From bb095aa99bb42ccd3d42ff392ca2c04154982a51 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Niels=20M=C3=B6ller?= Date: Thu, 30 Aug 2018 15:46:50 +0200 Subject: [PATCH] Allow send bitrate < start bitrate in RampUpTest. 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 Commit-Queue: Niels Moller Cr-Commit-Position: refs/heads/master@{#24523} --- call/rampup_tests.cc | 1 - 1 file changed, 1 deletion(-) diff --git a/call/rampup_tests.cc b/call/rampup_tests.cc index 1b2177258b..17b729629a 100644 --- a/call/rampup_tests.cc +++ b/call/rampup_tests.cc @@ -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 ||