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/video/video_quality_test.cc b/video/video_quality_test.cc
index 8f7d612..ad8c808 100644
--- a/video/video_quality_test.cc
+++ b/video/video_quality_test.cc
@@ -22,7 +22,6 @@
#include "api/rtc_event_log_output_file.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/task_queue/task_queue_base.h"
-#include "api/transport/media/media_transport_config.h"
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/video_codecs/video_encoder.h"
#include "call/fake_network_pipe.h"
@@ -1410,8 +1409,7 @@
}
void VideoQualityTest::SetupAudio(Transport* transport) {
- AudioSendStream::Config audio_send_config(transport,
- webrtc::MediaTransportConfig());
+ AudioSendStream::Config audio_send_config(transport);
audio_send_config.rtp.ssrc = kAudioSendSsrc;
// Add extension to enable audio send side BWE, and allow audio bit rate
