Switch voice transport to use Call and Stream instead of VoENetwork.

webrtc/media/engine/fakewebrtccall.cc

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 49 matching lines @@
 const webrtc::AudioReceiveStream::Config&
     FakeAudioReceiveStream::GetConfig() const {
   return config_;
 }
 
 void FakeAudioReceiveStream::SetStats(
     const webrtc::AudioReceiveStream::Stats& stats) {
   stats_ = stats;
 }
 
-void FakeAudioReceiveStream::IncrementReceivedPackets() {
-  received_packets_++;
+bool FakeAudioReceiveStream::VerifyLastPacket(const uint8_t* data,
+                                              size_t length) const {
+  return last_packet_ == rtc::Buffer(data, length);
+}
+
+bool FakeAudioReceiveStream::DeliverRtp(const uint8_t* packet,
+                                        size_t length,
+                                        const webrtc::PacketTime& packet_time) {
+  ++received_packets_;
+  last_packet_.SetData(packet, length);
+  return true;
+}
+
+bool FakeAudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
+  return true;
 }
 
 webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const {
   return stats_;
 }
 
 void FakeAudioReceiveStream::SetSink(
     std::unique_ptr<webrtc::AudioSinkInterface> sink) {
   sink_ = std::move(sink);
 }
@@ skipping 320 matching lines @@
       media_type == webrtc::MediaType::VIDEO) {
     for (auto receiver : video_receive_streams_) {
       if (receiver->GetConfig().rtp.remote_ssrc == ssrc)
         return DELIVERY_OK;
     }
   }
   if (media_type == webrtc::MediaType::ANY ||
       media_type == webrtc::MediaType::AUDIO) {
     for (auto receiver : audio_receive_streams_) {
       if (receiver->GetConfig().rtp.remote_ssrc == ssrc) {
-        receiver->IncrementReceivedPackets();
+        receiver->DeliverRtp(packet, length, packet_time);
         return DELIVERY_OK;
       }
     }
   }
   return DELIVERY_UNKNOWN_SSRC;
 }
 
 void FakeCall::SetStats(const webrtc::Call::Stats& stats) {
   stats_ = stats;
 }
@@ skipping 28 matching lines @@
     case webrtc::MediaType::ANY:
       ADD_FAILURE()
           << "SignalChannelNetworkState called with unknown parameter.";
   }
 }
 
 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
   last_sent_packet_ = sent_packet;
 }
 }  // namespace cricket