Add read support of RtpStreamId/RepairedRtpStreamId header extensions.

webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h

Index: webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h b/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
index 543688c75e8c91ec36006ce8026b7c6e8e0ca7a9..9a8e42e94d661f6f74a13b9f57bd5efdb3299827 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h
@@ -11,6 +11,7 @@
 #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"
@@ -98,5 +99,29 @@ class PlayoutDelayLimits {
   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_