Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket.

webrtc/call/call.cc

Index: webrtc/call/call.cc
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index e21b0762fe8429f21b87b38916c5c5b772ddb119..cad689533e1b59b2aca923674d81f6a8ed3944bd 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -1211,14 +1211,10 @@ PacketReceiver::DeliveryStatus Call::DeliverRtp(MediaType media_type,
     if (it != video_receive_ssrcs_.end()) {
       received_bytes_per_second_counter_.Add(static_cast<int>(length));
       received_video_bytes_per_second_counter_.Add(static_cast<int>(length));
-      // TODO(brandtr): Notify the BWE of received media packets here.
-      auto status = it->second->DeliverRtp(packet, length, packet_time)
+      auto status = it->second->OnRtpPacket(*parsed_packet)
                         ? DELIVERY_OK
                         : DELIVERY_PACKET_ERROR;
-      // Deliver media packets to FlexFEC subsystem. RTP header extensions need
-      // not be parsed, as FlexFEC is oblivious to the semantic meaning of the
-      // packet contents beyond the 12 byte RTP base header. The BWE is fed
-      // information about these media packets from the regular media pipeline.
+      // Deliver media packets to FlexFEC subsystem.
       if (parsed_packet) {
         auto it_bounds = flexfec_receive_ssrcs_media_.equal_range(ssrc);
         for (auto it = it_bounds.first; it != it_bounds.second; ++it)