Refactor AudioUnit code into its own class.

webrtc/modules/audio_device/ios/audio_device_ios.h

 /*
  *  Copyright (c) 2012 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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_
 #define WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_
 
 #include <memory>
 
-#include <AudioUnit/AudioUnit.h>
-
 #include "webrtc/base/asyncinvoker.h"
 #include "webrtc/base/objc/RTCMacros.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
 #include "webrtc/modules/audio_device/ios/audio_session_observer.h"
+#include "webrtc/modules/audio_device/ios/voice_processing_audio_unit.h"
 
 RTC_FWD_DECL_OBJC_CLASS(RTCAudioSessionDelegateAdapter);
 
 namespace webrtc {
 
 class FineAudioBuffer;
 
 // Implements full duplex 16-bit mono PCM audio support for iOS using a
 // Voice-Processing (VP) I/O audio unit in Core Audio. The VP I/O audio unit
 // supports audio echo cancellation. It also adds automatic gain control,
 // adjustment of voice-processing quality and muting.
 //
 // An instance must be created and destroyed on one and the same thread.
 // All supported public methods must also be called on the same thread.
 // A thread checker will RTC_DCHECK if any supported method is called on an
 // invalid thread.
 //
 // Recorded audio will be delivered on a real-time internal I/O thread in the
 // audio unit. The audio unit will also ask for audio data to play out on this
 // same thread.
 class AudioDeviceIOS : public AudioDeviceGeneric,
-                       public AudioSessionObserver {
+                       public AudioSessionObserver,
+                       public VoiceProcessingAudioUnitObserver {
  public:
   AudioDeviceIOS();
   ~AudioDeviceIOS();
 
   void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
 
   int32_t Init() override;
   int32_t Terminate() override;
   bool Initialized() const override { return initialized_; }
 
@@ skipping 100 matching lines @@
   void ClearPlayoutWarning() override {}
   void ClearPlayoutError() override {}
   void ClearRecordingWarning() override {}
   void ClearRecordingError() override {}
 
   // AudioSessionObserver methods. May be called from any thread.
   void OnInterruptionBegin() override;
   void OnInterruptionEnd() override;
   void OnValidRouteChange() override;
 
+  // VoiceProcessingAudioUnitObserver methods.
+  OSStatus OnDeliverRecordedData(AudioUnitRenderActionFlags* flags,
+                                 const AudioTimeStamp* time_stamp,
+                                 UInt32 bus_number,
+                                 UInt32 num_frames,
+                                 AudioBufferList* io_data) override;
+  OSStatus OnGetPlayoutData(AudioUnitRenderActionFlags* flags,
+                            const AudioTimeStamp* time_stamp,
+                            UInt32 bus_number,
+                            UInt32 num_frames,
+                            AudioBufferList* io_data) override;
+
  private:
   // Called by the relevant AudioSessionObserver methods on |thread_|.
   void HandleInterruptionBegin();
   void HandleInterruptionEnd();
   void HandleValidRouteChange();
 
   // Uses current |playout_parameters_| and |record_parameters_| to inform the
   // audio device buffer (ADB) about our internal audio parameters.
   void UpdateAudioDeviceBuffer();
 
   // Since the preferred audio parameters are only hints to the OS, the actual
   // values may be different once the AVAudioSession has been activated.
   // This method asks for the current hardware parameters and takes actions
   // if they should differ from what we have asked for initially. It also
   // defines |playout_parameters_| and |record_parameters_|.
   void SetupAudioBuffersForActiveAudioSession();
 
-  // 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.
-  // This method also initializes the created audio unit.
-  bool SetupAndInitializeVoiceProcessingAudioUnit();
+  // Creates the audio unit.
+  bool CreateAudioUnit();
 
   // Restarts active audio streams using a new sample rate. Required when e.g.
   // a BT headset is enabled or disabled.
-  bool RestartAudioUnitWithNewFormat(float sample_rate);
+  bool RestartAudioUnit(float sample_rate);
 
   // Activates our audio session, creates and initializes the voice-processing
   // audio unit and verifies that we got the preferred native audio parameters.
   bool InitPlayOrRecord();
 
   // Closes and deletes the voice-processing I/O unit.
   void ShutdownPlayOrRecord();
 
-  // Helper method for destroying the existing audio unit.
-  void DisposeAudioUnit();
-
-  // Callback function called on a real-time priority I/O thread from the audio
-  // unit. This method is used to signal that recorded audio is available.
-  static OSStatus RecordedDataIsAvailable(
-      void* in_ref_con,
-      AudioUnitRenderActionFlags* io_action_flags,
-      const AudioTimeStamp* time_stamp,
-      UInt32 in_bus_number,
-      UInt32 in_number_frames,
-      AudioBufferList* io_data);
-  OSStatus OnRecordedDataIsAvailable(
-      AudioUnitRenderActionFlags* io_action_flags,
-      const AudioTimeStamp* time_stamp,
-      UInt32 in_bus_number,
-      UInt32 in_number_frames);
-
-  // Callback function called on a real-time priority I/O thread from the audio
-  // unit. This method is used to provide audio samples to the audio unit.
-  static OSStatus GetPlayoutData(void* in_ref_con,
-                                 AudioUnitRenderActionFlags* io_action_flags,
-                                 const AudioTimeStamp* time_stamp,
-                                 UInt32 in_bus_number,
-                                 UInt32 in_number_frames,
-                                 AudioBufferList* io_data);
-  OSStatus OnGetPlayoutData(AudioUnitRenderActionFlags* io_action_flags,
-                            UInt32 in_number_frames,
-                            AudioBufferList* io_data);
-
   // Ensures that methods are called from the same thread as this object is
   // created on.
   rtc::ThreadChecker thread_checker_;
   // Thread that this object is created on.
   rtc::Thread* thread_;
   // Invoker used to execute methods on thread_.
   std::unique_ptr<rtc::AsyncInvoker> async_invoker_;
 
   // Raw pointer handle provided to us in AttachAudioBuffer(). Owned by the
   // AudioDeviceModuleImpl class and called by AudioDeviceModuleImpl::Create().
   // The AudioDeviceBuffer is a member of the AudioDeviceModuleImpl instance
   // and therefore outlives this object.
   AudioDeviceBuffer* audio_device_buffer_;
 
   // Contains audio parameters (sample rate, #channels, buffer size etc.) for
   // the playout and recording sides. These structure is set in two steps:
   // first, native sample rate and #channels are defined in Init(). Next, the
   // audio session is activated and we verify that the preferred parameters
   // were granted by the OS. At this stage it is also possible to add a third
   // component to the parameters; the native I/O buffer duration.
   // A RTC_CHECK will be hit if we for some reason fail to open an audio session
   // using the specified parameters.
   AudioParameters playout_parameters_;
   AudioParameters record_parameters_;
 
-  // The Voice-Processing I/O unit has the same characteristics 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.
-  AudioUnit vpio_unit_;
+  // The AudioUnit used to play and record audio.
+  std::unique_ptr<VoiceProcessingAudioUnit> audio_unit_;
 
   // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
   // in chunks of 10ms. It then allows for this data to be pulled in
   // a finer or coarser granularity. I.e. interacting with this class instead
   // of directly with the AudioDeviceBuffer one can ask for any number of
   // audio data samples. Is also supports a similar scheme for the recording
   // side.
   // Example: native buffer size can be 128 audio frames at 16kHz sample rate.
   // WebRTC will provide 480 audio frames per 10ms but iOS asks for 128
   // in each callback (one every 8ms). This class can then ask for 128 and the
   // FineAudioBuffer will ask WebRTC for new data only when needed and also
   // cache non-utilized audio between callbacks. On the recording side, iOS
   // can provide audio data frames of size 128 and these are accumulated until
   // enough data to supply one 10ms call exists. This 10ms chunk is then sent
   // to WebRTC and the remaining part is stored.
   std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
 
   // Extra audio buffer to be used by the playout side for rendering audio.
   // The buffer size is given by FineAudioBuffer::RequiredBufferSizeBytes().
-  std::unique_ptr<SInt8[]> playout_audio_buffer_;
+  std::unique_ptr<int8_t[]> playout_audio_buffer_;
 
   // Provides a mechanism for encapsulating one or more buffers of audio data.
   // Only used on the recording side.
   AudioBufferList audio_record_buffer_list_;
 
   // Temporary storage for recorded data. AudioUnitRender() renders into this
   // array as soon as a frame of the desired buffer size has been recorded.
-  std::unique_ptr<SInt8[]> record_audio_buffer_;
+  std::unique_ptr<int8_t[]> record_audio_buffer_;
 
   // Set to 1 when recording is active and 0 otherwise.
   volatile int recording_;
 
   // Set to 1 when playout is active and 0 otherwise.
   volatile int playing_;
 
   // Set to true after successful call to Init(), false otherwise.
   bool initialized_;
 
   // Set to true after successful call to InitRecording(), false otherwise.
   bool rec_is_initialized_;
 
   // Set to true after successful call to InitPlayout(), false otherwise.
   bool play_is_initialized_;
 
   // Set to true if audio session is interrupted, false otherwise.
   bool is_interrupted_;
 
   // Audio interruption observer instance.
   RTCAudioSessionDelegateAdapter* audio_session_observer_;
 
   // Contains the audio data format specification for a stream of audio.
   AudioStreamBasicDescription application_format_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_