webrtc/modules/audio_processing/aec3/render_delay_buffer.h
Per Åhgren 38e2d95bda AEC3 delay estimator refactoring and introducing ability to customize
This CL refactors the delay estimator in AEC3.
Furthermore, it adds:
1. Allow for a customized delay estimator behavior to simplify
development.
2. Exposes that behavior to clear configuration settings.
3. Adds logging of the delay range supported by the delay
estimator.

Bug: webrtc:8519
Change-Id: I1764a090519a78b021b2e7de565c52a6c02c848e
Reviewed-on: https://webrtc-review.googlesource.com/21166
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20733}
2017-11-17 17:51:37 +00:00

62 lines
2.1 KiB
C++

/*
* 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 MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
#define MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_
#include <stddef.h>
#include <array>
#include <vector>
#include "api/array_view.h"
#include "modules/audio_processing/aec3/aec3_common.h"
#include "modules/audio_processing/aec3/downsampled_render_buffer.h"
#include "modules/audio_processing/aec3/fft_data.h"
#include "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,
size_t down_sampling_factor,
size_t downsampled_render_buffer_size,
size_t render_delay_buffer_size);
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 // MODULES_AUDIO_PROCESSING_AEC3_RENDER_DELAY_BUFFER_H_