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
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
|
/*
* Copyright (c) 2013 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.
*/
#include "webrtc/test/fake_audio_device.h"
#include <algorithm>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/media_file/source/media_file_utility.h"
#include "webrtc/system_wrappers/include/clock.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
#include "webrtc/system_wrappers/include/file_wrapper.h"
#include "webrtc/system_wrappers/include/thread_wrapper.h"
namespace webrtc {
namespace test {
FakeAudioDevice::FakeAudioDevice(Clock* clock, const std::string& filename)
: audio_callback_(NULL),
capturing_(false),
captured_audio_(),
playout_buffer_(),
last_playout_ms_(-1),
clock_(clock),
tick_(EventTimerWrapper::Create()),
file_utility_(new ModuleFileUtility(0)),
input_stream_(FileWrapper::Create()) {
memset(captured_audio_, 0, sizeof(captured_audio_));
memset(playout_buffer_, 0, sizeof(playout_buffer_));
// Open audio input file as read-only and looping.
EXPECT_EQ(0, input_stream_->OpenFile(filename.c_str(), true, true))
<< filename;
}
FakeAudioDevice::~FakeAudioDevice() {
Stop();
if (thread_.get() != NULL)
thread_->Stop();
}
int32_t FakeAudioDevice::Init() {
rtc::CritScope cs(&lock_);
if (file_utility_->InitPCMReading(*input_stream_.get()) != 0)
return -1;
if (!tick_->StartTimer(true, 10))
return -1;
thread_ = ThreadWrapper::CreateThread(FakeAudioDevice::Run, this,
"FakeAudioDevice");
if (thread_.get() == NULL)
return -1;
if (!thread_->Start()) {
thread_.reset();
return -1;
}
thread_->SetPriority(webrtc::kHighPriority);
return 0;
}
int32_t FakeAudioDevice::RegisterAudioCallback(AudioTransport* callback) {
rtc::CritScope cs(&lock_);
audio_callback_ = callback;
return 0;
}
bool FakeAudioDevice::Playing() const {
rtc::CritScope cs(&lock_);
return capturing_;
}
int32_t FakeAudioDevice::PlayoutDelay(uint16_t* delay_ms) const {
*delay_ms = 0;
return 0;
}
bool FakeAudioDevice::Recording() const {
rtc::CritScope cs(&lock_);
return capturing_;
}
bool FakeAudioDevice::Run(void* obj) {
static_cast<FakeAudioDevice*>(obj)->CaptureAudio();
return true;
}
void FakeAudioDevice::CaptureAudio() {
{
rtc::CritScope cs(&lock_);
if (capturing_) {
int bytes_read = file_utility_->ReadPCMData(
*input_stream_.get(), captured_audio_, kBufferSizeBytes);
if (bytes_read <= 0)
return;
// 2 bytes per sample.
size_t num_samples = static_cast<size_t>(bytes_read / 2);
uint32_t new_mic_level;
EXPECT_EQ(0,
audio_callback_->RecordedDataIsAvailable(captured_audio_,
num_samples,
2,
1,
kFrequencyHz,
0,
0,
0,
false,
new_mic_level));
size_t samples_needed = kFrequencyHz / 100;
int64_t now_ms = clock_->TimeInMilliseconds();
uint32_t time_since_last_playout_ms = now_ms - last_playout_ms_;
if (last_playout_ms_ > 0 && time_since_last_playout_ms > 0) {
samples_needed = std::min(
static_cast<size_t>(kFrequencyHz / time_since_last_playout_ms),
kBufferSizeBytes / 2);
}
size_t samples_out = 0;
int64_t elapsed_time_ms = -1;
int64_t ntp_time_ms = -1;
EXPECT_EQ(0,
audio_callback_->NeedMorePlayData(samples_needed,
2,
1,
kFrequencyHz,
playout_buffer_,
samples_out,
&elapsed_time_ms,
&ntp_time_ms));
}
}
tick_->Wait(WEBRTC_EVENT_INFINITE);
}
void FakeAudioDevice::Start() {
rtc::CritScope cs(&lock_);
capturing_ = true;
}
void FakeAudioDevice::Stop() {
rtc::CritScope cs(&lock_);
capturing_ = false;
}
} // namespace test
} // namespace webrtc
|
