Fix issues with RestartingSendStreamPreservesRtpStatesWithRtx

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 248 matching lines @@
   while (!last) {
     uint8_t dataBuffer[IP_PACKET_SIZE] = {0};
     size_t payload_bytes_in_packet = 0;
 
     if (!packetizer->NextPacket(&dataBuffer[rtp_header_length],
                                 &payload_bytes_in_packet, &last)) {
       return -1;
     }
 
     // Write RTP header.
-    rtp_sender_->BuildRtpHeader(dataBuffer, payload_type, last,
-                                capture_timestamp, capture_time_ms);
+    int32_t header_length = rtp_sender_->BuildRtpHeader(
+        dataBuffer, payload_type, last, capture_timestamp, capture_time_ms);
+    if (header_length <= 0)
+      return -1;
 
     // 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
     // (HEVC)). The MTSI client may also add the payload bytes onto the last RTP
     // packet in each group of packets which make up another type of frame
     // (e.g. a P-Frame) only if the current value is different from the previous
@@ skipping 58 matching lines @@
   rtc::CritScope cs(&crit_);
   return retransmission_settings_;
 }
 
 void RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) {
   rtc::CritScope cs(&crit_);
   retransmission_settings_ = settings;
 }
 
 }  // namespace webrtc