Introduce MediaTransportConfig

Currently we pass media_transport from PeerConnection to media layers. The goal of this change is to replace media_transport with struct MediaTransportCondif, which will enable adding different transports (i.e. we plan to add DatagramTransport) as well as other media-transport related settings without changing 100s of files.

TODO: In the future we should consider also adding rtp_transport in the same config, but it will require a bit more work, so I did not include it in the same change.


Bug: webrtc:9719
Change-Id: Ie31e1faa3ed9e6beefe30a3da208130509ce00cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137181
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28016}

call/call_perf_tests.cc

@@ -244,7 +244,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
     CreateMatchingReceiveConfigs(receive_transport.get());
 
     AudioSendStream::Config audio_send_config(audio_send_transport.get(),
-                                              /*media_transport=*/nullptr);
+                                              MediaTransportConfig());
     audio_send_config.rtp.ssrc = kAudioSendSsrc;
     audio_send_config.send_codec_spec = AudioSendStream::Config::SendCodecSpec(
         kAudioSendPayloadType, {"ISAC", 16000, 1});