/* * 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_AEC3_BLOCK_DELAY_BUFFER_H_ #define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_DELAY_BUFFER_H_ #include #include #include "modules/audio_processing/audio_buffer.h" namespace webrtc { // Class for applying a fixed delay to the samples in a signal partitioned using // the audiobuffer band-splitting scheme. class BlockDelayBuffer { public: BlockDelayBuffer(size_t num_bands, size_t frame_length, size_t delay_samples); ~BlockDelayBuffer(); // Delays the samples by the specified delay. void DelaySignal(AudioBuffer* frame); private: const size_t frame_length_; const size_t delay_; std::vector> buf_; size_t last_insert_ = 0; }; } // namespace webrtc #endif // MODULES_AUDIO_PROCESSING_AEC3_BLOCK_DELAY_BUFFER_H_