mirror of
https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
118 lines
4.3 KiB
C++
118 lines
4.3 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/gain_controller2.h"
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#include "common_audio/include/audio_util.h"
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#include "modules/audio_processing/audio_buffer.h"
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#include "modules/audio_processing/include/audio_frame_view.h"
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#include "modules/audio_processing/logging/apm_data_dumper.h"
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#include "rtc_base/atomicops.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/strings/string_builder.h"
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namespace webrtc {
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int GainController2::instance_count_ = 0;
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GainController2::GainController2()
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: data_dumper_(
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new ApmDataDumper(rtc::AtomicOps::Increment(&instance_count_))),
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gain_applier_(/*hard_clip_samples=*/false,
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/*initial_gain_factor=*/0.f),
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adaptive_agc_(new AdaptiveAgc(data_dumper_.get())),
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limiter_(static_cast<size_t>(48000), data_dumper_.get(), "Agc2") {}
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GainController2::~GainController2() = default;
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void GainController2::Initialize(int sample_rate_hz) {
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RTC_DCHECK(sample_rate_hz == AudioProcessing::kSampleRate8kHz ||
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sample_rate_hz == AudioProcessing::kSampleRate16kHz ||
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sample_rate_hz == AudioProcessing::kSampleRate32kHz ||
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sample_rate_hz == AudioProcessing::kSampleRate48kHz);
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limiter_.SetSampleRate(sample_rate_hz);
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data_dumper_->InitiateNewSetOfRecordings();
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data_dumper_->DumpRaw("sample_rate_hz", sample_rate_hz);
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}
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void GainController2::Process(AudioBuffer* audio) {
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AudioFrameView<float> float_frame(audio->channels_f(), audio->num_channels(),
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audio->num_frames());
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// Apply fixed gain first, then the adaptive one.
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gain_applier_.ApplyGain(float_frame);
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if (adaptive_digital_mode_) {
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adaptive_agc_->Process(float_frame, limiter_.LastAudioLevel());
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}
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limiter_.Process(float_frame);
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}
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void GainController2::NotifyAnalogLevel(int level) {
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if (analog_level_ != level && adaptive_digital_mode_) {
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adaptive_agc_->Reset();
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}
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analog_level_ = level;
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}
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void GainController2::ApplyConfig(
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const AudioProcessing::Config::GainController2& config) {
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RTC_DCHECK(Validate(config))
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<< " the invalid config was " << ToString(config);
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config_ = config;
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if (config.fixed_digital.gain_db != config_.fixed_digital.gain_db) {
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// Reset the limiter to quickly react on abrupt level changes caused by
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// large changes of the fixed gain.
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limiter_.Reset();
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}
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gain_applier_.SetGainFactor(DbToRatio(config_.fixed_digital.gain_db));
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adaptive_digital_mode_ = config_.adaptive_digital.enabled;
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adaptive_agc_.reset(new AdaptiveAgc(data_dumper_.get(), config_));
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}
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bool GainController2::Validate(
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const AudioProcessing::Config::GainController2& config) {
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return config.fixed_digital.gain_db >= 0.f &&
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config.fixed_digital.gain_db < 50.f &&
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config.adaptive_digital.extra_saturation_margin_db >= 0.f &&
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config.adaptive_digital.extra_saturation_margin_db <= 100.f;
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}
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std::string GainController2::ToString(
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const AudioProcessing::Config::GainController2& config) {
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rtc::StringBuilder ss;
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std::string adaptive_digital_level_estimator;
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using LevelEstimatorType =
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AudioProcessing::Config::GainController2::LevelEstimator;
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switch (config.adaptive_digital.level_estimator) {
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case LevelEstimatorType::kRms:
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adaptive_digital_level_estimator = "RMS";
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break;
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case LevelEstimatorType::kPeak:
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adaptive_digital_level_estimator = "peak";
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break;
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}
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// clang-format off
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// clang formatting doesn't respect custom nested style.
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ss << "{"
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<< "enabled: " << (config.enabled ? "true" : "false") << ", "
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<< "fixed_digital: {gain_db: " << config.fixed_digital.gain_db << "}, "
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<< "adaptive_digital: {"
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<< "enabled: "
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<< (config.adaptive_digital.enabled ? "true" : "false") << ", "
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<< "level_estimator: " << adaptive_digital_level_estimator << ", "
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<< "extra_saturation_margin_db:"
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<< config.adaptive_digital.extra_saturation_margin_db << "}"
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<< "}";
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// clang-format on
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return ss.Release();
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}
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} // namespace webrtc
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