Make neteq_rtpplay parse RTP header extensions

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

 /*
  *  Copyright (c) 2016 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 42 matching lines @@
     packet_data->payload.SetData(packet_->payload(),
                                  packet_->payload_length_bytes());
   }
   packet_data->time_ms = packet_->time_ms();
 
   LoadNextPacket();
 
   return packet_data;
 }
 
-NetEqRtpDumpInput::NetEqRtpDumpInput(const std::string& file_name)
+NetEqRtpDumpInput::NetEqRtpDumpInput(const std::string& file_name,
+                                     const RtpHeaderExtensionMap& hdr_ext_map)
     : source_(RtpFileSource::Create(file_name)) {
+  for (const auto& ext_pair : hdr_ext_map) {
+    source_->RegisterRtpHeaderExtension(ext_pair.second, ext_pair.first);
+  }
   LoadNextPacket();
 }
 
 rtc::Optional<int64_t> NetEqRtpDumpInput::NextOutputEventTime() const {
   return next_output_event_ms_;
 }
 
 void NetEqRtpDumpInput::AdvanceOutputEvent() {
   if (next_output_event_ms_) {
     *next_output_event_ms_ += kOutputPeriodMs;
   }
   if (!NextPacketTime()) {
     next_output_event_ms_ = rtc::Optional<int64_t>();
   }
 }
 
 PacketSource* NetEqRtpDumpInput::source() {
   return source_.get();
 }
 
-NetEqEventLogInput::NetEqEventLogInput(const std::string& file_name)
+NetEqEventLogInput::NetEqEventLogInput(const std::string& file_name,
+                                       const RtpHeaderExtensionMap& hdr_ext_map)
     : source_(RtcEventLogSource::Create(file_name)) {
+  for (const auto& ext_pair : hdr_ext_map) {
+    source_->RegisterRtpHeaderExtension(ext_pair.second, ext_pair.first);
+  }
   LoadNextPacket();
   AdvanceOutputEvent();
 }
 
 rtc::Optional<int64_t> NetEqEventLogInput::NextOutputEventTime() const {
   return rtc::Optional<int64_t>(next_output_event_ms_);
 }
 
 void NetEqEventLogInput::AdvanceOutputEvent() {
   next_output_event_ms_ =
       rtc::Optional<int64_t>(source_->NextAudioOutputEventMs());
   if (*next_output_event_ms_ == std::numeric_limits<int64_t>::max()) {
     next_output_event_ms_ = rtc::Optional<int64_t>();
   }
 }
 
 PacketSource* NetEqEventLogInput::source() {
   return source_.get();
 }
 
 }  // namespace test
 }  // namespace webrtc