|  | /* | 
|  | *  Copyright (c) 2017 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 WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_ | 
|  | #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_ | 
|  |  | 
|  | #include <stddef.h> | 
|  | #include <array> | 
|  | #include <vector> | 
|  |  | 
|  | #include "webrtc/base/array_view.h" | 
|  | #include "webrtc/modules/audio_processing/aec3/aec3_common.h" | 
|  | #include "webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h" | 
|  | #include "webrtc/modules/audio_processing/aec3/fft_data.h" | 
|  | #include "webrtc/modules/audio_processing/aec3/render_buffer.h" | 
|  |  | 
|  | namespace webrtc { | 
|  |  | 
|  | // Class for buffering the incoming render blocks such that these may be | 
|  | // extracted with a specified delay. | 
|  | class RenderDelayBuffer { | 
|  | public: | 
|  | static RenderDelayBuffer* Create(size_t num_bands); | 
|  | virtual ~RenderDelayBuffer() = default; | 
|  |  | 
|  | // Resets the buffer data. | 
|  | virtual void Reset() = 0; | 
|  |  | 
|  | // Inserts a block into the buffer and returns true if the insert is | 
|  | // successful. | 
|  | virtual bool Insert(const std::vector<std::vector<float>>& block) = 0; | 
|  |  | 
|  | // Updates the buffers one step based on the specified buffer delay. Returns | 
|  | // true if there was no overrun, otherwise returns false. | 
|  | virtual bool UpdateBuffers() = 0; | 
|  |  | 
|  | // Sets the buffer delay. | 
|  | virtual void SetDelay(size_t delay) = 0; | 
|  |  | 
|  | // Gets the buffer delay. | 
|  | virtual size_t Delay() const = 0; | 
|  |  | 
|  | // Returns the render buffer for the echo remover. | 
|  | virtual const RenderBuffer& GetRenderBuffer() const = 0; | 
|  |  | 
|  | // Returns the downsampled render buffer. | 
|  | virtual const DownsampledRenderBuffer& GetDownsampledRenderBuffer() const = 0; | 
|  | }; | 
|  |  | 
|  | }  // namespace webrtc | 
|  |  | 
|  | #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_ |