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 14 matching lines @@
 #include "webrtc/modules/audio_device/audio_device_impl.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_device/include/mock_audio_transport.h"
 #include "webrtc/modules/audio_device/ios/audio_device_ios.h"
 #include "webrtc/system_wrappers/include/event_wrapper.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/gmock.h"
 #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::_;
 using ::testing::AtLeast;
 using ::testing::Gt;
 using ::testing::Invoke;
 using ::testing::NiceMock;
 using ::testing::NotNull;
 using ::testing::Return;
 
@@ skipping 770 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();
 }
 
+TEST_F(AudioDeviceTest, testInterruptedAudioSession) {

[henrika_webrtc, 2017/05/10 14:08:19]

Care to add some comments above this test to explain its purpose?

+  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.
+  [session notifyDidBeginInterruption];
+
+  // Wait for notification to propagate.
+  rtc::MessageQueueManager::ProcessAllMessageQueues();
+  EXPECT_TRUE(audio_device->is_interrupted_);
+
+  // Force it for testing.
+  audio_device->playing_ = false;
+  audio_device->ShutdownPlayOrRecord();
+  // Force it for testing.
+  audio_device->audio_is_initialized_ = false;
+
+  [session notifyDidEndInterruptionWithShouldResumeSession:YES];
+  // 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