/*
 *  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_ENCODEDECODETEST_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_

#include <stdio.h>

#include "ACMTest.h"
#include "audio_coding_module.h"
#include "RTPFile.h"
#include "PCMFile.h"
#include "typedefs.h"

namespace webrtc {

#define MAX_INCOMING_PAYLOAD 8096

// TestPacketization callback which writes the encoded payloads to file
class TestPacketization: public AudioPacketizationCallback {
 public:
  TestPacketization(RTPStream *rtpStream, WebRtc_UWord16 frequency);
  ~TestPacketization();
  virtual WebRtc_Word32 SendData(const FrameType frameType,
                                 const WebRtc_UWord8 payloadType,
                                 const WebRtc_UWord32 timeStamp,
                                 const WebRtc_UWord8* payloadData,
                                 const WebRtc_UWord16 payloadSize,
                                 const RTPFragmentationHeader* fragmentation);

 private:
  static void MakeRTPheader(WebRtc_UWord8* rtpHeader, WebRtc_UWord8 payloadType,
                            WebRtc_Word16 seqNo, WebRtc_UWord32 timeStamp,
                            WebRtc_UWord32 ssrc);
  RTPStream* _rtpStream;
  WebRtc_Word32 _frequency;
  WebRtc_Word16 _seqNo;
};

class Sender {
 public:
  Sender();
  void Setup(AudioCodingModule *acm, RTPStream *rtpStream);
  void Teardown();
  void Run();
  bool Add10MsData();
  bool Process();

  //for auto_test and logging
  WebRtc_UWord8 testMode;
  WebRtc_UWord8 codeId;

 private:
  AudioCodingModule* _acm;
  PCMFile _pcmFile;
  AudioFrame _audioFrame;
  WebRtc_UWord16 _payloadSize;
  WebRtc_UWord32 _timeStamp;
  TestPacketization* _packetization;
};

class Receiver {
 public:
  Receiver();
  void Setup(AudioCodingModule *acm, RTPStream *rtpStream);
  void Teardown();
  void Run();
  bool IncomingPacket();
  bool PlayoutData();

  //for auto_test and logging
  WebRtc_UWord8 codeId;
  WebRtc_UWord8 testMode;

 private:
  AudioCodingModule* _acm;
  bool _rtpEOF;
  RTPStream* _rtpStream;
  PCMFile _pcmFile;
  WebRtc_Word16* _playoutBuffer;
  WebRtc_UWord16 _playoutLengthSmpls;
  WebRtc_UWord8 _incomingPayload[MAX_INCOMING_PAYLOAD];
  WebRtc_UWord16 _payloadSizeBytes;
  WebRtc_UWord16 _realPayloadSizeBytes;
  WebRtc_Word32 _frequency;
  bool _firstTime;
  WebRtcRTPHeader _rtpInfo;
  WebRtc_UWord32 _nextTime;
};

class EncodeDecodeTest: public ACMTest {
 public:
  EncodeDecodeTest();
  EncodeDecodeTest(int testMode);
  virtual void Perform();

  WebRtc_UWord16 _playoutFreq;
  WebRtc_UWord8 _testMode;

 private:
  void EncodeToFile(int fileType, int codeId, int* codePars, int testMode);

 protected:
  Sender _sender;
  Receiver _receiver;
};      

} // namespace webrtc

#endif