webrtc_m130/modules/audio_processing/aec3/downsampled_render_buffer.cc

/*
 *  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() = default;

DownsampledRenderBuffer::~DownsampledRenderBuffer() = default;

}  // namespace webrtc
Major AEC3 render pipeline changes This CL adds major render pipeline changes to the AEC3 code. The reason for these are that 1) It allows the echo removal unit to receive information about the content in bands beyond band 0, thereby allowing removal of high-frequency echoes 2) It allows more controlled handling of the render buffers, allowing proper buffer behaviour during capture glitches and clock-drift. Unfortunately, the render pipeline caused a lot of related changes in much of the rest of the AEC3 files. Most of these are, however, caused by a change of class name. Another unfortunate effect of this CL, is that a number of unittest cease to compile. I chose to temporarily solve that by removing them from the build using #if/#endif. The reason for that is that those will anyway again need to be changed in the next review, and doing like this avoids them having to be reviewed twice. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2784023002 Cr-Commit-Position: refs/heads/master@{#17547} 2017-04-05 14:18:07 -07:00			`/*`
			`* 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.`
			`*/`

Fixing WebRTC after moving from src/webrtc to src/ In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846} 2017-09-15 06:47:31 +02:00			`#include "modules/audio_processing/aec3/downsampled_render_buffer.h"`
Major AEC3 render pipeline changes This CL adds major render pipeline changes to the AEC3 code. The reason for these are that 1) It allows the echo removal unit to receive information about the content in bands beyond band 0, thereby allowing removal of high-frequency echoes 2) It allows more controlled handling of the render buffers, allowing proper buffer behaviour during capture glitches and clock-drift. Unfortunately, the render pipeline caused a lot of related changes in much of the rest of the AEC3 files. Most of these are, however, caused by a change of class name. Another unfortunate effect of this CL, is that a number of unittest cease to compile. I chose to temporarily solve that by removing them from the build using #if/#endif. The reason for that is that those will anyway again need to be changed in the next review, and doing like this avoids them having to be reviewed twice. BUG=webrtc:6018 Review-Url: https://codereview.webrtc.org/2784023002 Cr-Commit-Position: refs/heads/master@{#17547} 2017-04-05 14:18:07 -07:00
			`namespace webrtc {`

			`DownsampledRenderBuffer::DownsampledRenderBuffer() = default;`

			`DownsampledRenderBuffer::~DownsampledRenderBuffer() = default;`

			`} // namespace webrtc`