Wire up PacketTime to ReceiveStreams.

talk/media/webrtc/webrtcvideoengine2.cc

 /*
  * libjingle
  * Copyright 2014 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 1417 matching lines @@
 
 bool WebRtcVideoChannel2::RequestIntraFrame() {
   // TODO(pbos): Implement.
   LOG(LS_VERBOSE) << "SendIntraFrame().";
   return true;
 }
 
 void WebRtcVideoChannel2::OnPacketReceived(
     rtc::Buffer* packet,
     const rtc::PacketTime& packet_time) {
+  const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
+                                              packet_time.not_before);
   const webrtc::PacketReceiver::DeliveryStatus delivery_result =
-      call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO,
-          reinterpret_cast<const uint8_t*>(packet->data()), packet->size());
+      call_->Receiver()->DeliverPacket(
+          webrtc::MediaType::VIDEO,
+          reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
+          webrtc_packet_time);
   switch (delivery_result) {
     case webrtc::PacketReceiver::DELIVERY_OK:
       return;
     case webrtc::PacketReceiver::DELIVERY_PACKET_ERROR:
       return;
     case webrtc::PacketReceiver::DELIVERY_UNKNOWN_SSRC:
       break;
   }
 
   uint32 ssrc = 0;
@@ skipping 19 matching lines @@
     }
   }
 
   switch (unsignalled_ssrc_handler_->OnUnsignalledSsrc(this, ssrc)) {
     case UnsignalledSsrcHandler::kDropPacket:
       return;
     case UnsignalledSsrcHandler::kDeliverPacket:
       break;
   }
 
-  if (call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO,
-          reinterpret_cast<const uint8_t*>(packet->data()), packet->size()) !=
-      webrtc::PacketReceiver::DELIVERY_OK) {
+  if (call_->Receiver()->DeliverPacket(
+          webrtc::MediaType::VIDEO,
+          reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
+          webrtc_packet_time) != webrtc::PacketReceiver::DELIVERY_OK) {
     LOG(LS_WARNING) << "Failed to deliver RTP packet on re-delivery.";
     return;
   }
 }
 
 void WebRtcVideoChannel2::OnRtcpReceived(
     rtc::Buffer* packet,
     const rtc::PacketTime& packet_time) {
-  if (call_->Receiver()->DeliverPacket(webrtc::MediaType::VIDEO,
-          reinterpret_cast<const uint8_t*>(packet->data()), packet->size()) !=
-      webrtc::PacketReceiver::DELIVERY_OK) {
+  const webrtc::PacketTime webrtc_packet_time(packet_time.timestamp,
+                                              packet_time.not_before);
+  if (call_->Receiver()->DeliverPacket(
+          webrtc::MediaType::VIDEO,
+          reinterpret_cast<const uint8_t*>(packet->data()), packet->size(),
+          webrtc_packet_time) != webrtc::PacketReceiver::DELIVERY_OK) {
     LOG(LS_WARNING) << "Failed to deliver RTCP packet.";
   }
 }
 
 void WebRtcVideoChannel2::OnReadyToSend(bool ready) {
   LOG(LS_VERBOSE) << "OnReadyToSend: " << (ready ? "Ready." : "Not ready.");
   call_->SignalNetworkState(ready ? webrtc::kNetworkUp : webrtc::kNetworkDown);
 }
 
 bool WebRtcVideoChannel2::MuteStream(uint32 ssrc, bool mute) {
@@ skipping 1245 matching lines @@
       video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
     }
   }
 
   return video_codecs;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VIDEO