mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-19 08:37:54 +01:00
2fa4618a3b
The class has also been renamed to better reflect its purpose. Bug: webrtc:7494 Change-Id: I223a364ab4f8b8a5fef765848bf05675d045cefd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236343 Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35277}
108 lines
4 KiB
C++
108 lines
4 KiB
C++
/*
|
|
* Copyright (c) 2018 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/agc2/adaptive_digital_gain_controller.h"
|
|
|
|
#include <algorithm>
|
|
|
|
#include "common_audio/include/audio_util.h"
|
|
#include "modules/audio_processing/agc2/vad_wrapper.h"
|
|
#include "modules/audio_processing/logging/apm_data_dumper.h"
|
|
#include "rtc_base/checks.h"
|
|
#include "rtc_base/logging.h"
|
|
|
|
namespace webrtc {
|
|
namespace {
|
|
|
|
// Peak and RMS audio levels in dBFS.
|
|
struct AudioLevels {
|
|
float peak_dbfs;
|
|
float rms_dbfs;
|
|
};
|
|
|
|
// Computes the audio levels for the first channel in `frame`.
|
|
AudioLevels ComputeAudioLevels(AudioFrameView<float> frame) {
|
|
float peak = 0.0f;
|
|
float rms = 0.0f;
|
|
for (const auto& x : frame.channel(0)) {
|
|
peak = std::max(std::fabs(x), peak);
|
|
rms += x * x;
|
|
}
|
|
return {FloatS16ToDbfs(peak),
|
|
FloatS16ToDbfs(std::sqrt(rms / frame.samples_per_channel()))};
|
|
}
|
|
|
|
} // namespace
|
|
|
|
AdaptiveDigitalGainController::AdaptiveDigitalGainController(
|
|
ApmDataDumper* apm_data_dumper,
|
|
const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
|
|
int sample_rate_hz,
|
|
int num_channels)
|
|
: speech_level_estimator_(apm_data_dumper, config),
|
|
gain_controller_(apm_data_dumper, config, sample_rate_hz, num_channels),
|
|
apm_data_dumper_(apm_data_dumper),
|
|
noise_level_estimator_(CreateNoiseFloorEstimator(apm_data_dumper)),
|
|
saturation_protector_(
|
|
CreateSaturationProtector(kSaturationProtectorInitialHeadroomDb,
|
|
config.adjacent_speech_frames_threshold,
|
|
apm_data_dumper)) {
|
|
RTC_DCHECK(apm_data_dumper);
|
|
RTC_DCHECK(noise_level_estimator_);
|
|
RTC_DCHECK(saturation_protector_);
|
|
}
|
|
|
|
AdaptiveDigitalGainController::~AdaptiveDigitalGainController() = default;
|
|
|
|
void AdaptiveDigitalGainController::Initialize(int sample_rate_hz,
|
|
int num_channels) {
|
|
gain_controller_.Initialize(sample_rate_hz, num_channels);
|
|
}
|
|
|
|
void AdaptiveDigitalGainController::Process(AudioFrameView<float> frame,
|
|
float speech_probability,
|
|
float limiter_envelope) {
|
|
AudioLevels levels = ComputeAudioLevels(frame);
|
|
apm_data_dumper_->DumpRaw("agc2_input_rms_dbfs", levels.rms_dbfs);
|
|
apm_data_dumper_->DumpRaw("agc2_input_peak_dbfs", levels.peak_dbfs);
|
|
|
|
AdaptiveDigitalGainApplier::FrameInfo info;
|
|
|
|
info.speech_probability = speech_probability;
|
|
|
|
speech_level_estimator_.Update(levels.rms_dbfs, levels.peak_dbfs,
|
|
info.speech_probability);
|
|
info.speech_level_dbfs = speech_level_estimator_.level_dbfs();
|
|
info.speech_level_reliable = speech_level_estimator_.IsConfident();
|
|
apm_data_dumper_->DumpRaw("agc2_speech_level_dbfs", info.speech_level_dbfs);
|
|
apm_data_dumper_->DumpRaw("agc2_speech_level_reliable",
|
|
info.speech_level_reliable);
|
|
|
|
info.noise_rms_dbfs = noise_level_estimator_->Analyze(frame);
|
|
apm_data_dumper_->DumpRaw("agc2_noise_rms_dbfs", info.noise_rms_dbfs);
|
|
|
|
saturation_protector_->Analyze(info.speech_probability, levels.peak_dbfs,
|
|
info.speech_level_dbfs);
|
|
info.headroom_db = saturation_protector_->HeadroomDb();
|
|
apm_data_dumper_->DumpRaw("agc2_headroom_db", info.headroom_db);
|
|
|
|
info.limiter_envelope_dbfs = FloatS16ToDbfs(limiter_envelope);
|
|
apm_data_dumper_->DumpRaw("agc2_limiter_envelope_dbfs",
|
|
info.limiter_envelope_dbfs);
|
|
|
|
gain_controller_.Process(info, frame);
|
|
}
|
|
|
|
void AdaptiveDigitalGainController::HandleInputGainChange() {
|
|
speech_level_estimator_.Reset();
|
|
saturation_protector_->Reset();
|
|
}
|
|
|
|
} // namespace webrtc
|