Update RtpSenderAudioTest to call methods on RTPSenderAudio rather than RTPSender, when possible. In particular, avoid RTPSender::SendOutgoingData. Drop parameterization on the WebRTC-SendSideBwe-WithOverhead field trial, since that appears unrelated to these tests. Also delete some unused parts of the RtpSender test. Bug: webrtc:7135 Change-Id: I535bf48bb1720e2727f4a62fa3e49b2bb84394a0 Reviewed-on: https://webrtc-review.googlesource.com/c/120920 Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26516}