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38e2d95bda
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}
30 lines
983 B
C++
30 lines
983 B
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_DOWNSAMPLED_RENDER_BUFFER_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_
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#include <vector>
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#include "modules/audio_processing/aec3/aec3_common.h"
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namespace webrtc {
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// Holds the circular buffer of the downsampled render data.
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struct DownsampledRenderBuffer {
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explicit DownsampledRenderBuffer(size_t downsampled_buffer_size);
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~DownsampledRenderBuffer();
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std::vector<float> buffer;
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int position = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_
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