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/*
* 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 "absl/base/nullability.h"
#include "absl/strings/string_view.h"
#include "api/task_queue/task_queue_base.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,
absl::Nonnull<TaskQueueBase*> worker_queue)
: debug_file_(std::move(debug_file)),
num_bytes_left_for_log_(max_log_size_bytes),
worker_queue_(worker_queue) {}
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 event = std::make_unique<audioproc::Event>();
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);
PostWriteToFileTask(std::move(event));
}
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() {
PostWriteToFileTask(capture_stream_info_.FetchEvent());
}
void AecDumpImpl::WriteRenderStreamMessage(const int16_t* const data,
int num_channels,
int samples_per_channel) {
auto event = std::make_unique<audioproc::Event>();
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);
PostWriteToFileTask(std::move(event));
}
void AecDumpImpl::WriteRenderStreamMessage(
const AudioFrameView<const float>& src) {
auto event = std::make_unique<audioproc::Event>();
event->set_type(audioproc::Event::REVERSE_STREAM);
audioproc::ReverseStream* msg = event->mutable_reverse_stream();
for (int 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());
}
PostWriteToFileTask(std::move(event));
}
void AecDumpImpl::WriteConfig(const InternalAPMConfig& config) {
RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
auto event = std::make_unique<audioproc::Event>();
event->set_type(audioproc::Event::CONFIG);
CopyFromConfigToEvent(config, event->mutable_config());
PostWriteToFileTask(std::move(event));
}
void AecDumpImpl::WriteRuntimeSetting(
const AudioProcessing::RuntimeSetting& runtime_setting) {
RTC_DCHECK_RUNS_SERIALIZED(&race_checker_);
auto event = std::make_unique<audioproc::Event>();
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::kCapturePostGain: {
float x;
runtime_setting.GetFloat(&x);
setting->set_capture_post_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.
break;
case AudioProcessing::RuntimeSetting::Type::kCaptureFixedPostGain: {
float x;
runtime_setting.GetFloat(&x);
setting->set_capture_fixed_post_gain(x);
break;
}
case AudioProcessing::RuntimeSetting::Type::kCaptureOutputUsed: {
bool x;
runtime_setting.GetBool(&x);
setting->set_capture_output_used(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_DCHECK_NOTREACHED();
break;
}
PostWriteToFileTask(std::move(event));
}
void AecDumpImpl::PostWriteToFileTask(std::unique_ptr<audioproc::Event> event) {
RTC_DCHECK(event);
worker_queue_->PostTask([event = std::move(event), this] {
std::string event_string = event->SerializeAsString();
const size_t event_byte_size = event_string.size();
if (num_bytes_left_for_log_ >= 0) {
const int64_t next_message_size = sizeof(int32_t) + event_byte_size;
if (num_bytes_left_for_log_ < next_message_size) {
// Ensure that no further events are written, even if they're smaller
// than the current event.
num_bytes_left_for_log_ = 0;
return;
}
num_bytes_left_for_log_ -= next_message_size;
}
// Write message preceded by its size.
if (!debug_file_.Write(&event_byte_size, sizeof(int32_t))) {
RTC_DCHECK_NOTREACHED();
}
if (!debug_file_.Write(event_string.data(), event_string.size())) {
RTC_DCHECK_NOTREACHED();
}
});
}
absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
FileWrapper file,
int64_t max_log_size_bytes,
absl::Nonnull<TaskQueueBase*> 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);
}
absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
absl::string_view file_name,
int64_t max_log_size_bytes,
absl::Nonnull<TaskQueueBase*> worker_queue) {
return Create(FileWrapper::OpenWriteOnly(file_name), max_log_size_bytes,
worker_queue);
}
absl::Nullable<std::unique_ptr<AecDump>> AecDumpFactory::Create(
absl::Nonnull<FILE*> handle,
int64_t max_log_size_bytes,
absl::Nonnull<TaskQueueBase*> worker_queue) {
return Create(FileWrapper(handle), max_log_size_bytes, worker_queue);
}
} // namespace webrtc