commit 77938e640907ee7e0e63be0e8b86e5732dcb43bb
author Amit Hilbuch <amithi@webrtc.org> Fri Dec 21 17:23:38 2018
committer Commit Bot <commit-bot@chromium.org> Fri Dec 21 20:59:23 2018
tree 327b6b58e8b6f5a76759957473d4a9c23da9ea56
parent 86079a45719423b0011cb6f7237665112f5bf3c4

Simulcast work to enable RID mux.

Rids can now be sent using rtp_sender.
Hooking up the rid values in the voice and video engine is still WIP.

Bug: webrtc:10074
Change-Id: I245c7ecb23b67fc0ba65caaa5dbb4fcfd60c81bb
Reviewed-on: https://webrtc-review.googlesource.com/c/114505
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26092}

api/rtpparameters.cc

diff --git a/api/rtpparameters.cc b/api/rtpparameters.cc
index 2b20c56..6b253dc 100644
--- a/api/rtpparameters.cc
+++ b/api/rtpparameters.cc
@@ -137,12 +137,20 @@
     "http://www.webrtc.org/experiments/rtp-hdrext/generic-frame-descriptor-00";
 const int RtpExtension::kGenericFrameDescriptorDefaultId = 11;
 
+const char RtpExtension::kEncryptHeaderExtensionsUri[] =
+    "urn:ietf:params:rtp-hdrext:encrypt";
+
 const char RtpExtension::kColorSpaceUri[] =
     "http://www.webrtc.org/experiments/rtp-hdrext/color-space";
 const int RtpExtension::kColorSpaceDefaultId = 12;
 
-const char RtpExtension::kEncryptHeaderExtensionsUri[] =
-    "urn:ietf:params:rtp-hdrext:encrypt";
+const char RtpExtension::kRidUri[] =
+    "urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id";
+const int RtpExtension::kRidDefaultId = 13;
+
+const char RtpExtension::kRepairedRidUri[] =
+    "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id";
+const int RtpExtension::kRepairedRidDefaultId = 14;
 
 constexpr int RtpExtension::kMinId;
 constexpr int RtpExtension::kMaxId;
@@ -153,7 +161,9 @@
 bool RtpExtension::IsSupportedForAudio(const std::string& uri) {
   return uri == webrtc::RtpExtension::kAudioLevelUri ||
          uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
-         uri == webrtc::RtpExtension::kMidUri;
+         uri == webrtc::RtpExtension::kMidUri ||
+         uri == webrtc::RtpExtension::kRidUri ||
+         uri == webrtc::RtpExtension::kRepairedRidUri;
 }
 
 bool RtpExtension::IsSupportedForVideo(const std::string& uri) {
@@ -167,7 +177,9 @@
          uri == webrtc::RtpExtension::kMidUri ||
          uri == webrtc::RtpExtension::kFrameMarkingUri ||
          uri == webrtc::RtpExtension::kGenericFrameDescriptorUri ||
-         uri == webrtc::RtpExtension::kColorSpaceUri;
+         uri == webrtc::RtpExtension::kColorSpaceUri ||
+         uri == webrtc::RtpExtension::kRidUri ||
+         uri == webrtc::RtpExtension::kRepairedRidUri;
 }
 
 bool RtpExtension::IsEncryptionSupported(const std::string& uri) {
@@ -185,7 +197,9 @@
          uri == webrtc::RtpExtension::kTransportSequenceNumberUri ||
          uri == webrtc::RtpExtension::kPlayoutDelayUri ||
          uri == webrtc::RtpExtension::kVideoContentTypeUri ||
-         uri == webrtc::RtpExtension::kMidUri;
+         uri == webrtc::RtpExtension::kMidUri ||
+         uri == webrtc::RtpExtension::kRidUri ||
+         uri == webrtc::RtpExtension::kRepairedRidUri;
 }
 
 const RtpExtension* RtpExtension::FindHeaderExtensionByUri(