Resolves issue with multiple calls to audio unit initialization

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 618 matching lines @@
 // which will require a new call to InitPlayout(). This test does not call
 // StartPlayout() while being uninitialized since doing so will hit a
 // RTC_DCHECK.
 TEST_F(AudioDeviceTest, StopPlayoutRequiresInitToRestart) {
   EXPECT_EQ(0, audio_device()->InitPlayout());
   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 instace shall deactivate the audio session.
+TEST_F(AudioDeviceTest, 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. Ignore the callback sequence.
+  NiceMock<MockAudioTransport> mock(kPlayout);
+  EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
+  StartPlayout();
+
+  // Initialize playout for the second ADM. If all is OK, the second ADM shall
+  // reuse the audio session activated when the first ADM started playing.
+  // This call will also ensure that we avoid a problem related to initializing
+  // two different audio unit instances back to back (see webrtc:5166 for
+  // details).
+  EXPECT_EQ(0, second_audio_device->InitPlayout());
+  EXPECT_TRUE(second_audio_device->PlayoutIsInitialized());
+
+  // Stop playout for the default ADM. The audio session shall not be
+  // deactivated since it is used by the second ADM.
+  StopPlayout();
+
+  // Start playout for the second ADM and verify that it starts as intended.
+  // Passing this test ensures that initialization of the second audio unit
+  // has been done successfully.
+  MockAudioTransport mock2(kPlayout);
+  mock2.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
+  EXPECT_CALL(
+      mock2, NeedMorePlayData(playout_frames_per_10ms_buffer(), kBytesPerSample,
+                              playout_channels(), playout_sample_rate(),
+                              NotNull(), _, _, _))
+      .Times(AtLeast(kNumCallbacks));
+  EXPECT_EQ(0, second_audio_device->RegisterAudioCallback(&mock2));
+  EXPECT_EQ(0, second_audio_device->StartPlayout());
+  EXPECT_TRUE(second_audio_device->Playing());
+  test_is_done_->Wait(kTestTimeOutInMilliseconds);
+  EXPECT_EQ(0, second_audio_device->StopPlayout());
+  EXPECT_FALSE(second_audio_device->Playing());
+  EXPECT_FALSE(second_audio_device->PlayoutIsInitialized());
+
+  EXPECT_EQ(0, second_audio_device->Terminate());
+}
+
 // Start playout and verify that the native audio layer starts asking for real
 // audio samples to play out using the NeedMorePlayData callback.
 TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
   MockAudioTransport mock(kPlayout);
   mock.HandleCallbacks(test_is_done_.get(), nullptr, kNumCallbacks);
   EXPECT_CALL(mock, NeedMorePlayData(playout_frames_per_10ms_buffer(),
                                      kBytesPerSample, playout_channels(),
                                      playout_sample_rate(), NotNull(), _, _, _))
       .Times(AtLeast(kNumCallbacks));
   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
@@ skipping 129 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