webrtc/modules/audio_coding/test/PacketLossTest.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_TEST_PACKETLOSSTEST_H_
#define WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_

#include <memory>
#include <string>
#include "webrtc/modules/audio_coding/test/EncodeDecodeTest.h"

namespace webrtc {

class ReceiverWithPacketLoss : public Receiver {
 public:
  ReceiverWithPacketLoss();
  void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
             std::string out_file_name, int channels, int loss_rate,
             int burst_length);
  bool IncomingPacket() override;

 protected:
  bool PacketLost();
  int loss_rate_;
  int burst_length_;
  int packet_counter_;
  int lost_packet_counter_;
  int burst_lost_counter_;
};

class SenderWithFEC : public Sender {
 public:
  SenderWithFEC();
  void Setup(AudioCodingModule *acm, RTPStream *rtpStream,
             std::string in_file_name, int sample_rate, int channels,
             int expected_loss_rate);
  bool SetPacketLossRate(int expected_loss_rate);
  bool SetFEC(bool enable_fec);
 protected:
  int expected_loss_rate_;
};

class PacketLossTest : public ACMTest {
 public:
  PacketLossTest(int channels, int expected_loss_rate_, int actual_loss_rate,
                 int burst_length);
  void Perform();
 protected:
  int channels_;
  std::string in_file_name_;
  int sample_rate_hz_;
  std::unique_ptr<SenderWithFEC> sender_;
  std::unique_ptr<ReceiverWithPacketLoss> receiver_;
  int expected_loss_rate_;
  int actual_loss_rate_;
  int burst_length_;
};

}  // namespace webrtc

#endif  // WEBRTC_MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_