mirror of
https://github.com/oxen-io/session-android.git
synced 2024-12-24 00:37:47 +00:00
67 lines
2.0 KiB
C
67 lines
2.0 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 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_
|