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06af5b5c64
Adopt DeinterleavedView and MonoView in the following classes and deprecate existing versions where external dependencies exist: * GainApplier * AdaptiveDigitalGainController * NoiseLevelEstimator * VoiceActivityDetectorWrapper (including MonoVad) Bug: chromium:335805780 Change-Id: I15dad833a87d31476d147dd2456bd1cc39f901ed Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355861 Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42611}
50 lines
1.6 KiB
C++
50 lines
1.6 KiB
C++
/*
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* Copyright (c) 2018 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_AGC2_GAIN_APPLIER_H_
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#define MODULES_AUDIO_PROCESSING_AGC2_GAIN_APPLIER_H_
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#include <stddef.h>
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#include "api/audio/audio_view.h"
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#include "modules/audio_processing/include/audio_frame_view.h"
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namespace webrtc {
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class GainApplier {
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public:
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GainApplier(bool hard_clip_samples, float initial_gain_factor);
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void ApplyGain(DeinterleavedView<float> signal);
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void SetGainFactor(float gain_factor);
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float GetGainFactor() const { return current_gain_factor_; }
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[[deprecated("Use DeinterleavedView<> version")]] void ApplyGain(
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AudioFrameView<float> signal) {
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ApplyGain(signal.view());
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}
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private:
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void Initialize(int samples_per_channel);
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// Whether to clip samples after gain is applied. If 'true', result
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// will fit in FloatS16 range.
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const bool hard_clip_samples_;
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float last_gain_factor_;
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// If this value is not equal to 'last_gain_factor', gain will be
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// ramped from 'last_gain_factor_' to this value during the next
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// 'ApplyGain'.
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float current_gain_factor_;
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int samples_per_channel_ = -1;
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float inverse_samples_per_channel_ = -1.f;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AGC2_GAIN_APPLIER_H_
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