Add read support of RtpStreamId/RepairedRtpStreamId header extensions.

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 <stdint.h>
+#include <string>
 
 #include "webrtc/api/video/video_rotation.h"
 #include "webrtc/base/array_view.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
 class AbsoluteSendTime {
  public:
   static constexpr RTPExtensionType kId = kRtpExtensionAbsoluteSendTime;
@@ skipping 67 matching lines @@
   // 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(rtc::ArrayView<const uint8_t> data,
                     PlayoutDelay* playout_delay);
   static bool Write(uint8_t* data, const PlayoutDelay& playout_delay);
 };
 
+class RtpStreamId {
+ public:
+  static constexpr RTPExtensionType kId = kRtpExtensionRtpStreamId;
+  // TODO(danilchap): Avoid using full 16 bytes when webrtc will set the
+  // extension.
+  static constexpr uint8_t kValueSizeBytes = 16;
+  static constexpr const char* kUri =
+      "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id";
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data, std::string* rid);
+};
+
+class RepairRtpStreamId {
+ public:
+  static constexpr RTPExtensionType kId = kRtpExtensionRepairRtpStreamId;
+  // TODO(danilchap): Avoid using full 16 bytes when webrtc will set the
+  // extension.

[pthatcher1, 2017/04/06 21:17:34]

I don't understand this TODO.  Only the WebRTC API has the "full 16", no the RTP
layer.  The RTP layer isn't bounded.  

And what does "when webrtc will set the extension" mean?

[danilchap, 2017/04/07 07:07:49]

Rewritten TODO from note to self to (hopefully) more readable note.

+  static constexpr uint8_t kValueSizeBytes = 16;
+  static constexpr const char* kUri =
+      "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id";
+
+  static bool Parse(rtc::ArrayView<const uint8_t> data, std::string* rid);
+};
+
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_HEADER_EXTENSIONS_H_