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..eefab0f4e1ca557db04785dc4540538d51f08207 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);
+
+  // 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