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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/gain_controller2.h"
#include <memory>
#include <utility>
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/agc2/cpu_features.h"
#include "modules/audio_processing/audio_buffer.h"
#include "modules/audio_processing/include/audio_frame_view.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_builder.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace {
using Agc2Config = AudioProcessing::Config::GainController2;
using InputVolumeControllerConfig = InputVolumeController::Config;
constexpr int kLogLimiterStatsPeriodMs = 30'000;
constexpr int kFrameLengthMs = 10;
constexpr int kLogLimiterStatsPeriodNumFrames =
kLogLimiterStatsPeriodMs / kFrameLengthMs;
// Detects the available CPU features and applies any kill-switches.
AvailableCpuFeatures GetAllowedCpuFeatures() {
AvailableCpuFeatures features = GetAvailableCpuFeatures();
if (field_trial::IsEnabled("WebRTC-Agc2SimdSse2KillSwitch")) {
features.sse2 = false;
}
if (field_trial::IsEnabled("WebRTC-Agc2SimdAvx2KillSwitch")) {
features.avx2 = false;
}
if (field_trial::IsEnabled("WebRTC-Agc2SimdNeonKillSwitch")) {
features.neon = false;
}
return features;
}
// Creates an adaptive digital gain controller if enabled.
std::unique_ptr<AdaptiveDigitalGainController> CreateAdaptiveDigitalController(
const Agc2Config::AdaptiveDigital& config,
int sample_rate_hz,
int num_channels,
ApmDataDumper* data_dumper) {
if (config.enabled) {
return std::make_unique<AdaptiveDigitalGainController>(
data_dumper, config, sample_rate_hz, num_channels);
}
return nullptr;
}
// Creates an input volume controller if `enabled` is true.
std::unique_ptr<InputVolumeController> CreateInputVolumeController(
bool enabled,
const InputVolumeControllerConfig& config,
int num_channels) {
if (enabled) {
return std::make_unique<InputVolumeController>(num_channels, config);
}
return nullptr;
}
} // namespace
std::atomic<int> GainController2::instance_count_(0);
GainController2::GainController2(
const Agc2Config& config,
const InputVolumeControllerConfig& input_volume_controller_config,
int sample_rate_hz,
int num_channels,
bool use_internal_vad)
: cpu_features_(GetAllowedCpuFeatures()),
data_dumper_(instance_count_.fetch_add(1) + 1),
fixed_gain_applier_(
/*hard_clip_samples=*/false,
/*initial_gain_factor=*/DbToRatio(config.fixed_digital.gain_db)),
adaptive_digital_controller_(
CreateAdaptiveDigitalController(config.adaptive_digital,
sample_rate_hz,
num_channels,
&data_dumper_)),
input_volume_controller_(
CreateInputVolumeController(config.input_volume_controller.enabled,
input_volume_controller_config,
num_channels)),
limiter_(sample_rate_hz, &data_dumper_, /*histogram_name_prefix=*/"Agc2"),
calls_since_last_limiter_log_(0) {
RTC_DCHECK(Validate(config));
data_dumper_.InitiateNewSetOfRecordings();
const bool use_vad = config.adaptive_digital.enabled;
if (use_vad && use_internal_vad) {
// TODO(bugs.webrtc.org/7494): Move `vad_reset_period_ms` from adaptive
// digital to gain controller 2 config.
vad_ = std::make_unique<VoiceActivityDetectorWrapper>(
config.adaptive_digital.vad_reset_period_ms, cpu_features_,
sample_rate_hz);
}
if (input_volume_controller_) {
input_volume_controller_->Initialize();
}
}
GainController2::~GainController2() = default;
// TODO(webrtc:7494): Pass the flag also to the other components.
void GainController2::SetCaptureOutputUsed(bool capture_output_used) {
if (input_volume_controller_) {
input_volume_controller_->HandleCaptureOutputUsedChange(
capture_output_used);
}
}
void GainController2::SetFixedGainDb(float gain_db) {
const float gain_factor = DbToRatio(gain_db);
if (fixed_gain_applier_.GetGainFactor() != gain_factor) {
// Reset the limiter to quickly react on abrupt level changes caused by
// large changes of the fixed gain.
limiter_.Reset();
}
fixed_gain_applier_.SetGainFactor(gain_factor);
}
void GainController2::Analyze(int applied_input_volume,
const AudioBuffer& audio_buffer) {
RTC_DCHECK_GE(applied_input_volume, 0);
RTC_DCHECK_LE(applied_input_volume, 255);
if (input_volume_controller_) {
input_volume_controller_->set_stream_analog_level(applied_input_volume);
input_volume_controller_->AnalyzePreProcess(audio_buffer);
}
}
absl::optional<int> GainController2::GetRecommendedInputVolume() const {
return input_volume_controller_
? absl::optional<int>(
input_volume_controller_->recommended_analog_level())
: absl::nullopt;
}
void GainController2::Process(absl::optional<float> speech_probability,
bool input_volume_changed,
AudioBuffer* audio) {
data_dumper_.DumpRaw("agc2_applied_input_volume_changed",
input_volume_changed);
if (input_volume_changed && !!adaptive_digital_controller_) {
adaptive_digital_controller_->HandleInputGainChange();
}
AudioFrameView<float> float_frame(audio->channels(), audio->num_channels(),
audio->num_frames());
if (vad_) {
speech_probability = vad_->Analyze(float_frame);
} else if (speech_probability.has_value()) {
RTC_DCHECK_GE(speech_probability.value(), 0.0f);
RTC_DCHECK_LE(speech_probability.value(), 1.0f);
}
if (speech_probability.has_value()) {
data_dumper_.DumpRaw("agc2_speech_probability", speech_probability.value());
}
if (input_volume_controller_) {
// TODO(bugs.webrtc.org/7494): A temprorary check, remove once not needed.
RTC_DCHECK(adaptive_digital_controller_);
absl::optional<float> speech_level;
if (adaptive_digital_controller_) {
speech_level =
adaptive_digital_controller_->GetSpeechLevelDbfsIfConfident();
}
RTC_DCHECK(speech_probability.has_value());
if (speech_probability.has_value()) {
input_volume_controller_->Process(*speech_probability, speech_level);
}
}
if (adaptive_digital_controller_) {
RTC_DCHECK(speech_probability.has_value());
adaptive_digital_controller_->Process(
float_frame, speech_probability.value(), limiter_.LastAudioLevel());
}
fixed_gain_applier_.ApplyGain(float_frame);
limiter_.Process(float_frame);
// Periodically log limiter stats.
if (++calls_since_last_limiter_log_ == kLogLimiterStatsPeriodNumFrames) {
calls_since_last_limiter_log_ = 0;
InterpolatedGainCurve::Stats stats = limiter_.GetGainCurveStats();
RTC_LOG(LS_INFO) << "AGC2 limiter stats"
<< " | identity: " << stats.look_ups_identity_region
<< " | knee: " << stats.look_ups_knee_region
<< " | limiter: " << stats.look_ups_limiter_region
<< " | saturation: " << stats.look_ups_saturation_region;
}
}
bool GainController2::Validate(
const AudioProcessing::Config::GainController2& config) {
const auto& fixed = config.fixed_digital;
const auto& adaptive = config.adaptive_digital;
return fixed.gain_db >= 0.0f && fixed.gain_db < 50.f &&
adaptive.headroom_db >= 0.0f && adaptive.max_gain_db > 0.0f &&
adaptive.initial_gain_db >= 0.0f &&
adaptive.max_gain_change_db_per_second > 0.0f &&
adaptive.max_output_noise_level_dbfs <= 0.0f;
}
} // namespace webrtc
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