commit f39c815a1d65c6b1c387a4f33f92bfc5fd143251
author Sebastian Jansson <srte@webrtc.org> Mon Oct 14 15:32:21 2019
committer Commit Bot <commit-bot@chromium.org> Tue Oct 15 12:55:46 2019
tree 29d903c5d31aaa883b560acad3d6c867a06e03ff
parent ffc84527309d73c2779ffad9b3910a83d03b5f06

Cleanup: Replacing set extension status bool with CHECK.

This was just checked in all places were it was used, moving the check
into RtpRtcp reduces the boiler plate required at the call sites.

Also changing to always register and unregister extensions by URI to
synchronize the code in AudioSendStream with the code in RtpVideoSender.

This prepares for reducing the scope of ChannelSend.

Bug: webrtc:9883
Change-Id: Ia64d79f20eb98f46cbbbe8318770e4fcf9caa1ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155620
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29490}

call/rtp_video_sender.cc

diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index 73e356d..2b2a0a0 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -381,7 +381,7 @@
     int id = rtp_config_.extensions[i].id;
     RTC_DCHECK(RtpExtension::IsSupportedForVideo(extension));
     for (const RtpStreamSender& stream : rtp_streams_) {
-      RTC_CHECK(stream.rtp_rtcp->RegisterRtpHeaderExtension(extension, id));
+      stream.rtp_rtcp->RegisterRtpHeaderExtension(extension, id);
     }
   }