Wire up PacketTime to ReceiveStreams.

talk/media/webrtc/fakewebrtccall.cc

 /*
  * libjingle
  * Copyright 2015 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 296 matching lines @@
   } else {
     delete *it;
     video_receive_streams_.erase(it);
   }
 }
 
 webrtc::PacketReceiver* FakeCall::Receiver() {
   return this;
 }
 
-FakeCall::DeliveryStatus FakeCall::DeliverPacket(webrtc::MediaType media_type,
-                                                 const uint8_t* packet,
-                                                 size_t length) {
+FakeCall::DeliveryStatus FakeCall::DeliverPacket(
+    webrtc::MediaType media_type,
+    const uint8_t* packet,
+    size_t length,
+    const webrtc::PacketTime& packet_time) {
   EXPECT_GE(length, 12u);
   uint32_t ssrc;
   if (!GetRtpSsrc(packet, length, &ssrc))
     return DELIVERY_PACKET_ERROR;
 
   if (media_type == webrtc::MediaType::ANY ||
       media_type == webrtc::MediaType::VIDEO) {
     for (auto receiver : video_receive_streams_) {
       if (receiver->GetConfig().rtp.remote_ssrc == ssrc)
         return DELIVERY_OK;
@@ skipping 29 matching lines @@
 
 void FakeCall::SetBitrateConfig(
     const webrtc::Call::Config::BitrateConfig& bitrate_config) {
   config_.bitrate_config = bitrate_config;
 }
 
 void FakeCall::SignalNetworkState(webrtc::NetworkState state) {
   network_state_ = state;
 }
 }  // namespace cricket