diff options
| author | Mirko Bonadei <mbonadei@webrtc.org> | 2017-09-15 06:47:31 +0200 |
|---|---|---|
| committer | Commit Bot <commit-bot@chromium.org> | 2017-09-15 05:02:56 +0000 |
| commit | 92ea95e34af5966555903026f45164afbd7e2088 (patch) | |
| tree | 3befb3fcfefff3b35da8a834052543338e15bcc0 /call/rtp_rtcp_demuxer_helper.h | |
| parent | bb547203bfebcc478b263c4e9ca173c6fd5a0c5d (diff) | |
| download | webrtc-92ea95e34af5966555903026f45164afbd7e2088.tar.gz | |
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}
Diffstat (limited to 'call/rtp_rtcp_demuxer_helper.h')
| -rw-r--r-- | call/rtp_rtcp_demuxer_helper.h | 12 |
1 files changed, 6 insertions, 6 deletions
diff --git a/call/rtp_rtcp_demuxer_helper.h b/call/rtp_rtcp_demuxer_helper.h index 0a4767d84a..32408e8a7a 100644 --- a/call/rtp_rtcp_demuxer_helper.h +++ b/call/rtp_rtcp_demuxer_helper.h @@ -8,16 +8,16 @@ * be found in the AUTHORS file in the root of the source tree. */ -#ifndef WEBRTC_CALL_RTP_RTCP_DEMUXER_HELPER_H_ -#define WEBRTC_CALL_RTP_RTCP_DEMUXER_HELPER_H_ +#ifndef CALL_RTP_RTCP_DEMUXER_HELPER_H_ +#define CALL_RTP_RTCP_DEMUXER_HELPER_H_ #include <algorithm> #include <map> #include <utility> -#include "webrtc/api/array_view.h" -#include "webrtc/api/optional.h" -#include "webrtc/rtc_base/basictypes.h" +#include "api/array_view.h" +#include "api/optional.h" +#include "rtc_base/basictypes.h" namespace webrtc { @@ -95,4 +95,4 @@ rtc::Optional<uint32_t> ParseRtcpPacketSenderSsrc( } // namespace webrtc -#endif // WEBRTC_CALL_RTP_RTCP_DEMUXER_HELPER_H_ +#endif // CALL_RTP_RTCP_DEMUXER_HELPER_H_ |