c8fa692ec4
I used a command like this to update the paths: perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"` The only manual edit is to add an include of webrtc/rtc_base/checks.h in webrtc/modules/audio_device/android/opensles_common.h, which likely was needed due to changed include paths due to 'git cl format'. BUG=webrtc:7634 NOTRY=True NOPRESUBMIT=True Review-Url: https://codereview.webrtc.org/2969653002 Cr-Commit-Position: refs/heads/master@{#18871}
43 lines
1.3 KiB
C++
43 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 WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
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#include "webrtc/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 // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_CONTROLLER_GAIN_APPLIER_H_
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