Provide isAudioEnabled flag to control audio unit.

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 "WebRTC/RTCMacros.h"
-#include "webrtc/base/asyncinvoker.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 VoiceProcessingAudioUnitObserver {
+                       public VoiceProcessingAudioUnitObserver,
+                       public rtc::MessageHandler {
  public:
   AudioDeviceIOS();
   ~AudioDeviceIOS();
 
   void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) override;
 
   int32_t Init() override;
   int32_t Terminate() override;
   bool Initialized() const override { return initialized_; }
 
@@ skipping 99 matching lines @@
   bool RecordingError() const override;
   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;
-  void OnConfiguredForWebRTC() override;
+  void OnCanPlayOrRecordChange(bool can_play_or_record) 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;
+  void OnSampleRateChange(float sample_rate) override;
+
+  // Handles messages from posts.
+  void OnMessage(rtc::Message *msg) override;
 
  private:
   // Called by the relevant AudioSessionObserver methods on |thread_|.
   void HandleInterruptionBegin();
   void HandleInterruptionEnd();
   void HandleValidRouteChange();
-  void HandleConfiguredForWebRTC();
+  void HandleCanPlayOrRecordChange(bool can_play_or_record);
+  void HandleSampleRateChange(float sample_rate);
 
   // 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 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 RestartAudioUnit(float sample_rate);
 
+  // Configures the audio session for WebRTC.
+  void ConfigureAudioSession();
+  // Unconfigures the audio session.
+  void UnconfigureAudioSession();
+
   // 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();
 
   // 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 AudioDeviceModule::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
@@ skipping 49 matching lines @@
   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_;
+
+  // Set to true if we've activated the audio session.
+  bool has_configured_session_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_