webrtc/modules/audio_processing/aec3/fft_buffer.cc
Per Åhgren c59a576c86 Corrections of the render buffering scheme in AEC3 to ensure causality
This CL modifies the refactored render buffering scheme in AEC3
so that:
-A non-causal state can never occur which means that situations with
 nonrecoverable echo should not occur.
-For a stable audio pipeline with a predefined API call jitter,
 render overruns and underruns can never occur.

Bug: webrtc:8629,chromium:793305
Change-Id: I06ba1c368f92db95274090b08475dd02dbb85145
Reviewed-on: https://webrtc-review.googlesource.com/29861
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21215}
2017-12-11 21:09:56 +00:00

23 lines
678 B
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.
*/
#include "modules/audio_processing/aec3/fft_buffer.h"
namespace webrtc {
FftBuffer::FftBuffer(size_t size) : size(static_cast<int>(size)), buffer(size) {
for (auto& b : buffer) {
b.Clear();
}
}
FftBuffer::~FftBuffer() = default;
} // namespace webrtc