Support delayed AudioUnit initialization.

webrtc/modules/audio_device/ios/voice_processing_audio_unit.h

 /*
  *  Copyright 2016 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 31 matching lines @@
 // as the Remote I/O unit (supports full duplex low-latency audio input and
 // output) and adds AEC for for two-way duplex communication. It also adds AGC,
 // adjustment of voice-processing quality, and muting. Hence, ideal for
 // VoIP applications.
 class VoiceProcessingAudioUnit {
  public:
   explicit VoiceProcessingAudioUnit(VoiceProcessingAudioUnitObserver* observer);
   ~VoiceProcessingAudioUnit();
 
   // TODO(tkchin): enum for state and state checking.
+  enum State : int32_t {
+    // Init() should be called.
+    kInitRequired,
+    // Audio unit created but not initialized.
+    kUninitialized,
+    // Initialized but not started. Equivalent to stopped.
+    kInitialized,
+    // Initialized and started.
+    kStarted,
+  };
 
   // Number of bytes per audio sample for 16-bit signed integer representation.
   static const UInt32 kBytesPerSample;
 
   // Initializes this class by creating the underlying audio unit instance.
   // Creates a Voice-Processing I/O unit and configures it for full-duplex
   // audio. The selected stream format is selected to avoid internal resampling
   // and to match the 10ms callback rate for WebRTC as well as possible.
   // Does not intialize the audio unit.
   bool Init();
 
+  VoiceProcessingAudioUnit::State GetState() const;
+
   // Initializes the underlying audio unit with the given sample rate.
   bool Initialize(Float64 sample_rate);
 
   // Starts the underlying audio unit.
   bool Start();
 
   // Stops the underlying audio unit.
   bool Stop();
 
   // Uninitializes the underlying audio unit.
@@ skipping 38 matching lines @@
 
   // Returns the predetermined format with a specific sample rate. See
   // implementation file for details on format.
   AudioStreamBasicDescription GetFormat(Float64 sample_rate) const;
 
   // Deletes the underlying audio unit.
   void DisposeAudioUnit();
 
   VoiceProcessingAudioUnitObserver* observer_;
   AudioUnit vpio_unit_;
+  VoiceProcessingAudioUnit::State state_;
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_IOS_VOICE_PROCESSING_AUDIO_UNIT_H_