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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
59 lines
1.9 KiB
C++
59 lines
1.9 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
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#include <stddef.h>
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#include <array>
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#include <vector>
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#include "api/array_view.h"
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#include "modules/audio_processing/aec3/aec3_common.h"
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#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
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#include "modules/audio_processing/aec3/fft_data.h"
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#include "modules/audio_processing/aec3/render_buffer.h"
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namespace webrtc {
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// Class for buffering the incoming render blocks such that these may be
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// extracted with a specified delay.
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class RenderDelayBuffer {
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public:
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static RenderDelayBuffer* Create(size_t num_bands);
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virtual ~RenderDelayBuffer() = default;
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// Resets the buffer data.
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virtual void Reset() = 0;
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// Inserts a block into the buffer and returns true if the insert is
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// successful.
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virtual bool Insert(const std::vector<std::vector<float>>& block) = 0;
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// Updates the buffers one step based on the specified buffer delay. Returns
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// true if there was no overrun, otherwise returns false.
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virtual bool UpdateBuffers() = 0;
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// Sets the buffer delay.
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virtual void SetDelay(size_t delay) = 0;
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// Gets the buffer delay.
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virtual size_t Delay() const = 0;
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// Returns the render buffer for the echo remover.
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virtual const RenderBuffer& GetRenderBuffer() const = 0;
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// Returns the downsampled render buffer.
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virtual const DownsampledRenderBuffer& GetDownsampledRenderBuffer() const = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
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