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 <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20183}diff --git a/modules/audio_device/audio_device_unittest.cc b/modules/audio_device/audio_device_unittest.cc
index daaefd4..e1f0c15 100644
--- a/modules/audio_device/audio_device_unittest.cc
+++ b/modules/audio_device/audio_device_unittest.cc
@@ -703,8 +703,9 @@
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");
}
