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Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/agc2/gain_applier.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/agc2/gain_applier.h | 44 |
1 files changed, 44 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier.h b/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier.h new file mode 100644 index 0000000000..ba8a4a4cd2 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/agc2/gain_applier.h @@ -0,0 +1,44 @@ +/* + * 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_AGC2_GAIN_APPLIER_H_ +#define MODULES_AUDIO_PROCESSING_AGC2_GAIN_APPLIER_H_ + +#include <stddef.h> + +#include "modules/audio_processing/include/audio_frame_view.h" + +namespace webrtc { +class GainApplier { + public: + GainApplier(bool hard_clip_samples, float initial_gain_factor); + + void ApplyGain(AudioFrameView<float> signal); + void SetGainFactor(float gain_factor); + float GetGainFactor() const { return current_gain_factor_; } + + private: + void Initialize(int samples_per_channel); + + // Whether to clip samples after gain is applied. If 'true', result + // will fit in FloatS16 range. + const bool hard_clip_samples_; + float last_gain_factor_; + + // If this value is not equal to 'last_gain_factor', gain will be + // ramped from 'last_gain_factor_' to this value during the next + // 'ApplyGain'. + float current_gain_factor_; + int samples_per_channel_ = -1; + float inverse_samples_per_channel_ = -1.f; +}; +} // namespace webrtc + +#endif // MODULES_AUDIO_PROCESSING_AGC2_GAIN_APPLIER_H_ |