Refactor the AudioDevice for iOS and improve the performance and stability

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 489 matching lines @@
     EXPECT_EQ(0, audio_device_->Init());
     EXPECT_EQ(0,
               audio_device()->GetPlayoutAudioParameters(&playout_parameters_));
     EXPECT_EQ(0, audio_device()->GetRecordAudioParameters(&record_parameters_));
   }
   virtual ~AudioDeviceTest() {
     EXPECT_EQ(0, audio_device_->Terminate());
     rtc::LogMessage::LogToDebug(old_sev_);
   }
 
-  // TODO(henrika): don't use hardcoded values below.
   int playout_sample_rate() const { return playout_parameters_.sample_rate(); }
   int record_sample_rate() const { return record_parameters_.sample_rate(); }
   int playout_channels() const { return playout_parameters_.channels(); }
   int record_channels() const { return record_parameters_.channels(); }
   size_t playout_frames_per_10ms_buffer() const {
     return playout_parameters_.frames_per_10ms_buffer();
   }
   size_t record_frames_per_10ms_buffer() const {
     return record_parameters_.frames_per_10ms_buffer();
   }
 
-  int total_delay_ms() const {
-    // TODO(henrika): improve this part.
-    return 100;
-  }
-
   rtc::scoped_refptr<AudioDeviceModule> audio_device() const {
     return audio_device_;
   }
 
   AudioDeviceModuleImpl* audio_device_impl() const {
     return static_cast<AudioDeviceModuleImpl*>(audio_device_.get());
   }
 
   AudioDeviceBuffer* audio_device_buffer() const {
     return audio_device_impl()->GetAudioDeviceBuffer();
@@ skipping 247 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