Add PeerConnectionInterface::UpdateCallBitrate with call tests.

webrtc/media/engine/fakewebrtccall.h

 /*
  *  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 273 matching lines @@
 
   DeliveryStatus DeliverPacket(webrtc::MediaType media_type,
                                const uint8_t* packet,
                                size_t length,
                                const webrtc::PacketTime& packet_time) override;
 
   webrtc::Call::Stats GetStats() const override;
 
   void SetBitrateConfig(
       const webrtc::Call::Config::BitrateConfig& bitrate_config) override;
+  webrtc::RTCError SetBitrateConfigMask(
+      const webrtc::Call::Config::BitrateConfigMask& mask) override;
   void OnNetworkRouteChanged(const std::string& transport_name,
                              const rtc::NetworkRoute& network_route) override {}
   void SignalChannelNetworkState(webrtc::MediaType media,
                                  webrtc::NetworkState state) override;
   void OnTransportOverheadChanged(webrtc::MediaType media,
                                   int transport_overhead_per_packet) override;
   void OnSentPacket(const rtc::SentPacket& sent_packet) override;
 
   webrtc::Call::Config config_;
   webrtc::NetworkState audio_network_state_;
@@ skipping 10 matching lines @@
 
   int num_created_send_streams_;
   int num_created_receive_streams_;
 
   int audio_transport_overhead_;
   int video_transport_overhead_;
 };
 
 }  // namespace cricket
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_