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93e5750a92
Hypothetical scenario: short weak speech at start of call, then high noise. The digital adaptive AGC2 would pick a high gain, and then continue to apply it on the noise. Unless the noise is detected by the noise estimator, the gain would never be reduced. This CL addresses the issue by sending limiter gain info to the adaptive digital AGC2. Bug: webrtc:7494 Change-Id: Idf5c2686af0f5e5bad981d39a95b8efc9ffb9d64 Reviewed-on: https://webrtc-review.googlesource.com/102641 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24922}
79 lines
2.7 KiB
C++
79 lines
2.7 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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#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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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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if (adaptive_digital_mode_) {
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adaptive_agc_.Process(float_frame, fixed_gain_controller_.LastAudioLevel());
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}
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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 && 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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fixed_gain_controller_.SetGain(config_.fixed_gain_db);
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adaptive_digital_mode_ = config_.adaptive_digital_mode;
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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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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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