session-android/jni/webrtc/modules/audio_coding/main/test/RTPFile.h

/*
 *  Copyright (c) 2012 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_MAIN_TEST_RTPFILE_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_

#include <stdio.h>
#include <queue>

#include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
#include "webrtc/modules/interface/module_common_types.h"
#include "webrtc/system_wrappers/interface/rw_lock_wrapper.h"
#include "webrtc/typedefs.h"

namespace webrtc {

class RTPStream {
 public:
  virtual ~RTPStream() {
  }

  virtual void Write(const uint8_t payloadType, const uint32_t timeStamp,
                     const int16_t seqNo, const uint8_t* payloadData,
                     const uint16_t payloadSize, uint32_t frequency) = 0;

  // Returns the packet's payload size. Zero should be treated as an
  // end-of-stream (in the case that EndOfFile() is true) or an error.
  virtual uint16_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
                        uint16_t payloadSize, uint32_t* offset) = 0;
  virtual bool EndOfFile() const = 0;

 protected:
  void MakeRTPheader(uint8_t* rtpHeader, uint8_t payloadType, int16_t seqNo,
                     uint32_t timeStamp, uint32_t ssrc);

  void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const uint8_t* rtpHeader);
};

class RTPPacket {
 public:
  RTPPacket(uint8_t payloadType, uint32_t timeStamp, int16_t seqNo,
            const uint8_t* payloadData, uint16_t payloadSize,
            uint32_t frequency);

  ~RTPPacket();

  uint8_t payloadType;
  uint32_t timeStamp;
  int16_t seqNo;
  uint8_t* payloadData;
  uint16_t payloadSize;
  uint32_t frequency;
};

class RTPBuffer : public RTPStream {
 public:
  RTPBuffer();

  ~RTPBuffer();

  void Write(const uint8_t payloadType, const uint32_t timeStamp,
             const int16_t seqNo, const uint8_t* payloadData,
             const uint16_t payloadSize, uint32_t frequency);

  uint16_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
                uint16_t payloadSize, uint32_t* offset);

  virtual bool EndOfFile() const;

 private:
  RWLockWrapper* _queueRWLock;
  std::queue<RTPPacket *> _rtpQueue;
};

class RTPFile : public RTPStream {
 public:
  ~RTPFile() {
  }

  RTPFile()
      : _rtpFile(NULL),
        _rtpEOF(false) {
  }

  void Open(const char *outFilename, const char *mode);

  void Close();

  void WriteHeader();

  void ReadHeader();

  void Write(const uint8_t payloadType, const uint32_t timeStamp,
             const int16_t seqNo, const uint8_t* payloadData,
             const uint16_t payloadSize, uint32_t frequency);

  uint16_t Read(WebRtcRTPHeader* rtpInfo, uint8_t* payloadData,
                uint16_t payloadSize, uint32_t* offset);

  bool EndOfFile() const {
    return _rtpEOF;
  }

 private:
  FILE* _rtpFile;
  bool _rtpEOF;
};

}  // namespace webrtc

#endif  // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_RTPFILE_H_