path: root/gr-audio-osx/src/audio_osx_source.cc

/* -*- c++ -*- */
/*
 * Copyright 2006 Free Software Foundation, Inc.
 * 
 * This file is part of GNU Radio.
 *
 * GNU Radio is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 3, or (at your option)
 * any later version.
 * 
 * GNU Radio is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 * 
 * You should have received a copy of the GNU General Public License
 * along with GNU Radio; see the file COPYING.  If not, write to
 * the Free Software Foundation, Inc., 51 Franklin Street,
 * Boston, MA 02110-1301, USA.
 */

#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

#define _USE_OMNI_THREADS_

#include <audio_osx_source.h>
#include <gr_io_signature.h>
#include <stdexcept>
#include <audio_osx.h>

#define _OSX_AU_DEBUG_ 0
#define _OSX_DO_LISTENERS_ 0

void PrintStreamDesc (AudioStreamBasicDescription *inDesc)
{
  if (inDesc == NULL) {
    fprintf (stderr, "PrintStreamDesc: Can't print a NULL desc!\n");
    return;
  }

  fprintf (stderr, "  Sample Rate        : %g\n", inDesc->mSampleRate);
  fprintf (stderr, "  Format ID          : %4s\n", (char*)&inDesc->mFormatID);
  fprintf (stderr, "  Format Flags       : %lX\n", inDesc->mFormatFlags);
  fprintf (stderr, "  Bytes per Packet   : %ld\n", inDesc->mBytesPerPacket);
  fprintf (stderr, "  Frames per Packet  : %ld\n", inDesc->mFramesPerPacket);
  fprintf (stderr, "  Bytes per Frame    : %ld\n", inDesc->mBytesPerFrame);
  fprintf (stderr, "  Channels per Frame : %ld\n", inDesc->mChannelsPerFrame);
  fprintf (stderr, "  Bits per Channel   : %ld\n", inDesc->mBitsPerChannel);
}

// FIXME these should query some kind of user preference

audio_osx_source::audio_osx_source (int sample_rate,
				    const std::string device_name,
				    bool do_block,
				    int channel_config,
				    int max_sample_count)
  : gr_sync_block ("audio_osx_source",
		   gr_make_io_signature (0, 0, 0),
		   gr_make_io_signature (0, 0, 0)),
    d_deviceSampleRate (0.0), d_outputSampleRate (0.0),
    d_channel_config (0),
    d_inputBufferSizeFrames (0), d_inputBufferSizeBytes (0),
    d_outputBufferSizeFrames (0), d_outputBufferSizeBytes (0),
    d_deviceBufferSizeFrames (0), d_deviceBufferSizeBytes (0),
    d_leadSizeFrames (0), d_leadSizeBytes (0),
    d_trailSizeFrames (0), d_trailSizeBytes (0),
    d_extraBufferSizeFrames (0), d_extraBufferSizeBytes (0),
    d_queueSampleCount (0), d_max_sample_count (0),
    d_n_AvailableInputFrames (0), d_n_ActualInputFrames (0),
    d_n_user_channels (0), d_n_max_channels (0), d_n_deviceChannels (0),
    d_do_block (do_block), d_passThrough (false),
    d_internal (0), d_cond_data (0),
    d_buffers (0),
    d_InputAU (0), d_InputBuffer (0), d_OutputBuffer (0),
    d_AudioConverter (0)
{
  if (sample_rate <= 0) {
    fprintf (stderr, "Invalid Sample Rate: %d\n", sample_rate);
    throw std::invalid_argument ("audio_osx_source::audio_osx_source");
  } else
    d_outputSampleRate = (Float64) sample_rate;

  if (channel_config <= 0 & channel_config != -1) {
    fprintf (stderr, "Invalid Channel Config: %d\n", channel_config);
    throw std::invalid_argument ("audio_osx_source::audio_osx_source");
  } else if (channel_config == -1) {
// no user input; try "device name" instead
    int l_n_channels = (int) strtol (device_name.data(), (char **)NULL, 10);
    if (l_n_channels == 0 & errno) {
      fprintf (stderr, "Error Converting Device Name: %d\n", errno);
      throw std::invalid_argument ("audio_osx_source::audio_osx_source");
    }
    if (l_n_channels <= 0)
      channel_config = 2;
    else
      channel_config = l_n_channels;
  }

  d_channel_config = channel_config;

// check that the max # of samples to store is valid

  if (max_sample_count == -1)
    max_sample_count = sample_rate;
  else if (max_sample_count <= 0) {
    fprintf (stderr, "Invalid Max Sample Count: %d\n", max_sample_count);
    throw std::invalid_argument ("audio_osx_source::audio_osx_source");
  }

  d_max_sample_count = max_sample_count;

#if _OSX_AU_DEBUG_
  fprintf (stderr, "source(): max # samples = %ld", d_max_sample_count);
#endif

  OSStatus err = noErr;

// create the default AudioUnit for input

// Open the default input unit
  ComponentDescription InputDesc;

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

  Component comp = FindNextComponent (NULL, &InputDesc);
  if (comp == NULL) {
    fprintf (stderr, "FindNextComponent Error\n");
    throw std::runtime_error ("audio_osx_source::audio_osx_source");
  }

  err = OpenAComponent (comp, &d_InputAU);
  CheckErrorAndThrow (err, "OpenAComponent",
		      "audio_osx_source::audio_osx_source");

  UInt32 enableIO;

// must enable the AUHAL for input and disable output 
// before setting the AUHAL's current device

// Enable input on the AUHAL
  enableIO = 1;
  err = AudioUnitSetProperty (d_InputAU,
			      kAudioOutputUnitProperty_EnableIO,
			      kAudioUnitScope_Input,
			      1, // input element
			      &enableIO,
			      sizeof (UInt32));
  CheckErrorAndThrow (err, "AudioUnitSetProperty Input Enable",
		      "audio_osx_source::audio_osx_source");

// Disable output on the AUHAL
  enableIO = 0;
  err = AudioUnitSetProperty (d_InputAU,
			      kAudioOutputUnitProperty_EnableIO,
			      kAudioUnitScope_Output,
			      0, // output element
			      &enableIO,
			      sizeof (UInt32));
  CheckErrorAndThrow (err, "AudioUnitSetProperty Output Disable",
		      "audio_osx_source::audio_osx_source");

// set the default input device for our input AU

  SetDefaultInputDeviceAsCurrent ();

#if _OSX_DO_LISTENERS_
// set up a listener if default hardware input device changes

  err = AudioHardwareAddPropertyListener
    (kAudioHardwarePropertyDefaultInputDevice,
     (AudioHardwarePropertyListenerProc) HardwareListener,
     this);

  CheckErrorAndThrow (err, "AudioHardwareAddPropertyListener",
		      "audio_osx_source::audio_osx_source");

// Add a listener for any changes in the input AU's output stream
// the function "UnitListener" will be called if the stream format
// changes for whatever reason

  err = AudioUnitAddPropertyListener
    (d_InputAU,
     kAudioUnitProperty_StreamFormat,
     (AudioUnitPropertyListenerProc) UnitListener,
     this);
  CheckErrorAndThrow (err, "Adding Unit Property Listener",
		      "audio_osx_source::audio_osx_source");
#endif

// Now find out if it actually can do input.

  UInt32 hasInput = 0;
  UInt32 dataSize = sizeof (hasInput);
  err = AudioUnitGetProperty (d_InputAU,
			      kAudioOutputUnitProperty_HasIO,
			      kAudioUnitScope_Input,
			      1,
			      &hasInput,
			      &dataSize);
  CheckErrorAndThrow (err, "AudioUnitGetProperty HasIO",
		      "audio_osx_source::audio_osx_source");
  if (hasInput == 0) {
    fprintf (stderr, "Selected Audio Device does not support Input.\n");
    throw std::runtime_error ("audio_osx_source::audio_osx_source");
  }

// Set up a callback function to retrieve input from the Audio Device

  AURenderCallbackStruct AUCallBack;

  AUCallBack.inputProc = (AURenderCallback)(audio_osx_source::AUInputCallback);
  AUCallBack.inputProcRefCon = this;

  err = AudioUnitSetProperty (d_InputAU,
			      kAudioOutputUnitProperty_SetInputCallback,
			      kAudioUnitScope_Global,
			      0,
			      &AUCallBack,
			      sizeof (AURenderCallbackStruct));
  CheckErrorAndThrow (err, "AudioUnitSetProperty Input Callback",
		      "audio_osx_source::audio_osx_source");

  UInt32 propertySize;
  AudioStreamBasicDescription asbd_device, asbd_client, asbd_user;

// asbd_device: ASBD of the device that is creating the input data stream
// asbd_client: ASBD of the client size (output) of the hardware device
// asbd_user:   ASBD of the user's arguments

// Get the Stream Format (device side)

  propertySize = sizeof (asbd_device);
  err = AudioUnitGetProperty (d_InputAU,
			      kAudioUnitProperty_StreamFormat,
			      kAudioUnitScope_Input,
			      1,
			      &asbd_device,
			      &propertySize);
  CheckErrorAndThrow (err, "AudioUnitGetProperty Device Input Stream Format",
		      "audio_osx_source::audio_osx_source");

#if _OSX_AU_DEBUG_
  fprintf (stderr, "\n---- Device Stream Format ----\n" );
  PrintStreamDesc (&asbd_device);
#endif

// Get the Stream Format (client side)
  propertySize = sizeof (asbd_client);
  err = AudioUnitGetProperty (d_InputAU,
			      kAudioUnitProperty_StreamFormat,
			      kAudioUnitScope_Output,
			      1,
			      &asbd_client,
			      &propertySize);
  CheckErrorAndThrow (err, "AudioUnitGetProperty Device Ouput Stream Format",
		      "audio_osx_source::audio_osx_source");

#if _OSX_AU_DEBUG_
  fprintf (stderr, "\n---- Client Stream Format ----\n");
  PrintStreamDesc (&asbd_client);
#endif

// Set the format of all the AUs to the input/output devices channel count

// get the max number of input (& thus output) channels supported by
// this device
  d_n_max_channels = asbd_client.mChannelsPerFrame;

// create the output io signature;
// no input siganture to set (source is hardware)
  set_output_signature (gr_make_io_signature (1,
					      d_n_max_channels,
					      sizeof (float)));

// allocate the output circular buffer(s), one per channel
  d_buffers = (circular_buffer<float>**) new
    circular_buffer<float>* [d_n_max_channels];
  UInt32 n_alloc = (UInt32) ceil ((double) d_max_sample_count);
  for (UInt32 n = 0; n < d_n_max_channels; n++) {
    d_buffers[n] = new circular_buffer<float> (n_alloc, false, false);
  }

  d_deviceSampleRate = asbd_device.mSampleRate;
  d_n_deviceChannels = asbd_device.mChannelsPerFrame;

// create an ASBD for the user's wants

  asbd_user.mSampleRate = d_outputSampleRate;
  asbd_user.mFormatID = kAudioFormatLinearPCM;
  asbd_user.mFormatFlags = (kLinearPCMFormatFlagIsFloat |
			    GR_PCM_ENDIANNESS |
			    kLinearPCMFormatFlagIsPacked |
			    kAudioFormatFlagIsNonInterleaved);
  asbd_user.mBytesPerPacket = 4;
  asbd_user.mFramesPerPacket = 1;
  asbd_user.mBytesPerFrame = 4;
  asbd_user.mChannelsPerFrame = d_n_max_channels;
  asbd_user.mBitsPerChannel = 32;

  if (d_deviceSampleRate == d_outputSampleRate) {
// no need to do conversion if asbd_client matches user wants
    d_passThrough = true;
    d_leadSizeFrames = d_trailSizeFrames = 0L;
  } else {
    d_passThrough = false;
// Create the audio converter

    err = AudioConverterNew (&asbd_client, &asbd_user, &d_AudioConverter);
    CheckErrorAndThrow (err, "AudioConverterNew",
			"audio_osx_source::audio_osx_source");

// Set the audio converter sample rate quality to "max" ...
// requires more samples, but should sound nicer

    UInt32 ACQuality = kAudioConverterQuality_Max;
    propertySize = sizeof (ACQuality);
    err = AudioConverterSetProperty (d_AudioConverter,
				     kAudioConverterSampleRateConverterQuality,
				     propertySize,
				     &ACQuality);
    CheckErrorAndThrow (err, "AudioConverterSetProperty "
			"SampleRateConverterQuality",
			"audio_osx_source::audio_osx_source");

// set the audio converter's prime method to "pre",
// which uses both leading and trailing frames
// from the "current input".  All of this is handled
// internally by the AudioConverter; we just supply
// the frames for conversion.

//   UInt32 ACPrimeMethod = kConverterPrimeMethod_None;
    UInt32 ACPrimeMethod = kConverterPrimeMethod_Pre;
    propertySize = sizeof (ACPrimeMethod);
    err = AudioConverterSetProperty (d_AudioConverter, 
				     kAudioConverterPrimeMethod,
				     propertySize,
				     &ACPrimeMethod);
    CheckErrorAndThrow (err, "AudioConverterSetProperty PrimeMethod",
			"audio_osx_source::audio_osx_source");

// Get the size of the I/O buffer(s) to allow for pre-allocated buffers
      
// lead frame info (trail frame info is ignored)

    AudioConverterPrimeInfo ACPrimeInfo = {0, 0};
    propertySize = sizeof (ACPrimeInfo);
    err = AudioConverterGetProperty (d_AudioConverter, 
				     kAudioConverterPrimeInfo,
				     &propertySize,
				     &ACPrimeInfo);
    CheckErrorAndThrow (err, "AudioConverterGetProperty PrimeInfo",
			"audio_osx_source::audio_osx_source");

    switch (ACPrimeMethod) {
    case (kConverterPrimeMethod_None):
      d_leadSizeFrames =
	d_trailSizeFrames = 0L;
      break;
    case (kConverterPrimeMethod_Normal):
      d_leadSizeFrames = 0L;
      d_trailSizeFrames = ACPrimeInfo.trailingFrames;
      break;
    default:
      d_leadSizeFrames = ACPrimeInfo.leadingFrames;
      d_trailSizeFrames = ACPrimeInfo.trailingFrames;
    }
  }
  d_leadSizeBytes = d_leadSizeFrames * sizeof (Float32);
  d_trailSizeBytes = d_trailSizeFrames * sizeof (Float32);

  propertySize = sizeof (d_deviceBufferSizeFrames);
  err = AudioUnitGetProperty (d_InputAU,
			      kAudioDevicePropertyBufferFrameSize,
			      kAudioUnitScope_Global,
			      0,
			      &d_deviceBufferSizeFrames,
			      &propertySize);
  CheckErrorAndThrow (err, "AudioUnitGetProperty Buffer Frame Size",
		      "audio_osx_source::audio_osx_source");

  d_deviceBufferSizeBytes = d_deviceBufferSizeFrames * sizeof (Float32);
  d_inputBufferSizeBytes = d_deviceBufferSizeBytes + d_leadSizeBytes;
  d_inputBufferSizeFrames = d_deviceBufferSizeFrames + d_leadSizeFrames;

// outBufSizeBytes = floor (inBufSizeBytes * rate_out / rate_in)
// since this is rarely exact, we need another buffer to hold
// "extra" samples not processed at any given sampling period
// this buffer must be at least 4 floats in size, but generally
// follows the rule that
// extraBufSize =  ceil (rate_in / rate_out)*sizeof(float)

  d_extraBufferSizeFrames = ((UInt32) ceil (d_deviceSampleRate
					    / d_outputSampleRate)
			     * sizeof (float));
  if (d_extraBufferSizeFrames < 4)
    d_extraBufferSizeFrames = 4;
  d_extraBufferSizeBytes = d_extraBufferSizeFrames * sizeof (float);

  d_outputBufferSizeFrames = (UInt32) ceil (((Float64) d_inputBufferSizeFrames)
					    * d_outputSampleRate
					    / d_deviceSampleRate);
  d_outputBufferSizeBytes = d_outputBufferSizeFrames * sizeof (float);
  d_inputBufferSizeFrames += d_extraBufferSizeFrames;

// pre-alloc all buffers

  AllocAudioBufferList (&d_InputBuffer, d_n_deviceChannels,
			d_inputBufferSizeBytes);
  if (d_passThrough == false) {
    AllocAudioBufferList (&d_OutputBuffer, d_n_max_channels,
			  d_outputBufferSizeBytes);
  } else {
    d_OutputBuffer = d_InputBuffer;
  }

// create the stuff to regulate I/O

  d_internal = new mld_mutex ();
  if (d_internal == NULL)
    CheckErrorAndThrow (errno, "new mld_mutex (internal)",
			"audio_osx_source::audio_osx_source");

  d_cond_data = new mld_condition ();
  if (d_cond_data == NULL)
    CheckErrorAndThrow (errno, "new mld_condition (data)",
			"audio_osx_source::audio_osx_source");

// initialize the AU for input

  err = AudioUnitInitialize (d_InputAU);
  CheckErrorAndThrow (err, "AudioUnitInitialize",
		      "audio_osx_source::audio_osx_source");

#if _OSX_AU_DEBUG_
  fprintf (stderr, "audio_osx_source Parameters:\n");
  fprintf (stderr, "  Device Sample Rate is %g\n", d_deviceSampleRate);
  fprintf (stderr, "  User Sample Rate is %g\n", d_outputSampleRate);
  fprintf (stderr, "  Max Sample Count is %ld\n", d_max_sample_count);
  fprintf (stderr, "  # Device Channels is %ld\n", d_n_deviceChannels);
  fprintf (stderr, "  # Max Channels is %ld\n", d_n_max_channels);
  fprintf (stderr, "  Device Buffer Size is Frames = %ld\n",
	   d_deviceBufferSizeFrames);
  fprintf (stderr, "  Lead Size is Frames = %ld\n",
	   d_leadSizeFrames);
  fprintf (stderr, "  Trail Size is Frames = %ld\n",
	   d_trailSizeFrames);
  fprintf (stderr, "  Input Buffer Size is Frames = %ld\n",
	   d_inputBufferSizeFrames);
  fprintf (stderr, "  Output Buffer Size is Frames = %ld\n",
	   d_outputBufferSizeFrames);
#endif
}

void
audio_osx_source::AllocAudioBufferList (AudioBufferList** t_ABL,
					UInt32 n_channels,
					UInt32 bufferSizeBytes)
{
  FreeAudioBufferList (t_ABL);
  UInt32 propertySize = (offsetof (AudioBufferList, mBuffers[0]) +
			 (sizeof (AudioBuffer) * n_channels));
  *t_ABL = (AudioBufferList*) calloc (1, propertySize);
  (*t_ABL)->mNumberBuffers = n_channels;

  int counter = n_channels;

  while (--counter >= 0) {
    (*t_ABL)->mBuffers[counter].mNumberChannels = 1;
    (*t_ABL)->mBuffers[counter].mDataByteSize = bufferSizeBytes;
    (*t_ABL)->mBuffers[counter].mData = calloc (1, bufferSizeBytes);
  }
}

void
audio_osx_source::FreeAudioBufferList (AudioBufferList** t_ABL)
{
// free pre-allocated audio buffer, if it exists
  if (*t_ABL != NULL) {
    int counter = (*t_ABL)->mNumberBuffers;
    while (--counter >= 0)
      free ((*t_ABL)->mBuffers[counter].mData);
    free (*t_ABL);
    (*t_ABL) = 0;
  }
}

bool audio_osx_source::IsRunning ()
{
  UInt32 AURunning = 0, AUSize = sizeof (UInt32);

  OSStatus err = AudioUnitGetProperty (d_InputAU,
				       kAudioOutputUnitProperty_IsRunning,
				       kAudioUnitScope_Global,
				       0,
				       &AURunning,
				       &AUSize);
  CheckErrorAndThrow (err, "AudioUnitGetProperty IsRunning",
		      "audio_osx_source::IsRunning");

  return (AURunning);
}

bool audio_osx_source::start ()
{
  if (! IsRunning ()) {
    OSStatus err = AudioOutputUnitStart (d_InputAU);
    CheckErrorAndThrow (err, "AudioOutputUnitStart",
			"audio_osx_source::start");
  }

  return (true);
}

bool audio_osx_source::stop ()
{
  if (IsRunning ()) {
    OSStatus err = AudioOutputUnitStop (d_InputAU);
    CheckErrorAndThrow (err, "AudioOutputUnitStart",
			"audio_osx_source::stop");
    for (UInt32 n = 0; n < d_n_user_channels; n++) {
      d_buffers[n]->abort ();
    }
  }

  return (true);
}

audio_osx_source::~audio_osx_source ()
{
  OSStatus err = noErr;

// stop the AudioUnit
  stop();

#if _OSX_DO_LISTENERS_
// remove the listeners

  err = AudioUnitRemovePropertyListener
    (d_InputAU,
     kAudioUnitProperty_StreamFormat,
     (AudioUnitPropertyListenerProc) UnitListener);
  CheckError (err, "~audio_osx_source: AudioUnitRemovePropertyListener");

  err = AudioHardwareRemovePropertyListener
    (kAudioHardwarePropertyDefaultInputDevice,
     (AudioHardwarePropertyListenerProc) HardwareListener);
  CheckError (err, "~audio_osx_source: AudioHardwareRemovePropertyListener");
#endif

// free pre-allocated audio buffers
  FreeAudioBufferList (&d_InputBuffer);

  if (d_passThrough == false) {
    err = AudioConverterDispose (d_AudioConverter);
    CheckError (err, "~audio_osx_source: AudioConverterDispose");
    FreeAudioBufferList (&d_OutputBuffer);
  }

// remove the audio unit
  err = AudioUnitUninitialize (d_InputAU);
  CheckError (err, "~audio_osx_source: AudioUnitUninitialize");

  err = CloseComponent (d_InputAU);
  CheckError (err, "~audio_osx_source: CloseComponent");

// empty and delete the queues
  for (UInt32 n = 0; n < d_n_max_channels; n++) {
    delete d_buffers[n];
    d_buffers[n] = 0;
  }
  delete [] d_buffers;
  d_buffers = 0;

// close and delete the control stuff
  delete d_internal;
  delete d_cond_data;
}

audio_osx_source_sptr
audio_osx_make_source (int sampling_freq,
		       const std::string device_name,
		       bool do_block,
		       int channel_config,
		       int max_sample_count)
{
  return audio_osx_source_sptr (new audio_osx_source (sampling_freq,
						      device_name,
						      do_block,
						      channel_config,
						      max_sample_count));
}

bool
audio_osx_source::check_topology (int ninputs, int noutputs)
{
// check # inputs to make sure it's valid
  if (ninputs != 0) {
    fprintf (stderr, "audio_osx_source::check_topology(): "
	     "number of input streams provided (%d) should be 0.\n",
	     ninputs);
    throw std::runtime_error ("audio_osx_source::check_topology()");
  }

// check # outputs to make sure it's valid
  if ((noutputs < 1) | (noutputs > (int) d_n_max_channels)) {
    fprintf (stderr, "audio_osx_source::check_topology(): "
	     "number of output streams provided (%d) should be in "
	     "[1,%ld] for the selected audio device.\n",
	     noutputs, d_n_max_channels);
    throw std::runtime_error ("audio_osx_source::check_topology()");
  }

// save the actual number of output (user) channels
  d_n_user_channels = noutputs;

#if _OSX_AU_DEBUG_
  fprintf (stderr, "chk_topo: Actual # user output channels = %d\n",
	   noutputs);
#endif

  return (true);
}

int
audio_osx_source::work (int noutput_items,
			gr_vector_const_void_star &input_items,
			gr_vector_void_star &output_items)
{
// acquire control to do processing here only
  d_internal->wait ();

#if _OSX_AU_DEBUG_
  fprintf (stderr, "work1: SC = %4ld, #OI = %4d, #Chan = %ld\n",
	   d_queueSampleCount, noutput_items, output_items.size());
#endif

// ?: always block until there is something to output from the source
//    or return anything that is available, even if it's less than desired?

  UInt32 actual_noutput_items = noutput_items;

  if (d_queueSampleCount < actual_noutput_items) {
    if (d_queueSampleCount == 0) {
// no data; do_block decides what to do
      if (d_do_block == true) {
	while (d_queueSampleCount == 0) {
// release control so-as to allow data to be retrieved
	  d_internal->post ();
// block until there is data to return
	  d_cond_data->wait ();
// the condition's signal() was called; acquire control
// to keep thread safe
	  d_internal->wait ();
	}
      } else {
// not enough data & not blocking; return nothing
	return (0);
      }
    }
    actual_noutput_items = d_queueSampleCount;
  }

  int l_counter = (int) output_items.size();

// get the items from the circular buffers
  while (--l_counter >= 0) {
    UInt32 t_n_output_items = actual_noutput_items;
    d_buffers[l_counter]->dequeue ((float*) output_items[l_counter],
				   &t_n_output_items);
    if (t_n_output_items != actual_noutput_items) {
      fprintf (stderr, "audio_osx_source::work(): "
	       "number of available items changing "
	       "unexpectedly; expecting %ld, got %ld.\n",
	       actual_noutput_items, t_n_output_items);
      throw std::runtime_error ("audio_osx_source::work()");
    }
  }

  d_queueSampleCount -= actual_noutput_items;

#if _OSX_AU_DEBUG_
  fprintf (stderr, "work2: SC = %4ld, act#OI = %4ld\n",
	   d_queueSampleCount, actual_noutput_items);
#endif

// release control to allow for other processing parts to run
  d_internal->post ();

  return (actual_noutput_items);
}

OSStatus
audio_osx_source::ConverterCallback (AudioConverterRef inAudioConverter,
				     UInt32* ioNumberDataPackets,
				     AudioBufferList* ioData,
				     AudioStreamPacketDescription** ioASPD,
				     void* inUserData)
{
// take current device buffers and copy them to the tail of the input buffers
// the lead buffer is already there in the first d_leadSizeFrames slots

  audio_osx_source* This = static_cast<audio_osx_source*>(inUserData);
  AudioBufferList* l_inputABL = This->d_InputBuffer;
  UInt32 totalInputBufferSizeBytes = ((*ioNumberDataPackets) * sizeof (float));
  int counter = This->d_n_deviceChannels;
  ioData->mNumberBuffers = This->d_n_deviceChannels;
  This->d_n_ActualInputFrames = (*ioNumberDataPackets);

#if _OSX_AU_DEBUG_
  fprintf (stderr, "cc1: io#DP = %ld, TIBSB = %ld, #C = %d\n",
	   *ioNumberDataPackets, totalInputBufferSizeBytes, counter);
#endif

  while (--counter >= 0)  {
    AudioBuffer* l_ioD_AB = &(ioData->mBuffers[counter]);
    l_ioD_AB->mNumberChannels = 1;
    l_ioD_AB->mData = (float*)(l_inputABL->mBuffers[counter].mData);
    l_ioD_AB->mDataByteSize = totalInputBufferSizeBytes;
  }

  return (noErr);
}

OSStatus
audio_osx_source::AUInputCallback (void* inRefCon,
				   AudioUnitRenderActionFlags* ioActionFlags,
				   const AudioTimeStamp* inTimeStamp,
				   UInt32 inBusNumber,
				   UInt32 inNumberFrames,
				   AudioBufferList* ioData)
{
  OSStatus err = noErr;
  audio_osx_source* This = static_cast<audio_osx_source*>(inRefCon);

  This->d_internal->wait ();

#if _OSX_AU_DEBUG_
  fprintf (stderr, "cb0: in#F = %4ld, inBN = %ld, SC = %4ld\n",
	   inNumberFrames, inBusNumber, This->d_queueSampleCount);
#endif

// Get the new audio data from the input device

  err = AudioUnitRender (This->d_InputAU,
			 ioActionFlags,
			 inTimeStamp,
			 1, //inBusNumber,
			 inNumberFrames,
			 This->d_InputBuffer);
  CheckErrorAndThrow (err, "AudioUnitRender",
		      "audio_osx_source::AUInputCallback");

  UInt32 AvailableInputFrames = inNumberFrames;
  This->d_n_AvailableInputFrames = inNumberFrames;

// get the number of actual output frames,
// either via converting the buffer or not

  UInt32 ActualOutputFrames;

  if (This->d_passThrough == true) {
    ActualOutputFrames = AvailableInputFrames;
  } else {
    UInt32 AvailableInputBytes = AvailableInputFrames * sizeof (float);
    UInt32 AvailableOutputBytes = AvailableInputBytes;
    UInt32 AvailableOutputFrames = AvailableOutputBytes / sizeof (float);
    UInt32 propertySize = sizeof (AvailableOutputBytes);
    err = AudioConverterGetProperty (This->d_AudioConverter,
		   kAudioConverterPropertyCalculateOutputBufferSize,
				     &propertySize,
				     &AvailableOutputBytes);
    CheckErrorAndThrow (err, "AudioConverterGetProperty CalculateOutputBufferSize", "audio_osx_source::audio_osx_source");

    AvailableOutputFrames = AvailableOutputBytes / sizeof (float);

#if 0
// when decimating too much, the output sounds warbly due to
// fluctuating # of output frames
// This should not be a surprise, but there's probably some
// clever programming that could lessed the effect ...
// like finding the "ideal" # of output frames, and keeping
// that number constant no matter the # of input frames
    UInt32 l_InputBytes = AvailableOutputBytes;
    propertySize = sizeof (AvailableOutputBytes);
    err = AudioConverterGetProperty (This->d_AudioConverter,
		     kAudioConverterPropertyCalculateInputBufferSize,
				     &propertySize,
				     &l_InputBytes);
    CheckErrorAndThrow (err, "AudioConverterGetProperty CalculateInputBufferSize", "audio_osx_source::audio_osx_source");

    if (l_InputBytes < AvailableInputBytes) {
// OK to zero pad the input a little
      AvailableOutputFrames += 1;
      AvailableOutputBytes = AvailableOutputFrames * sizeof (float);
    }
#endif

#if _OSX_AU_DEBUG_
    fprintf (stderr, "cb1:  avail: #IF = %ld, #OF = %ld\n",
	     AvailableInputFrames, AvailableOutputFrames);
#endif
    ActualOutputFrames = AvailableOutputFrames;

// convert the data to the correct rate
// on input, ActualOutputFrames is the number of available output frames

    err = AudioConverterFillComplexBuffer (This->d_AudioConverter,
	   (AudioConverterComplexInputDataProc)(This->ConverterCallback),
					   inRefCon,
					   &ActualOutputFrames,
					   This->d_OutputBuffer,
					   NULL);
    CheckErrorAndThrow (err, "AudioConverterFillComplexBuffer",
			"audio_osx_source::AUInputCallback");

// on output, ActualOutputFrames is the actual number of output frames

#if _OSX_AU_DEBUG_
    fprintf (stderr, "cb2: actual: #IF = %ld, #OF = %ld\n",
	     This->d_n_ActualInputFrames, AvailableOutputFrames);
    if (This->d_n_ActualInputFrames != AvailableInputFrames)
      fprintf (stderr, "cb2.1: avail#IF = %ld, actual#IF = %ld\n",
	       AvailableInputFrames, This->d_n_ActualInputFrames);
#endif
  }

// add the output frames to the buffers' queue, checking for overflow

  int l_counter = This->d_n_user_channels;
  int res = 0;

  while (--l_counter >= 0) {
    float* inBuffer = (float*) This->d_OutputBuffer->mBuffers[l_counter].mData;

#if _OSX_AU_DEBUG_
  fprintf (stderr, "cb2.5: enqueuing audio data.\n");
#endif

    int l_res = This->d_buffers[l_counter]->enqueue (inBuffer, ActualOutputFrames);
    if (l_res == -1)
      res = -1;
  }

  if (res == -1) {
// data coming in too fast
// drop oldest buffer
    fputs ("aO", stderr);
    fflush (stderr);
// set the local number of samples available to the max
    This->d_queueSampleCount = This->d_buffers[0]->buffer_length_items ();
  } else {
// keep up the local sample count
    This->d_queueSampleCount += ActualOutputFrames;
  }

#if _OSX_AU_DEBUG_
  fprintf (stderr, "cb3: #OI = %4ld, #Cnt = %4ld, mSC = %ld, \n",
	   ActualOutputFrames, This->d_queueSampleCount,
	   This->d_max_sample_count);
#endif

// signal that data is available, if appropraite
  This->d_cond_data->signal ();

#if _OSX_AU_DEBUG_
  fprintf (stderr, "cb4: releasing internal mutex.\n");
#endif

// release control to allow for other processing parts to run
  This->d_internal->post ();

#if _OSX_AU_DEBUG_
  fprintf (stderr, "cb5: returning.\n");
#endif

  return (err);
}

void
audio_osx_source::SetDefaultInputDeviceAsCurrent
()
{
// set the default input device
  AudioDeviceID deviceID;
  UInt32 dataSize = sizeof (AudioDeviceID);
  AudioHardwareGetProperty (kAudioHardwarePropertyDefaultInputDevice,
			    &dataSize,
			    &deviceID);
  OSStatus err = AudioUnitSetProperty (d_InputAU,
				       kAudioOutputUnitProperty_CurrentDevice,
				       kAudioUnitScope_Global,
				       0,
				       &deviceID,
				       sizeof (AudioDeviceID));
  CheckErrorAndThrow (err, "AudioUnitSetProperty Current Device",
		      "audio_osx_source::SetDefaultInputDeviceAsCurrent");
}

#if _OSX_DO_LISTENERS_
OSStatus
audio_osx_source::HardwareListener
(AudioHardwarePropertyID inPropertyID, 
 void *inClientData)
{
  OSStatus err = noErr;
  audio_osx_source* This = static_cast<audio_osx_source*>(inClientData);

  fprintf (stderr, "a_o_s::HardwareListener\n");

// set the new default hardware input device for use by our AU

  This->SetDefaultInputDeviceAsCurrent ();

// reset the converter to tell it that the stream has changed

  err = AudioConverterReset (This->d_AudioConverter);
  CheckErrorAndThrow (err, "AudioConverterReset",
		      "audio_osx_source::UnitListener");

  return (err);
}

OSStatus
audio_osx_source::UnitListener
(void *inRefCon,
 AudioUnit ci,
 AudioUnitPropertyID inID,
 AudioUnitScope inScope,
 AudioUnitElement inElement)
{
  OSStatus err = noErr;
  audio_osx_source* This = static_cast<audio_osx_source*>(inRefCon);
  AudioStreamBasicDescription asbd;			

  fprintf (stderr, "a_o_s::UnitListener\n");

// get the converter's input ASBD (for printing)

  UInt32 propertySize = sizeof (asbd);
  err = AudioConverterGetProperty (This->d_AudioConverter,
				   kAudioConverterCurrentInputStreamDescription,
				   &propertySize,
				   &asbd);
  CheckErrorAndThrow (err, "AudioConverterGetProperty "
		      "CurrentInputStreamDescription",
		      "audio_osx_source::UnitListener");

  fprintf (stderr, "UnitListener: Input Source changed.\n"
	   "Old Source Output Info:\n");
  PrintStreamDesc (&asbd);

// get the new input unit's output ASBD

  propertySize = sizeof (asbd);
  err = AudioUnitGetProperty (This->d_InputAU,
			      kAudioUnitProperty_StreamFormat,
			      kAudioUnitScope_Output, 1,
			      &asbd, &propertySize);
  CheckErrorAndThrow (err, "AudioUnitGetProperty StreamFormat",
		      "audio_osx_source::UnitListener");

  fprintf (stderr, "New Source Output Info:\n");
  PrintStreamDesc (&asbd);

// set the converter's input ASBD to this

  err = AudioConverterSetProperty (This->d_AudioConverter,
				   kAudioConverterCurrentInputStreamDescription,
				   propertySize,
				   &asbd);
  CheckErrorAndThrow (err, "AudioConverterSetProperty "
		      "CurrentInputStreamDescription",
		      "audio_osx_source::UnitListener");

// reset the converter to tell it that the stream has changed

  err = AudioConverterReset (This->d_AudioConverter);
  CheckErrorAndThrow (err, "AudioConverterReset",
		      "audio_osx_source::UnitListener");

  return (err);
}
#endif