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a837dd790d
Agc2 applies a digital gain to the nearend signal. When the analog level changes, the digital gain calculation is no longer valid. Therefore Agc2 should be notified to analog gain changes. This CL also allow audioproc_f to chain AGC1 and AGC2. In a dependent CL we will allow using AGC1 for analog gain and AGC2 for digital gain. Bug: webrtc:7494 Change-Id: Id75b3728fbf2de1d84b7fba005e4670c7a2985d9 Reviewed-on: https://webrtc-review.googlesource.com/89387 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24231}
75 lines
2.5 KiB
C++
75 lines
2.5 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 "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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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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fixed_gain_controller_(data_dumper_.get()),
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adaptive_agc_(data_dumper_.get()) {}
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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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fixed_gain_controller_.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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adaptive_agc_.Process(float_frame);
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fixed_gain_controller_.Process(float_frame);
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}
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void GainController2::NotifyAnalogLevel(int level) {
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if (analog_level_ != level) {
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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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fixed_gain_controller_.SetGain(config_.fixed_gain_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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}
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std::string GainController2::ToString(
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const AudioProcessing::Config::GainController2& config) {
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std::stringstream 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.str();
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}
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} // namespace webrtc
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