1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3 *
4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree.
9 */
10
11 #include "webrtc/test/fake_audio_device.h"
12
13 #include <algorithm>
14
15 #include "testing/gtest/include/gtest/gtest.h"
16 #include "webrtc/modules/media_file/source/media_file_utility.h"
17 #include "webrtc/system_wrappers/interface/clock.h"
18 #include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
19 #include "webrtc/system_wrappers/interface/event_wrapper.h"
20 #include "webrtc/system_wrappers/interface/file_wrapper.h"
21 #include "webrtc/system_wrappers/interface/thread_wrapper.h"
22
23 namespace webrtc {
24 namespace test {
25
FakeAudioDevice(Clock * clock,const std::string & filename)26 FakeAudioDevice::FakeAudioDevice(Clock* clock, const std::string& filename)
27 : audio_callback_(NULL),
28 capturing_(false),
29 captured_audio_(),
30 playout_buffer_(),
31 last_playout_ms_(-1),
32 clock_(clock),
33 tick_(EventWrapper::Create()),
34 lock_(CriticalSectionWrapper::CreateCriticalSection()),
35 file_utility_(new ModuleFileUtility(0)),
36 input_stream_(FileWrapper::Create()) {
37 memset(captured_audio_, 0, sizeof(captured_audio_));
38 memset(playout_buffer_, 0, sizeof(playout_buffer_));
39 // Open audio input file as read-only and looping.
40 EXPECT_EQ(0, input_stream_->OpenFile(filename.c_str(), true, true))
41 << filename;
42 }
43
~FakeAudioDevice()44 FakeAudioDevice::~FakeAudioDevice() {
45 Stop();
46
47 if (thread_.get() != NULL)
48 thread_->Stop();
49 }
50
Init()51 int32_t FakeAudioDevice::Init() {
52 CriticalSectionScoped cs(lock_.get());
53 if (file_utility_->InitPCMReading(*input_stream_.get()) != 0)
54 return -1;
55
56 if (!tick_->StartTimer(true, 10))
57 return -1;
58 thread_.reset(ThreadWrapper::CreateThread(
59 FakeAudioDevice::Run, this, webrtc::kHighPriority, "FakeAudioDevice"));
60 if (thread_.get() == NULL)
61 return -1;
62 unsigned int thread_id;
63 if (!thread_->Start(thread_id)) {
64 thread_.reset();
65 return -1;
66 }
67 return 0;
68 }
69
RegisterAudioCallback(AudioTransport * callback)70 int32_t FakeAudioDevice::RegisterAudioCallback(AudioTransport* callback) {
71 CriticalSectionScoped cs(lock_.get());
72 audio_callback_ = callback;
73 return 0;
74 }
75
Playing() const76 bool FakeAudioDevice::Playing() const {
77 CriticalSectionScoped cs(lock_.get());
78 return capturing_;
79 }
80
PlayoutDelay(uint16_t * delay_ms) const81 int32_t FakeAudioDevice::PlayoutDelay(uint16_t* delay_ms) const {
82 *delay_ms = 0;
83 return 0;
84 }
85
Recording() const86 bool FakeAudioDevice::Recording() const {
87 CriticalSectionScoped cs(lock_.get());
88 return capturing_;
89 }
90
Run(void * obj)91 bool FakeAudioDevice::Run(void* obj) {
92 static_cast<FakeAudioDevice*>(obj)->CaptureAudio();
93 return true;
94 }
95
CaptureAudio()96 void FakeAudioDevice::CaptureAudio() {
97 {
98 CriticalSectionScoped cs(lock_.get());
99 if (capturing_) {
100 int bytes_read = file_utility_->ReadPCMData(
101 *input_stream_.get(), captured_audio_, kBufferSizeBytes);
102 if (bytes_read <= 0)
103 return;
104 int num_samples = bytes_read / 2; // 2 bytes per sample.
105 uint32_t new_mic_level;
106 EXPECT_EQ(0,
107 audio_callback_->RecordedDataIsAvailable(captured_audio_,
108 num_samples,
109 2,
110 1,
111 kFrequencyHz,
112 0,
113 0,
114 0,
115 false,
116 new_mic_level));
117 uint32_t samples_needed = kFrequencyHz / 100;
118 int64_t now_ms = clock_->TimeInMilliseconds();
119 uint32_t time_since_last_playout_ms = now_ms - last_playout_ms_;
120 if (last_playout_ms_ > 0 && time_since_last_playout_ms > 0)
121 samples_needed = std::min(kFrequencyHz / time_since_last_playout_ms,
122 kBufferSizeBytes / 2);
123 uint32_t samples_out = 0;
124 int64_t elapsed_time_ms = -1;
125 int64_t ntp_time_ms = -1;
126 EXPECT_EQ(0,
127 audio_callback_->NeedMorePlayData(samples_needed,
128 2,
129 1,
130 kFrequencyHz,
131 playout_buffer_,
132 samples_out,
133 &elapsed_time_ms,
134 &ntp_time_ms));
135 }
136 }
137 tick_->Wait(WEBRTC_EVENT_INFINITE);
138 }
139
Start()140 void FakeAudioDevice::Start() {
141 CriticalSectionScoped cs(lock_.get());
142 capturing_ = true;
143 }
144
Stop()145 void FakeAudioDevice::Stop() {
146 CriticalSectionScoped cs(lock_.get());
147 capturing_ = false;
148 }
149 } // namespace test
150 } // namespace webrtc
151