Don't disable fec for packets without timing frames extension, even if

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 286 matching lines @@
     return false;
 
   // Create header that will be reused in all packets.
   std::unique_ptr<RtpPacketToSend> rtp_header = rtp_sender_->AllocatePacket();
   rtp_header->SetPayloadType(payload_type);
   rtp_header->SetTimestamp(rtp_timestamp);
   rtp_header->set_capture_time_ms(capture_time_ms);
   auto last_packet = rtc::MakeUnique<RtpPacketToSend>(*rtp_header);
 
   size_t fec_packet_overhead;
-  bool is_timing_frame = false;
   bool red_enabled;
   int32_t retransmission_settings;
   {
     rtc::CritScope cs(&crit_);
     // According to
     // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
     // ts_126114v120700p.pdf Section 7.4.5:
     // The MTSI client shall add the payload bytes as defined in this clause
     // onto the last RTP packet in each group of packets which make up a key
     // frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265
@@ skipping 11 matching lines @@
       last_rotation_ = current_rotation;
       // Report content type only for key frames.
       if (frame_type == kVideoFrameKey &&
           video_header->content_type != VideoContentType::UNSPECIFIED) {
         last_packet->SetExtension<VideoContentTypeExtension>(
             video_header->content_type);
       }
       if (video_header->video_timing.is_timing_frame) {
         last_packet->SetExtension<VideoTimingExtension>(
             video_header->video_timing);
-        is_timing_frame = true;
       }
     }
 
     // FEC settings.
     const FecProtectionParams& fec_params =
         frame_type == kVideoFrameKey ? key_fec_params_ : delta_fec_params_;
     if (flexfec_enabled())
       flexfec_sender_->SetFecParameters(fec_params);
     if (ulpfec_enabled())
       ulpfec_generator_.SetFecParameters(fec_params);
@@ skipping 39 matching lines @@
     if (!packetizer->NextPacket(packet.get()))
       return false;
     RTC_DCHECK_LE(packet->payload_size(),
                   last ? max_data_payload_length - last_packet_reduction_len
                        : max_data_payload_length);
     if (!rtp_sender_->AssignSequenceNumber(packet.get()))
       return false;
 
     bool protect_packet = (packetizer->GetProtectionType() == kProtectedPacket);
     // Put packetization finish timestamp into extension.
-    if (last && is_timing_frame) {
+    if (packet->HasExtension<VideoTimingExtension>()) {
       packet->set_packetization_finish_time_ms(clock_->TimeInMilliseconds());
       // TODO(ilnik): Due to webrtc:7859, packets with timing extensions are not
       // protected by FEC. It reduces FEC efficiency a bit. When FEC is moved
       // below the pacer, it can be re-enabled for these packets.
       // NOTE: Any RTP stream processor in the network, modifying 'network'
       // timestamps in the timing frames extension have to be an end-point for
       // FEC, otherwise recovered by FEC packets will be corrupted.
       protect_packet = false;
     }
 
@@ skipping 39 matching lines @@
   rtc::CritScope cs(&crit_);
   return retransmission_settings_;
 }
 
 void RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) {
   rtc::CritScope cs(&crit_);
   retransmission_settings_ = settings;
 }
 
 }  // namespace webrtc