Fix a bug in RtcEventLogSource

webrtc/modules/audio_coding/neteq/tools/packet.cc

 /*
  *  Copyright (c) 2014 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 111 matching lines @@
 void Packet::DeleteRedHeaders(std::list<RTPHeader*>* headers) {
   while (!headers->empty()) {
     delete headers->front();
     headers->pop_front();
   }
 }
 
 bool Packet::ParseHeader(const RtpHeaderParser& parser) {
   bool valid_header = parser.Parse(
       payload_memory_.get(), static_cast<int>(packet_length_bytes_), &header_);
-  assert(valid_header);
-  if (!valid_header) {
+  // Special case for dummy packets that have padding marked in the RTP header.
+  // This causes the RTP header parser to report failure, but is fine in this
+  // context.
+  const bool header_only_with_padding =
+      (header_.headerLength == packet_length_bytes_ &&
+       header_.paddingLength > 0);
+  if (!valid_header && !header_only_with_padding) {
     return false;
   }
   assert(header_.headerLength <= packet_length_bytes_);
   payload_ = &payload_memory_[header_.headerLength];
   assert(packet_length_bytes_ >= header_.headerLength);
   payload_length_bytes_ = packet_length_bytes_ - header_.headerLength;
   RTC_CHECK_GE(virtual_packet_length_bytes_, packet_length_bytes_);
   assert(virtual_packet_length_bytes_ >= header_.headerLength);
   virtual_payload_length_bytes_ =
       virtual_packet_length_bytes_ - header_.headerLength;
@@ skipping 12 matching lines @@
   destination->payload_type_frequency = header_.payload_type_frequency;
   memcpy(&destination->arrOfCSRCs,
          &header_.arrOfCSRCs,
          sizeof(header_.arrOfCSRCs));
   memcpy(
       &destination->extension, &header_.extension, sizeof(header_.extension));
 }
 
 }  // namespace test
 }  // namespace webrtc