Primarily, this is intended to reduce flakyness of
RampUpTest.AudioTransportSequenceNumber. We shouldn't expect audio
send rate >= 300 kbps at all time in these tests. And in general, if
it's at all relevant to test that bitrate doesn't drop below the start
bitrate, a perf test isn't the right place for that.
A run of
./third_party/gtest-parallel/gtest-parallel -r 1000 -w 1000 \
--gtest_filter=RampUpTest.AudioTransportSequenceNumber \
out/Release/webrtc_perf_tests
passes when I ran it locally after this change, but fails around 4 out
of 1000 times before the change.
Bug: webrtc:8878
Change-Id: I08614ce5683c9ba6fe4b72bfde83e6a81445a59b
Reviewed-on: https://webrtc-review.googlesource.com/96900
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24523}
bb095aa99b