Artem Titov | 2cf8eb9 | 2023-06-30 13:26:09 | [diff] [blame] | 1 | /* |
| 2 | * Copyright (c) 2023 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 | #include "modules/audio_device/test_audio_device_impl.h" |
| 11 | |
| 12 | #include <memory> |
| 13 | #include <utility> |
| 14 | |
| 15 | #include "absl/types/optional.h" |
| 16 | #include "api/array_view.h" |
| 17 | #include "api/task_queue/task_queue_factory.h" |
| 18 | #include "api/units/time_delta.h" |
| 19 | #include "modules/audio_device/include/test_audio_device.h" |
| 20 | #include "rtc_base/checks.h" |
| 21 | #include "rtc_base/synchronization/mutex.h" |
| 22 | #include "rtc_base/task_queue.h" |
| 23 | #include "rtc_base/task_utils/repeating_task.h" |
| 24 | |
| 25 | namespace webrtc { |
| 26 | namespace { |
| 27 | |
| 28 | constexpr int kFrameLengthUs = 10000; |
| 29 | |
| 30 | } |
| 31 | |
| 32 | TestAudioDevice::TestAudioDevice( |
| 33 | TaskQueueFactory* task_queue_factory, |
| 34 | std::unique_ptr<TestAudioDeviceModule::Capturer> capturer, |
| 35 | std::unique_ptr<TestAudioDeviceModule::Renderer> renderer, |
| 36 | float speed) |
| 37 | : task_queue_factory_(task_queue_factory), |
| 38 | capturer_(std::move(capturer)), |
| 39 | renderer_(std::move(renderer)), |
| 40 | process_interval_us_(kFrameLengthUs / speed), |
| 41 | audio_buffer_(nullptr), |
| 42 | rendering_(false), |
| 43 | capturing_(false) { |
| 44 | auto good_sample_rate = [](int sr) { |
| 45 | return sr == 8000 || sr == 16000 || sr == 32000 || sr == 44100 || |
| 46 | sr == 48000; |
| 47 | }; |
| 48 | |
| 49 | if (renderer_) { |
| 50 | const int sample_rate = renderer_->SamplingFrequency(); |
| 51 | playout_buffer_.resize(TestAudioDeviceModule::SamplesPerFrame(sample_rate) * |
| 52 | renderer_->NumChannels(), |
| 53 | 0); |
| 54 | RTC_CHECK(good_sample_rate(sample_rate)); |
| 55 | } |
| 56 | if (capturer_) { |
| 57 | RTC_CHECK(good_sample_rate(capturer_->SamplingFrequency())); |
| 58 | } |
| 59 | } |
| 60 | |
| 61 | AudioDeviceGeneric::InitStatus TestAudioDevice::Init() { |
| 62 | task_queue_ = |
| 63 | std::make_unique<rtc::TaskQueue>(task_queue_factory_->CreateTaskQueue( |
| 64 | "TestAudioDeviceModuleImpl", TaskQueueFactory::Priority::NORMAL)); |
| 65 | |
| 66 | RepeatingTaskHandle::Start(task_queue_->Get(), [this]() { |
| 67 | ProcessAudio(); |
| 68 | return TimeDelta::Micros(process_interval_us_); |
| 69 | }); |
| 70 | return InitStatus::OK; |
| 71 | } |
| 72 | |
| 73 | int32_t TestAudioDevice::PlayoutIsAvailable(bool& available) { |
| 74 | MutexLock lock(&lock_); |
| 75 | available = renderer_ != nullptr; |
| 76 | return 0; |
| 77 | } |
| 78 | |
| 79 | int32_t TestAudioDevice::InitPlayout() { |
| 80 | MutexLock lock(&lock_); |
| 81 | |
| 82 | if (rendering_) { |
| 83 | return -1; |
| 84 | } |
| 85 | |
| 86 | if (audio_buffer_ != nullptr && renderer_ != nullptr) { |
| 87 | // Update webrtc audio buffer with the selected parameters |
| 88 | audio_buffer_->SetPlayoutSampleRate(renderer_->SamplingFrequency()); |
| 89 | audio_buffer_->SetPlayoutChannels(renderer_->NumChannels()); |
| 90 | } |
| 91 | rendering_initialized_ = true; |
| 92 | return 0; |
| 93 | } |
| 94 | |
| 95 | bool TestAudioDevice::PlayoutIsInitialized() const { |
| 96 | MutexLock lock(&lock_); |
| 97 | return rendering_initialized_; |
| 98 | } |
| 99 | |
| 100 | int32_t TestAudioDevice::StartPlayout() { |
| 101 | MutexLock lock(&lock_); |
| 102 | RTC_CHECK(renderer_); |
| 103 | rendering_ = true; |
| 104 | return 0; |
| 105 | } |
| 106 | |
| 107 | int32_t TestAudioDevice::StopPlayout() { |
| 108 | MutexLock lock(&lock_); |
| 109 | rendering_ = false; |
| 110 | return 0; |
| 111 | } |
| 112 | |
| 113 | int32_t TestAudioDevice::RecordingIsAvailable(bool& available) { |
| 114 | MutexLock lock(&lock_); |
| 115 | available = capturer_ != nullptr; |
| 116 | return 0; |
| 117 | } |
| 118 | |
| 119 | int32_t TestAudioDevice::InitRecording() { |
| 120 | MutexLock lock(&lock_); |
| 121 | |
| 122 | if (capturing_) { |
| 123 | return -1; |
| 124 | } |
| 125 | |
| 126 | if (audio_buffer_ != nullptr && capturer_ != nullptr) { |
| 127 | // Update webrtc audio buffer with the selected parameters |
| 128 | audio_buffer_->SetRecordingSampleRate(capturer_->SamplingFrequency()); |
| 129 | audio_buffer_->SetRecordingChannels(capturer_->NumChannels()); |
| 130 | } |
| 131 | capturing_initialized_ = true; |
| 132 | return 0; |
| 133 | } |
| 134 | |
| 135 | bool TestAudioDevice::RecordingIsInitialized() const { |
| 136 | MutexLock lock(&lock_); |
| 137 | return capturing_initialized_; |
| 138 | } |
| 139 | |
| 140 | int32_t TestAudioDevice::StartRecording() { |
| 141 | MutexLock lock(&lock_); |
Artem Titov | 2cf8eb9 | 2023-06-30 13:26:09 | [diff] [blame] | 142 | capturing_ = true; |
| 143 | return 0; |
| 144 | } |
| 145 | |
| 146 | int32_t TestAudioDevice::StopRecording() { |
| 147 | MutexLock lock(&lock_); |
| 148 | capturing_ = false; |
| 149 | return 0; |
| 150 | } |
| 151 | |
| 152 | bool TestAudioDevice::Playing() const { |
| 153 | MutexLock lock(&lock_); |
| 154 | return rendering_; |
| 155 | } |
| 156 | |
| 157 | bool TestAudioDevice::Recording() const { |
| 158 | MutexLock lock(&lock_); |
| 159 | return capturing_; |
| 160 | } |
| 161 | |
| 162 | void TestAudioDevice::ProcessAudio() { |
| 163 | MutexLock lock(&lock_); |
| 164 | if (audio_buffer_ == nullptr) { |
| 165 | return; |
| 166 | } |
Artem Titov | 5993675 | 2023-07-07 13:14:54 | [diff] [blame] | 167 | if (capturing_ && capturer_ != nullptr) { |
Artem Titov | 2cf8eb9 | 2023-06-30 13:26:09 | [diff] [blame] | 168 | // Capture 10ms of audio. 2 bytes per sample. |
| 169 | const bool keep_capturing = capturer_->Capture(&recording_buffer_); |
| 170 | if (recording_buffer_.size() > 0) { |
| 171 | audio_buffer_->SetRecordedBuffer( |
| 172 | recording_buffer_.data(), |
| 173 | recording_buffer_.size() / capturer_->NumChannels(), |
| 174 | absl::make_optional(rtc::TimeNanos())); |
| 175 | audio_buffer_->DeliverRecordedData(); |
| 176 | } |
| 177 | if (!keep_capturing) { |
| 178 | capturing_ = false; |
| 179 | } |
| 180 | } |
| 181 | if (rendering_) { |
| 182 | const int sampling_frequency = renderer_->SamplingFrequency(); |
| 183 | int32_t samples_per_channel = audio_buffer_->RequestPlayoutData( |
| 184 | TestAudioDeviceModule::SamplesPerFrame(sampling_frequency)); |
| 185 | audio_buffer_->GetPlayoutData(playout_buffer_.data()); |
| 186 | size_t samples_out = samples_per_channel * renderer_->NumChannels(); |
| 187 | RTC_CHECK_LE(samples_out, playout_buffer_.size()); |
| 188 | const bool keep_rendering = renderer_->Render( |
| 189 | rtc::ArrayView<const int16_t>(playout_buffer_.data(), samples_out)); |
| 190 | if (!keep_rendering) { |
| 191 | rendering_ = false; |
| 192 | } |
| 193 | } |
| 194 | } |
| 195 | |
| 196 | void TestAudioDevice::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) { |
| 197 | MutexLock lock(&lock_); |
| 198 | RTC_DCHECK(audio_buffer || audio_buffer_); |
| 199 | audio_buffer_ = audio_buffer; |
| 200 | |
| 201 | if (renderer_ != nullptr) { |
| 202 | audio_buffer_->SetPlayoutSampleRate(renderer_->SamplingFrequency()); |
| 203 | audio_buffer_->SetPlayoutChannels(renderer_->NumChannels()); |
| 204 | } |
| 205 | if (capturer_ != nullptr) { |
| 206 | audio_buffer_->SetRecordingSampleRate(capturer_->SamplingFrequency()); |
| 207 | audio_buffer_->SetRecordingChannels(capturer_->NumChannels()); |
| 208 | } |
| 209 | } |
| 210 | |
| 211 | } // namespace webrtc |