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

webrtc/audio/audio_receive_stream.cc

Index: webrtc/audio/audio_receive_stream.cc
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index 17da10f35789eb4c864ca6f696d3cfd09da91e9e..0b2f3e0b09f58b976b5eb8a6ecdfee7c7807493b 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -68,19 +68,16 @@ std::string AudioReceiveStream::Config::ToString() const {
 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),
       rtp_header_parser_(RtpHeaderParser::Create()) {
   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);
   RTC_DCHECK(rtp_header_parser_);
 
   module_process_thread_checker_.DetachFromThread();
@@ -138,7 +135,6 @@ AudioReceiveStream::~AudioReceiveStream() {
   channel_proxy_->DeRegisterExternalTransport();
   channel_proxy_->ResetCongestionControlObjects();
   channel_proxy_->SetRtcEventLog(nullptr);
-  remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc);
 }
 
 void AudioReceiveStream::Start() {