EncodeDecodeTest.h

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/*

 *  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 MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_

#define MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_

#include <stdio.h>

#include <string.h>

#include "absl/strings/string_view.h"

#include "api/environment/environment.h"

#include "modules/audio_coding/acm2/acm_receiver.h"

#include "modules/audio_coding/include/audio_coding_module.h"

#include "modules/audio_coding/test/PCMFile.h"

#include "modules/audio_coding/test/RTPFile.h"

#include "modules/include/module_common_types.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, uint16_t frequency);

  ~TestPacketization();

  int32_t SendData(AudioFrameType frameType,

                   uint8_t payloadType,

                   uint32_t timeStamp,

                   const uint8_t* payloadData,

                   size_t payloadSize,

                   int64_t absolute_capture_timestamp_ms) override;

 private:

  static void MakeRTPheader(uint8_t* rtpHeader,

                            uint8_t payloadType,

                            int16_t seqNo,

                            uint32_t timeStamp,

                            uint32_t ssrc);

  RTPStream* _rtpStream;

  int32_t _frequency;

  int16_t _seqNo;

};

class Sender {

 public:

  Sender();

  void Setup(const Environment& env,

             AudioCodingModule* acm,

             RTPStream* rtpStream,

             absl::string_view in_file_name,

             int in_sample_rate,

             int payload_type,

             SdpAudioFormat format);

  void Teardown();

  void Run();

  bool Add10MsData();

 protected:

  AudioCodingModule* _acm;

 private:

  PCMFile _pcmFile;

  AudioFrame _audioFrame;

  TestPacketization* _packetization;

};

class Receiver {

 public:

  Receiver();

  virtual ~Receiver() {}

  void Setup(acm2::AcmReceiver* acm_receiver,

             RTPStream* rtpStream,

             absl::string_view out_file_name,

             size_t channels,

             int file_num);

  void Teardown();

  void Run();

  virtual bool IncomingPacket();

  bool PlayoutData();

 private:

  PCMFile _pcmFile;

  int16_t* _playoutBuffer;

  uint16_t _playoutLengthSmpls;

  int32_t _frequency;

  bool _firstTime;

 protected:

  acm2::AcmReceiver* _acm_receiver;

  uint8_t _incomingPayload[MAX_INCOMING_PAYLOAD];

  RTPStream* _rtpStream;

  RTPHeader _rtpHeader;

  size_t _realPayloadSizeBytes;

  size_t _payloadSizeBytes;

  uint32_t _nextTime;

};

class EncodeDecodeTest {

 public:

  EncodeDecodeTest();

  void Perform();

};

}  // namespace webrtc

#endif  // MODULES_AUDIO_CODING_TEST_ENCODEDECODETEST_H_