audio: wip: converting OSX audio to 3.7 style.

author: Tom Rondeau <trondeau@vt.edu> 2013-02-13 10:58:42 -0500
committer: Tom Rondeau <trondeau@vt.edu> 2013-02-13 10:58:42 -0500
commit: 9888a4d2d12dc874a75c114ba52c9956065ba923 (patch)
tree: 5d7e6972b444d4cae49c633e1943c69d2523392c
parent: 364404cdd1ddf6e9b8c128490f76bc95a384facd (diff)
23 files changed, 2846 insertions, 2768 deletions
diff --git a/gr-audio/lib/CMakeLists.txt b/gr-audio/lib/CMakeLists.txt
index b6e6a1f2db..030f0371a3 100644
--- a/gr-audio/lib/CMakeLists.txt
+++ b/gr-audio/lib/CMakeLists.txt
@@ -105,8 +105,8 @@ if(AUDIO_UNIT_H AND AUDIO_TOOLBOX_H)
         "-framework Carbon"
     )
     list(APPEND gr_audio_sources
-        ${CMAKE_CURRENT_SOURCE_DIR}/osx/audio_osx_source.cc
-        ${CMAKE_CURRENT_SOURCE_DIR}/osx/audio_osx_sink.cc
+        ${CMAKE_CURRENT_SOURCE_DIR}/osx/osx_source.cc
+        ${CMAKE_CURRENT_SOURCE_DIR}/osx/osx_sink.cc
     )
 
 endif(AUDIO_UNIT_H AND AUDIO_TOOLBOX_H)
@@ -122,9 +122,9 @@ if(PORTAUDIO_FOUND)
     list(APPEND gr_audio_libs ${PORTAUDIO_LIBRARIES})
     add_definitions(${PORTAUDIO_DEFINITIONS})
     list(APPEND gr_audio_sources
-        ${CMAKE_CURRENT_SOURCE_DIR}/portaudio/gri_portaudio.cc
-        ${CMAKE_CURRENT_SOURCE_DIR}/portaudio/audio_portaudio_source.cc
-        ${CMAKE_CURRENT_SOURCE_DIR}/portaudio/audio_portaudio_sink.cc
+        ${CMAKE_CURRENT_SOURCE_DIR}/portaudio/portaudio_impl.cc
+        ${CMAKE_CURRENT_SOURCE_DIR}/portaudio/portaudio_source.cc
+        ${CMAKE_CURRENT_SOURCE_DIR}/portaudio/portaudio_sink.cc
     )
     list(APPEND gr_audio_confs ${CMAKE_CURRENT_SOURCE_DIR}/portaudio/gr-audio-portaudio.conf)
 
diff --git a/gr-audio/lib/jack/jack_impl.h b/gr-audio/lib/jack/jack_impl.h
index 5dcd3b811e..178b6a1388 100644
--- a/gr-audio/lib/jack/jack_impl.h
+++ b/gr-audio/lib/jack/jack_impl.h
@@ -20,9 +20,9 @@
  * Boston, MA 02110-1301, USA.
  */
 
-#ifndef INCLUDED_GRI_JACK_H
-#define INCLUDED_GRI_JACK_H
+#ifndef INCLUDED_AUDIO_JACK_IMPL_H
+#define INCLUDED_AUDIO_JACK_IMPL_H
 
 #include <stdio.h>
 
-#endif /* INCLUDED_GRI_JACK_H */
+#endif /* INCLUDED_AUDIO_JACK_IMPL_H */
diff --git a/gr-audio/lib/osx/audio_osx.h b/gr-audio/lib/osx/audio_osx.h
deleted file mode 100644
index 8c9543d0d6..0000000000
--- a/gr-audio/lib/osx/audio_osx.h
+++ /dev/null
@@ -1,72 +0,0 @@
-/* -*- 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.
- */
-
-#ifndef INCLUDED_AUDIO_OSX_H
-#define INCLUDED_AUDIO_OSX_H
-
-#include <iostream>
-#include <string.h>
-
-#define CheckErrorAndThrow(err,what,throw_str)				\
-  if (err) {								\
-    OSStatus error = static_cast<OSStatus>(err);			\
-    char err_str[4];							\
-    strncpy (err_str, (char*)(&err), 4);				\
-    std::cerr << what << std::endl;					\
-    std::cerr << "  Error# " << error << " ('" << err_str		\
-	      << "')" << std::endl;					\
-    std::cerr << "  " << __FILE__ << ":" << __LINE__ << std::endl;	\
-    fflush (stderr);							\
-    throw std::runtime_error (throw_str);				\
-  }
-
-#define CheckError(err,what)						\
-  if (err) {								\
-    OSStatus error = static_cast<OSStatus>(err);			\
-    char err_str[4];							\
-    strncpy (err_str, (char*)(&err), 4);				\
-    std::cerr << what << std::endl;					\
-    std::cerr << "  Error# " << error << " ('" << err_str		\
-	      << "')" << std::endl;					\
-    std::cerr << "  " << __FILE__ << ":" << __LINE__ << std::endl;	\
-    fflush (stderr);							\
-  }
-
-#include <boost/detail/endian.hpp> //BOOST_BIG_ENDIAN
-#ifdef BOOST_BIG_ENDIAN
-#define GR_PCM_ENDIANNESS kLinearPCMFormatFlagIsBigEndian
-#else
-#define GR_PCM_ENDIANNESS 0
-#endif
-
-// Check the version of MacOSX being used
-#ifdef __APPLE_CC__
-#include <AvailabilityMacros.h>
-#ifndef MAC_OS_X_VERSION_10_6
-#define MAC_OS_X_VERSION_10_6 1060
-#endif
-#if MAC_OS_X_VERSION_MAX_ALLOWED < MAC_OS_X_VERSION_10_6
-#define GR_USE_OLD_AUDIO_UNIT
-#endif
-#endif
-
-#endif /* INCLUDED_AUDIO_OSX_H */
diff --git a/gr-audio/lib/osx/audio_osx_sink.cc b/gr-audio/lib/osx/audio_osx_sink.cc
deleted file mode 100644
index 939e5e0a1d..0000000000
--- a/gr-audio/lib/osx/audio_osx_sink.cc
+++ /dev/null
@@ -1,404 +0,0 @@
-/* -*- c++ -*- */
-/*
- * Copyright 2006-2011 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
-
-#include "gr_audio_registry.h"
-#include <audio_osx_sink.h>
-#include <gr_io_signature.h>
-#include <stdexcept>
-#include <audio_osx.h>
-
-#define _OSX_AU_DEBUG_ 0
-
-AUDIO_REGISTER_SINK(REG_PRIO_HIGH, osx)(
-    int sampling_rate, const std::string &device_name, bool ok_to_block
-){
-    return audio_sink::sptr(new audio_osx_sink(sampling_rate, device_name, ok_to_block));
-}
-
-audio_osx_sink::audio_osx_sink (int sample_rate,
-				const std::string device_name,
-				bool do_block,
-				int channel_config,
-				int max_sample_count)
-  : gr_sync_block ("audio_osx_sink",
-		   gr_make_io_signature (0, 0, 0),
-		   gr_make_io_signature (0, 0, 0)),
-    d_sample_rate (0.0), d_channel_config (0), d_n_channels (0),
-    d_queueSampleCount (0), d_max_sample_count (0),
-    d_do_block (do_block), d_internal (0), d_cond_data (0),
-    d_OutputAU (0)
-{
-  if (sample_rate <= 0) {
-    std::cerr << "Invalid Sample Rate: " << sample_rate << std::endl;
-    throw std::invalid_argument ("audio_osx_sink::audio_osx_sink");
-  } else
-    d_sample_rate = (Float64) sample_rate;
-
-  if (channel_config <= 0 & channel_config != -1) {
-    std::cerr << "Invalid Channel Config: " << channel_config << std::endl;
-    throw std::invalid_argument ("audio_osx_sink::audio_osx_sink");
-  } 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) {
-      std::cerr << "Error Converting Device Name: " << errno << std::endl;
-      throw std::invalid_argument ("audio_osx_sink::audio_osx_sink");
-    }
-    if (l_n_channels <= 0)
-      channel_config = 2;
-    else
-      channel_config = l_n_channels;
-  }
-
-  d_n_channels = d_channel_config = channel_config;
-
-// set the input signature
-
-  set_input_signature (gr_make_io_signature (1, d_n_channels, sizeof (float)));
-
-// 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) {
-    std::cerr << "Invalid Max Sample Count: " << max_sample_count << std::endl;
-    throw std::invalid_argument ("audio_osx_sink::audio_osx_sink");
-  }
-
-  d_max_sample_count = max_sample_count;
-
-// allocate the output circular buffer(s), one per channel
-
-  d_buffers = (circular_buffer<float>**) new
-    circular_buffer<float>* [d_n_channels];
-  UInt32 n_alloc = (UInt32) ceil ((double) d_max_sample_count);
-  for (UInt32 n = 0; n < d_n_channels; n++) {
-    d_buffers[n] = new circular_buffer<float> (n_alloc, false, false);
-  }
-
-// create the default AudioUnit for output
-  OSStatus err = noErr;
-
-// Open the default output unit
-#ifndef GR_USE_OLD_AUDIO_UNIT
-  AudioComponentDescription desc;
-#else
-  ComponentDescription desc;
-#endif
-
-  desc.componentType = kAudioUnitType_Output;
-  desc.componentSubType = kAudioUnitSubType_DefaultOutput;
-  desc.componentManufacturer = kAudioUnitManufacturer_Apple;
-  desc.componentFlags = 0;
-  desc.componentFlagsMask = 0;
-
-#ifndef GR_USE_OLD_AUDIO_UNIT
-  AudioComponent comp = AudioComponentFindNext(NULL, &desc);
-  if (comp == NULL) {
-    std::cerr << "AudioComponentFindNext Error" << std::endl;
-    throw std::runtime_error ("audio_osx_sink::audio_osx_sink");
-  }
-#else
-  Component comp = FindNextComponent (NULL, &desc);
-  if (comp == NULL) {
-    std::cerr << "FindNextComponent Error" << std::endl;
-    throw std::runtime_error ("audio_osx_sink::audio_osx_sink");
-  }
-#endif
-
-#ifndef GR_USE_OLD_AUDIO_UNIT
-  err = AudioComponentInstanceNew (comp, &d_OutputAU);
-  CheckErrorAndThrow (err, "AudioComponentInstanceNew", "audio_osx_sink::audio_osx_sink");
-#else
-  err = OpenAComponent (comp, &d_OutputAU);
-  CheckErrorAndThrow (err, "OpenAComponent", "audio_osx_sink::audio_osx_sink");
-#endif
-
-// Set up a callback function to generate output to the output unit
-
-  AURenderCallbackStruct input;
-  input.inputProc = (AURenderCallback)(audio_osx_sink::AUOutputCallback);
-  input.inputProcRefCon = this;
-
-  err = AudioUnitSetProperty (d_OutputAU,
-			      kAudioUnitProperty_SetRenderCallback,
-			      kAudioUnitScope_Input,
-			      0,
-			      &input,
-			      sizeof (input));
-  CheckErrorAndThrow (err, "AudioUnitSetProperty Render Callback", "audio_osx_sink::audio_osx_sink");
-
-// tell the Output Unit what format data will be supplied to it
-// so that it handles any format conversions
-
-  AudioStreamBasicDescription streamFormat;
-  streamFormat.mSampleRate = (Float64)(sample_rate);
-  streamFormat.mFormatID = kAudioFormatLinearPCM;
-  streamFormat.mFormatFlags = (kLinearPCMFormatFlagIsFloat |
-			       GR_PCM_ENDIANNESS |
-			       kLinearPCMFormatFlagIsPacked |
-			       kAudioFormatFlagIsNonInterleaved);
-  streamFormat.mBytesPerPacket = 4;
-  streamFormat.mFramesPerPacket = 1;
-  streamFormat.mBytesPerFrame = 4;
-  streamFormat.mChannelsPerFrame = d_n_channels;
-  streamFormat.mBitsPerChannel = 32;
-
-  err = AudioUnitSetProperty (d_OutputAU,
-			      kAudioUnitProperty_StreamFormat,
-			      kAudioUnitScope_Input,
-			      0,
-			      &streamFormat,
-			      sizeof (AudioStreamBasicDescription));
-  CheckErrorAndThrow (err, "AudioUnitSetProperty StreamFormat", "audio_osx_sink::audio_osx_sink");
-
-// create the stuff to regulate I/O
-
-  d_cond_data = new gruel::condition_variable ();
-  if (d_cond_data == NULL)
-    CheckErrorAndThrow (errno, "new condition (data)",
-			"audio_osx_sink::audio_osx_sink");
-
-  d_internal = new gruel::mutex ();
-  if (d_internal == NULL)
-    CheckErrorAndThrow (errno, "new mutex (internal)",
-			"audio_osx_sink::audio_osx_sink");
-
-// initialize the AU for output
-
-  err = AudioUnitInitialize (d_OutputAU);
-  CheckErrorAndThrow (err, "AudioUnitInitialize",
-		      "audio_osx_sink::audio_osx_sink");
-
-#if _OSX_AU_DEBUG_
-  std::cerr << "audio_osx_sink Parameters:" << std::endl;
-  std::cerr << "  Sample Rate is " << d_sample_rate << std::endl;
-  std::cerr << "  Number of Channels is " << d_n_channels << std::endl;
-  std::cerr << "  Max # samples to store per channel is " << d_max_sample_count << std::endl;
-#endif
-}
-
-bool audio_osx_sink::IsRunning ()
-{
-  UInt32 AURunning = 0, AUSize = sizeof (UInt32);
-
-  OSStatus err = AudioUnitGetProperty (d_OutputAU,
-				       kAudioOutputUnitProperty_IsRunning,
-				       kAudioUnitScope_Global,
-				       0,
-				       &AURunning,
-				       &AUSize);
-  CheckErrorAndThrow (err, "AudioUnitGetProperty IsRunning",
-		      "audio_osx_sink::IsRunning");
-
-  return (AURunning);
-}
-
-bool audio_osx_sink::start ()
-{
-  if (! IsRunning ()) {
-    OSStatus err = AudioOutputUnitStart (d_OutputAU);
-    CheckErrorAndThrow (err, "AudioOutputUnitStart", "audio_osx_sink::start");
-  }
-
-  return (true);
-}
-
-bool audio_osx_sink::stop ()
-{
-  if (IsRunning ()) {
-    OSStatus err = AudioOutputUnitStop (d_OutputAU);
-    CheckErrorAndThrow (err, "AudioOutputUnitStop", "audio_osx_sink::stop");
-
-    for (UInt32 n = 0; n < d_n_channels; n++) {
-      d_buffers[n]->abort ();
-    }
-  }
-
-  return (true);
-}
-
-audio_osx_sink::~audio_osx_sink ()
-{
-// stop and close the AudioUnit
-  stop ();
-  AudioUnitUninitialize (d_OutputAU);
-#ifndef GR_USE_OLD_AUDIO_UNIT
-  AudioComponentInstanceDispose (d_OutputAU);
-#else
-  CloseComponent (d_OutputAU);
-#endif
-
-// empty and delete the queues
-  for (UInt32 n = 0; n < d_n_channels; n++) {
-    delete d_buffers[n];
-    d_buffers[n] = 0;
-  }
-  delete [] d_buffers;
-  d_buffers = 0;
-
-// close and delete control stuff
-  delete d_cond_data;
-  d_cond_data = 0;
-  delete d_internal;
-  d_internal = 0;
-}
-
-int
-audio_osx_sink::work (int noutput_items,
-		      gr_vector_const_void_star &input_items,
-		      gr_vector_void_star &output_items)
-{
-  gruel::scoped_lock l (*d_internal);
-
-  /* take the input data, copy it, and push it to the bottom of the queue
-     mono input are pushed onto queue[0];
-     stereo input are pushed onto queue[1].
-     Start the AudioUnit if necessary. */
-
-  UInt32 l_max_count;
-  int diff_count = d_max_sample_count - noutput_items;
-  if (diff_count < 0)
-    l_max_count = 0;
-  else
-    l_max_count = (UInt32) diff_count;
-
-#if 0
-  if (l_max_count < d_queueItemLength->back()) {
-//  allow 2 buffers at a time, regardless of length
-    l_max_count = d_queueItemLength->back();
-  }
-#endif
-
-#if _OSX_AU_DEBUG_
-  std::cerr << "work1: qSC = " << d_queueSampleCount << ", lMC = "<< l_max_count
-	    << ", dmSC = " << d_max_sample_count << ", nOI = " << noutput_items << std::endl;
-#endif
-
-  if (d_queueSampleCount > l_max_count) {
-// data coming in too fast; do_block decides what to do
-    if (d_do_block == true) {
-// block until there is data to return
-      while (d_queueSampleCount > l_max_count) {
-// release control so-as to allow data to be retrieved;
-// block until there is data to return
-	d_cond_data->wait (l);
-// the condition's 'notify' was called; acquire control
-// to keep thread safe
-      }
-    }
-  }
-// not blocking case and overflow is handled by the circular buffer
-
-// add the input frames to the buffers' queue, checking for overflow
-
-  UInt32 l_counter;
-  int res = 0;
-  float* inBuffer = (float*) input_items[0];
-  const UInt32 l_size = input_items.size();
-  for (l_counter = 0; l_counter < l_size; l_counter++) {
-    inBuffer = (float*) input_items[l_counter];
-    int l_res = d_buffers[l_counter]->enqueue (inBuffer,
-					       noutput_items);
-    if (l_res == -1)
-      res = -1;
-  }
-  while (l_counter < d_n_channels) {
-// for extra channels, copy the last input's data
-    int l_res = d_buffers[l_counter++]->enqueue (inBuffer,
-						 noutput_items);
-    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
-    d_queueSampleCount = d_buffers[0]->buffer_length_items ();
-  } else {
-// keep up the local sample count
-    d_queueSampleCount += noutput_items;
-  }
-
-#if _OSX_AU_DEBUG_
-  std::cerr << "work2: #OI = " << noutput_items << ", #Cnt = "
-	    << d_queueSampleCount << ", mSC = " << d_max_sample_count << std::endl;
-#endif
-
-  return (noutput_items);
-}
-
-OSStatus audio_osx_sink::AUOutputCallback
-(void *inRefCon,
- AudioUnitRenderActionFlags *ioActionFlags,
- const AudioTimeStamp *inTimeStamp,
- UInt32 inBusNumber,
- UInt32 inNumberFrames,
- AudioBufferList *ioData)
-{
-  audio_osx_sink* This = (audio_osx_sink*) inRefCon;
-  OSStatus err = noErr;
-
-  gruel::scoped_lock l (*This->d_internal);
-
-#if _OSX_AU_DEBUG_
-  std::cerr << "cb_in: SC = " << This->d_queueSampleCount
-	    << ", in#F = " << inNumberFrames << std::endl;
-#endif
-
-  if (This->d_queueSampleCount < inNumberFrames) {
-// not enough data to fill request
-    err = -1;
-  } else {
-// enough data; remove data from our buffers into the AU's buffers
-    int l_counter = This->d_n_channels;
-
-    while (--l_counter >= 0) {
-      size_t t_n_output_items = inNumberFrames;
-      float* outBuffer = (float*) ioData->mBuffers[l_counter].mData;
-      This->d_buffers[l_counter]->dequeue (outBuffer, &t_n_output_items);
-      if (t_n_output_items != inNumberFrames) {
-	throw std::runtime_error ("audio_osx_sink::AUOutputCallback(): "
-				  "number of available items changing "
-				  "unexpectedly.\n");
-      }
-    }
-
-    This->d_queueSampleCount -= inNumberFrames;
-  }
-
-#if _OSX_AU_DEBUG_
-  std::cerr << "cb_out: SC = " << This->d_queueSampleCount << std::endl;
-#endif
-
-// signal that data is available
-  This->d_cond_data->notify_one ();
-
-  return (err);
-}
diff --git a/gr-audio/lib/osx/audio_osx_sink.h b/gr-audio/lib/osx/audio_osx_sink.h
deleted file mode 100644
index 0422e6f4b7..0000000000
--- a/gr-audio/lib/osx/audio_osx_sink.h
+++ /dev/null
@@ -1,80 +0,0 @@
-/* -*- c++ -*- */
-/*
- * Copyright 2006-2011,2013 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.
- */
-
-#ifndef INCLUDED_AUDIO_OSX_SINK_H
-#define INCLUDED_AUDIO_OSX_SINK_H
-
-#include <audio/sink.h>
-#include <string>
-#include <list>
-#include <AudioUnit/AudioUnit.h>
-#include <circular_buffer.h>
-
-/*!
- * \brief audio sink using OSX
- * \ingroup audio_blk
- *
- * input signature is one or two streams of floats.
- * Input samples must be in the range [-1,1].
- */
-
-class audio_osx_sink : public audio_sink {
-
-  Float64             d_sample_rate;
-  int                 d_channel_config;
-  UInt32              d_n_channels;
-  UInt32              d_queueSampleCount, d_max_sample_count;
-  bool                d_do_block;
-  gruel::mutex*       d_internal;
-  gruel::condition_variable* d_cond_data;
-  circular_buffer<float>** d_buffers;
-
-// AudioUnits and Such
-  AudioUnit           d_OutputAU;
-
-public:
-  audio_osx_sink (int sample_rate = 44100,
-		  const std::string device_name = "2",
-		  bool do_block = true,
-		  int channel_config = -1,
-		  int max_sample_count = -1);
-
-  ~audio_osx_sink ();
-
-  bool IsRunning ();
-  bool start ();
-  bool stop ();
-
-  int work (int noutput_items,
-	    gr_vector_const_void_star &input_items,
-	    gr_vector_void_star &output_items);
-
-private:
-  static OSStatus AUOutputCallback (void *inRefCon,
-				    AudioUnitRenderActionFlags *ioActionFlags,
-				    const AudioTimeStamp *inTimeStamp,
-				    UInt32 inBusNumber,
-				    UInt32 inNumberFrames,
-				    AudioBufferList *ioData);
-};
-
-#endif /* INCLUDED_AUDIO_OSX_SINK_H */
diff --git a/gr-audio/lib/osx/audio_osx_source.cc b/gr-audio/lib/osx/audio_osx_source.cc
deleted file mode 100644
index 29f0ac3811..0000000000
--- a/gr-audio/lib/osx/audio_osx_source.cc
+++ /dev/null
@@ -1,1065 +0,0 @@
-/* -*- c++ -*- */
-/*
- * Copyright 2006-2011 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
-
-#include "gr_audio_registry.h"
-#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
-
-AUDIO_REGISTER_SOURCE(REG_PRIO_HIGH, osx)(
-    int sampling_rate, const std::string &device_name, bool ok_to_block
-){
-    return audio_source::sptr(new audio_osx_source(sampling_rate, device_name, ok_to_block));
-}
-
-void PrintStreamDesc (AudioStreamBasicDescription *inDesc)
-{
-  if (inDesc == NULL) {
-    std::cerr << "PrintStreamDesc: Can't print a NULL desc!" << std::endl;
-    return;
-  }
-
-  std::cerr << "  Sample Rate        : " << inDesc->mSampleRate << std::endl;
-  char format_id[4];
-  strncpy (format_id, (char*)(&inDesc->mFormatID), 4);
-  std::cerr << "  Format ID          : " << format_id << std::endl;
-  std::cerr << "  Format Flags       : " << inDesc->mFormatFlags << std::endl;
-  std::cerr << "  Bytes per Packet   : " << inDesc->mBytesPerPacket << std::endl;
-  std::cerr << "  Frames per Packet  : " << inDesc->mFramesPerPacket << std::endl;
-  std::cerr << "  Bytes per Frame    : " << inDesc->mBytesPerFrame << std::endl;
-  std::cerr << "  Channels per Frame : " << inDesc->mChannelsPerFrame << std::endl;
-  std::cerr << "  Bits per Channel   : " << inDesc->mBitsPerChannel << std::endl;
-}
-
-// 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) {
-    std::cerr << "Invalid Sample Rate: " << sample_rate << std::endl;
-    throw std::invalid_argument ("audio_osx_source::audio_osx_source");
-  } else
-    d_outputSampleRate = (Float64) sample_rate;
-
-  if (channel_config <= 0 & channel_config != -1) {
-    std::cerr << "Invalid Channel Config: " << channel_config << std::endl;
-    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) {
-      std::cerr << "Error Converting Device Name: " << errno << std::endl;
-      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) {
-    std::cerr << "Invalid Max Sample Count: " << max_sample_count << std::endl;
-    throw std::invalid_argument ("audio_osx_source::audio_osx_source");
-  }
-
-  d_max_sample_count = max_sample_count;
-
-#if _OSX_AU_DEBUG_
-  std::cerr << "source(): max # samples = " << d_max_sample_count << std::endl;
-#endif
-
-  OSStatus err = noErr;
-
-// create the default AudioUnit for input
-
-// Open the default input unit
-#ifndef GR_USE_OLD_AUDIO_UNIT
-  AudioComponentDescription InputDesc;
-#else
-  ComponentDescription InputDesc;
-#endif
-
-
-  InputDesc.componentType = kAudioUnitType_Output;
-  InputDesc.componentSubType = kAudioUnitSubType_HALOutput;
-  InputDesc.componentManufacturer = kAudioUnitManufacturer_Apple;
-  InputDesc.componentFlags = 0;
-  InputDesc.componentFlagsMask = 0;
-
-#ifndef GR_USE_OLD_AUDIO_UNIT
-  AudioComponent comp = AudioComponentFindNext (NULL, &InputDesc);
-#else
-  Component comp = FindNextComponent (NULL, &InputDesc);
-#endif
-
-  if (comp == NULL) {
-#ifndef GR_USE_OLD_AUDIO_UNIT
-    std::cerr << "AudioComponentFindNext Error" << std::endl;
-#else
-    std::cerr << "FindNextComponent Error" << std::endl;
-#endif
-    throw std::runtime_error ("audio_osx_source::audio_osx_source");
-  }
-
-#ifndef GR_USE_OLD_AUDIO_UNIT
-  err = AudioComponentInstanceNew (comp, &d_InputAU);
-  CheckErrorAndThrow (err, "AudioComponentInstanceNew",
-		      "audio_osx_source::audio_osx_source");
-#else
-  err = OpenAComponent (comp, &d_InputAU);
-  CheckErrorAndThrow (err, "OpenAComponent",
-		      "audio_osx_source::audio_osx_source");
-#endif
-
-
-  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) {
-    std::cerr << "Selected Audio Device does not support Input." << std::endl;
-    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_
-  std::cerr << std::endl << "---- Device Stream Format ----" << std::endl;
-  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_
-  std::cerr << std::endl << "---- Client Stream Format ----" << std::endl;
-  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_device.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;
-
-  asbd_client.mSampleRate = asbd_device.mSampleRate;
-  asbd_client.mFormatID = kAudioFormatLinearPCM;
-  asbd_client.mFormatFlags = (kAudioFormatFlagIsFloat |
-			      kAudioFormatFlagIsPacked |
-			      kAudioFormatFlagIsNonInterleaved);
-  if ((asbd_client.mFormatID == kAudioFormatLinearPCM) &&
-      (d_n_deviceChannels == 1)) {
-      asbd_client.mFormatFlags &= ~kLinearPCMFormatFlagIsNonInterleaved;
-  }
-  asbd_client.mBytesPerFrame = sizeof (float);
-  asbd_client.mFramesPerPacket = 1;
-  asbd_client.mBitsPerChannel = asbd_client.mBytesPerFrame * 8;
-  asbd_client.mChannelsPerFrame = d_n_deviceChannels;
-  asbd_client.mBytesPerPacket = asbd_client.mBytesPerFrame;
-
-  propertySize = sizeof(AudioStreamBasicDescription);
-  err = AudioUnitSetProperty (d_InputAU,
-			      kAudioUnitProperty_StreamFormat,
-			      kAudioUnitScope_Output,
-			      1,
-			      &asbd_client,
-			      propertySize);
-  CheckErrorAndThrow (err, "AudioUnitSetProperty Device Ouput Stream Format",
-		      "audio_osx_source::audio_osx_source");
-
-// 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 = sizeof (float);
-  asbd_user.mFramesPerPacket = 1;
-  asbd_user.mBytesPerFrame = asbd_user.mBytesPerPacket;
-  asbd_user.mChannelsPerFrame = d_n_deviceChannels;
-  asbd_user.mBitsPerChannel = asbd_user.mBytesPerPacket * 8;
-
-  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_cond_data = new gruel::condition_variable ();
-  if (d_cond_data == NULL)
-    CheckErrorAndThrow (errno, "new condition (data)",
-			"audio_osx_source::audio_osx_source");
-
-  d_internal = new gruel::mutex ();
-  if (d_internal == NULL)
-    CheckErrorAndThrow (errno, "new mutex (internal)",
-			"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_
-  std::cerr << "audio_osx_source Parameters:" << std::endl;
-  std::cerr << "  Device Sample Rate is " << d_deviceSampleRate << std::endl;
-  std::cerr << "  User Sample Rate is " << d_outputSampleRate << std::endl;
-  std::cerr << "  Max Sample Count is " << d_max_sample_count << std::endl;
-  std::cerr << "  # Device Channels is " << d_n_deviceChannels << std::endl;
-  std::cerr << "  # Max Channels is " << d_n_max_channels << std::endl;
-  std::cerr << "  Device Buffer Size is Frames = " << d_deviceBufferSizeFrames << std::endl;
-  std::cerr << "  Lead Size is Frames = " << d_leadSizeFrames << std::endl;
-  std::cerr << "  Trail Size is Frames = " << d_trailSizeFrames << std::endl;
-  std::cerr << "  Input Buffer Size is Frames = " << d_inputBufferSizeFrames << std::endl;
-  std::cerr << "  Output Buffer Size is Frames = " << d_outputBufferSizeFrames << std::endl;
-#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");
-
-#ifndef GR_USE_OLD_AUDIO_UNIT
-  err = AudioComponentInstanceDispose (d_InputAU);
-  CheckError (err, "~audio_osx_source: AudioComponentInstanceDispose");
-#else
-  err = CloseComponent (d_InputAU);
-  CheckError (err, "~audio_osx_source: CloseComponent");
-#endif
-
-// 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_cond_data;
-  d_cond_data = 0;
-  delete d_internal;
-  d_internal = 0;
-}
-
-bool
-audio_osx_source::check_topology (int ninputs, int noutputs)
-{
-// check # inputs to make sure it's valid
-  if (ninputs != 0) {
-    std::cerr << "audio_osx_source::check_topology(): number of input "
-	      << "streams provided (" << ninputs
-	      << ") should be 0." << std::endl;
-    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)) {
-    std::cerr << "audio_osx_source::check_topology(): number of output "
-	      << "streams provided (" << noutputs << ") should be in [1,"
-	      << d_n_max_channels << "] for the selected audio device."
-	      << std::endl;
-    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_
-  std::cerr << "chk_topo: Actual # user output channels = "
-	    << noutputs << std::endl;
-#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
-  gruel::scoped_lock l (*d_internal);
-
-#if _OSX_AU_DEBUG_
-  std::cerr << "work1: SC = " << d_queueSampleCount
-	    << ", #OI = " << noutput_items
-	    << ", #Chan = " << output_items.size() << std::endl;
-#endif
-
-  // set the actual # of output items to the 'desired' amount then
-  // verify that data is available; if not enough data is available,
-  // either wait until it is (is "do_block" is true), return (0) is no
-  // data is available and "do_block" is false, or process the actual
-  // amount of available data.
-
-  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;
-	  // block until there is data to return
-	  d_cond_data->wait (l);
-	  // the condition's 'notify' was called; acquire control to
-	  // keep thread safe
-	}
-      } else {
-	// no data & not blocking; return nothing
-	return (0);
-      }
-    }
-    // use the actual amount of available data
-    actual_noutput_items = d_queueSampleCount;
-  }
-
-  // number of channels
-  int l_counter = (int) output_items.size();
-
-  // copy the items from the circular buffer(s) to 'work's output buffers
-  // verify that the number copied out is as expected.
-
-  while (--l_counter >= 0) {
-    size_t 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) {
-      std::cerr << "audio_osx_source::work(): ERROR: number of "
-		<< "available items changing unexpectedly; expecting "
-		<< actual_noutput_items << ", got "
-		<< t_n_output_items << "." << std::endl;
-      throw std::runtime_error ("audio_osx_source::work()");
-    }
-  }
-
-  // subtract the actual number of items removed from the buffer(s)
-  // from the local accounting of the number of available samples
-
-  d_queueSampleCount -= actual_noutput_items;
-
-#if _OSX_AU_DEBUG_
-  std::cerr << "work2: SC = " << d_queueSampleCount
-	    << ", act#OI = " << actual_noutput_items << std::endl
-	    << "Returning." << std::endl;
-#endif
-
-  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_
-  std::cerr << "cc1: io#DP = " << (*ioNumberDataPackets)
-	    << ", TIBSB = " << totalInputBufferSizeBytes
-	    << ", #C = " << counter << std::endl;
-#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;
-  }
-
-#if _OSX_AU_DEBUG_
-  std::cerr << "cc2: Returning." << std::endl;
-#endif
-
-  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);
-
-  gruel::scoped_lock l (*This->d_internal);
-
-#if _OSX_AU_DEBUG_
-  std::cerr << "cb0: in#F = " << inNumberFrames
-	    << ", inBN = " << inBusNumber
-	    << ", SC = " << This->d_queueSampleCount << std::endl;
-#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_
-    std::cerr << "cb1:  avail: #IF = " << AvailableInputFrames
-	      << ", #OF = " << AvailableOutputFrames << std::endl;
-#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_
-    std::cerr << "cb2: actual: #IF = " << This->d_n_ActualInputFrames
-	      << ", #OF = " << AvailableOutputFrames << std::endl;
-    if (This->d_n_ActualInputFrames != AvailableInputFrames)
-      std::cerr << "cb2.1: avail#IF = " << AvailableInputFrames
-		<< ", actual#IF = " << This->d_n_ActualInputFrames << std::endl;
-#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_
-    std::cerr << "cb3: enqueuing audio data." << std::endl;
-#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_
-  std::cerr << "cb4: #OI = " << ActualOutputFrames
-	    << ", #Cnt = " << This->d_queueSampleCount
-	    << ", mSC = " << This->d_max_sample_count << std::endl;
-#endif
-
-// signal that data is available, if appropraite
-  This->d_cond_data->notify_one ();
-
-#if _OSX_AU_DEBUG_
-  std::cerr << "cb5: returning." << std::endl;
-#endif
-
-  return (err);
-}
-
-void
-audio_osx_source::SetDefaultInputDeviceAsCurrent
-()
-{
-// set the default input device
-  AudioDeviceID deviceID = 0;
-  UInt32 dataSize = sizeof (AudioDeviceID);
-  OSStatus err = noErr;
-
-#ifndef GR_USE_OLD_AUDIO_UNIT
-  AudioObjectPropertyAddress theAddress =
-    { kAudioHardwarePropertyDefaultInputDevice,
-      kAudioObjectPropertyScopeGlobal,
-      kAudioObjectPropertyElementMaster };
-
-  err = AudioObjectGetPropertyData
-    (kAudioObjectSystemObject,
-     &theAddress,
-     0,
-     NULL,
-     &dataSize,
-     &deviceID);
-#else
-  err = AudioHardwareGetProperty
-    (kAudioHardwarePropertyDefaultInputDevice,
-     &dataSize,
-     &deviceID);
-#endif
-
-  CheckErrorAndThrow (err, "Get Audio Unit Property for Current Device",
-		      "audio_osx_source::SetDefaultInputDeviceAsCurrent");
-
-  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);
-
-  std::cerr << "a_o_s::HardwareListener" << std::endl;
-
-// 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;
-
-  std::cerr << "a_o_s::UnitListener" << std::endl;
-
-// 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");
-
-  std::cerr << "UnitListener: Input Source changed." << std::endl
-	    << "Old Source Output Info:" << std::endl;
-  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");
-
-  std::cerr << "New Source Output Info:" << std::endl;
-  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
diff --git a/gr-audio/lib/osx/audio_osx_source.h b/gr-audio/lib/osx/audio_osx_source.h
deleted file mode 100644
index f4a72f66fc..0000000000
--- a/gr-audio/lib/osx/audio_osx_source.h
+++ /dev/null
@@ -1,116 +0,0 @@
-/* -*- c++ -*- */
-/*
- * Copyright 2006-2011,2013 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.
- */
-
-#ifndef INCLUDED_AUDIO_OSX_SOURCE_H
-#define INCLUDED_AUDIO_OSX_SOURCE_H
-
-#include <audio/source.h>
-#include <string>
-#include <AudioToolbox/AudioToolbox.h>
-#include <AudioUnit/AudioUnit.h>
-#include <circular_buffer.h>
-
-/*!
- * \brief audio source using OSX
- * \ingroup audio_blk
- *
- * Input signature is one or two streams of floats.
- * Samples must be in the range [-1,1].
- */
-
-class audio_osx_source : public audio_source {
-
-  Float64             d_deviceSampleRate, d_outputSampleRate;
-  int                 d_channel_config;
-  UInt32              d_inputBufferSizeFrames, d_inputBufferSizeBytes;
-  UInt32              d_outputBufferSizeFrames, d_outputBufferSizeBytes;
-  UInt32              d_deviceBufferSizeFrames, d_deviceBufferSizeBytes;
-  UInt32              d_leadSizeFrames, d_leadSizeBytes;
-  UInt32              d_trailSizeFrames, d_trailSizeBytes;
-  UInt32              d_extraBufferSizeFrames, d_extraBufferSizeBytes;
-  UInt32              d_queueSampleCount, d_max_sample_count;
-  UInt32              d_n_AvailableInputFrames, d_n_ActualInputFrames;
-  UInt32              d_n_user_channels, d_n_max_channels, d_n_deviceChannels;
-  bool                d_do_block, d_passThrough, d_waiting_for_data;
-  gruel::mutex*       d_internal;
-  gruel::condition_variable* d_cond_data;
-  circular_buffer<float>** d_buffers;
-
-// AudioUnits and Such
-  AudioUnit           d_InputAU;
-  AudioBufferList*    d_InputBuffer;
-  AudioBufferList*    d_OutputBuffer;
-  AudioConverterRef   d_AudioConverter;
-
-public:
-  audio_osx_source (int sample_rate = 44100,
-		    const std::string device_name = "",
-		    bool do_block = true,
-		    int channel_config = -1,
-		    int max_sample_count = -1);
-
-  ~audio_osx_source ();
-
-  bool start ();
-  bool stop ();
-  bool IsRunning ();
-
-  bool check_topology (int ninputs, int noutputs);
-
-  int work (int noutput_items,
-	    gr_vector_const_void_star &input_items,
-	    gr_vector_void_star &output_items);
-
-private:
-  void SetDefaultInputDeviceAsCurrent ();
-
-  void AllocAudioBufferList (AudioBufferList** t_ABL,
-			     UInt32 n_channels,
-			     UInt32 inputBufferSizeBytes);
-
-  void FreeAudioBufferList (AudioBufferList** t_ABL);
-
-  static OSStatus ConverterCallback (AudioConverterRef inAudioConverter,
-				     UInt32* ioNumberDataPackets,
-				     AudioBufferList* ioData,
-				     AudioStreamPacketDescription** outASPD,
-				     void* inUserData);
-
-  static OSStatus AUInputCallback (void *inRefCon,
-				   AudioUnitRenderActionFlags *ioActionFlags,
-				   const AudioTimeStamp *inTimeStamp,
-				   UInt32 inBusNumber,
-				   UInt32 inNumberFrames,
-				   AudioBufferList *ioData);
-#if _OSX_DO_LISTENERS_
-  static OSStatus UnitListener (void *inRefCon,
-				AudioUnit ci,
-				AudioUnitPropertyID inID,
-				AudioUnitScope inScope,
-				AudioUnitElement inElement);
-
-  static OSStatus HardwareListener (AudioHardwarePropertyID inPropertyID,
-				    void *inClientData);
-#endif
-};
-
-#endif /* INCLUDED_AUDIO_OSX_SOURCE_H */
diff --git a/gr-audio/lib/osx/osx_impl.h b/gr-audio/lib/osx/osx_impl.h
new file mode 100644
index 0000000000..5a12bac71a
--- /dev/null
+++ b/gr-audio/lib/osx/osx_impl.h
@@ -0,0 +1,78 @@
+/* -*- c++ -*- */
+/*
+ * Copyright 2006, 2013 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.
+ */
+
+#ifndef INCLUDED_AUDIO_OSX_IMPL_H
+#define INCLUDED_AUDIO_OSX_IMPL_H
+
+#include <iostream>
+#include <string.h>
+
+namespace gr {
+  namespace audio {
+
+#define CheckErrorAndThrow(err,what,throw_str)                          \
+  if(err) {                                                             \
+    OSStatus error = static_cast<OSStatus>(err);                        \
+    char err_str[4];                                                    \
+    strncpy(err_str, (char*)(&err), 4);                                 \
+    std::cerr << what << std::endl;                                     \
+    std::cerr << "  Error# " << error << " ('" << err_str               \
+              << "')" << std::endl;                                     \
+    std::cerr << "  " << __FILE__ << ":" << __LINE__ << std::endl;      \
+    fflush(stderr);                                                     \
+    throw std::runtime_error(throw_str);                                \
+  }
+
+#define CheckError(err,what)                                            \
+  if(err) {                                                             \
+    OSStatus error = static_cast<OSStatus>(err);                        \
+    char err_str[4];                                                    \
+    strncpy(err_str, (char*)(&err), 4);                                 \
+    std::cerr << what << std::endl;                                     \
+    std::cerr << "  Error# " << error << " ('" << err_str               \
+              << "')" << std::endl;                                     \
+    std::cerr << "  " << __FILE__ << ":" << __LINE__ << std::endl;      \
+    fflush(stderr);                                                     \
+  }
+
+#include <boost/detail/endian.hpp> //BOOST_BIG_ENDIAN
+#ifdef BOOST_BIG_ENDIAN
+#define GR_PCM_ENDIANNESS kLinearPCMFormatFlagIsBigEndian
+#else
+#define GR_PCM_ENDIANNESS 0
+#endif
+
+// Check the version of MacOSX being used
+#ifdef __APPLE_CC__
+#include <AvailabilityMacros.h>
+#ifndef MAC_OS_X_VERSION_10_6
+#define MAC_OS_X_VERSION_10_6 1060
+#endif
+#if MAC_OS_X_VERSION_MAX_ALLOWED < MAC_OS_X_VERSION_10_6
+#define GR_USE_OLD_AUDIO_UNIT
+#endif
+#endif
+
+  } /* namespace audio */
+} /* namespace gr */
+
+#endif /* INCLUDED_AUDIO_OSX_IMPL_H */
diff --git a/gr-audio/lib/osx/osx_sink.cc b/gr-audio/lib/osx/osx_sink.cc
new file mode 100644
index 0000000000..3083bc9e96
--- /dev/null
+++ b/gr-audio/lib/osx/osx_sink.cc
@@ -0,0 +1,429 @@
+/* -*- c++ -*- */
+/*
+ * Copyright 2006-2011,2013 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
+
+#include "audio_registry.h"
+#include <osx_sink.h>
+#include <osx_impl.h>
+#include <gr_io_signature.h>
+#include <stdexcept>
+
+namespace gr {
+  namespace audio {
+
+#define _OSX_AU_DEBUG_ 0
+
+    AUDIO_REGISTER_SINK(REG_PRIO_HIGH, osx)(int sampling_rate,
+                                            const std::string &device_name,
+                                            bool ok_to_block)
+    {
+      return sink::sptr
+        (new osx_sink(sampling_rate, device_name, ok_to_block));
+    }
+
+    osx_sink::osx_sink(int sample_rate,
+                       const std::string device_name,
+                       bool do_block,
+                       int channel_config,
+                       int max_sample_count)
+      : gr_sync_block("audio_osx_sink",
+                      gr_make_io_signature(0, 0, 0),
+                      gr_make_io_signature(0, 0, 0)),
+        d_sample_rate(0.0), d_channel_config(0), d_n_channels(0),
+        d_queueSampleCount(0), d_max_sample_count(0),
+        d_do_block(do_block), d_internal(0), d_cond_data(0),
+        d_OutputAU(0)
+    {
+      if(sample_rate <= 0) {
+        std::cerr << "Invalid Sample Rate: " << sample_rate << std::endl;
+        throw std::invalid_argument ("audio_osx_sink::audio_osx_sink");
+      }
+      else
+        d_sample_rate = (Float64)sample_rate;
+
+      if(channel_config <= 0 & channel_config != -1) {
+        std::cerr << "Invalid Channel Config: " << channel_config << std::endl;
+        throw std::invalid_argument ("audio_osx_sink::audio_osx_sink");
+      }
+      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) {
+          std::cerr << "Error Converting Device Name: " << errno << std::endl;
+          throw std::invalid_argument("audio_osx_sink::audio_osx_sink");
+        }
+        if(l_n_channels <= 0)
+          channel_config = 2;
+        else
+          channel_config = l_n_channels;
+      }
+
+      d_n_channels = d_channel_config = channel_config;
+
+      // set the input signature
+
+      set_input_signature(gr_make_io_signature(1, d_n_channels, sizeof(float)));
+
+      // 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) {
+        std::cerr << "Invalid Max Sample Count: " << max_sample_count << std::endl;
+        throw std::invalid_argument ("audio_osx_sink::audio_osx_sink");
+      }
+
+      d_max_sample_count = max_sample_count;
+
+      // allocate the output circular buffer(s), one per channel
+
+      d_buffers = (circular_buffer<float>**) new
+        circular_buffer<float>* [d_n_channels];
+      UInt32 n_alloc = (UInt32) ceil((double)d_max_sample_count);
+      for(UInt32 n = 0; n < d_n_channels; n++) {
+        d_buffers[n] = new circular_buffer<float>(n_alloc, false, false);
+      }
+
+      // create the default AudioUnit for output
+      OSStatus err = noErr;
+
+      // Open the default output unit
+#ifndef GR_USE_OLD_AUDIO_UNIT
+      AudioComponentDescription desc;
+#else
+      ComponentDescription desc;
+#endif
+
+      desc.componentType = kAudioUnitType_Output;
+      desc.componentSubType = kAudioUnitSubType_DefaultOutput;
+      desc.componentManufacturer = kAudioUnitManufacturer_Apple;
+      desc.componentFlags = 0;
+      desc.componentFlagsMask = 0;
+
+#ifndef GR_USE_OLD_AUDIO_UNIT
+      AudioComponent comp = AudioComponentFindNext(NULL, &desc);
+      if(comp == NULL) {
+        std::cerr << "AudioComponentFindNext Error" << std::endl;
+        throw std::runtime_error("audio_osx_sink::audio_osx_sink");
+      }
+#else
+      Component comp = FindNextComponent(NULL, &desc);
+      if(comp == NULL) {
+        std::cerr << "FindNextComponent Error" << std::endl;
+        throw std::runtime_error("audio_osx_sink::audio_osx_sink");
+      }
+#endif
+
+#ifndef GR_USE_OLD_AUDIO_UNIT
+      err = AudioComponentInstanceNew(comp, &d_OutputAU);
+      CheckErrorAndThrow(err, "AudioComponentInstanceNew",
+                         "audio_osx_sink::audio_osx_sink");
+#else
+      err = OpenAComponent(comp, &d_OutputAU);
+      CheckErrorAndThrow(err, "OpenAComponent",
+                         "audio_osx_sink::audio_osx_sink");
+#endif
+
+      // Set up a callback function to generate output to the output unit
+
+      AURenderCallbackStruct input;
+      input.inputProc = (AURenderCallback)(audio_osx_sink::AUOutputCallback);
+      input.inputProcRefCon = this;
+
+      err = AudioUnitSetProperty(d_OutputAU,
+                                 kAudioUnitProperty_SetRenderCallback,
+                                 kAudioUnitScope_Input,
+                                 0,
+                                 &input,
+                                 sizeof (input));
+      CheckErrorAndThrow(err, "AudioUnitSetProperty Render Callback",
+                         "audio_osx_sink::audio_osx_sink");
+
+      // tell the Output Unit what format data will be supplied to it
+      // so that it handles any format conversions
+
+      AudioStreamBasicDescription streamFormat;
+      streamFormat.mSampleRate = (Float64)(sample_rate);
+      streamFormat.mFormatID = kAudioFormatLinearPCM;
+      streamFormat.mFormatFlags = (kLinearPCMFormatFlagIsFloat |
+                                   GR_PCM_ENDIANNESS |
+                                   kLinearPCMFormatFlagIsPacked |
+                                   kAudioFormatFlagIsNonInterleaved);
+      streamFormat.mBytesPerPacket = 4;
+      streamFormat.mFramesPerPacket = 1;
+      streamFormat.mBytesPerFrame = 4;
+      streamFormat.mChannelsPerFrame = d_n_channels;
+      streamFormat.mBitsPerChannel = 32;
+
+      err = AudioUnitSetProperty(d_OutputAU,
+                                 kAudioUnitProperty_StreamFormat,
+                                 kAudioUnitScope_Input,
+                                 0,
+                                 &streamFormat,
+                                 sizeof(AudioStreamBasicDescription));
+      CheckErrorAndThrow(err, "AudioUnitSetProperty StreamFormat",
+                         "audio_osx_sink::audio_osx_sink");
+
+      // create the stuff to regulate I/O
+
+      d_cond_data = new gruel::condition_variable();
+      if(d_cond_data == NULL)
+        CheckErrorAndThrow(errno, "new condition (data)",
+                           "audio_osx_sink::audio_osx_sink");
+
+      d_internal = new gruel::mutex();
+      if(d_internal == NULL)
+        CheckErrorAndThrow(errno, "new mutex (internal)",
+                           "audio_osx_sink::audio_osx_sink");
+
+      // initialize the AU for output
+
+      err = AudioUnitInitialize(d_OutputAU);
+      CheckErrorAndThrow(err, "AudioUnitInitialize",
+                         "audio_osx_sink::audio_osx_sink");
+
+#if _OSX_AU_DEBUG_
+      std::cerr << "audio_osx_sink Parameters:" << std::endl;
+      std::cerr << "  Sample Rate is " << d_sample_rate << std::endl;
+      std::cerr << "  Number of Channels is " << d_n_channels << std::endl;
+      std::cerr << "  Max # samples to store per channel is " << d_max_sample_count << std::endl;
+#endif
+}
+
+    bool
+    osx_sink::IsRunning()
+    {
+      UInt32 AURunning = 0, AUSize = sizeof(UInt32);
+
+      OSStatus err = AudioUnitGetProperty(d_OutputAU,
+                                          kAudioOutputUnitProperty_IsRunning,
+                                          kAudioUnitScope_Global,
+                                          0,
+                                          &AURunning,
+                                          &AUSize);
+      CheckErrorAndThrow(err, "AudioUnitGetProperty IsRunning",
+                         "audio_osx_sink::IsRunning");
+
+      return (AURunning);
+    }
+
+    bool
+    osx_sink::start()
+    {
+      if(!IsRunning()) {
+        OSStatus err = AudioOutputUnitStart(d_OutputAU);
+        CheckErrorAndThrow(err, "AudioOutputUnitStart",
+                           "audio_osx_sink::start");
+      }
+
+      return (true);
+    }
+
+    bool
+    osx_sink::stop()
+    {
+      if(IsRunning ()) {
+        OSStatus err = AudioOutputUnitStop(d_OutputAU);
+        CheckErrorAndThrow(err, "AudioOutputUnitStop",
+                           "audio_osx_sink::stop");
+
+        for(UInt32 n = 0; n < d_n_channels; n++) {
+          d_buffers[n]->abort();
+        }
+      }
+
+      return (true);
+    }
+
+    osx_sink::~osx_sink()
+    {
+      // stop and close the AudioUnit
+      stop();
+      AudioUnitUninitialize(d_OutputAU);
+#ifndef GR_USE_OLD_AUDIO_UNIT
+      AudioComponentInstanceDispose(d_OutputAU);
+#else
+      CloseComponent(d_OutputAU);
+#endif
+
+      // empty and delete the queues
+      for(UInt32 n = 0; n < d_n_channels; n++) {
+        delete d_buffers[n];
+        d_buffers[n] = 0;
+      }
+      delete [] d_buffers;
+      d_buffers = 0;
+
+      // close and delete control stuff
+      delete d_cond_data;
+      d_cond_data = 0;
+      delete d_internal;
+      d_internal = 0;
+    }
+
+    int
+    osx_sink::work(int noutput_items,
+                   gr_vector_const_void_star &input_items,
+                   gr_vector_void_star &output_items)
+    {
+      gruel::scoped_lock l(*d_internal);
+
+      /* take the input data, copy it, and push it to the bottom of the queue
+         mono input are pushed onto queue[0];
+         stereo input are pushed onto queue[1].
+         Start the AudioUnit if necessary. */
+
+      UInt32 l_max_count;
+      int diff_count = d_max_sample_count - noutput_items;
+      if(diff_count < 0)
+        l_max_count = 0;
+      else
+        l_max_count = (UInt32)diff_count;
+
+#if 0
+      if(l_max_count < d_queueItemLength->back()) {
+        // allow 2 buffers at a time, regardless of length
+        l_max_count = d_queueItemLength->back();
+      }
+#endif
+
+#if _OSX_AU_DEBUG_
+      std::cerr << "work1: qSC = " << d_queueSampleCount
+                << ", lMC = "<< l_max_count
+                << ", dmSC = " << d_max_sample_count
+                << ", nOI = " << noutput_items << std::endl;
+#endif
+
+      if(d_queueSampleCount > l_max_count) {
+        // data coming in too fast; do_block decides what to do
+        if(d_do_block == true) {
+          // block until there is data to return
+          while(d_queueSampleCount > l_max_count) {
+            // release control so-as to allow data to be retrieved;
+            // block until there is data to return
+            d_cond_data->wait(l);
+            // the condition's 'notify' was called; acquire control
+            // to keep thread safe
+          }
+        }
+      }
+      // not blocking case and overflow is handled by the circular buffer
+
+      // add the input frames to the buffers' queue, checking for overflow
+
+      UInt32 l_counter;
+      int res = 0;
+      float* inBuffer = (float*)input_items[0];
+      const UInt32 l_size = input_items.size();
+      for(l_counter = 0; l_counter < l_size; l_counter++) {
+        inBuffer = (float*)input_items[l_counter];
+        int l_res = d_buffers[l_counter]->enqueue(inBuffer,
+                                                  noutput_items);
+        if(l_res == -1)
+          res = -1;
+      }
+      while(l_counter < d_n_channels) {
+        // for extra channels, copy the last input's data
+        int l_res = d_buffers[l_counter++]->enqueue(inBuffer,
+                                                    noutput_items);
+        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
+        d_queueSampleCount = d_buffers[0]->buffer_length_items();
+      }
+      else {
+        // keep up the local sample count
+        d_queueSampleCount += noutput_items;
+      }
+
+#if _OSX_AU_DEBUG_
+      std::cerr << "work2: #OI = "
+                << noutput_items << ", #Cnt = "
+                << d_queueSampleCount << ", mSC = "
+                << d_max_sample_count << std::endl;
+#endif
+
+      return (noutput_items);
+    }
+
+    OSStatus
+    osx_sink::AUOutputCallback(void *inRefCon,
+                               AudioUnitRenderActionFlags *ioActionFlags,
+                               const AudioTimeStamp *inTimeStamp,
+                               UInt32 inBusNumber,
+                               UInt32 inNumberFrames,
+                               AudioBufferList *ioData)
+    {
+      audio_osx_sink* This = (audio_osx_sink*)inRefCon;
+      OSStatus err = noErr;
+
+      gruel::scoped_lock l(*This->d_internal);
+
+#if _OSX_AU_DEBUG_
+      std::cerr << "cb_in: SC = " << This->d_queueSampleCount
+                << ", in#F = " << inNumberFrames << std::endl;
+#endif
+
+      if(This->d_queueSampleCount < inNumberFrames) {
+        // not enough data to fill request
+        err = -1;
+      }
+      else {
+        // enough data; remove data from our buffers into the AU's buffers
+        int l_counter = This->d_n_channels;
+
+        while(--l_counter >= 0) {
+          size_t t_n_output_items = inNumberFrames;
+          float* outBuffer = (float*)ioData->mBuffers[l_counter].mData;
+          This->d_buffers[l_counter]->dequeue(outBuffer, &t_n_output_items);
+          if(t_n_output_items != inNumberFrames) {
+            throw std::runtime_error("audio_osx_sink::AUOutputCallback(): "
+                                     "number of available items changing "
+                                     "unexpectedly.\n");
+          }
+        }
+
+        This->d_queueSampleCount -= inNumberFrames;
+      }
+
+#if _OSX_AU_DEBUG_
+      std::cerr << "cb_out: SC = " << This->d_queueSampleCount << std::endl;
+#endif
+
+      // signal that data is available
+      This->d_cond_data->notify_one();
+
+      return (err);
+    }
+
+  } /* namespace audio */
+} /* namespace gr */
diff --git a/gr-audio/lib/osx/osx_sink.h b/gr-audio/lib/osx/osx_sink.h
new file mode 100644
index 0000000000..6bbd882239
--- /dev/null
+++ b/gr-audio/lib/osx/osx_sink.h
@@ -0,0 +1,86 @@
+/* -*- c++ -*- */
+/*
+ * Copyright 2006-2011,2013 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.
+ */
+
+#ifndef INCLUDED_AUDIO_OSX_SINK_H
+#define INCLUDED_AUDIO_OSX_SINK_H
+
+#include <audio/sink.h>
+#include <string>
+#include <list>
+#include <AudioUnit/AudioUnit.h>
+#include <circular_buffer.h>
+
+namespace gr {
+  namespace audio {
+
+    /*!
+     * \brief audio sink using OSX
+     * \ingroup audio_blk
+     *
+     * input signature is one or two streams of floats.
+     * Input samples must be in the range [-1,1].
+     */
+
+    class osx_sink : public sink
+    {
+      Float64 d_sample_rate;
+      int     d_channel_config;
+      UInt32  d_n_channels;
+      UInt32  d_queueSampleCount, d_max_sample_count;
+      bool    d_do_block;
+      gruel::mutex* d_internal;
+      gruel::condition_variable* d_cond_data;
+      circular_buffer<float>** d_buffers;
+
+      // AudioUnits and Such
+      AudioUnit d_OutputAU;
+
+    public:
+      osx_sink(int sample_rate = 44100,
+               const std::string device_name = "2",
+               bool do_block = true,
+               int channel_config = -1,
+               int max_sample_count = -1);
+
+      ~osx_sink();
+
+      bool IsRunning();
+      bool start();
+      bool stop();
+
+      int work(int noutput_items,
+               gr_vector_const_void_star &input_items,
+               gr_vector_void_star &output_items);
+
+    private:
+      static OSStatus AUOutputCallback(void *inRefCon,
+                                       AudioUnitRenderActionFlags *ioActionFlags,
+                                       const AudioTimeStamp *inTimeStamp,
+                                       UInt32 inBusNumber,
+                                       UInt32 inNumberFrames,
+                                       AudioBufferList *ioData);
+    };
+
+  } /* namespace audio */
+} /* namespace gr */
+
+#endif /* INCLUDED_AUDIO_OSX_SINK_H */
diff --git a/gr-audio/lib/osx/osx_source.cc b/gr-audio/lib/osx/osx_source.cc
new file mode 100644
index 0000000000..ee057c8337
--- /dev/null
+++ b/gr-audio/lib/osx/osx_source.cc
@@ -0,0 +1,1077 @@
+/* -*- c++ -*- */
+/*
+ * Copyright 2006-2011 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
+
+#include "audio_registry.h"
+#include <osx_source.h>
+#include <osx_impl.h>
+#include <gr_io_signature.h>
+#include <stdexcept>
+
+namespace gr {
+  namespace audio {
+
+#define _OSX_AU_DEBUG_ 0
+#define _OSX_DO_LISTENERS_ 0
+
+    AUDIO_REGISTER_SOURCE(REG_PRIO_HIGH, osx)(int sampling_rate,
+                                              const std::string &device_name,
+                                              bool ok_to_block)
+    {
+      return audio_source::sptr
+        (new audio_osx_source(sampling_rate, device_name, ok_to_block));
+    }
+
+    void
+    PrintStreamDesc(AudioStreamBasicDescription *inDesc)
+    {
+      if(inDesc == NULL) {
+        std::cerr << "PrintStreamDesc: Can't print a NULL desc!" << std::endl;
+        return;
+      }
+
+      std::cerr << "  Sample Rate        : " << inDesc->mSampleRate << std::endl;
+      char format_id[4];
+      strncpy(format_id, (char*)(&inDesc->mFormatID), 4);
+      std::cerr << "  Format ID          : " << format_id << std::endl;
+      std::cerr << "  Format Flags       : " << inDesc->mFormatFlags << std::endl;
+      std::cerr << "  Bytes per Packet   : " << inDesc->mBytesPerPacket << std::endl;
+      std::cerr << "  Frames per Packet  : " << inDesc->mFramesPerPacket << std::endl;
+      std::cerr << "  Bytes per Frame    : " << inDesc->mBytesPerFrame << std::endl;
+      std::cerr << "  Channels per Frame : " << inDesc->mChannelsPerFrame << std::endl;
+      std::cerr << "  Bits per Channel   : " << inDesc->mBitsPerChannel << std::endl;
+    }
+
+    // FIXME these should query some kind of user preference
+
+    osx_source::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) {
+        std::cerr << "Invalid Sample Rate: " << sample_rate << std::endl;
+        throw std::invalid_argument("audio_osx_source::audio_osx_source");
+      }
+      else
+        d_outputSampleRate = (Float64)sample_rate;
+
+      if(channel_config <= 0 & channel_config != -1) {
+        std::cerr << "Invalid Channel Config: " << channel_config << std::endl;
+        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) {
+          std::cerr << "Error Converting Device Name: " << errno << std::endl;
+          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) {
+        std::cerr << "Invalid Max Sample Count: " << max_sample_count << std::endl;
+        throw std::invalid_argument("audio_osx_source::audio_osx_source");
+      }
+
+      d_max_sample_count = max_sample_count;
+
+#if _OSX_AU_DEBUG_
+      std::cerr << "source(): max # samples = " << d_max_sample_count << std::endl;
+#endif
+
+      OSStatus err = noErr;
+
+      // create the default AudioUnit for input
+
+      // Open the default input unit
+#ifndef GR_USE_OLD_AUDIO_UNIT
+      AudioComponentDescription InputDesc;
+#else
+      ComponentDescription InputDesc;
+#endif
+
+      InputDesc.componentType = kAudioUnitType_Output;
+      InputDesc.componentSubType = kAudioUnitSubType_HALOutput;
+      InputDesc.componentManufacturer = kAudioUnitManufacturer_Apple;
+      InputDesc.componentFlags = 0;
+      InputDesc.componentFlagsMask = 0;
+
+#ifndef GR_USE_OLD_AUDIO_UNIT
+      AudioComponent comp = AudioComponentFindNext(NULL, &InputDesc);
+#else
+      Component comp = FindNextComponent(NULL, &InputDesc);
+#endif
+
+      if(comp == NULL) {
+#ifndef GR_USE_OLD_AUDIO_UNIT
+        std::cerr << "AudioComponentFindNext Error" << std::endl;
+#else
+        std::cerr << "FindNextComponent Error" << std::endl;
+#endif
+        throw std::runtime_error("audio_osx_source::audio_osx_source");
+  }
+
+#ifndef GR_USE_OLD_AUDIO_UNIT
+      err = AudioComponentInstanceNew(comp, &d_InputAU);
+      CheckErrorAndThrow(err, "AudioComponentInstanceNew",
+                         "audio_osx_source::audio_osx_source");
+#else
+      err = OpenAComponent(comp, &d_InputAU);
+      CheckErrorAndThrow(err, "OpenAComponent",
+                         "audio_osx_source::audio_osx_source");
+#endif
+
+      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) {
+        std::cerr << "Selected Audio Device does not support Input." << std::endl;
+        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_
+      std::cerr << std::endl << "---- Device Stream Format ----" << std::endl;
+      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_
+      std::cerr << std::endl << "---- Client Stream Format ----" << std::endl;
+      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_device.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;
+
+      asbd_client.mSampleRate = asbd_device.mSampleRate;
+      asbd_client.mFormatID = kAudioFormatLinearPCM;
+      asbd_client.mFormatFlags = (kAudioFormatFlagIsFloat |
+                                  kAudioFormatFlagIsPacked |
+                                  kAudioFormatFlagIsNonInterleaved);
+      if((asbd_client.mFormatID == kAudioFormatLinearPCM) &&
+         (d_n_deviceChannels == 1)) {
+        asbd_client.mFormatFlags &= ~kLinearPCMFormatFlagIsNonInterleaved;
+      }
+      asbd_client.mBytesPerFrame = sizeof(float);
+      asbd_client.mFramesPerPacket = 1;
+      asbd_client.mBitsPerChannel = asbd_client.mBytesPerFrame * 8;
+      asbd_client.mChannelsPerFrame = d_n_deviceChannels;
+      asbd_client.mBytesPerPacket = asbd_client.mBytesPerFrame;
+
+      propertySize = sizeof(AudioStreamBasicDescription);
+      err = AudioUnitSetProperty(d_InputAU,
+                                 kAudioUnitProperty_StreamFormat,
+                                 kAudioUnitScope_Output,
+                                 1,
+                                 &asbd_client,
+                                 propertySize);
+      CheckErrorAndThrow(err, "AudioUnitSetProperty Device Ouput Stream Format",
+                         "audio_osx_source::audio_osx_source");
+
+      // 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 = sizeof(float);
+      asbd_user.mFramesPerPacket = 1;
+      asbd_user.mBytesPerFrame = asbd_user.mBytesPerPacket;
+      asbd_user.mChannelsPerFrame = d_n_deviceChannels;
+      asbd_user.mBitsPerChannel = asbd_user.mBytesPerPacket * 8;
+
+      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_cond_data = new gruel::condition_variable();
+      if(d_cond_data == NULL)
+        CheckErrorAndThrow(errno, "new condition (data)",
+                           "audio_osx_source::audio_osx_source");
+
+      d_internal = new gruel::mutex();
+      if(d_internal == NULL)
+        CheckErrorAndThrow(errno, "new mutex (internal)",
+                           "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_
+      std::cerr << "audio_osx_source Parameters:" << std::endl;
+      std::cerr << "  Device Sample Rate is " << d_deviceSampleRate << std::endl;
+      std::cerr << "  User Sample Rate is " << d_outputSampleRate << std::endl;
+      std::cerr << "  Max Sample Count is " << d_max_sample_count << std::endl;
+      std::cerr << "  # Device Channels is " << d_n_deviceChannels << std::endl;
+      std::cerr << "  # Max Channels is " << d_n_max_channels << std::endl;
+      std::cerr << "  Device Buffer Size is Frames = " << d_deviceBufferSizeFrames << std::endl;
+      std::cerr << "  Lead Size is Frames = " << d_leadSizeFrames << std::endl;
+      std::cerr << "  Trail Size is Frames = " << d_trailSizeFrames << std::endl;
+      std::cerr << "  Input Buffer Size is Frames = " << d_inputBufferSizeFrames << std::endl;
+      std::cerr << "  Output Buffer Size is Frames = " << d_outputBufferSizeFrames << std::endl;
+#endif
+    }
+
+    void
+    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
+    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
+    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
+    osx_source::start()
+    {
+      if(! IsRunning ()) {
+        OSStatus err = AudioOutputUnitStart(d_InputAU);
+        CheckErrorAndThrow(err, "AudioOutputUnitStart",
+                           "audio_osx_source::start");
+      }
+
+      return (true);
+    }
+
+    bool
+    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);
+    }
+
+    osx_source::~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");
+
+#ifndef GR_USE_OLD_AUDIO_UNIT
+      err = AudioComponentInstanceDispose(d_InputAU);
+      CheckError(err, "~audio_osx_source: AudioComponentInstanceDispose");
+#else
+      err = CloseComponent(d_InputAU);
+      CheckError(err, "~audio_osx_source: CloseComponent");
+#endif
+
+      // 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_cond_data;
+      d_cond_data = 0;
+      delete d_internal;
+      d_internal = 0;
+    }
+
+    bool
+    osx_source::check_topology(int ninputs, int noutputs)
+    {
+      // check # inputs to make sure it's valid
+      if(ninputs != 0) {
+        std::cerr << "audio_osx_source::check_topology(): number of input "
+                  << "streams provided (" << ninputs
+                  << ") should be 0." << std::endl;
+        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)) {
+        std::cerr << "audio_osx_source::check_topology(): number of output "
+                  << "streams provided (" << noutputs << ") should be in [1,"
+                  << d_n_max_channels << "] for the selected audio device."
+                  << std::endl;
+        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_
+      std::cerr << "chk_topo: Actual # user output channels = "
+                << noutputs << std::endl;
+#endif
+
+      return (true);
+    }
+
+    int
+    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
+      gruel::scoped_lock l(*d_internal);
+
+#if _OSX_AU_DEBUG_
+      std::cerr << "work1: SC = " << d_queueSampleCount
+                << ", #OI = " << noutput_items
+                << ", #Chan = " << output_items.size() << std::endl;
+#endif
+
+      // set the actual # of output items to the 'desired' amount then
+      // verify that data is available; if not enough data is available,
+      // either wait until it is (is "do_block" is true), return (0) is no
+      // data is available and "do_block" is false, or process the actual
+      // amount of available data.
+
+      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;
+              // block until there is data to return
+              d_cond_data->wait(l);
+              // the condition's 'notify' was called; acquire control to
+              // keep thread safe
+            }
+          }
+          else {
+            // no data & not blocking; return nothing
+            return (0);
+          }
+        }
+        // use the actual amount of available data
+        actual_noutput_items = d_queueSampleCount;
+      }
+
+      // number of channels
+      int l_counter = (int)output_items.size();
+
+      // copy the items from the circular buffer(s) to 'work's output buffers
+      // verify that the number copied out is as expected.
+
+      while(--l_counter >= 0) {
+        size_t 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) {
+          std::cerr << "audio_osx_source::work(): ERROR: number of "
+                    << "available items changing unexpectedly; expecting "
+                    << actual_noutput_items << ", got "
+                    << t_n_output_items << "." << std::endl;
+          throw std::runtime_error("audio_osx_source::work()");
+        }
+      }
+
+      // subtract the actual number of items removed from the buffer(s)
+      // from the local accounting of the number of available samples
+
+      d_queueSampleCount -= actual_noutput_items;
+
+#if _OSX_AU_DEBUG_
+      std::cerr << "work2: SC = " << d_queueSampleCount
+                << ", act#OI = " << actual_noutput_items << std::endl
+                << "Returning." << std::endl;
+#endif
+
+      return (actual_noutput_items);
+    }
+
+    OSStatus
+    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
+
+      osx_source* This = static_cast<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_
+      std::cerr << "cc1: io#DP = " << (*ioNumberDataPackets)
+                << ", TIBSB = " << totalInputBufferSizeBytes
+                << ", #C = " << counter << std::endl;
+#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;
+      }
+
+#if _OSX_AU_DEBUG_
+      std::cerr << "cc2: Returning." << std::endl;
+#endif
+
+      return (noErr);
+    }
+
+    OSStatus
+    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);
+
+      gruel::scoped_lock l(*This->d_internal);
+
+#if _OSX_AU_DEBUG_
+      std::cerr << "cb0: in#F = " << inNumberFrames
+                << ", inBN = " << inBusNumber
+                << ", SC = " << This->d_queueSampleCount << std::endl;
+#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_
+        std::cerr << "cb1:  avail: #IF = " << AvailableInputFrames
+                  << ", #OF = " << AvailableOutputFrames << std::endl;
+#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_
+        std::cerr << "cb2: actual: #IF = " << This->d_n_ActualInputFrames
+                  << ", #OF = " << AvailableOutputFrames << std::endl;
+        if(This->d_n_ActualInputFrames != AvailableInputFrames)
+          std::cerr << "cb2.1: avail#IF = " << AvailableInputFrames
+                    << ", actual#IF = " << This->d_n_ActualInputFrames << std::endl;
+#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_
+        std::cerr << "cb3: enqueuing audio data." << std::endl;
+#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_
+      std::cerr << "cb4: #OI = " << ActualOutputFrames
+                << ", #Cnt = " << This->d_queueSampleCount
+                << ", mSC = " << This->d_max_sample_count << std::endl;
+#endif
+
+      // signal that data is available, if appropraite
+      This->d_cond_data->notify_one();
+
+#if _OSX_AU_DEBUG_
+      std::cerr << "cb5: returning." << std::endl;
+#endif
+
+      return (err);
+    }
+
+    void
+    osx_source::SetDefaultInputDeviceAsCurrent()
+    {
+      // set the default input device
+      AudioDeviceID deviceID = 0;
+      UInt32 dataSize = sizeof (AudioDeviceID);
+      OSStatus err = noErr;
+
+#ifndef GR_USE_OLD_AUDIO_UNIT
+      AudioObjectPropertyAddress theAddress =
+        { kAudioHardwarePropertyDefaultInputDevice,
+          kAudioObjectPropertyScopeGlobal,
+          kAudioObjectPropertyElementMaster };
+
+      err = AudioObjectGetPropertyData(kAudioObjectSystemObject,
+                                       &theAddress,
+                                       0,
+                                       NULL,
+                                       &dataSize,
+                                       &deviceID);
+#else
+      err = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice,
+                                     &dataSize,
+                                     &deviceID);
+#endif
+
+      CheckErrorAndThrow(err, "Get Audio Unit Property for Current Device",
+                         "audio_osx_source::SetDefaultInputDeviceAsCurrent");
+
+      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
+    osx_source::HardwareListener(AudioHardwarePropertyID inPropertyID,
+                                 void *inClientData)
+    {
+      OSStatus err = noErr;
+      osx_source* This = static_cast<osx_source*>(inClientData);
+
+      std::cerr << "a_o_s::HardwareListener" << std::endl;
+
+      // 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
+    osx_source::UnitListener(void *inRefCon,
+                             AudioUnit ci,
+                             AudioUnitPropertyID inID,
+                             AudioUnitScope inScope,
+                             AudioUnitElement inElement)
+    {
+      OSStatus err = noErr;
+      osx_source* This = static_cast<osx_source*>(inRefCon);
+      AudioStreamBasicDescription asbd;
+
+      std::cerr << "a_o_s::UnitListener" << std::endl;
+
+      // 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");
+
+      std::cerr << "UnitListener: Input Source changed." << std::endl
+                << "Old Source Output Info:" << std::endl;
+      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");
+
+      std::cerr << "New Source Output Info:" << std::endl;
+      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 /* _OSX_DO_LISTENERS_ */
+
+  } /* namespace audio */
+} /* namespace gr */
diff --git a/gr-audio/lib/osx/osx_source.h b/gr-audio/lib/osx/osx_source.h
new file mode 100644
index 0000000000..9315c8e44e
--- /dev/null
+++ b/gr-audio/lib/osx/osx_source.h
@@ -0,0 +1,121 @@
+/* -*- c++ -*- */
+/*
+ * Copyright 2006-2011,2013 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.
+ */
+
+#ifndef INCLUDED_AUDIO_OSX_SOURCE_H
+#define INCLUDED_AUDIO_OSX_SOURCE_H
+
+#include <audio/source.h>
+#include <string>
+#include <AudioToolbox/AudioToolbox.h>
+#include <AudioUnit/AudioUnit.h>
+#include <circular_buffer.h>
+
+namespace gr {
+  namespace audio {
+
+    /*!
+     * \brief audio source using OSX
+     * \ingroup audio_blk
+     *
+     * Input signature is one or two streams of floats.
+     * Samples must be in the range [-1,1].
+     */
+    class osx_source : public source
+    {
+      Float64             d_deviceSampleRate, d_outputSampleRate;
+      int                 d_channel_config;
+      UInt32              d_inputBufferSizeFrames, d_inputBufferSizeBytes;
+      UInt32              d_outputBufferSizeFrames, d_outputBufferSizeBytes;
+      UInt32              d_deviceBufferSizeFrames, d_deviceBufferSizeBytes;
+      UInt32              d_leadSizeFrames, d_leadSizeBytes;
+      UInt32              d_trailSizeFrames, d_trailSizeBytes;
+      UInt32              d_extraBufferSizeFrames, d_extraBufferSizeBytes;
+      UInt32              d_queueSampleCount, d_max_sample_count;
+      UInt32              d_n_AvailableInputFrames, d_n_ActualInputFrames;
+      UInt32              d_n_user_channels, d_n_max_channels, d_n_deviceChannels;
+      bool                d_do_block, d_passThrough, d_waiting_for_data;
+      gruel::mutex*       d_internal;
+      gruel::condition_variable* d_cond_data;
+      circular_buffer<float>** d_buffers;
+
+      // AudioUnits and Such
+      AudioUnit           d_InputAU;
+      AudioBufferList*    d_InputBuffer;
+      AudioBufferList*    d_OutputBuffer;
+      AudioConverterRef   d_AudioConverter;
+
+    public:
+      osx_source(int sample_rate = 44100,
+                 const std::string device_name = "",
+                 bool do_block = true,
+                 int channel_config = -1,
+                 int max_sample_count = -1);
+
+      ~osx_source();
+
+      bool start();
+      bool stop();
+      bool IsRunning();
+
+      bool check_topology(int ninputs, int noutputs);
+
+      int work(int noutput_items,
+               gr_vector_const_void_star &input_items,
+               gr_vector_void_star &output_items);
+
+    private:
+      void SetDefaultInputDeviceAsCurrent();
+
+      void AllocAudioBufferList(AudioBufferList** t_ABL,
+                                UInt32 n_channels,
+                                UInt32 inputBufferSizeBytes);
+
+      void FreeAudioBufferList(AudioBufferList** t_ABL);
+
+      static OSStatus ConverterCallback(AudioConverterRef inAudioConverter,
+                                        UInt32* ioNumberDataPackets,
+                                        AudioBufferList* ioData,
+                                        AudioStreamPacketDescription** outASPD,
+                                        void* inUserData);
+
+      static OSStatus AUInputCallback(void *inRefCon,
+                                      AudioUnitRenderActionFlags *ioActionFlags,
+                                      const AudioTimeStamp *inTimeStamp,
+                                      UInt32 inBusNumber,
+                                      UInt32 inNumberFrames,
+                                      AudioBufferList *ioData);
+#if _OSX_DO_LISTENERS_
+      static OSStatus UnitListener(void *inRefCon,
+                                   AudioUnit ci,
+                                   AudioUnitPropertyID inID,
+                                   AudioUnitScope inScope,
+                                   AudioUnitElement inElement);
+
+      static OSStatus HardwareListener(AudioHardwarePropertyID inPropertyID,
+                                       void *inClientData);
+#endif
+    };
+
+  } /* namespace audio */
+} /* namespace gr */
+
+#endif /* INCLUDED_AUDIO_OSX_SOURCE_H */
diff --git a/gr-audio/lib/portaudio/audio_portaudio_sink.cc b/gr-audio/lib/portaudio/audio_portaudio_sink.cc
deleted file mode 100644
index af7f1e48c5..0000000000
--- a/gr-audio/lib/portaudio/audio_portaudio_sink.cc
+++ /dev/null
@@ -1,362 +0,0 @@
-/* -*- c++ -*- */
-/*
- * Copyright 2006-2011 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 he 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
-
-#include "gr_audio_registry.h"
-#include <audio_portaudio_sink.h>
-#include <gr_io_signature.h>
-#include <gr_prefs.h>
-#include <stdio.h>
-#include <iostream>
-#include <unistd.h>
-#include <stdexcept>
-#include <gri_portaudio.h>
-#include <string.h>
-
-AUDIO_REGISTER_SINK(REG_PRIO_MED, portaudio)(
-    int sampling_rate, const std::string &device_name, bool ok_to_block
-){
-    return audio_sink::sptr(new audio_portaudio_sink(sampling_rate, device_name, ok_to_block));
-}
-
-//#define	LOGGING   0		// define to 0 or 1
-
-#define SAMPLE_FORMAT 		paFloat32
-typedef float sample_t;
-
-// Number of portaudio buffers in the ringbuffer
-static const unsigned int N_BUFFERS = 4;
-
-static std::string
-default_device_name ()
-{
-  return gr_prefs::singleton()->get_string("audio_portaudio", "default_output_device", "");
-}
-
-void
-audio_portaudio_sink::create_ringbuffer(void)
-{
-  int bufsize_samples = d_portaudio_buffer_size_frames * d_output_parameters.channelCount;
-
-  if (d_verbose)
-    fprintf(stderr,"ring buffer size  = %d frames\n",
-	    N_BUFFERS*bufsize_samples/d_output_parameters.channelCount);
-
-  // FYI, the buffer indicies are in units of samples.
-  d_writer = gr_make_buffer(N_BUFFERS * bufsize_samples, sizeof(sample_t));
-  d_reader = gr_buffer_add_reader(d_writer, 0);
-}
-
-/*
- * This routine will be called by the PortAudio engine when audio is needed.
- * It may called at interrupt level on some machines so don't do anything
- * that could mess up the system like calling malloc() or free().
- *
- * Our job is to write framesPerBuffer frames into outputBuffer.
- */
-int
-portaudio_sink_callback (const void *inputBuffer,
-			 void *outputBuffer,
-			 unsigned long framesPerBuffer,
-			 const PaStreamCallbackTimeInfo* timeInfo,
-			 PaStreamCallbackFlags statusFlags,
-			 void *arg)
-{
-  audio_portaudio_sink *self = (audio_portaudio_sink *)arg;
-  int nreqd_samples =
-    framesPerBuffer * self->d_output_parameters.channelCount;
-
-  int navail_samples = self->d_reader->items_available();
-
-  if (nreqd_samples <= navail_samples) {  // We've got enough data...
-    {
-      gruel::scoped_lock guard(self->d_ringbuffer_mutex);
-
-      memcpy(outputBuffer,
-	     self->d_reader->read_pointer(),
-	     nreqd_samples * sizeof(sample_t));
-      self->d_reader->update_read_pointer(nreqd_samples);
-
-      self->d_ringbuffer_ready = true;
-    }
-
-    // Tell the sink thread there is new room in the ringbuffer.
-    self->d_ringbuffer_cond.notify_one();
-    return paContinue;
-  }
-
-  else {			// underrun
-    self->d_nunderuns++;
-    ssize_t r = ::write(2, "aU", 2);	// FIXME change to non-blocking call
-    if(r == -1) {
-      perror("audio_portaudio_source::portaudio_source_callback write error to stderr.");
-    }
-
-    // FIXME we should transfer what we've got and pad the rest
-    memset(outputBuffer, 0, nreqd_samples * sizeof(sample_t));
-
-    self->d_ringbuffer_ready = true;
-    self->d_ringbuffer_cond.notify_one();  // Tell the sink to get going!
-
-    return paContinue;
-  }
-}
-
-
-// ----------------------------------------------------------------
-
-audio_portaudio_sink::audio_portaudio_sink(int sampling_rate,
-					   const std::string device_name,
-					   bool ok_to_block)
-  : gr_sync_block ("audio_portaudio_sink",
-		   gr_make_io_signature(0, 0, 0),
-		   gr_make_io_signature(0, 0, 0)),
-    d_sampling_rate(sampling_rate),
-    d_device_name(device_name.empty() ? default_device_name() : device_name),
-    d_ok_to_block(ok_to_block),
-    d_verbose(gr_prefs::singleton()->get_bool("audio_portaudio", "verbose", false)),
-    d_portaudio_buffer_size_frames(0),
-    d_stream(0),
-    d_ringbuffer_mutex(),
-    d_ringbuffer_cond(),
-    d_ringbuffer_ready(false),
-    d_nunderuns(0)
-{
-  memset(&d_output_parameters, 0, sizeof(d_output_parameters));
-  //if (LOGGING)
-  //  d_log = gri_logger::singleton();
-
-  PaError 	      err;
-  int     	      i, numDevices;
-  PaDeviceIndex	      device = 0;
-  const PaDeviceInfo *deviceInfo = NULL;
-
-  err = Pa_Initialize();
-  if (err != paNoError) {
-    bail ("Initialize failed", err);
-  }
-
-  if (d_verbose)
-    gri_print_devices();
-
-  numDevices = Pa_GetDeviceCount();
-  if (numDevices < 0)
-    bail("Pa Device count failed", 0);
-  if (numDevices == 0)
-    bail("no devices available", 0);
-
-  if (d_device_name.empty())
-  {
-    // FIXME Get smarter about picking something
-    fprintf(stderr,"\nUsing Default Device\n");
-    device = Pa_GetDefaultOutputDevice();
-    deviceInfo = Pa_GetDeviceInfo(device);
-    fprintf(stderr,"%s is the chosen device using %s as the host\n",
-	    deviceInfo->name, Pa_GetHostApiInfo(deviceInfo->hostApi)->name);
-  }
-  else
-  {
-    bool found = false;
-    fprintf(stderr,"\nTest Devices\n");
-    for (i=0;i<numDevices;i++) {
-      deviceInfo = Pa_GetDeviceInfo( i );
-      fprintf(stderr,"Testing device name: %s",deviceInfo->name);
-      if (deviceInfo->maxOutputChannels <= 0) {
-	fprintf(stderr,"\n");
-	continue;
-      }
-      if (strstr(deviceInfo->name, d_device_name.c_str())){
-	fprintf(stderr,"  Chosen!\n");
-	device = i;
-	fprintf(stderr,"%s using %s as the host\n",d_device_name.c_str(),
-		Pa_GetHostApiInfo(deviceInfo->hostApi)->name), fflush(stderr);
-	found = true;
-	deviceInfo = Pa_GetDeviceInfo(device);
-	i = numDevices; 	// force loop exit
-      }
-      else
-	fprintf(stderr,"\n"),fflush(stderr);
-    }
-
-    if (!found){
-      bail("Failed to find specified device name", 0);
-      exit(1);
-    }
-  }
-
-
-  d_output_parameters.device = device;
-  d_output_parameters.channelCount     = deviceInfo->maxOutputChannels;
-  d_output_parameters.sampleFormat     = SAMPLE_FORMAT;
-  d_output_parameters.suggestedLatency = deviceInfo->defaultLowOutputLatency;
-  d_output_parameters.hostApiSpecificStreamInfo = NULL;
-
-  // We fill in the real channelCount in check_topology when we know
-  // how many inputs are connected to us.
-
-  // Now that we know the maximum number of channels (allegedly)
-  // supported by the h/w, we can compute a reasonable input
-  // signature.  The portaudio specs say that they'll accept any
-  // number of channels from 1 to max.
-  set_input_signature(gr_make_io_signature(1, deviceInfo->maxOutputChannels,
-					   sizeof (sample_t)));
-}
-
-
-bool
-audio_portaudio_sink::check_topology (int ninputs, int noutputs)
-{
-  PaError err;
-
-  if (Pa_IsStreamActive(d_stream))
-  {
-      Pa_CloseStream(d_stream);
-      d_stream = 0;
-      d_reader.reset();		// boost::shared_ptr for d_reader = 0
-      d_writer.reset();		// boost::shared_ptr for d_write = 0
-  }
-
-  d_output_parameters.channelCount = ninputs;	// # of channels we're really using
-
-#if 1
-  d_portaudio_buffer_size_frames = (int)(0.0213333333  * d_sampling_rate + 0.5);  // Force 1024 frame buffers at 48000
-  fprintf(stderr, "Latency = %8.5f, requested sampling_rate = %g\n", // Force latency to 21.3333333.. ms
-	  0.0213333333, (double)d_sampling_rate);
-#endif
-  err = Pa_OpenStream(&d_stream,
-		      NULL,			// No input
-		      &d_output_parameters,
-		      d_sampling_rate,
-		      d_portaudio_buffer_size_frames,
-		      paClipOff,
-		      &portaudio_sink_callback,
-		      (void*)this);
-
-  if (err != paNoError) {
-    output_error_msg ("OpenStream failed", err);
-    return false;
-  }
-
-#if 0
-  const PaStreamInfo *psi = Pa_GetStreamInfo(d_stream);
-
-  d_portaudio_buffer_size_frames = (int)(d_output_parameters.suggestedLatency  * psi->sampleRate);
-  fprintf(stderr, "Latency = %7.4f, psi->sampleRate = %g\n",
-	  d_output_parameters.suggestedLatency, psi->sampleRate);
-#endif
-
-  fprintf(stderr, "d_portaudio_buffer_size_frames = %d\n", d_portaudio_buffer_size_frames);
-
-  assert(d_portaudio_buffer_size_frames != 0);
-
-  create_ringbuffer();
-
-  err = Pa_StartStream(d_stream);
-  if (err != paNoError) {
-    output_error_msg ("StartStream failed", err);
-    return false;
-  }
-
-  return true;
-}
-
-audio_portaudio_sink::~audio_portaudio_sink ()
-{
-  Pa_StopStream(d_stream);	// wait for output to drain
-  Pa_CloseStream(d_stream);
-  Pa_Terminate();
-}
-
-/*
- * This version consumes everything sent to it, blocking if required.
- * I think this will allow us better control of the total buffering/latency
- * in the audio path.
- */
-int
-audio_portaudio_sink::work (int noutput_items,
-			    gr_vector_const_void_star &input_items,
-			    gr_vector_void_star &output_items)
-{
-  const float **in = (const float **) &input_items[0];
-  const unsigned nchan = d_output_parameters.channelCount; // # of channels == samples/frame
-
-  int k;
-
-  for (k = 0; k < noutput_items; ){
-    int nframes = d_writer->space_available() / nchan;	// How much space in ringbuffer
-    if (nframes == 0){			// no room...
-      if (d_ok_to_block){
-	{
-	  gruel::scoped_lock guard(d_ringbuffer_mutex);
-	  while (!d_ringbuffer_ready)
-	    d_ringbuffer_cond.wait(guard);
-	}
-
-	continue;
-      }
-      else {
-	// There's no room and we're not allowed to block.
-	// (A USRP is most likely controlling the pacing through the pipeline.)
-	// We drop the samples on the ground, and say we processed them all ;)
-	//
-	// FIXME, there's probably room for a bit more finesse here.
-	return noutput_items;
-      }
-    }
-
-    // We can write the smaller of the request and the room we've got
-    {
-      gruel::scoped_lock guard(d_ringbuffer_mutex);
-
-      int nf = std::min(noutput_items - k, nframes);
-      float *p = (float *) d_writer->write_pointer();
-
-      for (int i = 0; i < nf; i++)
-	for (unsigned int c = 0; c < nchan; c++)
-	  *p++ = in[c][k + i];
-
-      d_writer->update_write_pointer(nf * nchan);
-      k += nf;
-
-      d_ringbuffer_ready = false;
-    }
-  }
-
-  return k;  // tell how many we actually did
-}
-
-void
-audio_portaudio_sink::output_error_msg (const char *msg, int err)
-{
-  fprintf (stderr, "audio_portaudio_sink[%s]: %s: %s\n",
-	   d_device_name.c_str (), msg, Pa_GetErrorText(err));
-}
-
-void
-audio_portaudio_sink::bail (const char *msg, int err) throw (std::runtime_error)
-{
-  output_error_msg (msg, err);
-  throw std::runtime_error ("audio_portaudio_sink");
-}
diff --git a/gr-audio/lib/portaudio/audio_portaudio_sink.h b/gr-audio/lib/portaudio/audio_portaudio_sink.h
deleted file mode 100644
index 4b21560d6c..0000000000
--- a/gr-audio/lib/portaudio/audio_portaudio_sink.h
+++ /dev/null
@@ -1,86 +0,0 @@
-/* -*- c++ -*- */
-/*
- * Copyright 2006-2011,2013 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.
- */
-#ifndef INCLUDED_AUDIO_PORTAUDIO_SINK_H
-#define INCLUDED_AUDIO_PORTAUDIO_SINK_H
-
-#include <audio/sink.h>
-#include <gr_buffer.h>
-#include <gruel/thread.h>
-#include <string>
-#include <portaudio.h>
-#include <stdexcept>
-//#include <gri_logger.h>
-
-PaStreamCallback portaudio_sink_callback;
-
-
-/*!
- * \brief Audio sink using PORTAUDIO
- * \ingroup audio_blk
- *
- * Input samples must be in the range [-1,1].
- */
-class audio_portaudio_sink : public audio_sink {
-
-  friend PaStreamCallback portaudio_sink_callback;
-
-
-  unsigned int		d_sampling_rate;
-  std::string		d_device_name;
-  bool			d_ok_to_block;
-  bool			d_verbose;
-
-  unsigned int		d_portaudio_buffer_size_frames;	// number of frames in a portaudio buffer
-
-  PaStream		*d_stream;
-  PaStreamParameters	d_output_parameters;
-
-  gr_buffer_sptr	d_writer;		// buffer used between work and callback
-  gr_buffer_reader_sptr	d_reader;
-
-  gruel::mutex          d_ringbuffer_mutex;
-  gruel::condition_variable d_ringbuffer_cond;
-  bool                  d_ringbuffer_ready;
-
-  // random stats
-  int			d_nunderuns;		// count of underruns
-  //gri_logger_sptr	d_log;			// handle to non-blocking logging instance
-
-  void output_error_msg (const char *msg, int err);
-  void bail (const char *msg, int err) throw (std::runtime_error);
-  void create_ringbuffer();
-
-
-public:
-  audio_portaudio_sink (int sampling_rate, const std::string device_name,
-			bool ok_to_block);
-
-  ~audio_portaudio_sink ();
-
-  bool check_topology (int ninputs, int noutputs);
-
-  int work (int noutput_items,
-	    gr_vector_const_void_star &input_items,
-	    gr_vector_void_star &output_items);
-};
-
-#endif /* INCLUDED_AUDIO_PORTAUDIO_SINK_H */
diff --git a/gr-audio/lib/portaudio/audio_portaudio_source.cc b/gr-audio/lib/portaudio/audio_portaudio_source.cc
deleted file mode 100644
index ddb1a6fb65..0000000000
--- a/gr-audio/lib/portaudio/audio_portaudio_source.cc
+++ /dev/null
@@ -1,374 +0,0 @@
-/* -*- c++ -*- */
-/*
- * Copyright 2006-2011 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 he 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
-
-#include "gr_audio_registry.h"
-#include <audio_portaudio_source.h>
-#include <gr_io_signature.h>
-#include <gr_prefs.h>
-#include <stdio.h>
-#include <iostream>
-#include <unistd.h>
-#include <stdexcept>
-#include <gri_portaudio.h>
-#include <string.h>
-
-AUDIO_REGISTER_SOURCE(REG_PRIO_MED, portaudio)(
-    int sampling_rate, const std::string &device_name, bool ok_to_block
-){
-    return audio_source::sptr(new audio_portaudio_source(sampling_rate, device_name, ok_to_block));
-}
-
-//#define	LOGGING 0		// define to 0 or 1
-
-#define SAMPLE_FORMAT 		paFloat32
-typedef float sample_t;
-
-// Number of portaudio buffers in the ringbuffer
-static const unsigned int N_BUFFERS = 4;
-
-static std::string
-default_device_name ()
-{
-  return gr_prefs::singleton()->get_string("audio_portaudio", "default_input_device", "");
-}
-
-void
-audio_portaudio_source::create_ringbuffer(void)
-{
-  int bufsize_samples = d_portaudio_buffer_size_frames * d_input_parameters.channelCount;
-
-  if (d_verbose)
-    fprintf(stderr, "ring buffer size  = %d frames\n",
-	    N_BUFFERS*bufsize_samples/d_input_parameters.channelCount);
-
-  // FYI, the buffer indicies are in units of samples.
-  d_writer = gr_make_buffer(N_BUFFERS * bufsize_samples, sizeof(sample_t));
-  d_reader = gr_buffer_add_reader(d_writer, 0);
-}
-
-/*
- * This routine will be called by the PortAudio engine when audio is needed.
- * It may called at interrupt level on some machines so don't do anything
- * that could mess up the system like calling malloc() or free().
- *
- * Our job is to copy framesPerBuffer frames from inputBuffer.
- */
-int
-portaudio_source_callback (const void *inputBuffer,
-			   void *outputBuffer,
-			   unsigned long framesPerBuffer,
-			   const PaStreamCallbackTimeInfo* timeInfo,
-			   PaStreamCallbackFlags statusFlags,
-			   void *arg)
-{
-  audio_portaudio_source *self = (audio_portaudio_source *)arg;
-  int nchan = self->d_input_parameters.channelCount;
-  int nframes_to_copy = framesPerBuffer;
-  int nframes_room = self->d_writer->space_available() / nchan;
-
-  if (nframes_to_copy <= nframes_room){  // We've got room for the data ..
-    //if (LOGGING)
-    //  self->d_log->printf("PAsrc  cb: f/b = %4ld\n", framesPerBuffer);
-
-    // copy from input buffer to ringbuffer
-    {
-      gruel::scoped_lock(d_ringbuffer_mutex);
-
-      memcpy(self->d_writer->write_pointer(),
-	     inputBuffer,
-	     nframes_to_copy * nchan * sizeof(sample_t));
-      self->d_writer->update_write_pointer(nframes_to_copy * nchan);
-
-      // Tell the source thread there is new data in the ringbuffer.
-      self->d_ringbuffer_ready = true;
-    }
-
-    self->d_ringbuffer_cond.notify_one();
-    return paContinue;
-  }
-
-  else {			// overrun
-    self->d_noverruns++;
-    ssize_t r = ::write(2, "aO", 2);	// FIXME change to non-blocking call
-    if(r == -1) {
-      perror("audio_portaudio_source::portaudio_source_callback write error to stderr.");
-    }
-
-    self->d_ringbuffer_ready = false;
-    self->d_ringbuffer_cond.notify_one();  // Tell the sink to get going!
-    return paContinue;
-  }
-}
-
-
-// ----------------------------------------------------------------
-
-audio_portaudio_source::audio_portaudio_source(int sampling_rate,
-					       const std::string device_name,
-					       bool ok_to_block)
-  : gr_sync_block ("audio_portaudio_source",
-		   gr_make_io_signature(0, 0, 0),
-		   gr_make_io_signature(0, 0, 0)),
-    d_sampling_rate(sampling_rate),
-    d_device_name(device_name.empty() ? default_device_name() : device_name),
-    d_ok_to_block(ok_to_block),
-    d_verbose(gr_prefs::singleton()->get_bool("audio_portaudio", "verbose", false)),
-    d_portaudio_buffer_size_frames(0),
-    d_stream(0),
-    d_ringbuffer_mutex(),
-    d_ringbuffer_cond(),
-    d_ringbuffer_ready(false),
-    d_noverruns(0)
-{
-  memset(&d_input_parameters, 0, sizeof(d_input_parameters));
-  //if (LOGGING)
-  //  d_log = gri_logger::singleton();
-
-  PaError 	      err;
-  int     	      i, numDevices;
-  PaDeviceIndex	      device = 0;
-  const PaDeviceInfo *deviceInfo = NULL;
-
-
-  err = Pa_Initialize();
-  if (err != paNoError) {
-    bail ("Initialize failed", err);
-  }
-
-  if (d_verbose)
-    gri_print_devices();
-
-  numDevices = Pa_GetDeviceCount();
-  if (numDevices < 0)
-    bail("Pa Device count failed", 0);
-  if (numDevices == 0)
-    bail("no devices available", 0);
-
-  if (d_device_name.empty())
-  {
-    // FIXME Get smarter about picking something
-    device = Pa_GetDefaultInputDevice();
-    deviceInfo = Pa_GetDeviceInfo(device);
-    fprintf(stderr,"%s is the chosen device using %s as the host\n",
-	    deviceInfo->name, Pa_GetHostApiInfo(deviceInfo->hostApi)->name);
-  }
-  else
-  {
-    bool found = false;
-
-    for (i=0;i<numDevices;i++) {
-      deviceInfo = Pa_GetDeviceInfo( i );
-      fprintf(stderr,"Testing device name: %s",deviceInfo->name);
-      if (deviceInfo->maxInputChannels <= 0) {
-	fprintf(stderr,"\n");
-	continue;
-      }
-      if (strstr(deviceInfo->name, d_device_name.c_str())){
-	fprintf(stderr,"  Chosen!\n");
-	device = i;
-	fprintf(stderr,"%s using %s as the host\n",d_device_name.c_str(),
-		Pa_GetHostApiInfo(deviceInfo->hostApi)->name), fflush(stderr);
-	found = true;
-	deviceInfo = Pa_GetDeviceInfo(device);
-	i = numDevices; 	// force loop exit
-      }
-      else
-	fprintf(stderr,"\n"),fflush(stderr);
-    }
-
-    if (!found){
-      bail("Failed to find specified device name", 0);
-    }
-  }
-
-
-  d_input_parameters.device = device;
-  d_input_parameters.channelCount     = deviceInfo->maxInputChannels;
-  d_input_parameters.sampleFormat     = SAMPLE_FORMAT;
-  d_input_parameters.suggestedLatency = deviceInfo->defaultLowInputLatency;
-  d_input_parameters.hostApiSpecificStreamInfo = NULL;
-
-  // We fill in the real channelCount in check_topology when we know
-  // how many inputs are connected to us.
-
-  // Now that we know the maximum number of channels (allegedly)
-  // supported by the h/w, we can compute a reasonable output
-  // signature.  The portaudio specs say that they'll accept any
-  // number of channels from 1 to max.
-  set_output_signature(gr_make_io_signature(1, deviceInfo->maxInputChannels,
-					    sizeof (sample_t)));
-}
-
-
-bool
-audio_portaudio_source::check_topology (int ninputs, int noutputs)
-{
-  PaError err;
-
-  if (Pa_IsStreamActive(d_stream))
-  {
-      Pa_CloseStream(d_stream);
-      d_stream = 0;
-      d_reader.reset();		// boost::shared_ptr for d_reader = 0
-      d_writer.reset();		// boost::shared_ptr for d_write = 0
-  }
-
-  d_input_parameters.channelCount = noutputs;	// # of channels we're really using
-
-#if 1
-  d_portaudio_buffer_size_frames = (int)(0.0213333333  * d_sampling_rate + 0.5);  // Force 512 frame buffers at 48000
-  fprintf(stderr, "Latency = %8.5f, requested sampling_rate = %g\n", // Force latency to 21.3333333.. ms
-	  0.0213333333, (double)d_sampling_rate);
-#endif
-  err = Pa_OpenStream(&d_stream,
-		      &d_input_parameters,
-		      NULL,			// No output
-		      d_sampling_rate,
-		      d_portaudio_buffer_size_frames,
-		      paClipOff,
-		      &portaudio_source_callback,
-		      (void*)this);
-
-  if (err != paNoError) {
-    output_error_msg ("OpenStream failed", err);
-    return false;
-  }
-
-#if 0
-  const PaStreamInfo *psi = Pa_GetStreamInfo(d_stream);
-
-  d_portaudio_buffer_size_frames = (int)(d_input_parameters.suggestedLatency  * psi->sampleRate);
-  fprintf(stderr, "Latency = %7.4f, psi->sampleRate = %g\n",
-	  d_input_parameters.suggestedLatency, psi->sampleRate);
-#endif
-
-  fprintf(stderr, "d_portaudio_buffer_size_frames = %d\n", d_portaudio_buffer_size_frames);
-
-  assert(d_portaudio_buffer_size_frames != 0);
-
-  create_ringbuffer();
-
-  err = Pa_StartStream(d_stream);
-  if (err != paNoError) {
-    output_error_msg ("StartStream failed", err);
-    return false;
-  }
-
-  return true;
-}
-
-audio_portaudio_source::~audio_portaudio_source ()
-{
-  Pa_StopStream(d_stream);	// wait for output to drain
-  Pa_CloseStream(d_stream);
-  Pa_Terminate();
-}
-
-int
-audio_portaudio_source::work (int noutput_items,
-			      gr_vector_const_void_star &input_items,
-			      gr_vector_void_star &output_items)
-{
-  float **out = (float **) &output_items[0];
-  const unsigned nchan = d_input_parameters.channelCount; // # of channels == samples/frame
-
-  int k;
-  for (k = 0; k < noutput_items; ){
-
-    int nframes = d_reader->items_available() / nchan;	// # of frames in ringbuffer
-    if (nframes == 0){		// no data right now...
-      if (k > 0)      		// If we've produced anything so far, return that
-	return k;
-
-      if (d_ok_to_block) {
-	gruel:: scoped_lock guard(d_ringbuffer_mutex);
-	while (d_ringbuffer_ready == false)
-	  d_ringbuffer_cond.wait(guard);	// block here, then try again
-	continue;
-      }
-
-      assert(k == 0);
-
-      // There's no data and we're not allowed to block.
-      // (A USRP is most likely controlling the pacing through the pipeline.)
-      // This is an underun.  The scheduler wouldn't have called us if it
-      // had anything better to do.  Thus we really need to produce some amount
-      // of "fill".
-      //
-      // There are lots of options for comfort noise, etc.
-      // FIXME We'll fill with zeros for now.  Yes, it will "click"...
-
-      // Fill with some frames of zeros
-      {
-	gruel::scoped_lock guard(d_ringbuffer_mutex);
-
-	int nf = std::min(noutput_items - k, (int) d_portaudio_buffer_size_frames);
-	for (int i = 0; i < nf; i++){
-	  for (unsigned int c = 0; c < nchan; c++){
-	    out[c][k + i] = 0;
-	  }
-	}
-	k += nf;
-
-	d_ringbuffer_ready = false;
-	return k;
-      }
-    }
-
-    // We can read the smaller of the request and what's in the buffer.
-    {
-      gruel::scoped_lock guard(d_ringbuffer_mutex);
-
-      int nf = std::min(noutput_items - k, nframes);
-
-      const float *p = (const float *) d_reader->read_pointer();
-      for (int i = 0; i < nf; i++){
-	for (unsigned int c = 0; c < nchan; c++){
-	  out[c][k + i] = *p++;
-	}
-      }
-      d_reader->update_read_pointer(nf * nchan);
-      k += nf;
-      d_ringbuffer_ready = false;
-    }
-  }
-
-  return k;  // tell how many we actually did
-}
-
-void
-audio_portaudio_source::output_error_msg (const char *msg, int err)
-{
-  fprintf (stderr, "audio_portaudio_source[%s]: %s: %s\n",
-	   d_device_name.c_str (), msg, Pa_GetErrorText(err));
-}
-
-void
-audio_portaudio_source::bail (const char *msg, int err) throw (std::runtime_error)
-{
-  output_error_msg (msg, err);
-  throw std::runtime_error ("audio_portaudio_source");
-}
diff --git a/gr-audio/lib/portaudio/audio_portaudio_source.h b/gr-audio/lib/portaudio/audio_portaudio_source.h
deleted file mode 100644
index 67a83f1fe5..0000000000
--- a/gr-audio/lib/portaudio/audio_portaudio_source.h
+++ /dev/null
@@ -1,84 +0,0 @@
-/* -*- c++ -*- */
-/*
- * Copyright 2006-2011,2013 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.
- */
-#ifndef INCLUDED_AUDIO_PORTAUDIO_SOURCE_H
-#define INCLUDED_AUDIO_PORTAUDIO_SOURCE_H
-
-#include <audio/source.h>
-#include <gr_buffer.h>
-#include <gruel/thread.h>
-#include <string>
-#include <portaudio.h>
-#include <stdexcept>
-
-PaStreamCallback portaudio_source_callback;
-
-
-/*!
- * \brief Audio source using PORTAUDIO
- * \ingroup audio_blk
- *
- * Input samples must be in the range [-1,1].
- */
-class audio_portaudio_source : public audio_source {
-
-  friend PaStreamCallback portaudio_source_callback;
-
-
-  unsigned int		d_sampling_rate;
-  std::string		d_device_name;
-  bool			d_ok_to_block;
-  bool			d_verbose;
-
-  unsigned int		d_portaudio_buffer_size_frames;	// number of frames in a portaudio buffer
-
-  PaStream		*d_stream;
-  PaStreamParameters	d_input_parameters;
-
-  gr_buffer_sptr	d_writer;		// buffer used between work and callback
-  gr_buffer_reader_sptr	d_reader;
-
-  gruel::mutex          d_ringbuffer_mutex;
-  gruel::condition_variable d_ringbuffer_cond;
-  bool                  d_ringbuffer_ready;
-
-  // random stats
-  int			d_noverruns;		// count of overruns
-
-  void output_error_msg (const char *msg, int err);
-  void bail (const char *msg, int err) throw (std::runtime_error);
-  void create_ringbuffer();
-
-
-public:
-  audio_portaudio_source (int sampling_rate, const std::string device_name,
-			  bool ok_to_block);
-
-  ~audio_portaudio_source ();
-
-  bool check_topology (int ninputs, int noutputs);
-
-  int work (int ninput_items,
-	    gr_vector_const_void_star &input_items,
-	    gr_vector_void_star &output_items);
-};
-
-#endif /* INCLUDED_AUDIO_PORTAUDIO_SOURCE_H */
diff --git a/gr-audio/lib/portaudio/gri_portaudio.cc b/gr-audio/lib/portaudio/gri_portaudio.cc
deleted file mode 100644
index 66f3d46472..0000000000
--- a/gr-audio/lib/portaudio/gri_portaudio.cc
+++ /dev/null
@@ -1,111 +0,0 @@
-/* -*- 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
-
-#include <gri_portaudio.h>
-#include <portaudio.h>
-#include <string.h>
-
-
-PaDeviceIndex
-gri_pa_find_device_by_name(const char *name)
-{
-  int   i;
-  int   numDevices;
-  const PaDeviceInfo *pdi;
-  int   len = strlen( name );
-  PaDeviceIndex result = paNoDevice;
-  numDevices = Pa_GetDeviceCount();
-  for( i=0; i<numDevices; i++ )
-  {
-    pdi = Pa_GetDeviceInfo( i );
-    if( strncmp( name, pdi->name, len ) == 0 )
-    {
-      result = i;
-      break;
-    }
-  }
-  return result;
-}
-
-
-void
-gri_print_devices()
-{
-  int     numDevices, defaultDisplayed, myDevice=0;
-  const   PaDeviceInfo *deviceInfo;
-
-  numDevices = Pa_GetDeviceCount();
-  if (numDevices < 0)
-    return;
-
-  printf("Number of devices found = %d\n", numDevices);
-
-  for (int i=0; i < numDevices; i++ ) {
-    deviceInfo = Pa_GetDeviceInfo( i );
-    printf( "--------------------------------------- device #%d\n", i );
-    /* Mark global and API specific default devices */
-    defaultDisplayed = 0;
-    if( i == Pa_GetDefaultInputDevice() )
-      {
-	myDevice = i;
-	printf( "[ Default Input" );
-	defaultDisplayed = 1;
-      }
-    else if( i == Pa_GetHostApiInfo( deviceInfo->hostApi )->defaultInputDevice )
-      {
-	const PaHostApiInfo *hostInfo = Pa_GetHostApiInfo( deviceInfo->hostApi );
-	printf( "[ Default %s Input", hostInfo->name );
-	defaultDisplayed = 1;
-      }
-
-    if( i == Pa_GetDefaultOutputDevice() )
-      {
-	printf( (defaultDisplayed ? "," : "[") );
-	printf( " Default Output" );
-	defaultDisplayed = 1;
-      }
-    else if( i == Pa_GetHostApiInfo( deviceInfo->hostApi )->defaultOutputDevice )
-      {
-	const PaHostApiInfo *hostInfo = Pa_GetHostApiInfo( deviceInfo->hostApi );
-	printf( (defaultDisplayed ? "," : "[") );
-	printf( " Default %s Output", hostInfo->name );
-	defaultDisplayed = 1;
-      }
-    if( defaultDisplayed )
-      printf( " ]\n" );
-
-    /* print device info fields */
-    printf( "Name                        = %s\n", deviceInfo->name );
-    printf( "Host API                    = %s\n",  Pa_GetHostApiInfo( deviceInfo->hostApi )->name );
-    printf( "Max inputs = %d", deviceInfo->maxInputChannels  );
-    printf( ", Max outputs = %d\n", deviceInfo->maxOutputChannels  );
-
-    printf( "Default low input latency   = %8.3f\n", deviceInfo->defaultLowInputLatency  );
-    printf( "Default low output latency  = %8.3f\n", deviceInfo->defaultLowOutputLatency  );
-    printf( "Default high input latency  = %8.3f\n", deviceInfo->defaultHighInputLatency  );
-    printf( "Default high output latency = %8.3f\n", deviceInfo->defaultHighOutputLatency  );
-  }
-}
diff --git a/gr-audio/lib/portaudio/portaudio_impl.cc b/gr-audio/lib/portaudio/portaudio_impl.cc
new file mode 100644
index 0000000000..ba37acc5fa
--- /dev/null
+++ b/gr-audio/lib/portaudio/portaudio_impl.cc
@@ -0,0 +1,108 @@
+/* -*- c++ -*- */
+/*
+ * Copyright 2006,2013 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
+
+#include <portaudio_impl.h>
+#include <portaudio.h>
+#include <string.h>
+
+namespace gr {
+  namespace audio {
+
+    PaDeviceIndex
+    pa_find_device_by_name(const char *name)
+    {
+      int i;
+      int numDevices;
+      const PaDeviceInfo *pdi;
+      int len = strlen(name);
+      PaDeviceIndex result = paNoDevice;
+      numDevices = Pa_GetDeviceCount();
+      for(i = 0; i < numDevices; i++) {
+        pdi = Pa_GetDeviceInfo(i);
+        if(strncmp(name, pdi->name, len) == 0) {
+          result = i;
+          break;
+        }
+      }
+      return result;
+    }
+
+    void
+    print_devices()
+    {
+      int numDevices, defaultDisplayed;
+      const PaDeviceInfo *deviceInfo;
+
+      numDevices = Pa_GetDeviceCount();
+      if(numDevices < 0)
+        return;
+
+      printf("Number of devices found = %d\n", numDevices);
+
+      for(int i = 0; i < numDevices; i++) {
+        deviceInfo = Pa_GetDeviceInfo(i);
+        printf("--------------------------------------- device #%d\n", i);
+        /* Mark global and API specific default devices */
+        defaultDisplayed = 0;
+        if(i == Pa_GetDefaultInputDevice()) {
+          printf("[ Default Input");
+          defaultDisplayed = 1;
+        }
+        else if(i == Pa_GetHostApiInfo(deviceInfo->hostApi)->defaultInputDevice) {
+          const PaHostApiInfo *hostInfo = Pa_GetHostApiInfo(deviceInfo->hostApi);
+          printf("[ Default %s Input", hostInfo->name);
+          defaultDisplayed = 1;
+        }
+
+        if(i == Pa_GetDefaultOutputDevice()) {
+          printf((defaultDisplayed ? "," : "["));
+          printf(" Default Output");
+          defaultDisplayed = 1;
+        }
+        else if(i == Pa_GetHostApiInfo(deviceInfo->hostApi)->defaultOutputDevice) {
+          const PaHostApiInfo *hostInfo = Pa_GetHostApiInfo(deviceInfo->hostApi);
+          printf((defaultDisplayed ? "," : "["));
+          printf(" Default %s Output", hostInfo->name);
+          defaultDisplayed = 1;
+        }
+        if(defaultDisplayed)
+          printf(" ]\n");
+
+        /* print device info fields */
+        printf("Name                        = %s\n", deviceInfo->name);
+        printf("Host API                    = %s\n", Pa_GetHostApiInfo(deviceInfo->hostApi)->name );
+        printf("Max inputs = %d", deviceInfo->maxInputChannels);
+        printf(", Max outputs = %d\n", deviceInfo->maxOutputChannels);
+
+        printf("Default low input latency   = %8.3f\n", deviceInfo->defaultLowInputLatency);
+        printf("Default low output latency  = %8.3f\n", deviceInfo->defaultLowOutputLatency);
+        printf("Default high input latency  = %8.3f\n", deviceInfo->defaultHighInputLatency);
+        printf("Default high output latency = %8.3f\n", deviceInfo->defaultHighOutputLatency);
+      }
+    }
+
+  } /* namespace audio */
+} /* namespace gr */
diff --git a/gr-audio/lib/portaudio/gri_portaudio.h b/gr-audio/lib/portaudio/portaudio_impl.h
index c3ea7d064d..0cb099e591 100644
--- a/gr-audio/lib/portaudio/gri_portaudio.h
+++ b/gr-audio/lib/portaudio/portaudio_impl.h
@@ -1,6 +1,6 @@
 /* -*- c++ -*- */
 /*
- * Copyright 2006 Free Software Foundation, Inc.
+ * Copyright 2006,2013 Free Software Foundation, Inc.
  *
  * This file is part of GNU Radio
  *
@@ -20,13 +20,19 @@
  * Boston, MA 02110-1301, USA.
  */
 
-#ifndef INCLUDED_GRI_PORTAUDIO_H
-#define INCLUDED_GRI_PORTAUDIO_H
+#ifndef INCLUDED_AUDIO_PORTAUDIO_IMPL_H
+#define INCLUDED_AUDIO_PORTAUDIO_IMPL_H
 
 #include <stdio.h>
 #include <portaudio.h>
 
-PaDeviceIndex gri_pa_find_device_by_name(const char *name);
-void gri_print_devices();
+namespace gr {
+  namespace audio {
 
-#endif /* INCLUDED_GRI_PORTAUDIO_H */
+    PaDeviceIndex pa_find_device_by_name(const char *name);
+    void print_devices();
+
+  } /* namespace audio */
+} /* namespace gr */
+
+#endif /* INCLUDED_AUDIO_PORTAUDIO_IMPL_H */
diff --git a/gr-audio/lib/portaudio/portaudio_sink.cc b/gr-audio/lib/portaudio/portaudio_sink.cc
new file mode 100644
index 0000000000..d1a3f5166c
--- /dev/null
+++ b/gr-audio/lib/portaudio/portaudio_sink.cc
@@ -0,0 +1,370 @@
+/* -*- c++ -*- */
+/*
+ * Copyright 2006-2011,2013 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 he 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
+
+#include "audio_registry.h"
+#include <portaudio_sink.h>
+#include <portaudio_impl.h>
+#include <gr_io_signature.h>
+#include <gr_prefs.h>
+#include <stdio.h>
+#include <iostream>
+#include <unistd.h>
+#include <stdexcept>
+#include <string.h>
+
+namespace gr {
+  namespace audio {
+
+    AUDIO_REGISTER_SINK(REG_PRIO_MED, portaudio)(int sampling_rate,
+                                                 const std::string &device_name,
+                                                 bool ok_to_block)
+    {
+      return sink::sptr
+        (new portaudio_sink(sampling_rate, device_name, ok_to_block));
+    }
+
+//#define LOGGING 0  // define to 0 or 1
+
+#define SAMPLE_FORMAT paFloat32
+
+    typedef float sample_t;
+
+    // Number of portaudio buffers in the ringbuffer
+    static const unsigned int N_BUFFERS = 4;
+
+    static std::string
+    default_device_name()
+    {
+      return gr_prefs::singleton()->get_string
+        ("audio_portaudio", "default_output_device", "");
+    }
+
+    void
+    portaudio_sink::create_ringbuffer(void)
+    {
+      int bufsize_samples = d_portaudio_buffer_size_frames * d_output_parameters.channelCount;
+
+      if(d_verbose) {
+        fprintf(stderr,"ring buffer size  = %d frames\n",
+                N_BUFFERS*bufsize_samples/d_output_parameters.channelCount);
+      }
+
+      // FYI, the buffer indicies are in units of samples.
+      d_writer = gr_make_buffer(N_BUFFERS * bufsize_samples, sizeof(sample_t));
+      d_reader = gr_buffer_add_reader(d_writer, 0);
+    }
+
+    /*
+     * This routine will be called by the PortAudio engine when audio is needed.
+     * It may called at interrupt level on some machines so don't do anything
+     * that could mess up the system like calling malloc() or free().
+     *
+     * Our job is to write framesPerBuffer frames into outputBuffer.
+     */
+    int
+    portaudio_sink_callback(const void *inputBuffer,
+                            void *outputBuffer,
+                            unsigned long framesPerBuffer,
+                            const PaStreamCallbackTimeInfo* timeInfo,
+                            PaStreamCallbackFlags statusFlags,
+                            void *arg)
+    {
+      portaudio_sink *self = (portaudio_sink *)arg;
+      int nreqd_samples =
+        framesPerBuffer * self->d_output_parameters.channelCount;
+
+      int navail_samples = self->d_reader->items_available();
+
+      if(nreqd_samples <= navail_samples) {  // We've got enough data...
+        {
+          gruel::scoped_lock guard(self->d_ringbuffer_mutex);
+
+          memcpy(outputBuffer,
+                 self->d_reader->read_pointer(),
+                 nreqd_samples * sizeof(sample_t));
+          self->d_reader->update_read_pointer(nreqd_samples);
+
+          self->d_ringbuffer_ready = true;
+        }
+
+        // Tell the sink thread there is new room in the ringbuffer.
+        self->d_ringbuffer_cond.notify_one();
+        return paContinue;
+      }
+
+      else {                   // underrun
+        self->d_nunderuns++;
+        ssize_t r = ::write(2, "aU", 2);  // FIXME change to non-blocking call
+        if(r == -1) {
+          perror("audio_portaudio_source::portaudio_source_callback write error to stderr.");
+        }
+
+        // FIXME we should transfer what we've got and pad the rest
+        memset(outputBuffer, 0, nreqd_samples * sizeof(sample_t));
+
+        self->d_ringbuffer_ready = true;
+        self->d_ringbuffer_cond.notify_one();  // Tell the sink to get going!
+
+        return paContinue;
+      }
+    }
+
+    // ----------------------------------------------------------------
+
+    portaudio_sink::portaudio_sink(int sampling_rate,
+                                   const std::string device_name,
+                                   bool ok_to_block)
+      : gr_sync_block("audio_portaudio_sink",
+                      gr_make_io_signature(0, 0, 0),
+                      gr_make_io_signature(0, 0, 0)),
+        d_sampling_rate(sampling_rate),
+        d_device_name(device_name.empty() ? default_device_name() : device_name),
+        d_ok_to_block(ok_to_block),
+        d_verbose(gr_prefs::singleton()->get_bool("audio_portaudio", "verbose", false)),
+        d_portaudio_buffer_size_frames(0),
+        d_stream(0),
+        d_ringbuffer_mutex(),
+        d_ringbuffer_cond(),
+        d_ringbuffer_ready(false),
+        d_nunderuns(0)
+    {
+      memset(&d_output_parameters, 0, sizeof(d_output_parameters));
+      //if(LOGGING)
+      //  d_log = gri_logger::singleton();
+
+      PaError err;
+      int i, numDevices;
+      PaDeviceIndex device = 0;
+      const PaDeviceInfo *deviceInfo = NULL;
+
+      err = Pa_Initialize();
+      if(err != paNoError) {
+        bail("Initialize failed", err);
+      }
+
+      if(d_verbose)
+        print_devices();
+
+      numDevices = Pa_GetDeviceCount();
+      if(numDevices < 0)
+        bail("Pa Device count failed", 0);
+      if(numDevices == 0)
+        bail("no devices available", 0);
+
+      if(d_device_name.empty()) {
+        // FIXME Get smarter about picking something
+        fprintf(stderr,"\nUsing Default Device\n");
+        device = Pa_GetDefaultOutputDevice();
+        deviceInfo = Pa_GetDeviceInfo(device);
+        fprintf(stderr,"%s is the chosen device using %s as the host\n",
+                deviceInfo->name, Pa_GetHostApiInfo(deviceInfo->hostApi)->name);
+      }
+      else {
+        bool found = false;
+        fprintf(stderr,"\nTest Devices\n");
+        for(i = 0; i < numDevices; i++) {
+          deviceInfo = Pa_GetDeviceInfo(i);
+          fprintf(stderr,"Testing device name: %s",deviceInfo->name);
+
+          if(deviceInfo->maxOutputChannels <= 0) {
+            fprintf(stderr,"\n");
+            continue;
+          }
+
+          if(strstr(deviceInfo->name, d_device_name.c_str())) {
+            fprintf(stderr,"  Chosen!\n");
+            device = i;
+            fprintf(stderr,"%s using %s as the host\n",d_device_name.c_str(),
+                    Pa_GetHostApiInfo(deviceInfo->hostApi)->name), fflush(stderr);
+            found = true;
+            deviceInfo = Pa_GetDeviceInfo(device);
+            i = numDevices;  // force loop exit
+          }
+          else
+            fprintf(stderr,"\n"), fflush(stderr);
+        }
+
+        if(!found) {
+          bail("Failed to find specified device name", 0);
+          exit(1);
+        }
+      }
+
+      d_output_parameters.device = device;
+      d_output_parameters.channelCount     = deviceInfo->maxOutputChannels;
+      d_output_parameters.sampleFormat     = SAMPLE_FORMAT;
+      d_output_parameters.suggestedLatency = deviceInfo->defaultLowOutputLatency;
+      d_output_parameters.hostApiSpecificStreamInfo = NULL;
+
+      // We fill in the real channelCount in check_topology when we know
+      // how many inputs are connected to us.
+
+      // Now that we know the maximum number of channels (allegedly)
+      // supported by the h/w, we can compute a reasonable input
+      // signature.  The portaudio specs say that they'll accept any
+      // number of channels from 1 to max.
+      set_input_signature(gr_make_io_signature(1, deviceInfo->maxOutputChannels,
+                                               sizeof(sample_t)));
+    }
+
+    bool
+    portaudio_sink::check_topology(int ninputs, int noutputs)
+    {
+      PaError err;
+
+      if(Pa_IsStreamActive(d_stream)) {
+        Pa_CloseStream(d_stream);
+        d_stream = 0;
+        d_reader.reset();  // boost::shared_ptr for d_reader = 0
+        d_writer.reset();  // boost::shared_ptr for d_write = 0
+      }
+
+      d_output_parameters.channelCount = ninputs;  // # of channels we're really using
+
+#if 1
+      d_portaudio_buffer_size_frames = (int)(0.0213333333  * d_sampling_rate + 0.5);  // Force 1024 frame buffers at 48000
+      fprintf(stderr, "Latency = %8.5f, requested sampling_rate = %g\n", // Force latency to 21.3333333.. ms
+              0.0213333333, (double)d_sampling_rate);
+#endif
+      err = Pa_OpenStream(&d_stream,
+                          NULL,			// No input
+                          &d_output_parameters,
+                          d_sampling_rate,
+                          d_portaudio_buffer_size_frames,
+                          paClipOff,
+                          &portaudio_sink_callback,
+                          (void*)this);
+
+      if(err != paNoError) {
+        output_error_msg("OpenStream failed", err);
+        return false;
+      }
+
+#if 0
+      const PaStreamInfo *psi = Pa_GetStreamInfo(d_stream);
+
+      d_portaudio_buffer_size_frames = (int)(d_output_parameters.suggestedLatency  * psi->sampleRate);
+      fprintf(stderr, "Latency = %7.4f, psi->sampleRate = %g\n",
+              d_output_parameters.suggestedLatency, psi->sampleRate);
+#endif
+
+      fprintf(stderr, "d_portaudio_buffer_size_frames = %d\n",
+              d_portaudio_buffer_size_frames);
+
+      assert(d_portaudio_buffer_size_frames != 0);
+
+      create_ringbuffer();
+
+      err = Pa_StartStream(d_stream);
+      if(err != paNoError) {
+        output_error_msg("StartStream failed", err);
+        return false;
+      }
+
+      return true;
+    }
+
+    portaudio_sink::~portaudio_sink()
+    {
+      Pa_StopStream(d_stream);	// wait for output to drain
+      Pa_CloseStream(d_stream);
+      Pa_Terminate();
+    }
+
+    /*
+     * This version consumes everything sent to it, blocking if required.
+     * I think this will allow us better control of the total buffering/latency
+     * in the audio path.
+     */
+    int
+    portaudio_sink::work(int noutput_items,
+                         gr_vector_const_void_star &input_items,
+                         gr_vector_void_star &output_items)
+    {
+      const float **in = (const float **)&input_items[0];
+      const unsigned nchan = d_output_parameters.channelCount; // # of channels == samples/frame
+  
+      int k;
+      for(k = 0; k < noutput_items;) {
+        int nframes = d_writer->space_available() / nchan;  // How much space in ringbuffer
+        if(nframes == 0) {      // no room...
+          if(d_ok_to_block) {
+            {
+              gruel::scoped_lock guard(d_ringbuffer_mutex);
+              while(!d_ringbuffer_ready)
+                d_ringbuffer_cond.wait(guard);
+            }
+            continue;
+          }
+          else {
+            // There's no room and we're not allowed to block.
+            // (A USRP is most likely controlling the pacing through the pipeline.)
+            // We drop the samples on the ground, and say we processed them all ;)
+            //
+            // FIXME, there's probably room for a bit more finesse here.
+            return noutput_items;
+          }
+        }
+
+        // We can write the smaller of the request and the room we've got
+        {
+          gruel::scoped_lock guard(d_ringbuffer_mutex);
+
+          int nf = std::min(noutput_items - k, nframes);
+          float *p = (float*)d_writer->write_pointer();
+
+          for(int i = 0; i < nf; i++) {
+            for(unsigned int c = 0; c < nchan; c++) {
+              *p++ = in[c][k + i];
+            }
+          }
+
+          d_writer->update_write_pointer(nf * nchan);
+          k += nf;
+
+          d_ringbuffer_ready = false;
+        }
+      }
+
+      return k;  // tell how many we actually did
+    }
+
+    void
+    portaudio_sink::output_error_msg(const char *msg, int err)
+    {
+      fprintf(stderr, "audio_portaudio_sink[%s]: %s: %s\n",
+              d_device_name.c_str(), msg, Pa_GetErrorText(err));
+    }
+
+    void
+    portaudio_sink::bail(const char *msg, int err) throw (std::runtime_error)
+    {
+      output_error_msg(msg, err);
+      throw std::runtime_error("audio_portaudio_sink");
+    }
+
+  } /* namespace audio */
+} /* namespace gr */
diff --git a/gr-audio/lib/portaudio/portaudio_sink.h b/gr-audio/lib/portaudio/portaudio_sink.h
new file mode 100644
index 0000000000..41a725b691
--- /dev/null
+++ b/gr-audio/lib/portaudio/portaudio_sink.h
@@ -0,0 +1,90 @@
+/* -*- c++ -*- */
+/*
+ * Copyright 2006-2011,2013 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.
+ */
+#ifndef INCLUDED_AUDIO_PORTAUDIO_SINK_H
+#define INCLUDED_AUDIO_PORTAUDIO_SINK_H
+
+#include <audio/sink.h>
+#include <gr_buffer.h>
+#include <gruel/thread.h>
+#include <string>
+#include <portaudio.h>
+#include <stdexcept>
+//#include <gri_logger.h>
+
+namespace gr {
+  namespace audio {
+
+    PaStreamCallback portaudio_sink_callback;
+
+    /*!
+     * \brief Audio sink using PORTAUDIO
+     * \ingroup audio_blk
+     *
+     * Input samples must be in the range [-1,1].
+     */
+    class portaudio_sink : public sink
+    {
+      friend PaStreamCallback portaudio_sink_callback;
+
+      unsigned int d_sampling_rate;
+      std::string  d_device_name;
+      bool         d_ok_to_block;
+      bool         d_verbose;
+
+      unsigned int d_portaudio_buffer_size_frames; // number of frames in a portaudio buffer
+
+      PaStream           *d_stream;
+      PaStreamParameters  d_output_parameters;
+
+      gr_buffer_sptr        d_writer;  // buffer used between work and callback
+      gr_buffer_reader_sptr d_reader;
+
+      gruel::mutex              d_ringbuffer_mutex;
+      gruel::condition_variable d_ringbuffer_cond;
+      bool                      d_ringbuffer_ready;
+
+      // random stats
+      int d_nunderuns;          // count of underruns
+      //gri_logger_sptr d_log;  // handle to non-blocking logging instance
+
+      void output_error_msg(const char *msg, int err);
+      void bail(const char *msg, int err) throw (std::runtime_error);
+      void create_ringbuffer();
+
+    public:
+      portaudio_sink(int sampling_rate,
+                     const std::string device_name,
+                     bool ok_to_block);
+
+      ~portaudio_sink();
+
+      bool check_topology(int ninputs, int noutputs);
+
+      int work(int noutput_items,
+               gr_vector_const_void_star &input_items,
+               gr_vector_void_star &output_items);
+    };
+
+  } /* namespace audio */
+} /* namespace gr */
+
+#endif /* INCLUDED_AUDIO_PORTAUDIO_SINK_H */
diff --git a/gr-audio/lib/portaudio/portaudio_source.cc b/gr-audio/lib/portaudio/portaudio_source.cc
new file mode 100644
index 0000000000..937c1d0dbf
--- /dev/null
+++ b/gr-audio/lib/portaudio/portaudio_source.cc
@@ -0,0 +1,378 @@
+/* -*- c++ -*- */
+/*
+ * Copyright 2006-2011,2013 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 he 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
+
+#include "audio_registry.h"
+#include <portaudio_source.h>
+#include <portaudio_impl.h>
+#include <gr_io_signature.h>
+#include <gr_prefs.h>
+#include <stdio.h>
+#include <iostream>
+#include <unistd.h>
+#include <stdexcept>
+#include <string.h>
+
+namespace gr {
+  namespace audio {
+
+    AUDIO_REGISTER_SOURCE(REG_PRIO_MED, portaudio)(int sampling_rate,
+                                                   const std::string &device_name,
+                                                   bool ok_to_block)
+    {
+      return source::sptr
+        (new portaudio_source(sampling_rate, device_name, ok_to_block));
+    }
+
+//#define LOGGING 0  // define to 0 or 1
+
+#define SAMPLE_FORMAT paFloat32
+
+    typedef float sample_t;
+
+    // Number of portaudio buffers in the ringbuffer
+    static const unsigned int N_BUFFERS = 4;
+
+    static std::string
+    default_device_name()
+    {
+      return gr_prefs::singleton()->get_string
+        ("audio_portaudio", "default_input_device", "");
+    }
+
+    void
+    portaudio_source::create_ringbuffer(void)
+    {
+      int bufsize_samples = d_portaudio_buffer_size_frames * d_input_parameters.channelCount;
+
+      if(d_verbose) {
+        fprintf(stderr, "ring buffer size  = %d frames\n",
+                N_BUFFERS*bufsize_samples/d_input_parameters.channelCount);
+      }
+
+      // FYI, the buffer indicies are in units of samples.
+      d_writer = gr_make_buffer(N_BUFFERS * bufsize_samples, sizeof(sample_t));
+      d_reader = gr_buffer_add_reader(d_writer, 0);
+    }
+
+    /*
+     * This routine will be called by the PortAudio engine when audio is needed.
+     * It may called at interrupt level on some machines so don't do anything
+     * that could mess up the system like calling malloc() or free().
+     *
+     * Our job is to copy framesPerBuffer frames from inputBuffer.
+     */
+    int
+    portaudio_source_callback(const void *inputBuffer,
+                              void *outputBuffer,
+                              unsigned long framesPerBuffer,
+                              const PaStreamCallbackTimeInfo* timeInfo,
+                              PaStreamCallbackFlags statusFlags,
+                              void *arg)
+    {
+      portaudio_source *self = (portaudio_source *)arg;
+      int nchan = self->d_input_parameters.channelCount;
+      int nframes_to_copy = framesPerBuffer;
+      int nframes_room = self->d_writer->space_available() / nchan;
+
+      if(nframes_to_copy <= nframes_room) {  // We've got room for the data ..
+        //if (LOGGING)
+        //  self->d_log->printf("PAsrc  cb: f/b = %4ld\n", framesPerBuffer);
+
+        // copy from input buffer to ringbuffer
+        {
+          gruel::scoped_lock(d_ringbuffer_mutex);
+
+          memcpy(self->d_writer->write_pointer(),
+                 inputBuffer,
+                 nframes_to_copy * nchan * sizeof(sample_t));
+          self->d_writer->update_write_pointer(nframes_to_copy * nchan);
+
+          // Tell the source thread there is new data in the ringbuffer.
+          self->d_ringbuffer_ready = true;
+        }
+
+        self->d_ringbuffer_cond.notify_one();
+        return paContinue;
+      }
+
+      else {			// overrun
+        self->d_noverruns++;
+        ssize_t r = ::write(2, "aO", 2);	// FIXME change to non-blocking call
+        if(r == -1) {
+          perror("audio_portaudio_source::portaudio_source_callback write error to stderr.");
+        }
+
+        self->d_ringbuffer_ready = false;
+        self->d_ringbuffer_cond.notify_one();  // Tell the sink to get going!
+        return paContinue;
+      }
+    }
+
+    // ----------------------------------------------------------------
+
+    portaudio_source::portaudio_source(int sampling_rate,
+                                       const std::string device_name,
+                                       bool ok_to_block)
+      : gr_sync_block("audio_portaudio_source",
+                      gr_make_io_signature(0, 0, 0),
+                      gr_make_io_signature(0, 0, 0)),
+        d_sampling_rate(sampling_rate),
+        d_device_name(device_name.empty() ? default_device_name() : device_name),
+        d_ok_to_block(ok_to_block),
+        d_verbose(gr_prefs::singleton()->get_bool("audio_portaudio", "verbose", false)),
+        d_portaudio_buffer_size_frames(0),
+        d_stream(0),
+        d_ringbuffer_mutex(),
+        d_ringbuffer_cond(),
+        d_ringbuffer_ready(false),
+        d_noverruns(0)
+    {
+      memset(&d_input_parameters, 0, sizeof(d_input_parameters));
+      //if(LOGGING)
+      //  d_log = gri_logger::singleton();
+
+      PaError err;
+      int i, numDevices;
+      PaDeviceIndex device = 0;
+      const PaDeviceInfo *deviceInfo = NULL;
+
+      err = Pa_Initialize();
+      if(err != paNoError) {
+        bail("Initialize failed", err);
+      }
+
+      if(d_verbose)
+        print_devices();
+
+      numDevices = Pa_GetDeviceCount();
+      if(numDevices < 0)
+        bail("Pa Device count failed", 0);
+      if(numDevices == 0)
+        bail("no devices available", 0);
+
+      if(d_device_name.empty()) {
+        // FIXME Get smarter about picking something
+        device = Pa_GetDefaultInputDevice();
+        deviceInfo = Pa_GetDeviceInfo(device);
+        fprintf(stderr,"%s is the chosen device using %s as the host\n",
+                deviceInfo->name, Pa_GetHostApiInfo(deviceInfo->hostApi)->name);
+      }
+      else {
+        bool found = false;
+
+        for(i = 0; i < numDevices; i++) {
+          deviceInfo = Pa_GetDeviceInfo(i);
+          fprintf(stderr,"Testing device name: %s",deviceInfo->name);
+          if(deviceInfo->maxInputChannels <= 0) {
+            fprintf(stderr,"\n");
+            continue;
+          }
+          if(strstr(deviceInfo->name, d_device_name.c_str())) {
+            fprintf(stderr,"  Chosen!\n");
+            device = i;
+            fprintf(stderr,"%s using %s as the host\n",d_device_name.c_str(),
+                    Pa_GetHostApiInfo(deviceInfo->hostApi)->name), fflush(stderr);
+            found = true;
+            deviceInfo = Pa_GetDeviceInfo(device);
+            i = numDevices; 	// force loop exit
+          }
+          else
+            fprintf(stderr,"\n"),fflush(stderr);
+        }
+
+        if(!found) {
+          bail("Failed to find specified device name", 0);
+        }
+      }
+
+      d_input_parameters.device = device;
+      d_input_parameters.channelCount = deviceInfo->maxInputChannels;
+      d_input_parameters.sampleFormat = SAMPLE_FORMAT;
+      d_input_parameters.suggestedLatency = deviceInfo->defaultLowInputLatency;
+      d_input_parameters.hostApiSpecificStreamInfo = NULL;
+
+      // We fill in the real channelCount in check_topology when we know
+      // how many inputs are connected to us.
+
+      // Now that we know the maximum number of channels (allegedly)
+      // supported by the h/w, we can compute a reasonable output
+      // signature.  The portaudio specs say that they'll accept any
+      // number of channels from 1 to max.
+      set_output_signature(gr_make_io_signature(1, deviceInfo->maxInputChannels,
+                                                sizeof (sample_t)));
+    }
+
+    bool
+    portaudio_source::check_topology(int ninputs, int noutputs)
+    {
+      PaError err;
+
+      if(Pa_IsStreamActive(d_stream)) {
+        Pa_CloseStream(d_stream);
+        d_stream = 0;
+        d_reader.reset();   // boost::shared_ptr for d_reader = 0
+        d_writer.reset();   // boost::shared_ptr for d_write = 0
+      }
+
+      d_input_parameters.channelCount = noutputs; // # of channels we're really using
+
+#if 1
+      d_portaudio_buffer_size_frames = (int)(0.0213333333  * d_sampling_rate + 0.5);  // Force 512 frame buffers at 48000
+      fprintf(stderr, "Latency = %8.5f, requested sampling_rate = %g\n", // Force latency to 21.3333333.. ms
+              0.0213333333, (double)d_sampling_rate);
+#endif
+      err = Pa_OpenStream(&d_stream,
+                          &d_input_parameters,
+                          NULL,			// No output
+                          d_sampling_rate,
+                          d_portaudio_buffer_size_frames,
+                          paClipOff,
+                          &portaudio_source_callback,
+                          (void*)this);
+
+      if(err != paNoError) {
+        output_error_msg("OpenStream failed", err);
+        return false;
+      }
+
+#if 0
+      const PaStreamInfo *psi = Pa_GetStreamInfo(d_stream);
+
+      d_portaudio_buffer_size_frames = (int)(d_input_parameters.suggestedLatency  * psi->sampleRate);
+      fprintf(stderr, "Latency = %7.4f, psi->sampleRate = %g\n",
+              d_input_parameters.suggestedLatency, psi->sampleRate);
+#endif
+
+      fprintf(stderr, "d_portaudio_buffer_size_frames = %d\n",
+              d_portaudio_buffer_size_frames);
+
+      assert(d_portaudio_buffer_size_frames != 0);
+
+      create_ringbuffer();
+
+      err = Pa_StartStream(d_stream);
+      if(err != paNoError) {
+        output_error_msg("StartStream failed", err);
+        return false;
+      }
+
+      return true;
+    }
+
+    portaudio_source::~portaudio_source()
+    {
+      Pa_StopStream(d_stream);	// wait for output to drain
+      Pa_CloseStream(d_stream);
+      Pa_Terminate();
+    }
+
+    int
+    portaudio_source::work(int noutput_items,
+                           gr_vector_const_void_star &input_items,
+                           gr_vector_void_star &output_items)
+    {
+      float **out = (float **)&output_items[0];
+      const unsigned nchan = d_input_parameters.channelCount; // # of channels == samples/frame
+  
+      int k;
+      for(k = 0; k < noutput_items;) {
+        int nframes = d_reader->items_available() / nchan;	// # of frames in ringbuffer
+        if(nframes == 0) {     // no data right now...
+          if(k > 0)            // If we've produced anything so far, return that
+            return k;
+
+          if(d_ok_to_block) {
+            gruel::scoped_lock guard(d_ringbuffer_mutex);
+            while(d_ringbuffer_ready == false)
+              d_ringbuffer_cond.wait(guard);  // block here, then try again
+            continue;
+          }
+
+          assert(k == 0);
+
+          // There's no data and we're not allowed to block.
+          // (A USRP is most likely controlling the pacing through the pipeline.)
+          // This is an underun.  The scheduler wouldn't have called us if it
+          // had anything better to do.  Thus we really need to produce some amount
+          // of "fill".
+          //
+          // There are lots of options for comfort noise, etc.
+          // FIXME We'll fill with zeros for now.  Yes, it will "click"...
+
+          // Fill with some frames of zeros
+          {
+            gruel::scoped_lock guard(d_ringbuffer_mutex);
+
+            int nf = std::min(noutput_items - k, (int)d_portaudio_buffer_size_frames);
+            for(int i = 0; i < nf; i++) {
+              for(unsigned int c = 0; c < nchan; c++) {
+                out[c][k + i] = 0;
+              }
+            }
+            k += nf;
+
+            d_ringbuffer_ready = false;
+            return k;
+          }
+        }
+
+        // We can read the smaller of the request and what's in the buffer.
+        {
+          gruel::scoped_lock guard(d_ringbuffer_mutex);
+
+          int nf = std::min(noutput_items - k, nframes);
+
+          const float *p = (const float*)d_reader->read_pointer();
+          for(int i = 0; i < nf; i++) {
+            for(unsigned int c = 0; c < nchan; c++) {
+              out[c][k + i] = *p++;
+            }
+          }
+          d_reader->update_read_pointer(nf * nchan);
+          k += nf;
+          d_ringbuffer_ready = false;
+        }
+      }
+
+      return k;  // tell how many we actually did
+    }
+
+    void
+    portaudio_source::output_error_msg(const char *msg, int err)
+    {
+      fprintf(stderr, "audio_portaudio_source[%s]: %s: %s\n",
+              d_device_name.c_str (), msg, Pa_GetErrorText(err));
+    }
+
+    void
+    portaudio_source::bail(const char *msg, int err) throw (std::runtime_error)
+    {
+      output_error_msg(msg, err);
+      throw std::runtime_error("audio_portaudio_source");
+    }
+
+  } /* namespace audio */
+} /* namespace gr */
diff --git a/gr-audio/lib/portaudio/portaudio_source.h b/gr-audio/lib/portaudio/portaudio_source.h
new file mode 100644
index 0000000000..d4f4f01d1f
--- /dev/null
+++ b/gr-audio/lib/portaudio/portaudio_source.h
@@ -0,0 +1,89 @@
+/* -*- c++ -*- */
+/*
+ * Copyright 2006-2011,2013 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.
+ */
+
+#ifndef INCLUDED_AUDIO_PORTAUDIO_SOURCE_H
+#define INCLUDED_AUDIO_PORTAUDIO_SOURCE_H
+
+#include <audio/source.h>
+#include <gr_buffer.h>
+#include <gruel/thread.h>
+#include <string>
+#include <portaudio.h>
+#include <stdexcept>
+
+namespace gr {
+  namespace audio {
+
+    PaStreamCallback portaudio_source_callback;
+
+    /*!
+     * \brief Audio source using PORTAUDIO
+     * \ingroup audio_blk
+     *
+     * Input samples must be in the range [-1,1].
+     */
+    class portaudio_source : public source
+    {
+      friend PaStreamCallback portaudio_source_callback;
+
+      unsigned int d_sampling_rate;
+      std::string  d_device_name;
+      bool         d_ok_to_block;
+      bool         d_verbose;
+
+      unsigned int d_portaudio_buffer_size_frames; // number of frames in a portaudio buffer
+
+      PaStream           *d_stream;
+      PaStreamParameters  d_input_parameters;
+
+      gr_buffer_sptr        d_writer;  // buffer used between work and callback
+      gr_buffer_reader_sptr d_reader;
+
+      gruel::mutex              d_ringbuffer_mutex;
+      gruel::condition_variable d_ringbuffer_cond;
+      bool                      d_ringbuffer_ready;
+
+      // random stats
+      int d_noverruns;  // count of overruns
+
+      void output_error_msg(const char *msg, int err);
+      void bail(const char *msg, int err) throw (std::runtime_error);
+      void create_ringbuffer();
+
+    public:
+      portaudio_source(int sampling_rate,
+                       const std::string device_name,
+                       bool ok_to_block);
+
+      ~portaudio_source();
+
+      bool check_topology(int ninputs, int noutputs);
+
+      int work(int ninput_items,
+               gr_vector_const_void_star &input_items,
+               gr_vector_void_star &output_items);
+    };
+
+  } /* namespace audio */
+} /* namespace gr */
+
+#endif /* INCLUDED_AUDIO_PORTAUDIO_SOURCE_H */
author	Tom Rondeau <trondeau@vt.edu>	2013-02-13 10:58:42 -0500
committer	Tom Rondeau <trondeau@vt.edu>	2013-02-13 10:58:42 -0500
commit	9888a4d2d12dc874a75c114ba52c9956065ba923 (patch)
tree	5d7e6972b444d4cae49c633e1943c69d2523392c
parent	364404cdd1ddf6e9b8c128490f76bc95a384facd (diff)