C++ AudioUnitInitialize函数代码示例

gboolean
gst_core_audio_open (GstCoreAudio * core_audio)
{
  OSStatus status;

  /* core_audio->osxbuf is already locked at this point */
  core_audio->cached_caps_valid = FALSE;
  gst_caps_replace (&core_audio->cached_caps, NULL);

  if (!gst_core_audio_open_impl (core_audio))
    return FALSE;

  /* Add property listener */
  status = AudioUnitAddPropertyListener (core_audio->audiounit,
      kAudioUnitProperty_AudioChannelLayout, _audio_unit_property_listener,
      core_audio);
  if (status != noErr) {
    GST_ERROR_OBJECT (core_audio, "Failed to add audio channel layout property "
        "listener for AudioUnit: %d", (int) status);
  }
  status = AudioUnitAddPropertyListener (core_audio->audiounit,
      kAudioUnitProperty_StreamFormat, _audio_unit_property_listener,
      core_audio);
  if (status != noErr) {
    GST_ERROR_OBJECT (core_audio, "Failed to add stream format property "
        "listener for AudioUnit: %d", (int) status);
  }

  /* Initialize the AudioUnit. We keep the audio unit initialized early so that
   * we can probe the underlying device. */
  status = AudioUnitInitialize (core_audio->audiounit);
  if (status) {
    GST_ERROR_OBJECT (core_audio, "Failed to initialize AudioUnit: %d",
        (int) status);
    return FALSE;
  }

  return TRUE;
}

gboolean
gst_core_audio_initialize (GstCoreAudio * core_audio,
    AudioStreamBasicDescription format, GstCaps * caps, gboolean is_passthrough)
{
  guint32 frame_size;
  OSStatus status;

  GST_DEBUG_OBJECT (core_audio,
      "Initializing: passthrough:%d caps:%" GST_PTR_FORMAT, is_passthrough,
      caps);

  if (!gst_core_audio_initialize_impl (core_audio, format, caps,
          is_passthrough, &frame_size)) {
    goto error;
  }

  if (core_audio->is_src) {
    /* create AudioBufferList needed for recording */
    core_audio->recBufferList =
        buffer_list_alloc (format.mChannelsPerFrame,
        frame_size * format.mBytesPerFrame);
  }

  /* Initialize the AudioUnit */
  status = AudioUnitInitialize (core_audio->audiounit);
  if (status) {
    GST_ERROR_OBJECT (core_audio, "Failed to initialise AudioUnit: %d",
        (int) status);
    goto error;
  }
  return TRUE;

error:
  if (core_audio->is_src && core_audio->recBufferList) {
    buffer_list_free (core_audio->recBufferList);
    core_audio->recBufferList = NULL;
  }
  return FALSE;
}

bool auLoader::initialize()
{
  /** Algorithm: **/
  /** Call the AU's Initialize method **/
  OSStatus err = AudioUnitInitialize(m_plugin);
  if(err != noErr)
  {
    debug(LOG_ERROR, "Could not initialize plugin");
    return false;
  }
  
  /** Set up output buffers **/
  m_buffer_list = (AudioBufferList *)malloc(offsetof(AudioBufferList, mBuffers[MAX_CHANNELS]));
  m_buffer_list->mNumberBuffers = MAX_CHANNELS;
  
  /** Connect input properties **/
  AURenderCallbackStruct callback;
  callback.inputProc = this->inputCallback;
  callback.inputProcRefCon = this;
  
  /** Set up render notifications **/
  err = AudioUnitSetProperty(m_plugin, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input,
                             0, &callback, sizeof(callback));
  if(err != noErr)
  {
    debug(LOG_WARN, "Could not configure inputs");
  }
  
  err = AudioUnitAddRenderNotify(m_plugin, this->renderNotify, NULL);
  if(err != noErr)
  {
    debug(LOG_ERROR, "Could not set up render notification");
  }
  
  debug(LOG_INFO, "AU initialized");
  return true;
}

void iOSCoreAudioInit()
{
    if (!audioInstance) {
        OSErr err;
        
        // first, grab the default output
        AudioComponentDescription defaultOutputDescription;
        defaultOutputDescription.componentType = kAudioUnitType_Output;
        defaultOutputDescription.componentSubType = kAudioUnitSubType_RemoteIO;
        defaultOutputDescription.componentManufacturer = kAudioUnitManufacturer_Apple;
        defaultOutputDescription.componentFlags = 0;
        defaultOutputDescription.componentFlagsMask = 0;
        AudioComponent defaultOutput = AudioComponentFindNext(NULL, &defaultOutputDescription);
        
        // create our instance
        err = AudioComponentInstanceNew(defaultOutput, &audioInstance);
        if (err != noErr) {
            audioInstance = nil;
            return;
        }
        
        // create our callback so we can give it the audio data
        AURenderCallbackStruct input;
        input.inputProc = iOSCoreAudioCallback;
        input.inputProcRefCon = NULL;
        err = AudioUnitSetProperty(audioInstance,
                                   kAudioUnitProperty_SetRenderCallback,
                                   kAudioUnitScope_Input,
                                   0,
                                   &input,
                                   sizeof(input));
        if (err != noErr) {
            AudioComponentInstanceDispose(audioInstance);
            audioInstance = nil;
            return;
        }
        
        // setup the audio format we'll be using (stereo pcm)
        AudioStreamBasicDescription streamFormat;
        memset(&streamFormat, 0, sizeof(streamFormat));
        streamFormat.mSampleRate = SAMPLE_RATE;
        streamFormat.mFormatID = kAudioFormatLinearPCM;
        streamFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked;
        streamFormat.mBitsPerChannel = sizeof(AudioSampleType) * 8;
        streamFormat.mChannelsPerFrame = 2;
        streamFormat.mFramesPerPacket = 1;
        streamFormat.mBytesPerFrame = (streamFormat.mBitsPerChannel / 8) * streamFormat.mChannelsPerFrame;
        streamFormat.mBytesPerPacket = streamFormat.mBytesPerFrame * streamFormat.mFramesPerPacket;
        err = AudioUnitSetProperty(audioInstance,
                                   kAudioUnitProperty_StreamFormat,
                                   kAudioUnitScope_Input,
                                   0,
                                   &streamFormat,
                                   sizeof(AudioStreamBasicDescription));
        if (err != noErr) {
            AudioComponentInstanceDispose(audioInstance);
            audioInstance = nil;
            return;
        }
        
        // k, all setup, so init
        err = AudioUnitInitialize(audioInstance);
        if (err != noErr) {
            AudioComponentInstanceDispose(audioInstance);
            audioInstance = nil;
            return;
        }
        
        // finally start playback
        err = AudioOutputUnitStart(audioInstance);
        if (err != noErr) {
            AudioUnitUninitialize(audioInstance);
            AudioComponentInstanceDispose(audioInstance);
            audioInstance = nil;
            return;
        }
        
        // we're good to go
    }
}

//.........这里部分代码省略.........
      psize = sizeof(AudioDeviceID);
      /* for input, select device AFTER enabling IO */
      AudioUnitSetProperty(cdata->inunit,kAudioOutputUnitProperty_CurrentDevice,
                           kAudioUnitScope_Global, isInput, &dev, psize);
      aunit = &(cdata->inunit);
    }
    else {
      AudioComponentInstanceNew(HALOutput, &(cdata->outunit));
      psize = sizeof(AudioDeviceID);
      /* for output, select device BEFORE enabling IO */
      AudioUnitSetProperty(cdata->outunit, kAudioOutputUnitProperty_CurrentDevice,
                           kAudioUnitScope_Global, isInput, &dev, psize);
      enableIO = 1;
      AudioUnitSetProperty(cdata->outunit, kAudioOutputUnitProperty_EnableIO,
                           kAudioUnitScope_Output, 0, &enableIO, sizeof(enableIO));
      enableIO = 0;
      AudioUnitSetProperty(cdata->outunit, kAudioOutputUnitProperty_EnableIO,
                           kAudioUnitScope_Input, 1, &enableIO, sizeof(enableIO));
      aunit = &(cdata->outunit);
    }
    /* now set the buffer size */
    psize = sizeof(AudioDeviceID);
    AudioUnitGetProperty(*aunit, kAudioOutputUnitProperty_CurrentDevice,
                         kAudioUnitScope_Global, isInput, &dev, &psize);
    prop.mSelector = kAudioDevicePropertyBufferFrameSize;
    psize = 4;
    AudioObjectSetPropertyData(dev, &prop, 0, NULL, psize, &bufframes);
    psize = sizeof(maxFPS);
    AudioUnitGetProperty(*aunit, kAudioUnitProperty_MaximumFramesPerSlice,
                         kAudioUnitScope_Global, isInput, &maxFPS, &psize);
    AudioUnitSetProperty(*aunit, kAudioUnitProperty_MaximumFramesPerSlice,
                         kAudioUnitScope_Global, isInput, &bufframes,
                         sizeof(UInt32));
    /* set the stream properties */
    psize = sizeof(AudioStreamBasicDescription);
    AudioUnitGetProperty(*aunit, kAudioUnitProperty_StreamFormat,
                         (isInput ? kAudioUnitScope_Output : kAudioUnitScope_Input),
                         isInput, &format, &psize);
    format.mSampleRate    = srate;
    format.mFormatID = kAudioFormatLinearPCM;
    format.mFormatFlags = kAudioFormatFlagsCanonical |
      kLinearPCMFormatFlagIsNonInterleaved;
    format.mBytesPerPacket = sizeof(AudioUnitSampleType);
    format.mFramesPerPacket = 1;
    format.mBytesPerFrame = sizeof(AudioUnitSampleType);
    format.mChannelsPerFrame = nchnls;
    format.mBitsPerChannel = sizeof(AudioUnitSampleType)*8;
    AudioUnitSetProperty(*aunit, kAudioUnitProperty_StreamFormat,
                         (isInput ? kAudioUnitScope_Output : kAudioUnitScope_Input),
                         isInput, &format,
                         sizeof(AudioStreamBasicDescription));
    /* set the callbacks and open the device */
    if(!isInput) {
      AURenderCallbackStruct output;
      output.inputProc = Csound_Render;
      output.inputProcRefCon = cdata;
      AudioUnitSetProperty(*aunit, kAudioUnitProperty_SetRenderCallback,
                           kAudioUnitScope_Input, isInput, &output, sizeof(output));
      AudioUnitInitialize(*aunit);
      AudioOutputUnitStart(*aunit);

      csound->Message(csound,
                      Str("***** AuHAL module: output device open with %d "
                          "buffer frames\n"),
                          bufframes);
      cdata->disp = 0;
    }
    else {
      AURenderCallbackStruct input;
      AudioBufferList *CAInputData =
        (AudioBufferList*)malloc(sizeof(UInt32)
                                 + cdata->inchnls * sizeof(AudioBuffer));
      CAInputData->mNumberBuffers = cdata->inchnls;
      for (i = 0; i < cdata->inchnls; i++) {
        CAInputData->mBuffers[i].mNumberChannels = 1;
        CAInputData->mBuffers[i].mDataByteSize =
          bufframes * sizeof(AudioUnitSampleType);
        CAInputData->mBuffers[i].mData =
          calloc(bufframes, sizeof(AudioUnitSampleType));
      }
      cdata->inputdata = CAInputData;

      input.inputProc = Csound_Input;
      input.inputProcRefCon = cdata;
      AudioUnitSetProperty(*aunit, kAudioOutputUnitProperty_SetInputCallback,
                           kAudioUnitScope_Input, isInput, &input, sizeof(input));
      AudioUnitInitialize(*aunit);
      AudioOutputUnitStart(*aunit);
      csound->Message(csound,
              "***** AuHAL module: input device open with %d buffer frames\n",
                      bufframes);
    }
    if(!cdata->disp)
      csound->Message(csound,
              "==========================================================\n");

    cdata->disp = 0;
    return 0;

}

int CoreAudioDriver::init( unsigned bufferSize )
{
	OSStatus err = noErr;

	m_pOut_L = new float[ m_nBufferSize ];
	m_pOut_R = new float[ m_nBufferSize ];

	memset ( m_pOut_L, 0, m_nBufferSize * sizeof( float ) );
	memset ( m_pOut_R, 0, m_nBufferSize * sizeof( float ) );

	// Get Component
	AudioComponent compOutput;
	AudioComponentDescription descAUHAL;

	descAUHAL.componentType = kAudioUnitType_Output;
	descAUHAL.componentSubType = kAudioUnitSubType_HALOutput;
	descAUHAL.componentManufacturer = kAudioUnitManufacturer_Apple;
	descAUHAL.componentFlags = 0;
	descAUHAL.componentFlagsMask = 0;

	compOutput = AudioComponentFindNext( NULL, &descAUHAL );
	if ( compOutput == NULL ) {
		ERRORLOG( "Error in FindNextComponent" );
		//exit (-1);
	}

	err = AudioComponentInstanceNew( compOutput, &m_outputUnit );
	if ( err != noErr ) {
		ERRORLOG( "Error Opening Component" );
	}

	// Get Current Output Device
	retrieveDefaultDevice();

	// Set AUHAL to Current Device
	err = AudioUnitSetProperty(
			  m_outputUnit,
			  kAudioOutputUnitProperty_CurrentDevice,
			  kAudioUnitScope_Global,
			  0,
			  &m_outputDevice,
			  sizeof( m_outputDevice )
		  );
	if ( err != noErr ) {
		ERRORLOG( "Could not set Current Device" );
	}

	AudioStreamBasicDescription asbdesc;
	asbdesc.mSampleRate = ( Float64 )m_nSampleRate;
	asbdesc.mFormatID = kAudioFormatLinearPCM;
	asbdesc.mFormatFlags = kAudioFormatFlagsNativeFloatPacked | kAudioFormatFlagIsNonInterleaved;
	asbdesc.mBytesPerPacket = sizeof( Float32 );
	asbdesc.mFramesPerPacket = 1;
	asbdesc.mBytesPerFrame = sizeof( Float32 );
	asbdesc.mChannelsPerFrame = 2;	// comix: was set to 1
	asbdesc.mBitsPerChannel = 32;



	err = AudioUnitSetProperty(
				m_outputUnit,
				kAudioUnitProperty_StreamFormat,
				kAudioUnitScope_Input,
				0,
				&asbdesc,
				sizeof( AudioStreamBasicDescription )
			);

// Set Render Callback
	AURenderCallbackStruct out;
	out.inputProc = renderProc;
	out.inputProcRefCon = ( void * )this;

	err = AudioUnitSetProperty(
				m_outputUnit,
				kAudioUnitProperty_SetRenderCallback,
				kAudioUnitScope_Global,
				0,
				&out,
				sizeof( out )
			);
	if ( err != noErr ) {
		ERRORLOG( "Could not Set Render Callback" );
	}

	//Initialize AUHAL
	err = AudioUnitInitialize( m_outputUnit );
	if ( err != noErr ) {
		ERRORLOG( "Could not Initialize AudioUnit" );
	}

	return 0;
}

//.........这里部分代码省略.........
		if (audioComponent == NULL)
		{
			return;
		}
		
		error = AudioComponentInstanceNew(audioComponent, &_au);
		if (error != noErr)
		{
			return;
		}
	}
	else
	{
		ComponentDescription audioDesc;
		audioDesc.componentType = kAudioUnitType_Output;
		audioDesc.componentSubType = kAudioUnitSubType_DefaultOutput;
		audioDesc.componentManufacturer = kAudioUnitManufacturer_Apple;
		audioDesc.componentFlags = 0;
		audioDesc.componentFlagsMask = 0;
		
		Component audioComponent = FindNextComponent(NULL, &audioDesc);
		if (audioComponent == NULL)
		{
			return;
		}
		
		error = OpenAComponent(audioComponent, &_au);
		if (error != noErr)
		{
			return;
		}
	}
#else
	ComponentDescription audioDesc;
	audioDesc.componentType = kAudioUnitType_Output;
	audioDesc.componentSubType = kAudioUnitSubType_DefaultOutput;
	audioDesc.componentManufacturer = kAudioUnitManufacturer_Apple;
	audioDesc.componentFlags = 0;
	audioDesc.componentFlagsMask = 0;
	
	Component audioComponent = FindNextComponent(NULL, &audioDesc);
	if (audioComponent == NULL)
	{
		return;
	}
	
	error = OpenAComponent(audioComponent, &_au);
	if (error != noErr)
	{
		return;
	}
#endif
	
	// Set the render callback
	AURenderCallbackStruct callback;
	callback.inputProc = &CoreAudioOutputRenderCallback;
	callback.inputProcRefCon = _buffer;
	
	error = AudioUnitSetProperty(_au,
								 kAudioUnitProperty_SetRenderCallback,
								 kAudioUnitScope_Input,
								 0,
								 &callback,
								 sizeof(callback) );
	
	if(error != noErr)
	{
		return;
	}
	
	// Set up the audio unit for audio streaming
	AudioStreamBasicDescription outputFormat;
	outputFormat.mSampleRate = SPU_SAMPLE_RATE;
	outputFormat.mFormatID = kAudioFormatLinearPCM;
	outputFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kLinearPCMFormatFlagIsPacked;
	outputFormat.mBytesPerPacket = SPU_SAMPLE_SIZE;
	outputFormat.mFramesPerPacket = 1;
	outputFormat.mBytesPerFrame = SPU_SAMPLE_SIZE;
	outputFormat.mChannelsPerFrame = SPU_NUMBER_CHANNELS;
	outputFormat.mBitsPerChannel = SPU_SAMPLE_RESOLUTION;
	
	error = AudioUnitSetProperty(_au,
								 kAudioUnitProperty_StreamFormat,
								 kAudioUnitScope_Input,
								 0,
								 &outputFormat,
								 sizeof(outputFormat) );
	
	if(error != noErr)
	{
		return;
	}
	
	// Initialize our new audio unit
	error = AudioUnitInitialize(_au);
	if(error != noErr)
	{
		return;
	}
}

int
sa_stream_open(sa_stream_t *s) {

  ComponentDescription desc;
  Component comp;
  AURenderCallbackStruct input;
  AudioStreamBasicDescription fmt;

  if (s == NULL) {
    return SA_ERROR_NO_INIT;
  }
  if (s->output_unit != NULL) {
    return SA_ERROR_INVALID;
  }

  /*
   * Open the default audio output unit.
   */
  desc.componentType         = kAudioUnitType_Output;
  desc.componentSubType      = kAudioUnitSubType_DefaultOutput;
  desc.componentManufacturer = kAudioUnitManufacturer_Apple;
  desc.componentFlags        = 0;
  desc.componentFlagsMask    = 0;

  comp = FindNextComponent(NULL, &desc);
  if (comp == NULL) {
    return SA_ERROR_NO_DEVICE;
  }

  if (OpenAComponent(comp, &s->output_unit) != noErr) {
    return SA_ERROR_NO_DEVICE;
  }

  /*
   * Set up the render callback used to feed audio data into the output unit.
   */
  input.inputProc       = audio_callback;
  input.inputProcRefCon = s;
  if (AudioUnitSetProperty(s->output_unit, kAudioUnitProperty_SetRenderCallback,
      kAudioUnitScope_Input, 0, &input, sizeof(input)) != 0) {
    return SA_ERROR_SYSTEM;
  }

  /*
   * Set up the format description for our audio data. Apple uses the
   * following terminology:
   *
   * sample = a single data value for one channel
   * frame  = a set of samples that includes one sample for each channel
   * packet = the smallest indivisible block of audio data; for uncompressed
   *          audio (which is what we have), this is one frame
   * rate   = the number of complete frames per second
   *
   * Note that this definition of frame differs from, well, pretty much everyone
   * else's. See this really long link for more info:
   *
   * http://developer.apple.com/documentation/MusicAudio/Reference/CoreAudioDataTypesRef/Reference/reference.html#//apple_ref/c/tdef/AudioStreamBasicDescription
   */
  fmt.mFormatID         = kAudioFormatLinearPCM;
  fmt.mFormatFlags      = kLinearPCMFormatFlagIsSignedInteger |
#ifdef __BIG_ENDIAN__
                          kLinearPCMFormatFlagIsBigEndian |
#endif
                          kLinearPCMFormatFlagIsPacked;
  fmt.mSampleRate       = s->rate;
  fmt.mChannelsPerFrame = s->n_channels;
  fmt.mBitsPerChannel   = s->bytes_per_ch * 8;
  fmt.mFramesPerPacket  = 1;  /* uncompressed audio */
  fmt.mBytesPerFrame    = fmt.mChannelsPerFrame * fmt.mBitsPerChannel / 8;
  fmt.mBytesPerPacket   = fmt.mBytesPerFrame * fmt.mFramesPerPacket;

  /*
   * We're feeding data in to the output bus of the audio system, so we set
   * the format description on the input scope of the device, using the very
   * obvious element value of 0 to indicate the output bus.
   *
   * http://developer.apple.com/technotes/tn2002/tn2091.html
   */
  if (AudioUnitSetProperty(s->output_unit, kAudioUnitProperty_StreamFormat,
      kAudioUnitScope_Input, 0, &fmt, sizeof(AudioStreamBasicDescription)) != 0) {
    return SA_ERROR_NOT_SUPPORTED;
  }

  if (AudioUnitInitialize(s->output_unit) != 0) {
    return SA_ERROR_SYSTEM;
  }

  return SA_SUCCESS;
}

//.........这里部分代码省略.........
    ss.mBitsPerChannel = 16;
    ss.mFormatFlags |= kAudioFormatFlagIsSignedInteger;
    break;
  case CUBEB_SAMPLE_S16BE:
    ss.mBitsPerChannel = 16;
    ss.mFormatFlags |= kAudioFormatFlagIsSignedInteger |
                       kAudioFormatFlagIsBigEndian;
    break;
  case CUBEB_SAMPLE_FLOAT32LE:
    ss.mBitsPerChannel = 32;
    ss.mFormatFlags |= kAudioFormatFlagIsFloat;
    break;
  case CUBEB_SAMPLE_FLOAT32BE:
    ss.mBitsPerChannel = 32;
    ss.mFormatFlags |= kAudioFormatFlagIsFloat |
                       kAudioFormatFlagIsBigEndian;
    break;
  default:
    return CUBEB_ERROR_INVALID_FORMAT;
  }

  ss.mFormatID = kAudioFormatLinearPCM;
  ss.mFormatFlags |= kLinearPCMFormatFlagIsPacked;
  ss.mSampleRate = stream_params.rate;
  ss.mChannelsPerFrame = stream_params.channels;

  ss.mBytesPerFrame = (ss.mBitsPerChannel / 8) * ss.mChannelsPerFrame;
  ss.mFramesPerPacket = 1;
  ss.mBytesPerPacket = ss.mBytesPerFrame * ss.mFramesPerPacket;

  desc.componentType = kAudioUnitType_Output;
  desc.componentSubType = kAudioUnitSubType_DefaultOutput;
  desc.componentManufacturer = kAudioUnitManufacturer_Apple;
  desc.componentFlags = 0;
  desc.componentFlagsMask = 0;
#if MAC_OS_X_VERSION_MIN_REQUIRED < 1060
  comp = FindNextComponent(NULL, &desc);
#else
  comp = AudioComponentFindNext(NULL, &desc);
#endif
  assert(comp);

  stm = calloc(1, sizeof(*stm));
  assert(stm);

  stm->context = context;
  stm->data_callback = data_callback;
  stm->state_callback = state_callback;
  stm->user_ptr = user_ptr;

  stm->sample_spec = ss;

  r = pthread_mutex_init(&stm->mutex, NULL);
  assert(r == 0);

  stm->frames_played = 0;
  stm->frames_queued = 0;

#if MAC_OS_X_VERSION_MIN_REQUIRED < 1060
  r = OpenAComponent(comp, &stm->unit);
#else
  r = AudioComponentInstanceNew(comp, &stm->unit);
#endif
  if (r != 0) {
    audiounit_stream_destroy(stm);
    return CUBEB_ERROR;
  }

  input.inputProc = audiounit_output_callback;
  input.inputProcRefCon = stm;
  r = AudioUnitSetProperty(stm->unit, kAudioUnitProperty_SetRenderCallback,
                           kAudioUnitScope_Global, 0, &input, sizeof(input));
  if (r != 0) {
    audiounit_stream_destroy(stm);
    return CUBEB_ERROR;
  }

  r = AudioUnitSetProperty(stm->unit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input,
                           0, &ss, sizeof(ss));
  if (r != 0) {
    audiounit_stream_destroy(stm);
    return CUBEB_ERROR;
  }

  buffer_size = ss.mSampleRate / 1000.0 * latency * ss.mBytesPerFrame / NBUFS;
  if (buffer_size % ss.mBytesPerFrame != 0) {
    buffer_size += ss.mBytesPerFrame - (buffer_size % ss.mBytesPerFrame);
  }
  assert(buffer_size % ss.mBytesPerFrame == 0);

  r = AudioUnitInitialize(stm->unit);
  if (r != 0) {
    audiounit_stream_destroy(stm);
    return CUBEB_ERROR;
  }

  *stream = stm;

  return CUBEB_OK;
}

//.........这里部分代码省略.........
#endif

	AudioStreamBasicDescription strdesc;

	zeromem(&strdesc, sizeof(strdesc));
	UInt32 size = sizeof(strdesc);
	result = AudioUnitGetProperty(audio_unit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, kOutputBus, &strdesc, &size);
	ERR_FAIL_COND_V(result != noErr, FAILED);

	switch (strdesc.mChannelsPerFrame) {
		case 2: // Stereo
		case 4: // Surround 3.1
		case 6: // Surround 5.1
		case 8: // Surround 7.1
			channels = strdesc.mChannelsPerFrame;
			break;

		default:
			// Unknown number of channels, default to stereo
			channels = 2;
			break;
	}

	zeromem(&strdesc, sizeof(strdesc));
	size = sizeof(strdesc);
	result = AudioUnitGetProperty(audio_unit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, kInputBus, &strdesc, &size);
	ERR_FAIL_COND_V(result != noErr, FAILED);

	switch (strdesc.mChannelsPerFrame) {
		case 1: // Mono
			capture_channels = 1;
			break;

		case 2: // Stereo
			capture_channels = 2;
			break;

		default:
			// Unknown number of channels, default to stereo
			capture_channels = 2;
			break;
	}

	mix_rate = GLOBAL_DEF_RST("audio/mix_rate", DEFAULT_MIX_RATE);

	zeromem(&strdesc, sizeof(strdesc));
	strdesc.mFormatID = kAudioFormatLinearPCM;
	strdesc.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;
	strdesc.mChannelsPerFrame = channels;
	strdesc.mSampleRate = mix_rate;
	strdesc.mFramesPerPacket = 1;
	strdesc.mBitsPerChannel = 16;
	strdesc.mBytesPerFrame = strdesc.mBitsPerChannel * strdesc.mChannelsPerFrame / 8;
	strdesc.mBytesPerPacket = strdesc.mBytesPerFrame * strdesc.mFramesPerPacket;

	result = AudioUnitSetProperty(audio_unit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, kOutputBus, &strdesc, sizeof(strdesc));
	ERR_FAIL_COND_V(result != noErr, FAILED);

	strdesc.mChannelsPerFrame = capture_channels;

	result = AudioUnitSetProperty(audio_unit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, kInputBus, &strdesc, sizeof(strdesc));
	ERR_FAIL_COND_V(result != noErr, FAILED);

	int latency = GLOBAL_DEF_RST("audio/output_latency", DEFAULT_OUTPUT_LATENCY);
	// Sample rate is independent of channels (ref: https://stackoverflow.com/questions/11048825/audio-sample-frequency-rely-on-channels)
	buffer_frames = closest_power_of_2(latency * mix_rate / 1000);

#ifdef OSX_ENABLED
	result = AudioUnitSetProperty(audio_unit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, kOutputBus, &buffer_frames, sizeof(UInt32));
	ERR_FAIL_COND_V(result != noErr, FAILED);
#endif

	unsigned int buffer_size = buffer_frames * channels;
	samples_in.resize(buffer_size);
	input_buf.resize(buffer_size);
	input_buffer.resize(buffer_size * 8);
	input_position = 0;
	input_size = 0;

	print_verbose("CoreAudio: detected " + itos(channels) + " channels");
	print_verbose("CoreAudio: audio buffer frames: " + itos(buffer_frames) + " calculated latency: " + itos(buffer_frames * 1000 / mix_rate) + "ms");

	AURenderCallbackStruct callback;
	zeromem(&callback, sizeof(AURenderCallbackStruct));
	callback.inputProc = &AudioDriverCoreAudio::output_callback;
	callback.inputProcRefCon = this;
	result = AudioUnitSetProperty(audio_unit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, kOutputBus, &callback, sizeof(callback));
	ERR_FAIL_COND_V(result != noErr, FAILED);

	zeromem(&callback, sizeof(AURenderCallbackStruct));
	callback.inputProc = &AudioDriverCoreAudio::input_callback;
	callback.inputProcRefCon = this;
	result = AudioUnitSetProperty(audio_unit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &callback, sizeof(callback));
	ERR_FAIL_COND_V(result != noErr, FAILED);

	result = AudioUnitInitialize(audio_unit);
	ERR_FAIL_COND_V(result != noErr, FAILED);

	return OK;
}

void *runPluginLoop(void *plug) {
  AudioUnit outputUnit;
  OSStatus err = noErr;
  
	// Open the default output unit
	ComponentDescription desc;
	desc.componentType = kAudioUnitType_Output;
	desc.componentSubType = kAudioUnitSubType_DefaultOutput;
	desc.componentManufacturer = kAudioUnitManufacturer_Apple;
	desc.componentFlags = 0;
	desc.componentFlagsMask = 0;
	
	Component comp = FindNextComponent(NULL, &desc);
	if(comp == NULL) { 
    debug(LOG_ERROR, "FindNextComponent failed");
    return NULL;
  }
	
	err = OpenAComponent(comp, &outputUnit);
	if(comp == NULL) {
    debug(LOG_ERROR, "OpenAComponent failed with error code %ld\n", err);
    return NULL;
  }
  
	// Set up a callback function to generate output to the output unit
  AURenderCallbackStruct input;
	input.inputProc = processData;
	input.inputProcRefCon = plug;
  
	err = AudioUnitSetProperty(outputUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input,
                             0, &input, sizeof(input));
  
  AudioStreamBasicDescription streamFormat;
  streamFormat.mSampleRate = DEF_SAMPLE_RATE;
  streamFormat.mFormatID = kAudioFormatLinearPCM;
  streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger 
    | kLinearPCMFormatFlagIsBigEndian
    | kLinearPCMFormatFlagIsPacked
    | kAudioFormatFlagIsNonInterleaved;
  streamFormat.mBytesPerPacket = 2;
  streamFormat.mFramesPerPacket = 1;
  streamFormat.mBytesPerFrame = 2;
  streamFormat.mChannelsPerFrame = 2;	
  streamFormat.mBitsPerChannel = 16;	
  
  err = AudioUnitSetProperty(outputUnit, kAudioUnitProperty_StreamFormat,
                             kAudioUnitScope_Input, 0, &streamFormat,
                             sizeof(AudioStreamBasicDescription));
  if(err) {
    debug(LOG_ERROR, "AudioUnitSetProperty-SF failed with code %4.4s, %ld\n", (char*)&err, err);
    return NULL;
  }
  
  // Initialize unit
  err = AudioUnitInitialize(outputUnit);
  if(err) {
    debug(LOG_ERROR, "AudioUnitInitialize failed with code %ld\n", err);
    return NULL;
  }
  
  Float64 outSampleRate;
  UInt32 size = sizeof(Float64);
  err = AudioUnitGetProperty(outputUnit, kAudioUnitProperty_SampleRate,
                             kAudioUnitScope_Output, 0, &outSampleRate, &size);
  if(err) {
    debug(LOG_ERROR, "AudioUnitSetProperty-GF failed with code %4.4s, %ld\n", (char*)&err, err);
    return NULL;
  }
  
  // Start the rendering
  // The DefaultOutputUnit will do any format conversions to the format of the default device
  err = AudioOutputUnitStart(outputUnit);
  if(err) {
    debug(LOG_ERROR, "AudioOutputUnitStart failed with code %ld\n", err);
    return NULL;
  }
  
  // Loop until this thread is killed
  CFRunLoopRun();
  
  // REALLY after you're finished playing STOP THE AUDIO OUTPUT UNIT!!!!!!	
  // but we never get here because we're running until the process is nuked...	
  AudioOutputUnitStop(outputUnit);
  
  err = AudioUnitUninitialize(outputUnit);
  if(err) {
    debug(LOG_ERROR, "AudioUnitUninitialize failed with code %ld\n", err);
    return NULL;
  }
  
  return NULL;
}

void InputImplAudioUnit::setup() 
{
	if( mIsSetup ) return;
	
	OSStatus err = noErr;

	//get default input device
	if( ! mDevice ) {
		mDevice = InputImplAudioUnit::getDefaultDevice();
	}

	//create AudioOutputUnit
	AudioComponent component;
	AudioComponentDescription description;
	
	description.componentType = kAudioUnitType_Output;
#if defined( CINDER_MAC )
	description.componentSubType = kAudioUnitSubType_HALOutput;
#elif defined( CINDER_COCOA_TOUCH )
	description.componentSubType = kAudioUnitSubType_RemoteIO;
#endif
	description.componentManufacturer = kAudioUnitManufacturer_Apple;
	description.componentFlags = 0;
	description.componentFlagsMask = 0;
	component = AudioComponentFindNext( NULL, &description );
	if( ! component ) {
		std::cout << "Error finding next component" << std::endl;
		throw;
	}
	
	err = AudioComponentInstanceNew( component, &mInputUnit );
	if( err != noErr ) {
		mInputUnit = NULL;
		std::cout << "Error getting output unit" << std::endl;
		throw;
	}
	
	
	// Initialize the AU
	/*err = AudioUnitInitialize( mInputUnit );
	if(err != noErr)
	{
		std::cout << "failed to initialize HAL Output AU" << std::endl;
		throw;
	}*/
	
	UInt32 param;
	//enable IO on AudioUnit's input scope
	param = 1;
	err = AudioUnitSetProperty( mInputUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &param, sizeof( UInt32 ) );
	if( err != noErr ) {
		std::cout << "Error enable IO on Output unit input" << std::endl;
		throw;
	}
	
	//disable IO on AudioUnit's output scope
	param = 0;
	err = AudioUnitSetProperty( mInputUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &param, sizeof( UInt32 ) );
	if( err != noErr ) {
		std::cout << "Error disabling IO on Output unit output" << std::endl;
		throw;
	}
	
#if defined( CINDER_MAC )
	AudioDeviceID nativeDeviceId = static_cast<AudioDeviceID>( mDevice->getDeviceId() );
	
	// Set the current device to the default input unit.
	err = AudioUnitSetProperty( mInputUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &nativeDeviceId, sizeof(AudioDeviceID) );
	if( err != noErr ) {
		std::cout << "failed to set AU input device" << std::endl;
		throw;
	}
#endif
	
	AURenderCallbackStruct callback;
	callback.inputProc = InputImplAudioUnit::inputCallback;
	callback.inputProcRefCon = this;
	err = AudioUnitSetProperty( mInputUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &callback, sizeof(AURenderCallbackStruct) );
	
	//Don't setup buffers until you know what the 
	//input and output device audio streams look like.
	
	// Initialize the AudioUnit
	err = AudioUnitInitialize( mInputUnit );
	if(err != noErr) {
		std::cout << "failed to initialize HAL Output AU" << std::endl;
		throw;
	}
	
	//Get Size of IO Buffers
	uint32_t sampleCount;
	param = sizeof(UInt32);
#if defined( CINDER_MAC )
	err = AudioUnitGetProperty( mInputUnit, kAudioDevicePropertyBufferFrameSize, kAudioUnitScope_Global, 0, &sampleCount, &param );
	if( err != noErr ) {
		std::cout << "Error getting buffer frame size" << std::endl;
		throw;
	}
#elif defined( CINDER_COCOA_TOUCH )
	AudioUnitGetProperty( mInputUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Global, 0, &sampleCount, &param );
//.........这里部分代码省略.........

//.........这里部分代码省略.........
    size = sizeof(AudioDeviceID);

    AudioDeviceID inputDevice;
    err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice,
                                                  &size, 
                                                  &inputDevice);

    if (err)
        exit(err);

    err =AudioUnitSetProperty(auHAL,
                         kAudioOutputUnitProperty_CurrentDevice, 
                         kAudioUnitScope_Global, 
                         0, 
                         &inputDevice, 
                         sizeof(inputDevice));
    if (err)
        exit(err);

    AudioStreamBasicDescription DeviceFormat;
    AudioStreamBasicDescription DesiredFormat;
   //Use CAStreamBasicDescriptions instead of 'naked' 
   //AudioStreamBasicDescriptions to minimize errors.
   //CAStreamBasicDescription.h can be found in the CoreAudio SDK.

    size = sizeof(AudioStreamBasicDescription);

     //Get the input device format
    AudioUnitGetProperty (auHAL,
                                   kAudioUnitProperty_StreamFormat,
                                   kAudioUnitScope_Input,
                                   1,
                                   &DeviceFormat,
                                   &size);

    //set the desired format to the device's sample rate
    memcpy(&DesiredFormat, &DeviceFormat, sizeof(AudioStreamBasicDescription));

    sampling_rate = DeviceFormat.mSampleRate;  // for laser-emulating filters
    DesiredFormat.mSampleRate =  DeviceFormat.mSampleRate;
    DesiredFormat.mChannelsPerFrame = 4;
    DesiredFormat.mBitsPerChannel = 16;
    DesiredFormat.mBytesPerPacket = DesiredFormat.mBytesPerFrame =
        DesiredFormat.mChannelsPerFrame * 2;
    DesiredFormat.mFramesPerPacket = 1;
    DesiredFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;

     //set format to output scope
    err = AudioUnitSetProperty(
                            auHAL,
                            kAudioUnitProperty_StreamFormat,
                            kAudioUnitScope_Output,
                            1,
                            &DesiredFormat,
                            sizeof(AudioStreamBasicDescription));
    if (err)
        exit(err);

    SInt32 *channelMap =NULL;
    UInt32 numOfChannels = DesiredFormat.mChannelsPerFrame; //2 channels
    UInt32 mapSize = numOfChannels *sizeof(SInt32);
    channelMap = (SInt32 *)malloc(mapSize);

    //for each channel of desired input, map the channel from
    //the device's output channel. 
    for(i=0;i<numOfChannels;i++)
    {
        channelMap[i]=i;
    }
    err = AudioUnitSetProperty(auHAL,
                                        kAudioOutputUnitProperty_ChannelMap,
                                        kAudioUnitScope_Output,
                                        1,
                                        channelMap,
                                        size);
    if (err)
        exit(err);
    free(channelMap);

    AURenderCallbackStruct input;
    input.inputProc = callback;
    input.inputProcRefCon = 0;

    err = AudioUnitSetProperty(
            auHAL, 
            kAudioOutputUnitProperty_SetInputCallback, 
            kAudioUnitScope_Global,
            0,
            &input, 
            sizeof(input));
    if (err)
        exit(err);

     err = AudioUnitInitialize(auHAL);
    if (err)
        exit(err);
     err = AudioOutputUnitStart(auHAL);
    if (err)
        exit(err);
}

//.........这里部分代码省略.........
            latency < 1 || latency > 2000) {
        return CUBEB_ERROR_INVALID_FORMAT;
    }

    memset(&ss, 0, sizeof(ss));
    ss.mFormatFlags = 0;

    switch (stream_params.format) {
    case CUBEB_SAMPLE_S16LE:
        ss.mBitsPerChannel = 16;
        ss.mFormatFlags |= kAudioFormatFlagIsSignedInteger;
        break;
    case CUBEB_SAMPLE_S16BE:
        ss.mBitsPerChannel = 16;
        ss.mFormatFlags |= kAudioFormatFlagIsSignedInteger |
                           kAudioFormatFlagIsBigEndian;
        break;
    case CUBEB_SAMPLE_FLOAT32LE:
        ss.mBitsPerChannel = 32;
        ss.mFormatFlags |= kAudioFormatFlagIsFloat;
        break;
    case CUBEB_SAMPLE_FLOAT32BE:
        ss.mBitsPerChannel = 32;
        ss.mFormatFlags |= kAudioFormatFlagIsFloat |
                           kAudioFormatFlagIsBigEndian;
        break;
    default:
        return CUBEB_ERROR_INVALID_FORMAT;
    }

    ss.mFormatID = kAudioFormatLinearPCM;
    ss.mFormatFlags |= kLinearPCMFormatFlagIsPacked;
    ss.mSampleRate = stream_params.rate;
    ss.mChannelsPerFrame = stream_params.channels;

    ss.mBytesPerFrame = (ss.mBitsPerChannel / 8) * ss.mChannelsPerFrame;
    ss.mFramesPerPacket = 1;
    ss.mBytesPerPacket = ss.mBytesPerFrame * ss.mFramesPerPacket;

    desc.componentType = kAudioUnitType_Output;
    desc.componentSubType = kAudioUnitSubType_DefaultOutput;
    desc.componentManufacturer = kAudioUnitManufacturer_Apple;
    desc.componentFlags = 0;
    desc.componentFlagsMask = 0;
    comp = FindNextComponent(NULL, &desc);
    assert(comp);

    stm = calloc(1, sizeof(*stm));
    assert(stm);

    stm->data_callback = data_callback;
    stm->state_callback = state_callback;
    stm->user_ptr = user_ptr;

    stm->sample_spec = ss;

    r = pthread_mutex_init(&stm->mutex, NULL);
    assert(r == 0);

    stm->frames_played = 0;
    stm->frames_queued = 0;

    r = OpenAComponent(comp, &stm->unit);
    if (r != 0) {
        fprintf(stderr, "cubeb_audiounit: FATAL: OpenAComponent returned %ld\n", (long) r);
    }
    assert(r == 0);

    input.inputProc = audio_unit_output_callback;
    input.inputProcRefCon = stm;
    r = AudioUnitSetProperty(stm->unit, kAudioUnitProperty_SetRenderCallback,
                             kAudioUnitScope_Global, 0, &input, sizeof(input));
    if (r != 0) {
        fprintf(stderr, "cubeb_audiounit: FATAL: AudioUnitSetProperty(SetRenderCallback) returned %ld\n", (long) r);
    }
    assert(r == 0);

    r = AudioUnitSetProperty(stm->unit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input,
                             0, &ss, sizeof(ss));
    if (r != 0) {
        fprintf(stderr, "cubeb_audiounit: FATAL: AudioUnitSetProperty(StreamFormat) returned %ld\n", (long) r);
    }
    assert(r == 0);

    buffer_size = ss.mSampleRate / 1000.0 * latency * ss.mBytesPerFrame / NBUFS;
    if (buffer_size % ss.mBytesPerFrame != 0) {
        buffer_size += ss.mBytesPerFrame - (buffer_size % ss.mBytesPerFrame);
    }
    assert(buffer_size % ss.mBytesPerFrame == 0);

    r = AudioUnitInitialize(stm->unit);
    if (r != 0) {
        fprintf(stderr, "cubeb_audiounit: FATAL: AudioUnitInitialize returned %ld\n", (long) r);
    }
    assert(r == 0);

    *stream = stm;

    return CUBEB_OK;
}

//.........这里部分代码省略.........
#endif
  assert(comp);

  stm = calloc(1, sizeof(*stm));
  assert(stm);

  stm->context = context;
  stm->data_callback = data_callback;
  stm->state_callback = state_callback;
  stm->user_ptr = user_ptr;

  stm->sample_spec = ss;

  r = pthread_mutex_init(&stm->mutex, NULL);
  assert(r == 0);

  stm->frames_played = 0;
  stm->frames_queued = 0;
  stm->current_latency_frames = 0;
  stm->hw_latency_frames = UINT64_MAX;

#if MAC_OS_X_VERSION_MIN_REQUIRED < 1060
  r = OpenAComponent(comp, &stm->unit);
#else
  r = AudioComponentInstanceNew(comp, &stm->unit);
#endif
  if (r != 0) {
    audiounit_stream_destroy(stm);
    return CUBEB_ERROR;
  }

  input.inputProc = audiounit_output_callback;
  input.inputProcRefCon = stm;
  r = AudioUnitSetProperty(stm->unit, kAudioUnitProperty_SetRenderCallback,
                           kAudioUnitScope_Global, 0, &input, sizeof(input));
  if (r != 0) {
    audiounit_stream_destroy(stm);
    return CUBEB_ERROR;
  }

  buffer_size = latency / 1000.0 * ss.mSampleRate;

  /* Get the range of latency this particular device can work with, and clamp
   * the requested latency to this acceptable range. */
  if (audiounit_get_acceptable_latency_range(&latency_range) != CUBEB_OK) {
    audiounit_stream_destroy(stm);
    return CUBEB_ERROR;
  }

  if (buffer_size < (unsigned int) latency_range.mMinimum) {
    buffer_size = (unsigned int) latency_range.mMinimum;
  } else if (buffer_size > (unsigned int) latency_range.mMaximum) {
    buffer_size = (unsigned int) latency_range.mMaximum;
  }

  /**
   * Get the default buffer size. If our latency request is below the default,
   * set it. Otherwise, use the default latency.
   **/
  size = sizeof(default_buffer_size);
  r = AudioUnitGetProperty(stm->unit, kAudioDevicePropertyBufferFrameSize,
                           kAudioUnitScope_Output, 0, &default_buffer_size, &size);

  if (r != 0) {
    audiounit_stream_destroy(stm);
    return CUBEB_ERROR;
  }

  // Setting the latency doesn't work well for USB headsets (eg. plantronics).
  // Keep the default latency for now.
#if 0
  if (buffer_size < default_buffer_size) {
    /* Set the maximum number of frame that the render callback will ask for,
     * effectively setting the latency of the stream. This is process-wide. */
    r = AudioUnitSetProperty(stm->unit, kAudioDevicePropertyBufferFrameSize,
                             kAudioUnitScope_Output, 0, &buffer_size, sizeof(buffer_size));
    if (r != 0) {
      audiounit_stream_destroy(stm);
      return CUBEB_ERROR;
    }
  }
#endif

  r = AudioUnitSetProperty(stm->unit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input,
                           0, &ss, sizeof(ss));
  if (r != 0) {
    audiounit_stream_destroy(stm);
    return CUBEB_ERROR;
  }

  r = AudioUnitInitialize(stm->unit);
  if (r != 0) {
    audiounit_stream_destroy(stm);
    return CUBEB_ERROR;
  }

  *stream = stm;

  return CUBEB_OK;
}