Check before send/receive rtp header extensions.
BUG=1788
R=pbos@webrtc.org, tommi@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/13949004
Patch from Changbin Shao <changbin.shao@intel.com>.
git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6744 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/config.h b/config.h
index 2e96ec1..e4bccf9 100644
--- a/config.h
+++ b/config.h
@@ -10,8 +10,8 @@
// TODO(pbos): Move Config from common.h to here.
-#ifndef WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CONFIG_H_
-#define WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CONFIG_H_
+#ifndef WEBRTC_CONFIG_H_
+#define WEBRTC_CONFIG_H_
#include <string>
#include <vector>
@@ -73,9 +73,10 @@
// RTP header extension to use for the video stream, see RFC 5285.
struct RtpExtension {
- RtpExtension(const char* name, int id) : name(name), id(id) {}
+ RtpExtension(const std::string& name, int id) : name(name), id(id) {}
std::string ToString() const;
- // TODO(mflodman) Add API to query supported extensions.
+ static bool IsSupported(const std::string& name);
+
static const char* kTOffset;
static const char* kAbsSendTime;
std::string name;
@@ -109,4 +110,4 @@
} // namespace webrtc
-#endif // WEBRTC_VIDEO_ENGINE_NEW_INCLUDE_CONFIG_H_
+#endif // WEBRTC_CONFIG_H_
