Add PeerConnectionInterface::UpdateCallBitrate with call tests.

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 570 matching lines @@
 
 webrtc::Call::Stats FakeCall::GetStats() const {
   return stats_;
 }
 
 void FakeCall::SetBitrateConfig(
     const webrtc::Call::Config::BitrateConfig& bitrate_config) {
   config_.bitrate_config = bitrate_config;
 }
 
+void FakeCall::SetBitrateConfigMask(
+    const webrtc::Call::Config::BitrateConfigMask& mask) {
+  // TODO(zstein): not implemented
+}
+
 void FakeCall::SignalChannelNetworkState(webrtc::MediaType media,
                                          webrtc::NetworkState state) {
   switch (media) {
     case webrtc::MediaType::AUDIO:
       audio_network_state_ = state;
       break;
     case webrtc::MediaType::VIDEO:
       video_network_state_ = state;
       break;
     case webrtc::MediaType::DATA:
@@ skipping 20 matching lines @@
 }
 
 void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
   last_sent_packet_ = sent_packet;
   if (sent_packet.packet_id >= 0) {
     last_sent_nonnegative_packet_id_ = sent_packet.packet_id;
   }
 }
 
 }  // namespace cricket