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}
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@ -1031,18 +1031,6 @@ PeerConnection::AddTransceiver(
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return AddTransceiver(track, RtpTransceiverInit());
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}
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RtpTransportInternal* PeerConnection::GetRtpTransport(const std::string& mid) {
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// TODO(bugs.webrtc.org/9987): Avoid the thread jump.
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// This might be done by caching the value on the signaling thread.
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RTC_DCHECK_RUN_ON(signaling_thread());
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return network_thread()->BlockingCall([this, &mid] {
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RTC_DCHECK_RUN_ON(network_thread());
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auto rtp_transport = transport_controller_->GetRtpTransport(mid);
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RTC_DCHECK(rtp_transport);
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return rtp_transport;
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});
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}
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RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
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PeerConnection::AddTransceiver(
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rtc::scoped_refptr<MediaStreamTrackInterface> track,
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@ -417,9 +417,6 @@ class PeerConnection : public PeerConnectionInternal,
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const RtpTransceiverInit& init,
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bool fire_callback = true) override;
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// Returns rtp transport, result can not be nullptr.
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RtpTransportInternal* GetRtpTransport(const std::string& mid);
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// Returns true if SRTP (either using DTLS-SRTP or SDES) is required by
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// this session.
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bool SrtpRequired() const override;
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