Fixes minor issue in AudioDeviceTest.RunPlayoutAndRecordingInFullDuplex for iOS

webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc

 /*
  *  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 565 matching lines @@
     const int seconds =
         static_cast<int>(bytes / (sample_rate * kBytesPerSample));
     PRINTD("file size: %d [secs]\n", seconds);
     PRINTD("file size: %" PRIuS " [callbacks]\n",
            seconds * kNumCallbacksPerSecond);
 #endif
     return file_name;
   }
 
   void StartPlayout() {
-    EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
     EXPECT_FALSE(audio_device()->Playing());
     EXPECT_EQ(0, audio_device()->InitPlayout());
     EXPECT_TRUE(audio_device()->PlayoutIsInitialized());
     EXPECT_EQ(0, audio_device()->StartPlayout());
     EXPECT_TRUE(audio_device()->Playing());
   }
 
   void StopPlayout() {
     EXPECT_EQ(0, audio_device()->StopPlayout());
     EXPECT_FALSE(audio_device()->Playing());
-    EXPECT_FALSE(audio_device()->PlayoutIsInitialized());
   }
 
   void StartRecording() {
-    EXPECT_FALSE(audio_device()->RecordingIsInitialized());
     EXPECT_FALSE(audio_device()->Recording());
     EXPECT_EQ(0, audio_device()->InitRecording());
     EXPECT_TRUE(audio_device()->RecordingIsInitialized());
     EXPECT_EQ(0, audio_device()->StartRecording());
     EXPECT_TRUE(audio_device()->Recording());
   }
 
   void StopRecording() {
     EXPECT_EQ(0, audio_device()->StopRecording());
     EXPECT_FALSE(audio_device()->Recording());
@@ skipping 244 matching lines @@
   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();
 }
 
 }  // namespace webrtc