New parser for event log. Manually parse the outermost EventStream

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

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
 
 #include <memory>
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
+#include "webrtc/call/rtc_event_log_parser.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
 class RtpHeaderParser;
 
-namespace rtclog {
-class EventStream;
-}  // namespace rtclog
-
 namespace test {
 
 class Packet;
 
 class RtcEventLogSource : public PacketSource {
  public:
   // Creates an RtcEventLogSource reading from |file_name|. If the file cannot
   // be opened, or has the wrong format, NULL will be returned.
   static RtcEventLogSource* Create(const std::string& file_name);
 
   virtual ~RtcEventLogSource();
 
   // Registers an RTP header extension and binds it to |id|.
   virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
 
   // Returns a pointer to the next packet. Returns NULL if end of file was
   // reached.
   Packet* NextPacket() override;
 
   // Returns the timestamp of the next audio output event, in milliseconds. The
   // maximum value of int64_t is returned if there are no more audio output
   // events available.
   int64_t NextAudioOutputEventMs();
 
  private:
   RtcEventLogSource();
 
   bool OpenFile(const std::string& file_name);
 
-  int rtp_packet_index_ = 0;
-  int audio_output_index_ = 0;
+  size_t rtp_packet_index_ = 0;
+  size_t audio_output_index_ = 0;
 
-  std::unique_ptr<rtclog::EventStream> event_log_;
+  ParsedRtcEventLog parsed_stream_;
   std::unique_ptr<RtpHeaderParser> parser_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource);
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_