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}
diff --git a/audio/BUILD.gn b/audio/BUILD.gn
index 4646eb1..7539f37 100644
--- a/audio/BUILD.gn
+++ b/audio/BUILD.gn
@@ -123,6 +123,7 @@
deps = [
":audio",
":audio_end_to_end_test",
+ "../api:libjingle_peerconnection_api",
"../api:loopback_media_transport",
"../api:mock_audio_mixer",
"../api:mock_frame_decryptor",