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libtgvoip/webrtc_dsp/modules/audio_processing/aec3/suppression_gain_limiter.h
Grishka 5caaaafa42 Updated WebRTC APM
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.
2018-11-23 04:02:53 +03:00

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1.9 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.
*/
#ifndef MODULES_AUDIO_PROCESSING_AEC3_SUPPRESSION_GAIN_LIMITER_H_
#define MODULES_AUDIO_PROCESSING_AEC3_SUPPRESSION_GAIN_LIMITER_H_
#include "api/array_view.h"
#include "api/audio/echo_canceller3_config.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
// Class for applying a smoothly increasing limit for the suppression gain
// during call startup and after in-call resets.
class SuppressionGainUpperLimiter {
public:
explicit SuppressionGainUpperLimiter(const EchoCanceller3Config& config);
// Reset the limiting behavior.
void Reset();
// Updates the limiting behavior for the current capture bloc.
void Update(bool render_activity, bool transparent_mode);
// Returns the current suppressor gain limit.
float Limit() const { return suppressor_gain_limit_; }
// Return whether the suppressor gain limit is active.
bool IsActive() const { return (realignment_counter_ > 0); }
// Inactivate limiter.
void Deactivate() {
realignment_counter_ = 0;
suppressor_gain_limit_ = 1.f;
}
private:
const EchoCanceller3Config::EchoRemovalControl::GainRampup rampup_config_;
const float gain_rampup_increase_;
bool call_startup_phase_ = true;
int realignment_counter_ = 0;
bool active_render_seen_ = false;
float suppressor_gain_limit_ = 1.f;
bool recent_reset_ = false;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(SuppressionGainUpperLimiter);
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AEC3_SUPPRESSION_GAIN_LIMITER_H_