Re-enables AudioDeviceTest.StartStopPlayout on Android

webrtc/modules/audio_device/android/audio_device_unittest.cc

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 798 matching lines @@
   const int default_volume = GetSpeakerVolume();
   const int max_volume = GetMaxSpeakerVolume();
   EXPECT_EQ(0, audio_device()->SetSpeakerVolume(max_volume));
   int new_volume = GetSpeakerVolume();
   EXPECT_EQ(new_volume, max_volume);
   EXPECT_EQ(0, audio_device()->SetSpeakerVolume(default_volume));
 }
 
 // Tests that playout can be initiated, started and stopped. No audio callback
 // is registered in this test.
-// Flaky on our trybots makes this test unusable.
-// https://code.google.com/p/webrtc/issues/detail?id=5046
-TEST_F(AudioDeviceTest, DISABLED_StartStopPlayout) {
+TEST_F(AudioDeviceTest, StartStopPlayout) {
   StartPlayout();
   StopPlayout();
   StartPlayout();
   StopPlayout();
 }
 
 // Tests that recording can be initiated, started and stopped. No audio callback
 // is registered in this test.
 TEST_F(AudioDeviceTest, StartStopRecording) {
   StartRecording();
@@ skipping 184 matching lines @@
   StopPlayout();
   StopRecording();
   // Verify that the correct number of transmitted impulses are detected.
   EXPECT_EQ(latency_audio_stream->num_latency_values(),
             static_cast<size_t>(
                 kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1));
   latency_audio_stream->PrintResults();
 }
 
 }  // namespace webrtc