From b6ca72154f3e646f932d17c36d89901ec5763517 Mon Sep 17 00:00:00 2001 From: henrika Date: Fri, 6 Oct 2017 12:47:52 +0200 Subject: [PATCH] Changes test condition in AudioDeviceTest.RunPlayoutAndRecordingInFullDuplex TBR=henrik.lundin Bug: webrtc:7744 Change-Id: I8e8adc666e4734a9333dc3eaa273f68d66a8d0af Reviewed-on: https://webrtc-review.googlesource.com/7260 Commit-Queue: Henrik Andreassson Reviewed-by: Henrik Andreassson Cr-Commit-Position: refs/heads/master@{#20183} --- modules/audio_device/audio_device_unittest.cc | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/modules/audio_device/audio_device_unittest.cc b/modules/audio_device/audio_device_unittest.cc index daaefd4c17..e1f0c1564e 100644 --- a/modules/audio_device/audio_device_unittest.cc +++ b/modules/audio_device/audio_device_unittest.cc @@ -703,8 +703,9 @@ TEST_F(AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) { StopPlayout(); // This thresholds is set rather high to accommodate differences in hardware // in several devices. The main idea is to capture cases where a very large - // latency is built up. - EXPECT_LE(audio_stream.average_size(), 5u); + // latency is built up. See http://bugs.webrtc.org/7744 for examples on + // bots where relatively large average latencies can happen. + EXPECT_LE(audio_stream.average_size(), 25u); PRINT("\n"); }