stefan@webrtc.org | e028410 | 2013-11-18 11:45:11 +0000 | [diff] [blame] | 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 | |
| 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 | |
| 44 | FakeAudioDevice::~FakeAudioDevice() { |
| 45 | Stop(); |
| 46 | |
| 47 | if (thread_.get() != NULL) |
| 48 | thread_->Stop(); |
| 49 | } |
| 50 | |
| 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 | |
| 70 | int32_t FakeAudioDevice::RegisterAudioCallback(AudioTransport* callback) { |
| 71 | CriticalSectionScoped cs(lock_.get()); |
| 72 | audio_callback_ = callback; |
| 73 | return 0; |
| 74 | } |
| 75 | |
| 76 | bool FakeAudioDevice::Playing() const { |
| 77 | CriticalSectionScoped cs(lock_.get()); |
| 78 | return capturing_; |
| 79 | } |
| 80 | |
| 81 | int32_t FakeAudioDevice::PlayoutDelay(uint16_t* delay_ms) const { |
| 82 | *delay_ms = 0; |
| 83 | return 0; |
| 84 | } |
| 85 | |
| 86 | bool FakeAudioDevice::Recording() const { |
| 87 | CriticalSectionScoped cs(lock_.get()); |
| 88 | return capturing_; |
| 89 | } |
| 90 | |
| 91 | bool FakeAudioDevice::Run(void* obj) { |
| 92 | static_cast<FakeAudioDevice*>(obj)->CaptureAudio(); |
| 93 | return true; |
| 94 | } |
| 95 | |
| 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 | EXPECT_EQ(0, |
| 125 | audio_callback_->NeedMorePlayData(samples_needed, |
| 126 | 2, |
| 127 | 1, |
| 128 | kFrequencyHz, |
| 129 | playout_buffer_, |
| 130 | samples_out)); |
| 131 | } |
| 132 | } |
| 133 | tick_->Wait(WEBRTC_EVENT_INFINITE); |
| 134 | } |
| 135 | |
| 136 | void FakeAudioDevice::Start() { |
| 137 | CriticalSectionScoped cs(lock_.get()); |
| 138 | capturing_ = true; |
| 139 | } |
| 140 | |
| 141 | void FakeAudioDevice::Stop() { |
| 142 | CriticalSectionScoped cs(lock_.get()); |
| 143 | capturing_ = false; |
| 144 | } |
| 145 | } // namespace test |
| 146 | } // namespace webrtc |