Delete media type check in Call::NotifyBweOfReceivedPacket.

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 1259 matching lines @@
   if (!use_send_side_bwe && header.extension.hasTransportSequenceNumber) {
     // Inconsistent configuration of send side BWE. Do nothing.
     // TODO(nisse): Without this check, we may produce RTCP feedback
     // packets even when not negotiated. But it would be cleaner to
     // move the check down to RTCPSender::SendFeedbackPacket, which
     // would also help the PacketRouter to select an appropriate rtp
     // module in the case that some, but not all, have RTCP feedback
     // enabled.
     return;
   }
-  // For audio, we only support send side BWE.
-  if (media_type == MediaType::VIDEO ||

[stefan-webrtc, 2017/05/05 12:26:39]

I don't think it's safe to remove this.

Let's say we're receiving audio without header extensions and video with
abs-send-time. When receiving the video packets
RemoteBitrateEstimatorAbsSendTime will be picked in
ReceiveSideCongestionController::WrappingBitrateEstimator::PickEstimatorFromHeader.
When we later receive audio packets without header extension
RemoteBitrateEstimatorSingleStream will be picked instead. Therefore we may end
up toggling between the two, when what we really want is to not include the
audio packets in the estimation.

-      (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
-    receive_side_cc_.OnReceivedPacket(
-        packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
-        header);
-  }
+  receive_side_cc_.OnReceivedPacket(
+      packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
+      header);
 }
 
 }  // namespace internal
 
 }  // namespace webrtc