Update to the neteq_rtpplay utility to support RtcEventLog input files.

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

 /*
- *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
+ *  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_

[minyue-webrtc, 2015/08/28 14:50:21]

a line break before

[ivoc, 2015/09/01 10:03:50]

Done.

+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
 
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
-
-#include <stdio.h>
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/scoped_ptr.h"
-#include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
 #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
 class RtpHeaderParser;
 
+namespace rtclog {
+class EventStream;
+}  // namespace rtclog
+
 namespace test {
 
-class RtpFileReader;
+class Packet;
 
-class RtpFileSource : public PacketSource {
+class RtcEventLogSource : public PacketSource {
  public:
-  // Creates an RtpFileSource reading from |file_name|. If the file cannot be
-  // opened, or has the wrong format, NULL will be returned.
-  static RtpFileSource* Create(const std::string& file_name);
+  // 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 ~RtpFileSource();
+  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, or if a the data was corrupt.
+  // reached.
   Packet* NextPacket() override;
 
+  // Returns the timestamp of the next audio output event, in milliseconds. A
+  // value of -1 is returned if the end of the file is reached.

[hlundin-webrtc, 2015/08/28 12:06:25]

Is it end of file that is reached, or are there simply no more audio output
events?

[ivoc, 2015/09/01 10:03:50]

Updated to say that there are no more audio output events.

+  int NextAudioOutputEventMs();
+
  private:
-  static const int kFirstLineLength = 40;
-  static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2;
-  static const size_t kPacketHeaderSize = 8;
-
-  RtpFileSource();
+  RtcEventLogSource();
 
   bool OpenFile(const std::string& file_name);
 
-  rtc::scoped_ptr<RtpFileReader> rtp_reader_;
+  int rtp_packet_index_ = 0;
+  int audio_output_index_ = 0;
+
+  rtc::scoped_ptr<rtclog::EventStream> event_log_;
   rtc::scoped_ptr<RtpHeaderParser> parser_;
 
-  DISALLOW_COPY_AND_ASSIGN(RtpFileSource);
+  DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource);
 };
 
 }  // namespace test
 }  // namespace webrtc
-#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_