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

webrtc/audio/audio_receive_stream.h

Index: webrtc/audio/audio_receive_stream.h
diff --git a/webrtc/audio/audio_receive_stream.h b/webrtc/audio/audio_receive_stream.h
index 6721c7ee6531e4af9d1299074983e65ce697ef6b..679a6251cc2fce3f688e39cc25b0f4eb834d1187 100644
--- a/webrtc/audio/audio_receive_stream.h
+++ b/webrtc/audio/audio_receive_stream.h
@@ -23,7 +23,6 @@
 
 namespace webrtc {
 class PacketRouter;
-class RemoteBitrateEstimator;
 class RtcEventLog;
 
 namespace voe {
@@ -38,7 +37,6 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
                                  public Syncable {
  public:
   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);
@@ -78,7 +76,6 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream,
 
   rtc::ThreadChecker worker_thread_checker_;
   rtc::ThreadChecker module_process_thread_checker_;
-  RemoteBitrateEstimator* const remote_bitrate_estimator_;
   const webrtc::AudioReceiveStream::Config config_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
   std::unique_ptr<RtpHeaderParser> rtp_header_parser_;