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

[henrika_webrtc, 2017/05/11 08:00:28]

Please use 80 characters per line as in rest of the file.

+// AVAudioSessionInterruptionTypeEnded notification event.
+// When AudioDeviceIOS is interrupted, is_interrupted_ set to true.

[henrika_webrtc, 2017/05/11 08:00:28]

'is_interrupted is set to true'

+// When the audio is stopped, the RTCAudioSessionDelegate would be removed.
+// Then when the iOS AVAudioSessionInterruptionTypeEnded notification is received, the event
+// is not passed to the delegate as it is no longer in the delegates list.

[henrika_webrtc, 2017/05/11 08:00:28]

Could you make it more clear that this is the expected behaviour, e.g. by using
'should' instead of 'is' here?

 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