Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call.

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 45 matching lines @@
   if (!sync_group.empty()) {
     ss << ", sync_group: " << sync_group;
   }
   ss << '}';
   return ss.str();
 }
 
 namespace internal {
 AudioReceiveStream::AudioReceiveStream(
     PacketRouter* packet_router,
-    RemoteBitrateEstimator* remote_bitrate_estimator,
     const webrtc::AudioReceiveStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
     webrtc::RtcEventLog* event_log)
-    : remote_bitrate_estimator_(remote_bitrate_estimator),
-      config_(config),
+    : 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);
-  RTC_DCHECK(remote_bitrate_estimator);
 
   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);
   channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
   // TODO(solenberg): Config NACK history window (which is a packet count),
   // using the actual packet size for the configured codec.
   channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0,
@@ skipping 30 matching lines @@
 AudioReceiveStream::~AudioReceiveStream() {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
   if (playing_) {
     Stop();
   }
   channel_proxy_->DisassociateSendChannel();
   channel_proxy_->DeRegisterExternalTransport();
   channel_proxy_->ResetCongestionControlObjects();
   channel_proxy_->SetRtcEventLog(nullptr);
-  remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc);
 }
 
 void AudioReceiveStream::Start() {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   if (playing_) {
     return;
   }
 
   int error = SetVoiceEnginePlayout(true);
   if (error != 0) {
@@ skipping 197 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