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Fixing WebRTC after moving from src/webrtc to src/

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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}
This commit is contained in:
Mirko Bonadei
2017-09-15 06:47:31 +02:00
committed by Commit Bot
parent bb547203bf
commit 92ea95e34a
3635 changed files with 19692 additions and 19645 deletions
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4
modules/audio_coding/codecs/ilbc/create_augmented_vec.c
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@ -18,8 +18,8 @@
#include "defines.h"
#include "constants.h"
#include "webrtc/rtc_base/sanitizer.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "rtc_base/sanitizer.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
/*----------------------------------------------------------------*
* Recreate a specific codebook vector from the augmented part.