Remove use of RtpHeaderExtension and clean up

webrtc/config.h

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  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.
  */
 
@@ skipping 34 matching lines @@
 
   // Payload type used for RED packets.
   int red_payload_type;
 
   // RTX payload type for RED payload.
   int red_rtx_payload_type;
 };
 
 // RTP header extension, see RFC 5285.
 struct RtpExtension {
-  RtpExtension(const std::string& name, int id) : name(name), id(id) {}
+  RtpExtension() : id(0) {}
+  RtpExtension(const std::string& uri, int id) : uri(uri), id(id) {}
   std::string ToString() const;
   bool operator==(const RtpExtension& rhs) const {
-    return name == rhs.name && id == rhs.id;
+    return uri == rhs.uri && id == rhs.id;
   }
-  static bool IsSupportedForAudio(const std::string& name);
-  static bool IsSupportedForVideo(const std::string& name);
+  static bool IsSupportedForAudio(const std::string& uri);
+  static bool IsSupportedForVideo(const std::string& uri);
 
-  static const char* kTOffset;
-  static const char* kAbsSendTime;
-  static const char* kVideoRotation;
-  static const char* kAudioLevel;
-  static const char* kTransportSequenceNumber;
-  std::string name;
+  // Header extension for audio levels, as defined in:
+  // http://tools.ietf.org/html/draft-ietf-avtext-client-to-mixer-audio-level-03
+  static const char* kAudioLevelUri;
+  static const int kAudioLevelDefaultId;
+
+  // Header extension for RTP timestamp offset, see RFC 5450 for details:
+  // http://tools.ietf.org/html/rfc5450
+  static const char* kTimestampOffsetUri;
+  static const int kTimestampOffsetDefaultId;
+
+  // Header extension for absolute send time, see url for details:
+  // http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
+  static const char* kAbsSendTimeUri;
+  static const int kAbsSendTimeDefaultId;
+
+  // Header extension for coordination of video orientation, see url for
+  // details:
+  // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/ts_126114v120700p.pdf
+  static const char* kVideoRotationUri;
+  static const int kVideoRotationDefaultId;
+
+  // Header extension for transport sequence number, see url for details:
+  // http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions
+  static const char* kTransportSequenceNumberUri;
+  static const int kTransportSequenceNumberDefaultId;
+
+  std::string uri;
   int id;
 };
 
 struct VideoStream {
   VideoStream();
   ~VideoStream();
   std::string ToString() const;
 
   size_t width;
   size_t height;
@@ skipping 65 matching lines @@
 struct VoicePacing {
   VoicePacing() : enabled(false) {}
   explicit VoicePacing(bool value) : enabled(value) {}
   static const ConfigOptionID identifier = ConfigOptionID::kVoicePacing;
   bool enabled;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_CONFIG_H_