| /* |
| * Copyright (c) 2013 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 "test/fake_audio_device.h" |
| |
| #include <algorithm> |
| #include <utility> |
| |
| #include "common_audio/wav_file.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/random.h" |
| #include "system_wrappers/include/event_wrapper.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| |
| constexpr int kFrameLengthMs = 10; |
| constexpr int kFramesPerSecond = 1000 / kFrameLengthMs; |
| |
| // Assuming 10ms audio packets.. |
| class PulsedNoiseCapturer final : public test::FakeAudioDevice::Capturer { |
| public: |
| PulsedNoiseCapturer(int16_t max_amplitude, int sampling_frequency_in_hz) |
| : sampling_frequency_in_hz_(sampling_frequency_in_hz), |
| fill_with_zero_(false), |
| random_generator_(1), |
| max_amplitude_(max_amplitude) { |
| RTC_DCHECK_GT(max_amplitude, 0); |
| } |
| |
| int SamplingFrequency() const override { |
| return sampling_frequency_in_hz_; |
| } |
| |
| bool Capture(rtc::BufferT<int16_t>* buffer) override { |
| fill_with_zero_ = !fill_with_zero_; |
| buffer->SetData( |
| test::FakeAudioDevice::SamplesPerFrame(sampling_frequency_in_hz_), |
| [&](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; |
| } |
| |
| private: |
| int sampling_frequency_in_hz_; |
| bool fill_with_zero_; |
| Random random_generator_; |
| const int16_t max_amplitude_; |
| }; |
| |
| class WavFileReader final : public test::FakeAudioDevice::Capturer { |
| public: |
| WavFileReader(std::string filename, int sampling_frequency_in_hz) |
| : sampling_frequency_in_hz_(sampling_frequency_in_hz), |
| wav_reader_(filename) { |
| RTC_CHECK_EQ(wav_reader_.sample_rate(), sampling_frequency_in_hz); |
| RTC_CHECK_EQ(wav_reader_.num_channels(), 1); |
| } |
| |
| int SamplingFrequency() const override { |
| return sampling_frequency_in_hz_; |
| } |
| |
| bool Capture(rtc::BufferT<int16_t>* buffer) override { |
| buffer->SetData( |
| test::FakeAudioDevice::SamplesPerFrame(sampling_frequency_in_hz_), |
| [&](rtc::ArrayView<int16_t> data) { |
| return wav_reader_.ReadSamples(data.size(), data.data()); |
| }); |
| return buffer->size() > 0; |
| } |
| |
| private: |
| int sampling_frequency_in_hz_; |
| WavReader wav_reader_; |
| }; |
| |
| class WavFileWriter final : public test::FakeAudioDevice::Renderer { |
| public: |
| WavFileWriter(std::string filename, int sampling_frequency_in_hz) |
| : sampling_frequency_in_hz_(sampling_frequency_in_hz), |
| wav_writer_(filename, sampling_frequency_in_hz, 1) {} |
| |
| int SamplingFrequency() const override { |
| return sampling_frequency_in_hz_; |
| } |
| |
| bool Render(rtc::ArrayView<const int16_t> data) override { |
| wav_writer_.WriteSamples(data.data(), data.size()); |
| return true; |
| } |
| |
| private: |
| int sampling_frequency_in_hz_; |
| WavWriter wav_writer_; |
| }; |
| |
| class BoundedWavFileWriter : public test::FakeAudioDevice::Renderer { |
| public: |
| BoundedWavFileWriter(std::string filename, int sampling_frequency_in_hz) |
| : sampling_frequency_in_hz_(sampling_frequency_in_hz), |
| wav_writer_(filename, sampling_frequency_in_hz, 1), |
| silent_audio_(test::FakeAudioDevice::SamplesPerFrame( |
| sampling_frequency_in_hz), 0), |
| started_writing_(false), |
| trailing_zeros_(0) {} |
| |
| int SamplingFrequency() const override { |
| return sampling_frequency_in_hz_; |
| } |
| |
| 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_; |
| std::vector<int16_t> silent_audio_; |
| bool started_writing_; |
| size_t trailing_zeros_; |
| }; |
| |
| |
| class DiscardRenderer final : public test::FakeAudioDevice::Renderer { |
| public: |
| explicit DiscardRenderer(int sampling_frequency_in_hz) |
| : sampling_frequency_in_hz_(sampling_frequency_in_hz) {} |
| |
| int SamplingFrequency() const override { |
| return sampling_frequency_in_hz_; |
| } |
| |
| bool Render(rtc::ArrayView<const int16_t> data) override { |
| return true; |
| } |
| |
| private: |
| int sampling_frequency_in_hz_; |
| }; |
| |
| } // namespace |
| namespace test { |
| |
| size_t FakeAudioDevice::SamplesPerFrame(int sampling_frequency_in_hz) { |
| return rtc::CheckedDivExact(sampling_frequency_in_hz, kFramesPerSecond); |
| } |
| |
| std::unique_ptr<FakeAudioDevice::Capturer> |
| FakeAudioDevice::CreatePulsedNoiseCapturer( |
| int16_t max_amplitude, int sampling_frequency_in_hz) { |
| return std::unique_ptr<FakeAudioDevice::Capturer>( |
| new PulsedNoiseCapturer(max_amplitude, sampling_frequency_in_hz)); |
| } |
| |
| std::unique_ptr<FakeAudioDevice::Capturer> FakeAudioDevice::CreateWavFileReader( |
| std::string filename, int sampling_frequency_in_hz) { |
| return std::unique_ptr<FakeAudioDevice::Capturer>( |
| new WavFileReader(filename, sampling_frequency_in_hz)); |
| } |
| |
| std::unique_ptr<FakeAudioDevice::Capturer> FakeAudioDevice::CreateWavFileReader( |
| std::string filename) { |
| int sampling_frequency_in_hz = WavReader(filename).sample_rate(); |
| return std::unique_ptr<FakeAudioDevice::Capturer>( |
| new WavFileReader(filename, sampling_frequency_in_hz)); |
| } |
| |
| std::unique_ptr<FakeAudioDevice::Renderer> FakeAudioDevice::CreateWavFileWriter( |
| std::string filename, int sampling_frequency_in_hz) { |
| return std::unique_ptr<FakeAudioDevice::Renderer>( |
| new WavFileWriter(filename, sampling_frequency_in_hz)); |
| } |
| |
| std::unique_ptr<FakeAudioDevice::Renderer> |
| FakeAudioDevice::CreateBoundedWavFileWriter( |
| std::string filename, int sampling_frequency_in_hz) { |
| return std::unique_ptr<FakeAudioDevice::Renderer>( |
| new BoundedWavFileWriter(filename, sampling_frequency_in_hz)); |
| } |
| |
| std::unique_ptr<FakeAudioDevice::Renderer> |
| FakeAudioDevice::CreateDiscardRenderer(int sampling_frequency_in_hz) { |
| return std::unique_ptr<FakeAudioDevice::Renderer>( |
| new DiscardRenderer(sampling_frequency_in_hz)); |
| } |
| |
| |
| FakeAudioDevice::FakeAudioDevice(std::unique_ptr<Capturer> capturer, |
| std::unique_ptr<Renderer> renderer, |
| float speed) |
| : capturer_(std::move(capturer)), |
| renderer_(std::move(renderer)), |
| speed_(speed), |
| audio_callback_(nullptr), |
| rendering_(false), |
| capturing_(false), |
| done_rendering_(true, true), |
| done_capturing_(true, true), |
| tick_(EventTimerWrapper::Create()), |
| thread_(FakeAudioDevice::Run, this, "FakeAudioDevice") { |
| auto good_sample_rate = [](int sr) { |
| return sr == 8000 || sr == 16000 || sr == 32000 |
| || sr == 44100 || sr == 48000; |
| }; |
| |
| if (renderer_) { |
| const int sample_rate = renderer_->SamplingFrequency(); |
| playout_buffer_.resize(SamplesPerFrame(sample_rate), 0); |
| RTC_CHECK(good_sample_rate(sample_rate)); |
| } |
| if (capturer_) { |
| RTC_CHECK(good_sample_rate(capturer_->SamplingFrequency())); |
| } |
| } |
| |
| FakeAudioDevice::~FakeAudioDevice() { |
| StopPlayout(); |
| StopRecording(); |
| thread_.Stop(); |
| } |
| |
| int32_t FakeAudioDevice::StartPlayout() { |
| rtc::CritScope cs(&lock_); |
| RTC_CHECK(renderer_); |
| rendering_ = true; |
| done_rendering_.Reset(); |
| return 0; |
| } |
| |
| int32_t FakeAudioDevice::StopPlayout() { |
| rtc::CritScope cs(&lock_); |
| rendering_ = false; |
| done_rendering_.Set(); |
| return 0; |
| } |
| |
| int32_t FakeAudioDevice::StartRecording() { |
| rtc::CritScope cs(&lock_); |
| RTC_CHECK(capturer_); |
| capturing_ = true; |
| done_capturing_.Reset(); |
| return 0; |
| } |
| |
| int32_t FakeAudioDevice::StopRecording() { |
| rtc::CritScope cs(&lock_); |
| capturing_ = false; |
| done_capturing_.Set(); |
| return 0; |
| } |
| |
| int32_t FakeAudioDevice::Init() { |
| // TODO(solenberg): Temporarily allow multiple init calls. |
| if (!inited_) { |
| RTC_CHECK(tick_->StartTimer(true, kFrameLengthMs / speed_)); |
| thread_.Start(); |
| thread_.SetPriority(rtc::kHighPriority); |
| inited_ = true; |
| } |
| return 0; |
| } |
| |
| int32_t FakeAudioDevice::RegisterAudioCallback(AudioTransport* callback) { |
| rtc::CritScope cs(&lock_); |
| RTC_DCHECK(callback || audio_callback_); |
| audio_callback_ = callback; |
| return 0; |
| } |
| |
| bool FakeAudioDevice::Playing() const { |
| rtc::CritScope cs(&lock_); |
| return rendering_; |
| } |
| |
| bool FakeAudioDevice::Recording() const { |
| rtc::CritScope cs(&lock_); |
| return capturing_; |
| } |
| |
| bool FakeAudioDevice::WaitForPlayoutEnd(int timeout_ms) { |
| return done_rendering_.Wait(timeout_ms); |
| } |
| |
| bool FakeAudioDevice::WaitForRecordingEnd(int timeout_ms) { |
| return done_capturing_.Wait(timeout_ms); |
| } |
| |
| bool FakeAudioDevice::Run(void* obj) { |
| static_cast<FakeAudioDevice*>(obj)->ProcessAudio(); |
| return true; |
| } |
| |
| void FakeAudioDevice::ProcessAudio() { |
| { |
| rtc::CritScope cs(&lock_); |
| if (capturing_) { |
| // Capture 10ms of audio. 2 bytes per sample. |
| const bool keep_capturing = capturer_->Capture(&recording_buffer_); |
| uint32_t new_mic_level; |
| if (recording_buffer_.size() > 0) { |
| audio_callback_->RecordedDataIsAvailable( |
| recording_buffer_.data(), recording_buffer_.size(), 2, 1, |
| capturer_->SamplingFrequency(), 0, 0, 0, false, new_mic_level); |
| } |
| if (!keep_capturing) { |
| capturing_ = false; |
| done_capturing_.Set(); |
| } |
| } |
| if (rendering_) { |
| size_t samples_out; |
| int64_t elapsed_time_ms; |
| int64_t ntp_time_ms; |
| const int sampling_frequency = renderer_->SamplingFrequency(); |
| audio_callback_->NeedMorePlayData( |
| SamplesPerFrame(sampling_frequency), 2, 1, sampling_frequency, |
| playout_buffer_.data(), samples_out, &elapsed_time_ms, &ntp_time_ms); |
| const bool keep_rendering = renderer_->Render( |
| rtc::ArrayView<const int16_t>(playout_buffer_.data(), samples_out)); |
| if (!keep_rendering) { |
| rendering_ = false; |
| done_rendering_.Set(); |
| } |
| } |
| } |
| tick_->Wait(WEBRTC_EVENT_INFINITE); |
| } |
| |
| |
| } // namespace test |
| } // namespace webrtc |