Revert of 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 17da10f35789eb4c864ca6f696d3cfd09da91e9e..1f24b2ca2726c099cef35f215457d08425e6e19a 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -330,6 +330,19 @@
     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);
 }