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
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
|
/*
* Copyright (c) 2012 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_TESTSTEREO_H_
#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_
#include <math.h>
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/main/test/ACMTest.h"
#include "webrtc/modules/audio_coding/main/test/Channel.h"
#include "webrtc/modules/audio_coding/main/test/PCMFile.h"
#define PCMA_AND_PCMU
namespace webrtc {
enum StereoMonoMode {
kNotSet,
kMono,
kStereo
};
class TestPackStereo : public AudioPacketizationCallback {
public:
TestPackStereo();
~TestPackStereo();
void RegisterReceiverACM(AudioCodingModule* acm);
int32_t SendData(const FrameType frame_type,
const uint8_t payload_type,
const uint32_t timestamp,
const uint8_t* payload_data,
const size_t payload_size,
const RTPFragmentationHeader* fragmentation) override;
uint16_t payload_size();
uint32_t timestamp_diff();
void reset_payload_size();
void set_codec_mode(StereoMonoMode mode);
void set_lost_packet(bool lost);
private:
AudioCodingModule* receiver_acm_;
int16_t seq_no_;
uint32_t timestamp_diff_;
uint32_t last_in_timestamp_;
uint64_t total_bytes_;
int payload_size_;
StereoMonoMode codec_mode_;
// Simulate packet losses
bool lost_packet_;
};
class TestStereo : public ACMTest {
public:
explicit TestStereo(int test_mode);
~TestStereo();
void Perform() override;
private:
// The default value of '-1' indicates that the registration is based only on
// codec name and a sampling frequncy matching is not required. This is useful
// for codecs which support several sampling frequency.
void RegisterSendCodec(char side, char* codec_name, int32_t samp_freq_hz,
int rate, int pack_size, int channels,
int payload_type);
void Run(TestPackStereo* channel, int in_channels, int out_channels,
int percent_loss = 0);
void OpenOutFile(int16_t test_number);
void DisplaySendReceiveCodec();
int test_mode_;
rtc::scoped_ptr<AudioCodingModule> acm_a_;
rtc::scoped_ptr<AudioCodingModule> acm_b_;
TestPackStereo* channel_a2b_;
PCMFile* in_file_stereo_;
PCMFile* in_file_mono_;
PCMFile out_file_;
int16_t test_cntr_;
uint16_t pack_size_samp_;
uint16_t pack_size_bytes_;
int counter_;
char* send_codec_name_;
// Payload types for stereo codecs and CNG
#ifdef WEBRTC_CODEC_G722
int g722_pltype_;
#endif
int l16_8khz_pltype_;
int l16_16khz_pltype_;
int l16_32khz_pltype_;
#ifdef PCMA_AND_PCMU
int pcma_pltype_;
int pcmu_pltype_;
#endif
#ifdef WEBRTC_CODEC_OPUS
int opus_pltype_;
#endif
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_TESTSTEREO_H_
|
