| /* |
| * 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/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" |
| #include "webrtc/rtc_base/array_view.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_ |