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Cleanup: Replacing set extension status bool with CHECK.

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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}
This commit is contained in:
Sebastian Jansson
2019-10-14 17:32:21 +02:00
committed by Commit Bot
parent ffc8452730
commit f39c815a1d
11 changed files with 53 additions and 41 deletions
Download Patch File Download Diff File Expand all files Collapse all files

9
modules/rtp_rtcp/source/rtp_rtcp_impl.cc
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@ -538,15 +538,20 @@ int32_t ModuleRtpRtcpImpl::RegisterSendRtpHeaderExtension(
return rtp_sender_->RegisterRtpHeaderExtension(type, id);
}
bool ModuleRtpRtcpImpl::RegisterRtpHeaderExtension(const std::string& uri,
void ModuleRtpRtcpImpl::RegisterRtpHeaderExtension(absl::string_view uri,
int id) {
return rtp_sender_->RegisterRtpHeaderExtension(uri, id);
bool registered = rtp_sender_->RegisterRtpHeaderExtension(uri, id);
RTC_CHECK(registered);
}
int32_t ModuleRtpRtcpImpl::DeregisterSendRtpHeaderExtension(
const RTPExtensionType type) {
return rtp_sender_->DeregisterRtpHeaderExtension(type);
}
void ModuleRtpRtcpImpl::DeregisterSendRtpHeaderExtension(
absl::string_view uri) {
rtp_sender_->DeregisterRtpHeaderExtension(uri);
}
// (TMMBR) Temporary Max Media Bit Rate.
bool ModuleRtpRtcpImpl::TMMBR() const {