modules/audio_processing/aec_dump/aec_dump_impl.cc - src - Git at Google

 /*
  *  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 "modules/audio_processing/aec_dump/aec_dump_impl.h"

 #include <memory>
 #include <utility>

 #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 int16_t* const data,
                                         int num_channels,
                                         int samples_per_channel) {
   capture_stream_info_.AddInput(data, num_channels, samples_per_channel);
 }

 void AecDumpImpl::AddCaptureStreamOutput(const int16_t* const data,
                                          int num_channels,
                                          int samples_per_channel) {
   capture_stream_info_.AddOutput(data, num_channels, samples_per_channel);
 }

 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 int16_t* const data,
                                            int num_channels,
                                            int samples_per_channel) {
   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) * samples_per_channel * num_channels;
   msg->set_data(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::kCaptureFixedPostGain: {
       float x;
       runtime_setting.GetFloat(&x);
       setting->set_capture_fixed_post_gain(x);
       break;
     }
     case AudioProcessing::RuntimeSetting::Type::kPlayoutVolumeChange: {
       int x;
       runtime_setting.GetInt(&x);
       setting->set_playout_volume_change(x);
       break;
     }
     case AudioProcessing::RuntimeSetting::Type::kPlayoutAudioDeviceChange: {
       AudioProcessing::RuntimeSetting::PlayoutAudioDeviceInfo src;
       runtime_setting.GetPlayoutAudioDeviceInfo(&src);
       auto* dst = setting->mutable_playout_audio_device_change();
       dst->set_id(src.id);
       dst->set_max_volume(src.max_volume);
       break;
     }
     case AudioProcessing::RuntimeSetting::Type::kNotSpecified:
       RTC_NOTREACHED();
       break;
   }
   worker_queue_->PostTask(std::move(task));
 }

 std::unique_ptr<WriteToFileTask> AecDumpImpl::CreateWriteToFileTask() {
   return std::make_unique<WriteToFileTask>(&debug_file_,
                                            &num_bytes_left_for_log_);
 }

 std::unique_ptr<AecDump> AecDumpFactory::Create(webrtc::FileWrapper file,
                                                 int64_t max_log_size_bytes,
                                                 rtc::TaskQueue* worker_queue) {
   RTC_DCHECK(worker_queue);
   if (!file.is_open())
     return nullptr;

   return std::make_unique<AecDumpImpl>(std::move(file), 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) {
   return Create(FileWrapper::OpenWriteOnly(file_name.c_str()),
                 max_log_size_bytes, worker_queue);
 }

 std::unique_ptr<AecDump> AecDumpFactory::Create(FILE* handle,
                                                 int64_t max_log_size_bytes,
                                                 rtc::TaskQueue* worker_queue) {
   return Create(FileWrapper(handle), max_log_size_bytes, worker_queue);
 }

 }  // namespace webrtc
	/*
	* 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 "modules/audio_processing/aec_dump/aec_dump_impl.h"

	#include <memory>
	#include <utility>

	#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 int16_t* const data,
	int num_channels,
	int samples_per_channel) {
	capture_stream_info_.AddInput(data, num_channels, samples_per_channel);
	}

	void AecDumpImpl::AddCaptureStreamOutput(const int16_t* const data,
	int num_channels,
	int samples_per_channel) {
	capture_stream_info_.AddOutput(data, num_channels, samples_per_channel);
	}

	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 int16_t* const data,
	int num_channels,
	int samples_per_channel) {
	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) * samples_per_channel * num_channels;
	msg->set_data(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::kCaptureFixedPostGain: {
	float x;
	runtime_setting.GetFloat(&x);
	setting->set_capture_fixed_post_gain(x);
	break;
	}
	case AudioProcessing::RuntimeSetting::Type::kPlayoutVolumeChange: {
	int x;
	runtime_setting.GetInt(&x);
	setting->set_playout_volume_change(x);
	break;
	}
	case AudioProcessing::RuntimeSetting::Type::kPlayoutAudioDeviceChange: {
	AudioProcessing::RuntimeSetting::PlayoutAudioDeviceInfo src;
	runtime_setting.GetPlayoutAudioDeviceInfo(&src);
	auto* dst = setting->mutable_playout_audio_device_change();
	dst->set_id(src.id);
	dst->set_max_volume(src.max_volume);
	break;
	}
	case AudioProcessing::RuntimeSetting::Type::kNotSpecified:
	RTC_NOTREACHED();
	break;
	}
	worker_queue_->PostTask(std::move(task));
	}

	std::unique_ptr<WriteToFileTask> AecDumpImpl::CreateWriteToFileTask() {
	return std::make_unique<WriteToFileTask>(&debug_file_,
	&num_bytes_left_for_log_);
	}

	std::unique_ptr<AecDump> AecDumpFactory::Create(webrtc::FileWrapper file,
	int64_t max_log_size_bytes,
	rtc::TaskQueue* worker_queue) {
	RTC_DCHECK(worker_queue);
	if (!file.is_open())
	return nullptr;

	return std::make_unique<AecDumpImpl>(std::move(file), 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) {
	return Create(FileWrapper::OpenWriteOnly(file_name.c_str()),
	max_log_size_bytes, worker_queue);
	}

	std::unique_ptr<AecDump> AecDumpFactory::Create(FILE* handle,
	int64_t max_log_size_bytes,
	rtc::TaskQueue* worker_queue) {
	return Create(FileWrapper(handle), max_log_size_bytes, worker_queue);
	}

	} // namespace webrtc