Count FlexFEC packets in |fec_bitrate_| in RTPSenderVideo.

webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc

 /*
  *  Copyright (c) 2012 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 180 matching lines @@
 
   if (protect_media_packet)
     flexfec_sender_->AddRtpPacketAndGenerateFec(*media_packet);
 
   SendVideoPacket(std::move(media_packet), media_packet_storage);
 
   if (flexfec_sender_->FecAvailable()) {
     std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets =
         flexfec_sender_->GetFecPackets();
     for (auto& fec_packet : fec_packets) {
+      size_t packet_length = fec_packet->size();
       uint32_t timestamp = fec_packet->Timestamp();
       uint16_t seq_num = fec_packet->SequenceNumber();
       if (rtp_sender_->SendToNetwork(std::move(fec_packet), kDontRetransmit,
                                      RtpPacketSender::kLowPriority)) {
-        // TODO(brandtr): Wire up stats here.
+        rtc::CritScope cs(&stats_crit_);
+        fec_bitrate_.Update(packet_length, clock_->TimeInMilliseconds());

[danilchap, 2017/01/23 12:42:39]

add/extend a unittest for this.

[brandtr, 2017/01/24 09:42:27]

Done.

         TRACE_EVENT_INSTANT2(TRACE_DISABLED_BY_DEFAULT("webrtc_rtp"),
                              "Video::PacketFlexfec", "timestamp", timestamp,
                              "seqnum", seq_num);
       } else {
         LOG(LS_WARNING) << "Failed to send FlexFEC packet " << seq_num;
       }
     }
   }
 }
 
@@ skipping 199 matching lines @@
   rtc::CritScope cs(&crit_);
   return retransmission_settings_;
 }
 
 void RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) {
   rtc::CritScope cs(&crit_);
   retransmission_settings_ = settings;
 }
 
 }  // namespace webrtc