text:
/*
 * The text above constitutes the entire PortAudio license; however,
 * the PortAudio community also makes the following non-binding requests:
 *
 * Any person wishing to distribute modifications to the Software is
 * requested to send the modifications to the original developer so that
 * they can be incorporated into the canonical version. It is also
 * requested that these non-binding requests be included along with the
 * license above.
 */

/**
 @file pa_mac_core
 @ingroup hostapi_src
 @author Bjorn Roche
 @brief AUHAL implementation of PortAudio
*/

/* FIXME: not all error conditions call PaUtil_SetLastHostErrorInfo()
 * PaMacCore_SetError() will do this.
 */

#include "pa_mac_core_internal.h"

#include <string.h> /* strlen(), memcmp() etc. */
#include <libkern/OSAtomic.h>

#include "pa_mac_core.h"
#include "pa_mac_core_utilities.h"
#include "pa_mac_core_blocking.h"

#ifdef __cplusplus
extern "C"
{
#endif /* __cplusplus */

/* prototypes for functions declared in this file */

PaError PaMacCore_Initialize( PaUtilHostApiRepresentation **hostApi, PaHostApiIndex index );

/*
 * Function declared in pa_mac_core.h. Sets up a PaMacCoreStreamInfoStruct
 * with the requested flags and initializes channel map.
 */
void PaMacCore_SetupStreamInfo(  PaMacCoreStreamInfo *data, const unsigned long flags )
{
   bzero( data, sizeof( PaMacCoreStreamInfo ) );
   data->size = sizeof( PaMacCoreStreamInfo );
   data->hostApiType = paCoreAudio;
   data->version = 0x01;
   data->flags = flags;
   data->channelMap = NULL;
   data->channelMapSize = 0;
}

/*
 * Function declared in pa_mac_core.h. Adds channel mapping to a PaMacCoreStreamInfoStruct
 */
void PaMacCore_SetupChannelMap( PaMacCoreStreamInfo *data, const SInt32 * const channelMap, const unsigned long channelMapSize )
{
   data->channelMap = channelMap;
   data->channelMapSize = channelMapSize;
}
static char *channelName = NULL;
static int channelNameSize = 0;
static bool ensureChannelNameSize( int size )
{
   if( size >= channelNameSize ) {
      free( channelName );
      channelName = (char *) malloc( ( channelNameSize = size ) + 1 );
      if( !channelName ) {
         channelNameSize = 0;
         return false;
      }
   }
   return true;
}
/*
 * Function declared in pa_mac_core.h. retrives channel names.
 */
const char *PaMacCore_GetChannelName( int device, int channelIndex, bool input )
{
   struct PaUtilHostApiRepresentation *hostApi;
   PaError err;
   OSStatus error;
   err = PaUtil_GetHostApiRepresentation( &hostApi, paCoreAudio );
   assert(err == paNoError);
   PaMacAUHAL *macCoreHostApi = (PaMacAUHAL*)hostApi;
   AudioDeviceID hostApiDevice = macCoreHostApi->devIds[device];

   UInt32 size = 0;

   error = AudioDeviceGetPropertyInfo( hostApiDevice,
                                       channelIndex + 1,
                                       input,
                                       kAudioDevicePropertyChannelName,
                                       &size,
                                       NULL );
   if( error ) {
      //try the CFString
      CFStringRef name;
      bool isDeviceName = false;
      size = sizeof( name );
      error = AudioDeviceGetProperty( hostApiDevice,
                                      channelIndex + 1,
                                      input,
                                      kAudioDevicePropertyChannelNameCFString,
                                      &size,
                                      &name );
      if( error ) { //as a last-ditch effort, get the device name. Later we'll append the channel number.
         size = sizeof( name );
         error = AudioDeviceGetProperty( hostApiDevice,
                                      channelIndex + 1,
                                      input,
                                      kAudioDevicePropertyDeviceNameCFString,
                                      &size,
                                      &name );
         if( error )
            return NULL;
         isDeviceName = true;
      }
      if( isDeviceName ) {
         name = CFStringCreateWithFormat( NULL, NULL, CFSTR( "%@: %d"), name, channelIndex + 1 );
      }

      CFIndex length = CFStringGetLength(name);
      while( ensureChannelNameSize( length * sizeof(UniChar) + 1 ) ) {
         if( CFStringGetCString( name, channelName, channelNameSize, kCFStringEncodingUTF8 ) ) {
            if( isDeviceName )
               CFRelease( name );
            return channelName;
         }
         if( length == 0 )
            ++length;
         length *= 2;
      }
      if( isDeviceName )
         CFRelease( name );
      return NULL;
   }

   //continue with C string:
   if( !ensureChannelNameSize( size ) )
      return NULL;

   error = AudioDeviceGetProperty( hostApiDevice,
                                   channelIndex + 1,
                                   input,
                                   kAudioDevicePropertyChannelName,
                                   &size,
                                   channelName );

   if( error ) {
      ERR( error );
      return NULL;
   }
   return channelName;
}

AudioDeviceID PaMacCore_GetStreamInputDevice( PaStream* s )
{
    PaMacCoreStream *stream = (PaMacCoreStream*)s;
    VVDBUG(("PaMacCore_GetStreamInputHandle()\n"));

    return ( stream->inputDevice );
}

AudioDeviceID PaMacCore_GetStreamOutputDevice( PaStream* s )
{
    PaMacCoreStream *stream = (PaMacCoreStream*)s;
    VVDBUG(("PaMacCore_GetStreamOutputHandle()\n"));

    return ( stream->outputDevice );
}

#ifdef __cplusplus
}
#endif /* __cplusplus */

#define RING_BUFFER_ADVANCE_DENOMINATOR (4)

static void Terminate( struct PaUtilHostApiRepresentation *hostApi );
static PaError IsFormatSupported( struct PaUtilHostApiRepresentation *hostApi,
                                  const PaStreamParameters *inputParameters,
                                  const PaStreamParameters *outputParameters,
                                  double sampleRate );
static PaError OpenStream( struct PaUtilHostApiRepresentation *hostApi,
                           PaStream** s,
                           const PaStreamParameters *inputParameters,
                           const PaStreamParameters *outputParameters,
                           double sampleRate,
                           unsigned long framesPerBuffer,
                           PaStreamFlags streamFlags,
                           PaStreamCallback *streamCallback,
                           void *userData );
static PaError CloseStream( PaStream* stream );
static PaError StartStream( PaStream *stream );
static PaError StopStream( PaStream *stream );
static PaError AbortStream( PaStream *stream );
static PaError IsStreamStopped( PaStream *s );
static PaError IsStreamActive( PaStream *stream );
static PaTime GetStreamTime( PaStream *stream );
static void setStreamStartTime( PaStream *stream );
static OSStatus AudioIOProc( void *inRefCon,
                               AudioUnitRenderActionFlags *ioActionFlags,
                               const AudioTimeStamp *inTimeStamp,
                               UInt32 inBusNumber,
                               UInt32 inNumberFrames,
                               AudioBufferList *ioData );
static double GetStreamCpuLoad( PaStream* stream );

static PaError GetChannelInfo( PaMacAUHAL *auhalHostApi,
                               PaDeviceInfo *deviceInfo,
                               AudioDeviceID macCoreDeviceId,
                               int isInput);

static PaError OpenAndSetupOneAudioUnit(
                                   const PaMacCoreStream *stream,
                                   const PaStreamParameters *inStreamParams,
                                   const PaStreamParameters *outStreamParams,
                                   const UInt32 requestedFramesPerBuffer,
                                   UInt32 *actualInputFramesPerBuffer,
                                   UInt32 *actualOutputFramesPerBuffer,
                                   const PaMacAUHAL *auhalHostApi,
                                   AudioUnit *audioUnit,
                                   AudioConverterRef *srConverter,
                                   AudioDeviceID *audioDevice,
                                   const double sampleRate,
                                   void *refCon );

/* for setting errors. */
#define PA_AUHAL_SET_LAST_HOST_ERROR( errorCode, errorText ) \
    PaUtil_SetLastHostErrorInfo( paInDevelopment, errorCode, errorText )

/*
 * Callback for setting over/underrun flags.
 *
 */
static OSStatus xrunCallback(
    AudioDeviceID inDevice,
    UInt32 inChannel,
    Boolean isInput,
    AudioDevicePropertyID inPropertyID,
    void* inClientData)
{
   PaMacCoreStream *stream = (PaMacCoreStream *) inClientData;
   if( stream->state != ACTIVE )
      return 0; //if the stream isn't active, we don't care if the device is dropping
   if( isInput )
      OSAtomicOr32( paInputUnderflow, (uint32_t *)&(stream->xrunFlags) );
   else
      OSAtomicOr32( paOutputOverflow, (uint32_t *)&(stream->xrunFlags) );

   return 0;
}

/*
 * Callback called when starting or stopping a stream.
 */
static void startStopCallback(
   void *               inRefCon,
   AudioUnit            ci,
   AudioUnitPropertyID  inID,
   AudioUnitScope       inScope,
   AudioUnitElement     inElement )
{
   PaMacCoreStream *stream = (PaMacCoreStream *) inRefCon;
   UInt32 isRunning;
   UInt32 size = sizeof( isRunning );
   assert( !AudioUnitGetProperty( ci, kAudioOutputUnitProperty_IsRunning, inScope, inElement, &isRunning, &size ) );
   if( isRunning )
      return; //We are only interested in when we are stopping
   // -- if we are using 2 I/O units, we only need one notification!
   if( stream->inputUnit && stream->outputUnit && stream->inputUnit != stream->outputUnit && ci == stream->inputUnit )
      return;
   PaStreamFinishedCallback *sfc = stream->streamRepresentation.streamFinishedCallback;
   if( stream->state == STOPPING )
      stream->state = STOPPED ;
   if( sfc )
      sfc( stream->streamRepresentation.userData );
}

/*currently, this is only used in initialization, but it might be modified
  to be used when the list of devices changes.*/
static PaError gatherDeviceInfo(PaMacAUHAL *auhalHostApi)
{
    UInt32 size;
    UInt32 propsize;
    VVDBUG(("gatherDeviceInfo()\n"));
    /* -- free any previous allocations -- */
    if( auhalHostApi->devIds )
        PaUtil_GroupFreeMemory(auhalHostApi->allocations, auhalHostApi->devIds);
    auhalHostApi->devIds = NULL;

    /* -- figure out how many devices there are -- */
    AudioHardwareGetPropertyInfo( kAudioHardwarePropertyDevices,
                                  &propsize,
                                  NULL );
    auhalHostApi->devCount = propsize / sizeof( AudioDeviceID );

    VDBUG( ( "Found %ld device(s).\n", auhalHostApi->devCount ) );

    /* -- copy the device IDs -- */
    auhalHostApi->devIds = (AudioDeviceID *)PaUtil_GroupAllocateMemory(
                             auhalHostApi->allocations,
                             propsize );
    if( !auhalHostApi->devIds )
        return paInsufficientMemory;
    AudioHardwareGetProperty( kAudioHardwarePropertyDevices,
                                  &propsize,
                                  auhalHostApi->devIds );
#ifdef MAC_CORE_VERBOSE_DEBUG
    {
       int i;
       for( i=0; i<auhalHostApi->devCount; ++i )
          printf( "Device %d\t: %ld\n", i, auhalHostApi->devIds[i] );
    }
#endif

    size = sizeof(AudioDeviceID);
    auhalHostApi->defaultIn  = kAudioDeviceUnknown;
    auhalHostApi->defaultOut = kAudioDeviceUnknown;

    /* determine the default device. */
    /* I am not sure how these calls to AudioHardwareGetProperty()
       could fail, but in case they do, we use the first available
       device as the default. */
    if( 0 != AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice,
                     &size,
                     &auhalHostApi->defaultIn) ) {
       int i;
       auhalHostApi->defaultIn  = kAudioDeviceUnknown;
       VDBUG(("Failed to get default input device from OS."));
       VDBUG((" I will substitute the first available input Device."));
       for( i=0; i<auhalHostApi->devCount; ++i ) {
          PaDeviceInfo devInfo;
          if( 0 != GetChannelInfo( auhalHostApi, &devInfo,
                                   auhalHostApi->devIds[i], TRUE ) )
             if( devInfo.maxInputChannels ) {
                auhalHostApi->defaultIn = auhalHostApi->devIds[i];
                break;
             }
       }
    }
    if( 0 != AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
                     &size,
                     &auhalHostApi->defaultOut) ) {
       int i;
       auhalHostApi->defaultIn  = kAudioDeviceUnknown;
       VDBUG(("Failed to get default output device from OS."));
       VDBUG((" I will substitute the first available output Device."));
       for( i=0; i<auhalHostApi->devCount; ++i ) {
          PaDeviceInfo devInfo;
          if( 0 != GetChannelInfo( auhalHostApi, &devInfo,
                                   auhalHostApi->devIds[i], FALSE ) )
             if( devInfo.maxOutputChannels ) {
                auhalHostApi->defaultOut = auhalHostApi->devIds[i];
                break;
             }
       }
    }

    VDBUG( ( "Default in : %ld\n", auhalHostApi->defaultIn  ) );
    VDBUG( ( "Default out: %ld\n", auhalHostApi->defaultOut ) );

    return paNoError;
}

static PaError GetChannelInfo( PaMacAUHAL *auhalHostApi,
                               PaDeviceInfo *deviceInfo,
                               AudioDeviceID macCoreDeviceId,
                               int isInput)
{
    UInt32 propSize;
    PaError err = paNoError;
    UInt32 i;
    int numChannels = 0;
    AudioBufferList *buflist = NULL;
    UInt32 frameLatency;

    VVDBUG(("GetChannelInfo()\n"));

    /* Get the number of channels from the stream configuration.
       Fail if we can't get this. */

    err = ERR(AudioDeviceGetPropertyInfo(macCoreDeviceId, 0, isInput, kAudioDevicePropertyStreamConfiguration, &propSize, NULL));
    if (err)
        return err;

    buflist = PaUtil_AllocateMemory(propSize);
    if( !buflist )
       return paInsufficientMemory;
    err = ERR(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyStreamConfiguration, &propSize, buflist));
    if (err)
        goto error;

    for (i = 0; i < buflist->mNumberBuffers; ++i)
        numChannels += buflist->mBuffers[i].mNumberChannels;

    if (isInput)
        deviceInfo->maxInputChannels = numChannels;
    else
        deviceInfo->maxOutputChannels = numChannels;

    if (numChannels > 0) /* do not try to retrieve the latency if there is no channels. */
    {
       /* Get the latency.  Don't fail if we can't get this. */
       /* default to something reasonable */
       deviceInfo->defaultLowInputLatency = .01;
       deviceInfo->defaultHighInputLatency = .10;
       deviceInfo->defaultLowOutputLatency = .01;
       deviceInfo->defaultHighOutputLatency = .10;
       propSize = sizeof(UInt32);
       err = WARNING(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyLatency, &propSize, &frameLatency));
       if (!err)
       {
          /** FEEDBACK:
           * This code was arrived at by trial and error, and some extentive, but not exhaustive
           * testing. Sebastien Beaulieu <seb@plogue.com> has suggested using
           * kAudioDevicePropertyLatency + kAudioDevicePropertySafetyOffset + buffer size instead.
           * At the time this code was written, many users were reporting dropouts with audio
           * programs that probably used this formula. This was probably
           * around 10.4.4, and the problem is probably fixed now. So perhaps
           * his formula should be reviewed and used.
           * */
          double secondLatency = frameLatency / deviceInfo->defaultSampleRate;
          if (isInput)
          {
             deviceInfo->defaultLowInputLatency = 3 * secondLatency;
             deviceInfo->defaultHighInputLatency = 3 * 10 * secondLatency;
          }
          else
          {
             deviceInfo->defaultLowOutputLatency = 3 * secondLatency;
             deviceInfo->defaultHighOutputLatency = 3 * 10 * secondLatency;
          }
       }
    }
    PaUtil_FreeMemory( buflist );
    return paNoError;
 error:
    PaUtil_FreeMemory( buflist );
    return err;
}

static PaError InitializeDeviceInfo( PaMacAUHAL *auhalHostApi,
                                     PaDeviceInfo *deviceInfo,
                                     AudioDeviceID macCoreDeviceId,
                                     PaHostApiIndex hostApiIndex )
{
    Float64 sampleRate;
    char *name;
    PaError err = paNoError;
    UInt32 propSize;

    VVDBUG(("InitializeDeviceInfo(): macCoreDeviceId=%ld\n", macCoreDeviceId));

    memset(deviceInfo, 0, sizeof(deviceInfo));

    deviceInfo->structVersion = 2;
    deviceInfo->hostApi = hostApiIndex;

    /* Get the device name.  Fail if we can't get it. */
    err = ERR(AudioDeviceGetPropertyInfo(macCoreDeviceId, 0, 0, kAudioDevicePropertyDeviceName, &propSize, NULL));
    if (err)
        return err;

    name = PaUtil_GroupAllocateMemory(auhalHostApi->allocations,propSize);
    if ( !name )
        return paInsufficientMemory;
    err = ERR(AudioDeviceGetProperty(macCoreDeviceId, 0, 0, kAudioDevicePropertyDeviceName, &propSize, name));
    if (err)
        return err;
    deviceInfo->name = name;

    /* Try to get the default sample rate.  Don't fail if we can't get this. */
    propSize = sizeof(Float64);
    err = ERR(AudioDeviceGetProperty(macCoreDeviceId, 0, 0, kAudioDevicePropertyNominalSampleRate, &propSize, &sampleRate));
    if (err)
        deviceInfo->defaultSampleRate = 0.0;
    else
        deviceInfo->defaultSampleRate = sampleRate;

    /* Get the maximum number of input and output channels.  Fail if we can't get this. */

    err = GetChannelInfo(auhalHostApi, deviceInfo, macCoreDeviceId, 1);
    if (err)
        return err;

    err = GetChannelInfo(auhalHostApi, deviceInfo, macCoreDeviceId, 0);
    if (err)
        return err;

    return paNoError;
}

PaError PaMacCore_Initialize( PaUtilHostApiRepresentation **hostApi, PaHostApiIndex hostApiIndex )
{
    PaError result = paNoError;
    int i;
    PaMacAUHAL *auhalHostApi;
    PaDeviceInfo *deviceInfoArray;

    VVDBUG(("PaMacCore_Initialize(): hostApiIndex=%d\n", hostApiIndex));

    auhalHostApi = (PaMacAUHAL*)PaUtil_AllocateMemory( sizeof(PaMacAUHAL) );
    if( !auhalHostApi )
    {
        result = paInsufficientMemory;
        goto error;
    }

    auhalHostApi->allocations = PaUtil_CreateAllocationGroup();
    if( !auhalHostApi->allocations )
    {
        result = paInsufficientMemory;
        goto error;
    }

    auhalHostApi->devIds = NULL;
    auhalHostApi->devCount = 0;

    /* get the info we need about the devices */
    result = gatherDeviceInfo( auhalHostApi );
    if( result != paNoError )
       goto error;

    *hostApi = &auhalHostApi->inheritedHostApiRep;
    (*hostApi)->info.structVersion = 1;
    (*hostApi)->info.type = paCoreAudio;
    (*hostApi)->info.name = "Core Audio";

    (*hostApi)->info.defaultInputDevice = paNoDevice;
    (*hostApi)->info.defaultOutputDevice = paNoDevice;

    (*hostApi)->info.deviceCount = 0;

    if( auhalHostApi->devCount > 0 )
    {
        (*hostApi)->deviceInfos = (PaDeviceInfo**)PaUtil_GroupAllocateMemory(
                auhalHostApi->allocations, sizeof(PaDeviceInfo*) * auhalHostApi->devCount);
        if( !(*hostApi)->deviceInfos )
        {
            result = paInsufficientMemory;
            goto error;
        }

        /* allocate all device info structs in a contiguous block */
        deviceInfoArray = (PaDeviceInfo*)PaUtil_GroupAllocateMemory(
                auhalHostApi->allocations, sizeof(PaDeviceInfo) * auhalHostApi->devCount );
        if( !deviceInfoArray )
        {
            result = paInsufficientMemory;
            goto error;
        }

        for( i=0; i < auhalHostApi->devCount; ++i )
        {
            int err;
            err = InitializeDeviceInfo( auhalHostApi, &deviceInfoArray[i],
                                      auhalHostApi->devIds[i],
                                      hostApiIndex );
            if (err == paNoError)
            { /* copy some info and set the defaults */
                (*hostApi)->deviceInfos[(*hostApi)->info.deviceCount] = &deviceInfoArray[i];
                if (auhalHostApi->devIds[i] == auhalHostApi->defaultIn)
                    (*hostApi)->info.defaultInputDevice = (*hostApi)->info.deviceCount;
                if (auhalHostApi->devIds[i] == auhalHostApi->defaultOut)
                    (*hostApi)->info.defaultOutputDevice = (*hostApi)->info.deviceCount;
                (*hostApi)->info.deviceCount++;
            }
            else
            { /* there was an error. we need to shift the devices down, so we ignore this one */
                int j;
                auhalHostApi->devCount--;
                for( j=i; j<auhalHostApi->devCount; ++j )
                   auhalHostApi->devIds[j] = auhalHostApi->devIds[j+1];
                i--;
            }
        }
    }

    (*hostApi)->Terminate = Terminate;
    (*hostApi)->OpenStream = OpenStream;
    (*hostApi)->IsFormatSupported = IsFormatSupported;

    PaUtil_InitializeStreamInterface( &auhalHostApi->callbackStreamInterface,
                                      CloseStream, StartStream,
                                      StopStream, AbortStream, IsStreamStopped,
                                      IsStreamActive,
                                      GetStreamTime, GetStreamCpuLoad,
                                      PaUtil_DummyRead, PaUtil_DummyWrite,
                                      PaUtil_DummyGetReadAvailable,
                                      PaUtil_DummyGetWriteAvailable );

    PaUtil_InitializeStreamInterface( &auhalHostApi->blockingStreamInterface,
                                      CloseStream, StartStream,
                                      StopStream, AbortStream, IsStreamStopped,
                                      IsStreamActive,
                                      GetStreamTime, PaUtil_DummyGetCpuLoad,
                                      ReadStream, WriteStream,
                                      GetStreamReadAvailable,
                                      GetStreamWriteAvailable );

    return result;

error:
    if( auhalHostApi )
    {
        if( auhalHostApi->allocations )
        {
            PaUtil_FreeAllAllocations( auhalHostApi->allocations );
            PaUtil_DestroyAllocationGroup( auhalHostApi->allocations );
        }

        PaUtil_FreeMemory( auhalHostApi );
    }
    return result;
}

static void Terminate( struct PaUtilHostApiRepresentation *hostApi )
{
    PaMacAUHAL *auhalHostApi = (PaMacAUHAL*)hostApi;

    VVDBUG(("Terminate()\n"));

    /*
        IMPLEMENT ME:
            - clean up any resources not handled by the allocation group
        TODO: Double check that everything is handled by alloc group
    */

    if( auhalHostApi->allocations )
    {
        PaUtil_FreeAllAllocations( auhalHostApi->allocations );
        PaUtil_DestroyAllocationGroup( auhalHostApi->allocations );
    }

    PaUtil_FreeMemory( auhalHostApi );
}

static PaError IsFormatSupported( struct PaUtilHostApiRepresentation *hostApi,
                                  const PaStreamParameters *inputParameters,
                                  const PaStreamParameters *outputParameters,
                                  double sampleRate )
{
    int inputChannelCount, outputChannelCount;
    PaSampleFormat inputSampleFormat, outputSampleFormat;

    VVDBUG(("IsFormatSupported(): in chan=%d, in fmt=%ld, out chan=%d, out fmt=%ld sampleRate=%g\n",
                inputParameters  ? inputParameters->channelCount  : -1,
                inputParameters  ? inputParameters->sampleFormat  : -1,
                outputParameters ? outputParameters->channelCount : -1,
                outputParameters ? outputParameters->sampleFormat : -1,
                (float) sampleRate ));

    /** These first checks are standard PA checks. We do some fancier checks
        later. */
    if( inputParameters )
    {
        inputChannelCount = inputParameters->channelCount;
        inputSampleFormat = inputParameters->sampleFormat;

        /* all standard sample formats are supported by the buffer adapter,
            this implementation doesn't support any custom sample formats */
        if( inputSampleFormat & paCustomFormat )
            return paSampleFormatNotSupported;

        /* unless alternate device specification is supported, reject the use of
            paUseHostApiSpecificDeviceSpecification */

        if( inputParameters->device == paUseHostApiSpecificDeviceSpecification )
            return paInvalidDevice;

        /* check that input device can support inputChannelCount */
        if( inputChannelCount > hostApi->deviceInfos[ inputParameters->device ]->maxInputChannels )
            return paInvalidChannelCount;
    }
    else
    {
        inputChannelCount = 0;
    }

    if( outputParameters )
    {
        outputChannelCount = outputParameters->channelCount;
        outputSampleFormat = outputParameters->sampleFormat;

        /* all standard sample formats are supported by the buffer adapter,
            this implementation doesn't support any custom sample formats */
        if( outputSampleFormat & paCustomFormat )
            return paSampleFormatNotSupported;

        /* unless alternate device specification is supported, reject the use of
            paUseHostApiSpecificDeviceSpecification */

        if( outputParameters->device == paUseHostApiSpecificDeviceSpecification )
            return paInvalidDevice;

        /* check that output device can support outputChannelCount */
        if( outputChannelCount > hostApi->deviceInfos[ outputParameters->device ]->maxOutputChannels )
            return paInvalidChannelCount;

    }
    else
    {
        outputChannelCount = 0;
    }

    /* FEEDBACK */
    /*        I think the only way to check a given format SR combo is     */
    /*        to try opening it. This could be disruptive, is that Okay?   */
    /*        The alternative is to just read off available sample rates,  */
    /*        but this will not work %100 of the time (eg, a device that   */
    /*        supports N output at one rate but only N/2 at a higher rate.)*/

    /* The following code opens the device with the requested parameters to
       see if it works. */
    {
       PaError err;
       PaStream *s;
       err = OpenStream( hostApi, &s, inputParameters, outputParameters,
                           sampleRate, 1024, 0, (PaStreamCallback *)1, NULL );
       if( err != paNoError && err != paInvalidSampleRate )
          DBUG( ( "OpenStream @ %g returned: %d: %s\n",
                  (float) sampleRate, err, Pa_GetErrorText( err ) ) );
       if( err )
          return err;
       err = CloseStream( s );
       if( err ) {
          /* FEEDBACK: is this more serious? should we assert? */
          DBUG( ( "WARNING: could not close Stream. %d: %s\n",
                  err, Pa_GetErrorText( err ) ) );
       }
    }

    return paFormatIsSupported;
}

static PaError OpenAndSetupOneAudioUnit(
                                   const PaMacCoreStream *stream,
                                   const PaStreamParameters *inStreamParams,
                                   const PaStreamParameters *outStreamParams,
                                   const UInt32 requestedFramesPerBuffer,
                                   UInt32 *actualInputFramesPerBuffer,
                                   UInt32 *actualOutputFramesPerBuffer,
                                   const PaMacAUHAL *auhalHostApi,
                                   AudioUnit *audioUnit,
                                   AudioConverterRef *srConverter,
                                   AudioDeviceID *audioDevice,
                                   const double sampleRate,
                                   void *refCon )
{
    ComponentDescription desc;
    Component comp;
    /*An Apple TN suggests using CAStreamBasicDescription, but that is C++*/
    AudioStreamBasicDescription desiredFormat;
    OSStatus result = noErr;
    PaError paResult = paNoError;
    int line = 0;
    UInt32 callbackKey;
    AURenderCallbackStruct rcbs;
    unsigned long macInputStreamFlags  = paMacCorePlayNice;
    unsigned long macOutputStreamFlags = paMacCorePlayNice;
    SInt32 const *inChannelMap = NULL;
    SInt32 const *outChannelMap = NULL;
    unsigned long inChannelMapSize = 0;
    unsigned long outChannelMapSize = 0;

    VVDBUG(("OpenAndSetupOneAudioUnit(): in chan=%d, in fmt=%ld, out chan=%d, out fmt=%ld, requestedFramesPerBuffer=%ld\n",
                inStreamParams  ? inStreamParams->channelCount  : -1,
                inStreamParams  ? inStreamParams->sampleFormat  : -1,
                outStreamParams ? outStreamParams->channelCount : -1,
                outStreamParams ? outStreamParams->sampleFormat : -1,
                requestedFramesPerBuffer ));

    /* -- handle the degenerate case  -- */
    if( !inStreamParams && !outStreamParams ) {
       *audioUnit = NULL;
       *audioDevice = kAudioDeviceUnknown;
       return paNoError;
    }

    /* -- get the user's api specific info, if they set any -- */
    if( inStreamParams && inStreamParams->hostApiSpecificStreamInfo )
    {
       macInputStreamFlags=
            ((PaMacCoreStreamInfo*)inStreamParams->hostApiSpecificStreamInfo)
                  ->flags;
       inChannelMap = ((PaMacCoreStreamInfo*)inStreamParams->hostApiSpecificStreamInfo)
                  ->channelMap;
       inChannelMapSize = ((PaMacCoreStreamInfo*)inStreamParams->hostApiSpecificStreamInfo)
                  ->channelMapSize;
    }
    if( outStreamParams && outStreamParams->hostApiSpecificStreamInfo )
    {
       macOutputStreamFlags=
            ((PaMacCoreStreamInfo*)outStreamParams->hostApiSpecificStreamInfo)
                  ->flags;
       outChannelMap = ((PaMacCoreStreamInfo*)outStreamParams->hostApiSpecificStreamInfo)
                  ->channelMap;
       outChannelMapSize = ((PaMacCoreStreamInfo*)outStreamParams->hostApiSpecificStreamInfo)
                  ->channelMapSize;
    }
    /* Override user's flags here, if desired for testing. */

    /*
     * The HAL AU is a Mac OS style "component".
     * the first few steps deal with that.
     * Later steps work on a combination of Mac OS
     * components and the slightly lower level
     * HAL.
     */

    /* -- describe the output type AudioUnit -- */
    /*  Note: for the default AudioUnit, we could use the
     *  componentSubType value kAudioUnitSubType_DefaultOutput;
     *  but I don't think that's relevant here.
     */
    desc.componentType         = kAudioUnitType_Output;
    desc.componentSubType      = kAudioUnitSubType_HALOutput;
    desc.componentManufacturer = kAudioUnitManufacturer_Apple;
    desc.componentFlags        = 0;
    desc.componentFlagsMask    = 0;
    /* -- find the component -- */
    comp = FindNextComponent( NULL, &desc );
    if( !comp )
    {
       DBUG( ( "AUHAL component not found." ) );
       *audioUnit = NULL;
       *audioDevice = kAudioDeviceUnknown;
       return paUnanticipatedHostError;
    }
    /* -- open it -- */
    result = OpenAComponent( comp, audioUnit );
    if( result )
    {
       DBUG( ( "Failed to open AUHAL component." ) );
       *audioUnit = NULL;
       *audioDevice = kAudioDeviceUnknown;
       return ERR( result );
    }
    /* -- prepare a little error handling logic / hackery -- */
#define ERR_WRAP(mac_err) do { result = mac_err ; line = __LINE__ ; if ( result != noErr ) goto error ; } while(0)

    /* -- if there is input, we have to explicitly enable input -- */
    if( inStreamParams )
    {
       UInt32 enableIO = 1;
       ERR_WRAP( AudioUnitSetProperty( *audioUnit,
                 kAudioOutputUnitProperty_EnableIO,
                 kAudioUnitScope_Input,
                 INPUT_ELEMENT,
                 &enableIO,
                 sizeof(enableIO) ) );
    }
    /* -- if there is no output, we must explicitly disable output -- */
    if( !outStreamParams )
    {
       UInt32 enableIO = 0;
       ERR_WRAP( AudioUnitSetProperty( *audioUnit,
                 kAudioOutputUnitProperty_EnableIO,
                 kAudioUnitScope_Output,
                 OUTPUT_ELEMENT,
                 &enableIO,
                 sizeof(enableIO) ) );
    }

    /* -- set the devices -- */
    /* make sure input and output are the same device if we are doing input and
       output. */
    if( inStreamParams && outStreamParams )
    {
       assert( outStreamParams->device == inStreamParams->device );
    }
    if( inStreamParams )
    {
       *audioDevice = auhalHostApi->devIds[inStreamParams->device] ;
       ERR_WRAP( AudioUnitSetProperty( *audioUnit,
                    kAudioOutputUnitProperty_CurrentDevice,
                    kAudioUnitScope_Global,
                    INPUT_ELEMENT,
                    audioDevice,
                    sizeof(AudioDeviceID) ) );
    }
    if( outStreamParams && outStreamParams != inStreamParams )
    {
       *audioDevice = auhalHostApi->devIds[outStreamParams->device] ;
       ERR_WRAP( AudioUnitSetProperty( *audioUnit,
                    kAudioOutputUnitProperty_CurrentDevice,
                    kAudioUnitScope_Global,
                    OUTPUT_ELEMENT,
                    audioDevice,
                    sizeof(AudioDeviceID) ) );
    }
    /* -- add listener for dropouts -- */
    ERR_WRAP( AudioDeviceAddPropertyListener( *audioDevice,
                                              0,
                                              outStreamParams ? false : true,
                                              kAudioDeviceProcessorOverload,
                                              xrunCallback,
                                              (void *)stream) );

    /* -- listen for stream start and stop -- */
    ERR_WRAP( AudioUnitAddPropertyListener( *audioUnit,
                                            kAudioOutputUnitProperty_IsRunning,
                                            startStopCallback,
                                            (void *)stream ) );

    /* -- set format -- */
    bzero( &desiredFormat, sizeof(desiredFormat) );
    desiredFormat.mFormatID         = kAudioFormatLinearPCM ;
    desiredFormat.mFormatFlags      = kAudioFormatFlagsNativeFloatPacked;
    desiredFormat.mFramesPerPacket  = 1;
    desiredFormat.mBitsPerChannel   = sizeof( float ) * 8;

    result = 0;
    /*  set device format first, but only touch the device if the user asked */
    if( inStreamParams ) {
       /*The callback never calls back if we don't set the FPB */
       /*This seems wierd, because I would think setting anything on the device
         would be disruptive.*/
       paResult = setBestFramesPerBuffer( *audioDevice, FALSE,
                                          requestedFramesPerBuffer,
                                          actualInputFramesPerBuffer );
       if( paResult ) goto error;
       if( macInputStreamFlags & paMacCoreChangeDeviceParameters ) {
          bool requireExact;
          requireExact=macInputStreamFlags & paMacCoreFailIfConversionRequired;
          paResult = setBestSampleRateForDevice( *audioDevice, FALSE,
                                                 requireExact, sampleRate );
          if( paResult ) goto error;
       }
       if( actualInputFramesPerBuffer && actualOutputFramesPerBuffer )
          *actualOutputFramesPerBuffer = *actualInputFramesPerBuffer ;
    }
    if( outStreamParams && !inStreamParams ) {
       /*The callback never calls back if we don't set the FPB */
       /*This seems wierd, because I would think setting anything on the device
         would be disruptive.*/
       paResult = setBestFramesPerBuffer( *audioDevice, TRUE,
                                          requestedFramesPerBuffer,
                                          actualOutputFramesPerBuffer );
       if( paResult ) goto error;
       if( macOutputStreamFlags & paMacCoreChangeDeviceParameters ) {
          bool requireExact;
          requireExact=macOutputStreamFlags & paMacCoreFailIfConversionRequired;
          paResult = setBestSampleRateForDevice( *audioDevice, TRUE,
                                                 requireExact, sampleRate );
          if( paResult ) goto error;
       }
    }

    /* -- set the quality of the output converter -- */
    if( outStreamParams ) {
       UInt32 value = kAudioConverterQuality_Max;
       switch( macOutputStreamFlags & 0x0700 ) {
       case 0x0100: /*paMacCore_ConversionQualityMin:*/
          value=kRenderQuality_Min;
          break;
       case 0x0200: /*paMacCore_ConversionQualityLow:*/
          value=kRenderQuality_Low;
          break;
       case 0x0300: /*paMacCore_ConversionQualityMedium:*/
          value=kRenderQuality_Medium;
          break;
       case 0x0400: /*paMacCore_ConversionQualityHigh:*/
          value=kRenderQuality_High;
          break;
       }
       ERR_WRAP( AudioUnitSetProperty( *audioUnit,
                    kAudioUnitProperty_RenderQuality,
                    kAudioUnitScope_Global,
                    OUTPUT_ELEMENT,
                    &value,
                    sizeof(value) ) );
    }
    /* now set the format on the Audio Units. */
    if( outStreamParams )
    {
       desiredFormat.mSampleRate    =sampleRate;
       desiredFormat.mBytesPerPacket=sizeof(float)*outStreamParams->channelCount;
       desiredFormat.mBytesPerFrame =sizeof(float)*outStreamParams->channelCount;
       desiredFormat.mChannelsPerFrame = outStreamParams->channelCount;
       ERR_WRAP( AudioUnitSetProperty( *audioUnit,
                            kAudioUnitProperty_StreamFormat,
                            kAudioUnitScope_Input,
                            OUTPUT_ELEMENT,
                            &desiredFormat,
                            sizeof(AudioStreamBasicDescription) ) );
    }
    if( inStreamParams )
    {
       AudioStreamBasicDescription sourceFormat;
       UInt32 size = sizeof( AudioStreamBasicDescription );

       /* keep the sample rate of the device, or we confuse AUHAL */
       ERR_WRAP( AudioUnitGetProperty( *audioUnit,
                            kAudioUnitProperty_StreamFormat,
                            kAudioUnitScope_Input,
                            INPUT_ELEMENT,
                            &sourceFormat,
                            &size ) );
       desiredFormat.mSampleRate = sourceFormat.mSampleRate;
       desiredFormat.mBytesPerPacket=sizeof(float)*inStreamParams->channelCount;
       desiredFormat.mBytesPerFrame =sizeof(float)*inStreamParams->channelCount;
       desiredFormat.mChannelsPerFrame = inStreamParams->channelCount;
       ERR_WRAP( AudioUnitSetProperty( *audioUnit,
                            kAudioUnitProperty_StreamFormat,
                            kAudioUnitScope_Output,
                            INPUT_ELEMENT,
                            &desiredFormat,
                            sizeof(AudioStreamBasicDescription) ) );
    }
    /* set the maximumFramesPerSlice */
    /* not doing this causes real problems
       (eg. the callback might not be called). The idea of setting both this
       and the frames per buffer on the device is that we'll be most likely
       to actually get the frame size we requested in the callback with the
       minimum latency. */
    if( outStreamParams ) {
       UInt32 size = sizeof( *actualOutputFramesPerBuffer );
       ERR_WRAP( AudioUnitSetProperty( *audioUnit,
                            kAudioUnitProperty_MaximumFramesPerSlice,
                            kAudioUnitScope_Input,
                            OUTPUT_ELEMENT,
                            actualOutputFramesPerBuffer,
                            sizeof(*actualOutputFramesPerBuffer) ) );
       ERR_WRAP( AudioUnitGetProperty( *audioUnit,
                            kAudioUnitProperty_MaximumFramesPerSlice,
                            kAudioUnitScope_Global,
                            OUTPUT_ELEMENT,
                            actualOutputFramesPerBuffer,
                            &size ) );
    }
    if( inStreamParams ) {
       /*UInt32 size = sizeof( *actualInputFramesPerBuffer );*/
       ERR_WRAP( AudioUnitSetProperty( *audioUnit,
                            kAudioUnitProperty_MaximumFramesPerSlice,
                            kAudioUnitScope_Output,
                            INPUT_ELEMENT,
                            actualInputFramesPerBuffer,
                            sizeof(*actualInputFramesPerBuffer) ) );
/* Don't know why this causes problems
       ERR_WRAP( AudioUnitGetProperty( *audioUnit,
                            kAudioUnitProperty_MaximumFramesPerSlice,
                            kAudioUnitScope_Global, //Output,
                            INPUT_ELEMENT,
                            actualInputFramesPerBuffer,
                            &size ) );
*/
    }

    /* -- if we have input, we may need to setup an SR converter -- */
    /* even if we got the sample rate we asked for, we need to do
       the conversion in case another program changes the underlying SR. */
    /* FIXME: I think we need to monitor stream and change the converter if the incoming format changes. */
    if( inStreamParams ) {
       AudioStreamBasicDescription desiredFormat;
       AudioStreamBasicDescription sourceFormat;
       UInt32 sourceSize = sizeof( sourceFormat );
       bzero( &desiredFormat, sizeof(desiredFormat) );
       desiredFormat.mSampleRate       = sampleRate;
       desiredFormat.mFormatID         = kAudioFormatLinearPCM ;
       desiredFormat.mFormatFlags      = kAudioFormatFlagsNativeFloatPacked;
       desiredFormat.mFramesPerPacket  = 1;
       desiredFormat.mBitsPerChannel   = sizeof( float ) * 8;
       desiredFormat.mBytesPerPacket=sizeof(float)*inStreamParams->channelCount;
       desiredFormat.mBytesPerFrame =sizeof(float)*inStreamParams->channelCount;
       desiredFormat.mChannelsPerFrame = inStreamParams->channelCount;

       /* get the source format */
       ERR_WRAP( AudioUnitGetProperty(
                         *audioUnit,
                         kAudioUnitProperty_StreamFormat,
                         kAudioUnitScope_Output,
                         INPUT_ELEMENT,
                         &sourceFormat,
                         &sourceSize ) );

       if( desiredFormat.mSampleRate != sourceFormat.mSampleRate )
       {
          UInt32 value = kAudioConverterQuality_Max;
          switch( macInputStreamFlags & 0x0700 ) {
          case 0x0100: /*paMacCore_ConversionQualityMin:*/
             value=kAudioConverterQuality_Min;
             break;
          case 0x0200: /*paMacCore_ConversionQualityLow:*/
             value=kAudioConverterQuality_Low;
             break;
          case 0x0300: /*paMacCore_ConversionQualityMedium:*/
             value=kAudioConverterQuality_Medium;
             break;
          case 0x0400: /*paMacCore_ConversionQualityHigh:*/
             value=kAudioConverterQuality_High;
             break;
          }
          VDBUG(( "Creating sample rate converter for input"
                  " to convert from %g to %g\n",
                  (float)sourceFormat.mSampleRate,
                  (float)desiredFormat.mSampleRate ) );
          /* create our converter */
          ERR_WRAP( AudioConverterNew(
                             &sourceFormat,
                             &desiredFormat,
                             srConverter ) );
          /* Set quality */
          ERR_WRAP( AudioConverterSetProperty(
                             *srConverter,
                             kAudioConverterSampleRateConverterQuality,
                             sizeof( value ),
                             &value ) );
       }
    }
    /* -- set IOProc (callback) -- */
    callbackKey = outStreamParams ? kAudioUnitProperty_SetRenderCallback
                                  : kAudioOutputUnitProperty_SetInputCallback ;
    rcbs.inputProc = AudioIOProc;
    rcbs.inputProcRefCon = refCon;
    ERR_WRAP( AudioUnitSetProperty(
                               *audioUnit,
                               callbackKey,
                               kAudioUnitScope_Output,
                               outStreamParams ? OUTPUT_ELEMENT : INPUT_ELEMENT,
                               &rcbs,
                               sizeof(rcbs)) );

    if( inStreamParams && outStreamParams && *srConverter )
           ERR_WRAP( AudioUnitSetProperty(
                               *audioUnit,
                               kAudioOutputUnitProperty_SetInputCallback,
                               kAudioUnitScope_Output,
                               INPUT_ELEMENT,
                               &rcbs,
                               sizeof(rcbs)) );

    /* channel mapping. */
    if(inChannelMap)
    {
        UInt32 mapSize = inChannelMapSize *sizeof(SInt32);

        //for each channel of desired input, map the channel from
        //the device's output channel.
        ERR_WRAP( AudioUnitSetProperty(*audioUnit,
                                kAudioOutputUnitProperty_ChannelMap,
                                kAudioUnitScope_Output,
                                INPUT_ELEMENT,
                                inChannelMap,
                                mapSize));
    }
    if(outChannelMap)
    {
        UInt32 mapSize = outChannelMapSize *sizeof(SInt32);

        //for each channel of desired output, map the channel from
        //the device's output channel.
        ERR_WRAP(AudioUnitSetProperty(*audioUnit,
                                kAudioOutputUnitProperty_ChannelMap,
                                kAudioUnitScope_Output,
                                OUTPUT_ELEMENT,
                                outChannelMap,
                                mapSize));
    }
    /* initialize the audio unit */
    ERR_WRAP( AudioUnitInitialize(*audioUnit) );

    if( inStreamParams && outStreamParams )
       VDBUG( ("Opened device %ld for input and output.\n", *audioDevice ) );
    else if( inStreamParams )
       VDBUG( ("Opened device %ld for input.\n", *audioDevice ) );
    else if( outStreamParams )
       VDBUG( ("Opened device %ld for output.\n", *audioDevice ) );
    return paNoError;
#undef ERR_WRAP

    error:
       CloseComponent( *audioUnit );
       *audioUnit = NULL;
       if( result )
          return PaMacCore_SetError( result, line, 1 );
       return paResult;
}

/* see pa_hostapi.h for a list of validity guarantees made about OpenStream parameters */
static PaError OpenStream( struct PaUtilHostApiRepresentation *hostApi,
                           PaStream** s,
                           const PaStreamParameters *inputParameters,
                           const PaStreamParameters *outputParameters,
                           double sampleRate,
                           unsigned long framesPerBuffer,
                           PaStreamFlags streamFlags,
                           PaStreamCallback *streamCallback,
                           void *userData )
{
    PaError result = paNoError;
    PaMacAUHAL *auhalHostApi = (PaMacAUHAL*)hostApi;
    PaMacCoreStream *stream = 0;
    int inputChannelCount, outputChannelCount;
    PaSampleFormat inputSampleFormat, outputSampleFormat;
    PaSampleFormat hostInputSampleFormat, hostOutputSampleFormat;
    VVDBUG(("OpenStream(): in chan=%d, in fmt=%ld, out chan=%d, out fmt=%ld SR=%g, FPB=%ld\n",
                inputParameters  ? inputParameters->channelCount  : -1,
                inputParameters  ? inputParameters->sampleFormat  : -1,
                outputParameters ? outputParameters->channelCount : -1,
                outputParameters ? outputParameters->sampleFormat : -1,
                (float) sampleRate,
                framesPerBuffer ));
    VDBUG( ("Opening Stream.\n") );

    /*These first few bits of code are from paSkeleton with few modifications.*/
    if( inputParameters )
    {
        inputChannelCount = inputParameters->channelCount;
        inputSampleFormat = inputParameters->sampleFormat;

        /* unless alternate device specification is supported, reject the use of
            paUseHostApiSpecificDeviceSpecification */

        if( inputParameters->device == paUseHostApiSpecificDeviceSpecification )
            return paInvalidDevice;

        /* check that input device can support inputChannelCount */
        if( inputChannelCount > hostApi->deviceInfos[ inputParameters->device ]->maxInputChannels )
            return paInvalidChannelCount;

        /* Host supports interleaved float32 */
        hostInputSampleFormat = paFloat32;
    }
    else
    {
        inputChannelCount = 0;
        inputSampleFormat = hostInputSampleFormat = paFloat32; /* Surpress 'uninitialised var' warnings. */
    }

    if( outputParameters )
    {
        outputChannelCount = outputParameters->channelCount;
        outputSampleFormat = outputParameters->sampleFormat;

        /* unless alternate device specification is supported, reject the use of
            paUseHostApiSpecificDeviceSpecification */

        if( outputParameters->device == paUseHostApiSpecificDeviceSpecification )
            return paInvalidDevice;

        /* check that output device can support inputChannelCount */
        if( outputChannelCount > hostApi->deviceInfos[ outputParameters->device ]->maxOutputChannels )
            return paInvalidChannelCount;

        /* Host supports interleaved float32 */
        hostOutputSampleFormat = paFloat32;
    }
    else
    {
        outputChannelCount = 0;
        outputSampleFormat = hostOutputSampleFormat = paFloat32; /* Surpress 'uninitialized var' warnings. */
    }

    /* validate platform specific flags */
    if( (streamFlags & paPlatformSpecificFlags) != 0 )
        return paInvalidFlag; /* unexpected platform specific flag */

    stream = (PaMacCoreStream*)PaUtil_AllocateMemory( sizeof(PaMacCoreStream) );
    if( !stream )
    {
        result = paInsufficientMemory;
        goto error;
    }

    /* If we fail after this point, we my be left in a bad state, with
       some data structures setup and others not. So, first thing we
       do is initialize everything so that if we fail, we know what hasn't
       been touched.
     */

    stream->inputAudioBufferList.mBuffers[0].mData = NULL;
    stream->inputRingBuffer.buffer = NULL;
    bzero( &stream->blio, sizeof( PaMacBlio ) );
/*
    stream->blio.inputRingBuffer.buffer = NULL;
    stream->blio.outputRingBuffer.buffer = NULL;
    stream->blio.inputSampleFormat = inputParameters?inputParameters->sampleFormat:0;
    stream->blio.inputSampleSize = computeSampleSizeFromFormat(stream->blio.inputSampleFormat);
    stream->blio.outputSampleFormat=outputParameters?outputParameters->sampleFormat:0;
    stream->blio.outputSampleSize = computeSampleSizeFromFormat(stream->blio.outputSampleFormat);
*/
    stream->inputSRConverter = NULL;
    stream->inputUnit = NULL;
    stream->outputUnit = NULL;
    stream->inputFramesPerBuffer = 0;
    stream->outputFramesPerBuffer = 0;
    stream->bufferProcessorIsInitialized = FALSE;

    /* assert( streamCallback ) ; */ /* only callback mode is implemented */
    if( streamCallback )
    {
        PaUtil_InitializeStreamRepresentation( &stream->streamRepresentation,
                                        &auhalHostApi->callbackStreamInterface,
                                        streamCallback, userData );
    }
    else
    {
        PaUtil_InitializeStreamRepresentation( &stream->streamRepresentation,
                                        &auhalHostApi->blockingStreamInterface,
                                        BlioCallback, &stream->blio );
    }

    PaUtil_InitializeCpuLoadMeasurer( &stream->cpuLoadMeasurer, sampleRate );

    /* -- handle paFramesPerBufferUnspecified -- */
    if( framesPerBuffer == paFramesPerBufferUnspecified ) {
       long requested = 64;
       if( inputParameters )
          requested = MAX( requested, inputParameters->suggestedLatency * sampleRate / 2 );
       if( outputParameters )
          requested = MAX( requested, outputParameters->suggestedLatency *sampleRate / 2 );
       VDBUG( ("Block Size unspecified. Based on Latency, the user wants a Block Size near: %ld.\n",
              requested ) );
       if( requested <= 64 ) {
          /*requested a realtively low latency. make sure this is in range of devices */
          /*try to get the device's min natural buffer size and use that (but no smaller than 64).*/
          AudioValueRange audioRange;
          UInt32 size = sizeof( audioRange );
          if( inputParameters ) {
             WARNING( result = AudioDeviceGetProperty( auhalHostApi->devIds[inputParameters->device],
                                          0,
                                          false,
                                          kAudioDevicePropertyBufferFrameSizeRange,
                                          &size, &audioRange ) );
             if( result )
                requested = MAX( requested, audioRange.mMinimum );
          }
          size = sizeof( audioRange );
          if( outputParameters ) {
             WARNING( result = AudioDeviceGetProperty( auhalHostApi->devIds[outputParameters->device],
                                          0,
                                          false,
                                          kAudioDevicePropertyBufferFrameSizeRange,
                                          &size, &audioRange ) );
             if( result )
                requested = MAX( requested, audioRange.mMinimum );
          }
       } else {
          /* requested a realtively high latency. make sure this is in range of devices */
          /*try to get the device's max natural buffer size and use that (but no larger than 1024).*/
          AudioValueRange audioRange;
          UInt32 size = sizeof( audioRange );
          requested = MIN( requested, 1024 );
          if( inputParameters ) {
             WARNING( result = AudioDeviceGetProperty( auhalHostApi->devIds[inputParameters->device],
                                          0,
                                          false,
                                          kAudioDevicePropertyBufferFrameSizeRange,
                                          &size, &audioRange ) );
             if( result )
                requested = MIN( requested, audioRange.mMaximum );
          }
          size = sizeof( audioRange );
          if( outputParameters ) {
             WARNING( result = AudioDeviceGetProperty( auhalHostApi->devIds[outputParameters->device],
                                          0,
                                          false,
                                          kAudioDevicePropertyBufferFrameSizeRange,
                                          &size, &audioRange ) );
             if( result )
                requested = MIN( requested, audioRange.mMaximum );
          }
       }
       /* -- double check ranges -- */
       if( requested > 1024 ) requested = 1024;
       if( requested < 64 ) requested = 64;
       VDBUG(("After querying hardware, setting block size to %ld.\n", requested));
       framesPerBuffer = requested;
    }

    /* -- Now we actually open and setup streams. -- */
    if( inputParameters && outputParameters && outputParameters->device == inputParameters->device )
    { /* full duplex. One device. */
       UInt32 inputFramesPerBuffer  = (UInt32) stream->inputFramesPerBuffer;
       UInt32 outputFramesPerBuffer = (UInt32) stream->outputFramesPerBuffer;
       result = OpenAndSetupOneAudioUnit( stream,
                                          inputParameters,
                                          outputParameters,
                                          framesPerBuffer,
                                          &inputFramesPerBuffer,
                                          &outputFramesPerBuffer,
                                          auhalHostApi,
                                          &(stream->inputUnit),
                                          &(stream->inputSRConverter),
                                          &(stream->inputDevice),
                                          sampleRate,
                                          stream );
       stream->inputFramesPerBuffer = inputFramesPerBuffer;
       stream->outputFramesPerBuffer = outputFramesPerBuffer;
       stream->outputUnit = stream->inputUnit;
       stream->outputDevice = stream->inputDevice;
       if( result != paNoError )
           goto error;
    }
    else
    { /* full duplex, different devices OR simplex */
       UInt32 outputFramesPerBuffer = (UInt32) stream->outputFramesPerBuffer;
       UInt32 inputFramesPerBuffer  = (UInt32) stream->inputFramesPerBuffer;
       result = OpenAndSetupOneAudioUnit( stream,
                                          NULL,
                                          outputParameters,
                                          framesPerBuffer,
                                          NULL,
                                          &outputFramesPerBuffer,
                                          auhalHostApi,
                                          &(stream->outputUnit),
                                          NULL,
                                          &(stream->outputDevice),
                                          sampleRate,
                                          stream );
       if( result != paNoError )
           goto error;
       result = OpenAndSetupOneAudioUnit( stream,
                                          inputParameters,
                                          NULL,
                                          framesPerBuffer,
                                          &inputFramesPerBuffer,
                                          NULL,
                                          auhalHostApi,
                                          &(stream->inputUnit),
                                          &(stream->inputSRConverter),
                                          &(stream->inputDevice),
                                          sampleRate,
                                          stream );
       if( result != paNoError )
           goto error;
       stream->inputFramesPerBuffer = inputFramesPerBuffer;
       stream->outputFramesPerBuffer = outputFramesPerBuffer;
    }

    if( stream->inputUnit ) {
       const size_t szfl = sizeof(float);
       /* setup the AudioBufferList used for input */
       bzero( &stream->inputAudioBufferList, sizeof( AudioBufferList ) );
       stream->inputAudioBufferList.mNumberBuffers = 1;
       stream->inputAudioBufferList.mBuffers[0].mNumberChannels
                 = inputChannelCount;
       stream->inputAudioBufferList.mBuffers[0].mDataByteSize
                 = stream->inputFramesPerBuffer*inputChannelCount*szfl;
       stream->inputAudioBufferList.mBuffers[0].mData
                 = (float *) calloc(
                               stream->inputFramesPerBuffer*inputChannelCount,
                               szfl );
       if( !stream->inputAudioBufferList.mBuffers[0].mData )
       {
          result = paInsufficientMemory;
          goto error;
       }

       /*
        * If input and output devs are different or we are doing SR conversion,
        * we also need a
        * ring buffer to store inpt data while waiting for output
        * data.
        */
       if( (stream->outputUnit && stream->inputUnit != stream->outputUnit)
           || stream->inputSRConverter )
       {
          /* May want the ringSize ot initial position in
             ring buffer to depend somewhat on sample rate change */

          void *data;
          long ringSize;

          ringSize = computeRingBufferSize( inputParameters,
                                            outputParameters,
                                            stream->inputFramesPerBuffer,
                                            stream->outputFramesPerBuffer,
                                            sampleRate );
          /*ringSize <<= 4; *//*16x bigger, for testing */

          /*now, we need to allocate memory for the ring buffer*/
          data = calloc( ringSize, szfl );
          if( !data )
          {
             result = paInsufficientMemory;
             goto error;
          }

          /* now we can initialize the ring buffer */
          PaUtil_InitializeRingBuffer( &stream->inputRingBuffer,
                                   ringSize*szfl, data ) ;
          /* advance the read point a little, so we are reading from the
             middle of the buffer */
          if( stream->outputUnit )
             PaUtil_AdvanceRingBufferWriteIndex( &stream->inputRingBuffer, ringSize*szfl / RING_BUFFER_ADVANCE_DENOMINATOR );
       }
    }

    /* -- initialize Blio Buffer Processors -- */
    if( !streamCallback )
    {
       long ringSize;

       ringSize = computeRingBufferSize( inputParameters,
                                         outputParameters,
                                         stream->inputFramesPerBuffer,
                                         stream->outputFramesPerBuffer,
                                         sampleRate );
       result = initializeBlioRingBuffers( &stream->blio,
              inputParameters?inputParameters->sampleFormat:0 ,
              outputParameters?outputParameters->sampleFormat:0 ,
              MAX(stream->inputFramesPerBuffer,stream->outputFramesPerBuffer),
              ringSize,
              inputParameters?inputChannelCount:0 ,
              outputParameters?outputChannelCount:0 ) ;
       if( result != paNoError )
          goto error;
    }

    /* -- initialize Buffer Processor -- */
    {
       unsigned long maxHostFrames = stream->inputFramesPerBuffer;
       if( stream->outputFramesPerBuffer > maxHostFrames )
          maxHostFrames = stream->outputFramesPerBuffer;
       result = PaUtil_InitializeBufferProcessor( &stream->bufferProcessor,
                 inputChannelCount, inputSampleFormat,
                 hostInputSampleFormat,
                 outputChannelCount, outputSampleFormat,
                 hostOutputSampleFormat,
                 sampleRate,
                 streamFlags,
                 framesPerBuffer,
                 /* If sample rate conversion takes place, the buffer size
                    will not be known. */
                 maxHostFrames,
                 stream->inputSRConverter
                              ? paUtilUnknownHostBufferSize
                              : paUtilBoundedHostBufferSize,
                 streamCallback ? streamCallback : BlioCallback,
                 streamCallback ? userData : &stream->blio );
       if( result != paNoError )
           goto error;
    }
    stream->bufferProcessorIsInitialized = TRUE;

    /*
        IMPLEMENT ME: initialise the following fields with estimated or actual
        values.
        I think this is okay the way it is br 12/1/05
        maybe need to change input latency estimate if IO devs differ
    */
    stream->streamRepresentation.streamInfo.inputLatency =
            PaUtil_GetBufferProcessorInputLatency(&stream->bufferProcessor)/sampleRate;
    stream->streamRepresentation.streamInfo.outputLatency =
            PaUtil_GetBufferProcessorOutputLatency(&stream->bufferProcessor)/sampleRate;
    stream->streamRepresentation.streamInfo.sampleRate = sampleRate;

    stream->sampleRate  = sampleRate;
    stream->outDeviceSampleRate = 0;
    if( stream->outputUnit ) {
       Float64 rate;
       UInt32 size = sizeof( rate );
       result = ERR( AudioDeviceGetProperty( stream->outputDevice,
                                    0,
                                    FALSE,
                                    kAudioDevicePropertyNominalSampleRate,
                                    &size, &rate ) );
       if( result )
          goto error;
       stream->outDeviceSampleRate = rate;
    }
    stream->inDeviceSampleRate = 0;
    if( stream->inputUnit ) {
       Float64 rate;
       UInt32 size = sizeof( rate );
       result = ERR( AudioDeviceGetProperty( stream->inputDevice,
                                    0,
                                    TRUE,
                                    kAudioDevicePropertyNominalSampleRate,
                                    &size, &rate ) );
       if( result )
          goto error;
       stream->inDeviceSampleRate = rate;
    }
    stream->userInChan  = inputChannelCount;
    stream->userOutChan = outputChannelCount;

    stream->isTimeSet   = FALSE;
    stream->state = STOPPED;
    stream->xrunFlags = 0;

    *s = (PaStream*)stream;

    return result;

error:
    CloseStream( stream );
    return result;
}

PaTime GetStreamTime( PaStream *s )
{
   /* FIXME: I am not at all sure this timing info stuff is right.
             patest_sine_time reports negative latencies, which is wierd.*/
    PaMacCoreStream *stream = (PaMacCoreStream*)s;
    AudioTimeStamp timeStamp;

    VVDBUG(("GetStreamTime()\n"));

    if ( !stream->isTimeSet )
        return (PaTime)0;

    if ( stream->outputDevice ) {
        AudioDeviceGetCurrentTime( stream->outputDevice, &timeStamp);
        return (PaTime)(timeStamp.mSampleTime - stream->startTime.mSampleTime)/stream->outDeviceSampleRate;
    } else if ( stream->inputDevice ) {
        AudioDeviceGetCurrentTime( stream->inputDevice, &timeStamp);
    return (PaTime)(timeStamp.mSampleTime - stream->startTime.mSampleTime)/stream->inDeviceSampleRate;
    } else {
        return (PaTime)0;
    }
}

static void setStreamStartTime( PaStream *stream )
{
   /* FIXME: I am not at all sure this timing info stuff is right.
             patest_sine_time reports negative latencies, which is wierd.*/
   PaMacCoreStream *s = (PaMacCoreStream *) stream;
   VVDBUG(("setStreamStartTime()\n"));
   if( s->outputDevice )
      AudioDeviceGetCurrentTime( s->outputDevice, &s->startTime);
   else if( s->inputDevice )
      AudioDeviceGetCurrentTime( s->inputDevice, &s->startTime);
   else
      bzero( &s->startTime, sizeof( s->startTime ) );

   //FIXME: we need a memory barier here

   s->isTimeSet = TRUE;
}

static PaTime TimeStampToSecs(PaMacCoreStream *stream, const AudioTimeStamp* timeStamp)
{
    VVDBUG(("TimeStampToSecs()\n"));
    //printf( "ATS: %lu, %g, %g\n", timeStamp->mFlags, timeStamp->mSampleTime, timeStamp->mRateScalar );
    if (timeStamp->mFlags & kAudioTimeStampSampleTimeValid)
        return (timeStamp->mSampleTime / stream->sampleRate);
    else
        return 0;
}

#define RING_BUFFER_EMPTY (1000)

static OSStatus ringBufferIOProc( AudioConverterRef inAudioConverter,
                             UInt32*ioDataSize,
                             void** outData,
                             void*inUserData )
{
   void *dummyData;
   long dummySize;
   PaUtilRingBuffer *rb = (PaUtilRingBuffer *) inUserData;

   VVDBUG(("ringBufferIOProc()\n"));

   assert( sizeof( UInt32 ) == sizeof( long ) );
   if( PaUtil_GetRingBufferReadAvailable( rb ) == 0 ) {
      *outData = NULL;
      *ioDataSize = 0;
      return RING_BUFFER_EMPTY;
   }
   PaUtil_GetRingBufferReadRegions( rb, *ioDataSize,
                                    outData, (long *)ioDataSize,
                                    &dummyData, &dummySize );

   assert( *ioDataSize );
   PaUtil_AdvanceRingBufferReadIndex( rb, *ioDataSize );

   return noErr;
}

/*
 * Called by the AudioUnit API to process audio from the sound card.
 * This is where the magic happens.
 */
/* FEEDBACK: there is a lot of redundant code here because of how all the cases differ. This makes it hard to maintain, so if there are suggestinos for cleaning it up, I'm all ears. */
static OSStatus AudioIOProc( void *inRefCon,
                               AudioUnitRenderActionFlags *ioActionFlags,
                               const AudioTimeStamp *inTimeStamp,
                               UInt32 inBusNumber,
                               UInt32 inNumberFrames,
                               AudioBufferList *ioData )
{
   unsigned long framesProcessed     = 0;
   PaStreamCallbackTimeInfo timeInfo = {0,0,0};
   PaMacCoreStream *stream           = (PaMacCoreStream*)inRefCon;
   const bool isRender               = inBusNumber == OUTPUT_ELEMENT;
   int callbackResult                = paContinue ;

   VVDBUG(("AudioIOProc()\n"));

   PaUtil_BeginCpuLoadMeasurement( &stream->cpuLoadMeasurer );

   /* -----------------------------------------------------------------*\
      This output may be useful for debugging,
      But printing durring the callback is a bad enough idea that
      this is not enabled by enableing the usual debugging calls.
   \* -----------------------------------------------------------------*/
   /*
   static int renderCount = 0;
   static int inputCount = 0;
   printf( "-------------------  starting reder/input\n" );
   if( isRender )
      printf("Render callback (%d):\t", ++renderCount);
   else
      printf("Input callback  (%d):\t", ++inputCount);
   printf( "Call totals: %d (input), %d (render)\n", inputCount, renderCount );

   printf( "--- inBusNumber: %lu\n", inBusNumber );
   printf( "--- inNumberFrames: %lu\n", inNumberFrames );
   printf( "--- %x ioData\n", (unsigned) ioData );
   if( ioData )
   {
      int i=0;
      printf( "--- ioData.mNumBuffers %lu: \n", ioData->mNumberBuffers );
      for( i=0; i<ioData->mNumberBuffers; ++i )
         printf( "--- ioData buffer %d size: %lu.\n", i, ioData->mBuffers[i].mDataByteSize );
   }
      ----------------------------------------------------------------- */

   if( !stream->isTimeSet )
      setStreamStartTime( stream );

   if( isRender ) {
      AudioTimeStamp currentTime;
      timeInfo.outputBufferDacTime = TimeStampToSecs(stream, inTimeStamp);
      AudioDeviceGetCurrentTime(stream->outputDevice, &currentTime);
      timeInfo.currentTime = TimeStampToSecs(stream, &currentTime);
   }
   if( isRender && stream->inputUnit == stream->outputUnit )
      timeInfo.inputBufferAdcTime = TimeStampToSecs(stream, inTimeStamp);
   if( !isRender ) {
      AudioTimeStamp currentTime;
      timeInfo.inputBufferAdcTime = TimeStampToSecs(stream, inTimeStamp);
      AudioDeviceGetCurrentTime(stream->inputDevice, &currentTime);
      timeInfo.currentTime = TimeStampToSecs(stream, &currentTime);
   }

   //printf( "---%g, %g, %g\n", timeInfo.inputBufferAdcTime, timeInfo.currentTime, timeInfo.outputBufferDacTime );

   if( isRender && stream->inputUnit == stream->outputUnit
                && !stream->inputSRConverter )
   {
      /* --------- Full Duplex, One Device, no SR Conversion -------
       *
       * This is the lowest latency case, and also the simplest.
       * Input data and output data are available at the same time.
       * we do not use the input SR converter or the input ring buffer.
       *
       */
      OSStatus err = 0;
      unsigned long frames;

      /* -- start processing -- */
      PaUtil_BeginBufferProcessing( &(stream->bufferProcessor),
                                    &timeInfo,
                                    stream->xrunFlags );
      stream->xrunFlags = 0; //FIXME: this flag also gets set outside by a callback, which calls the xrunCallback function. It should be in the same thread as the main audio callback, but the apple docs just use the word "usually" so it may be possible to loose an xrun notification, if that callback happens here.

      /* -- compute frames. do some checks -- */
      assert( ioData->mNumberBuffers == 1 );
      assert( ioData->mBuffers[0].mNumberChannels == stream->userOutChan );
      frames = ioData->mBuffers[0].mDataByteSize;
      frames /= sizeof( float ) * ioData->mBuffers[0].mNumberChannels;
      /* -- copy and process input data -- */
      err= AudioUnitRender(stream->inputUnit,
                    ioActionFlags,
                    inTimeStamp,
                    INPUT_ELEMENT,
                    inNumberFrames,
                    &stream->inputAudioBufferList );
      /* FEEDBACK: I'm not sure what to do when this call fails */
      assert( !err );

      PaUtil_SetInputFrameCount( &(stream->bufferProcessor), frames );
      PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor),
                          0,
                          stream->inputAudioBufferList.mBuffers[0].mData,
                          stream->inputAudioBufferList.mBuffers[0].mNumberChannels);
      /* -- Copy and process output data -- */
      PaUtil_SetOutputFrameCount( &(stream->bufferProcessor), frames );
      PaUtil_SetInterleavedOutputChannels( &(stream->bufferProcessor),
                                        0,
                                        ioData->mBuffers[0].mData,
                                        ioData->mBuffers[0].mNumberChannels);
      /* -- complete processing -- */
      framesProcessed =
                 PaUtil_EndBufferProcessing( &(stream->bufferProcessor),
                                             &callbackResult );
   }
   else if( isRender )
   {
      /* -------- Output Side of Full Duplex (Separate Devices or SR Conversion)
       *       -- OR Simplex Output
       *
       * This case handles output data as in the full duplex case,
       * and, if there is input data, reads it off the ring buffer
       * and into the PA buffer processor. If sample rate conversion
       * is required on input, that is done here as well.
       */
      unsigned long frames;

      /* Sometimes, when stopping a duplex stream we get erroneous
         xrun flags, so if this is our last run, clear the flags. */
      int xrunFlags = stream->xrunFlags;
/*
      if( xrunFlags & paInputUnderflow )
         printf( "input underflow.\n" );
      if( xrunFlags & paInputOverflow )
         printf( "input overflow.\n" );
*/
      if( stream->state == STOPPING || stream->state == CALLBACK_STOPPED )
         xrunFlags = 0;

      /* -- start processing -- */
      PaUtil_BeginBufferProcessing( &(stream->bufferProcessor),
                                    &timeInfo,
                                    xrunFlags );
      stream->xrunFlags = 0; /* FEEDBACK: we only send flags to Buf Proc once */

      /* -- Copy and process output data -- */
      assert( ioData->mNumberBuffers == 1 );
      frames = ioData->mBuffers[0].mDataByteSize;
      frames /= sizeof( float ) * ioData->mBuffers[0].mNumberChannels;
      assert( ioData->mBuffers[0].mNumberChannels == stream->userOutChan );
      PaUtil_SetOutputFrameCount( &(stream->bufferProcessor), frames );
      PaUtil_SetInterleavedOutputChannels( &(stream->bufferProcessor),
                                     0,
                                     ioData->mBuffers[0].mData,
                                     ioData->mBuffers[0].mNumberChannels);

      /* -- copy and process input data, and complete processing -- */
      if( stream->inputUnit ) {
         const int flsz = sizeof( float );
         /* Here, we read the data out of the ring buffer, through the
            audio converter. */
         int inChan = stream->inputAudioBufferList.mBuffers[0].mNumberChannels;
         if( stream->inputSRConverter )
         {
               OSStatus err;
               UInt32 size;
               float data[ inChan * frames ];
               size = sizeof( data );
               err = AudioConverterFillBuffer(
                             stream->inputSRConverter,
                             ringBufferIOProc,
                             &stream->inputRingBuffer,
                             &size,
                             (void *)&data );
               if( err == RING_BUFFER_EMPTY )
               { /*the ring buffer callback underflowed */
                  err = 0;
                  bzero( ((char *)data) + size, sizeof(data)-size );
                  stream->xrunFlags |= paInputUnderflow;
               }
               ERR( err );
               assert( !err );

               PaUtil_SetInputFrameCount( &(stream->bufferProcessor), frames );
               PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor),
                                   0,
                                   data,
                                   inChan );
               framesProcessed =
                    PaUtil_EndBufferProcessing( &(stream->bufferProcessor),
                                                &callbackResult );
         }
         else
         {
            /* Without the AudioConverter is actually a bit more complex
               because we have to do a little buffer processing that the
               AudioConverter would otherwise handle for us. */
            void *data1, *data2;
            long size1, size2;
            PaUtil_GetRingBufferReadRegions( &stream->inputRingBuffer,
                                             inChan*frames*flsz,
                                             &data1, &size1,
                                             &data2, &size2 );
            if( size1 / ( flsz * inChan ) == frames ) {
               /* simplest case: all in first buffer */
               PaUtil_SetInputFrameCount( &(stream->bufferProcessor), frames );
               PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor),
                                   0,
                                   data1,
                                   inChan );
               framesProcessed =
                    PaUtil_EndBufferProcessing( &(stream->bufferProcessor),
                                                &callbackResult );
               PaUtil_AdvanceRingBufferReadIndex(&stream->inputRingBuffer, size1 );
            } else if( ( size1 + size2 ) / ( flsz * inChan ) < frames ) {
               /*we underflowed. take what data we can, zero the rest.*/
               unsigned char data[frames*inChan*flsz];
               if( size1 )
                  memcpy( data, data1, size1 );
               if( size2 )
                  memcpy( data+size1, data2, size2 );
               bzero( data+size1+size2, frames*flsz*inChan - size1 - size2 );

               PaUtil_SetInputFrameCount( &(stream->bufferProcessor), frames );
               PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor),
                                   0,
                                   data,
                                   inChan );
               framesProcessed =
                    PaUtil_EndBufferProcessing( &(stream->bufferProcessor),
                                                &callbackResult );
               PaUtil_AdvanceRingBufferReadIndex( &stream->inputRingBuffer,
                                                  size1+size2 );
               /* flag underflow */
               stream->xrunFlags |= paInputUnderflow;
            } else {
               /*we got all the data, but split between buffers*/
               PaUtil_SetInputFrameCount( &(stream->bufferProcessor),
                                          size1 / ( flsz * inChan ) );
               PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor),
                                   0,
                                   data1,
                                   inChan );
               PaUtil_Set2ndInputFrameCount( &(stream->bufferProcessor),
                                             size2 / ( flsz * inChan ) );
               PaUtil_Set2ndInterleavedInputChannels( &(stream->bufferProcessor),
                                   0,
                                   data2,
                                   inChan );
               framesProcessed =
                    PaUtil_EndBufferProcessing( &(stream->bufferProcessor),
                                                &callbackResult );
               PaUtil_AdvanceRingBufferReadIndex(&stream->inputRingBuffer, size1+size2 );
            }
         }
      } else {
         framesProcessed =
                 PaUtil_EndBufferProcessing( &(stream->bufferProcessor),
                                             &callbackResult );
      }

   }
   else
   {
      /* ------------------ Input
       *
       * First, we read off the audio data and put it in the ring buffer.
       * if this is an input-only stream, we need to process it more,
       * otherwise, we let the output case deal with it.
       */
      OSStatus err = 0;
      int chan = stream->inputAudioBufferList.mBuffers[0].mNumberChannels ;
      /* FIXME: looping here may not actually be necessary, but it was something I tried in testing. */
      do {
         err= AudioUnitRender(stream->inputUnit,
                 ioActionFlags,
                 inTimeStamp,
                 INPUT_ELEMENT,
                 inNumberFrames,
                 &stream->inputAudioBufferList );
         if( err == -10874 )
            inNumberFrames /= 2;
      } while( err == -10874 && inNumberFrames > 1 );
      /* FEEDBACK: I'm not sure what to do when this call fails */
      ERR( err );
      assert( !err );
      if( stream->inputSRConverter || stream->outputUnit )
      {
         /* If this is duplex or we use a converter, put the data
            into the ring buffer. */
         long bytesIn, bytesOut;
         bytesIn = sizeof( float ) * inNumberFrames * chan;
         bytesOut = PaUtil_WriteRingBuffer( &stream->inputRingBuffer,
                                            stream->inputAudioBufferList.mBuffers[0].mData,
                                            bytesIn );
         if( bytesIn != bytesOut )
            stream->xrunFlags |= paInputOverflow ;
      }
      else
      {
         /* for simplex input w/o SR conversion,
            just pop the data into the buffer processor.*/
         PaUtil_BeginBufferProcessing( &(stream->bufferProcessor),
                              &timeInfo,
                              stream->xrunFlags );
         stream->xrunFlags = 0;

         PaUtil_SetInputFrameCount( &(stream->bufferProcessor), inNumberFrames);
         PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor),
                             0,
                             stream->inputAudioBufferList.mBuffers[0].mData,
                             chan );
         framesProcessed =
              PaUtil_EndBufferProcessing( &(stream->bufferProcessor),
                                          &callbackResult );
      }
      if( !stream->outputUnit && stream->inputSRConverter )
      {
         /* ------------------ Simplex Input w/ SR Conversion
          *
          * if this is a simplex input stream, we need to read off the buffer,
          * do our sample rate conversion and pass the results to the buffer
          * processor.
          * The logic here is complicated somewhat by the fact that we don't
          * know how much data is available, so we loop on reasonably sized
          * chunks, and let the BufferProcessor deal with the rest.
          *
          */
         /*This might be too big or small depending on SR conversion*/
         float data[ chan * inNumberFrames ];
         OSStatus err;
         do
         { /*Run the buffer processor until we are out of data*/
            UInt32 size;
            long f;

            size = sizeof( data );
            err = AudioConverterFillBuffer(
                          stream->inputSRConverter,
                          ringBufferIOProc,
                          &stream->inputRingBuffer,
                          &size,
                          (void *)data );
            if( err != RING_BUFFER_EMPTY )
               ERR( err );
            assert( err == 0 || err == RING_BUFFER_EMPTY );

            f = size / ( chan * sizeof(float) );
            PaUtil_SetInputFrameCount( &(stream->bufferProcessor), f );
            if( f )
            {
               PaUtil_BeginBufferProcessing( &(stream->bufferProcessor),
                                             &timeInfo,
                                             stream->xrunFlags );
               stream->xrunFlags = 0;

               PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor),
                                0,
                                data,
                                chan );
               framesProcessed =
                    PaUtil_EndBufferProcessing( &(stream->bufferProcessor),
                                                &callbackResult );
            }
         } while( callbackResult == paContinue && !err );
      }
   }

   switch( callbackResult )
   {
   case paContinue: break;
   case paComplete:
   case paAbort:
      stream->isTimeSet = FALSE;
      stream->state = CALLBACK_STOPPED ;
      if( stream->outputUnit )
         AudioOutputUnitStop(stream->outputUnit);
      if( stream->inputUnit )
         AudioOutputUnitStop(stream->inputUnit);
      break;
   }

   PaUtil_EndCpuLoadMeasurement( &stream->cpuLoadMeasurer, framesProcessed );
   return noErr;
}

/*
    When CloseStream() is called, the multi-api layer ensures that
    the stream has already been stopped or aborted.
*/
static PaError CloseStream( PaStream* s )
{
    /* This may be called from a failed OpenStream.
       Therefore, each piece of info is treated seperately. */
    PaError result = paNoError;
    PaMacCoreStream *stream = (PaMacCoreStream*)s;

    VVDBUG(("CloseStream()\n"));
    VDBUG( ( "Closing stream.\n" ) );

    if( stream ) {
       if( stream->outputUnit )
          AudioDeviceRemovePropertyListener( stream->outputDevice,
                                             0,
                                             false,
                                             kAudioDeviceProcessorOverload,
                                             xrunCallback );
       if( stream->inputUnit && stream->outputUnit != stream->inputUnit )
          AudioDeviceRemovePropertyListener( stream->inputDevice,
                                             0,
                                             true,
                                             kAudioDeviceProcessorOverload,
                                             xrunCallback );
       if( stream->outputUnit && stream->outputUnit != stream->inputUnit ) {
          AudioUnitUninitialize( stream->outputUnit );
          CloseComponent( stream->outputUnit );
       }
       stream->outputUnit = NULL;
       if( stream->inputUnit )
       {
          AudioUnitUninitialize( stream->inputUnit );
          CloseComponent( stream->inputUnit );
          stream->inputUnit = NULL;
       }
       if( stream->inputRingBuffer.buffer )
          free( (void *) stream->inputRingBuffer.buffer );
       stream->inputRingBuffer.buffer = NULL;
       /*TODO: is there more that needs to be done on error
               from AudioConverterDispose?*/
       if( stream->inputSRConverter )
          ERR( AudioConverterDispose( stream->inputSRConverter ) );
       stream->inputSRConverter = NULL;
       if( stream->inputAudioBufferList.mBuffers[0].mData )
          free( stream->inputAudioBufferList.mBuffers[0].mData );
       stream->inputAudioBufferList.mBuffers[0].mData = NULL;

       result = destroyBlioRingBuffers( &stream->blio );
       if( result )
          return result;
       if( stream->bufferProcessorIsInitialized )
          PaUtil_TerminateBufferProcessor( &stream->bufferProcessor );
       PaUtil_TerminateStreamRepresentation( &stream->streamRepresentation );
       PaUtil_FreeMemory( stream );
    }

    return result;
}

static PaError StartStream( PaStream *s )
{
    PaMacCoreStream *stream = (PaMacCoreStream*)s;
    OSStatus result = noErr;
    VVDBUG(("StartStream()\n"));
    VDBUG( ( "Starting stream.\n" ) );

#define ERR_WRAP(mac_err) do { result = mac_err ; if ( result != noErr ) return ERR(result) ; } while(0)

    /*FIXME: maybe want to do this on close/abort for faster start? */
    PaUtil_ResetBufferProcessor( &stream->bufferProcessor );
    if(  stream->inputSRConverter )
       ERR_WRAP( AudioConverterReset( stream->inputSRConverter ) );

    /* -- start -- */
    stream->state = ACTIVE;
    if( stream->inputUnit ) {
       ERR_WRAP( AudioOutputUnitStart(stream->inputUnit) );
    }
    if( stream->outputUnit && stream->outputUnit != stream->inputUnit ) {
       ERR_WRAP( AudioOutputUnitStart(stream->outputUnit) );
    }

    //setStreamStartTime( stream );
    //stream->isTimeSet = TRUE;

    return paNoError;
#undef ERR_WRAP
}

// it's not clear from appl's docs that this really waits
// until all data is flushed.
static ComponentResult BlockWhileAudioUnitIsRunning( AudioUnit audioUnit, AudioUnitElement element )
{
    Boolean isRunning = 1;
    while( isRunning ) {
       UInt32 s = sizeof( isRunning );
       ComponentResult err = AudioUnitGetProperty( audioUnit, kAudioOutputUnitProperty_IsRunning, kAudioUnitScope_Global, element,  &isRunning, &s );
       if( err )
          return err;
       Pa_Sleep( 100 );
    }
    return noErr;
}

static PaError StopStream( PaStream *s )
{
    PaMacCoreStream *stream = (PaMacCoreStream*)s;
    OSStatus result = noErr;
    PaError paErr;
    VVDBUG(("StopStream()\n"));

    VDBUG( ("Waiting for BLIO.\n") );
    waitUntilBlioWriteBufferIsFlushed( &stream->blio );
    VDBUG( ( "Stopping stream.\n" ) );

    stream->isTimeSet = FALSE;
    stream->state = STOPPING;

#define ERR_WRAP(mac_err) do { result = mac_err ; if ( result != noErr ) return ERR(result) ; } while(0)
    /* -- stop and reset -- */
    if( stream->inputUnit == stream->outputUnit && stream->inputUnit )
    {
       ERR_WRAP( AudioOutputUnitStop(stream->inputUnit) );
       ERR_WRAP( BlockWhileAudioUnitIsRunning(stream->inputUnit,0) );
       ERR_WRAP( BlockWhileAudioUnitIsRunning(stream->inputUnit,1) );
       ERR_WRAP( AudioUnitReset(stream->inputUnit, kAudioUnitScope_Global, 1) );
       ERR_WRAP( AudioUnitReset(stream->inputUnit, kAudioUnitScope_Global, 0) );
    }
    else
    {
       if( stream->inputUnit )
       {
          ERR_WRAP(AudioOutputUnitStop(stream->inputUnit) );
          ERR_WRAP( BlockWhileAudioUnitIsRunning(stream->inputUnit,1) );
          ERR_WRAP(AudioUnitReset(stream->inputUnit,kAudioUnitScope_Global,1));
       }
       if( stream->outputUnit )
       {
          ERR_WRAP(AudioOutputUnitStop(stream->outputUnit));
          ERR_WRAP( BlockWhileAudioUnitIsRunning(stream->outputUnit,0) );
          ERR_WRAP(AudioUnitReset(stream->outputUnit,kAudioUnitScope_Global,0));
       }
    }
    if( stream->inputRingBuffer.buffer ) {
       PaUtil_FlushRingBuffer( &stream->inputRingBuffer );
       bzero( (void *)stream->inputRingBuffer.buffer,
              stream->inputRingBuffer.bufferSize );
       /* advance the write point a little, so we are reading from the
          middle of the buffer. We'll need extra at the end because
          testing has shown that this helps. */
       if( stream->outputUnit )
          PaUtil_AdvanceRingBufferWriteIndex( &stream->inputRingBuffer,
                                              stream->inputRingBuffer.bufferSize
                                              / RING_BUFFER_ADVANCE_DENOMINATOR );
    }

    stream->xrunFlags = 0;
    stream->state = STOPPED;

    paErr = resetBlioRingBuffers( &stream->blio );
    if( paErr )
       return paErr;

/*
    //stream->isTimeSet = FALSE;
*/

    VDBUG( ( "Stream Stopped.\n" ) );
    return paNoError;
#undef ERR_WRAP
}

static PaError AbortStream( PaStream *s )
{
    VVDBUG(("AbortStream()->StopStream()\n"));
    VDBUG( ( "Aborting stream.\n" ) );
    /* We have nothing faster than StopStream. */
    return StopStream(s);
}

static PaError IsStreamStopped( PaStream *s )
{
    PaMacCoreStream *stream = (PaMacCoreStream*)s;
    VVDBUG(("IsStreamStopped()\n"));

    return stream->state == STOPPED ? 1 : 0;
}

static PaError IsStreamActive( PaStream *s )
{
    PaMacCoreStream *stream = (PaMacCoreStream*)s;
    VVDBUG(("IsStreamActive()\n"));
    return ( stream->state == ACTIVE || stream->state == STOPPING );
}

static double GetStreamCpuLoad( PaStream* s )
{
    PaMacCoreStream *stream = (PaMacCoreStream*)s;
    VVDBUG(("GetStreamCpuLoad()\n"));

    return PaUtil_GetCpuLoad( &stream->cpuLoadMeasurer );
}


text:
Enter Gordon Brown Who Reminds me of …
I have mixed feelings about the Prime Minister to be. I was put off the first time I ever saw him by how much he reminded me physically of Richard Nixon – the jowly face, the body language, even the way he had of speaking. That was a long time ago, but makeovers later and the Nixonian aura still persists for me
Gordon Brown is a hard fellow to warm to, as was the disgraced President. But one must try to give the man a chance. But I do think he's a bit of a "strange" one, though that does not mean he won't be a great leader. I wish him well and I hope we find him to be a leader of wisdom and depth who can rise to the challenge of a country suspicious of its leadership and worried about the future and the direction we seem to be heading.
It is a strange system that can simply change Prime Ministers at the whim, or not, of one man. Doesn't seem very democratic. I really do wonder if Gordon Brown is 'electable' without becoming Prime Minister first. The United Kingdom, like the United States does not elect it's political leader directly. The United Kingdom is, however, electing a government. Americans really don't think that way. First of all, it's complicated because the Head of State in the UK is a sovereign born to the role for life. The closest we have to a Prime Minister is the Speaker of the House of Representatives and the Majority Leader of the Senate.
It is very valuable for Americans to study how politics in the UK works. It explains why our own government is the way it is. Our government was created to correct what were deemed to be the faults of Parliamentary Sovereign Britain. The most evident method of doing this was the system we have of 'checks and balances'. This is an important concept to have developed, but there is a price to pay – government stagnation.
For example, in Britain, the government of the day can raise taxes within hours! (Un-American!!) In Britain the government of the day is elected for 5 years – in the U.S. the government can change dramatically every two years. The Queen has an interesting way of putting things to a new Prime Minister when she asks him if he or she can 'form a government'. In Britain 'government' means do you have enough of a consensus to do the job? A Government is 'of the day'.
I love the concept here of the 'loyal opposition'. For every 'government' there is a 'shadow government' and every minister in the government must stand up and defend the policies of the day and is answerable directly to his 'shadow minister' in particular and must also be able to respond to questions from other honourable parliamentarians. The most striking thing to me about Parliament is its intimacy and almost informality amidst all the tradition and politeness of the 'honourable friends'.
Most Americans believe that ours is the greatest democracy in the world and our Constitution is the best in the world. I don't believe this – other countries do have democracies that can rival ours. Somehow all the safeguards that the founding fathers came up with didn't stop big government. It is a fact that when the Constitution was written , it was assumed there would be Constitutional Conventions held that would bring forth Constitutional reform – when the amendments to the Constitution are longer than the original document, it seems to me perhaps it is about time to start over again.
But I cannot in my wildest imagination come up with who we find to perform such a task and how we would ever be able to get 50 states to agree to its adoption. And so comes some more wisdom from the British! They have an 'Unwritten Constitution'. I used to joke about this as an instance of British arrogance. Now I think it a mark of genius!
Labels: democracy, Gordon Brown, government, Politics

redpajama_set_name: RedPajamaCommonCrawl


Andrew's been eating at this one particular Polish Restaurant every day since he's been visiting NY the past few weeks. We all had to see what all the fuss was about. Pretty awesome Polish food. I got the White Borscht and the Hungarian Potato Pie. That was a beast. I was so stuffed. It was truly a Happy End.


Regular readers might have noticed posts to this site have been sporadic over the past couple of months. I've been quite busy finishing up this semester's research projects along with preparing for the big move and haven't had time to write new content. Today, I knock the dust off of the blog with few bits of news I'm excited to share.
First, I'm happy to report the move is mostly done and the family and I are now Missouri residents. Last Saturday the moving truck was loaded and we made the 40-hour trip across the country over the span of four days. We have greatly been looking forward to this new chapter of life and anticipate God doing spectacular things in the coming years.
For those of you that don't know, we moved to Missouri to be part of The Crossings Church – a growing, healthy congregation planted in Wentzville, MO (a suburb of St. Louis) by good friends of ours a few years ago. Those who know Airiel and I know we are passionate about seeing people come to know Jesus and maturing in Him (i.e. making and developing disciples). That same passion is part of the DNA of The Crossings, and I encourage you to visit this congregation and spend a bit of time talking with the people making it up. You will hear story after story of God redeeming broken people and transforming them into the hands and feet of Jesus – the fruit of disciple-making!
This atmosphere is very attractive to Airiel and I, and it's because of it that we turned down other work offers choosing instead to move here. We hope to use our gifts and talents however we can to eventually help spread this type of community to another place in the form of a church plant. For now, we will become observers and learners in order to soak up as much knowledge as we can from our friends here, and I am 100% positive this time will be formative.
Second, I'm approaching the completion of my first Masters Degree from Fuller Seminary (and by "first" I mean "last", but first sounds much more impressive). While on the west coast I was able to complete the majority of the required courses and now have just seven more to finish online. I will be taking most of those this summer and fall, and plan to spend many work days doing something I enjoy most of the time – studying theology.
I've also been investigating doctoral programs, and, while I'm pretty sure a terminal degree is something I will eventually tackle, I'm not quite settled on when or where I should enroll. After looking into a number of institutions to study through, I'm currently leaning toward Covenant Seminary simply because 1) it's near St Louis, 2) it's reputable, and 3) it would probably be the most convenient option. We will see.
I'm hoping to also make a presentation at some point in the fall to the Downtown Church of Christ in Searcy, AR (my sponsoring congregation) to inform everyone about the new work, but have not yet gotten confirmation on when exactly that's going to happen or what the format will be. We will see.
That's all for now – thanks for reading.
Congrats on the big move Wes and family. God bless y'all as y'all do this amazing work.