Fixes issue in ADM on Mac OSX when audio is renegotiated

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.
  */
 
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 // Tests Start/Stop recording without any registered audio callback.
 TEST_F(AudioDeviceTest, StartStopRecording) {
   SKIP_TEST_IF_NOT(requirements_satisfied());
   StartRecording();
   StopRecording();
   StartRecording();
   StopRecording();
 }
 
+// Tests Init/Stop/Init recording without any registered audio callback.
+// See https://bugs.chromium.org/p/webrtc/issues/detail?id=8041 for details
+// on why this test is useful.
+TEST_F(AudioDeviceTest, InitStopInitRecording) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  EXPECT_EQ(0, audio_device()->InitRecording());
+  EXPECT_TRUE(audio_device()->RecordingIsInitialized());
+  StopRecording();
+  EXPECT_EQ(0, audio_device()->InitRecording());
+  StopRecording();
+}
+
+// Tests Init/Stop/Init recording while playout is active.
+TEST_F(AudioDeviceTest, InitStopInitRecordingWhilePlaying) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  StartPlayout();
+  EXPECT_EQ(0, audio_device()->InitRecording());
+  EXPECT_TRUE(audio_device()->RecordingIsInitialized());
+  StopRecording();
+  EXPECT_EQ(0, audio_device()->InitRecording());
+  StopRecording();
+  StopPlayout();
+}
+
+// Tests Init/Stop/Init playout without any registered audio callback.
+TEST_F(AudioDeviceTest, InitStopInitPlayout) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  EXPECT_EQ(0, audio_device()->InitPlayout());
+  EXPECT_TRUE(audio_device()->PlayoutIsInitialized());
+  StopPlayout();
+  EXPECT_EQ(0, audio_device()->InitPlayout());
+  StopPlayout();
+}
+
+// Tests Init/Stop/Init playout while recording is active.
+TEST_F(AudioDeviceTest, InitStopInitPlayoutWhileRecording) {
+  SKIP_TEST_IF_NOT(requirements_satisfied());
+  StartRecording();
+  EXPECT_EQ(0, audio_device()->InitPlayout());
+  EXPECT_TRUE(audio_device()->PlayoutIsInitialized());
+  StopPlayout();
+  EXPECT_EQ(0, audio_device()->InitPlayout());
+  StopPlayout();
+  StopRecording();
+}
+
 // Start playout and verify that the native audio layer starts asking for real
 // audio samples to play out using the NeedMorePlayData() callback.
 // Note that we can't add expectations on audio parameters in EXPECT_CALL
 // since parameter are not provided in the each callback. We therefore test and
 // verify the parameters in the fake audio transport implementation instead.
 TEST_F(AudioDeviceTest, StartPlayoutVerifyCallbacks) {
   SKIP_TEST_IF_NOT(requirements_satisfied());
   MockAudioTransport mock(TransportType::kPlay);
   mock.HandleCallbacks(event(), nullptr, kNumCallbacks);
   EXPECT_CALL(mock, NeedMorePlayData(_, _, _, _, NotNull(), _, _, _))
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   StopPlayout();
   // Verify that the correct number of transmitted impulses are detected.
   EXPECT_EQ(audio_stream.num_latency_values(),
             static_cast<size_t>(
                 kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1));
   // Print out min, max and average delay values for debugging purposes.
   audio_stream.PrintResults();
 }
 
 }  // namespace webrtc