Delete rtp_sender_ check in ModuleRtpRtcpImpl2::SetSendingMediaStatus
Analogous to https://webrtc-review.googlesource.com/c/src/+/267845/
Bug: webrtc:10198
Change-Id: Ib7d5e9b2a456486a419c61e7b2ce36df8960c67a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268762
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37550}
diff --git a/audio/channel_receive.cc b/audio/channel_receive.cc
index 7a4ceb2..6b8d1c1 100644
--- a/audio/channel_receive.cc
+++ b/audio/channel_receive.cc
@@ -578,7 +578,6 @@
InitFrameTransformerDelegate(std::move(frame_transformer));
rtp_rtcp_ = ModuleRtpRtcpImpl2::Create(configuration);
- rtp_rtcp_->SetSendingMediaStatus(false);
rtp_rtcp_->SetRemoteSSRC(remote_ssrc_);
// Ensure that RTCP is enabled for the created channel.
diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
index b1729c0..7a35e47 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
@@ -286,15 +286,8 @@
return rtcp_sender_.Sending();
}
-// TODO(nisse): This method shouldn't be called for a receive-only
-// stream. Delete rtp_sender_ check as soon as all applications are
-// updated.
void ModuleRtpRtcpImpl2::SetSendingMediaStatus(const bool sending) {
- if (rtp_sender_) {
- rtp_sender_->packet_generator.SetSendingMediaStatus(sending);
- } else {
- RTC_DCHECK(!sending);
- }
+ rtp_sender_->packet_generator.SetSendingMediaStatus(sending);
}
bool ModuleRtpRtcpImpl2::SendingMedia() const {