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3e4c77f1c1
This CL refactors AGC2 and fixes the order with which the fixed and the adaptive digital gain controllers are applied - i.e., fixed first, then adaptive and finally limiter. FixedGainController has been removed since we need to split the processing done by the gain applier and the limiter. Also, GainApplier and Limiter are easy enough to be used without a wrapper and a wrapper would need 2 separated calls in the right order - i.e., error prone. FrameCombiner in audio mixer has been adapted and now only uses the limiter (which is what is needed since no gain is applied). The unit tests for FixedGainController have been moved to gain_controller2_unittests. They have been re-adapted and ChangeFixedGainShouldBeFastAndTimeInvariant has been re-tuned. Bug: webrtc:7494 Change-Id: I4d7daeae917257ac019a645b74deba6642f77322 Reviewed-on: https://webrtc-review.googlesource.com/c/108624 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25477}
93 lines
3.4 KiB
C++
93 lines
3.4 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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config_ = config;
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if (gain_applier_.GetGainFactor() != config_.fixed_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_gain_db));
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adaptive_digital_mode_ = config_.adaptive_digital_mode;
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adaptive_agc_.reset(
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new AdaptiveAgc(data_dumper_.get(), config_.extra_saturation_margin_db));
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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_gain_db >= 0.f &&
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config.extra_saturation_margin_db >= 0.f &&
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config.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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ss << "{enabled: " << (config.enabled ? "true" : "false") << ", "
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<< "fixed_gain_dB: " << config.fixed_gain_db << "}";
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return ss.Release();
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}
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} // namespace webrtc
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