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
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
|
/*
* Copyright (c) 2011 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_NETEQ_TEST_NETEQ_DECODING_TEST_H_
#define MODULES_AUDIO_CODING_NETEQ_TEST_NETEQ_DECODING_TEST_H_
#include <cstddef>
#include <cstdint>
#include <memory>
#include <set>
#include "absl/strings/string_view.h"
#include "api/audio/audio_frame.h"
#include "api/environment/environment.h"
#include "api/neteq/neteq.h"
#include "api/rtp_headers.h"
#include "modules/audio_coding/neteq/tools/packet.h"
#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
#include "system_wrappers/include/clock.h"
#include "test/gtest.h"
namespace webrtc {
class NetEqDecodingTest : public ::testing::Test {
protected:
// NetEQ must be polled for data once every 10 ms.
// Thus, none of the constants below can be changed.
static constexpr int kTimeStepMs = 10;
static constexpr size_t kBlockSize8kHz = kTimeStepMs * 8;
static constexpr size_t kBlockSize16kHz = kTimeStepMs * 16;
static constexpr size_t kBlockSize32kHz = kTimeStepMs * 32;
static constexpr size_t kBlockSize48kHz = kTimeStepMs * 48;
static constexpr int kInitSampleRateHz = 8000;
NetEqDecodingTest();
virtual void SetUp();
virtual void TearDown();
void OpenInputFile(absl::string_view rtp_file);
void Process();
void DecodeAndCompare(absl::string_view rtp_file,
absl::string_view output_checksum,
absl::string_view network_stats_checksum,
bool gen_ref);
static void PopulateRtpInfo(int frame_index,
int timestamp,
RTPHeader* rtp_info);
static void PopulateCng(int frame_index,
int timestamp,
RTPHeader* rtp_info,
uint8_t* payload,
size_t* payload_len);
void WrapTest(uint16_t start_seq_no,
uint32_t start_timestamp,
const std::set<uint16_t>& drop_seq_numbers,
bool expect_seq_no_wrap,
bool expect_timestamp_wrap);
void LongCngWithClockDrift(double drift_factor,
double network_freeze_ms,
bool pull_audio_during_freeze,
int delay_tolerance_ms,
int max_time_to_speech_ms);
SimulatedClock clock_;
const Environment env_;
std::unique_ptr<NetEq> neteq_;
NetEq::Config config_;
std::unique_ptr<test::RtpFileSource> rtp_source_;
std::unique_ptr<test::Packet> packet_;
AudioFrame out_frame_;
int output_sample_rate_;
int algorithmic_delay_ms_;
};
class NetEqDecodingTestTwoInstances : public NetEqDecodingTest {
public:
NetEqDecodingTestTwoInstances() : NetEqDecodingTest() {}
void SetUp() override;
void CreateSecondInstance();
protected:
std::unique_ptr<NetEq> neteq2_;
NetEq::Config config2_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_NETEQ_TEST_NETEQ_DECODING_TEST_H_
|
