacket(bool rtcp, rtc::Buffer* packet,
-                             rtc::DiffServCodePoint dscp) {
+bool BaseChannel::SendPacket(bool rtcp,
+                             rtc::Buffer* packet,
+                             const rtc::PacketOptions& options) {
   // SendPacket gets called from MediaEngine, typically on an encoder thread.
   // If the thread is not our worker thread, we will post to our worker
   // so that the real work happens on our worker. This avoids us having to
   // synchronize access to all the pieces of the send path, including
   // SRTP and the inner workings of the transport channels.
   // The only downside is that we can't return a proper failure code if
   // needed. Since UDP is unreliable anyway, this should be a non-issue.
   if (rtc::Thread::Current() != worker_thread_) {
     // Avoid a copy by transferring the ownership of the packet data.
     int message_id = (!rtcp) ? MSG_RTPPACKET : MSG_RTCPPACKET;
     PacketMessageData* data = new PacketMessageData;
     data->packet = packet->Pass();
-    data->dscp = dscp;
+    data->options = options;
     worker_thread_->Post(this, message_id, data);
     return true;
   }
 
   // Now that we are on the correct thread, ensure we have a place to send this
   // packet before doing anything. (We might get RTCP packets that we don't
   // intend to send.) If we've negotiated RTCP mux, send RTCP over the RTP
   // transport.
   TransportChannel* channel = (!rtcp || rtcp_mux_filter_.IsActive()) ?
       transport_channel_ : rtcp_transport_channel_;
   if (!channel || !channel->writable()) {
     return false;
   }
 
   // Protect ourselves against crazy data.
   if (!ValidPacket(rtcp, packet)) {
     LOG(LS_ERROR) << "Dropping outgoing " << content_name_ << " "
                   << PacketType(rtcp)
                   << " packet: wrong size=" << packet->size();
     return false;
   }
 
-  rtc::PacketOptions options(dscp);
+  rtc::PacketOptions updated_options;
+  updated_options = options;
   // Protect if needed.
   if (srtp_filter_.IsActive()) {
     bool res;
     uint8_t* data = packet->data();
     int len = static_cast<int>(packet->size());
     if (!rtcp) {
     // If ENABLE_EXTERNAL_AUTH flag is on then packet authentication is not done
     // inside libsrtp for a RTP packet. A external HMAC module will be writing
     // a fake HMAC value. This is ONLY done for a RTP packet.
     // Socket layer will update rtp sendtime extension header if present in
     // packet with current time before updating the HMAC.
 #if !defined(ENABLE_EXTERNAL_AUTH)
       res = srtp_filter_.ProtectRtp(
           data, len, static_cast<int>(packet->capacity()), &len);
 #else
-      options.packet_time_params.rtp_sendtime_extension_id =
+      updated_options.packet_time_params.rtp_sendtime_extension_id =
           rtp_abs_sendtime_extn_id_;
       res = srtp_filter_.ProtectRtp(
           data, len, static_cast<int>(packet->capacity()), &len,
-          &options.packet_time_params.srtp_packet_index);
+          &updated_options.packet_time_params.srtp_packet_index);
       // If protection succeeds, let's get auth params from srtp.
       if (res) {
         uint8_t* auth_key = NULL;
         int key_len;
         res = srtp_filter_.GetRtpAuthParams(
-            &auth_key, &key_len, &options.packet_time_params.srtp_auth_tag_len);
+            &auth_key, &key_len,
+            &updated_options.packet_time_params.srtp_auth_tag_len);
         if (res) {
-          options.packet_time_params.srtp_auth_key.resize(key_len);
-          options.packet_time_params.srtp_auth_key.assign(auth_key,
-                                                          auth_key + key_len);
+          updated_options.packet_time_params.srtp_auth_key.resize(key_len);
+          updated_options.packet_time_params.srtp_auth_key.assign(
+              auth_key, auth_key + key_len);
         }
       }
 #endif
       if (!res) {
         int seq_num = -1;
         uint32_t ssrc = 0;
         GetRtpSeqNum(data, len, &seq_num);
         GetRtpSsrc(data, len, &ssrc);
         LOG(LS_ERROR) << "Failed to protect " << content_name_
                       << " RTP packet: size=" << len
@@ skipping 557 matching lines @@
         SafeSetError(desc.str(), error_desc);
         ret = false;
       }
     }
   }
   // Check for new streams.
   for (StreamParamsVec::const_iterator it = streams.begin();
        it != streams.end(); ++it) {
     if (!GetStreamBySsrc(local_streams_, it->first_ssrc())) {
       if (media_channel()->AddSendStream(*it)) {
-        LOG(LS_INFO) << "Add send ssrc: " << it->ssrcs[0];
+        LOG(LS_INFO) << "Add send stream ssrc: " << it->ssrcs[0];
       } else {
         std::ostringstream desc;
         desc << "Failed to add send stream ssrc: " << it->first_ssrc();
         SafeSetError(desc.str(), error_desc);
         ret = false;
       }
     }
   }
   local_streams_ = streams;
   return ret;
@@ skipping 80 matching lines @@
       FindHeaderExtension(extensions, kRtpAbsoluteSenderTimeHeaderExtension);
   rtp_abs_sendtime_extn_id_ =
       send_time_extension ? send_time_extension->id : -1;
 }
 
 void BaseChannel::OnMessage(rtc::Message *pmsg) {
   switch (pmsg->message_id) {
     case MSG_RTPPACKET:
     case MSG_RTCPPACKET: {
       PacketMessageData* data = static_cast<PacketMessageData*>(pmsg->pdata);
-      SendPacket(pmsg->message_id == MSG_RTCPPACKET, &data->packet, data->dscp);
+      SendPacket(pmsg->message_id == MSG_RTCPPACKET, &data->packet,
+                 data->options);
       delete data;  // because it is Posted
       break;
     }
     case MSG_FIRSTPACKETRECEIVED: {
       SignalFirstPacketReceived(this);
       break;
     }
   }
 }
 
@@ skipping 1034 matching lines @@
   return (data_channel_type_ == DCT_RTP);
 }
 
 void DataChannel::OnStreamClosedRemotely(uint32_t sid) {
   rtc::TypedMessageData<uint32_t>* message =
       new rtc::TypedMessageData<uint32_t>(sid);
   signaling_thread()->Post(this, MSG_STREAMCLOSEDREMOTELY, message);
 }
 
 }  // namespace cricket