Delete media type check in Call::NotifyBweOfReceivedPacket.

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 0e64269d4e4c70391a00be65da2440778bb14197..241a6be6607f0381b25abbca867a236fba9f2d90 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -1277,13 +1277,9 @@ void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
     // 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