Skip audio_device_unittest when platform audio isn't used.

webrtc/modules/audio_device/audio_device_unittest.cc

 /*
  *  Copyright (c) 2017 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 188 matching lines @@
   AudioDeviceTest() : event_(false, false) {
 #if !defined(ADDRESS_SANITIZER) && !defined(MEMORY_SANITIZER)
     rtc::LogMessage::LogToDebug(rtc::LS_INFO);
     // Add extra logging fields here if needed for debugging.
     // rtc::LogMessage::LogTimestamps();
     // rtc::LogMessage::LogThreads();
     audio_device_ =
         AudioDeviceModule::Create(0, AudioDeviceModule::kPlatformDefaultAudio);
     EXPECT_NE(audio_device_.get(), nullptr);
     AudioDeviceModule::AudioLayer audio_layer;
-    EXPECT_EQ(0, audio_device_->ActiveAudioLayer(&audio_layer));
-    if (audio_layer == AudioDeviceModule::kLinuxAlsaAudio) {
+    int got_platform_audio_layer =
+        audio_device_->ActiveAudioLayer(&audio_layer);
+    if (got_platform_audio_layer != 0 ||
+        audio_layer == AudioDeviceModule::kLinuxAlsaAudio) {
       requirements_satisfied_ = false;
     }
     if (requirements_satisfied_) {
       EXPECT_EQ(0, audio_device_->Init());
       const int16_t num_playout_devices = audio_device_->PlayoutDevices();
       const int16_t num_record_devices = audio_device_->RecordingDevices();
       requirements_satisfied_ =
           num_playout_devices > 0 && num_record_devices > 0;
     }
 #else
@@ skipping 137 matching lines @@
       .Times(AtLeast(kNumCallbacks));
   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
   StartPlayout();
   StartRecording();
   event()->Wait(kTestTimeOutInMilliseconds);
   StopRecording();
   StopPlayout();
 }
 
 }  // namespace webrtc