bb547203bf
In order to eliminate the WebRTC Subtree mirror in Chromium, WebRTC is moving the content of the src/webrtc directory up to the src/ directory. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38 Reviewed-on: https://webrtc-review.googlesource.com/1560 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19845}
31 lines
1005 B
C++
31 lines
1005 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 WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_
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#include <array>
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#include "webrtc/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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DownsampledRenderBuffer();
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~DownsampledRenderBuffer();
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std::array<float, kDownsampledRenderBufferSize> buffer = {};
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int position = 0;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_
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