Simplify and extend RtpHeaderExtensionMap

webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h

 /*
  *  Copyright (c) 2016 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.
  */
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_
 
 #include "webrtc/base/basictypes.h"
 #include "webrtc/common_video/rotation.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
 class AbsoluteSendTime {
  public:
   static constexpr RTPExtensionType kId = kRtpExtensionAbsoluteSendTime;
   static constexpr uint8_t kValueSizeBytes = 3;
-  static const char* const kUri;
+  static constexpr const char* kUri =
+      "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time";
 
   static bool Parse(const uint8_t* data, uint32_t* time_24bits);
   static bool Write(uint8_t* data, int64_t time_ms);
 
   static constexpr uint32_t MsTo24Bits(int64_t time_ms) {
     return static_cast<uint32_t>(((time_ms << 18) + 500) / 1000) & 0x00FFFFFF;
   }
 };
 
 class AudioLevel {
  public:
   static constexpr RTPExtensionType kId = kRtpExtensionAudioLevel;
   static constexpr uint8_t kValueSizeBytes = 1;
-  static const char* const kUri;
+  static constexpr const char* kUri =
+      "urn:ietf:params:rtp-hdrext:ssrc-audio-level";
 
   static bool Parse(const uint8_t* data,
                     bool* voice_activity,
                     uint8_t* audio_level);
   static bool Write(uint8_t* data, bool voice_activity, uint8_t audio_level);
 };
 
 class TransmissionOffset {
  public:
   static constexpr RTPExtensionType kId = kRtpExtensionTransmissionTimeOffset;
   static constexpr uint8_t kValueSizeBytes = 3;
-  static const char* const kUri;
+  static constexpr const char* kUri = "urn:ietf:params:rtp-hdrext:toffset";
 
   static bool Parse(const uint8_t* data, int32_t* rtp_time);
   static bool Write(uint8_t* data, int32_t rtp_time);
 };
 
 class TransportSequenceNumber {
  public:
   static constexpr RTPExtensionType kId = kRtpExtensionTransportSequenceNumber;
   static constexpr uint8_t kValueSizeBytes = 2;
-  static const char* const kUri;
-
+  static constexpr const char* kUri =
+      "http://www.ietf.org/id/"
+      "draft-holmer-rmcat-transport-wide-cc-extensions-01";
   static bool Parse(const uint8_t* data, uint16_t* value);
   static bool Write(uint8_t* data, uint16_t value);
 };
 
 class VideoOrientation {
  public:
   static constexpr RTPExtensionType kId = kRtpExtensionVideoRotation;
   static constexpr uint8_t kValueSizeBytes = 1;
-  static const char* const kUri;
+  static constexpr const char* kUri = "urn:3gpp:video-orientation";
 
   static bool Parse(const uint8_t* data, VideoRotation* value);
   static bool Write(uint8_t* data, VideoRotation value);
   static bool Parse(const uint8_t* data, uint8_t* value);
   static bool Write(uint8_t* data, uint8_t value);
 };
 
 class PlayoutDelayLimits {
  public:
   static constexpr RTPExtensionType kId = kRtpExtensionPlayoutDelay;
   static constexpr uint8_t kValueSizeBytes = 3;
-  static const char* const kUri;
+  static constexpr const char* kUri =
+      "http://www.webrtc.org/experiments/rtp-hdrext/playout-delay";
 
   // Playout delay in milliseconds. A playout delay limit (min or max)
   // has 12 bits allocated. This allows a range of 0-4095 values which
   // translates to a range of 0-40950 in milliseconds.
   static constexpr int kGranularityMs = 10;
   // Maximum playout delay value in milliseconds.
   static constexpr int kMaxMs = 0xfff * kGranularityMs;  // 40950.
 
   static bool Parse(const uint8_t* data, PlayoutDelay* playout_delay);
   static bool Write(uint8_t* data, const PlayoutDelay& playout_delay);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_