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/base/platform_thread.h"
17#include "webrtc/modules/media_file/media_file_utility.h"
18#include "webrtc/system_wrappers/include/clock.h"
19#include "webrtc/system_wrappers/include/event_wrapper.h"
20#include "webrtc/system_wrappers/include/file_wrapper.h"
21
22namespace webrtc {
23namespace test {
24
25FakeAudioDevice::FakeAudioDevice(Clock* clock, const std::string& filename)
26    : audio_callback_(NULL),
27      capturing_(false),
28      captured_audio_(),
29      playout_buffer_(),
30      last_playout_ms_(-1),
31      clock_(clock),
32      tick_(EventTimerWrapper::Create()),
33      thread_(FakeAudioDevice::Run, this, "FakeAudioDevice"),
34      file_utility_(new ModuleFileUtility(0)),
35      input_stream_(FileWrapper::Create()) {
36  memset(captured_audio_, 0, sizeof(captured_audio_));
37  memset(playout_buffer_, 0, sizeof(playout_buffer_));
38  // Open audio input file as read-only and looping.
39  EXPECT_EQ(0, input_stream_->OpenFile(filename.c_str(), true, true))
40      << filename;
41}
42
43FakeAudioDevice::~FakeAudioDevice() {
44  Stop();
45
46  thread_.Stop();
47}
48
49int32_t FakeAudioDevice::Init() {
50  rtc::CritScope cs(&lock_);
51  if (file_utility_->InitPCMReading(*input_stream_.get()) != 0)
52    return -1;
53
54  if (!tick_->StartTimer(true, 10))
55    return -1;
56  thread_.Start();
57  thread_.SetPriority(rtc::kHighPriority);
58  return 0;
59}
60
61int32_t FakeAudioDevice::RegisterAudioCallback(AudioTransport* callback) {
62  rtc::CritScope cs(&lock_);
63  audio_callback_ = callback;
64  return 0;
65}
66
67bool FakeAudioDevice::Playing() const {
68  rtc::CritScope cs(&lock_);
69  return capturing_;
70}
71
72int32_t FakeAudioDevice::PlayoutDelay(uint16_t* delay_ms) const {
73  *delay_ms = 0;
74  return 0;
75}
76
77bool FakeAudioDevice::Recording() const {
78  rtc::CritScope cs(&lock_);
79  return capturing_;
80}
81
82bool FakeAudioDevice::Run(void* obj) {
83  static_cast<FakeAudioDevice*>(obj)->CaptureAudio();
84  return true;
85}
86
87void FakeAudioDevice::CaptureAudio() {
88  {
89    rtc::CritScope cs(&lock_);
90    if (capturing_) {
91      int bytes_read = file_utility_->ReadPCMData(
92          *input_stream_.get(), captured_audio_, kBufferSizeBytes);
93      if (bytes_read <= 0)
94        return;
95      // 2 bytes per sample.
96      size_t num_samples = static_cast<size_t>(bytes_read / 2);
97      uint32_t new_mic_level;
98      EXPECT_EQ(0,
99                audio_callback_->RecordedDataIsAvailable(captured_audio_,
100                                                         num_samples,
101                                                         2,
102                                                         1,
103                                                         kFrequencyHz,
104                                                         0,
105                                                         0,
106                                                         0,
107                                                         false,
108                                                         new_mic_level));
109      size_t samples_needed = kFrequencyHz / 100;
110      int64_t now_ms = clock_->TimeInMilliseconds();
111      uint32_t time_since_last_playout_ms = now_ms - last_playout_ms_;
112      if (last_playout_ms_ > 0 && time_since_last_playout_ms > 0) {
113        samples_needed = std::min(
114            static_cast<size_t>(kFrequencyHz / time_since_last_playout_ms),
115            kBufferSizeBytes / 2);
116      }
117      size_t samples_out = 0;
118      int64_t elapsed_time_ms = -1;
119      int64_t ntp_time_ms = -1;
120      EXPECT_EQ(0,
121                audio_callback_->NeedMorePlayData(samples_needed,
122                                                  2,
123                                                  1,
124                                                  kFrequencyHz,
125                                                  playout_buffer_,
126                                                  samples_out,
127                                                  &elapsed_time_ms,
128                                                  &ntp_time_ms));
129    }
130  }
131  tick_->Wait(WEBRTC_EVENT_INFINITE);
132}
133
134void FakeAudioDevice::Start() {
135  rtc::CritScope cs(&lock_);
136  capturing_ = true;
137}
138
139void FakeAudioDevice::Stop() {
140  rtc::CritScope cs(&lock_);
141  capturing_ = false;
142}
143}  // namespace test
144}  // namespace webrtc
145