Disable AudioDeviceTest.StartPlayoutOnTwoInstances on iOS

webrtc/modules/audio_device/ios/audio_device_unittest_ios.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 602 matching lines @@
   EXPECT_EQ(0, audio_device()->StartPlayout());
   EXPECT_EQ(0, audio_device()->StopPlayout());
   EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
 }
 
 // Verify that we can create two ADMs and start playing on the second ADM.
 // Only the first active instance shall activate an audio session and the
 // last active instance shall deactivate the audio session. The test does not
 // explicitly verify correct audio session calls but instead focuses on
 // ensuring that audio starts for both ADMs.
-TEST_F(AudioDeviceTest, StartPlayoutOnTwoInstances) {
+
+// Failing when running on real iOS devices: bugs.webrtc.org/6889.
+TEST_F(AudioDeviceTest, DISABLED_StartPlayoutOnTwoInstances) {
   // Create and initialize a second/extra ADM instance. The default ADM is
   // created by the test harness.
   rtc::scoped_refptr<AudioDeviceModule> second_audio_device =
       CreateAudioDevice(AudioDeviceModule::kPlatformDefaultAudio);
   EXPECT_NE(second_audio_device.get(), nullptr);
   EXPECT_EQ(0, second_audio_device->Init());
 
   // Start playout for the default ADM but don't wait here. Instead use the
   // upcoming second stream for that. We set the same expectation on number
   // of callbacks as for the second stream.
@@ skipping 179 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