| /* |
| * Copyright (c) 2017 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 <utility> |
| |
| #include "modules/audio_processing/aec_dump/aec_dump_impl.h" |
| |
| #include "absl/memory/memory.h" |
| #include "modules/audio_processing/aec_dump/aec_dump_factory.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/event.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| void CopyFromConfigToEvent(const webrtc::InternalAPMConfig& config, |
| webrtc::audioproc::Config* pb_cfg) { |
| pb_cfg->set_aec_enabled(config.aec_enabled); |
| pb_cfg->set_aec_delay_agnostic_enabled(config.aec_delay_agnostic_enabled); |
| pb_cfg->set_aec_drift_compensation_enabled( |
| config.aec_drift_compensation_enabled); |
| pb_cfg->set_aec_extended_filter_enabled(config.aec_extended_filter_enabled); |
| pb_cfg->set_aec_suppression_level(config.aec_suppression_level); |
| |
| pb_cfg->set_aecm_enabled(config.aecm_enabled); |
| pb_cfg->set_aecm_comfort_noise_enabled(config.aecm_comfort_noise_enabled); |
| pb_cfg->set_aecm_routing_mode(config.aecm_routing_mode); |
| |
| pb_cfg->set_agc_enabled(config.agc_enabled); |
| pb_cfg->set_agc_mode(config.agc_mode); |
| pb_cfg->set_agc_limiter_enabled(config.agc_limiter_enabled); |
| pb_cfg->set_noise_robust_agc_enabled(config.noise_robust_agc_enabled); |
| |
| pb_cfg->set_hpf_enabled(config.hpf_enabled); |
| |
| pb_cfg->set_ns_enabled(config.ns_enabled); |
| pb_cfg->set_ns_level(config.ns_level); |
| |
| pb_cfg->set_transient_suppression_enabled( |
| config.transient_suppression_enabled); |
| |
| pb_cfg->set_pre_amplifier_enabled(config.pre_amplifier_enabled); |
| pb_cfg->set_pre_amplifier_fixed_gain_factor( |
| config.pre_amplifier_fixed_gain_factor); |
| |
| pb_cfg->set_experiments_description(config.experiments_description); |
| } |
| |
| } // namespace |
| |
| AecDumpImpl::AecDumpImpl(FileWrapper debug_file, |
| int64_t max_log_size_bytes, |
| rtc::TaskQueue* worker_queue) |
| : debug_file_(std::move(debug_file)), |
| num_bytes_left_for_log_(max_log_size_bytes), |
| worker_queue_(worker_queue), |
| capture_stream_info_(CreateWriteToFileTask()) {} |
| |
| AecDumpImpl::~AecDumpImpl() { |
| // Block until all tasks have finished running. |
| rtc::Event thread_sync_event; |
| worker_queue_->PostTask([&thread_sync_event] { thread_sync_event.Set(); }); |
| // Wait until the event has been signaled with .Set(). By then all |
| // pending tasks will have finished. |
| thread_sync_event.Wait(rtc::Event::kForever); |
| } |
| |
| void AecDumpImpl::WriteInitMessage(const ProcessingConfig& api_format, |
| int64_t time_now_ms) { |
| auto task = CreateWriteToFileTask(); |
| auto* event = task->GetEvent(); |
| event->set_type(audioproc::Event::INIT); |
| audioproc::Init* msg = event->mutable_init(); |
| |
| msg->set_sample_rate(api_format.input_stream().sample_rate_hz()); |
| msg->set_output_sample_rate(api_format.output_stream().sample_rate_hz()); |
| msg->set_reverse_sample_rate( |
| api_format.reverse_input_stream().sample_rate_hz()); |
| msg->set_reverse_output_sample_rate( |
| api_format.reverse_output_stream().sample_rate_hz()); |
| |
| msg->set_num_input_channels( |
| static_cast<int32_t>(api_format.input_stream().num_channels())); |
| msg->set_num_output_channels( |
| static_cast<int32_t>(api_format.output_stream().num_channels())); |
| msg->set_num_reverse_channels( |
| static_cast<int32_t>(api_format.reverse_input_stream().num_channels())); |
| msg->set_num_reverse_output_channels( |
| api_format.reverse_output_stream().num_channels()); |
| msg->set_timestamp_ms(time_now_ms); |
| |
| worker_queue_->PostTask(std::move(task)); |
| } |
| |
| void AecDumpImpl::AddCaptureStreamInput( |
| const AudioFrameView<const float>& src) { |
| capture_stream_info_.AddInput(src); |
| } |
| |
| void AecDumpImpl::AddCaptureStreamOutput( |
| const AudioFrameView<const float>& src) { |
| capture_stream_info_.AddOutput(src); |
| } |
| |
| void AecDumpImpl::AddCaptureStreamInput(const AudioFrame& frame) { |
| capture_stream_info_.AddInput(frame); |
| } |
| |
| void AecDumpImpl::AddCaptureStreamOutput(const AudioFrame& frame) { |
| capture_stream_info_.AddOutput(frame); |
| } |
| |
| void AecDumpImpl::AddAudioProcessingState(const AudioProcessingState& state) { |
| capture_stream_info_.AddAudioProcessingState(state); |
| } |
| |
| void AecDumpImpl::WriteCaptureStreamMessage() { |
| auto task = capture_stream_info_.GetTask(); |
| RTC_DCHECK(task); |
| worker_queue_->PostTask(std::move(task)); |
| capture_stream_info_.SetTask(CreateWriteToFileTask()); |
| } |
| |
| void AecDumpImpl::WriteRenderStreamMessage(const AudioFrame& frame) { |
| auto task = CreateWriteToFileTask(); |
| auto* event = task->GetEvent(); |
| |
| event->set_type(audioproc::Event::REVERSE_STREAM); |
| audioproc::ReverseStream* msg = event->mutable_reverse_stream(); |
| const size_t data_size = |
| sizeof(int16_t) * frame.samples_per_channel_ * frame.num_channels_; |
| msg->set_data(frame.data(), data_size); |
| |
| worker_queue_->PostTask(std::move(task)); |
| } |
| |
| void AecDumpImpl::WriteRenderStreamMessage( |
| const AudioFrameView<const float>& src) { |
| auto task = CreateWriteToFileTask(); |
| auto* event = task->GetEvent(); |
| |
| event->set_type(audioproc::Event::REVERSE_STREAM); |
| |
| audioproc::ReverseStream* msg = event->mutable_reverse_stream(); |
| |
| for (size_t i = 0; i < src.num_channels(); ++i) { |
| const auto& channel_view = src.channel(i); |
| msg->add_channel(channel_view.begin(), sizeof(float) * channel_view.size()); |
| } |
| |
| worker_queue_->PostTask(std::move(task)); |
| } |
| |
| void AecDumpImpl::WriteConfig(const InternalAPMConfig& config) { |
| RTC_DCHECK_RUNS_SERIALIZED(&race_checker_); |
| auto task = CreateWriteToFileTask(); |
| auto* event = task->GetEvent(); |
| event->set_type(audioproc::Event::CONFIG); |
| CopyFromConfigToEvent(config, event->mutable_config()); |
| worker_queue_->PostTask(std::move(task)); |
| } |
| |
| void AecDumpImpl::WriteRuntimeSetting( |
| const AudioProcessing::RuntimeSetting& runtime_setting) { |
| RTC_DCHECK_RUNS_SERIALIZED(&race_checker_); |
| auto task = CreateWriteToFileTask(); |
| auto* event = task->GetEvent(); |
| event->set_type(audioproc::Event::RUNTIME_SETTING); |
| audioproc::RuntimeSetting* setting = event->mutable_runtime_setting(); |
| switch (runtime_setting.type()) { |
| case AudioProcessing::RuntimeSetting::Type::kCapturePreGain: { |
| float x; |
| runtime_setting.GetFloat(&x); |
| setting->set_capture_pre_gain(x); |
| break; |
| } |
| case AudioProcessing::RuntimeSetting::Type:: |
| kCustomRenderProcessingRuntimeSetting: { |
| float x; |
| runtime_setting.GetFloat(&x); |
| setting->set_custom_render_processing_setting(x); |
| break; |
| } |
| case AudioProcessing::RuntimeSetting::Type::kCaptureCompressionGain: |
| // Runtime AGC1 compression gain is ignored. |
| // TODO(http://bugs.webrtc.org/10432): Store compression gain in aecdumps. |
| break; |
| case AudioProcessing::RuntimeSetting::Type::kNotSpecified: |
| RTC_NOTREACHED(); |
| break; |
| } |
| worker_queue_->PostTask(std::move(task)); |
| } |
| |
| std::unique_ptr<WriteToFileTask> AecDumpImpl::CreateWriteToFileTask() { |
| return absl::make_unique<WriteToFileTask>(&debug_file_, |
| &num_bytes_left_for_log_); |
| } |
| |
| std::unique_ptr<AecDump> AecDumpFactory::Create(rtc::PlatformFile file, |
| int64_t max_log_size_bytes, |
| rtc::TaskQueue* worker_queue) { |
| RTC_DCHECK(worker_queue); |
| FILE* handle = rtc::FdopenPlatformFileForWriting(file); |
| if (!handle) { |
| return nullptr; |
| } |
| return absl::make_unique<AecDumpImpl>(FileWrapper(handle), max_log_size_bytes, |
| worker_queue); |
| } |
| |
| std::unique_ptr<AecDump> AecDumpFactory::Create(std::string file_name, |
| int64_t max_log_size_bytes, |
| rtc::TaskQueue* worker_queue) { |
| RTC_DCHECK(worker_queue); |
| FileWrapper debug_file = FileWrapper::OpenWriteOnly(file_name.c_str()); |
| if (!debug_file.is_open()) { |
| return nullptr; |
| } |
| return absl::make_unique<AecDumpImpl>(std::move(debug_file), |
| max_log_size_bytes, worker_queue); |
| } |
| |
| std::unique_ptr<AecDump> AecDumpFactory::Create(FILE* handle, |
| int64_t max_log_size_bytes, |
| rtc::TaskQueue* worker_queue) { |
| RTC_DCHECK(worker_queue); |
| RTC_DCHECK(handle); |
| return absl::make_unique<AecDumpImpl>(FileWrapper(handle), max_log_size_bytes, |
| worker_queue); |
| } |
| } // namespace webrtc |