Introduce RtpStreamReceiver and RtpStreamReceiverControllerInterface.

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.
  */
 
 #include "webrtc/audio/audio_receive_stream.h"
 
 #include <string>
 #include <utility>
 
 #include "webrtc/api/call/audio_sink.h"
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/timeutils.h"
+#include "webrtc/call/rtp_transport_controller_receive_interface.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace webrtc {
 
 std::string AudioReceiveStream::Config::Rtp::ToString() const {
@@ skipping 22 matching lines @@
   ss << ", voe_channel_id: " << voe_channel_id;
   if (!sync_group.empty()) {
     ss << ", sync_group: " << sync_group;
   }
   ss << '}';
   return ss.str();
 }
 
 namespace internal {
 AudioReceiveStream::AudioReceiveStream(
+    RtpTransportControllerReceiveInterface* transport,
     PacketRouter* packet_router,
     const webrtc::AudioReceiveStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
     webrtc::RtcEventLog* event_log)
-    : config_(config),
-      audio_state_(audio_state) {
+    : config_(config), audio_state_(audio_state) {
   LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
   RTC_DCHECK_NE(config_.voe_channel_id, -1);
   RTC_DCHECK(audio_state_.get());
   RTC_DCHECK(packet_router);
 
   module_process_thread_checker_.DetachFromThread();
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
   channel_proxy_->SetRtcEventLog(event_log);
@@ skipping 19 matching lines @@
     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_transport_receiver_ =
+      transport->CreateReceiver(config_.rtp.remote_ssrc, this);

[pthatcher1, 2017/05/17 23:24:53]

So CreateReceiver really means "add sink with a context that removes itself upon
destruction"?  

[nisse-webrtc, 2017/05/18 08:45:42]

Yes. The plan is that it should get additional responsibilities later, like
ownership of rtp state and stats methods.

 }
 
 AudioReceiveStream::~AudioReceiveStream() {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
   if (playing_) {
     Stop();
   }
   channel_proxy_->DisassociateSendChannel();
   channel_proxy_->DeRegisterExternalTransport();
@@ skipping 213 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