cf02cf13a7
This CL adds major render pipeline changes to the AEC3 code. The reason for these are that 1) It allows the echo removal unit to receive information about the content in bands beyond band 0, thereby allowing removal of high-frequency echoes 2) It allows more controlled handling of the render buffers, allowing proper buffer behaviour during capture glitches and clock-drift. Unfortunately, the render pipeline caused a lot of related changes in much of the rest of the AEC3 files. Most of these are, however, caused by a change of class name. Another unfortunate effect of this CL, is that a number of unittest cease to compile. I chose to temporarily solve that by removing them from the build using #if/#endif. The reason for that is that those will anyway again need to be changed in the next review, and doing like this avoids them having to be reviewed twice. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2784023002 Cr-Commit-Position: refs/heads/master@{#17547}
60 lines
2.0 KiB
C++
60 lines
2.0 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 WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
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#define WEBRTC_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 "webrtc/base/array_view.h"
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#include "webrtc/modules/audio_processing/aec3/aec3_common.h"
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#include "webrtc/modules/audio_processing/aec3/downsampled_render_buffer.h"
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#include "webrtc/modules/audio_processing/aec3/fft_data.h"
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#include "webrtc/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 // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
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