Always call RemoteBitrateEstimator::IncomingPacket from 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 1f24b2ca2726c099cef35f215457d08425e6e19a..17da10f35789eb4c864ca6f696d3cfd09da91e9e 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -330,19 +330,6 @@ bool AudioReceiveStream::DeliverRtp(const uint8_t* packet,
     return false;
   }
 
-  // Only forward if the parsed header has one of the headers necessary for
-  // bandwidth estimation. RTP timestamps has different rates for audio and
-  // video and shouldn't be mixed.
-  if (config_.rtp.transport_cc &&
-      header.extension.hasTransportSequenceNumber) {
-    int64_t arrival_time_ms = rtc::TimeMillis();
-    if (packet_time.timestamp >= 0)
-      arrival_time_ms = (packet_time.timestamp + 500) / 1000;
-    size_t payload_size = length - header.headerLength;
-    remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
-                                              header);
-  }
-
   return channel_proxy_->ReceivedRTPPacket(packet, length, packet_time);
 }