| commit | 92ea95e34af5966555903026f45164afbd7e2088 | [log] [tgz] |
|---|---|---|
| author | Mirko Bonadei <mbonadei@webrtc.org> | Fri Sep 15 06:47:31 2017 +0200 |
| committer | Commit Bot <commit-bot@chromium.org> | Fri Sep 15 05:02:56 2017 +0000 |
| tree | 3befb3fcfefff3b35da8a834052543338e15bcc0 | |
| parent | bb547203bfebcc478b263c4e9ca173c6fd5a0c5d [diff] [blame] |
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}
diff --git a/modules/audio_coding/codecs/g711/g711_interface.c b/modules/audio_coding/codecs/g711/g711_interface.c index 5b96a9c..52f73fb 100644 --- a/modules/audio_coding/codecs/g711/g711_interface.c +++ b/modules/audio_coding/codecs/g711/g711_interface.c
@@ -10,7 +10,7 @@ #include <string.h> #include "g711.h" #include "g711_interface.h" -#include "webrtc/typedefs.h" +#include "typedefs.h" size_t WebRtcG711_EncodeA(const int16_t* speechIn, size_t len,