Initial implementation of RtpTransportControllerReceive and related interfaces.

webrtc/audio/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 79 matching lines @@
   // cannot do that until we're able to move Channel ownership into the
   // Audio{Send,Receive}Streams.  The best we can do is check that we're not
   // trying to use two different factories using the different interfaces.
   RTC_CHECK(config.decoder_factory);
   RTC_CHECK_EQ(config.decoder_factory,
                channel_proxy_->GetAudioDecoderFactory());
 
   channel_proxy_->RegisterExternalTransport(config.rtcp_send_transport);
   channel_proxy_->SetReceiveCodecs(config.decoder_map);
 
-  for (const auto& extension : config.rtp.extensions) {
-    if (extension.uri == RtpExtension::kAudioLevelUri) {
-      channel_proxy_->SetReceiveAudioLevelIndicationStatus(true, extension.id);
-    } else if (extension.uri == RtpExtension::kTransportSequenceNumberUri) {
-      channel_proxy_->EnableReceiveTransportSequenceNumber(extension.id);
-    } else {
-      RTC_NOTREACHED() << "Unsupported RTP extension.";
-    }
-  }
   // Configure bandwidth estimation.
   channel_proxy_->RegisterReceiverCongestionControlObjects(packet_router);
 
   // Register with transport.
   rtp_stream_receiver_ =
       receiver_controller->CreateReceiver(config_.rtp.remote_ssrc,
                                           channel_proxy_.get());
 }
 
 AudioReceiveStream::~AudioReceiveStream() {
@@ skipping 225 matching lines @@
 int AudioReceiveStream::SetVoiceEnginePlayout(bool playout) {
   ScopedVoEInterface<VoEBase> base(voice_engine());
   if (playout) {
     return base->StartPlayout(config_.voe_channel_id);
   } else {
     return base->StopPlayout(config_.voe_channel_id);
   }
 }
 }  // namespace internal
 }  // namespace webrtc