Move ADM Create() method to public interface.

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 519 matching lines @@
     return static_cast<AudioDeviceModuleImpl*>(audio_device_.get());
   }
 
   AudioDeviceBuffer* audio_device_buffer() const {
     return audio_device_impl()->GetAudioDeviceBuffer();
   }
 
   rtc::scoped_refptr<AudioDeviceModule> CreateAudioDevice(
       AudioDeviceModule::AudioLayer audio_layer) {
     rtc::scoped_refptr<AudioDeviceModule> module(
-        AudioDeviceModuleImpl::Create(0, audio_layer));
+        AudioDeviceModule::Create(0, audio_layer));
     return module;
   }
 
   // Returns file name relative to the resource root given a sample rate.
   std::string GetFileName(int sample_rate) {
     EXPECT_TRUE(sample_rate == 48000 || sample_rate == 44100 ||
                 sample_rate == 16000);
     char fname[64];
     snprintf(fname, sizeof(fname), "audio_device/audio_short%d",
              sample_rate / 1000);
@@ skipping 287 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