iOS audio session isInterrupted flag does not get reset correctly:

webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm

Index: webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm
diff --git a/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc b/webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm
similarity index 96%
rename from webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc
rename to webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm
index 1b7ff5780f9177be6132f184b0b6e4713e904d24..b889642858373590b870aad969ad99f440d116aa 100644
--- a/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc
+++ b/webrtc/modules/audio_device/ios/audio_device_unittest_ios.mm
@@ -32,6 +32,9 @@
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
+#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession.h"
+#import "webrtc/modules/audio_device/ios/objc/RTCAudioSession+Private.h"
+
 using std::cout;
 using std::endl;
 using ::testing::_;
@@ -822,4 +825,33 @@ TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
   latency_audio_stream->PrintResults();
 }
 
+TEST_F(AudioDeviceTest, testInterruptedAudioSession) {
+  RTCAudioSession *session = [RTCAudioSession sharedInstance];
+  std::unique_ptr<webrtc::AudioDeviceIOS> audioDevice;

[tkchin_webrtc, 2017/05/09 21:07:02]

use C++ style in C++ file. underscores.

+  audioDevice.reset(new webrtc::AudioDeviceIOS());
+  std::unique_ptr<webrtc::AudioDeviceBuffer> audioBuffer;

[tkchin_webrtc, 2017/05/09 21:07:02]

There are several methods in this test file already that create buffers / start
playout etc. Reuse those?

+  audioBuffer.reset(new webrtc::AudioDeviceBuffer());
+  audioDevice->AttachAudioBuffer(audioBuffer.get());
+  audioDevice->Init();
+  audioDevice->InitPlayout();
+  [session notifyDidBeginInterruption]; // force interruption

[tkchin_webrtc, 2017/05/09 21:07:02]

nit: we typically do not comment inline, move comment before, use periods and
capitalize

+
+  // Wait for notification to propagate
+  rtc::MessageQueueManager::ProcessAllMessageQueues();
+  EXPECT_TRUE(audioDevice->is_interrupted_);
+
+  audioDevice->playing_ = false;  // force it for testing
+  audioDevice->ShutdownPlayOrRecord();
+  audioDevice->audio_is_initialized_ = false; // force it for testing
+
+  [session notifyDidEndInterruptionWithShouldResumeSession:YES];
+  // Wait for notification to propagate
+  rtc::MessageQueueManager::ProcessAllMessageQueues();
+  EXPECT_TRUE(audioDevice->is_interrupted_);
+
+  audioDevice->Init();
+  audioDevice->InitPlayout();
+  EXPECT_FALSE(audioDevice->is_interrupted_);
+}
+
 }  // namespace webrtc