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/*
* Copyright (c) 2023 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/test_audio_device_impl.h"
#include <memory>
#include <optional>
#include <utility>
#include "api/array_view.h"
#include "api/task_queue/task_queue_factory.h"
#include "api/units/time_delta.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "rtc_base/checks.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_utils/repeating_task.h"
namespace webrtc {
namespace {
constexpr int kFrameLengthUs = 10000;
}
TestAudioDevice::TestAudioDevice(
TaskQueueFactory* task_queue_factory,
std::unique_ptr<TestAudioDeviceModule::Capturer> capturer,
std::unique_ptr<TestAudioDeviceModule::Renderer> renderer,
float speed)
: task_queue_factory_(task_queue_factory),
capturer_(std::move(capturer)),
renderer_(std::move(renderer)),
process_interval_us_(kFrameLengthUs / speed),
audio_buffer_(nullptr),
rendering_(false),
capturing_(false) {
auto good_sample_rate = [](int sr) {
return sr == 8000 || sr == 16000 || sr == 24000 || sr == 32000 ||
sr == 44100 || sr == 48000;
};
if (renderer_) {
const int sample_rate = renderer_->SamplingFrequency();
playout_buffer_.resize(TestAudioDeviceModule::SamplesPerFrame(sample_rate) *
renderer_->NumChannels(),
0);
RTC_CHECK(good_sample_rate(sample_rate));
}
if (capturer_) {
RTC_CHECK(good_sample_rate(capturer_->SamplingFrequency()));
}
}
AudioDeviceGeneric::InitStatus TestAudioDevice::Init() {
task_queue_ = task_queue_factory_->CreateTaskQueue(
"TestAudioDeviceModuleImpl", TaskQueueFactory::Priority::NORMAL);
RepeatingTaskHandle::Start(task_queue_.get(), [this]() {
ProcessAudio();
return TimeDelta::Micros(process_interval_us_);
});
return InitStatus::OK;
}
int32_t TestAudioDevice::PlayoutIsAvailable(bool& available) {
MutexLock lock(&lock_);
available = renderer_ != nullptr;
return 0;
}
int32_t TestAudioDevice::InitPlayout() {
MutexLock lock(&lock_);
if (rendering_) {
return -1;
}
if (audio_buffer_ != nullptr && renderer_ != nullptr) {
// Update webrtc audio buffer with the selected parameters
audio_buffer_->SetPlayoutSampleRate(renderer_->SamplingFrequency());
audio_buffer_->SetPlayoutChannels(renderer_->NumChannels());
}
rendering_initialized_ = true;
return 0;
}
bool TestAudioDevice::PlayoutIsInitialized() const {
MutexLock lock(&lock_);
return rendering_initialized_;
}
int32_t TestAudioDevice::StartPlayout() {
MutexLock lock(&lock_);
RTC_CHECK(renderer_);
rendering_ = true;
return 0;
}
int32_t TestAudioDevice::StopPlayout() {
MutexLock lock(&lock_);
rendering_ = false;
return 0;
}
int32_t TestAudioDevice::RecordingIsAvailable(bool& available) {
MutexLock lock(&lock_);
available = capturer_ != nullptr;
return 0;
}
int32_t TestAudioDevice::InitRecording() {
MutexLock lock(&lock_);
if (capturing_) {
return -1;
}
if (audio_buffer_ != nullptr && capturer_ != nullptr) {
// Update webrtc audio buffer with the selected parameters
audio_buffer_->SetRecordingSampleRate(capturer_->SamplingFrequency());
audio_buffer_->SetRecordingChannels(capturer_->NumChannels());
}
capturing_initialized_ = true;
return 0;
}
bool TestAudioDevice::RecordingIsInitialized() const {
MutexLock lock(&lock_);
return capturing_initialized_;
}
int32_t TestAudioDevice::StartRecording() {
MutexLock lock(&lock_);
capturing_ = true;
return 0;
}
int32_t TestAudioDevice::StopRecording() {
MutexLock lock(&lock_);
capturing_ = false;
return 0;
}
bool TestAudioDevice::Playing() const {
MutexLock lock(&lock_);
return rendering_;
}
bool TestAudioDevice::Recording() const {
MutexLock lock(&lock_);
return capturing_;
}
void TestAudioDevice::ProcessAudio() {
MutexLock lock(&lock_);
if (audio_buffer_ == nullptr) {
return;
}
if (capturing_ && capturer_ != nullptr) {
// Capture 10ms of audio. 2 bytes per sample.
const bool keep_capturing = capturer_->Capture(&recording_buffer_);
if (recording_buffer_.size() > 0) {
audio_buffer_->SetRecordedBuffer(
recording_buffer_.data(),
recording_buffer_.size() / capturer_->NumChannels(),
std::make_optional(TimeNanos()));
audio_buffer_->DeliverRecordedData();
}
if (!keep_capturing) {
capturing_ = false;
}
}
if (rendering_) {
const int sampling_frequency = renderer_->SamplingFrequency();
int32_t samples_per_channel = audio_buffer_->RequestPlayoutData(
TestAudioDeviceModule::SamplesPerFrame(sampling_frequency));
audio_buffer_->GetPlayoutData(playout_buffer_.data());
size_t samples_out = samples_per_channel * renderer_->NumChannels();
RTC_CHECK_LE(samples_out, playout_buffer_.size());
const bool keep_rendering = renderer_->Render(
ArrayView<const int16_t>(playout_buffer_.data(), samples_out));
if (!keep_rendering) {
rendering_ = false;
}
}
}
void TestAudioDevice::AttachAudioBuffer(AudioDeviceBuffer* audio_buffer) {
MutexLock lock(&lock_);
RTC_DCHECK(audio_buffer || audio_buffer_);
audio_buffer_ = audio_buffer;
if (renderer_ != nullptr) {
audio_buffer_->SetPlayoutSampleRate(renderer_->SamplingFrequency());
audio_buffer_->SetPlayoutChannels(renderer_->NumChannels());
}
if (capturer_ != nullptr) {
audio_buffer_->SetRecordingSampleRate(capturer_->SamplingFrequency());
audio_buffer_->SetRecordingChannels(capturer_->NumChannels());
}
}
} // namespace webrtc
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