Pass the media transport from JsepTransportController to Call.

Add TargetRateObservers for media transport in the call object.



Bug: webrtc:9719
Change-Id: I5448d05359cf09b8cd2a678b2ac876aa8f8970e7
Reviewed-on: https://webrtc-review.googlesource.com/c/110622
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25662}

media/engine/fakewebrtccall.h

diff --git a/media/engine/fakewebrtccall.h b/media/engine/fakewebrtccall.h
index dbcedb8..1b6deb0 100644
--- a/media/engine/fakewebrtccall.h
+++ b/media/engine/fakewebrtccall.h
@@ -273,6 +273,9 @@
   int GetNumCreatedReceiveStreams() const;
   void SetStats(const webrtc::Call::Stats& stats);
 
+  void MediaTransportChange(
+      webrtc::MediaTransportInterface* media_transport_interface) override;
+
  private:
   webrtc::AudioSendStream* CreateAudioSendStream(
       const webrtc::AudioSendStream::Config& config) override;