| /* |
| * Copyright (c) 2018 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 "modules/audio_device/include/test_audio_device.h" |
| |
| #include <algorithm> |
| #include <cstdint> |
| #include <cstdlib> |
| #include <memory> |
| #include <string> |
| #include <type_traits> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/strings/string_view.h" |
| #include "api/array_view.h" |
| #include "api/make_ref_counted.h" |
| #include "common_audio/wav_file.h" |
| #include "modules/audio_device/audio_device_impl.h" |
| #include "modules/audio_device/include/audio_device_default.h" |
| #include "modules/audio_device/test_audio_device_impl.h" |
| #include "rtc_base/buffer.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/event.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| #include "rtc_base/platform_thread.h" |
| #include "rtc_base/random.h" |
| #include "rtc_base/synchronization/mutex.h" |
| #include "rtc_base/task_queue.h" |
| #include "rtc_base/task_utils/repeating_task.h" |
| #include "rtc_base/thread_annotations.h" |
| #include "rtc_base/time_utils.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| |
| constexpr int kFrameLengthUs = 10000; |
| constexpr int kFramesPerSecond = rtc::kNumMicrosecsPerSec / kFrameLengthUs; |
| |
| class TestAudioDeviceModuleImpl : public AudioDeviceModuleImpl { |
| public: |
| TestAudioDeviceModuleImpl( |
| TaskQueueFactory* task_queue_factory, |
| std::unique_ptr<TestAudioDeviceModule::Capturer> capturer, |
| std::unique_ptr<TestAudioDeviceModule::Renderer> renderer, |
| float speed = 1) |
| : AudioDeviceModuleImpl( |
| AudioLayer::kDummyAudio, |
| std::make_unique<TestAudioDevice>(task_queue_factory, |
| std::move(capturer), |
| std::move(renderer), |
| speed), |
| task_queue_factory, |
| /*create_detached=*/true) {} |
| |
| ~TestAudioDeviceModuleImpl() override = default; |
| }; |
| |
| // A fake capturer that generates pulses with random samples between |
| // -max_amplitude and +max_amplitude. |
| class PulsedNoiseCapturerImpl final |
| : public TestAudioDeviceModule::PulsedNoiseCapturer { |
| public: |
| // Assuming 10ms audio packets. |
| PulsedNoiseCapturerImpl(int16_t max_amplitude, |
| int sampling_frequency_in_hz, |
| int num_channels) |
| : sampling_frequency_in_hz_(sampling_frequency_in_hz), |
| fill_with_zero_(false), |
| random_generator_(1), |
| max_amplitude_(max_amplitude), |
| num_channels_(num_channels) { |
| RTC_DCHECK_GT(max_amplitude, 0); |
| } |
| |
| int SamplingFrequency() const override { return sampling_frequency_in_hz_; } |
| |
| int NumChannels() const override { return num_channels_; } |
| |
| bool Capture(rtc::BufferT<int16_t>* buffer) override { |
| fill_with_zero_ = !fill_with_zero_; |
| int16_t max_amplitude; |
| { |
| MutexLock lock(&lock_); |
| max_amplitude = max_amplitude_; |
| } |
| buffer->SetData( |
| TestAudioDeviceModule::SamplesPerFrame(sampling_frequency_in_hz_) * |
| num_channels_, |
| [&](rtc::ArrayView<int16_t> data) { |
| if (fill_with_zero_) { |
| std::fill(data.begin(), data.end(), 0); |
| } else { |
| std::generate(data.begin(), data.end(), [&]() { |
| return random_generator_.Rand(-max_amplitude, max_amplitude); |
| }); |
| } |
| return data.size(); |
| }); |
| return true; |
| } |
| |
| void SetMaxAmplitude(int16_t amplitude) override { |
| MutexLock lock(&lock_); |
| max_amplitude_ = amplitude; |
| } |
| |
| private: |
| int sampling_frequency_in_hz_; |
| bool fill_with_zero_; |
| Random random_generator_; |
| Mutex lock_; |
| int16_t max_amplitude_ RTC_GUARDED_BY(lock_); |
| const int num_channels_; |
| }; |
| |
| class WavFileReader final : public TestAudioDeviceModule::Capturer { |
| public: |
| WavFileReader(absl::string_view filename, |
| int sampling_frequency_in_hz, |
| int num_channels, |
| bool repeat) |
| : WavFileReader(std::make_unique<WavReader>(filename), |
| sampling_frequency_in_hz, |
| num_channels, |
| repeat) {} |
| |
| int SamplingFrequency() const override { return sampling_frequency_in_hz_; } |
| |
| int NumChannels() const override { return num_channels_; } |
| |
| bool Capture(rtc::BufferT<int16_t>* buffer) override { |
| buffer->SetData( |
| TestAudioDeviceModule::SamplesPerFrame(sampling_frequency_in_hz_) * |
| num_channels_, |
| [&](rtc::ArrayView<int16_t> data) { |
| size_t read = wav_reader_->ReadSamples(data.size(), data.data()); |
| if (read < data.size() && repeat_) { |
| do { |
| wav_reader_->Reset(); |
| size_t delta = wav_reader_->ReadSamples( |
| data.size() - read, data.subview(read).data()); |
| RTC_CHECK_GT(delta, 0) << "No new data read from file"; |
| read += delta; |
| } while (read < data.size()); |
| } |
| return read; |
| }); |
| return buffer->size() > 0; |
| } |
| |
| private: |
| WavFileReader(std::unique_ptr<WavReader> wav_reader, |
| int sampling_frequency_in_hz, |
| int num_channels, |
| bool repeat) |
| : sampling_frequency_in_hz_(sampling_frequency_in_hz), |
| num_channels_(num_channels), |
| wav_reader_(std::move(wav_reader)), |
| repeat_(repeat) { |
| RTC_CHECK_EQ(wav_reader_->sample_rate(), sampling_frequency_in_hz); |
| RTC_CHECK_EQ(wav_reader_->num_channels(), num_channels); |
| } |
| |
| const int sampling_frequency_in_hz_; |
| const int num_channels_; |
| std::unique_ptr<WavReader> wav_reader_; |
| const bool repeat_; |
| }; |
| |
| class WavFileWriter final : public TestAudioDeviceModule::Renderer { |
| public: |
| WavFileWriter(absl::string_view filename, |
| int sampling_frequency_in_hz, |
| int num_channels) |
| : WavFileWriter(std::make_unique<WavWriter>(filename, |
| sampling_frequency_in_hz, |
| num_channels), |
| sampling_frequency_in_hz, |
| num_channels) {} |
| |
| int SamplingFrequency() const override { return sampling_frequency_in_hz_; } |
| |
| int NumChannels() const override { return num_channels_; } |
| |
| bool Render(rtc::ArrayView<const int16_t> data) override { |
| wav_writer_->WriteSamples(data.data(), data.size()); |
| return true; |
| } |
| |
| private: |
| WavFileWriter(std::unique_ptr<WavWriter> wav_writer, |
| int sampling_frequency_in_hz, |
| int num_channels) |
| : sampling_frequency_in_hz_(sampling_frequency_in_hz), |
| wav_writer_(std::move(wav_writer)), |
| num_channels_(num_channels) {} |
| |
| int sampling_frequency_in_hz_; |
| std::unique_ptr<WavWriter> wav_writer_; |
| const int num_channels_; |
| }; |
| |
| class BoundedWavFileWriter : public TestAudioDeviceModule::Renderer { |
| public: |
| BoundedWavFileWriter(absl::string_view filename, |
| int sampling_frequency_in_hz, |
| int num_channels) |
| : sampling_frequency_in_hz_(sampling_frequency_in_hz), |
| wav_writer_(filename, sampling_frequency_in_hz, num_channels), |
| num_channels_(num_channels), |
| silent_audio_( |
| TestAudioDeviceModule::SamplesPerFrame(sampling_frequency_in_hz) * |
| num_channels, |
| 0), |
| started_writing_(false), |
| trailing_zeros_(0) {} |
| |
| int SamplingFrequency() const override { return sampling_frequency_in_hz_; } |
| |
| int NumChannels() const override { return num_channels_; } |
| |
| bool Render(rtc::ArrayView<const int16_t> data) override { |
| const int16_t kAmplitudeThreshold = 5; |
| |
| const int16_t* begin = data.begin(); |
| const int16_t* end = data.end(); |
| if (!started_writing_) { |
| // Cut off silence at the beginning. |
| while (begin < end) { |
| if (std::abs(*begin) > kAmplitudeThreshold) { |
| started_writing_ = true; |
| break; |
| } |
| ++begin; |
| } |
| } |
| if (started_writing_) { |
| // Cut off silence at the end. |
| while (begin < end) { |
| if (*(end - 1) != 0) { |
| break; |
| } |
| --end; |
| } |
| if (begin < end) { |
| // If it turns out that the silence was not final, need to write all the |
| // skipped zeros and continue writing audio. |
| while (trailing_zeros_ > 0) { |
| const size_t zeros_to_write = |
| std::min(trailing_zeros_, silent_audio_.size()); |
| wav_writer_.WriteSamples(silent_audio_.data(), zeros_to_write); |
| trailing_zeros_ -= zeros_to_write; |
| } |
| wav_writer_.WriteSamples(begin, end - begin); |
| } |
| // Save the number of zeros we skipped in case this needs to be restored. |
| trailing_zeros_ += data.end() - end; |
| } |
| return true; |
| } |
| |
| private: |
| int sampling_frequency_in_hz_; |
| WavWriter wav_writer_; |
| const int num_channels_; |
| std::vector<int16_t> silent_audio_; |
| bool started_writing_; |
| size_t trailing_zeros_; |
| }; |
| |
| class DiscardRenderer final : public TestAudioDeviceModule::Renderer { |
| public: |
| explicit DiscardRenderer(int sampling_frequency_in_hz, int num_channels) |
| : sampling_frequency_in_hz_(sampling_frequency_in_hz), |
| num_channels_(num_channels) {} |
| |
| int SamplingFrequency() const override { return sampling_frequency_in_hz_; } |
| |
| int NumChannels() const override { return num_channels_; } |
| |
| bool Render(rtc::ArrayView<const int16_t> data) override { return true; } |
| |
| private: |
| int sampling_frequency_in_hz_; |
| const int num_channels_; |
| }; |
| |
| } // namespace |
| |
| size_t TestAudioDeviceModule::SamplesPerFrame(int sampling_frequency_in_hz) { |
| return rtc::CheckedDivExact(sampling_frequency_in_hz, kFramesPerSecond); |
| } |
| |
| rtc::scoped_refptr<AudioDeviceModule> TestAudioDeviceModule::Create( |
| TaskQueueFactory* task_queue_factory, |
| std::unique_ptr<TestAudioDeviceModule::Capturer> capturer, |
| std::unique_ptr<TestAudioDeviceModule::Renderer> renderer, |
| float speed) { |
| auto audio_device = rtc::make_ref_counted<TestAudioDeviceModuleImpl>( |
| task_queue_factory, std::move(capturer), std::move(renderer), speed); |
| |
| // Ensure that the current platform is supported. |
| if (audio_device->CheckPlatform() == -1) { |
| return nullptr; |
| } |
| |
| // Create the platform-dependent implementation. |
| if (audio_device->CreatePlatformSpecificObjects() == -1) { |
| return nullptr; |
| } |
| |
| // Ensure that the generic audio buffer can communicate with the platform |
| // specific parts. |
| if (audio_device->AttachAudioBuffer() == -1) { |
| return nullptr; |
| } |
| |
| return audio_device; |
| } |
| |
| std::unique_ptr<TestAudioDeviceModule::PulsedNoiseCapturer> |
| TestAudioDeviceModule::CreatePulsedNoiseCapturer(int16_t max_amplitude, |
| int sampling_frequency_in_hz, |
| int num_channels) { |
| return std::make_unique<PulsedNoiseCapturerImpl>( |
| max_amplitude, sampling_frequency_in_hz, num_channels); |
| } |
| |
| std::unique_ptr<TestAudioDeviceModule::Renderer> |
| TestAudioDeviceModule::CreateDiscardRenderer(int sampling_frequency_in_hz, |
| int num_channels) { |
| return std::make_unique<DiscardRenderer>(sampling_frequency_in_hz, |
| num_channels); |
| } |
| |
| std::unique_ptr<TestAudioDeviceModule::Capturer> |
| TestAudioDeviceModule::CreateWavFileReader(absl::string_view filename, |
| int sampling_frequency_in_hz, |
| int num_channels) { |
| return std::make_unique<WavFileReader>(filename, sampling_frequency_in_hz, |
| num_channels, false); |
| } |
| |
| std::unique_ptr<TestAudioDeviceModule::Capturer> |
| TestAudioDeviceModule::CreateWavFileReader(absl::string_view filename, |
| bool repeat) { |
| WavReader reader(filename); |
| int sampling_frequency_in_hz = reader.sample_rate(); |
| int num_channels = rtc::checked_cast<int>(reader.num_channels()); |
| return std::make_unique<WavFileReader>(filename, sampling_frequency_in_hz, |
| num_channels, repeat); |
| } |
| |
| std::unique_ptr<TestAudioDeviceModule::Renderer> |
| TestAudioDeviceModule::CreateWavFileWriter(absl::string_view filename, |
| int sampling_frequency_in_hz, |
| int num_channels) { |
| return std::make_unique<WavFileWriter>(filename, sampling_frequency_in_hz, |
| num_channels); |
| } |
| |
| std::unique_ptr<TestAudioDeviceModule::Renderer> |
| TestAudioDeviceModule::CreateBoundedWavFileWriter(absl::string_view filename, |
| int sampling_frequency_in_hz, |
| int num_channels) { |
| return std::make_unique<BoundedWavFileWriter>( |
| filename, sampling_frequency_in_hz, num_channels); |
| } |
| |
| } // namespace webrtc |