webrtc/modules/audio_coding/test/PacketLossTest.h
Karl Wiberg 3ff52ffa22 Remove the useless ACMTest base class
Bug: webrtc:8396
Change-Id: I021a2429910b21ffe4829e0ed51b9290bc715c0c
Reviewed-on: https://webrtc-review.googlesource.com/102884
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24907}
2018-10-01 12:01:44 +00:00

77 lines
2 KiB
C++

/*
* 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 MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
#define MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
#include <memory>
#include <string>
#include "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:
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 // MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_