Delete media transport integration.
MediaTransport is deprecated and the code is unused.
No-Try: True
Bug: webrtc:9719
Change-Id: I5b864c1e74bf04df16c15f51b8fac3d407331dcd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160620
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29923}
diff --git a/test/scenario/audio_stream.cc b/test/scenario/audio_stream.cc
index f5d2116..2738f69 100644
--- a/test/scenario/audio_stream.cc
+++ b/test/scenario/audio_stream.cc
@@ -73,8 +73,7 @@
rtc::scoped_refptr<AudioEncoderFactory> encoder_factory,
Transport* send_transport)
: sender_(sender), config_(config) {
- AudioSendStream::Config send_config(send_transport,
- webrtc::MediaTransportConfig());
+ AudioSendStream::Config send_config(send_transport);
ssrc_ = sender->GetNextAudioSsrc();
send_config.rtp.ssrc = ssrc_;
SdpAudioFormat::Parameters sdp_params;
