Update to the RtcEventLog protobuf to remove the DebugEvent message.

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

 /*
  *  Copyright (c) 2015 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 29 matching lines @@
   const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
   if (!rtp_packet.has_type() || rtp_packet.type() != rtclog::AUDIO ||
       !rtp_packet.has_incoming() || !rtp_packet.incoming() ||
       !rtp_packet.has_packet_length() || rtp_packet.packet_length() == 0 ||
       !rtp_packet.has_header() || rtp_packet.header().size() == 0 ||
       rtp_packet.packet_length() < rtp_packet.header().size())
     return nullptr;
   return &rtp_packet;
 }
 
-const rtclog::DebugEvent* GetAudioOutputEvent(const rtclog::Event& event) {
-  if (!event.has_type() || event.type() != rtclog::Event::DEBUG_EVENT)
+const rtclog::AudioPlayoutEvent* GetAudioPlayoutEvent(
+    const rtclog::Event& event) {
+  if (!event.has_type() || event.type() != rtclog::Event::AUDIO_PLAYOUT_EVENT)
     return nullptr;
-  if (!event.has_timestamp_us() || !event.has_debug_event())
+  if (!event.has_timestamp_us() || !event.has_audio_playout_event())
     return nullptr;
-  const rtclog::DebugEvent& debug_event = event.debug_event();
-  if (!debug_event.has_type() ||
-      debug_event.type() != rtclog::DebugEvent::AUDIO_PLAYOUT)
+  const rtclog::AudioPlayoutEvent& playout_event = event.audio_playout_event();
+  if (!playout_event.has_local_ssrc())
     return nullptr;
-  return &debug_event;
+  return &playout_event;
 }
 
 }  // namespace
 
 RtcEventLogSource* RtcEventLogSource::Create(const std::string& file_name) {
   RtcEventLogSource* source = new RtcEventLogSource();
   RTC_CHECK(source->OpenFile(file_name));
   return source;
 }
 
@@ skipping 30 matching lines @@
       // it can be deleted.
       delete packet;
     }
   }
   return nullptr;
 }
 
 int64_t RtcEventLogSource::NextAudioOutputEventMs() {
   while (audio_output_index_ < event_log_->stream_size()) {
     const rtclog::Event& event = event_log_->stream(audio_output_index_);
-    const rtclog::DebugEvent* debug_event = GetAudioOutputEvent(event);
+    const rtclog::AudioPlayoutEvent* playout_event =
+        GetAudioPlayoutEvent(event);
     audio_output_index_++;
-    if (debug_event)
+    if (playout_event)
       return event.timestamp_us() / 1000;
   }
   return std::numeric_limits<int64_t>::max();
 }
 
 RtcEventLogSource::RtcEventLogSource()
     : PacketSource(), parser_(RtpHeaderParser::Create()) {}
 
 bool RtcEventLogSource::OpenFile(const std::string& file_name) {
   event_log_.reset(new rtclog::EventStream());
   return RtcEventLog::ParseRtcEventLog(file_name, event_log_.get());
 }
 
 }  // namespace test
 }  // namespace webrtc