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

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Unified Diff: webrtc/api/sctputils.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

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Index: webrtc/api/sctputils.h

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

similarity index 93%

rename from talk/app/webrtc/sctputils.h

rename to webrtc/api/sctputils.h

index f16873c4c3d66d59a56b65f60890e125b64d690c..a3bdb5c16a313cd32133dcc9fea06ab916537990 100644

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

+++ b/webrtc/api/sctputils.h

@@ -25,12 +25,12 @@

* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.

-#ifndef TALK_APP_WEBRTC_SCTPUTILS_H_

-#define TALK_APP_WEBRTC_SCTPUTILS_H_

+#ifndef WEBRTC_API_SCTPUTILS_H_

+#define WEBRTC_API_SCTPUTILS_H_

#include <string>

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

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

namespace rtc {

class Buffer;

@@ -55,4 +55,4 @@ bool WriteDataChannelOpenMessage(const std::string& label,

void WriteDataChannelOpenAckMessage(rtc::Buffer* payload);

} // namespace webrtc

-#endif // TALK_APP_WEBRTC_SCTPUTILS_H_

+#endif // WEBRTC_API_SCTPUTILS_H_

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