Remove webrtc/stream.h and unutilized inheritance.

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 62 matching lines @@
 }
 
 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);
 }
 
 FakeVideoSendStream::FakeVideoSendStream(
@@ skipping 367 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