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

/*
 *  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.
 */

#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 test {

class RtpFileReader;

class RtpFileSource : 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);

  virtual ~RtpFileSource();

  // 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.
  Packet* NextPacket() override;

 private:
  static const int kFirstLineLength = 40;
  static const int kRtpFileHeaderSize = 4 + 4 + 4 + 2 + 2;
  static const size_t kPacketHeaderSize = 8;

  RtpFileSource();

  bool OpenFile(const std::string& file_name);

  rtc::scoped_ptr<RtpFileReader> rtp_reader_;
  rtc::scoped_ptr<RtpHeaderParser> parser_;

  DISALLOW_COPY_AND_ASSIGN(RtpFileSource);
};

}  // namespace test
}  // namespace webrtc
#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTP_FILE_SOURCE_H_