Moving src/webrtc into src/.

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}
This commit is contained in:
Mirko Bonadei
2017-09-15 06:15:48 +02:00
committed by Commit Bot
parent 6674846b4a
commit bb547203bf
4576 changed files with 1092 additions and 1196 deletions

View File

@ -0,0 +1,35 @@
/*
* Copyright (c) 2015 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.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_BUFFER_TOOLS_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_BUFFER_TOOLS_H_
#include <vector>
#include "webrtc/api/array_view.h"
#include "webrtc/modules/audio_processing/audio_buffer.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
namespace webrtc {
namespace test {
// Copies a vector into an audiobuffer.
void CopyVectorToAudioBuffer(const StreamConfig& stream_config,
rtc::ArrayView<const float> source,
AudioBuffer* destination);
// Extracts a vector from an audiobuffer.
void ExtractVectorFromAudioBuffer(const StreamConfig& stream_config,
AudioBuffer* source,
std::vector<float>* destination);
} // namespace test
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AUDIO_BUFFER_TOOLS_H_