commit 4cd92d88ea1015d20052fb245f103746f52971ee
author Jakob Ivarsson <jakobi@webrtc.org> Sat Oct 31 11:40:43 2020
committer Commit Bot <commit-bot@chromium.org> Mon Nov 02 09:57:41 2020
tree c9802d3c5b75743dc0964fac39daddc043c8e82e
parent 4258df38e642ea2f0808865343cfcb46ceb2b3dc

Use non-null transport for RTCP in AV sync test.

This fixes a bug where TWCC feedback messages were not forwarded to the
sender which results in BWE dropping down to the minimum bitrate.

This is blocking landing of:
https://webrtc-review.googlesource.com/c/src/+/188801
since it causes excessive pacing at low bitrates.

Bug: webrtc:6762
Change-Id: I34947967a60c2a09937df33e9d6f17b51a644152
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191220
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32532}

call/call_perf_tests.cc

diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index ac1d29e..7ddf547 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -182,7 +182,6 @@
   std::unique_ptr<test::PacketTransport> audio_send_transport;
   std::unique_ptr<test::PacketTransport> video_send_transport;
   std::unique_ptr<test::PacketTransport> receive_transport;
-  test::NullTransport rtcp_send_transport;
 
   AudioSendStream* audio_send_stream;
   AudioReceiveStream* audio_receive_stream;
@@ -271,7 +270,7 @@
     AudioReceiveStream::Config audio_recv_config;
     audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc;
     audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc;
-    audio_recv_config.rtcp_send_transport = &rtcp_send_transport;
+    audio_recv_config.rtcp_send_transport = receive_transport.get();
     audio_recv_config.sync_group = kSyncGroup;
     audio_recv_config.decoder_factory = audio_decoder_factory_;
     audio_recv_config.decoder_map = {