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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
42 lines
1.3 KiB
C++
42 lines
1.3 KiB
C++
/*
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* Copyright (c) 2016 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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#ifndef MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
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#define MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
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#include "rtc_base/constructormagic.h"
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namespace webrtc {
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class ApmDataDumper;
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class AudioBuffer;
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class GainApplier {
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public:
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explicit GainApplier(ApmDataDumper* data_dumper);
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void Initialize(int sample_rate_hz);
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// Applies the specified gain to the audio frame and returns the resulting
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// number of saturated sample values.
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int Process(float new_gain, AudioBuffer* audio);
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private:
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ApmDataDumper* const data_dumper_;
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float old_gain_ = 1.f;
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float gain_increase_step_size_ = 0.f;
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float gain_normal_decrease_step_size_ = 0.f;
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float gain_saturated_decrease_step_size_ = 0.f;
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bool last_frame_was_saturated_;
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RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(GainApplier);
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
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