Remove PeerConnection::GetRtpTransport

This function isn't used anymore.

Bug: webrtc:9987
Change-Id: I37f1c86cc4802950347db302e8a9207b9dd370bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330261
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41327}
This commit is contained in:
Tommi 2023-12-06 15:14:04 +01:00 committed by WebRTC LUCI CQ
parent 8a29d89e99
commit d6601ce66b
2 changed files with 0 additions and 15 deletions

View File

@ -1031,18 +1031,6 @@ PeerConnection::AddTransceiver(
return AddTransceiver(track, RtpTransceiverInit());
}
RtpTransportInternal* PeerConnection::GetRtpTransport(const std::string& mid) {
// TODO(bugs.webrtc.org/9987): Avoid the thread jump.
// This might be done by caching the value on the signaling thread.
RTC_DCHECK_RUN_ON(signaling_thread());
return network_thread()->BlockingCall([this, &mid] {
RTC_DCHECK_RUN_ON(network_thread());
auto rtp_transport = transport_controller_->GetRtpTransport(mid);
RTC_DCHECK(rtp_transport);
return rtp_transport;
});
}
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
PeerConnection::AddTransceiver(
rtc::scoped_refptr<MediaStreamTrackInterface> track,

View File

@ -417,9 +417,6 @@ class PeerConnection : public PeerConnectionInternal,
const RtpTransceiverInit& init,
bool fire_callback = true) override;
// Returns rtp transport, result can not be nullptr.
RtpTransportInternal* GetRtpTransport(const std::string& mid);
// Returns true if SRTP (either using DTLS-SRTP or SDES) is required by
// this session.
bool SrtpRequired() const override;