commit dd2eebef5e578877b2ddaccef4aafa360c7e288f
author Artem Titov <titovartem@webrtc.org> Mon Aug 20 13:27:45 2018 +0200
committer Commit Bot <commit-bot@chromium.org> Mon Aug 20 11:47:28 2018 +0000
tree 06b6246c09bd3392d6ff6c26f21c1fca7df7c0b5
parent 908689d04718c02f7e29f12431faf447187cc713

Deprecate two DirectTransport ctors and remove their direct usage.

Because DirectTransport is switched on SimulatedPacketReceiverInterface
we can't create it from some specific config in ctor, so all ctors,
that accept specific config are deprecated and you should pass concrete
implementation of underlying implememntation instead.

Bug: webrtc:9630
Change-Id: I7f241f310c993d8136b40898e55a6915924d61bd
Reviewed-on: https://webrtc-review.googlesource.com/94841
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24344}

video/BUILD.gn

diff --git a/video/BUILD.gn b/video/BUILD.gn
index 5901fb9..d2a52fb 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -206,6 +206,7 @@
       "../api:fec_controller_api",
       "../api:video_quality_test_fixture_api",
       "../call:fake_network",
+      "../call:simulated_network",
       "../logging:rtc_event_log_api",
       "../logging:rtc_event_log_impl_output",
       "../media:rtc_audio_video",
@@ -429,10 +430,12 @@
       "../api/video:video_frame_i420",
       "../api/video_codecs:video_codecs_api",
       "../call:call_interfaces",
+      "../call:fake_network",
       "../call:mock_bitrate_allocator",
       "../call:mock_rtp_interfaces",
       "../call:rtp_receiver",
       "../call:rtp_sender",
+      "../call:simulated_network",
       "../call:video_stream_api",
       "../common_video",
       "../logging:rtc_event_log_api",