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Allow send bitrate < start bitrate in RampUpTest.

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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}
This commit is contained in:
Niels Möller 2018-08-30 15:46:50 +02:00 committed by Commit Bot
parent 689b5874d4
commit bb095aa99b
1 changed files with 0 additions and 1 deletions
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1
call/rampup_tests.cc
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@ -306,7 +306,6 @@ void RampUpTester::PollStats() {
if (sender_call_) {
Call::Stats stats = sender_call_->GetStats();
EXPECT_GE(stats.send_bandwidth_bps, start_bitrate_bps_);
EXPECT_GE(expected_bitrate_bps_, 0);
if (stats.send_bandwidth_bps >= expected_bitrate_bps_ &&
(min_run_time_ms_ == -1 ||