mirror of
https://github.com/mollyim/webrtc.git
synced 2025-05-17 15:47:53 +01:00
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}
20 lines
719 B
C++
20 lines
719 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/downsampled_render_buffer.h"
|
|
|
|
namespace webrtc {
|
|
|
|
DownsampledRenderBuffer::DownsampledRenderBuffer(size_t downsampled_buffer_size)
|
|
: buffer(downsampled_buffer_size, 0.f) {}
|
|
|
|
DownsampledRenderBuffer::~DownsampledRenderBuffer() = default;
|
|
|
|
} // namespace webrtc
|