iOS audio session isInterrupted flag does not get reset correctly:

webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm

 /*
  *  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 807 matching lines @@
       std::max(kTestTimeOutInMilliseconds, 1000 * kMeasureLatencyTimeInSec));
   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();
 }
 
+// Verifies that the AudioDeviceIOS is_interrupted_ flag is reset correctly
+// after an iOS AVAudioSessionInterruptionTypeEnded notification event.
+// When AudioDeviceIOS is interrupted, is_interrupted_ is set to true.
+// When the audio device is stopped, the AudioDeviceIOS's
+// audio_session_observer_ is removed from RTCAudioSession's delegate list.
+// Then when the iOS AVAudioSessionInterruptionTypeEnded notification is
+// received by RTCAudioSession, RTCAudioSession can't propagate the event
+// back to AudioDeviceIOS as AudioDeviceIOS's audio_session_observer_ is no
+// longer in RTCAudioSession's delegate list.
+// The fix is to set AudioDeviceIOS's is_interrupted_ flag to RTCAudioSession's
+// isInterrupted flag when AudioDeviceIOS is initialized the next time audio
+// is to be started.
 TEST_F(AudioDeviceTest, testInterruptedAudioSession) {
   RTCAudioSession *session = [RTCAudioSession sharedInstance];
   std::unique_ptr<webrtc::AudioDeviceIOS> audio_device;
   audio_device.reset(new webrtc::AudioDeviceIOS());
   std::unique_ptr<webrtc::AudioDeviceBuffer> audio_buffer;
   audio_buffer.reset(new webrtc::AudioDeviceBuffer());
   audio_device->AttachAudioBuffer(audio_buffer.get());
   audio_device->Init();
   audio_device->InitPlayout();
   // Force interruption.
@@ skipping 13 matching lines @@
   // Wait for notification to propagate.
   rtc::MessageQueueManager::ProcessAllMessageQueues();
   EXPECT_TRUE(audio_device->is_interrupted_);
 
   audio_device->Init();
   audio_device->InitPlayout();
   EXPECT_FALSE(audio_device->is_interrupted_);
 }
 
 }  // namespace webrtc