1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
|
/*
* 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 <string>
#include "absl/strings/string_view.h"
#include "api/audio_codecs/audio_format.h"
#include "api/environment/environment.h"
#include "api/neteq/neteq.h"
#include "modules/audio_coding/include/audio_coding_module.h"
#include "modules/audio_coding/test/EncodeDecodeTest.h"
#include "modules/audio_coding/test/RTPFile.h"
namespace webrtc {
class ReceiverWithPacketLoss : public Receiver {
public:
ReceiverWithPacketLoss();
void Setup(NetEq* neteq,
RTPStream* rtpStream,
absl::string_view out_file_name,
int channels,
int file_num,
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(const Environment& env,
AudioCodingModule* acm,
RTPStream* rtpStream,
absl::string_view in_file_name,
int payload_type,
SdpAudioFormat format,
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_;
int expected_loss_rate_;
int actual_loss_rate_;
int burst_length_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_TEST_PACKETLOSSTEST_H_
|
