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

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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}
This commit is contained in:
Jakob Ivarsson 2020-10-31 12:40:43 +01:00 committed by Commit Bot
parent 4258df38e6
commit 4cd92d88ea
1 changed files with 1 additions and 2 deletions
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3
call/call_perf_tests.cc
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@ -182,7 +182,6 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
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 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
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 = {