webrtc/api/rtpsender.h - Issue 1610243002: Move talk/app/webrtc to webrtc/api

Keyboard Shortcuts

	File
u :	up to issue
j / k :	jump to file after / before current file
J / K :	jump to next file with a comment after / before current file
	Side-by-side diff
i :	toggle intra-line diffs
e :	expand all comments
c :	collapse all comments
s :	toggle showing all comments
n / p :	next / previous diff chunk or comment
N / P :	next / previous comment
<Up> / <Down> :	next / previous line

	Issue
u :	up to list of issues
j / k :	jump to patch after / before current patch
o / <Enter> :	open current patch in side-by-side view
i :	open current patch in unified diff view

	Issue List
j / k :	jump to issue after / before current issue
o / <Enter> :	open current issue

Unified Diff: webrtc/api/rtpsender.h

Issue 1610243002: Move talk/app/webrtc to webrtc/api (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master

Patch Set: Removed processing of api.gyp for Chromium builds Created 4 years, 10 months ago

Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.

Jump to:

Index: webrtc/api/rtpsender.h

diff --git a/talk/app/webrtc/rtpsender.h b/webrtc/api/rtpsender.h

similarity index 96%

rename from talk/app/webrtc/rtpsender.h

rename to webrtc/api/rtpsender.h

index c68f64be40b29ab673c2f507d15856d01f92434c..45b765de9f4a25b2c3c64ae4e766d5489a8b8f1a 100644

--- a/talk/app/webrtc/rtpsender.h

+++ b/webrtc/api/rtpsender.h

@@ -29,14 +29,14 @@

// An RtpSender associates a MediaStreamTrackInterface with an underlying

// transport (provided by AudioProviderInterface/VideoProviderInterface)

-#ifndef TALK_APP_WEBRTC_RTPSENDER_H_

-#define TALK_APP_WEBRTC_RTPSENDER_H_

+#ifndef WEBRTC_API_RTPSENDER_H_

+#define WEBRTC_API_RTPSENDER_H_

#include <string>

-#include "talk/app/webrtc/mediastreamprovider.h"

-#include "talk/app/webrtc/rtpsenderinterface.h"

-#include "talk/app/webrtc/statscollector.h"

+#include "webrtc/api/mediastreamprovider.h"

+#include "webrtc/api/rtpsenderinterface.h"

+#include "webrtc/api/statscollector.h"

#include "webrtc/base/basictypes.h"

#include "webrtc/base/criticalsection.h"

#include "webrtc/base/scoped_ptr.h"

@@ -192,4 +192,4 @@ class VideoRtpSender : public ObserverInterface,

} // namespace webrtc

-#endif // TALK_APP_WEBRTC_RTPSENDER_H_

+#endif // WEBRTC_API_RTPSENDER_H_

« no previous file with comments | « webrtc/api/rtpreceiverinterface.h ('k') | webrtc/api/rtpsender.cc » ('j') | no next file with comments »