1 /* 2 * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_ 12 #define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_ 13 14 #include <stddef.h> 15 #include <array> 16 #include <vector> 17 18 #include "api/array_view.h" 19 #include "modules/audio_processing/aec3/aec3_common.h" 20 #include "modules/audio_processing/aec3/downsampled_render_buffer.h" 21 #include "modules/audio_processing/aec3/fft_data.h" 22 #include "modules/audio_processing/aec3/render_buffer.h" 23 24 namespace webrtc { 25 26 // Class for buffering the incoming render blocks such that these may be 27 // extracted with a specified delay. 28 class RenderDelayBuffer { 29 public: 30 static RenderDelayBuffer* Create(size_t num_bands, 31 size_t down_sampling_factor, 32 size_t downsampled_render_buffer_size, 33 size_t render_delay_buffer_size); 34 virtual ~RenderDelayBuffer() = default; 35 36 // Resets the buffer data. 37 virtual void Reset() = 0; 38 39 // Inserts a block into the buffer and returns true if the insert is 40 // successful. 41 virtual bool Insert(const std::vector<std::vector<float>>& block) = 0; 42 43 // Updates the buffers one step based on the specified buffer delay. Returns 44 // true if there was no overrun, otherwise returns false. 45 virtual bool UpdateBuffers() = 0; 46 47 // Sets the buffer delay. 48 virtual void SetDelay(size_t delay) = 0; 49 50 // Gets the buffer delay. 51 virtual size_t Delay() const = 0; 52 53 // Returns the render buffer for the echo remover. 54 virtual const RenderBuffer& GetRenderBuffer() const = 0; 55 56 // Returns the downsampled render buffer. 57 virtual const DownsampledRenderBuffer& GetDownsampledRenderBuffer() const = 0; 58 }; 59 60 } // namespace webrtc 61 62 #endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_ 63