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/*
* 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_MAIN_TEST_PACKETLOSSTEST_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_
#include <string>
#include "webrtc/modules/audio_coding/main/test/EncodeDecodeTest.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.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_;
scoped_ptr<SenderWithFEC> sender_;
scoped_ptr<ReceiverWithPacketLoss> receiver_;
int expected_loss_rate_;
int actual_loss_rate_;
int burst_length_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_PACKETLOSSTEST_H_
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