Define Stream base classes

webrtc/video/audio_receive_stream.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 @@
     } else {
       RTC_NOTREACHED() << "Unsupported RTP extension.";
     }
   }
 }
 
 webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
   return webrtc::AudioReceiveStream::Stats();
 }
 
+void AudioReceiveStream::Start() {
+}
+
+void AudioReceiveStream::Stop() {
+}
+
+void AudioReceiveStream::SignalNetworkState(NetworkState state) {
+}
+
 bool AudioReceiveStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   return false;
 }
 
 bool AudioReceiveStream::DeliverRtp(const uint8_t* packet, size_t length) {
   RTPHeader header;
   if (!rtp_header_parser_->Parse(packet, length, &header)) {
     return false;
   }
 
   // Only forward if the parsed header has absolute sender time. RTP time stamps
   // may have different rates for audio and video and shouldn't be mixed.
   if (header.extension.hasAbsoluteSendTime) {
     int64_t arrival_time_ms = TickTime::MillisecondTimestamp();
     size_t payload_size = length - header.headerLength;
     remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
                                               header, false);
   }
   return true;
 }
 }  // namespace internal
 }  // namespace webrtc