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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: Updated location for peerconnection_unittests.isolate Created 4 years, 11 months ago
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Index: webrtc/api/rtpsender.h
diff --git a/talk/app/webrtc/rtpsender.h b/webrtc/api/rtpsender.h
similarity index 74%
rename from talk/app/webrtc/rtpsender.h
rename to webrtc/api/rtpsender.h
index 9e055c14956766e6801a9a836cfbc6117ff5d7bb..948fcd92dd266122e2adaa0d2a9a05310e8b1e5b 100644
--- a/talk/app/webrtc/rtpsender.h
+++ b/webrtc/api/rtpsender.h
@@ -1,43 +1,26 @@
/*
- * libjingle
- * Copyright 2015 Google Inc.
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
*
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ * 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.
*/
// This file contains classes that implement RtpSenderInterface.
// 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 "talk/media/base/audiorenderer.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 +175,4 @@ class VideoRtpSender : public ObserverInterface,
} // namespace webrtc
-#endif // TALK_APP_WEBRTC_RTPSENDER_H_
+#endif // WEBRTC_API_RTPSENDER_H_

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