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Increase the threshold for RunPlayoutAndRecordingInFullDuplex.

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RunPlayoutAndRecordingInFullDuplex fails sometimes on Android swarming
bots, presumably because the timing is hardware dependent.

This test ensures that audio starts pumping. The exact performance is
not that important.

R=kjellander@webrtc.org, henrika@webrtc.org
BUG=webrtc:6464
NOTRY=True

Review-Url: https://codereview.webrtc.org/2391563002
Cr-Commit-Position: refs/heads/master@{#14492}
This commit is contained in:
ehmaldonado 2016-10-04 01:58:57 -07:00 committed by Commit bot
parent c07ebb30c5
commit ebb0b8ec9a
1 changed files with 6 additions and 2 deletions
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8
webrtc/modules/audio_device/android/audio_device_unittest.cc
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@ -1020,8 +1020,12 @@ TEST_F(AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) {
1000 * kFullDuplexTimeInSec));
StopPlayout();
StopRecording();
EXPECT_LE(fifo_audio_stream->average_size(), 10u);
EXPECT_LE(fifo_audio_stream->largest_size(), 20u);
// These thresholds are set rather high to accomodate differences in hardware
// in several devices, so this test can be used in swarming.
// See http://bugs.webrtc.org/6464
EXPECT_LE(fifo_audio_stream->average_size(), 30u);
EXPECT_LE(fifo_audio_stream->largest_size(), 40u);
}
// Measures loopback latency and reports the min, max and average values for