Increase the threshold for RunPlayoutAndRecordingInFullDuplex.

webrtc/modules/audio_device/android/audio_device_unittest.cc

Index: webrtc/modules/audio_device/android/audio_device_unittest.cc
diff --git a/webrtc/modules/audio_device/android/audio_device_unittest.cc b/webrtc/modules/audio_device/android/audio_device_unittest.cc
index 56fb209d0253d09d192a346ac72fea3841b3f1f8..db837241a5c24eefaa7c111c1ef284b5c80bc50a 100644
--- a/webrtc/modules/audio_device/android/audio_device_unittest.cc
+++ b/webrtc/modules/audio_device/android/audio_device_unittest.cc
@@ -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);
+
+  // This threshold are set particularly high to accomodate differences in

[henrika_webrtc, 2016/10/03 14:48:44]

Nit. These thresholds are set rather high ....

And http://bugs.webrtc.org/6464.

+  // hardware in several devices, so this test can be used in swarming.
+  // See 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