webrtc/modules/audio_processing/level_controller/gain_applier.h
Mirko Bonadei bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00

42 lines
1.3 KiB
C++

/*
* Copyright (c) 2016 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 WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
#include "webrtc/rtc_base/constructormagic.h"
namespace webrtc {
class ApmDataDumper;
class AudioBuffer;
class GainApplier {
public:
explicit GainApplier(ApmDataDumper* data_dumper);
void Initialize(int sample_rate_hz);
// Applies the specified gain to the audio frame and returns the resulting
// number of saturated sample values.
int Process(float new_gain, AudioBuffer* audio);
private:
ApmDataDumper* const data_dumper_;
float old_gain_ = 1.f;
float gain_increase_step_size_ = 0.f;
float gain_normal_decrease_step_size_ = 0.f;
float gain_saturated_decrease_step_size_ = 0.f;
bool last_frame_was_saturated_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainApplier);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_