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
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
|
/*
* Copyright 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.
*/
#include "pc/test/fake_audio_capture_module.h"
#include <string.h>
#include <algorithm>
#include <cstdint>
#include "api/audio/audio_device_defines.h"
#include "api/scoped_refptr.h"
#include "api/test/rtc_error_matchers.h"
#include "api/units/time_delta.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/wait_until.h"
class FakeAdmTest : public ::testing::Test, public webrtc::AudioTransport {
protected:
static const int kMsInSecond = 1000;
FakeAdmTest()
: push_iterations_(0), pull_iterations_(0), rec_buffer_bytes_(0) {
memset(rec_buffer_, 0, sizeof(rec_buffer_));
}
void SetUp() override {
fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
EXPECT_TRUE(fake_audio_capture_module_.get() != nullptr);
}
// Callbacks inherited from webrtc::AudioTransport.
// ADM is pushing data.
int32_t RecordedDataIsAvailable(const void* audioSamples,
const size_t nSamples,
const size_t nBytesPerSample,
const size_t nChannels,
const uint32_t samplesPerSec,
const uint32_t totalDelayMS,
const int32_t clockDrift,
const uint32_t currentMicLevel,
const bool keyPressed,
uint32_t& newMicLevel) override {
webrtc::MutexLock lock(&mutex_);
rec_buffer_bytes_ = nSamples * nBytesPerSample;
if ((rec_buffer_bytes_ == 0) ||
(rec_buffer_bytes_ >
FakeAudioCaptureModule::kNumberSamples *
FakeAudioCaptureModule::kNumberBytesPerSample)) {
ADD_FAILURE();
return -1;
}
memcpy(rec_buffer_, audioSamples, rec_buffer_bytes_);
++push_iterations_;
newMicLevel = currentMicLevel;
return 0;
}
void PullRenderData(int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames,
void* audio_data,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) override {}
// ADM is pulling data.
int32_t NeedMorePlayData(const size_t nSamples,
const size_t nBytesPerSample,
const size_t nChannels,
const uint32_t samplesPerSec,
void* audioSamples,
size_t& nSamplesOut,
int64_t* elapsed_time_ms,
int64_t* ntp_time_ms) override {
webrtc::MutexLock lock(&mutex_);
++pull_iterations_;
const size_t audio_buffer_size = nSamples * nBytesPerSample;
const size_t bytes_out =
RecordedDataReceived()
? CopyFromRecBuffer(audioSamples, audio_buffer_size)
: GenerateZeroBuffer(audioSamples, audio_buffer_size);
nSamplesOut = bytes_out / nBytesPerSample;
*elapsed_time_ms = 0;
*ntp_time_ms = 0;
return 0;
}
int push_iterations() const {
webrtc::MutexLock lock(&mutex_);
return push_iterations_;
}
int pull_iterations() const {
webrtc::MutexLock lock(&mutex_);
return pull_iterations_;
}
webrtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
private:
bool RecordedDataReceived() const { return rec_buffer_bytes_ != 0; }
size_t GenerateZeroBuffer(void* audio_buffer, size_t audio_buffer_size) {
memset(audio_buffer, 0, audio_buffer_size);
return audio_buffer_size;
}
size_t CopyFromRecBuffer(void* audio_buffer, size_t audio_buffer_size) {
EXPECT_EQ(audio_buffer_size, rec_buffer_bytes_);
const size_t min_buffer_size =
std::min(audio_buffer_size, rec_buffer_bytes_);
memcpy(audio_buffer, rec_buffer_, min_buffer_size);
return min_buffer_size;
}
webrtc::AutoThread main_thread_;
mutable webrtc::Mutex mutex_;
int push_iterations_;
int pull_iterations_;
char rec_buffer_[FakeAudioCaptureModule::kNumberSamples *
FakeAudioCaptureModule::kNumberBytesPerSample];
size_t rec_buffer_bytes_;
};
TEST_F(FakeAdmTest, PlayoutTest) {
EXPECT_EQ(0, fake_audio_capture_module_->RegisterAudioCallback(this));
bool stereo_available = false;
EXPECT_EQ(0, fake_audio_capture_module_->StereoPlayoutIsAvailable(
&stereo_available));
EXPECT_TRUE(stereo_available);
EXPECT_NE(0, fake_audio_capture_module_->StartPlayout());
EXPECT_FALSE(fake_audio_capture_module_->PlayoutIsInitialized());
EXPECT_FALSE(fake_audio_capture_module_->Playing());
EXPECT_EQ(0, fake_audio_capture_module_->StopPlayout());
EXPECT_EQ(0, fake_audio_capture_module_->InitPlayout());
EXPECT_TRUE(fake_audio_capture_module_->PlayoutIsInitialized());
EXPECT_FALSE(fake_audio_capture_module_->Playing());
EXPECT_EQ(0, fake_audio_capture_module_->StartPlayout());
EXPECT_TRUE(fake_audio_capture_module_->Playing());
uint16_t delay_ms = 10;
EXPECT_EQ(0, fake_audio_capture_module_->PlayoutDelay(&delay_ms));
EXPECT_EQ(0, delay_ms);
EXPECT_THAT(
webrtc::WaitUntil([&] { return pull_iterations(); }, ::testing::Gt(0),
{.timeout = webrtc::TimeDelta::Millis(kMsInSecond)}),
webrtc::IsRtcOk());
EXPECT_GE(0, push_iterations());
EXPECT_EQ(0, fake_audio_capture_module_->StopPlayout());
EXPECT_FALSE(fake_audio_capture_module_->Playing());
}
TEST_F(FakeAdmTest, RecordTest) {
EXPECT_EQ(0, fake_audio_capture_module_->RegisterAudioCallback(this));
bool stereo_available = false;
EXPECT_EQ(0, fake_audio_capture_module_->StereoRecordingIsAvailable(
&stereo_available));
EXPECT_FALSE(stereo_available);
EXPECT_NE(0, fake_audio_capture_module_->StartRecording());
EXPECT_FALSE(fake_audio_capture_module_->Recording());
EXPECT_EQ(0, fake_audio_capture_module_->StopRecording());
EXPECT_EQ(0, fake_audio_capture_module_->InitRecording());
EXPECT_EQ(0, fake_audio_capture_module_->StartRecording());
EXPECT_TRUE(fake_audio_capture_module_->Recording());
EXPECT_THAT(
webrtc::WaitUntil([&] { return push_iterations(); }, ::testing::Gt(0),
{.timeout = webrtc::TimeDelta::Millis(kMsInSecond)}),
webrtc::IsRtcOk());
EXPECT_GE(0, pull_iterations());
EXPECT_EQ(0, fake_audio_capture_module_->StopRecording());
EXPECT_FALSE(fake_audio_capture_module_->Recording());
}
TEST_F(FakeAdmTest, DuplexTest) {
EXPECT_EQ(0, fake_audio_capture_module_->RegisterAudioCallback(this));
EXPECT_EQ(0, fake_audio_capture_module_->InitPlayout());
EXPECT_EQ(0, fake_audio_capture_module_->StartPlayout());
EXPECT_EQ(0, fake_audio_capture_module_->InitRecording());
EXPECT_EQ(0, fake_audio_capture_module_->StartRecording());
EXPECT_THAT(
webrtc::WaitUntil([&] { return push_iterations(); }, ::testing::Gt(0),
{.timeout = webrtc::TimeDelta::Millis(kMsInSecond)}),
webrtc::IsRtcOk());
EXPECT_THAT(
webrtc::WaitUntil([&] { return pull_iterations(); }, ::testing::Gt(0),
{.timeout = webrtc::TimeDelta::Millis(kMsInSecond)}),
webrtc::IsRtcOk());
EXPECT_EQ(0, fake_audio_capture_module_->StopPlayout());
EXPECT_EQ(0, fake_audio_capture_module_->StopRecording());
}
|
