Updated PortAudio to latest version for Mac

git-svn-id: http://svn.pjsip.org/repos/pjproject/trunk@770 74dad513-b988-da41-8d7b-12977e46ad98

13 files changed, 4867 insertions, 747 deletions

diff --git a/pjmedia/build/os-auto.mak.in b/pjmedia/build/os-auto.mak.in
index e35cebd3..9f444690 100644
--- a/pjmedia/build/os-auto.mak.in
+++ b/pjmedia/build/os-auto.mak.in
@@ -5,16 +5,19 @@ export CFLAGS += @ac_no_small_filter@ @ac_no_large_filter@ @ac_no_g711_plc@ @ac_
 
 # Define the desired sound device backend
 # Valid values are:
-#   - pa_unix:	    PortAudio on Unix (OSS or ALSA)
-#   - pa_darwinos:  PortAudio on MacOSX (CoreAudio)
-#   - pa_win32:	    PortAudio on Win32 (WMME)
-#   - ds:	    Win32 DirectSound (dsound.c)
-#   - null:	    Null sound device (nullsound.c)
+#   - pa_unix:	    	PortAudio on Unix (OSS or ALSA)
+#   - pa_darwinos:  	PortAudio on MacOSX (CoreAudio)
+#   - pa_old_darwinos:  PortAudio on MacOSX (old CoreAudio, for OSX 10.2)
+#   - pa_win32:	    	PortAudio on Win32 (WMME)
+#   - ds:	    	Win32 DirectSound (dsound.c)
+#   - null:	    	Null sound device (nullsound.c)
 AC_PJMEDIA_SND=@ac_pjmedia_snd@
 
 # For Unix, specify if ALSA should be supported
 AC_PA_USE_ALSA=@ac_pa_use_alsa@
 
+# Additional PortAudio CFLAGS are in @ac_pa_cflags@
+
 #
 # Codecs
 #
@@ -77,16 +80,34 @@ endif
 
 
 #
-# PortAudio on MacOS X
+# PortAudio on MacOS X (using current PortAudio)
 #
 ifeq ($(AC_PJMEDIA_SND),pa_darwinos)
-export PJMEDIA_OBJS += $(PA_DIR)/pa_mac_hostapis.o $(PA_DIR)/pa_unix_util.o \
-		       $(PA_DIR)/pa_mac_core.o
-export CFLAGS += -DPA_USE_COREAUDIO=1  \
+export PJMEDIA_OBJS +=	$(PA_DIR)/pa_mac_hostapis.o \
+			$(PA_DIR)/pa_unix_util.o \
+			$(PA_DIR)/pa_mac_core.o \
+			$(PA_DIR)/pa_mac_core_blocking.o \
+			$(PA_DIR)/pa_mac_core_utilities.o \
+			$(PA_DIR)/ringbuffer.o
+export CFLAGS += -DPA_USE_COREAUDIO=1 \
 	         -DPJMEDIA_SOUND_IMPLEMENTATION=PJMEDIA_SOUND_PORTAUDIO_SOUND
+export CFLAGS += @ac_pa_cflags@
 endif
 
 #
+# PortAudio on MacOS X (using old PortAudio, for MacOS X 10.2.x)
+#
+ifeq ($(AC_PJMEDIA_SND),pa_old_darwinos)
+export PJMEDIA_OBJS +=	$(PA_DIR)/pa_mac_hostapis.o \
+			$(PA_DIR)/pa_unix_util.o \
+			$(PA_DIR)/pa_mac_core_old.o 
+export CFLAGS += -DPA_USE_COREAUDIO=1 \
+	         -DPJMEDIA_SOUND_IMPLEMENTATION=PJMEDIA_SOUND_PORTAUDIO_SOUND
+export CFLAGS += @ac_pa_cflags@
+endif
+
+#
+#
 # PortAudio on Win32 (WMME)
 #
 ifeq ($(AC_PJMEDIA_SND),pa_win32)
diff --git a/pjmedia/src/pjmedia/portaudio/notes.txt b/pjmedia/src/pjmedia/portaudio/notes.txt
new file mode 100644
index 00000000..ad66f358
--- /dev/null
+++ b/pjmedia/src/pjmedia/portaudio/notes.txt
@@ -0,0 +1,145 @@
+Notes on status of CoreAudio Implementation of PortAudio
+
+Document Last Updated December 9, 2005
+
+There are currently two implementations of PortAudio for Mac Core Audio.
+
+The original is in pa_mac_core_old.c, and the newer, default implementation
+is in pa_mac_core.c.
+Only pa_mac_core.c is currently developed and supported as it uses apple's
+current core audio technology. To select use the old implementation, replace
+pa_mac_core.c with pa_mac_core_old.c (eg. "cp pa_mac_core_auhal.c
+pa_mac_core.c"), then run configure and make as usual.
+
+----------------------------------------
+
+Notes on Original implementation:
+
+by Phil Burk and Darren Gibbs
+
+Last updated March 20, 2002
+
+WHAT WORKS
+
+Output with very low latency, <10 msec.
+Half duplex input or output.
+Full duplex on the same CoreAudio device.
+The paFLoat32, paInt16, paInt8, paUInt8 sample formats.
+Pa_GetCPULoad()
+Pa_StreamTime()
+
+KNOWN BUGS OR LIMITATIONS
+
+We do not yet support simultaneous input and output on different 
+devices. Note that some CoreAudio devices like the Roland UH30 look 
+like one device but are actually two different CoreAudio devices. The 
+Built-In audio is typically one CoreAudio device.
+
+Mono doesn't work.
+
+DEVICE MAPPING
+
+CoreAudio devices can support both input and output. But the sample 
+rates supported may be different. So we have map one or two PortAudio 
+device to each CoreAudio device depending on whether it supports 
+input, output or both.
+
+When we query devices, we first get a list of CoreAudio devices. Then 
+we scan the list and add a PortAudio device for each CoreAudio device 
+that supports input. Then we make a scan for output devices.
+
+-------------------------------------------
+
+Notes on Newer/Default AUHAL implementation:
+
+by Bjorn Roche
+
+Last Updated December 9, 2005
+
+Principle of Operation:
+
+This implementation uses AUHAL for audio I/O. To some extent, it also
+operates at the "HAL" Layer, though this behavior can be limited by
+platform specific flags (see pa_mac_core.h for details). The default
+settings should be reasonable: they don't change the SR of the device and
+don't cause interruptions if other devices are using the device.
+
+Major Software Elements Used: Apple's HAL AUs provide output SR
+conversion transparently, however, only on output, so this
+implementation uses AudioConverters to convert the sample rate on input.
+A PortAudio ring buffer is used to buffer input when sample rate
+conversion is required or when separate audio units are used for duplex
+IO. Finally, a PortAudio buffer processor is used to convert formats and
+provide additional buffers if needed. Internally, interleaved floating
+point data streams are used exclusively - the audio unit converts from
+the audio hardware's native format to interleaved float PCM and
+PortAudio's Buffer processor is used for conversion to user formats.
+
+Simplex Input: Simplex input uses a single callback. If sample rate
+conversion is required, a ring buffer and AudioConverter are used as
+well.
+
+Simplex output: Simplex output uses a single callback. No ring buffer or
+audio converter is used because AUHAL does its own output SR conversion.
+
+Duplex, one device (no SR conversion): When one device is used, a single
+callback is used. This achieves very low latency.
+
+Duplex, separate devices or SR conversion: When SR conversion is
+required, data must be buffered before it is converted and data is not
+always available at the same times on input and output, so SR conversion
+requires the same treatment as separate devices. The input callback
+reads data and puts it in the ring buffer. The output callback reads the
+data off the ring buffer, into an audio converter and finally to the
+buffer processor.
+
+Platform Specific Options:
+
+By using the flags in pa_mac_core.h, the user may specify several options.
+For example, the user can specify the sample-rate conversion quality, and
+the extent to which PA will attempt to "play nice" and to what extent it
+will interrupt other apps to improve performance. For example, if 44100 Hz
+sample rate is requested but the device is set at 48000 Hz, PA can either
+change the device for optimal playback ("Pro" mode), which may interrupt
+other programs playing back audio, or simple use a sample-rate coversion,
+which allows for friendlier sharing of the device ("Play Nice" mode).
+
+
+Known issues:
+
+- Latency: Latency settings are ignored in most cases. Exceptions are when
+doing I/O between different devices and as a hint for selecting a realtively
+low or relatively high latency in conjunction with
+paHostFramesPerBufferUnspecified. Latency settings are always automatically
+bound to "safe" values, however, so setting extreme values here should not be
+an issue.
+
+- Buffer Size: paHostFramesPerBufferUnspecified and specific host buffer sizes
+are supported. paHostFramesPerBufferUnspecified works best in "pro" mode,
+where the buffer size and sample rate of the audio device is most likely
+to match the expected values.
+
+- Timing info. It reports on stream time, but I'm probably doing something
+wrong since patest_sine_time often reports negative latency numbers.
+
+- xrun detection: The only xrun detection performed is when reading
+and writing the ring buffer. There is probably more that can be done.
+
+- abort/stop issues: stopping a stream is always a complete operation,
+but latency should be low enough to make the lack of a separate abort
+unnecessary. Apple clarifies its AudioOutputUnitStop() call here:
+http://lists.apple.com/archives/coreaudio-api/2005/Dec/msg00055.html
+
+- blocking interface: Not implemented.
+
+- multichannel: It has been tested successfully on multichannel hardware
+from MOTU: traveler and 896HD.
+
+- sample rate conversion quality: By default, SR conversion is the maximum
+available. This can be tweaked using flags pa_mac_core.h. Note that the AU
+render quyality property is used to set the sample rat conversion quality
+as "documented" here:
+http://lists.apple.com/archives/coreaudio-api/2004/Jan/msg00141.html
+
+- x86: I haven't tested it on an x86 Mac myself, but users have reported 
+being able to comiple and run it.
diff --git a/pjmedia/src/pjmedia/portaudio/pa_mac_core.c b/pjmedia/src/pjmedia/portaudio/pa_mac_core.c
index ce55b477..2b2b9cc1 100644
--- a/pjmedia/src/pjmedia/portaudio/pa_mac_core.c
+++ b/pjmedia/src/pjmedia/portaudio/pa_mac_core.c
@@ -1,13 +1,24 @@
 /*
- * $Id$
- * pa_mac_core.c
- * Implementation of PortAudio for Mac OS X CoreAudio       
- *                                                                                         
+ * Implementation of the PortAudio API for Apple AUHAL
+ *
  * PortAudio Portable Real-Time Audio Library
  * Latest Version at: http://www.portaudio.com
  *
- * Authors: Ross Bencina and Phil Burk
- * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
+ * Written by Bjorn Roche of XO Audio LLC, from PA skeleton code.
+ * Portions copied from code by Dominic Mazzoni (who wrote a HAL implementation)
+ *
+ * Dominic's code was based on code by Phil Burk, Darren Gibbs,
+ * Gord Peters, Stephane Letz, and Greg Pfiel.
+ *
+ * The following people also deserve acknowledgements:
+ *
+ * Olivier Tristan for feedback and testing
+ * Glenn Zelniker and Z-Systems engineering for sponsoring the Blocking I/O
+ * interface.
+ * 
+ *
+ * Based on the Open Source API proposed by Ross Bencina
+ * Copyright (c) 1999-2002 Ross Bencina, Phil Burk
  *
  * Permission is hereby granted, free of charge, to any person obtaining
  * a copy of this software and associated documentation files
@@ -20,10 +31,6 @@
  * The above copyright notice and this permission notice shall be
  * included in all copies or substantial portions of the Software.
  *
- * Any person wishing to distribute modifications to the Software is
- * requested to send the modifications to the original developer so that
- * they can be incorporated into the canonical version.
- *
  * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
  * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
  * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
@@ -31,603 +38,956 @@
  * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
  * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
  * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+/*
+ * The text above constitutes the entire PortAudio license; however, 
+ * the PortAudio community also makes the following non-binding requests:
  *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version. It is also 
+ * requested that these non-binding requests be included along with the 
+ * license above.
  */
 
-#include <CoreAudio/CoreAudio.h>
-#include <AudioToolbox/AudioToolbox.h>
-#include <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-#include <assert.h>
-
-#include "portaudio.h"
-#include "pa_trace.h"
-#include "pa_util.h"
-#include "pa_allocation.h"
-#include "pa_hostapi.h"
-#include "pa_stream.h"
-#include "pa_cpuload.h"
-#include "pa_process.h"
-
-// =====  constants  =====
-
-// =====  structs  =====
-#pragma mark structs
-
-// PaMacCoreHostApiRepresentation - host api datastructure specific to this implementation
-typedef struct PaMacCore_HAR
-{
-    PaUtilHostApiRepresentation inheritedHostApiRep;
-    PaUtilStreamInterface callbackStreamInterface;
-    PaUtilStreamInterface blockingStreamInterface;
-    
-    PaUtilAllocationGroup *allocations;
-    AudioDeviceID *macCoreDeviceIds;
-}
-PaMacCoreHostApiRepresentation;
+/**
+ @file pa_mac_core
+ @ingroup hostapi_src
+ @author Bjorn Roche
+ @brief AUHAL implementation of PortAudio
+*/
 
-typedef struct PaMacCore_DI
-{
-    PaDeviceInfo inheritedDeviceInfo;
-}
-PaMacCoreDeviceInfo;
+/* FIXME: not all error conditions call PaUtil_SetLastHostErrorInfo()
+ * PaMacCore_SetError() will do this.
+ */
 
-// PaMacCoreStream - a stream data structure specifically for this implementation
-typedef struct PaMacCore_S
-{
-    PaUtilStreamRepresentation streamRepresentation;
-    PaUtilCpuLoadMeasurer cpuLoadMeasurer;
-    PaUtilBufferProcessor bufferProcessor;
-    
-    int primeStreamUsingCallback;
-    
-    AudioDeviceID inputDevice;
-    AudioDeviceID outputDevice;
-    
-    // Processing thread management --------------
-//    HANDLE abortEvent;
-//    HANDLE processingThread;
-//    DWORD processingThreadId;
-    
-    char throttleProcessingThreadOnOverload; // 0 -> don't throtte, non-0 -> throttle
-    int processingThreadPriority;
-    int highThreadPriority;
-    int throttledThreadPriority;
-    unsigned long throttledSleepMsecs;
-    
-    int isStopped;
-    volatile int isActive;
-    volatile int stopProcessing; // stop thread once existing buffers have been returned
-    volatile int abortProcessing; // stop thread immediately
-    
-//    DWORD allBuffersDurationMs; // used to calculate timeouts
-}
-PaMacCoreStream;
+#include "pa_mac_core_internal.h"
 
-// Data needed by the CoreAudio callback functions
-typedef struct PaMacCore_CD
-{
-    PaMacCoreStream *stream;
-    PaStreamCallback *callback;
-    void *userData;
-    PaUtilConverter *inputConverter;
-    PaUtilConverter *outputConverter;
-    void *inputBuffer;
-    void *outputBuffer;
-    int inputChannelCount;
-    int outputChannelCount;
-    PaSampleFormat inputSampleFormat;
-    PaSampleFormat outputSampleFormat;
-    PaUtilTriangularDitherGenerator *ditherGenerator;
-}
-PaMacClientData;
+#include <string.h> /* strlen(), memcmp() etc. */
 
-// =====  CoreAudio-PortAudio bridge functions =====
-#pragma mark CoreAudio-PortAudio bridge functions
+#include "pa_mac_core.h"
+#include "pa_mac_core_utilities.h"
+#include "pa_mac_core_blocking.h"
 
-// Maps CoreAudio OSStatus codes to PortAudio PaError codes
-static PaError conv_err(OSStatus error)
-{
-    PaError result;
-    
-    switch (error) {
-        case kAudioHardwareNoError:
-            result = paNoError; break;
-        case kAudioHardwareNotRunningError:
-            result = paInternalError; break;
-        case kAudioHardwareUnspecifiedError:
-            result = paInternalError; break;
-        case kAudioHardwareUnknownPropertyError:
-            result = paInternalError; break;
-        case kAudioHardwareBadPropertySizeError:
-            result = paInternalError; break;
-        case kAudioHardwareIllegalOperationError:
-            result = paInternalError; break;
-        case kAudioHardwareBadDeviceError:
-            result = paInvalidDevice; break;
-        case kAudioHardwareBadStreamError:
-            result = paBadStreamPtr; break;
-//        case kAudioHardwareUnsupportedOperationError:
-//            result = paInternalError; break;
-        case kAudioDeviceUnsupportedFormatError:
-            result = paSampleFormatNotSupported; break;
-        case kAudioDevicePermissionsError:
-            result = paDeviceUnavailable; break;
-        default:
-            result = paInternalError;
-    }
-    
-    return result;
-}
 
-static AudioStreamBasicDescription *InitializeStreamDescription(const PaStreamParameters *parameters, double sampleRate)
-{
-    struct AudioStreamBasicDescription *streamDescription = PaUtil_AllocateMemory(sizeof(AudioStreamBasicDescription));
-    streamDescription->mSampleRate = sampleRate;
-    streamDescription->mFormatID = kAudioFormatLinearPCM;
-    streamDescription->mFormatFlags = 0;
-    streamDescription->mFramesPerPacket = 1;
-    
-    if (parameters->sampleFormat & paNonInterleaved) {
-        streamDescription->mFormatFlags |= kLinearPCMFormatFlagIsNonInterleaved;
-        streamDescription->mChannelsPerFrame = 1;
-        streamDescription->mBytesPerFrame = Pa_GetSampleSize(parameters->sampleFormat);
-        streamDescription->mBytesPerPacket = Pa_GetSampleSize(parameters->sampleFormat);
-    }
-    else {
-        streamDescription->mChannelsPerFrame = parameters->channelCount;
-    }
-    
-    streamDescription->mBytesPerFrame = Pa_GetSampleSize(parameters->sampleFormat) * streamDescription->mChannelsPerFrame;
-    streamDescription->mBytesPerPacket = streamDescription->mBytesPerFrame * streamDescription->mFramesPerPacket;
-    
-    if (parameters->sampleFormat & paFloat32) {
-        streamDescription->mFormatFlags |= kLinearPCMFormatFlagIsFloat;
-        streamDescription->mBitsPerChannel = 32;
-    }
-    else if (parameters->sampleFormat & paInt32) {
-        streamDescription->mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger;
-        streamDescription->mBitsPerChannel = 32;
-    }
-    else if (parameters->sampleFormat & paInt24) {
-        streamDescription->mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger;
-        streamDescription->mBitsPerChannel = 24;
-    }
-    else if (parameters->sampleFormat & paInt16) {
-        streamDescription->mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger;
-        streamDescription->mBitsPerChannel = 16;
-    }
-    else if (parameters->sampleFormat & paInt8) {
-        streamDescription->mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger;
-        streamDescription->mBitsPerChannel = 8;
-    }    
-    else if (parameters->sampleFormat & paInt32) {
-        streamDescription->mBitsPerChannel = 8;
-    }
-    
-    return streamDescription;
-}
 
-static PaStreamCallbackTimeInfo *InitializeTimeInfo(const AudioTimeStamp* now, const AudioTimeStamp* inputTime, const AudioTimeStamp* outputTime)
+/* prototypes for functions declared in this file */
+
+#ifdef __cplusplus
+extern "C"
 {
-    PaStreamCallbackTimeInfo *timeInfo = PaUtil_AllocateMemory(sizeof(PaStreamCallbackTimeInfo));
-    
-    timeInfo->inputBufferAdcTime = inputTime->mSampleTime;
-    timeInfo->currentTime = now->mSampleTime;
-    timeInfo->outputBufferDacTime = outputTime->mSampleTime;
-    
-    return timeInfo;
+#endif /* __cplusplus */
+
+PaError PaMacCore_Initialize( PaUtilHostApiRepresentation **hostApi, PaHostApiIndex index );
+
+#ifdef __cplusplus
 }
+#endif /* __cplusplus */
 
-// =====  support functions  =====
-#pragma mark support functions
+#define RING_BUFFER_ADVANCE_DENOMINATOR (4)
 
-static void CleanUp(PaMacCoreHostApiRepresentation *macCoreHostApi)
+static void Terminate( struct PaUtilHostApiRepresentation *hostApi );
+static PaError IsFormatSupported( struct PaUtilHostApiRepresentation *hostApi,
+                                  const PaStreamParameters *inputParameters,
+                                  const PaStreamParameters *outputParameters,
+                                  double sampleRate );
+static PaError OpenStream( struct PaUtilHostApiRepresentation *hostApi,
+                           PaStream** s,
+                           const PaStreamParameters *inputParameters,
+                           const PaStreamParameters *outputParameters,
+                           double sampleRate,
+                           unsigned long framesPerBuffer,
+                           PaStreamFlags streamFlags,
+                           PaStreamCallback *streamCallback,
+                           void *userData );
+static PaError CloseStream( PaStream* stream );
+static PaError StartStream( PaStream *stream );
+static PaError StopStream( PaStream *stream );
+static PaError AbortStream( PaStream *stream );
+static PaError IsStreamStopped( PaStream *s );
+static PaError IsStreamActive( PaStream *stream );
+static PaTime GetStreamTime( PaStream *stream );
+static void setStreamStartTime( PaStream *stream );
+static OSStatus AudioIOProc( void *inRefCon,
+                               AudioUnitRenderActionFlags *ioActionFlags,
+                               const AudioTimeStamp *inTimeStamp,
+                               UInt32 inBusNumber,
+                               UInt32 inNumberFrames,
+                               AudioBufferList *ioData );
+static double GetStreamCpuLoad( PaStream* stream );
+
+static PaError GetChannelInfo( PaMacAUHAL *auhalHostApi,
+                               PaDeviceInfo *deviceInfo,
+                               AudioDeviceID macCoreDeviceId,
+                               int isInput);
+
+static PaError OpenAndSetupOneAudioUnit(
+                                   const PaStreamParameters *inStreamParams,
+                                   const PaStreamParameters *outStreamParams,
+                                   const unsigned long requestedFramesPerBuffer,
+                                   unsigned long *actualInputFramesPerBuffer,
+                                   unsigned long *actualOutputFramesPerBuffer,
+                                   const PaMacAUHAL *auhalHostApi,
+                                   AudioUnit *audioUnit,
+                                   AudioConverterRef *srConverter,
+                                   AudioDeviceID *audioDevice,
+                                   const double sampleRate,
+                                   void *refCon );
+
+/* for setting errors. */
+#define PA_AUHAL_SET_LAST_HOST_ERROR( errorCode, errorText ) \
+    PaUtil_SetLastHostErrorInfo( paInDevelopment, errorCode, errorText )
+
+
+/*currently, this is only used in initialization, but it might be modified
+  to be used when the list of devices changes.*/
+static PaError gatherDeviceInfo(PaMacAUHAL *auhalHostApi)
 {
-    if( macCoreHostApi->allocations )
+    UInt32 size;
+    UInt32 propsize;
+    VVDBUG(("gatherDeviceInfo()\n"));
+    /* -- free any previous allocations -- */
+    if( auhalHostApi->devIds )
+        PaUtil_GroupFreeMemory(auhalHostApi->allocations, auhalHostApi->devIds);
+    auhalHostApi->devIds = NULL;
+
+    /* -- figure out how many devices there are -- */
+    AudioHardwareGetPropertyInfo( kAudioHardwarePropertyDevices,
+                                  &propsize,
+                                  NULL );
+    auhalHostApi->devCount = propsize / sizeof( AudioDeviceID );
+
+    VDBUG( ( "Found %ld device(s).\n", auhalHostApi->devCount ) );
+
+    /* -- copy the device IDs -- */
+    auhalHostApi->devIds = (AudioDeviceID *)PaUtil_GroupAllocateMemory(
+                             auhalHostApi->allocations,
+                             propsize );
+    if( !auhalHostApi->devIds )
+        return paInsufficientMemory;
+    AudioHardwareGetProperty( kAudioHardwarePropertyDevices,
+                                  &propsize,
+                                  auhalHostApi->devIds );
+#ifdef MAC_CORE_VERBOSE_DEBUG
     {
-        PaUtil_FreeAllAllocations( macCoreHostApi->allocations );
-        PaUtil_DestroyAllocationGroup( macCoreHostApi->allocations );
+       int i;
+       for( i=0; i<auhalHostApi->devCount; ++i )
+          printf( "Device %d\t: %ld\n", i, auhalHostApi->devIds[i] );
     }
-    
-    PaUtil_FreeMemory( macCoreHostApi );    
+#endif
+
+    size = sizeof(AudioDeviceID);
+    auhalHostApi->defaultIn  = kAudioDeviceUnknown;
+    auhalHostApi->defaultOut = kAudioDeviceUnknown;
+
+    /* determine the default device. */
+    /* I am not sure how these calls to AudioHardwareGetProperty()
+       could fail, but in case they do, we use the first available
+       device as the default. */
+    if( 0 != AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice,
+                     &size,
+                     &auhalHostApi->defaultIn) ) {
+       int i;
+       auhalHostApi->defaultIn  = kAudioDeviceUnknown;
+       VDBUG(("Failed to get default input device from OS."));
+       VDBUG((" I will substitute the first available input Device."));
+       for( i=0; i<auhalHostApi->devCount; ++i ) {
+          PaDeviceInfo devInfo;
+          if( 0 != GetChannelInfo( auhalHostApi, &devInfo,
+                                   auhalHostApi->devIds[i], TRUE ) )
+             if( devInfo.maxInputChannels ) {
+                auhalHostApi->defaultIn = auhalHostApi->devIds[i];
+                break;
+             }
+       }
+    }   
+    if( 0 != AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice,
+                     &size,
+                     &auhalHostApi->defaultOut) ) {
+       int i;
+       auhalHostApi->defaultIn  = kAudioDeviceUnknown;
+       VDBUG(("Failed to get default output device from OS."));
+       VDBUG((" I will substitute the first available output Device."));
+       for( i=0; i<auhalHostApi->devCount; ++i ) {
+          PaDeviceInfo devInfo;
+          if( 0 != GetChannelInfo( auhalHostApi, &devInfo,
+                                   auhalHostApi->devIds[i], FALSE ) )
+             if( devInfo.maxOutputChannels ) {
+                auhalHostApi->defaultOut = auhalHostApi->devIds[i];
+                break;
+             }
+       }
+    }   
+
+    VDBUG( ( "Default in : %ld\n", auhalHostApi->defaultIn  ) );
+    VDBUG( ( "Default out: %ld\n", auhalHostApi->defaultOut ) );
+
+    return paNoError;
 }
 
-static PaError GetChannelInfo(PaDeviceInfo *deviceInfo, AudioDeviceID macCoreDeviceId, int isInput)
+static PaError GetChannelInfo( PaMacAUHAL *auhalHostApi,
+                               PaDeviceInfo *deviceInfo,
+                               AudioDeviceID macCoreDeviceId,
+                               int isInput)
 {
     UInt32 propSize;
     PaError err = paNoError;
     UInt32 i;
     int numChannels = 0;
-    AudioBufferList *buflist;
+    AudioBufferList *buflist = NULL;
+    UInt32 frameLatency;
+
+    VVDBUG(("GetChannelInfo()\n"));
+
+    /* Get the number of channels from the stream configuration.
+       Fail if we can't get this. */
+
+    err = ERR(AudioDeviceGetPropertyInfo(macCoreDeviceId, 0, isInput, kAudioDevicePropertyStreamConfiguration, &propSize, NULL));
+    if (err)
+        return err;
 
-    err = conv_err(AudioDeviceGetPropertyInfo(macCoreDeviceId, 0, isInput, kAudioDevicePropertyStreamConfiguration, &propSize, NULL));
     buflist = PaUtil_AllocateMemory(propSize);
-    err = conv_err(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyStreamConfiguration, &propSize, buflist));
-    if (!err) {
-        for (i = 0; i < buflist->mNumberBuffers; ++i) {
-            numChannels += buflist->mBuffers[i].mNumberChannels;
-        }
-		
-		if (isInput)
-			deviceInfo->maxInputChannels = numChannels;
-		else
-			deviceInfo->maxOutputChannels = numChannels;
-		
-        int frameLatency;
-        propSize = sizeof(UInt32);
-        err = conv_err(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyLatency, &propSize, &frameLatency));
-        if (!err) {
-            double secondLatency = frameLatency / deviceInfo->defaultSampleRate;
-            if (isInput) {
-                deviceInfo->defaultLowInputLatency = secondLatency;
-                deviceInfo->defaultHighInputLatency = secondLatency;
-            }
-            else {
-                deviceInfo->defaultLowOutputLatency = secondLatency;
-                deviceInfo->defaultHighOutputLatency = secondLatency;
-            }
-        }
+    if( !buflist )
+       return paInsufficientMemory;
+    err = ERR(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyStreamConfiguration, &propSize, buflist));
+    if (err)
+        goto error;
+
+    for (i = 0; i < buflist->mNumberBuffers; ++i)
+        numChannels += buflist->mBuffers[i].mNumberChannels;
+
+    if (isInput)
+        deviceInfo->maxInputChannels = numChannels;
+    else
+        deviceInfo->maxOutputChannels = numChannels;
+      
+    if (numChannels > 0) /* do not try to retrieve the latency if there is no channels. */
+    {
+       /* Get the latency.  Don't fail if we can't get this. */
+       /* default to something reasonable */
+       deviceInfo->defaultLowInputLatency = .01;
+       deviceInfo->defaultHighInputLatency = .10;
+       deviceInfo->defaultLowOutputLatency = .01;
+       deviceInfo->defaultHighOutputLatency = .10;
+       propSize = sizeof(UInt32);
+       err = WARNING(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyLatency, &propSize, &frameLatency));
+       if (!err)
+       {
+          /** FEEDBACK:
+           * This code was arrived at by trial and error, and some extentive, but not exhaustive
+           * testing. Sebastien Beaulieu <seb@plogue.com> has suggested using
+           * kAudioDevicePropertyLatency + kAudioDevicePropertySafetyOffset + buffer size instead.
+           * At the time this code was written, many users were reporting dropouts with audio
+           * programs that probably used this formula. This was probably
+           * around 10.4.4, and the problem is probably fixed now. So perhaps
+           * his formula should be reviewed and used.
+           * */
+          double secondLatency = frameLatency / deviceInfo->defaultSampleRate;
+          if (isInput)
+          {
+             deviceInfo->defaultLowInputLatency = 3 * secondLatency;
+             deviceInfo->defaultHighInputLatency = 3 * 10 * secondLatency;
+          }
+          else
+          {
+             deviceInfo->defaultLowOutputLatency = 3 * secondLatency;
+             deviceInfo->defaultHighOutputLatency = 3 * 10 * secondLatency;
+          }
+       }
     }
-    PaUtil_FreeMemory(buflist);
-    
+    PaUtil_FreeMemory( buflist );
+    return paNoError;
+ error:
+    PaUtil_FreeMemory( buflist );
     return err;
 }
 
-static PaError InitializeDeviceInfo(PaMacCoreDeviceInfo *macCoreDeviceInfo,  AudioDeviceID macCoreDeviceId, PaHostApiIndex hostApiIndex )
+static PaError InitializeDeviceInfo( PaMacAUHAL *auhalHostApi,
+                                     PaDeviceInfo *deviceInfo,
+                                     AudioDeviceID macCoreDeviceId,
+                                     PaHostApiIndex hostApiIndex )
 {
-    PaDeviceInfo *deviceInfo = &macCoreDeviceInfo->inheritedDeviceInfo;
-    deviceInfo->structVersion = 2;
-    deviceInfo->hostApi = hostApiIndex;
-    
+    Float64 sampleRate;
+    char *name;
     PaError err = paNoError;
     UInt32 propSize;
 
-    err = conv_err(AudioDeviceGetPropertyInfo(macCoreDeviceId, 0, 0, kAudioDevicePropertyDeviceName, &propSize, NULL));
-    // FIXME: this allocation should be part of the allocations group
-    char *name = PaUtil_AllocateMemory(propSize);
-    err = conv_err(AudioDeviceGetProperty(macCoreDeviceId, 0, 0, kAudioDevicePropertyDeviceName, &propSize, name));
-    if (!err) {
-        deviceInfo->name = name;
-    }
-    
-    Float64 sampleRate;
+    VVDBUG(("InitializeDeviceInfo(): macCoreDeviceId=%ld\n", macCoreDeviceId));
+
+    memset(deviceInfo, 0, sizeof(deviceInfo));
+
+    deviceInfo->structVersion = 2;
+    deviceInfo->hostApi = hostApiIndex;
+
+    /* Get the device name.  Fail if we can't get it. */
+    err = ERR(AudioDeviceGetPropertyInfo(macCoreDeviceId, 0, 0, kAudioDevicePropertyDeviceName, &propSize, NULL));
+    if (err)
+        return err;
+
+    name = PaUtil_GroupAllocateMemory(auhalHostApi->allocations,propSize);
+    if ( !name )
+        return paInsufficientMemory;
+    err = ERR(AudioDeviceGetProperty(macCoreDeviceId, 0, 0, kAudioDevicePropertyDeviceName, &propSize, name));
+    if (err)
+        return err;
+    deviceInfo->name = name;
+
+    /* Try to get the default sample rate.  Don't fail if we can't get this. */
     propSize = sizeof(Float64);
-    err = conv_err(AudioDeviceGetProperty(macCoreDeviceId, 0, 0, kAudioDevicePropertyNominalSampleRate, &propSize, &sampleRate));
-    if (!err) {
+    err = ERR(AudioDeviceGetProperty(macCoreDeviceId, 0, 0, kAudioDevicePropertyNominalSampleRate, &propSize, &sampleRate));
+    if (err)
+        deviceInfo->defaultSampleRate = 0.0;
+    else
         deviceInfo->defaultSampleRate = sampleRate;
-    }
 
+    /* Get the maximum number of input and output channels.  Fail if we can't get this. */
 
-    // Get channel info
-    err = GetChannelInfo(deviceInfo, macCoreDeviceId, 1);
-    err = GetChannelInfo(deviceInfo, macCoreDeviceId, 0);
+    err = GetChannelInfo(auhalHostApi, deviceInfo, macCoreDeviceId, 1);
+    if (err)
+        return err;
 
-    return err;
+    err = GetChannelInfo(auhalHostApi, deviceInfo, macCoreDeviceId, 0);
+    if (err)
+        return err;
+
+    return paNoError;
 }
 
-static PaError InitializeDeviceInfos( PaMacCoreHostApiRepresentation *macCoreHostApi, PaHostApiIndex hostApiIndex )
+PaError PaMacCore_Initialize( PaUtilHostApiRepresentation **hostApi, PaHostApiIndex hostApiIndex )
 {
     PaError result = paNoError;
-    PaUtilHostApiRepresentation *hostApi;
-    PaMacCoreDeviceInfo *deviceInfoArray;
-
-    // initialise device counts and default devices under the assumption that there are no devices. These values are incremented below if and when devices are successfully initialized.
-    hostApi = &macCoreHostApi->inheritedHostApiRep;
-    hostApi->info.deviceCount = 0;
-    hostApi->info.defaultInputDevice = paNoDevice;
-    hostApi->info.defaultOutputDevice = paNoDevice;
-    
-    UInt32 propsize;
-    AudioHardwareGetPropertyInfo(kAudioHardwarePropertyDevices, &propsize, NULL);
-    int numDevices = propsize / sizeof(AudioDeviceID);
-    hostApi->info.deviceCount = numDevices;
-    if (numDevices > 0) {
-        hostApi->deviceInfos = (PaDeviceInfo**)PaUtil_GroupAllocateMemory(
-                                            macCoreHostApi->allocations, sizeof(PaDeviceInfo*) * numDevices );
-        if( !hostApi->deviceInfos )
+    int i;
+    PaMacAUHAL *auhalHostApi;
+    PaDeviceInfo *deviceInfoArray;
+
+    VVDBUG(("PaMacCore_Initialize(): hostApiIndex=%d\n", hostApiIndex));
+
+    auhalHostApi = (PaMacAUHAL*)PaUtil_AllocateMemory( sizeof(PaMacAUHAL) );
+    if( !auhalHostApi )
+    {
+        result = paInsufficientMemory;
+        goto error;
+    }
+
+    auhalHostApi->allocations = PaUtil_CreateAllocationGroup();
+    if( !auhalHostApi->allocations )
+    {
+        result = paInsufficientMemory;
+        goto error;
+    }
+
+    auhalHostApi->devIds = NULL;
+    auhalHostApi->devCount = 0;
+
+    /* get the info we need about the devices */
+    result = gatherDeviceInfo( auhalHostApi );
+    if( result != paNoError )
+       goto error;
+
+    *hostApi = &auhalHostApi->inheritedHostApiRep;
+    (*hostApi)->info.structVersion = 1;
+    (*hostApi)->info.type = paCoreAudio;
+    (*hostApi)->info.name = "Core Audio";
+
+    (*hostApi)->info.defaultInputDevice = paNoDevice;
+    (*hostApi)->info.defaultOutputDevice = paNoDevice;
+
+    (*hostApi)->info.deviceCount = 0;  
+
+    if( auhalHostApi->devCount > 0 )
+    {
+        (*hostApi)->deviceInfos = (PaDeviceInfo**)PaUtil_GroupAllocateMemory(
+                auhalHostApi->allocations, sizeof(PaDeviceInfo*) * auhalHostApi->devCount);
+        if( !(*hostApi)->deviceInfos )
         {
-            return paInsufficientMemory;
+            result = paInsufficientMemory;
+            goto error;
         }
 
-        // allocate all device info structs in a contiguous block
-        deviceInfoArray = (PaMacCoreDeviceInfo*)PaUtil_GroupAllocateMemory(
-                                macCoreHostApi->allocations, sizeof(PaMacCoreDeviceInfo) * numDevices );
+        /* allocate all device info structs in a contiguous block */
+        deviceInfoArray = (PaDeviceInfo*)PaUtil_GroupAllocateMemory(
+                auhalHostApi->allocations, sizeof(PaDeviceInfo) * auhalHostApi->devCount );
         if( !deviceInfoArray )
         {
-            return paInsufficientMemory;
+            result = paInsufficientMemory;
+            goto error;
         }
-        
-        macCoreHostApi->macCoreDeviceIds = PaUtil_GroupAllocateMemory(macCoreHostApi->allocations, propsize);
-        AudioHardwareGetProperty(kAudioHardwarePropertyDevices, &propsize, macCoreHostApi->macCoreDeviceIds);
 
-        AudioDeviceID defaultInputDevice, defaultOutputDevice;
-        propsize = sizeof(AudioDeviceID);
-        AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice, &propsize, &defaultInputDevice);
-        AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, &propsize, &defaultOutputDevice);
-        
-        UInt32 i;
-        for (i = 0; i < numDevices; ++i) {
-            if (macCoreHostApi->macCoreDeviceIds[i] == defaultInputDevice) {
-                hostApi->info.defaultInputDevice = i;
+        for( i=0; i < auhalHostApi->devCount; ++i )
+        {
+            int err;
+            err = InitializeDeviceInfo( auhalHostApi, &deviceInfoArray[i],
+                                      auhalHostApi->devIds[i],
+                                      hostApiIndex );
+            if (err == paNoError)
+            { /* copy some info and set the defaults */
+                (*hostApi)->deviceInfos[(*hostApi)->info.deviceCount] = &deviceInfoArray[i];
+                if (auhalHostApi->devIds[i] == auhalHostApi->defaultIn)
+                    (*hostApi)->info.defaultInputDevice = (*hostApi)->info.deviceCount;
+                if (auhalHostApi->devIds[i] == auhalHostApi->defaultOut)
+                    (*hostApi)->info.defaultOutputDevice = (*hostApi)->info.deviceCount;
+                (*hostApi)->info.deviceCount++;
             }
-            if (macCoreHostApi->macCoreDeviceIds[i] == defaultOutputDevice) {
-                hostApi->info.defaultOutputDevice = i;
+            else
+            { /* there was an error. we need to shift the devices down, so we ignore this one */
+                int j;
+                auhalHostApi->devCount--;
+                for( j=i; j<auhalHostApi->devCount; ++j )
+                   auhalHostApi->devIds[j] = auhalHostApi->devIds[j+1];
+                i--;
             }
-            InitializeDeviceInfo(&deviceInfoArray[i], macCoreHostApi->macCoreDeviceIds[i], hostApiIndex);
-            hostApi->deviceInfos[i] = &(deviceInfoArray[i].inheritedDeviceInfo);      
         }
     }
 
-    return result;
-}
+    (*hostApi)->Terminate = Terminate;
+    (*hostApi)->OpenStream = OpenStream;
+    (*hostApi)->IsFormatSupported = IsFormatSupported;
+
+    PaUtil_InitializeStreamInterface( &auhalHostApi->callbackStreamInterface,
+                                      CloseStream, StartStream,
+                                      StopStream, AbortStream, IsStreamStopped,
+                                      IsStreamActive,
+                                      GetStreamTime, GetStreamCpuLoad,
+                                      PaUtil_DummyRead, PaUtil_DummyWrite,
+                                      PaUtil_DummyGetReadAvailable,
+                                      PaUtil_DummyGetWriteAvailable );
+
+    PaUtil_InitializeStreamInterface( &auhalHostApi->blockingStreamInterface,
+                                      CloseStream, StartStream,
+                                      StopStream, AbortStream, IsStreamStopped,
+                                      IsStreamActive,
+                                      GetStreamTime, PaUtil_DummyGetCpuLoad,
+                                      ReadStream, WriteStream,
+                                      GetStreamReadAvailable,
+                                      GetStreamWriteAvailable );
 
-static OSStatus CheckFormat(AudioDeviceID macCoreDeviceId, const PaStreamParameters *parameters, double sampleRate, int isInput)
-{
-    UInt32 propSize = sizeof(AudioStreamBasicDescription);
-    AudioStreamBasicDescription *streamDescription = PaUtil_AllocateMemory(propSize);
-
-    streamDescription->mSampleRate = sampleRate;
-    streamDescription->mFormatID = 0;
-    streamDescription->mFormatFlags = 0;
-    streamDescription->mBytesPerPacket = 0;
-    streamDescription->mFramesPerPacket = 0;
-    streamDescription->mBytesPerFrame = 0;
-    streamDescription->mChannelsPerFrame = 0;
-    streamDescription->mBitsPerChannel = 0;
-    streamDescription->mReserved = 0;
-
-    OSStatus result = AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyStreamFormatSupported, &propSize, streamDescription);
-    PaUtil_FreeMemory(streamDescription);
     return result;
-}
 
-static OSStatus CopyInputData(PaMacClientData* destination, const AudioBufferList *source, unsigned long frameCount)
-{
-    int frameSpacing, channelSpacing;
-    if (destination->inputSampleFormat & paNonInterleaved) {
-        frameSpacing = 1;
-        channelSpacing = destination->inputChannelCount;
-    }
-    else {
-        frameSpacing = destination->inputChannelCount;
-        channelSpacing = 1;
-    }
-    
-    AudioBuffer const *inputBuffer = &source->mBuffers[0];
-    void *coreAudioBuffer = inputBuffer->mData;
-    void *portAudioBuffer = destination->inputBuffer;
-    UInt32 i, streamNumber, streamChannel;
-    for (i = streamNumber = streamChannel = 0; i < destination->inputChannelCount; ++i, ++streamChannel) {
-        if (streamChannel >= inputBuffer->mNumberChannels) {
-            ++streamNumber;
-            inputBuffer = &source->mBuffers[streamNumber];
-            coreAudioBuffer = inputBuffer->mData;
-            streamChannel = 0;
+error:
+    if( auhalHostApi )
+    {
+        if( auhalHostApi->allocations )
+        {
+            PaUtil_FreeAllAllocations( auhalHostApi->allocations );
+            PaUtil_DestroyAllocationGroup( auhalHostApi->allocations );
         }
-        destination->inputConverter(portAudioBuffer, frameSpacing, coreAudioBuffer, inputBuffer->mNumberChannels, frameCount, destination->ditherGenerator);
-        coreAudioBuffer += sizeof(Float32);
-        portAudioBuffer += Pa_GetSampleSize(destination->inputSampleFormat) * channelSpacing;
+                
+        PaUtil_FreeMemory( auhalHostApi );
     }
-    return 0;
+    return result;
 }
 
-static OSStatus CopyOutputData(AudioBufferList* destination, PaMacClientData *source, unsigned long frameCount)
-{
-    int frameSpacing, channelSpacing;
-    if (source->outputSampleFormat & paNonInterleaved) {
-        frameSpacing = 1;
-        channelSpacing = source->outputChannelCount;
-    }
-    else {
-        frameSpacing = source->outputChannelCount;
-        channelSpacing = 1;
-    }
-    
-    AudioBuffer *outputBuffer = &destination->mBuffers[0];
-    void *coreAudioBuffer = outputBuffer->mData;
-    void *portAudioBuffer = source->outputBuffer;
-    UInt32 i, streamNumber, streamChannel;
-    for (i = streamNumber = streamChannel = 0; i < source->outputChannelCount; ++i, ++streamChannel) {
-        if (streamChannel >= outputBuffer->mNumberChannels) {
-            ++streamNumber;
-            outputBuffer = &destination->mBuffers[streamNumber];
-            coreAudioBuffer = outputBuffer->mData;
-            streamChannel = 0;
-        }
-        source->outputConverter(coreAudioBuffer, outputBuffer->mNumberChannels, portAudioBuffer, frameSpacing, frameCount, NULL);
-        coreAudioBuffer += sizeof(Float32);
-        portAudioBuffer += Pa_GetSampleSize(source->outputSampleFormat) * channelSpacing;
-    }    
-    return 0;
-}
 
-static OSStatus AudioIOProc( AudioDeviceID inDevice,
-                      const AudioTimeStamp* inNow,
-                      const AudioBufferList* inInputData,
-                      const AudioTimeStamp* inInputTime,
-                      AudioBufferList* outOutputData, 
-                      const AudioTimeStamp* inOutputTime,
-                      void* inClientData)
+static void Terminate( struct PaUtilHostApiRepresentation *hostApi )
 {
-    PaMacClientData *clientData = (PaMacClientData *)inClientData;
-    PaStreamCallbackTimeInfo *timeInfo = InitializeTimeInfo(inNow, inInputTime, inOutputTime);
-    
-    PaUtil_BeginCpuLoadMeasurement( &clientData->stream->cpuLoadMeasurer );
-    
-    AudioBuffer *outputBuffer = &outOutputData->mBuffers[0];
-    unsigned long frameCount = outputBuffer->mDataByteSize / (outputBuffer->mNumberChannels * sizeof(Float32));
+    PaMacAUHAL *auhalHostApi = (PaMacAUHAL*)hostApi;
 
-    if (clientData->inputBuffer) {
-        CopyInputData(clientData, inInputData, frameCount);
-    }
-    PaStreamCallbackResult result = clientData->callback(clientData->inputBuffer, clientData->outputBuffer, frameCount, timeInfo, paNoFlag, clientData->userData);
-    if (clientData->outputBuffer) {
-        CopyOutputData(outOutputData, clientData, frameCount);
-    }
+    VVDBUG(("Terminate()\n"));
 
-    PaUtil_EndCpuLoadMeasurement( &clientData->stream->cpuLoadMeasurer, frameCount );
-    
-    if (result == paComplete || result == paAbort) {
-        Pa_StopStream(clientData->stream);
-    }
-    return 0;
-}
-
-// This is not for input-only streams, this is for streams where the input device is different from the output device
-// TODO: This needs to store the output data in a buffer, to be written to the device the next time AudioOutputProc is called
-static OSStatus AudioInputProc( AudioDeviceID inDevice,
-                         const AudioTimeStamp* inNow,
-                         const AudioBufferList* inInputData,
-                         const AudioTimeStamp* inInputTime,
-                         AudioBufferList* outOutputData, 
-                         const AudioTimeStamp* inOutputTime,
-                         void* inClientData)
-{
-    PaMacClientData *clientData = (PaMacClientData *)inClientData;
-    PaStreamCallbackTimeInfo *timeInfo = InitializeTimeInfo(inNow, inInputTime, inOutputTime);
-
-    PaUtil_BeginCpuLoadMeasurement( &clientData->stream->cpuLoadMeasurer );
+    /*
+        IMPLEMENT ME:
+            - clean up any resources not handled by the allocation group
+        TODO: Double check that everything is handled by alloc group
+    */
 
-    AudioBuffer const *inputBuffer = &inInputData->mBuffers[0];
-    unsigned long frameCount = inputBuffer->mDataByteSize / (inputBuffer->mNumberChannels * sizeof(Float32));
+    if( auhalHostApi->allocations )
+    {
+        PaUtil_FreeAllAllocations( auhalHostApi->allocations );
+        PaUtil_DestroyAllocationGroup( auhalHostApi->allocations );
+    }
 
-    CopyInputData(clientData, inInputData, frameCount);
-    clientData->callback(clientData->inputBuffer, NULL, frameCount, timeInfo, paNoFlag, clientData->userData);
-    
-    PaUtil_EndCpuLoadMeasurement( &clientData->stream->cpuLoadMeasurer, frameCount );
-    return 0;
+    PaUtil_FreeMemory( auhalHostApi );
 }
 
-// This is not for output-only streams, this is for streams where the input device is different from the output device
-static OSStatus AudioOutputProc( AudioDeviceID inDevice,
-                          const AudioTimeStamp* inNow,
-                          const AudioBufferList* inInputData,
-                          const AudioTimeStamp* inInputTime,
-                          AudioBufferList* outOutputData, 
-                          const AudioTimeStamp* inOutputTime,
-                          void* inClientData)
+
+static PaError IsFormatSupported( struct PaUtilHostApiRepresentation *hostApi,
+                                  const PaStreamParameters *inputParameters,
+                                  const PaStreamParameters *outputParameters,
+                                  double sampleRate )
 {
-    PaMacClientData *clientData = (PaMacClientData *)inClientData;
-    PaStreamCallbackTimeInfo *timeInfo = InitializeTimeInfo(inNow, inInputTime, inOutputTime);
+    int inputChannelCount, outputChannelCount;
+    PaSampleFormat inputSampleFormat, outputSampleFormat;
+
+    VVDBUG(("IsFormatSupported(): in chan=%d, in fmt=%ld, out chan=%d, out fmt=%ld sampleRate=%g\n",
+                inputParameters  ? inputParameters->channelCount  : -1,
+                inputParameters  ? inputParameters->sampleFormat  : -1,
+                outputParameters ? outputParameters->channelCount : -1,
+                outputParameters ? outputParameters->sampleFormat : -1,
+                (float) sampleRate ));
+ 
+    /** These first checks are standard PA checks. We do some fancier checks
+        later. */
+    if( inputParameters )
+    {
+        inputChannelCount = inputParameters->channelCount;
+        inputSampleFormat = inputParameters->sampleFormat;
+
+        /* all standard sample formats are supported by the buffer adapter,
+            this implementation doesn't support any custom sample formats */
+        if( inputSampleFormat & paCustomFormat )
+            return paSampleFormatNotSupported;
+            
+        /* unless alternate device specification is supported, reject the use of
+            paUseHostApiSpecificDeviceSpecification */
+
+        if( inputParameters->device == paUseHostApiSpecificDeviceSpecification )
+            return paInvalidDevice;
+
+        /* check that input device can support inputChannelCount */
+        if( inputChannelCount > hostApi->deviceInfos[ inputParameters->device ]->maxInputChannels )
+            return paInvalidChannelCount;
+    }
+    else
+    {
+        inputChannelCount = 0;
+    }
 
-    PaUtil_BeginCpuLoadMeasurement( &clientData->stream->cpuLoadMeasurer );
+    if( outputParameters )
+    {
+        outputChannelCount = outputParameters->channelCount;
+        outputSampleFormat = outputParameters->sampleFormat;
 
-    AudioBuffer *outputBuffer = &outOutputData->mBuffers[0];
-    unsigned long frameCount = outputBuffer->mDataByteSize / (outputBuffer->mNumberChannels * sizeof(Float32));
+        /* all standard sample formats are supported by the buffer adapter,
+            this implementation doesn't support any custom sample formats */
+        if( outputSampleFormat & paCustomFormat )
+            return paSampleFormatNotSupported;
+            
+        /* unless alternate device specification is supported, reject the use of
+            paUseHostApiSpecificDeviceSpecification */
 
-    clientData->callback(NULL, clientData->outputBuffer, frameCount, timeInfo, paNoFlag, clientData->userData);
+        if( outputParameters->device == paUseHostApiSpecificDeviceSpecification )
+            return paInvalidDevice;
 
-    CopyOutputData(outOutputData, clientData, frameCount);
+        /* check that output device can support outputChannelCount */
+        if( outputChannelCount > hostApi->deviceInfos[ outputParameters->device ]->maxOutputChannels )
+            return paInvalidChannelCount;
 
-    PaUtil_EndCpuLoadMeasurement( &clientData->stream->cpuLoadMeasurer, frameCount );
-    return 0;
-}
+    }
+    else
+    {
+        outputChannelCount = 0;
+    }
+ 
+    /* FEEDBACK */
+    /*        I think the only way to check a given format SR combo is     */
+    /*        to try opening it. This could be disruptive, is that Okay?   */
+    /*        The alternative is to just read off available sample rates,  */
+    /*        but this will not work %100 of the time (eg, a device that   */
+    /*        supports N output at one rate but only N/2 at a higher rate.)*/
+
+    /* The following code opens the device with the requested parameters to
+       see if it works. */
+    {
+       PaError err;
+       PaStream *s;
+       err = OpenStream( hostApi, &s, inputParameters, outputParameters,
+                           sampleRate, 1024, 0, (PaStreamCallback *)1, NULL );
+       if( err != paNoError && err != paInvalidSampleRate )
+          DBUG( ( "OpenStream @ %g returned: %d: %s\n",
+                  (float) sampleRate, err, Pa_GetErrorText( err ) ) );
+       if( err ) 
+          return err;
+       err = CloseStream( s );
+       if( err ) {
+          /* FEEDBACK: is this more serious? should we assert? */
+          DBUG( ( "WARNING: could not close Stream. %d: %s\n",
+                  err, Pa_GetErrorText( err ) ) );
+       }
+    }
 
-static PaError SetSampleRate(AudioDeviceID device, double sampleRate, int isInput)
-{
-    PaError result = paNoError;
-    
-    double actualSampleRate;
-    UInt32 propSize = sizeof(double);
-    result = conv_err(AudioDeviceSetProperty(device, NULL, 0, isInput, kAudioDevicePropertyNominalSampleRate, propSize, &sampleRate));
-    
-    result = conv_err(AudioDeviceGetProperty(device, 0, isInput, kAudioDevicePropertyNominalSampleRate, &propSize, &actualSampleRate));
-    
-    if (result == paNoError && actualSampleRate != sampleRate) {
-        result = paInvalidSampleRate;
-    }
-    
-    return result;    
+    return paFormatIsSupported;
 }
 
-static PaError SetFramesPerBuffer(AudioDeviceID device, unsigned long framesPerBuffer, int isInput)
+static PaError OpenAndSetupOneAudioUnit(
+                                   const PaStreamParameters *inStreamParams,
+                                   const PaStreamParameters *outStreamParams,
+                                   const unsigned long requestedFramesPerBuffer,
+                                   unsigned long *actualInputFramesPerBuffer,
+                                   unsigned long *actualOutputFramesPerBuffer,
+                                   const PaMacAUHAL *auhalHostApi,
+                                   AudioUnit *audioUnit,
+                                   AudioConverterRef *srConverter,
+                                   AudioDeviceID *audioDevice,
+                                   const double sampleRate,
+                                   void *refCon )
 {
-    PaError result = paNoError;
-    UInt32 preferredFramesPerBuffer = framesPerBuffer;
-    //    while (preferredFramesPerBuffer > UINT32_MAX) {
-    //        preferredFramesPerBuffer /= 2;
-    //    }
-    
-    UInt32 actualFramesPerBuffer;
-    UInt32 propSize = sizeof(UInt32);
-    result = conv_err(AudioDeviceSetProperty(device, NULL, 0, isInput, kAudioDevicePropertyBufferFrameSize, propSize, &preferredFramesPerBuffer));
-    
-    result = conv_err(AudioDeviceGetProperty(device, 0, isInput, kAudioDevicePropertyBufferFrameSize, &propSize, &actualFramesPerBuffer));
-    
-    if (result != paNoError) {
-        // do nothing
-    }
-    else if (actualFramesPerBuffer > framesPerBuffer) {
-        result = paBufferTooSmall;
-    }
-    else if (actualFramesPerBuffer < framesPerBuffer) {
-        result = paBufferTooBig;
-    }
-    
-    return result;    
-}
-    
-static PaError SetUpUnidirectionalStream(AudioDeviceID device, double sampleRate, unsigned long framesPerBuffer, int isInput)
-{
-    PaError err = paNoError;
-    err = SetSampleRate(device, sampleRate, isInput);
-    if( err == paNoError )
-        err = SetFramesPerBuffer(device, framesPerBuffer, isInput);
-    return err;
-}
+    ComponentDescription desc;
+    Component comp;
+    /*An Apple TN suggests using CAStreamBasicDescription, but that is C++*/
+    AudioStreamBasicDescription desiredFormat;
+    OSErr result = noErr;
+    PaError paResult = paNoError;
+    int line = 0;
+    UInt32 callbackKey;
+    AURenderCallbackStruct rcbs;
+    unsigned long macInputStreamFlags  = paMacCorePlayNice;
+    unsigned long macOutputStreamFlags = paMacCorePlayNice;
+
+    VVDBUG(("OpenAndSetupOneAudioUnit(): in chan=%d, in fmt=%ld, out chan=%d, out fmt=%ld, requestedFramesPerBuffer=%ld\n",
+                inStreamParams  ? inStreamParams->channelCount  : -1,
+                inStreamParams  ? inStreamParams->sampleFormat  : -1,
+                outStreamParams ? outStreamParams->channelCount : -1,
+                outStreamParams ? outStreamParams->sampleFormat : -1,
+                requestedFramesPerBuffer ));
+
+    /* -- handle the degenerate case  -- */
+    if( !inStreamParams && !outStreamParams ) {
+       *audioUnit = NULL;
+       *audioDevice = kAudioDeviceUnknown;
+       return paNoError;
+    }
 
-// =====  PortAudio functions  =====
-#pragma mark PortAudio functions
+    /* -- get the user's api specific info, if they set any -- */
+    if( inStreamParams && inStreamParams->hostApiSpecificStreamInfo )
+       macInputStreamFlags=
+            ((paMacCoreStreamInfo*)inStreamParams->hostApiSpecificStreamInfo)
+                  ->flags;
+    if( outStreamParams && outStreamParams->hostApiSpecificStreamInfo )
+       macOutputStreamFlags=
+            ((paMacCoreStreamInfo*)outStreamParams->hostApiSpecificStreamInfo)
+                  ->flags;
+    /* Override user's flags here, if desired for testing. */
+
+    /*
+     * The HAL AU is a Mac OS style "component".
+     * the first few steps deal with that.
+     * Later steps work on a combination of Mac OS
+     * components and the slightly lower level
+     * HAL.
+     */
+
+    /* -- describe the output type AudioUnit -- */
+    /*  Note: for the default AudioUnit, we could use the
+     *  componentSubType value kAudioUnitSubType_DefaultOutput;
+     *  but I don't think that's relevant here.
+     */
+    desc.componentType         = kAudioUnitType_Output;
+    desc.componentSubType      = kAudioUnitSubType_HALOutput;
+    desc.componentManufacturer = kAudioUnitManufacturer_Apple;
+    desc.componentFlags        = 0;
+    desc.componentFlagsMask    = 0;
+    /* -- find the component -- */
+    comp = FindNextComponent( NULL, &desc );
+    if( !comp )
+    {
+       DBUG( ( "AUHAL component not found." ) );
+       *audioUnit = NULL;
+       *audioDevice = kAudioDeviceUnknown;
+       return paUnanticipatedHostError;
+    }
+    /* -- open it -- */
+    result = OpenAComponent( comp, audioUnit );
+    if( result )
+    {
+       DBUG( ( "Failed to open AUHAL component." ) );
+       *audioUnit = NULL;
+       *audioDevice = kAudioDeviceUnknown;
+       return ERR( result );
+    }
+    /* -- prepare a little error handling logic / hackery -- */
+#define ERR_WRAP(mac_err) do { result = mac_err ; line = __LINE__ ; if ( result != noErr ) goto error ; } while(0)
 
-#ifdef __cplusplus
-extern "C"
-{
-#endif // __cplusplus
-    
-    PaError PaMacCore_Initialize( PaUtilHostApiRepresentation **hostApi, PaHostApiIndex index );
-    
-#ifdef __cplusplus
-}
-#endif // __cplusplus
+    /* -- if there is input, we have to explicitly enable input -- */
+    if( inStreamParams )
+    {
+       UInt32 enableIO;
+       enableIO = 1;
+       ERR_WRAP( AudioUnitSetProperty( *audioUnit,
+                 kAudioOutputUnitProperty_EnableIO,
+                 kAudioUnitScope_Input,
+                 INPUT_ELEMENT,
+                 &enableIO,
+                 sizeof(enableIO) ) );
+    }
+    /* -- if there is no output, we must explicitly disable output -- */
+    if( !outStreamParams )
+    {
+       UInt32 enableIO;
+       enableIO = 0;
+       ERR_WRAP( AudioUnitSetProperty( *audioUnit,
+                 kAudioOutputUnitProperty_EnableIO,
+                 kAudioUnitScope_Output,
+                 OUTPUT_ELEMENT,
+                 &enableIO,
+                 sizeof(enableIO) ) );
+    }
+    /* -- set the devices -- */
+    /* make sure input and output are the same device if we are doing input and
+       output. */
+    if( inStreamParams && outStreamParams )
+       assert( outStreamParams->device == inStreamParams->device );
+    if( inStreamParams )
+    {
+       *audioDevice = auhalHostApi->devIds[inStreamParams->device] ;
+       ERR_WRAP( AudioUnitSetProperty( *audioUnit,
+                    kAudioOutputUnitProperty_CurrentDevice,
+                    kAudioUnitScope_Global,
+                    INPUT_ELEMENT,
+                    audioDevice,
+                    sizeof(AudioDeviceID) ) );
+    }
+    if( outStreamParams )
+    {
+       *audioDevice = auhalHostApi->devIds[outStreamParams->device] ;
+       ERR_WRAP( AudioUnitSetProperty( *audioUnit,
+                    kAudioOutputUnitProperty_CurrentDevice,
+                    kAudioUnitScope_Global,
+                    OUTPUT_ELEMENT,
+                    audioDevice,
+                    sizeof(AudioDeviceID) ) );
+    }
 
-static void Terminate( struct PaUtilHostApiRepresentation *hostApi )
-{
-    PaMacCoreHostApiRepresentation *macCoreHostApi = (PaMacCoreHostApiRepresentation*)hostApi;
-    
-    CleanUp(macCoreHostApi);
-}
+    /* -- set format -- */
+    bzero( &desiredFormat, sizeof(desiredFormat) );
+    desiredFormat.mFormatID         = kAudioFormatLinearPCM ;
+    desiredFormat.mFormatFlags      = kAudioFormatFlagsNativeFloatPacked;
+    desiredFormat.mFramesPerPacket  = 1;
+    desiredFormat.mBitsPerChannel   = sizeof( float ) * 8;
+
+    result = 0;
+    /*  set device format first, but only touch the device if the user asked */
+    if( inStreamParams ) {
+       /*The callback never calls back if we don't set the FPB */
+       /*This seems wierd, because I would think setting anything on the device
+         would be disruptive.*/
+       paResult = setBestFramesPerBuffer( *audioDevice, FALSE,
+                                          requestedFramesPerBuffer,
+                                          actualInputFramesPerBuffer );
+       if( paResult ) goto error;
+       if( macInputStreamFlags & paMacCore_ChangeDeviceParameters ) {
+          bool requireExact;
+          requireExact=macInputStreamFlags&paMacCore_FailIfConversionRequired;
+          paResult = setBestSampleRateForDevice( *audioDevice, FALSE,
+                                                 requireExact, sampleRate );
+          if( paResult ) goto error;
+       }
+       if( actualInputFramesPerBuffer && actualOutputFramesPerBuffer )
+          *actualOutputFramesPerBuffer = *actualInputFramesPerBuffer ;
+    }
+    if( outStreamParams && !inStreamParams ) {
+       /*The callback never calls back if we don't set the FPB */
+       /*This seems wierd, because I would think setting anything on the device
+         would be disruptive.*/
+       paResult = setBestFramesPerBuffer( *audioDevice, TRUE,
+                                          requestedFramesPerBuffer,
+                                          actualOutputFramesPerBuffer );
+       if( paResult ) goto error;
+       if( macOutputStreamFlags & paMacCore_ChangeDeviceParameters ) {
+          bool requireExact;
+          requireExact=macOutputStreamFlags&paMacCore_FailIfConversionRequired;
+          paResult = setBestSampleRateForDevice( *audioDevice, TRUE,
+                                                 requireExact, sampleRate );
+          if( paResult ) goto error;
+       }
+    }
 
-static PaError IsFormatSupported( struct PaUtilHostApiRepresentation *hostApi,
-                                  const PaStreamParameters *inputParameters,
-                                  const PaStreamParameters *outputParameters,
-                                  double sampleRate )
-{
-    PaMacCoreHostApiRepresentation *macCoreHostApi = (PaMacCoreHostApiRepresentation*)hostApi;
-    PaDeviceInfo *deviceInfo;
-    
-    PaError result = paNoError;
-    if (inputParameters) {
-        deviceInfo = macCoreHostApi->inheritedHostApiRep.deviceInfos[inputParameters->device];
-        if (inputParameters->channelCount > deviceInfo->maxInputChannels)
-            result = paInvalidChannelCount;
-        else if (CheckFormat(macCoreHostApi->macCoreDeviceIds[inputParameters->device], inputParameters, sampleRate, 1) != kAudioHardwareNoError) {
-            result = paInvalidSampleRate;
-        }
+    /* -- set the quality of the output converter -- */
+    if( outStreamParams ) {
+       UInt32 value = kAudioConverterQuality_Max;
+       switch( macOutputStreamFlags & 0x0700 ) {
+       case 0x0100: /*paMacCore_ConversionQualityMin:*/
+          value=kRenderQuality_Min;
+          break;
+       case 0x0200: /*paMacCore_ConversionQualityLow:*/
+          value=kRenderQuality_Low;
+          break;
+       case 0x0300: /*paMacCore_ConversionQualityMedium:*/
+          value=kRenderQuality_Medium;
+          break;
+       case 0x0400: /*paMacCore_ConversionQualityHigh:*/
+          value=kRenderQuality_High;
+          break;
+       }
+       ERR_WRAP( AudioUnitSetProperty( *audioUnit,
+                    kAudioUnitProperty_RenderQuality,
+                    kAudioUnitScope_Global,
+                    OUTPUT_ELEMENT,
+                    &value,
+                    sizeof(value) ) );
     }
-    if (outputParameters && result == paNoError) {
-        deviceInfo = macCoreHostApi->inheritedHostApiRep.deviceInfos[outputParameters->device];
-        if (outputParameters->channelCount > deviceInfo->maxOutputChannels)
-            result = paInvalidChannelCount;
-        else if (CheckFormat(macCoreHostApi->macCoreDeviceIds[outputParameters->device], outputParameters, sampleRate, 0) != kAudioHardwareNoError) {
-            result = paInvalidSampleRate;
-        }
+    /* now set the format on the Audio Units. */
+    if( outStreamParams )
+    {
+       desiredFormat.mSampleRate    =sampleRate;
+       desiredFormat.mBytesPerPacket=sizeof(float)*outStreamParams->channelCount;
+       desiredFormat.mBytesPerFrame =sizeof(float)*outStreamParams->channelCount;
+       desiredFormat.mChannelsPerFrame = outStreamParams->channelCount;
+       ERR_WRAP( AudioUnitSetProperty( *audioUnit,
+                            kAudioUnitProperty_StreamFormat,
+                            kAudioUnitScope_Input,
+                            OUTPUT_ELEMENT,
+                            &desiredFormat,
+                            sizeof(AudioStreamBasicDescription) ) );
+    }
+    if( inStreamParams )
+    {
+       AudioStreamBasicDescription sourceFormat;
+       UInt32 size = sizeof( AudioStreamBasicDescription );
+
+       /* keep the sample rate of the device, or we confuse AUHAL */
+       ERR_WRAP( AudioUnitGetProperty( *audioUnit,
+                            kAudioUnitProperty_StreamFormat,
+                            kAudioUnitScope_Input,
+                            INPUT_ELEMENT,
+                            &sourceFormat,
+                            &size ) );
+       desiredFormat.mSampleRate = sourceFormat.mSampleRate;
+       desiredFormat.mBytesPerPacket=sizeof(float)*inStreamParams->channelCount;
+       desiredFormat.mBytesPerFrame =sizeof(float)*inStreamParams->channelCount;
+       desiredFormat.mChannelsPerFrame = inStreamParams->channelCount;
+       ERR_WRAP( AudioUnitSetProperty( *audioUnit,
+                            kAudioUnitProperty_StreamFormat,
+                            kAudioUnitScope_Output,
+                            INPUT_ELEMENT,
+                            &desiredFormat,
+                            sizeof(AudioStreamBasicDescription) ) );
+    }
+    /* set the maximumFramesPerSlice */
+    /* not doing this causes real problems
+       (eg. the callback might not be called). The idea of setting both this
+       and the frames per buffer on the device is that we'll be most likely
+       to actually get the frame size we requested in the callback with the
+       minimum latency. */
+    if( outStreamParams ) {
+       UInt32 size = sizeof( *actualOutputFramesPerBuffer );
+       ERR_WRAP( AudioUnitSetProperty( *audioUnit,
+                            kAudioUnitProperty_MaximumFramesPerSlice,
+                            kAudioUnitScope_Input,
+                            OUTPUT_ELEMENT,
+                            actualOutputFramesPerBuffer,
+                            sizeof(unsigned long) ) );
+       ERR_WRAP( AudioUnitGetProperty( *audioUnit,
+                            kAudioUnitProperty_MaximumFramesPerSlice,
+                            kAudioUnitScope_Global,
+                            OUTPUT_ELEMENT,
+                            actualOutputFramesPerBuffer,
+                            &size ) );
+    }
+    if( inStreamParams ) {
+       /*UInt32 size = sizeof( *actualInputFramesPerBuffer );*/
+       ERR_WRAP( AudioUnitSetProperty( *audioUnit,
+                            kAudioUnitProperty_MaximumFramesPerSlice,
+                            kAudioUnitScope_Output,
+                            INPUT_ELEMENT,
+                            actualInputFramesPerBuffer,
+                            sizeof(unsigned long) ) );
+/* Don't know why this causes problems
+       ERR_WRAP( AudioUnitGetProperty( *audioUnit,
+                            kAudioUnitProperty_MaximumFramesPerSlice,
+                            kAudioUnitScope_Global, //Output,
+                            INPUT_ELEMENT,
+                            actualInputFramesPerBuffer,
+                            &size ) );
+*/
     }
 
-    return result;
+    /* -- if we have input, we may need to setup an SR converter -- */
+    /* even if we got the sample rate we asked for, we need to do
+       the conversion in case another program changes the underlying SR. */
+    /* FIXME: I think we need to monitor stream and change the converter if the incoming format changes. */
+    if( inStreamParams ) {
+       AudioStreamBasicDescription desiredFormat;
+       AudioStreamBasicDescription sourceFormat;
+       UInt32 sourceSize = sizeof( sourceFormat );
+       bzero( &desiredFormat, sizeof(desiredFormat) );
+       desiredFormat.mSampleRate       = sampleRate;
+       desiredFormat.mFormatID         = kAudioFormatLinearPCM ;
+       desiredFormat.mFormatFlags      = kAudioFormatFlagsNativeFloatPacked;
+       desiredFormat.mFramesPerPacket  = 1;
+       desiredFormat.mBitsPerChannel   = sizeof( float ) * 8;
+       desiredFormat.mBytesPerPacket=sizeof(float)*inStreamParams->channelCount;
+       desiredFormat.mBytesPerFrame =sizeof(float)*inStreamParams->channelCount;
+       desiredFormat.mChannelsPerFrame = inStreamParams->channelCount;
+
+       /* get the source format */
+       ERR_WRAP( AudioUnitGetProperty(
+                         *audioUnit,
+                         kAudioUnitProperty_StreamFormat,
+                         kAudioUnitScope_Output,
+                         INPUT_ELEMENT,
+                         &sourceFormat,
+                         &sourceSize ) );
+
+       if( desiredFormat.mSampleRate != sourceFormat.mSampleRate )
+       {
+          UInt32 value = kAudioConverterQuality_Max;
+          switch( macInputStreamFlags & 0x0700 ) {
+          case 0x0100: /*paMacCore_ConversionQualityMin:*/
+             value=kAudioConverterQuality_Min;
+             break;
+          case 0x0200: /*paMacCore_ConversionQualityLow:*/
+             value=kAudioConverterQuality_Low;
+             break;
+          case 0x0300: /*paMacCore_ConversionQualityMedium:*/
+             value=kAudioConverterQuality_Medium;
+             break;
+          case 0x0400: /*paMacCore_ConversionQualityHigh:*/
+             value=kAudioConverterQuality_High;
+             break;
+          }
+          VDBUG(( "Creating sample rate converter for input"
+                  " to convert from %g to %g\n",
+                  (float)sourceFormat.mSampleRate,
+                  (float)desiredFormat.mSampleRate ) );
+          /* create our converter */
+          ERR_WRAP( AudioConverterNew( 
+                             &sourceFormat,
+                             &desiredFormat,
+                             srConverter ) );
+          /* Set quality */
+          ERR_WRAP( AudioConverterSetProperty(
+                             *srConverter,
+                             kAudioConverterSampleRateConverterQuality,
+                             sizeof( value ),
+                             &value ) );
+       }
+    }
+    /* -- set IOProc (callback) -- */
+    callbackKey = outStreamParams ? kAudioUnitProperty_SetRenderCallback
+                                  : kAudioOutputUnitProperty_SetInputCallback ;
+    rcbs.inputProc = AudioIOProc;
+    rcbs.inputProcRefCon = refCon;
+    ERR_WRAP( AudioUnitSetProperty(
+                               *audioUnit,
+                               callbackKey,
+                               kAudioUnitScope_Output,
+                               outStreamParams ? OUTPUT_ELEMENT : INPUT_ELEMENT,
+                               &rcbs,
+                               sizeof(rcbs)) );
+
+    if( inStreamParams && outStreamParams && *srConverter )
+           ERR_WRAP( AudioUnitSetProperty(
+                               *audioUnit,
+                               kAudioOutputUnitProperty_SetInputCallback,
+                               kAudioUnitScope_Output,
+                               INPUT_ELEMENT,
+                               &rcbs,
+                               sizeof(rcbs)) );
+
+    /*IMPLEMENTME: may need to worry about channel mapping.*/
+
+    /* initialize the audio unit */
+    ERR_WRAP( AudioUnitInitialize(*audioUnit) );
+
+    if( inStreamParams && outStreamParams )
+       VDBUG( ("Opened device %ld for input and output.\n", *audioDevice ) );
+    else if( inStreamParams )
+       VDBUG( ("Opened device %ld for input.\n", *audioDevice ) );
+    else if( outStreamParams )
+       VDBUG( ("Opened device %ld for output.\n", *audioDevice ) );
+    return paNoError;
+#undef ERR_WRAP
+
+    error:
+       CloseComponent( *audioUnit );
+       *audioUnit = NULL;
+       if( result )
+          return PaMacCore_SetError( result, line, 1 );
+       return paResult;
 }
 
+/* see pa_hostapi.h for a list of validity guarantees made about OpenStream parameters */
 static PaError OpenStream( struct PaUtilHostApiRepresentation *hostApi,
                            PaStream** s,
                            const PaStreamParameters *inputParameters,
@@ -638,264 +998,1045 @@ static PaError OpenStream( struct PaUtilHostApiRepresentation *hostApi,
                            PaStreamCallback *streamCallback,
                            void *userData )
 {
-    PaError err = paNoError;
-    PaMacCoreHostApiRepresentation *macCoreHostApi = (PaMacCoreHostApiRepresentation *)hostApi;
-    PaMacCoreStream *stream = PaUtil_AllocateMemory(sizeof(PaMacCoreStream));
-    stream->isActive = 0;
-    stream->isStopped = 1;
-    stream->inputDevice = kAudioDeviceUnknown;
-    stream->outputDevice = kAudioDeviceUnknown;
-    
-    PaUtil_InitializeStreamRepresentation( &stream->streamRepresentation,
-                                           ( (streamCallback)
-                                             ? &macCoreHostApi->callbackStreamInterface
-                                             : &macCoreHostApi->blockingStreamInterface ),
-                                           streamCallback, userData );
-    PaUtil_InitializeCpuLoadMeasurer( &stream->cpuLoadMeasurer, sampleRate );
-    
-    *s = (PaStream*)stream;
-    PaMacClientData *clientData = PaUtil_AllocateMemory(sizeof(PaMacClientData));
-    clientData->stream = stream;
-    clientData->callback = streamCallback;
-    clientData->userData = userData;
-    clientData->inputBuffer = 0;
-    clientData->outputBuffer = 0;
-    clientData->ditherGenerator = PaUtil_AllocateMemory(sizeof(PaUtilTriangularDitherGenerator));
-    PaUtil_InitializeTriangularDitherState(clientData->ditherGenerator);
-    
-    if (inputParameters != NULL) {
-        stream->inputDevice = macCoreHostApi->macCoreDeviceIds[inputParameters->device];
-        clientData->inputConverter = PaUtil_SelectConverter(paFloat32, inputParameters->sampleFormat, streamFlags);
-        clientData->inputBuffer = PaUtil_AllocateMemory(Pa_GetSampleSize(inputParameters->sampleFormat) * framesPerBuffer * inputParameters->channelCount);
-        clientData->inputChannelCount = inputParameters->channelCount;
-        clientData->inputSampleFormat = inputParameters->sampleFormat;
-        err = SetUpUnidirectionalStream(stream->inputDevice, sampleRate, framesPerBuffer, 1);
-    }
-    
-    if (err == paNoError && outputParameters != NULL) {
-        stream->outputDevice = macCoreHostApi->macCoreDeviceIds[outputParameters->device];
-        clientData->outputConverter = PaUtil_SelectConverter(outputParameters->sampleFormat, paFloat32, streamFlags);
-        clientData->outputBuffer = PaUtil_AllocateMemory(Pa_GetSampleSize(outputParameters->sampleFormat) * framesPerBuffer * outputParameters->channelCount);
-        clientData->outputChannelCount = outputParameters->channelCount;
-        clientData->outputSampleFormat = outputParameters->sampleFormat;
-        err = SetUpUnidirectionalStream(stream->outputDevice, sampleRate, framesPerBuffer, 0);
-    }
-
-    if (inputParameters == NULL || outputParameters == NULL || stream->inputDevice == stream->outputDevice) {
-        AudioDeviceID device = (inputParameters == NULL) ? stream->outputDevice : stream->inputDevice;
-
-        AudioDeviceAddIOProc(device, AudioIOProc, clientData);
-    }
-    else {
-        // using different devices for input and output
-        AudioDeviceAddIOProc(stream->inputDevice, AudioInputProc, clientData);
-        AudioDeviceAddIOProc(stream->outputDevice, AudioOutputProc, clientData);
-    }
-    
-    return err;
-}
+    PaError result = paNoError;
+    PaMacAUHAL *auhalHostApi = (PaMacAUHAL*)hostApi;
+    PaMacCoreStream *stream = 0;
+    int inputChannelCount, outputChannelCount;
+    PaSampleFormat inputSampleFormat, outputSampleFormat;
+    PaSampleFormat hostInputSampleFormat, hostOutputSampleFormat;
+    VVDBUG(("OpenStream(): in chan=%d, in fmt=%ld, out chan=%d, out fmt=%ld SR=%g, FPB=%ld\n",
+                inputParameters  ? inputParameters->channelCount  : -1,
+                inputParameters  ? inputParameters->sampleFormat  : -1,
+                outputParameters ? outputParameters->channelCount : -1,
+                outputParameters ? outputParameters->sampleFormat : -1,
+                (float) sampleRate,
+                framesPerBuffer ));
+    VDBUG( ("Opening Stream.\n") );
+
+    /*These first few bits of code are from paSkeleton with few modifications.*/
+    if( inputParameters )
+    {
+        inputChannelCount = inputParameters->channelCount;
+        inputSampleFormat = inputParameters->sampleFormat;
 
-// Note: When CloseStream() is called, the multi-api layer ensures that the stream has already been stopped or aborted.
-static PaError CloseStream( PaStream* s )
-{
-    PaError err = paNoError;
-    PaMacCoreStream *stream = (PaMacCoreStream*)s;
+        /* unless alternate device specification is supported, reject the use of
+            paUseHostApiSpecificDeviceSpecification */
 
-    PaUtil_ResetCpuLoadMeasurer( &stream->cpuLoadMeasurer );
+        if( inputParameters->device == paUseHostApiSpecificDeviceSpecification )
+            return paInvalidDevice;
 
-    if (stream->inputDevice != kAudioDeviceUnknown) {
-        if (stream->outputDevice == kAudioDeviceUnknown || stream->outputDevice == stream->inputDevice) {
-            err = conv_err(AudioDeviceRemoveIOProc(stream->inputDevice, AudioIOProc));
-        }
-        else {
-            err = conv_err(AudioDeviceRemoveIOProc(stream->inputDevice, AudioInputProc));
-            err = conv_err(AudioDeviceRemoveIOProc(stream->outputDevice, AudioOutputProc));
-        }
+        /* check that input device can support inputChannelCount */
+        if( inputChannelCount > hostApi->deviceInfos[ inputParameters->device ]->maxInputChannels )
+            return paInvalidChannelCount;
+
+        /* Host supports interleaved float32 */
+        hostInputSampleFormat = paFloat32;
     }
-    else {
-        err = conv_err(AudioDeviceRemoveIOProc(stream->outputDevice, AudioIOProc));
+    else
+    {
+        inputChannelCount = 0;
+        inputSampleFormat = hostInputSampleFormat = paFloat32; /* Surpress 'uninitialised var' warnings. */
     }
-    
-    return err;
-}
 
+    if( outputParameters )
+    {
+        outputChannelCount = outputParameters->channelCount;
+        outputSampleFormat = outputParameters->sampleFormat;
+        
+        /* unless alternate device specification is supported, reject the use of
+            paUseHostApiSpecificDeviceSpecification */
 
-static PaError StartStream( PaStream *s )
-{
-    PaError err = paNoError;
-    PaMacCoreStream *stream = (PaMacCoreStream*)s;
+        if( outputParameters->device == paUseHostApiSpecificDeviceSpecification )
+            return paInvalidDevice;
 
-    if (stream->inputDevice != kAudioDeviceUnknown) {
-        if (stream->outputDevice == kAudioDeviceUnknown || stream->outputDevice == stream->inputDevice) {
-            err = conv_err(AudioDeviceStart(stream->inputDevice, AudioIOProc));
-        }
-        else {
-            err = conv_err(AudioDeviceStart(stream->inputDevice, AudioInputProc));
-            err = conv_err(AudioDeviceStart(stream->outputDevice, AudioOutputProc));
-        }
+        /* check that output device can support inputChannelCount */
+        if( outputChannelCount > hostApi->deviceInfos[ outputParameters->device ]->maxOutputChannels )
+            return paInvalidChannelCount;
+
+        /* Host supports interleaved float32 */
+        hostOutputSampleFormat = paFloat32;
+    }
+    else
+    {
+        outputChannelCount = 0;
+        outputSampleFormat = hostOutputSampleFormat = paFloat32; /* Surpress 'uninitialized var' warnings. */
     }
-    else {
-        err = conv_err(AudioDeviceStart(stream->outputDevice, AudioIOProc));
+
+    /* validate platform specific flags */
+    if( (streamFlags & paPlatformSpecificFlags) != 0 )
+        return paInvalidFlag; /* unexpected platform specific flag */
+
+    stream = (PaMacCoreStream*)PaUtil_AllocateMemory( sizeof(PaMacCoreStream) );
+    if( !stream )
+    {
+        result = paInsufficientMemory;
+        goto error;
     }
-    
-    stream->isActive = 1;
-    stream->isStopped = 0;
-    return err;
+
+    /* If we fail after this point, we my be left in a bad state, with
+       some data structures setup and others not. So, first thing we
+       do is initialize everything so that if we fail, we know what hasn't
+       been touched.
+     */
+
+    stream->inputAudioBufferList.mBuffers[0].mData = NULL;
+    stream->inputRingBuffer.buffer = NULL;
+    bzero( &stream->blio, sizeof( PaMacBlio ) );
+/*
+    stream->blio.inputRingBuffer.buffer = NULL;
+    stream->blio.outputRingBuffer.buffer = NULL;
+    stream->blio.inputSampleFormat = inputParameters?inputParameters->sampleFormat:0;
+    stream->blio.inputSampleSize = computeSampleSizeFromFormat(stream->blio.inputSampleFormat);
+    stream->blio.outputSampleFormat=outputParameters?outputParameters->sampleFormat:0;
+    stream->blio.outputSampleSize = computeSampleSizeFromFormat(stream->blio.outputSampleFormat);
+*/
+    stream->inputSRConverter = NULL;
+    stream->inputUnit = NULL;
+    stream->outputUnit = NULL;
+    stream->inputFramesPerBuffer = 0;
+    stream->outputFramesPerBuffer = 0;
+    stream->bufferProcessorIsInitialized = FALSE;
+
+    /* assert( streamCallback ) ; */ /* only callback mode is implemented */
+    if( streamCallback )
+    {
+        PaUtil_InitializeStreamRepresentation( &stream->streamRepresentation,
+                                        &auhalHostApi->callbackStreamInterface,
+                                        streamCallback, userData );
+    }
+    else
+    {
+        PaUtil_InitializeStreamRepresentation( &stream->streamRepresentation,
+                                        &auhalHostApi->blockingStreamInterface,
+					BlioCallback, &stream->blio );
+    }
+
+    PaUtil_InitializeCpuLoadMeasurer( &stream->cpuLoadMeasurer, sampleRate );
+
+    /* -- handle paFramesPerBufferUnspecified -- */
+    if( framesPerBuffer == paFramesPerBufferUnspecified ) {
+       long requested = 64;
+       if( inputParameters )
+          requested = MAX( requested, inputParameters->suggestedLatency * sampleRate / 2 );
+       if( outputParameters )
+          requested = MAX( requested, outputParameters->suggestedLatency *sampleRate / 2 );
+       VDBUG( ("Block Size unspecified. Based on Latency, the user wants a Block Size near: %ld.\n",
+              requested ) );
+       if( requested <= 64 ) {
+          /*requested a realtively low latency. make sure this is in range of devices */
+          /*try to get the device's min natural buffer size and use that (but no smaller than 64).*/
+          AudioValueRange audioRange;
+          size_t size = sizeof( audioRange );
+          if( inputParameters ) {
+             WARNING( result = AudioDeviceGetProperty( auhalHostApi->devIds[inputParameters->device],
+                                          0,
+                                          false,
+                                          kAudioDevicePropertyBufferFrameSizeRange,
+                                          &size, &audioRange ) );
+             if( result )
+                requested = MAX( requested, audioRange.mMinimum );
+          }
+          if( outputParameters ) {
+             WARNING( result = AudioDeviceGetProperty( auhalHostApi->devIds[outputParameters->device],
+                                          0,
+                                          false,
+                                          kAudioDevicePropertyBufferFrameSizeRange,
+                                          &size, &audioRange ) );
+             if( result )
+                requested = MAX( requested, audioRange.mMinimum );
+          }
+       } else {
+          /* requested a realtively high latency. make sure this is in range of devices */
+          /*try to get the device's max natural buffer size and use that (but no larger than 1024).*/
+          AudioValueRange audioRange;
+          size_t size = sizeof( audioRange );
+          requested = MIN( requested, 1024 );
+          if( inputParameters ) {
+             WARNING( result = AudioDeviceGetProperty( auhalHostApi->devIds[inputParameters->device],
+                                          0,
+                                          false,
+                                          kAudioDevicePropertyBufferFrameSizeRange,
+                                          &size, &audioRange ) );
+             if( result )
+                requested = MIN( requested, audioRange.mMaximum );
+          }
+          if( outputParameters ) {
+             WARNING( result = AudioDeviceGetProperty( auhalHostApi->devIds[outputParameters->device],
+                                          0,
+                                          false,
+                                          kAudioDevicePropertyBufferFrameSizeRange,
+                                          &size, &audioRange ) );
+             if( result )
+                requested = MIN( requested, audioRange.mMaximum );
+          }
+       }
+       /* -- double check ranges -- */
+       if( requested > 1024 ) requested = 1024;
+       if( requested < 64 ) requested = 64;
+       VDBUG(("After querying hardware, setting block size to %ld.\n", requested));
+       framesPerBuffer = requested;
+    }
+
+    /* -- Now we actually open and setup streams. -- */
+    if( inputParameters && outputParameters && outputParameters->device == inputParameters->device )
+    { /* full duplex. One device. */
+       result = OpenAndSetupOneAudioUnit( inputParameters,
+                                          outputParameters,
+                                          framesPerBuffer,
+                                          &(stream->inputFramesPerBuffer),
+                                          &(stream->outputFramesPerBuffer),
+                                          auhalHostApi,
+                                          &(stream->inputUnit),
+                                          &(stream->inputSRConverter),
+                                          &(stream->inputDevice),
+                                          sampleRate,
+                                          stream );
+       stream->outputUnit = stream->inputUnit;
+       stream->outputDevice = stream->inputDevice;
+       if( result != paNoError )
+           goto error;
+    }
+    else
+    { /* full duplex, different devices OR simplex */
+       result = OpenAndSetupOneAudioUnit( NULL,
+                                          outputParameters,
+                                          framesPerBuffer,
+                                          NULL,
+                                          &(stream->outputFramesPerBuffer),
+                                          auhalHostApi,
+                                          &(stream->outputUnit),
+                                          NULL,
+                                          &(stream->outputDevice),
+                                          sampleRate,
+                                          stream );
+       if( result != paNoError )
+           goto error;
+       result = OpenAndSetupOneAudioUnit( inputParameters,
+                                          NULL,
+                                          framesPerBuffer,
+                                          &(stream->inputFramesPerBuffer),
+                                          NULL,
+                                          auhalHostApi,
+                                          &(stream->inputUnit),
+                                          &(stream->inputSRConverter),
+                                          &(stream->inputDevice),
+                                          sampleRate,
+                                          stream );
+       if( result != paNoError )
+           goto error;
+    }
+
+    if( stream->inputUnit ) {
+       const size_t szfl = sizeof(float);
+       /* setup the AudioBufferList used for input */
+       bzero( &stream->inputAudioBufferList, sizeof( AudioBufferList ) );
+       stream->inputAudioBufferList.mNumberBuffers = 1;
+       stream->inputAudioBufferList.mBuffers[0].mNumberChannels
+                 = inputChannelCount;
+       stream->inputAudioBufferList.mBuffers[0].mDataByteSize
+                 = stream->inputFramesPerBuffer*inputChannelCount*szfl;
+       stream->inputAudioBufferList.mBuffers[0].mData
+                 = (float *) calloc(
+                               stream->inputFramesPerBuffer*inputChannelCount,
+                               szfl );
+       if( !stream->inputAudioBufferList.mBuffers[0].mData )
+       {
+          result = paInsufficientMemory;
+          goto error;
+       }
+        
+       /*
+        * If input and output devs are different or we are doing SR conversion,
+        * we also need a
+        * ring buffer to store inpt data while waiting for output
+        * data.
+        */
+       if( (stream->outputUnit && stream->inputUnit != stream->outputUnit)
+           || stream->inputSRConverter )
+       {
+          /* May want the ringSize ot initial position in
+             ring buffer to depend somewhat on sample rate change */
+
+          void *data;
+          long ringSize;
+
+          ringSize = computeRingBufferSize( inputParameters,
+                                            outputParameters,
+                                            stream->inputFramesPerBuffer,
+                                            stream->outputFramesPerBuffer,
+                                            sampleRate );
+          /*ringSize <<= 4; *//*16x bigger, for testing */
+
+
+          /*now, we need to allocate memory for the ring buffer*/
+          data = calloc( ringSize, szfl );
+          if( !data )
+          {
+             result = paInsufficientMemory;
+             goto error;
+          }
+
+          /* now we can initialize the ring buffer */
+          assert( 0 ==
+            RingBuffer_Init( &stream->inputRingBuffer,
+                             ringSize*szfl, data ) );
+          /* advance the read point a little, so we are reading from the
+             middle of the buffer */
+          if( stream->outputUnit )
+             RingBuffer_AdvanceWriteIndex( &stream->inputRingBuffer, ringSize*szfl / RING_BUFFER_ADVANCE_DENOMINATOR );
+       }
+    }
+
+    /* -- initialize Blio Buffer Processors -- */
+    if( !streamCallback )
+    {
+       long ringSize;
+
+       ringSize = computeRingBufferSize( inputParameters,
+                                         outputParameters,
+                                         stream->inputFramesPerBuffer,
+                                         stream->outputFramesPerBuffer,
+                                         sampleRate );
+       result = initializeBlioRingBuffers( &stream->blio,
+              inputParameters?inputParameters->sampleFormat:0 ,
+              outputParameters?outputParameters->sampleFormat:0 ,
+              MAX(stream->inputFramesPerBuffer,stream->outputFramesPerBuffer),
+              ringSize,
+              inputParameters?inputChannelCount:0 ,
+              outputParameters?outputChannelCount:0 ) ;
+       if( result != paNoError )
+          goto error;
+    }
+
+    /* -- initialize Buffer Processor -- */
+    {
+       unsigned long maxHostFrames = stream->inputFramesPerBuffer;
+       if( stream->outputFramesPerBuffer > maxHostFrames )
+          maxHostFrames = stream->outputFramesPerBuffer;
+       result = PaUtil_InitializeBufferProcessor( &stream->bufferProcessor,
+                 inputChannelCount, inputSampleFormat,
+                 hostInputSampleFormat,
+                 outputChannelCount, outputSampleFormat,
+                 hostOutputSampleFormat,
+                 sampleRate,
+                 streamFlags,
+                 framesPerBuffer,
+                 /* If sample rate conversion takes place, the buffer size
+                    will not be known. */
+                 maxHostFrames,
+                 stream->inputSRConverter
+                              ? paUtilUnknownHostBufferSize
+                              : paUtilBoundedHostBufferSize,
+                 streamCallback ? streamCallback : BlioCallback,
+                 streamCallback ? userData : &stream->blio );
+       if( result != paNoError )
+           goto error;
+    }
+    stream->bufferProcessorIsInitialized = TRUE;
+
+    /*
+        IMPLEMENT ME: initialise the following fields with estimated or actual
+        values.
+        I think this is okay the way it is br 12/1/05
+        maybe need to change input latency estimate if IO devs differ
+    */
+    stream->streamRepresentation.streamInfo.inputLatency =
+            PaUtil_GetBufferProcessorInputLatency(&stream->bufferProcessor)/sampleRate;
+    stream->streamRepresentation.streamInfo.outputLatency =
+            PaUtil_GetBufferProcessorOutputLatency(&stream->bufferProcessor)/sampleRate;
+    stream->streamRepresentation.streamInfo.sampleRate = sampleRate;
+
+    stream->sampleRate  = sampleRate;
+    stream->outDeviceSampleRate = 0;
+    if( stream->outputUnit ) {
+       Float64 rate;
+       UInt32 size = sizeof( rate );
+       result = ERR( AudioDeviceGetProperty( stream->outputDevice,
+                                    0,
+                                    FALSE,
+                                    kAudioDevicePropertyNominalSampleRate,
+                                    &size, &rate ) );
+       if( result )
+          goto error;
+       stream->outDeviceSampleRate = rate;
+    }
+    stream->inDeviceSampleRate = 0;
+    if( stream->inputUnit ) {
+       Float64 rate;
+       UInt32 size = sizeof( rate );
+       result = ERR( AudioDeviceGetProperty( stream->inputDevice,
+                                    0,
+                                    TRUE,
+                                    kAudioDevicePropertyNominalSampleRate,
+                                    &size, &rate ) );
+       if( result )
+          goto error;
+       stream->inDeviceSampleRate = rate;
+    }
+    stream->userInChan  = inputChannelCount;
+    stream->userOutChan = outputChannelCount;
+
+    stream->isTimeSet   = FALSE;
+    stream->state = STOPPED;
+    stream->xrunFlags = 0;
+
+    *s = (PaStream*)stream;
+
+    return result;
+
+error:
+    CloseStream( stream );
+    return result;
 }
 
-static PaError AbortStream( PaStream *s )
+PaTime GetStreamTime( PaStream *s )
 {
-    PaError err = paNoError;
+   /* FIXME: I am not at all sure this timing info stuff is right.
+             patest_sine_time reports negative latencies, which is wierd.*/
     PaMacCoreStream *stream = (PaMacCoreStream*)s;
-    
-    if (stream->inputDevice != kAudioDeviceUnknown) {
-        if (stream->outputDevice == kAudioDeviceUnknown || stream->outputDevice == stream->inputDevice) {
-            err = conv_err(AudioDeviceStop(stream->inputDevice, AudioIOProc));
-        }
-        else {
-            err = conv_err(AudioDeviceStop(stream->inputDevice, AudioInputProc));
-            err = conv_err(AudioDeviceStop(stream->outputDevice, AudioOutputProc));
-        }
-    }
-    else {
-        err = conv_err(AudioDeviceStop(stream->outputDevice, AudioIOProc));
+    AudioTimeStamp timeStamp;
+
+    VVDBUG(("GetStreamTime()\n"));
+
+    if ( !stream->isTimeSet )
+        return (PaTime)0;
+
+    if ( stream->outputDevice ) {
+        AudioDeviceGetCurrentTime( stream->outputDevice, &timeStamp);
+        return (PaTime)(timeStamp.mSampleTime - stream->startTime.mSampleTime)/stream->outDeviceSampleRate;
+    } else if ( stream->inputDevice ) {
+        AudioDeviceGetCurrentTime( stream->inputDevice, &timeStamp);
+    return (PaTime)(timeStamp.mSampleTime - stream->startTime.mSampleTime)/stream->inDeviceSampleRate;
+    } else {
+        return (PaTime)0;
     }
-    
-    stream->isActive = 0;
-    stream->isStopped = 1;
-    return err;
-}    
+}
 
-static PaError StopStream( PaStream *s )
+static void setStreamStartTime( PaStream *stream )
 {
-    // TODO: this should be nicer than abort
-    return AbortStream(s);
+   /* FIXME: I am not at all sure this timing info stuff is right.
+             patest_sine_time reports negative latencies, which is wierd.*/
+   PaMacCoreStream *s = (PaMacCoreStream *) stream;
+   VVDBUG(("setStreamStartTime()\n"));
+   if( s->outputDevice )
+      AudioDeviceGetCurrentTime( s->outputDevice, &s->startTime);
+   else if( s->inputDevice )
+      AudioDeviceGetCurrentTime( s->inputDevice, &s->startTime);
+   else
+      bzero( &s->startTime, sizeof( s->startTime ) );
+
+   //FIXME: we need a memory barier here
+
+   s->isTimeSet = TRUE;
 }
 
-static PaError IsStreamStopped( PaStream *s )
+
+static PaTime TimeStampToSecs(PaMacCoreStream *stream, const AudioTimeStamp* timeStamp)
 {
-    PaMacCoreStream *stream = (PaMacCoreStream*)s;
-    
-    return stream->isStopped;
+    VVDBUG(("TimeStampToSecs()\n"));
+    //printf( "ATS: %lu, %g, %g\n", timeStamp->mFlags, timeStamp->mSampleTime, timeStamp->mRateScalar );
+    if (timeStamp->mFlags & kAudioTimeStampSampleTimeValid)
+        return (timeStamp->mSampleTime / stream->sampleRate);
+    else
+        return 0;
 }
 
+#define RING_BUFFER_EMPTY (1000)
 
-static PaError IsStreamActive( PaStream *s )
+static OSStatus ringBufferIOProc( AudioConverterRef inAudioConverter, 
+                             UInt32*ioDataSize, 
+                             void** outData, 
+                             void*inUserData )
 {
-    PaMacCoreStream *stream = (PaMacCoreStream*)s;
+   void *dummyData;
+   long dummySize;
+   RingBuffer *rb = (RingBuffer *) inUserData;
+
+   VVDBUG(("ringBufferIOProc()\n"));
+
+   assert( sizeof( UInt32 ) == sizeof( long ) );
+   if( RingBuffer_GetReadAvailable( rb ) == 0 ) {
+      *outData = NULL;
+      *ioDataSize = 0;
+      return RING_BUFFER_EMPTY;
+   }
+   RingBuffer_GetReadRegions( rb, *ioDataSize,
+                              outData, (long *)ioDataSize, 
+                              &dummyData, &dummySize );
+      
+   assert( *ioDataSize );
+   RingBuffer_AdvanceReadIndex( rb, *ioDataSize );
+
+   return noErr;
+}
 
-    return stream->isActive;
+/*
+ * Called by the AudioUnit API to process audio from the sound card.
+ * This is where the magic happens.
+ */
+/* FEEDBACK: there is a lot of redundant code here because of how all the cases differ. This makes it hard to maintain, so if there are suggestinos for cleaning it up, I'm all ears. */
+static OSStatus AudioIOProc( void *inRefCon,
+                               AudioUnitRenderActionFlags *ioActionFlags,
+                               const AudioTimeStamp *inTimeStamp,
+                               UInt32 inBusNumber,
+                               UInt32 inNumberFrames,
+                               AudioBufferList *ioData )
+{
+   unsigned long framesProcessed     = 0;
+   PaStreamCallbackTimeInfo timeInfo = {0,0,0};
+   PaMacCoreStream *stream           = (PaMacCoreStream*)inRefCon;
+   const bool isRender               = inBusNumber == OUTPUT_ELEMENT;
+   int callbackResult                = paContinue ;
+
+   VVDBUG(("AudioIOProc()\n"));
+
+   PaUtil_BeginCpuLoadMeasurement( &stream->cpuLoadMeasurer );
+
+   /* -----------------------------------------------------------------*\
+      This output may be useful for debugging,
+      But printing durring the callback is a bad enough idea that
+      this is not enabled by enableing the usual debugging calls.
+   \* -----------------------------------------------------------------*/
+   /*
+   static int renderCount = 0;
+   static int inputCount = 0;
+   printf( "-------------------  starting reder/input\n" );
+   if( isRender )
+      printf("Render callback (%d):\t", ++renderCount);
+   else
+      printf("Input callback  (%d):\t", ++inputCount);
+   printf( "Call totals: %d (input), %d (render)\n", inputCount, renderCount );
+
+   printf( "--- inBusNumber: %lu\n", inBusNumber );
+   printf( "--- inNumberFrames: %lu\n", inNumberFrames );
+   printf( "--- %x ioData\n", (unsigned) ioData );
+   if( ioData )
+   {
+      int i=0;
+      printf( "--- ioData.mNumBuffers %lu: \n", ioData->mNumberBuffers );
+      for( i=0; i<ioData->mNumberBuffers; ++i )
+         printf( "--- ioData buffer %d size: %lu.\n", i, ioData->mBuffers[i].mDataByteSize );
+   }
+      ----------------------------------------------------------------- */
+
+   if( !stream->isTimeSet )
+      setStreamStartTime( stream );
+
+   if( isRender ) {
+      AudioTimeStamp currentTime;
+      timeInfo.outputBufferDacTime = TimeStampToSecs(stream, inTimeStamp);
+      AudioDeviceGetCurrentTime(stream->outputDevice, &currentTime);
+      timeInfo.currentTime = TimeStampToSecs(stream, &currentTime);
+   }
+   if( isRender && stream->inputUnit == stream->outputUnit )
+      timeInfo.inputBufferAdcTime = TimeStampToSecs(stream, inTimeStamp);
+   if( !isRender ) {
+      AudioTimeStamp currentTime;
+      timeInfo.inputBufferAdcTime = TimeStampToSecs(stream, inTimeStamp);
+      AudioDeviceGetCurrentTime(stream->inputDevice, &currentTime);
+      timeInfo.currentTime = TimeStampToSecs(stream, &currentTime);
+   }
+
+   //printf( "---%g, %g, %g\n", timeInfo.inputBufferAdcTime, timeInfo.currentTime, timeInfo.outputBufferDacTime );
+
+   if( isRender && stream->inputUnit == stream->outputUnit
+                && !stream->inputSRConverter )
+   {
+      /* --------- Full Duplex, One Device, no SR Conversion -------
+       *
+       * This is the lowest latency case, and also the simplest.
+       * Input data and output data are available at the same time.
+       * we do not use the input SR converter or the input ring buffer.
+       *
+       */
+      OSErr err = 0;
+      unsigned long frames;
+
+      /* -- start processing -- */
+      PaUtil_BeginBufferProcessing( &(stream->bufferProcessor),
+                                    &timeInfo,
+                                    stream->xrunFlags );
+      stream->xrunFlags = 0;
+
+      /* -- compute frames. do some checks -- */
+      assert( ioData->mNumberBuffers == 1 );
+      assert( ioData->mBuffers[0].mNumberChannels == stream->userOutChan );
+      frames = ioData->mBuffers[0].mDataByteSize;
+      frames /= sizeof( float ) * ioData->mBuffers[0].mNumberChannels;
+      /* -- copy and process input data -- */
+      err= AudioUnitRender(stream->inputUnit,
+                    ioActionFlags,
+                    inTimeStamp,
+                    INPUT_ELEMENT,
+                    inNumberFrames,
+                    &stream->inputAudioBufferList );
+      /* FEEDBACK: I'm not sure what to do when this call fails */
+      assert( !err );
+
+      PaUtil_SetInputFrameCount( &(stream->bufferProcessor), frames );
+      PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor),
+                          0,
+                          stream->inputAudioBufferList.mBuffers[0].mData,
+                          stream->inputAudioBufferList.mBuffers[0].mNumberChannels);
+      /* -- Copy and process output data -- */
+      PaUtil_SetOutputFrameCount( &(stream->bufferProcessor), frames );
+      PaUtil_SetInterleavedOutputChannels( &(stream->bufferProcessor),
+                                        0,
+                                        ioData->mBuffers[0].mData,
+                                        ioData->mBuffers[0].mNumberChannels);
+      /* -- complete processing -- */
+      framesProcessed =
+                 PaUtil_EndBufferProcessing( &(stream->bufferProcessor),
+                                             &callbackResult );
+   }
+   else if( isRender )
+   {
+      /* -------- Output Side of Full Duplex (Separate Devices or SR Conversion)
+       *       -- OR Simplex Output
+       *
+       * This case handles output data as in the full duplex case,
+       * and, if there is input data, reads it off the ring buffer 
+       * and into the PA buffer processor. If sample rate conversion
+       * is required on input, that is done here as well.
+       */
+      unsigned long frames;
+
+      /* Sometimes, when stopping a duplex stream we get erroneous
+         xrun flags, so if this is our last run, clear the flags. */
+      int xrunFlags = stream->xrunFlags;
+/*
+      if( xrunFlags & paInputUnderflow )
+         printf( "input underflow.\n" );
+      if( xrunFlags & paInputOverflow )
+         printf( "input overflow.\n" );
+*/
+      if( stream->state == STOPPING || stream->state == CALLBACK_STOPPED )
+         xrunFlags = 0;
+
+      /* -- start processing -- */
+      PaUtil_BeginBufferProcessing( &(stream->bufferProcessor),
+                                    &timeInfo,
+                                    xrunFlags );
+      stream->xrunFlags = 0; /* FEEDBACK: we only send flags to Buf Proc once */
+
+      /* -- Copy and process output data -- */
+      assert( ioData->mNumberBuffers == 1 );
+      frames = ioData->mBuffers[0].mDataByteSize;
+      frames /= sizeof( float ) * ioData->mBuffers[0].mNumberChannels;
+      assert( ioData->mBuffers[0].mNumberChannels == stream->userOutChan );
+      PaUtil_SetOutputFrameCount( &(stream->bufferProcessor), frames );
+      PaUtil_SetInterleavedOutputChannels( &(stream->bufferProcessor),
+                                     0,
+                                     ioData->mBuffers[0].mData,
+                                     ioData->mBuffers[0].mNumberChannels);
+
+      /* -- copy and process input data, and complete processing -- */
+      if( stream->inputUnit ) {
+         const int flsz = sizeof( float );
+         /* Here, we read the data out of the ring buffer, through the
+            audio converter. */
+         int inChan = stream->inputAudioBufferList.mBuffers[0].mNumberChannels;
+         if( stream->inputSRConverter )
+         {
+               OSStatus err;
+               UInt32 size;
+               float data[ inChan * frames ];
+               size = sizeof( data );
+               err = AudioConverterFillBuffer( 
+                             stream->inputSRConverter,
+                             ringBufferIOProc,
+                             &stream->inputRingBuffer,
+                             &size,
+                             (void *)&data );
+               if( err == RING_BUFFER_EMPTY )
+               { /*the ring buffer callback underflowed */
+                  err = 0;
+                  bzero( ((char *)data) + size, sizeof(data)-size );
+                  stream->xrunFlags |= paInputUnderflow;
+               }
+               ERR( err );
+               assert( !err );
+               
+               PaUtil_SetInputFrameCount( &(stream->bufferProcessor), frames );
+               PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor),
+                                   0,
+                                   data,
+                                   inChan );
+               framesProcessed =
+                    PaUtil_EndBufferProcessing( &(stream->bufferProcessor),
+                                                &callbackResult );
+         }
+         else
+         {
+            /* Without the AudioConverter is actually a bit more complex
+               because we have to do a little buffer processing that the
+               AudioConverter would otherwise handle for us. */
+            void *data1, *data2;
+            long size1, size2;
+            RingBuffer_GetReadRegions( &stream->inputRingBuffer,
+                                       inChan*frames*flsz,
+                                       &data1, &size1,
+                                       &data2, &size2 );
+            if( size1 / ( flsz * inChan ) == frames ) {
+               /* simplest case: all in first buffer */
+               PaUtil_SetInputFrameCount( &(stream->bufferProcessor), frames );
+               PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor),
+                                   0,
+                                   data1,
+                                   inChan );
+               framesProcessed =
+                    PaUtil_EndBufferProcessing( &(stream->bufferProcessor),
+                                                &callbackResult );
+               RingBuffer_AdvanceReadIndex(&stream->inputRingBuffer, size1 );
+            } else if( ( size1 + size2 ) / ( flsz * inChan ) < frames ) {
+               /*we underflowed. take what data we can, zero the rest.*/
+               float data[frames*inChan];
+               if( size1 )
+                  memcpy( data, data1, size1 );
+               if( size2 )
+                  memcpy( data+size1, data2, size2 );
+               bzero( data+size1+size2, frames*flsz*inChan - size1 - size2 );
+
+               PaUtil_SetInputFrameCount( &(stream->bufferProcessor), frames );
+               PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor),
+                                   0,
+                                   data,
+                                   inChan );
+               framesProcessed =
+                    PaUtil_EndBufferProcessing( &(stream->bufferProcessor),
+                                                &callbackResult );
+               RingBuffer_AdvanceReadIndex( &stream->inputRingBuffer,
+                                            size1+size2 );
+               /* flag underflow */
+               stream->xrunFlags |= paInputUnderflow;
+            } else {
+               /*we got all the data, but split between buffers*/
+               PaUtil_SetInputFrameCount( &(stream->bufferProcessor),
+                                          size1 / ( flsz * inChan ) );
+               PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor),
+                                   0,
+                                   data1,
+                                   inChan );
+               PaUtil_Set2ndInputFrameCount( &(stream->bufferProcessor),
+                                             size2 / ( flsz * inChan ) );
+               PaUtil_Set2ndInterleavedInputChannels( &(stream->bufferProcessor),
+                                   0,
+                                   data2,
+                                   inChan );
+               framesProcessed =
+                    PaUtil_EndBufferProcessing( &(stream->bufferProcessor),
+                                                &callbackResult );
+               RingBuffer_AdvanceReadIndex(&stream->inputRingBuffer, size1+size2 );
+            }
+         }
+      } else {
+         framesProcessed =
+                 PaUtil_EndBufferProcessing( &(stream->bufferProcessor),
+                                             &callbackResult );
+      }
+
+   }
+   else
+   {
+      /* ------------------ Input
+       *
+       * First, we read off the audio data and put it in the ring buffer.
+       * if this is an input-only stream, we need to process it more,
+       * otherwise, we let the output case deal with it.
+       */
+      OSErr err = 0;
+      int chan = stream->inputAudioBufferList.mBuffers[0].mNumberChannels ;
+      /* FIXME: looping here may not actually be necessary, but it was something I tried in testing. */
+      do {
+         err= AudioUnitRender(stream->inputUnit,
+                 ioActionFlags,
+                 inTimeStamp,
+                 INPUT_ELEMENT,
+                 inNumberFrames,
+                 &stream->inputAudioBufferList );
+         if( err == -10874 )
+            inNumberFrames /= 2;
+      } while( err == -10874 && inNumberFrames > 1 );
+      /* FEEDBACK: I'm not sure what to do when this call fails */
+      ERR( err );
+      assert( !err );
+      if( stream->inputSRConverter || stream->outputUnit )
+      {
+         /* If this is duplex or we use a converter, put the data
+            into the ring buffer. */
+         long bytesIn, bytesOut;
+         bytesIn = sizeof( float ) * inNumberFrames * chan;
+         bytesOut = RingBuffer_Write( &stream->inputRingBuffer,
+                                stream->inputAudioBufferList.mBuffers[0].mData,
+                                bytesIn );
+         if( bytesIn != bytesOut )
+            stream->xrunFlags |= paInputOverflow ;
+      }
+      else
+      {
+         /* for simplex input w/o SR conversion,
+            just pop the data into the buffer processor.*/
+         PaUtil_BeginBufferProcessing( &(stream->bufferProcessor),
+                              &timeInfo,
+                              stream->xrunFlags );
+         stream->xrunFlags = 0;
+
+         PaUtil_SetInputFrameCount( &(stream->bufferProcessor), inNumberFrames);
+         PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor),
+                             0,
+                             stream->inputAudioBufferList.mBuffers[0].mData,
+                             chan );
+         framesProcessed =
+              PaUtil_EndBufferProcessing( &(stream->bufferProcessor),
+                                          &callbackResult );
+      }
+      if( !stream->outputUnit && stream->inputSRConverter )
+      {
+         /* ------------------ Simplex Input w/ SR Conversion
+          *
+          * if this is a simplex input stream, we need to read off the buffer,
+          * do our sample rate conversion and pass the results to the buffer
+          * processor.
+          * The logic here is complicated somewhat by the fact that we don't
+          * know how much data is available, so we loop on reasonably sized
+          * chunks, and let the BufferProcessor deal with the rest.
+          *
+          */
+         /*This might be too big or small depending on SR conversion*/
+         float data[ chan * inNumberFrames ];
+         OSStatus err;
+         do
+         { /*Run the buffer processor until we are out of data*/
+            UInt32 size;
+            long f;
+
+            size = sizeof( data );
+            err = AudioConverterFillBuffer( 
+                          stream->inputSRConverter,
+                          ringBufferIOProc,
+                          &stream->inputRingBuffer,
+                          &size,
+                          (void *)data );
+            if( err != RING_BUFFER_EMPTY )
+               ERR( err );
+            assert( err == 0 || err == RING_BUFFER_EMPTY );
+
+            f = size / ( chan * sizeof(float) );
+            PaUtil_SetInputFrameCount( &(stream->bufferProcessor), f );
+            if( f )
+            {
+               PaUtil_BeginBufferProcessing( &(stream->bufferProcessor),
+                                             &timeInfo,
+                                             stream->xrunFlags );
+               stream->xrunFlags = 0;
+
+               PaUtil_SetInterleavedInputChannels( &(stream->bufferProcessor),
+                                0,
+                                data,
+                                chan );
+               framesProcessed =
+                    PaUtil_EndBufferProcessing( &(stream->bufferProcessor),
+                                                &callbackResult );
+            }
+         } while( callbackResult == paContinue && !err );
+      }
+   }
+
+   switch( callbackResult )
+   {
+   case paContinue: break;
+   case paComplete:
+   case paAbort:
+      stream->isTimeSet = FALSE;
+      stream->state = CALLBACK_STOPPED ;
+      if( stream->outputUnit )
+         AudioOutputUnitStop(stream->outputUnit);
+      if( stream->inputUnit )
+         AudioOutputUnitStop(stream->inputUnit);
+      break;
+   }
+
+   PaUtil_EndCpuLoadMeasurement( &stream->cpuLoadMeasurer, framesProcessed );
+   return noErr;
 }
 
 
-static PaTime GetStreamTime( PaStream *s )
+/*
+    When CloseStream() is called, the multi-api layer ensures that
+    the stream has already been stopped or aborted.
+*/
+static PaError CloseStream( PaStream* s )
 {
-    OSStatus err;
-    PaTime result;
+    /* This may be called from a failed OpenStream.
+       Therefore, each piece of info is treated seperately. */
+    PaError result = paNoError;
     PaMacCoreStream *stream = (PaMacCoreStream*)s;
 
-    AudioTimeStamp *timeStamp = PaUtil_AllocateMemory(sizeof(AudioTimeStamp));
-    if (stream->inputDevice != kAudioDeviceUnknown) {
-        err = AudioDeviceGetCurrentTime(stream->inputDevice, timeStamp);
-    }
-    else {
-        err = AudioDeviceGetCurrentTime(stream->outputDevice, timeStamp);
+    VVDBUG(("CloseStream()\n"));
+    VDBUG( ( "Closing stream.\n" ) );
+
+    if( stream ) {
+       if( stream->outputUnit && stream->outputUnit != stream->inputUnit ) {
+          AudioUnitUninitialize( stream->outputUnit );
+          CloseComponent( stream->outputUnit );
+       }
+       stream->outputUnit = NULL;
+       if( stream->inputUnit )
+       {
+          AudioUnitUninitialize( stream->inputUnit );
+          CloseComponent( stream->inputUnit );
+          stream->inputUnit = NULL;
+       }
+       if( stream->inputRingBuffer.buffer )
+          free( (void *) stream->inputRingBuffer.buffer );
+       stream->inputRingBuffer.buffer = NULL;
+       /*TODO: is there more that needs to be done on error
+               from AudioConverterDispose?*/
+       if( stream->inputSRConverter )
+          ERR( AudioConverterDispose( stream->inputSRConverter ) );
+       stream->inputSRConverter = NULL;
+       if( stream->inputAudioBufferList.mBuffers[0].mData )
+          free( stream->inputAudioBufferList.mBuffers[0].mData );
+       stream->inputAudioBufferList.mBuffers[0].mData = NULL;
+
+       result = destroyBlioRingBuffers( &stream->blio );
+       if( result )
+          return result;
+       if( stream->bufferProcessorIsInitialized )
+          PaUtil_TerminateBufferProcessor( &stream->bufferProcessor );
+       PaUtil_TerminateStreamRepresentation( &stream->streamRepresentation );
+       PaUtil_FreeMemory( stream );
     }
-    
-    result = err ? 0 : timeStamp->mSampleTime;
-    PaUtil_FreeMemory(timeStamp);
 
     return result;
 }
 
 
-static double GetStreamCpuLoad( PaStream* s )
+static PaError StartStream( PaStream *s )
 {
     PaMacCoreStream *stream = (PaMacCoreStream*)s;
-    
-    return PaUtil_GetCpuLoad( &stream->cpuLoadMeasurer );
-}
+    OSErr result = noErr;
+    VVDBUG(("StartStream()\n"));
+    VDBUG( ( "Starting stream.\n" ) );
 
+#define ERR_WRAP(mac_err) do { result = mac_err ; if ( result != noErr ) return ERR(result) ; } while(0)
 
-// As separate stream interfaces are used for blocking and callback streams, the following functions can be guaranteed to only be called for blocking streams.
+    /*FIXME: maybe want to do this on close/abort for faster start? */
+    PaUtil_ResetBufferProcessor( &stream->bufferProcessor );
+    if(  stream->inputSRConverter )
+       ERR_WRAP( AudioConverterReset( stream->inputSRConverter ) );
 
-static PaError ReadStream( PaStream* s,
-                           void *buffer,
-                           unsigned long frames )
-{
-    return paInternalError;
+    /* -- start -- */
+    stream->state = ACTIVE;
+    if( stream->inputUnit ) {
+       ERR_WRAP( AudioOutputUnitStart(stream->inputUnit) );
+    }
+    if( stream->outputUnit && stream->outputUnit != stream->inputUnit ) {
+       ERR_WRAP( AudioOutputUnitStart(stream->outputUnit) );
+    }
+
+    //setStreamStartTime( stream );
+    //stream->isTimeSet = TRUE;
+
+    return paNoError;
+#undef ERR_WRAP
 }
 
 
-static PaError WriteStream( PaStream* s,
-                            const void *buffer,
-                            unsigned long frames )
+static PaError StopStream( PaStream *s )
 {
-    return paInternalError;
+    PaMacCoreStream *stream = (PaMacCoreStream*)s;
+    OSErr result = noErr;
+    PaError paErr;
+    VVDBUG(("StopStream()\n"));
+
+    VDBUG( ("Waiting for BLIO.\n") );
+    waitUntilBlioWriteBufferIsFlushed( &stream->blio );
+    VDBUG( ( "Stopping stream.\n" ) );
+
+    stream->isTimeSet = FALSE;
+    stream->state = STOPPING;
+
+#define ERR_WRAP(mac_err) do { result = mac_err ; if ( result != noErr ) return ERR(result) ; } while(0)
+    /* -- stop and reset -- */
+    if( stream->inputUnit == stream->outputUnit && stream->inputUnit )
+    {
+       ERR_WRAP( AudioOutputUnitStop(stream->inputUnit) );
+       ERR_WRAP( AudioUnitReset(stream->inputUnit, kAudioUnitScope_Global, 1) );
+       ERR_WRAP( AudioUnitReset(stream->inputUnit, kAudioUnitScope_Global, 0) );
+    }
+    else
+    {
+       if( stream->inputUnit )
+       {
+          ERR_WRAP(AudioOutputUnitStop(stream->inputUnit) );
+          ERR_WRAP(AudioUnitReset(stream->inputUnit,kAudioUnitScope_Global,1));
+       }
+       if( stream->outputUnit )
+       {
+          ERR_WRAP(AudioOutputUnitStop(stream->outputUnit));
+          ERR_WRAP(AudioUnitReset(stream->outputUnit,kAudioUnitScope_Global,0));
+       }
+    }
+    if( stream->inputRingBuffer.buffer ) {
+       RingBuffer_Flush( &stream->inputRingBuffer );
+       bzero( (void *)stream->inputRingBuffer.buffer,
+              stream->inputRingBuffer.bufferSize );
+       /* advance the write point a little, so we are reading from the
+          middle of the buffer. We'll need extra at the end because
+          testing has shown that this helps. */
+       if( stream->outputUnit )
+          RingBuffer_AdvanceWriteIndex( &stream->inputRingBuffer,
+                                    stream->inputRingBuffer.bufferSize
+                                           / RING_BUFFER_ADVANCE_DENOMINATOR );
+    }
+
+    stream->xrunFlags = 0;
+    stream->state = STOPPED;
+
+    paErr = resetBlioRingBuffers( &stream->blio );
+    if( paErr )
+       return paErr;
+
+/*
+    //stream->isTimeSet = FALSE;
+*/
+
+    VDBUG( ( "Stream Stopped.\n" ) );
+    return paNoError;
+#undef ERR_WRAP
+}
+
+static PaError AbortStream( PaStream *s )
+{
+    VVDBUG(("AbortStream()->StopStream()\n"));
+    VDBUG( ( "Aborting stream.\n" ) );
+    /* We have nothing faster than StopStream. */
+    return StopStream(s);
 }
 
 
-static signed long GetStreamReadAvailable( PaStream* s )
+static PaError IsStreamStopped( PaStream *s )
 {
-    return paInternalError;
+    PaMacCoreStream *stream = (PaMacCoreStream*)s;
+    VVDBUG(("IsStreamStopped()\n"));
+
+    return stream->state == STOPPED ? 1 : 0;
 }
 
 
-static signed long GetStreamWriteAvailable( PaStream* s )
+static PaError IsStreamActive( PaStream *s )
 {
-    return paInternalError;
+    PaMacCoreStream *stream = (PaMacCoreStream*)s;
+    VVDBUG(("IsStreamActive()\n"));
+    return ( stream->state == ACTIVE || stream->state == STOPPING );
 }
 
-// HostAPI-specific initialization function
-PaError PaMacCore_Initialize( PaUtilHostApiRepresentation **hostApi, PaHostApiIndex hostApiIndex )
+
+static double GetStreamCpuLoad( PaStream* s )
 {
-    PaError result = paNoError;
-    PaMacCoreHostApiRepresentation *macCoreHostApi = (PaMacCoreHostApiRepresentation *)PaUtil_AllocateMemory( sizeof(PaMacCoreHostApiRepresentation) );
-    if( !macCoreHostApi )
-    {
-        result = paInsufficientMemory;
-        goto error;
-    }
-    
-    macCoreHostApi->allocations = PaUtil_CreateAllocationGroup();
-    if( !macCoreHostApi->allocations )
-    {
-        result = paInsufficientMemory;
-        goto error;
-    }
-    
-    *hostApi = &macCoreHostApi->inheritedHostApiRep;
-    (*hostApi)->info.structVersion = 1;
-    (*hostApi)->info.type = paCoreAudio;
-    (*hostApi)->info.name = "CoreAudio";
+    PaMacCoreStream *stream = (PaMacCoreStream*)s;
+    VVDBUG(("GetStreamCpuLoad()\n"));
 
-    result = InitializeDeviceInfos(macCoreHostApi, hostApiIndex);
-    if (result != paNoError) {
-        goto error;
-    }
-    
-    // Set up the proper callbacks to this HostApi's functions
-    (*hostApi)->Terminate = Terminate;
-    (*hostApi)->OpenStream = OpenStream;
-    (*hostApi)->IsFormatSupported = IsFormatSupported;
-    
-    PaUtil_InitializeStreamInterface( &macCoreHostApi->callbackStreamInterface, CloseStream, StartStream,
-                                      StopStream, AbortStream, IsStreamStopped, IsStreamActive,
-                                      GetStreamTime, GetStreamCpuLoad,
-                                      PaUtil_DummyRead, PaUtil_DummyWrite,
-                                      PaUtil_DummyGetReadAvailable, PaUtil_DummyGetWriteAvailable );
-    
-    PaUtil_InitializeStreamInterface( &macCoreHostApi->blockingStreamInterface, CloseStream, StartStream,
-                                      StopStream, AbortStream, IsStreamStopped, IsStreamActive,
-                                      GetStreamTime, PaUtil_DummyGetCpuLoad,
-                                      ReadStream, WriteStream, GetStreamReadAvailable, GetStreamWriteAvailable );
-    
-    return result;
-    
-error:
-        if( macCoreHostApi ) {
-            CleanUp(macCoreHostApi);
-        }
-    
-    return result;
+    return PaUtil_GetCpuLoad( &stream->cpuLoadMeasurer );
 }
diff --git a/pjmedia/src/pjmedia/portaudio/pa_mac_core.h b/pjmedia/src/pjmedia/portaudio/pa_mac_core.h
new file mode 100644
index 00000000..f810e2f6
--- /dev/null
+++ b/pjmedia/src/pjmedia/portaudio/pa_mac_core.h
@@ -0,0 +1,118 @@
+/*
+ * PortAudio Portable Real-Time Audio Library
+ * Macintosh Core Audio specific extensions
+ * portaudio.h should be included before this file.
+ *
+ * Copyright (c) 2005-2006 Bjorn Roche
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files
+ * (the "Software"), to deal in the Software without restriction,
+ * including without limitation the rights to use, copy, modify, merge,
+ * publish, distribute, sublicense, and/or sell copies of the Software,
+ * and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+/*
+ * The text above constitutes the entire PortAudio license; however, 
+ * the PortAudio community also makes the following non-binding requests:
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version. It is also 
+ * requested that these non-binding requests be included along with the 
+ * license above.
+ */
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+
+/*
+ * A pointer to a paMacCoreStreamInfo may be passed as
+ * the hostApiSpecificStreamInfo in the PaStreamParameters struct
+ * when opening a stream or querying the format. Use NULL, for the
+ * defaults. Note that for duplex streams, both infos should be the
+ * same or behaviour is undefined.
+ */
+typedef struct paMacCoreStreamInfo
+{
+    unsigned long size;         /**size of whole structure including this header */
+    PaHostApiTypeId hostApiType;/**host API for which this data is intended */
+    unsigned long version;      /**structure version */
+    unsigned long flags;        /* flags to modify behaviour */
+} paMacCoreStreamInfo;
+
+/* Use this function to initialize a paMacCoreStreamInfo struct
+   using the requested flags. */
+void paSetupMacCoreStreamInfo( paMacCoreStreamInfo *data, unsigned long flags )
+{
+   bzero( data, sizeof( paMacCoreStreamInfo ) );
+   data->size = sizeof( paMacCoreStreamInfo );
+   data->hostApiType = paCoreAudio;
+   data->version = 0x01;
+   data->flags = flags;
+}
+
+/*
+ * The following flags alter the behaviour of PA on the mac platform.
+ * they can be ORed together. These should work both for opening and
+ * checking a device.
+ */
+/* Allows PortAudio to change things like the device's frame size,
+ * which allows for much lower latency, but might disrupt the device
+ * if other programs are using it, even when you are just Querying
+ * the device. */
+const unsigned long paMacCore_ChangeDeviceParameters      = 0x01;
+
+/* In combination with the above flag,
+ * causes the stream opening to fail, unless the exact sample rates
+ * are supported by the device. */
+const unsigned long paMacCore_FailIfConversionRequired    = 0x02;
+
+/* These flags set the SR conversion quality, if required. The wierd ordering
+ * allows Maximum Quality to be the default.*/
+const unsigned long paMacCore_ConversionQualityMin    = 0x0100;
+const unsigned long paMacCore_ConversionQualityMedium = 0x0200;
+const unsigned long paMacCore_ConversionQualityLow    = 0x0300;
+const unsigned long paMacCore_ConversionQualityHigh   = 0x0400;
+const unsigned long paMacCore_ConversionQualityMax    = 0x0000;
+
+/*
+ * Here are some "preset" combinations of flags (above) to get to some
+ * common configurations. THIS IS OVERKILL, but if more flags are added
+ * it won't be.
+ */
+/*This is the default setting: do as much sample rate conversion as possible
+ * and as little mucking with the device as possible. */
+const unsigned long paMacCorePlayNice = 0x00;
+/*This setting is tuned for pro audio apps. It allows SR conversion on input
+  and output, but it tries to set the appropriate SR on the device.*/
+const unsigned long paMacCorePro      = 0x01;
+/*This is a setting to minimize CPU usage and still play nice.*/
+const unsigned long paMacCoreMinimizeCPUButPlayNice = 0x0100;
+/*This is a setting to minimize CPU usage, even if that means interrupting the device. */
+const unsigned long paMacCoreMinimizeCPU = 0x0101;
+
+#ifdef PA_OLD_CORE_AUDIO
+#	define kAudioOutputUnitProperty_EnableIO         	2003
+#	define kAudioOutputUnitProperty_SetInputCallback 	2005
+#endif
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+
diff --git a/pjmedia/src/pjmedia/portaudio/pa_mac_core_blocking.c b/pjmedia/src/pjmedia/portaudio/pa_mac_core_blocking.c
new file mode 100644
index 00000000..f474e7b0
--- /dev/null
+++ b/pjmedia/src/pjmedia/portaudio/pa_mac_core_blocking.c
@@ -0,0 +1,564 @@
+/*
+ * Implementation of the PortAudio API for Apple AUHAL
+ *
+ * PortAudio Portable Real-Time Audio Library
+ * Latest Version at: http://www.portaudio.com
+ *
+ * Written by Bjorn Roche of XO Audio LLC, from PA skeleton code.
+ * Portions copied from code by Dominic Mazzoni (who wrote a HAL implementation)
+ *
+ * Dominic's code was based on code by Phil Burk, Darren Gibbs,
+ * Gord Peters, Stephane Letz, and Greg Pfiel.
+ *
+ * The following people also deserve acknowledgements:
+ *
+ * Olivier Tristan for feedback and testing
+ * Glenn Zelniker and Z-Systems engineering for sponsoring the Blocking I/O
+ * interface.
+ * 
+ *
+ * Based on the Open Source API proposed by Ross Bencina
+ * Copyright (c) 1999-2002 Ross Bencina, Phil Burk
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files
+ * (the "Software"), to deal in the Software without restriction,
+ * including without limitation the rights to use, copy, modify, merge,
+ * publish, distribute, sublicense, and/or sell copies of the Software,
+ * and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+/*
+ * The text above constitutes the entire PortAudio license; however, 
+ * the PortAudio community also makes the following non-binding requests:
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version. It is also 
+ * requested that these non-binding requests be included along with the 
+ * license above.
+ */
+
+/**
+ @file
+ @ingroup hostaip_src
+
+ This file contains the implementation
+ required for blocking I/O. It is separated from pa_mac_core.c simply to ease
+ development.
+*/
+
+#include "pa_mac_core_blocking.h"
+#include "pa_mac_core_internal.h"
+#include <assert.h>
+#ifdef MOSX_USE_NON_ATOMIC_FLAG_BITS
+# define OSAtomicOr32( a, b ) ( (*(b)) |= (a) )
+# define OSAtomicAnd32( a, b ) ( (*(b)) &= (a) )
+#else
+# include <libkern/OSAtomic.h>
+#endif
+
+/*
+ * This fnuction determines the size of a particular sample format.
+ * if the format is not recognized, this returns zero.
+ */
+static size_t computeSampleSizeFromFormat( PaSampleFormat format )
+{
+   switch( format ) {
+   case paFloat32: return 4;
+   case paInt32: return 4;
+   case paInt24: return 3;
+   case paInt16: return 2;
+   case paInt8: case paUInt8: return 1;
+   default: return 0;
+   }
+}
+
+
+/*
+ * Functions for initializing, resetting, and destroying BLIO structures.
+ *
+ */
+
+/* This should be called with the relevant info when initializing a stream for
+   callback. */
+PaError initializeBlioRingBuffers(
+                                       PaMacBlio *blio,
+                                       PaSampleFormat inputSampleFormat,
+                                       PaSampleFormat outputSampleFormat,
+                                       size_t framesPerBuffer,
+                                       long ringBufferSize,
+                                       int inChan,
+                                       int outChan )
+{
+   void *data;
+   int result;
+
+   /* zeroify things */
+   bzero( blio, sizeof( PaMacBlio ) );
+   /* this is redundant, but the buffers are used to check
+      if the bufffers have been initialized, so we do it explicitly. */
+   blio->inputRingBuffer.buffer = NULL;
+   blio->outputRingBuffer.buffer = NULL;
+
+   /* initialize simple data */
+   blio->inputSampleFormat = inputSampleFormat;
+   blio->inputSampleSize = computeSampleSizeFromFormat(inputSampleFormat);
+   blio->outputSampleFormat = outputSampleFormat;
+   blio->outputSampleSize = computeSampleSizeFromFormat(outputSampleFormat);
+   blio->framesPerBuffer = framesPerBuffer;
+   blio->inChan = inChan;
+   blio->outChan = outChan;
+   blio->statusFlags = 0;
+   blio->errors = paNoError;
+#ifdef PA_MAC_BLIO_MUTEX
+   blio->isInputEmpty = false;
+   blio->isOutputFull = false;
+#endif
+
+   /* setup ring buffers */
+#ifdef PA_MAC_BLIO_MUTEX
+   result = PaMacCore_SetUnixError( pthread_mutex_init(&(blio->inputMutex),NULL), 0 );
+   if( result )
+      goto error;
+   result = UNIX_ERR( pthread_cond_init( &(blio->inputCond), NULL ) );
+   if( result )
+      goto error;
+   result = UNIX_ERR( pthread_mutex_init(&(blio->outputMutex),NULL) );
+   if( result )
+      goto error;
+   result = UNIX_ERR( pthread_cond_init( &(blio->outputCond), NULL ) );
+#endif
+   if( inChan ) {
+      data = calloc( ringBufferSize, blio->inputSampleSize );
+      if( !data )
+      {
+         result = paInsufficientMemory;
+         goto error;
+      }
+
+      assert( 0 == RingBuffer_Init(
+            &blio->inputRingBuffer,
+            ringBufferSize*blio->inputSampleSize,
+            data ) );
+   }
+   if( outChan ) {
+      data = calloc( ringBufferSize, blio->outputSampleSize );
+      if( !data )
+      {
+         result = paInsufficientMemory;
+         goto error;
+      }
+
+      assert( 0 == RingBuffer_Init(
+            &blio->outputRingBuffer,
+            ringBufferSize*blio->outputSampleSize,
+            data ) );
+   }
+
+   result = resetBlioRingBuffers( blio );
+   if( result )
+      goto error;
+
+   return 0;
+
+ error:
+   destroyBlioRingBuffers( blio );
+   return result;
+}
+
+#ifdef PA_MAC_BLIO_MUTEX
+PaError blioSetIsInputEmpty( PaMacBlio *blio, bool isEmpty )
+{
+   PaError result = paNoError;
+   if( isEmpty == blio->isInputEmpty )
+      goto done;
+
+   /* we need to update the value. Here's what we do:
+    * - Lock the mutex, so noone else can write.
+    * - update the value.
+    * - unlock.
+    * - broadcast to all listeners.
+    */
+   result = UNIX_ERR( pthread_mutex_lock( &blio->inputMutex ) );
+   if( result )
+      goto done;
+   blio->isInputEmpty = isEmpty;
+   result = UNIX_ERR( pthread_mutex_unlock( &blio->inputMutex ) );
+   if( result )
+      goto done;
+   result = UNIX_ERR( pthread_cond_broadcast( &blio->inputCond ) );
+   if( result )
+      goto done;
+
+ done:
+   return result;
+}
+PaError blioSetIsOutputFull( PaMacBlio *blio, bool isFull )
+{
+   PaError result = paNoError;
+   if( isFull == blio->isOutputFull )
+      goto done;
+
+   /* we need to update the value. Here's what we do:
+    * - Lock the mutex, so noone else can write.
+    * - update the value.
+    * - unlock.
+    * - broadcast to all listeners.
+    */
+   result = UNIX_ERR( pthread_mutex_lock( &blio->outputMutex ) );
+   if( result )
+      goto done;
+   blio->isOutputFull = isFull;
+   result = UNIX_ERR( pthread_mutex_unlock( &blio->outputMutex ) );
+   if( result )
+      goto done;
+   result = UNIX_ERR( pthread_cond_broadcast( &blio->outputCond ) );
+   if( result )
+      goto done;
+
+ done:
+   return result;
+}
+#endif
+
+/* This should be called after stopping or aborting the stream, so that on next
+   start, the buffers will be ready. */
+PaError resetBlioRingBuffers( PaMacBlio *blio )
+{
+#ifdef PA_MAC__BLIO_MUTEX
+   int result;
+#endif
+   blio->statusFlags = 0;
+   if( blio->outputRingBuffer.buffer ) {
+      RingBuffer_Flush( &blio->outputRingBuffer );
+      bzero( blio->outputRingBuffer.buffer,
+             blio->outputRingBuffer.bufferSize );
+      /* Advance buffer */
+      RingBuffer_AdvanceWriteIndex( &blio->outputRingBuffer, blio->outputRingBuffer.bufferSize );
+
+      /* Update isOutputFull. */
+#ifdef PA_MAC__BLIO_MUTEX
+      result = blioSetIsOutputFull( blio, toAdvance == blio->outputRingBuffer.bufferSize );
+      if( result )
+         goto error;
+#endif
+/*
+      printf( "------%d\n" ,  blio->framesPerBuffer );
+      printf( "------%d\n" ,  blio->outChan );
+      printf( "------%d\n" ,  blio->outputSampleSize );
+      printf( "------%d\n" ,  blio->framesPerBuffer*blio->outChan*blio->outputSampleSize );
+*/
+   }
+   if( blio->inputRingBuffer.buffer ) {
+      RingBuffer_Flush( &blio->inputRingBuffer );
+      bzero( blio->inputRingBuffer.buffer,
+             blio->inputRingBuffer.bufferSize );
+      /* Update isInputEmpty. */
+#ifdef PA_MAC__BLIO_MUTEX
+      result = blioSetIsInputEmpty( blio, true );
+      if( result )
+         goto error;
+#endif
+   }
+   return paNoError;
+#ifdef PA_MAC__BLIO_MUTEX
+ error:
+   return result;
+#endif
+}
+
+/*This should be called when you are done with the blio. It can safely be called
+  multiple times if there are no exceptions. */
+PaError destroyBlioRingBuffers( PaMacBlio *blio )
+{
+   PaError result = paNoError;
+   if( blio->inputRingBuffer.buffer ) {
+      free( blio->inputRingBuffer.buffer );
+#ifdef PA_MAC__BLIO_MUTEX
+      result = UNIX_ERR( pthread_mutex_destroy( & blio->inputMutex ) );
+      if( result ) return result;
+      result = UNIX_ERR( pthread_cond_destroy( & blio->inputCond ) );
+      if( result ) return result;
+#endif
+   }
+   blio->inputRingBuffer.buffer = NULL;
+   if( blio->outputRingBuffer.buffer ) {
+      free( blio->outputRingBuffer.buffer );
+#ifdef PA_MAC__BLIO_MUTEX
+      result = UNIX_ERR( pthread_mutex_destroy( & blio->outputMutex ) );
+      if( result ) return result;
+      result = UNIX_ERR( pthread_cond_destroy( & blio->outputCond ) );
+      if( result ) return result;
+#endif
+   }
+   blio->outputRingBuffer.buffer = NULL;
+
+   return result;
+}
+
+/*
+ * this is the BlioCallback function. It expects to recieve a PaMacBlio Object
+ * pointer as userData.
+ *
+ */
+int BlioCallback( const void *input, void *output, unsigned long frameCount,
+	const PaStreamCallbackTimeInfo* timeInfo,
+        PaStreamCallbackFlags statusFlags,
+        void *userData )
+{
+   PaMacBlio *blio = (PaMacBlio*)userData;
+   long avail;
+   long toRead;
+   long toWrite;
+
+   /* set flags returned by OS: */
+   OSAtomicOr32( statusFlags, &blio->statusFlags ) ;
+
+   /* --- Handle Input Buffer --- */
+   if( blio->inChan ) {
+      avail = RingBuffer_GetWriteAvailable( &blio->inputRingBuffer );
+
+      /* check for underflow */
+      if( avail < frameCount * blio->inputSampleSize * blio->inChan )
+         OSAtomicOr32( paInputOverflow, &blio->statusFlags );
+
+      toRead = MIN( avail, frameCount * blio->inputSampleSize * blio->inChan );
+
+      /* copy the data */
+      /*printf( "reading %d\n", toRead );*/
+      assert( toRead == RingBuffer_Write( &blio->inputRingBuffer, input, toRead ) );
+#ifdef PA_MAC__BLIO_MUTEX
+      /* Priority inversion. See notes below. */
+      blioSetIsInputEmpty( blio, false );
+#endif
+   }
+
+
+   /* --- Handle Output Buffer --- */
+   if( blio->outChan ) {
+      avail = RingBuffer_GetReadAvailable( &blio->outputRingBuffer );
+
+      /* check for underflow */
+      if( avail < frameCount * blio->outputSampleSize * blio->outChan )
+         OSAtomicOr32( paOutputUnderflow, &blio->statusFlags );
+
+      toWrite = MIN( avail, frameCount * blio->outputSampleSize * blio->outChan );
+
+      if( toWrite != frameCount * blio->outputSampleSize * blio->outChan )
+         bzero( ((char *)output)+toWrite,
+                frameCount * blio->outputSampleSize * blio->outChan - toWrite );
+      /* copy the data */
+      /*printf( "writing %d\n", toWrite );*/
+      assert( toWrite == RingBuffer_Read( &blio->outputRingBuffer, output, toWrite ) );
+#ifdef PA_MAC__BLIO_MUTEX
+      /* We have a priority inversion here. However, we will only have to
+         wait if this was true and is now false, which means we've got
+         some room in the buffer.
+         Hopefully problems will be minimized. */
+      blioSetIsOutputFull( blio, false );
+#endif
+   }
+
+   return paContinue;
+}
+
+PaError ReadStream( PaStream* stream,
+                           void *buffer,
+                           unsigned long frames )
+{
+    PaMacBlio *blio = & ((PaMacCoreStream*)stream) -> blio;
+    char *cbuf = (char *) buffer;
+    PaError ret = paNoError;
+    VVDBUG(("ReadStream()\n"));
+
+    while( frames > 0 ) {
+       long avail;
+       long toRead;
+       do {
+          avail = RingBuffer_GetReadAvailable( &blio->inputRingBuffer );
+/*
+          printf( "Read Buffer is %%%g full: %ld of %ld.\n",
+                  100 * (float)avail / (float) blio->inputRingBuffer.bufferSize,
+                  avail, blio->inputRingBuffer.bufferSize );
+*/
+          if( avail == 0 ) {
+#ifdef PA_MAC_BLIO_MUTEX
+             /**block when empty*/
+             ret = UNIX_ERR( pthread_mutex_lock( &blio->inputMutex ) );
+             if( ret )
+                return ret;
+             while( blio->isInputEmpty ) {
+                ret = UNIX_ERR( pthread_cond_wait( &blio->inputCond, &blio->inputMutex ) );
+                if( ret )
+                   return ret;
+             }
+             ret = UNIX_ERR( pthread_mutex_unlock( &blio->inputMutex ) );
+             if( ret )
+                return ret;
+#else
+             Pa_Sleep( PA_MAC_BLIO_BUSY_WAIT_SLEEP_INTERVAL );
+#endif
+          }
+       } while( avail == 0 );
+       toRead = MIN( avail, frames * blio->inputSampleSize * blio->inChan );
+       toRead -= toRead % blio->inputSampleSize * blio->inChan ;
+       RingBuffer_Read( &blio->inputRingBuffer, (void *)cbuf, toRead );
+       cbuf += toRead;
+       frames -= toRead / ( blio->inputSampleSize * blio->inChan );
+
+       if( toRead == avail ) {
+#ifdef PA_MAC_BLIO_MUTEX
+          /* we just emptied the buffer, so we need to mark it as empty. */
+          ret = blioSetIsInputEmpty( blio, true );
+          if( ret )
+             return ret;
+          /* of course, in the meantime, the callback may have put some sats
+             in, so
+             so check for that, too, to avoid a race condition. */
+          if( RingBuffer_GetReadAvailable( &blio->inputRingBuffer ) ) {
+             blioSetIsInputEmpty( blio, false );
+             if( ret )
+                return ret;
+          }
+#endif
+       }
+    }
+
+    /*   Report either paNoError or paInputOverflowed. */
+    /*   may also want to report other errors, but this is non-standard. */
+    ret = blio->statusFlags & paInputOverflow;
+
+    /* report underflow only once: */
+    if( ret ) {
+       OSAtomicAnd32( ~paInputOverflow, &blio->statusFlags );
+       ret = paInputOverflowed;
+    }
+
+    return ret;
+}
+
+
+PaError WriteStream( PaStream* stream,
+                            const void *buffer,
+                            unsigned long frames )
+{
+    PaMacBlio *blio = & ((PaMacCoreStream*)stream) -> blio;
+    char *cbuf = (char *) buffer;
+    PaError ret = paNoError;
+    VVDBUG(("WriteStream()\n"));
+
+    while( frames > 0 ) {
+       long avail = 0;
+       long toWrite;
+
+       do {
+          avail = RingBuffer_GetWriteAvailable( &blio->outputRingBuffer );
+/*
+          printf( "Write Buffer is %%%g full: %ld of %ld.\n",
+                  100 - 100 * (float)avail / (float) blio->outputRingBuffer.bufferSize,
+                  avail, blio->outputRingBuffer.bufferSize );
+*/
+          if( avail == 0 ) {
+#ifdef PA_MAC_BLIO_MUTEX
+             /*block while full*/
+             ret = UNIX_ERR( pthread_mutex_lock( &blio->outputMutex ) );
+             if( ret )
+                return ret;
+             while( blio->isOutputFull ) {
+                ret = UNIX_ERR( pthread_cond_wait( &blio->outputCond, &blio->outputMutex ) );
+                if( ret )
+                   return ret;
+             }
+             ret = UNIX_ERR( pthread_mutex_unlock( &blio->outputMutex ) );
+             if( ret )
+                return ret;
+#else
+             Pa_Sleep( PA_MAC_BLIO_BUSY_WAIT_SLEEP_INTERVAL );
+#endif
+          }
+       } while( avail == 0 );
+
+       toWrite = MIN( avail, frames * blio->outputSampleSize * blio->outChan );
+       toWrite -= toWrite % blio->outputSampleSize * blio->outChan ;
+       RingBuffer_Write( &blio->outputRingBuffer, (void *)cbuf, toWrite );
+       cbuf += toWrite;
+       frames -= toWrite / ( blio->outputSampleSize * blio->outChan );
+
+#ifdef PA_MAC_BLIO_MUTEX
+       if( toWrite == avail ) {
+          /* we just filled up the buffer, so we need to mark it as filled. */
+          ret = blioSetIsOutputFull( blio, true );
+          if( ret )
+             return ret;
+          /* of course, in the meantime, we may have emptied the buffer, so
+             so check for that, too, to avoid a race condition. */
+          if( RingBuffer_GetWriteAvailable( &blio->outputRingBuffer ) ) {
+             blioSetIsOutputFull( blio, false );
+             if( ret )
+                return ret;
+          }
+       }
+#endif
+    }
+
+    /*   Report either paNoError or paOutputUnderflowed. */
+    /*   may also want to report other errors, but this is non-standard. */
+    ret = blio->statusFlags & paOutputUnderflow;
+
+    /* report underflow only once: */
+    if( ret ) {
+      OSAtomicAnd32( ~paOutputUnderflow, &blio->statusFlags );
+      ret = paOutputUnderflowed;
+    }
+
+    return ret;
+}
+
+/*
+ *
+ */
+void waitUntilBlioWriteBufferIsFlushed( PaMacBlio *blio )
+{
+    if( blio->outputRingBuffer.buffer ) {
+       long avail = RingBuffer_GetWriteAvailable( &blio->outputRingBuffer );
+       while( avail != blio->outputRingBuffer.bufferSize ) {
+          if( avail == 0 )
+             Pa_Sleep( PA_MAC_BLIO_BUSY_WAIT_SLEEP_INTERVAL );
+          avail = RingBuffer_GetWriteAvailable( &blio->outputRingBuffer );
+       }
+    }
+}
+
+
+signed long GetStreamReadAvailable( PaStream* stream )
+{
+    PaMacBlio *blio = & ((PaMacCoreStream*)stream) -> blio;
+    VVDBUG(("GetStreamReadAvailable()\n"));
+
+    return RingBuffer_GetReadAvailable( &blio->inputRingBuffer )
+                         / ( blio->outputSampleSize * blio->outChan );
+}
+
+
+signed long GetStreamWriteAvailable( PaStream* stream )
+{
+    PaMacBlio *blio = & ((PaMacCoreStream*)stream) -> blio;
+    VVDBUG(("GetStreamWriteAvailable()\n"));
+
+    return RingBuffer_GetWriteAvailable( &blio->outputRingBuffer )
+                         / ( blio->outputSampleSize * blio->outChan );
+}
+
diff --git a/pjmedia/src/pjmedia/portaudio/pa_mac_core_blocking.h b/pjmedia/src/pjmedia/portaudio/pa_mac_core_blocking.h
new file mode 100644
index 00000000..8185e203
--- /dev/null
+++ b/pjmedia/src/pjmedia/portaudio/pa_mac_core_blocking.h
@@ -0,0 +1,133 @@
+/*
+ * Internal blocking interfaces for PortAudio Apple AUHAL implementation
+ *
+ * PortAudio Portable Real-Time Audio Library
+ * Latest Version at: http://www.portaudio.com
+ *
+ * Written by Bjorn Roche of XO Audio LLC, from PA skeleton code.
+ * Portions copied from code by Dominic Mazzoni (who wrote a HAL implementation)
+ *
+ * Dominic's code was based on code by Phil Burk, Darren Gibbs,
+ * Gord Peters, Stephane Letz, and Greg Pfiel.
+ *
+ * The following people also deserve acknowledgements:
+ *
+ * Olivier Tristan for feedback and testing
+ * Glenn Zelniker and Z-Systems engineering for sponsoring the Blocking I/O
+ * interface.
+ * 
+ *
+ * Based on the Open Source API proposed by Ross Bencina
+ * Copyright (c) 1999-2002 Ross Bencina, Phil Burk
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files
+ * (the "Software"), to deal in the Software without restriction,
+ * including without limitation the rights to use, copy, modify, merge,
+ * publish, distribute, sublicense, and/or sell copies of the Software,
+ * and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+/*
+ * The text above constitutes the entire PortAudio license; however, 
+ * the PortAudio community also makes the following non-binding requests:
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version. It is also 
+ * requested that these non-binding requests be included along with the 
+ * license above.
+ */
+
+/**
+ @file
+ @ingroup hostaip_src
+*/
+
+#ifndef PA_MAC_CORE_BLOCKING_H_
+#define PA_MAC_CORE_BLOCKING_H_
+
+#include "ringbuffer.h"
+#include "portaudio.h"
+#include "pa_mac_core_utilities.h"
+
+/*
+ * Number of miliseconds to busy wait whil waiting for data in blocking calls.
+ */
+#define PA_MAC_BLIO_BUSY_WAIT_SLEEP_INTERVAL (5)
+/*
+ * Define exactly one of these blocking methods
+ * PA_MAC_BLIO_MUTEX is not actively maintained.
+ */
+#define PA_MAC_BLIO_BUSY_WAIT
+/*
+#define PA_MAC_BLIO_MUTEX
+*/
+
+typedef struct {
+    RingBuffer inputRingBuffer;
+    RingBuffer outputRingBuffer;
+    PaSampleFormat inputSampleFormat;
+    size_t inputSampleSize;
+    PaSampleFormat outputSampleFormat;
+    size_t outputSampleSize;
+
+    size_t framesPerBuffer;
+
+    int inChan;
+    int outChan;
+
+    //PaStreamCallbackFlags statusFlags;
+    uint32_t statusFlags;
+    PaError errors;
+
+    /* Here we handle blocking, using condition variables. */
+#ifdef  PA_MAC_BLIO_MUTEX
+    volatile bool isInputEmpty;
+    pthread_mutex_t inputMutex;
+    pthread_cond_t inputCond;
+
+    volatile bool isOutputFull;
+    pthread_mutex_t outputMutex;
+    pthread_cond_t outputCond;
+#endif
+}
+PaMacBlio;
+
+/*
+ * These functions operate on condition and related variables.
+ */
+
+PaError initializeBlioRingBuffers(
+                                       PaMacBlio *blio,
+                                       PaSampleFormat inputSampleFormat,
+                                       PaSampleFormat outputSampleFormat,
+                                       size_t framesPerBuffer,
+                                       long ringBufferSize,
+                                       int inChan,
+                                       int outChan );
+PaError destroyBlioRingBuffers( PaMacBlio *blio );
+PaError resetBlioRingBuffers( PaMacBlio *blio );
+
+int BlioCallback(
+        const void *input, void *output,
+        unsigned long frameCount,
+        const PaStreamCallbackTimeInfo* timeInfo,
+        PaStreamCallbackFlags statusFlags,
+        void *userData );
+
+void waitUntilBlioWriteBufferIsFlushed( PaMacBlio *blio );
+
+#endif /*PA_MAC_CORE_BLOCKING_H_*/
diff --git a/pjmedia/src/pjmedia/portaudio/pa_mac_core_internal.h b/pjmedia/src/pjmedia/portaudio/pa_mac_core_internal.h
new file mode 100644
index 00000000..095b3b38
--- /dev/null
+++ b/pjmedia/src/pjmedia/portaudio/pa_mac_core_internal.h
@@ -0,0 +1,162 @@
+/*
+ * Internal interfaces for PortAudio Apple AUHAL implementation
+ *
+ * PortAudio Portable Real-Time Audio Library
+ * Latest Version at: http://www.portaudio.com
+ *
+ * Written by Bjorn Roche of XO Audio LLC, from PA skeleton code.
+ * Portions copied from code by Dominic Mazzoni (who wrote a HAL implementation)
+ *
+ * Dominic's code was based on code by Phil Burk, Darren Gibbs,
+ * Gord Peters, Stephane Letz, and Greg Pfiel.
+ *
+ * The following people also deserve acknowledgements:
+ *
+ * Olivier Tristan for feedback and testing
+ * Glenn Zelniker and Z-Systems engineering for sponsoring the Blocking I/O
+ * interface.
+ * 
+ *
+ * Based on the Open Source API proposed by Ross Bencina
+ * Copyright (c) 1999-2002 Ross Bencina, Phil Burk
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files
+ * (the "Software"), to deal in the Software without restriction,
+ * including without limitation the rights to use, copy, modify, merge,
+ * publish, distribute, sublicense, and/or sell copies of the Software,
+ * and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+/*
+ * The text above constitutes the entire PortAudio license; however, 
+ * the PortAudio community also makes the following non-binding requests:
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version. It is also 
+ * requested that these non-binding requests be included along with the 
+ * license above.
+ */
+
+/**
+ @file pa_mac_core
+ @ingroup hostapi_src
+ @author Bjorn Roche
+ @brief AUHAL implementation of PortAudio
+*/
+
+#ifndef PA_MAC_CORE_INTERNAL_H__
+#define PA_MAC_CORE_INTERNAL_H__
+
+#include <AudioUnit/AudioUnit.h>
+#include <AudioToolbox/AudioToolbox.h>
+
+
+#include "portaudio.h"
+#include "pa_util.h"
+#include "pa_hostapi.h"
+#include "pa_stream.h"
+#include "pa_allocation.h"
+#include "pa_cpuload.h"
+#include "pa_process.h"
+#include "ringbuffer.h"
+
+#include "pa_mac_core_blocking.h"
+
+/* function prototypes */
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif /* __cplusplus */
+
+PaError PaMacCore_Initialize( PaUtilHostApiRepresentation **hostApi, PaHostApiIndex index );
+
+#ifdef __cplusplus
+}
+#endif /* __cplusplus */
+
+#define RING_BUFFER_ADVANCE_DENOMINATOR (4)
+
+PaError ReadStream( PaStream* stream, void *buffer, unsigned long frames );
+PaError WriteStream( PaStream* stream, const void *buffer, unsigned long frames );
+signed long GetStreamReadAvailable( PaStream* stream );
+signed long GetStreamWriteAvailable( PaStream* stream );
+/* PaMacAUHAL - host api datastructure specific to this implementation */
+typedef struct
+{
+    PaUtilHostApiRepresentation inheritedHostApiRep;
+    PaUtilStreamInterface callbackStreamInterface;
+    PaUtilStreamInterface blockingStreamInterface;
+
+    PaUtilAllocationGroup *allocations;
+
+    /* implementation specific data goes here */
+    long devCount;
+    AudioDeviceID *devIds; /*array of all audio devices*/
+    AudioDeviceID defaultIn;
+    AudioDeviceID defaultOut;
+}
+PaMacAUHAL;
+
+
+
+/* stream data structure specifically for this implementation */
+typedef struct PaMacCoreStream
+{
+    PaUtilStreamRepresentation streamRepresentation;
+    PaUtilCpuLoadMeasurer cpuLoadMeasurer;
+    PaUtilBufferProcessor bufferProcessor;
+
+    /* implementation specific data goes here */
+    bool bufferProcessorIsInitialized;
+    AudioUnit inputUnit;
+    AudioUnit outputUnit;
+    AudioDeviceID inputDevice;
+    AudioDeviceID outputDevice;
+    size_t userInChan;
+    size_t userOutChan;
+    size_t inputFramesPerBuffer;
+    size_t outputFramesPerBuffer;
+    PaMacBlio blio;
+    /* We use this ring buffer when input and out devs are different. */
+    RingBuffer inputRingBuffer;
+    /* We may need to do SR conversion on input. */
+    AudioConverterRef inputSRConverter;
+    /* We need to preallocate an inputBuffer for reading data. */
+    AudioBufferList inputAudioBufferList;
+    AudioTimeStamp startTime;
+    volatile PaStreamCallbackFlags xrunFlags;
+    volatile bool isTimeSet;
+    volatile enum {
+       STOPPED          = 0, /* playback is completely stopped,
+                                and the user has called StopStream(). */
+       CALLBACK_STOPPED = 1, /* callback has requested stop,
+                                but user has not yet called StopStream(). */
+       STOPPING         = 2, /* The stream is in the process of closing.
+                                This state is just used internally;
+                                externally it is indistinguishable from
+                                ACTIVE.*/
+       ACTIVE           = 3  /* The stream is active and running. */
+    } state;
+    double sampleRate;
+    //these may be different from the stream sample rate due to SR conversion:
+    double outDeviceSampleRate;
+    double inDeviceSampleRate;
+}
+PaMacCoreStream;
+
+#endif /* PA_MAC_CORE_INTERNAL_H__ */
diff --git a/pjmedia/src/pjmedia/portaudio/pa_mac_core_old.c b/pjmedia/src/pjmedia/portaudio/pa_mac_core_old.c
new file mode 100644
index 00000000..acb02c4a
--- /dev/null
+++ b/pjmedia/src/pjmedia/portaudio/pa_mac_core_old.c
@@ -0,0 +1,914 @@
+/*
+ * $Id$
+ * pa_mac_core.c
+ * Implementation of PortAudio for Mac OS X CoreAudio       
+ *                                                                                         
+ * PortAudio Portable Real-Time Audio Library
+ * Latest Version at: http://www.portaudio.com
+ *
+ * Authors: Ross Bencina and Phil Burk
+ * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files
+ * (the "Software"), to deal in the Software without restriction,
+ * including without limitation the rights to use, copy, modify, merge,
+ * publish, distribute, sublicense, and/or sell copies of the Software,
+ * and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+/*
+ * The text above constitutes the entire PortAudio license; however, 
+ * the PortAudio community also makes the following non-binding requests:
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version. It is also 
+ * requested that these non-binding requests be included along with the 
+ * license above.
+ */
+
+#include <CoreAudio/CoreAudio.h>
+#include <AudioToolbox/AudioToolbox.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <math.h>
+#include <assert.h>
+
+#include "portaudio.h"
+#include "pa_trace.h"
+#include "pa_util.h"
+#include "pa_allocation.h"
+#include "pa_hostapi.h"
+#include "pa_stream.h"
+#include "pa_cpuload.h"
+#include "pa_process.h"
+
+// =====  constants  =====
+
+// =====  structs  =====
+#pragma mark structs
+
+// PaMacCoreHostApiRepresentation - host api datastructure specific to this implementation
+typedef struct PaMacCore_HAR
+{
+    PaUtilHostApiRepresentation inheritedHostApiRep;
+    PaUtilStreamInterface callbackStreamInterface;
+    PaUtilStreamInterface blockingStreamInterface;
+    
+    PaUtilAllocationGroup *allocations;
+    AudioDeviceID *macCoreDeviceIds;
+}
+PaMacCoreHostApiRepresentation;
+
+typedef struct PaMacCore_DI
+{
+    PaDeviceInfo inheritedDeviceInfo;
+}
+PaMacCoreDeviceInfo;
+
+// PaMacCoreStream - a stream data structure specifically for this implementation
+typedef struct PaMacCore_S
+{
+    PaUtilStreamRepresentation streamRepresentation;
+    PaUtilCpuLoadMeasurer cpuLoadMeasurer;
+    PaUtilBufferProcessor bufferProcessor;
+    
+    int primeStreamUsingCallback;
+    
+    AudioDeviceID inputDevice;
+    AudioDeviceID outputDevice;
+    
+    // Processing thread management --------------
+//    HANDLE abortEvent;
+//    HANDLE processingThread;
+//    DWORD processingThreadId;
+    
+    char throttleProcessingThreadOnOverload; // 0 -> don't throtte, non-0 -> throttle
+    int processingThreadPriority;
+    int highThreadPriority;
+    int throttledThreadPriority;
+    unsigned long throttledSleepMsecs;
+    
+    int isStopped;
+    volatile int isActive;
+    volatile int stopProcessing; // stop thread once existing buffers have been returned
+    volatile int abortProcessing; // stop thread immediately
+    
+//    DWORD allBuffersDurationMs; // used to calculate timeouts
+}
+PaMacCoreStream;
+
+// Data needed by the CoreAudio callback functions
+typedef struct PaMacCore_CD
+{
+    PaMacCoreStream *stream;
+    PaStreamCallback *callback;
+    void *userData;
+    PaUtilConverter *inputConverter;
+    PaUtilConverter *outputConverter;
+    void *inputBuffer;
+    void *outputBuffer;
+    int inputChannelCount;
+    int outputChannelCount;
+    PaSampleFormat inputSampleFormat;
+    PaSampleFormat outputSampleFormat;
+    PaUtilTriangularDitherGenerator *ditherGenerator;
+}
+PaMacClientData;
+
+// =====  CoreAudio-PortAudio bridge functions =====
+#pragma mark CoreAudio-PortAudio bridge functions
+
+// Maps CoreAudio OSStatus codes to PortAudio PaError codes
+static PaError conv_err(OSStatus error)
+{
+    PaError result;
+    
+    switch (error) {
+        case kAudioHardwareNoError:
+            result = paNoError; break;
+        case kAudioHardwareNotRunningError:
+            result = paInternalError; break;
+        case kAudioHardwareUnspecifiedError:
+            result = paInternalError; break;
+        case kAudioHardwareUnknownPropertyError:
+            result = paInternalError; break;
+        case kAudioHardwareBadPropertySizeError:
+            result = paInternalError; break;
+        case kAudioHardwareIllegalOperationError:
+            result = paInternalError; break;
+        case kAudioHardwareBadDeviceError:
+            result = paInvalidDevice; break;
+        case kAudioHardwareBadStreamError:
+            result = paBadStreamPtr; break;
+	//bennylp:
+        //case kAudioHardwareUnsupportedOperationError:
+        //    result = paInternalError; break;
+        case kAudioDeviceUnsupportedFormatError:
+            result = paSampleFormatNotSupported; break;
+        case kAudioDevicePermissionsError:
+            result = paDeviceUnavailable; break;
+        default:
+            result = paInternalError;
+    }
+    
+    return result;
+}
+
+/* This function is unused
+static AudioStreamBasicDescription *InitializeStreamDescription(const PaStreamParameters *parameters, double sampleRate)
+{
+    struct AudioStreamBasicDescription *streamDescription = PaUtil_AllocateMemory(sizeof(AudioStreamBasicDescription));
+    streamDescription->mSampleRate = sampleRate;
+    streamDescription->mFormatID = kAudioFormatLinearPCM;
+    streamDescription->mFormatFlags = 0;
+    streamDescription->mFramesPerPacket = 1;
+    
+    if (parameters->sampleFormat & paNonInterleaved) {
+        streamDescription->mFormatFlags |= kLinearPCMFormatFlagIsNonInterleaved;
+        streamDescription->mChannelsPerFrame = 1;
+        streamDescription->mBytesPerFrame = Pa_GetSampleSize(parameters->sampleFormat);
+        streamDescription->mBytesPerPacket = Pa_GetSampleSize(parameters->sampleFormat);
+    }
+    else {
+        streamDescription->mChannelsPerFrame = parameters->channelCount;
+    }
+    
+    streamDescription->mBytesPerFrame = Pa_GetSampleSize(parameters->sampleFormat) * streamDescription->mChannelsPerFrame;
+    streamDescription->mBytesPerPacket = streamDescription->mBytesPerFrame * streamDescription->mFramesPerPacket;
+    
+    if (parameters->sampleFormat & paFloat32) {
+        streamDescription->mFormatFlags |= kLinearPCMFormatFlagIsFloat;
+        streamDescription->mBitsPerChannel = 32;
+    }
+    else if (parameters->sampleFormat & paInt32) {
+        streamDescription->mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger;
+        streamDescription->mBitsPerChannel = 32;
+    }
+    else if (parameters->sampleFormat & paInt24) {
+        streamDescription->mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger;
+        streamDescription->mBitsPerChannel = 24;
+    }
+    else if (parameters->sampleFormat & paInt16) {
+        streamDescription->mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger;
+        streamDescription->mBitsPerChannel = 16;
+    }
+    else if (parameters->sampleFormat & paInt8) {
+        streamDescription->mFormatFlags |= kLinearPCMFormatFlagIsSignedInteger;
+        streamDescription->mBitsPerChannel = 8;
+    }    
+    else if (parameters->sampleFormat & paInt32) {
+        streamDescription->mBitsPerChannel = 8;
+    }
+    
+    return streamDescription;
+}
+*/
+
+static PaStreamCallbackTimeInfo *InitializeTimeInfo(const AudioTimeStamp* now, const AudioTimeStamp* inputTime, const AudioTimeStamp* outputTime)
+{
+    PaStreamCallbackTimeInfo *timeInfo = PaUtil_AllocateMemory(sizeof(PaStreamCallbackTimeInfo));
+    
+    timeInfo->inputBufferAdcTime = inputTime->mSampleTime;
+    timeInfo->currentTime = now->mSampleTime;
+    timeInfo->outputBufferDacTime = outputTime->mSampleTime;
+    
+    return timeInfo;
+}
+
+// =====  support functions  =====
+#pragma mark support functions
+
+static void CleanUp(PaMacCoreHostApiRepresentation *macCoreHostApi)
+{
+    if( macCoreHostApi->allocations )
+    {
+        PaUtil_FreeAllAllocations( macCoreHostApi->allocations );
+        PaUtil_DestroyAllocationGroup( macCoreHostApi->allocations );
+    }
+    
+    PaUtil_FreeMemory( macCoreHostApi );    
+}
+
+static PaError GetChannelInfo(PaDeviceInfo *deviceInfo, AudioDeviceID macCoreDeviceId, int isInput)
+{
+    UInt32 propSize;
+    PaError err = paNoError;
+    UInt32 i;
+    int numChannels = 0;
+    AudioBufferList *buflist;
+
+    err = conv_err(AudioDeviceGetPropertyInfo(macCoreDeviceId, 0, isInput, kAudioDevicePropertyStreamConfiguration, &propSize, NULL));
+    buflist = PaUtil_AllocateMemory(propSize);
+    err = conv_err(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyStreamConfiguration, &propSize, buflist));
+    if (!err) {
+        for (i = 0; i < buflist->mNumberBuffers; ++i) {
+            numChannels += buflist->mBuffers[i].mNumberChannels;
+        }
+		
+		if (isInput)
+			deviceInfo->maxInputChannels = numChannels;
+		else
+			deviceInfo->maxOutputChannels = numChannels;
+		
+        int frameLatency;
+        propSize = sizeof(UInt32);
+        err = conv_err(AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyLatency, &propSize, &frameLatency));
+        if (!err) {
+            double secondLatency = frameLatency / deviceInfo->defaultSampleRate;
+            if (isInput) {
+                deviceInfo->defaultLowInputLatency = secondLatency;
+                deviceInfo->defaultHighInputLatency = secondLatency;
+            }
+            else {
+                deviceInfo->defaultLowOutputLatency = secondLatency;
+                deviceInfo->defaultHighOutputLatency = secondLatency;
+            }
+        }
+    }
+    PaUtil_FreeMemory(buflist);
+    
+    return err;
+}
+
+static PaError InitializeDeviceInfo(PaMacCoreDeviceInfo *macCoreDeviceInfo,  AudioDeviceID macCoreDeviceId, PaHostApiIndex hostApiIndex )
+{
+    PaDeviceInfo *deviceInfo = &macCoreDeviceInfo->inheritedDeviceInfo;
+    deviceInfo->structVersion = 2;
+    deviceInfo->hostApi = hostApiIndex;
+    
+    PaError err = paNoError;
+    UInt32 propSize;
+
+    err = conv_err(AudioDeviceGetPropertyInfo(macCoreDeviceId, 0, 0, kAudioDevicePropertyDeviceName, &propSize, NULL));
+    // FIXME: this allocation should be part of the allocations group
+    char *name = PaUtil_AllocateMemory(propSize);
+    err = conv_err(AudioDeviceGetProperty(macCoreDeviceId, 0, 0, kAudioDevicePropertyDeviceName, &propSize, name));
+    if (!err) {
+        deviceInfo->name = name;
+    }
+    
+    Float64 sampleRate;
+    propSize = sizeof(Float64);
+    err = conv_err(AudioDeviceGetProperty(macCoreDeviceId, 0, 0, kAudioDevicePropertyNominalSampleRate, &propSize, &sampleRate));
+    if (!err) {
+        deviceInfo->defaultSampleRate = sampleRate;
+    }
+
+
+    // Get channel info
+    err = GetChannelInfo(deviceInfo, macCoreDeviceId, 1);
+    err = GetChannelInfo(deviceInfo, macCoreDeviceId, 0);
+
+    return err;
+}
+
+static PaError InitializeDeviceInfos( PaMacCoreHostApiRepresentation *macCoreHostApi, PaHostApiIndex hostApiIndex )
+{
+    PaError result = paNoError;
+    PaUtilHostApiRepresentation *hostApi;
+    PaMacCoreDeviceInfo *deviceInfoArray;
+
+    // initialise device counts and default devices under the assumption that there are no devices. These values are incremented below if and when devices are successfully initialized.
+    hostApi = &macCoreHostApi->inheritedHostApiRep;
+    hostApi->info.deviceCount = 0;
+    hostApi->info.defaultInputDevice = paNoDevice;
+    hostApi->info.defaultOutputDevice = paNoDevice;
+    
+    UInt32 propsize;
+    AudioHardwareGetPropertyInfo(kAudioHardwarePropertyDevices, &propsize, NULL);
+    int numDevices = propsize / sizeof(AudioDeviceID);
+    hostApi->info.deviceCount = numDevices;
+    if (numDevices > 0) {
+        hostApi->deviceInfos = (PaDeviceInfo**)PaUtil_GroupAllocateMemory(
+                                            macCoreHostApi->allocations, sizeof(PaDeviceInfo*) * numDevices );
+        if( !hostApi->deviceInfos )
+        {
+            return paInsufficientMemory;
+        }
+
+        // allocate all device info structs in a contiguous block
+        deviceInfoArray = (PaMacCoreDeviceInfo*)PaUtil_GroupAllocateMemory(
+                                macCoreHostApi->allocations, sizeof(PaMacCoreDeviceInfo) * numDevices );
+        if( !deviceInfoArray )
+        {
+            return paInsufficientMemory;
+        }
+        
+        macCoreHostApi->macCoreDeviceIds = PaUtil_GroupAllocateMemory(macCoreHostApi->allocations, propsize);
+        AudioHardwareGetProperty(kAudioHardwarePropertyDevices, &propsize, macCoreHostApi->macCoreDeviceIds);
+
+        AudioDeviceID defaultInputDevice, defaultOutputDevice;
+        propsize = sizeof(AudioDeviceID);
+        AudioHardwareGetProperty(kAudioHardwarePropertyDefaultInputDevice, &propsize, &defaultInputDevice);
+        AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, &propsize, &defaultOutputDevice);
+        
+        UInt32 i;
+        for (i = 0; i < numDevices; ++i) {
+            if (macCoreHostApi->macCoreDeviceIds[i] == defaultInputDevice) {
+                hostApi->info.defaultInputDevice = i;
+            }
+            if (macCoreHostApi->macCoreDeviceIds[i] == defaultOutputDevice) {
+                hostApi->info.defaultOutputDevice = i;
+            }
+            InitializeDeviceInfo(&deviceInfoArray[i], macCoreHostApi->macCoreDeviceIds[i], hostApiIndex);
+            hostApi->deviceInfos[i] = &(deviceInfoArray[i].inheritedDeviceInfo);      
+        }
+    }
+
+    return result;
+}
+
+static OSStatus CheckFormat(AudioDeviceID macCoreDeviceId, const PaStreamParameters *parameters, double sampleRate, int isInput)
+{
+    UInt32 propSize = sizeof(AudioStreamBasicDescription);
+    AudioStreamBasicDescription *streamDescription = PaUtil_AllocateMemory(propSize);
+
+    streamDescription->mSampleRate = sampleRate;
+    streamDescription->mFormatID = 0;
+    streamDescription->mFormatFlags = 0;
+    streamDescription->mBytesPerPacket = 0;
+    streamDescription->mFramesPerPacket = 0;
+    streamDescription->mBytesPerFrame = 0;
+    streamDescription->mChannelsPerFrame = 0;
+    streamDescription->mBitsPerChannel = 0;
+    streamDescription->mReserved = 0;
+
+    OSStatus result = AudioDeviceGetProperty(macCoreDeviceId, 0, isInput, kAudioDevicePropertyStreamFormatSupported, &propSize, streamDescription);
+    PaUtil_FreeMemory(streamDescription);
+    return result;
+}
+
+static OSStatus CopyInputData(PaMacClientData* destination, const AudioBufferList *source, unsigned long frameCount)
+{
+    int frameSpacing, channelSpacing;
+    if (destination->inputSampleFormat & paNonInterleaved) {
+        frameSpacing = 1;
+        channelSpacing = destination->inputChannelCount;
+    }
+    else {
+        frameSpacing = destination->inputChannelCount;
+        channelSpacing = 1;
+    }
+    
+    AudioBuffer const *inputBuffer = &source->mBuffers[0];
+    void *coreAudioBuffer = inputBuffer->mData;
+    void *portAudioBuffer = destination->inputBuffer;
+    UInt32 i, streamNumber, streamChannel;
+    for (i = streamNumber = streamChannel = 0; i < destination->inputChannelCount; ++i, ++streamChannel) {
+        if (streamChannel >= inputBuffer->mNumberChannels) {
+            ++streamNumber;
+            inputBuffer = &source->mBuffers[streamNumber];
+            coreAudioBuffer = inputBuffer->mData;
+            streamChannel = 0;
+        }
+        destination->inputConverter(portAudioBuffer, frameSpacing, coreAudioBuffer, inputBuffer->mNumberChannels, frameCount, destination->ditherGenerator);
+        coreAudioBuffer += sizeof(Float32);
+        portAudioBuffer += Pa_GetSampleSize(destination->inputSampleFormat) * channelSpacing;
+    }
+    return noErr;
+}
+
+static OSStatus CopyOutputData(AudioBufferList* destination, PaMacClientData *source, unsigned long frameCount)
+{
+    int frameSpacing, channelSpacing;
+    if (source->outputSampleFormat & paNonInterleaved) {
+        frameSpacing = 1;
+        channelSpacing = source->outputChannelCount;
+    }
+    else {
+        frameSpacing = source->outputChannelCount;
+        channelSpacing = 1;
+    }
+    
+    AudioBuffer *outputBuffer = &destination->mBuffers[0];
+    void *coreAudioBuffer = outputBuffer->mData;
+    void *portAudioBuffer = source->outputBuffer;
+    UInt32 i, streamNumber, streamChannel;
+    for (i = streamNumber = streamChannel = 0; i < source->outputChannelCount; ++i, ++streamChannel) {
+        if (streamChannel >= outputBuffer->mNumberChannels) {
+            ++streamNumber;
+            outputBuffer = &destination->mBuffers[streamNumber];
+            coreAudioBuffer = outputBuffer->mData;
+            streamChannel = 0;
+        }
+        source->outputConverter(coreAudioBuffer, outputBuffer->mNumberChannels, portAudioBuffer, frameSpacing, frameCount, NULL);
+        coreAudioBuffer += sizeof(Float32);
+        portAudioBuffer += Pa_GetSampleSize(source->outputSampleFormat) * channelSpacing;
+    }
+    return noErr;
+}
+
+static OSStatus AudioIOProc( AudioDeviceID inDevice,
+                      const AudioTimeStamp* inNow,
+                      const AudioBufferList* inInputData,
+                      const AudioTimeStamp* inInputTime,
+                      AudioBufferList* outOutputData, 
+                      const AudioTimeStamp* inOutputTime,
+                      void* inClientData)
+{
+    PaMacClientData *clientData = (PaMacClientData *)inClientData;
+    PaStreamCallbackTimeInfo *timeInfo = InitializeTimeInfo(inNow, inInputTime, inOutputTime);
+    
+    PaUtil_BeginCpuLoadMeasurement( &clientData->stream->cpuLoadMeasurer );
+    
+    AudioBuffer *outputBuffer = &outOutputData->mBuffers[0];
+    unsigned long frameCount = outputBuffer->mDataByteSize / (outputBuffer->mNumberChannels * sizeof(Float32));
+
+    if (clientData->inputBuffer) {
+        CopyInputData(clientData, inInputData, frameCount);
+    }
+    PaStreamCallbackResult result = clientData->callback(clientData->inputBuffer, clientData->outputBuffer, frameCount, timeInfo, paNoFlag, clientData->userData);
+    if (clientData->outputBuffer) {
+        CopyOutputData(outOutputData, clientData, frameCount);
+    }
+
+    PaUtil_EndCpuLoadMeasurement( &clientData->stream->cpuLoadMeasurer, frameCount );
+    
+    if (result == paComplete || result == paAbort) {
+        Pa_StopStream(clientData->stream);
+    }
+
+    PaUtil_FreeMemory( timeInfo );
+    return noErr;
+}
+
+// This is not for input-only streams, this is for streams where the input device is different from the output device
+static OSStatus AudioInputProc( AudioDeviceID inDevice,
+                         const AudioTimeStamp* inNow,
+                         const AudioBufferList* inInputData,
+                         const AudioTimeStamp* inInputTime,
+                         AudioBufferList* outOutputData, 
+                         const AudioTimeStamp* inOutputTime,
+                         void* inClientData)
+{
+    PaMacClientData *clientData = (PaMacClientData *)inClientData;
+    PaStreamCallbackTimeInfo *timeInfo = InitializeTimeInfo(inNow, inInputTime, inOutputTime);
+
+    PaUtil_BeginCpuLoadMeasurement( &clientData->stream->cpuLoadMeasurer );
+
+    AudioBuffer const *inputBuffer = &inInputData->mBuffers[0];
+    unsigned long frameCount = inputBuffer->mDataByteSize / (inputBuffer->mNumberChannels * sizeof(Float32));
+
+    CopyInputData(clientData, inInputData, frameCount);
+    PaStreamCallbackResult result = clientData->callback(clientData->inputBuffer, clientData->outputBuffer, frameCount, timeInfo, paNoFlag, clientData->userData);
+    
+    PaUtil_EndCpuLoadMeasurement( &clientData->stream->cpuLoadMeasurer, frameCount );
+    if( result == paComplete || result == paAbort )
+       Pa_StopStream(clientData->stream);
+    PaUtil_FreeMemory( timeInfo );
+    return noErr;
+}
+
+// This is not for output-only streams, this is for streams where the input device is different from the output device
+static OSStatus AudioOutputProc( AudioDeviceID inDevice,
+                          const AudioTimeStamp* inNow,
+                          const AudioBufferList* inInputData,
+                          const AudioTimeStamp* inInputTime,
+                          AudioBufferList* outOutputData, 
+                          const AudioTimeStamp* inOutputTime,
+                          void* inClientData)
+{
+    PaMacClientData *clientData = (PaMacClientData *)inClientData;
+    //PaStreamCallbackTimeInfo *timeInfo = InitializeTimeInfo(inNow, inInputTime, inOutputTime);
+
+    PaUtil_BeginCpuLoadMeasurement( &clientData->stream->cpuLoadMeasurer );
+
+    AudioBuffer *outputBuffer = &outOutputData->mBuffers[0];
+    unsigned long frameCount = outputBuffer->mDataByteSize / (outputBuffer->mNumberChannels * sizeof(Float32));
+
+    //clientData->callback(NULL, clientData->outputBuffer, frameCount, timeInfo, paNoFlag, clientData->userData);
+
+    CopyOutputData(outOutputData, clientData, frameCount);
+
+    PaUtil_EndCpuLoadMeasurement( &clientData->stream->cpuLoadMeasurer, frameCount );
+    return noErr;
+}
+
+static PaError SetSampleRate(AudioDeviceID device, double sampleRate, int isInput)
+{
+    PaError result = paNoError;
+    
+    double actualSampleRate;
+    UInt32 propSize = sizeof(double);
+    result = conv_err(AudioDeviceSetProperty(device, NULL, 0, isInput, kAudioDevicePropertyNominalSampleRate, propSize, &sampleRate));
+    
+    result = conv_err(AudioDeviceGetProperty(device, 0, isInput, kAudioDevicePropertyNominalSampleRate, &propSize, &actualSampleRate));
+    
+    if (result == paNoError && actualSampleRate != sampleRate) {
+        result = paInvalidSampleRate;
+    }
+    
+    return result;    
+}
+
+static PaError SetFramesPerBuffer(AudioDeviceID device, unsigned long framesPerBuffer, int isInput)
+{
+    PaError result = paNoError;
+    UInt32 preferredFramesPerBuffer = framesPerBuffer;
+    //    while (preferredFramesPerBuffer > UINT32_MAX) {
+    //        preferredFramesPerBuffer /= 2;
+    //    }
+    
+    UInt32 actualFramesPerBuffer;
+    UInt32 propSize = sizeof(UInt32);
+    result = conv_err(AudioDeviceSetProperty(device, NULL, 0, isInput, kAudioDevicePropertyBufferFrameSize, propSize, &preferredFramesPerBuffer));
+    
+    result = conv_err(AudioDeviceGetProperty(device, 0, isInput, kAudioDevicePropertyBufferFrameSize, &propSize, &actualFramesPerBuffer));
+    
+    if (result != paNoError) {
+        // do nothing
+    }
+    else if (actualFramesPerBuffer > framesPerBuffer) {
+        result = paBufferTooSmall;
+    }
+    else if (actualFramesPerBuffer < framesPerBuffer) {
+        result = paBufferTooBig;
+    }
+    
+    return result;    
+}
+    
+static PaError SetUpUnidirectionalStream(AudioDeviceID device, double sampleRate, unsigned long framesPerBuffer, int isInput)
+{
+    PaError err = paNoError;
+    err = SetSampleRate(device, sampleRate, isInput);
+    if( err == paNoError )
+        err = SetFramesPerBuffer(device, framesPerBuffer, isInput);
+    return err;
+}
+
+// =====  PortAudio functions  =====
+#pragma mark PortAudio functions
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif // __cplusplus
+    
+    PaError PaMacCore_Initialize( PaUtilHostApiRepresentation **hostApi, PaHostApiIndex index );
+    
+#ifdef __cplusplus
+}
+#endif // __cplusplus
+
+static void Terminate( struct PaUtilHostApiRepresentation *hostApi )
+{
+    PaMacCoreHostApiRepresentation *macCoreHostApi = (PaMacCoreHostApiRepresentation*)hostApi;
+    
+    CleanUp(macCoreHostApi);
+}
+
+static PaError IsFormatSupported( struct PaUtilHostApiRepresentation *hostApi,
+                                  const PaStreamParameters *inputParameters,
+                                  const PaStreamParameters *outputParameters,
+                                  double sampleRate )
+{
+    PaMacCoreHostApiRepresentation *macCoreHostApi = (PaMacCoreHostApiRepresentation*)hostApi;
+    PaDeviceInfo *deviceInfo;
+    
+    PaError result = paNoError;
+    if (inputParameters) {
+        deviceInfo = macCoreHostApi->inheritedHostApiRep.deviceInfos[inputParameters->device];
+        if (inputParameters->channelCount > deviceInfo->maxInputChannels)
+            result = paInvalidChannelCount;
+        else if (CheckFormat(macCoreHostApi->macCoreDeviceIds[inputParameters->device], inputParameters, sampleRate, 1) != kAudioHardwareNoError) {
+            result = paInvalidSampleRate;
+        }
+    }
+    if (outputParameters && result == paNoError) {
+        deviceInfo = macCoreHostApi->inheritedHostApiRep.deviceInfos[outputParameters->device];
+        if (outputParameters->channelCount > deviceInfo->maxOutputChannels)
+            result = paInvalidChannelCount;
+        else if (CheckFormat(macCoreHostApi->macCoreDeviceIds[outputParameters->device], outputParameters, sampleRate, 0) != kAudioHardwareNoError) {
+            result = paInvalidSampleRate;
+        }
+    }
+
+    return result;
+}
+
+static PaError OpenStream( struct PaUtilHostApiRepresentation *hostApi,
+                           PaStream** s,
+                           const PaStreamParameters *inputParameters,
+                           const PaStreamParameters *outputParameters,
+                           double sampleRate,
+                           unsigned long framesPerBuffer,
+                           PaStreamFlags streamFlags,
+                           PaStreamCallback *streamCallback,
+                           void *userData )
+{
+    PaError err = paNoError;
+    PaMacCoreHostApiRepresentation *macCoreHostApi = (PaMacCoreHostApiRepresentation *)hostApi;
+    PaMacCoreStream *stream = PaUtil_AllocateMemory(sizeof(PaMacCoreStream));
+    stream->isActive = 0;
+    stream->isStopped = 1;
+    stream->inputDevice = kAudioDeviceUnknown;
+    stream->outputDevice = kAudioDeviceUnknown;
+    
+    PaUtil_InitializeStreamRepresentation( &stream->streamRepresentation,
+                                           ( (streamCallback)
+                                             ? &macCoreHostApi->callbackStreamInterface
+                                             : &macCoreHostApi->blockingStreamInterface ),
+                                           streamCallback, userData );
+    PaUtil_InitializeCpuLoadMeasurer( &stream->cpuLoadMeasurer, sampleRate );
+    
+    *s = (PaStream*)stream;
+    PaMacClientData *clientData = PaUtil_AllocateMemory(sizeof(PaMacClientData));
+    clientData->stream = stream;
+    clientData->callback = streamCallback;
+    clientData->userData = userData;
+    clientData->inputBuffer = 0;
+    clientData->outputBuffer = 0;
+    clientData->ditherGenerator = PaUtil_AllocateMemory(sizeof(PaUtilTriangularDitherGenerator));
+    PaUtil_InitializeTriangularDitherState(clientData->ditherGenerator);
+    
+    if (inputParameters != NULL) {
+        stream->inputDevice = macCoreHostApi->macCoreDeviceIds[inputParameters->device];
+        clientData->inputConverter = PaUtil_SelectConverter(paFloat32, inputParameters->sampleFormat, streamFlags);
+        clientData->inputBuffer = PaUtil_AllocateMemory(Pa_GetSampleSize(inputParameters->sampleFormat) * framesPerBuffer * inputParameters->channelCount);
+        clientData->inputChannelCount = inputParameters->channelCount;
+        clientData->inputSampleFormat = inputParameters->sampleFormat;
+        err = SetUpUnidirectionalStream(stream->inputDevice, sampleRate, framesPerBuffer, 1);
+    }
+    
+    if (err == paNoError && outputParameters != NULL) {
+        stream->outputDevice = macCoreHostApi->macCoreDeviceIds[outputParameters->device];
+        clientData->outputConverter = PaUtil_SelectConverter(outputParameters->sampleFormat, paFloat32, streamFlags);
+        clientData->outputBuffer = PaUtil_AllocateMemory(Pa_GetSampleSize(outputParameters->sampleFormat) * framesPerBuffer * outputParameters->channelCount);
+        clientData->outputChannelCount = outputParameters->channelCount;
+        clientData->outputSampleFormat = outputParameters->sampleFormat;
+        err = SetUpUnidirectionalStream(stream->outputDevice, sampleRate, framesPerBuffer, 0);
+    }
+
+    if (inputParameters == NULL || outputParameters == NULL || stream->inputDevice == stream->outputDevice) {
+        AudioDeviceID device = (inputParameters == NULL) ? stream->outputDevice : stream->inputDevice;
+
+        AudioDeviceAddIOProc(device, AudioIOProc, clientData);
+    }
+    else {
+        // using different devices for input and output
+        AudioDeviceAddIOProc(stream->inputDevice, AudioInputProc, clientData);
+        AudioDeviceAddIOProc(stream->outputDevice, AudioOutputProc, clientData);
+    }
+    
+    return err;
+}
+
+// Note: When CloseStream() is called, the multi-api layer ensures that the stream has already been stopped or aborted.
+static PaError CloseStream( PaStream* s )
+{
+    PaError err = paNoError;
+    PaMacCoreStream *stream = (PaMacCoreStream*)s;
+
+    PaUtil_ResetCpuLoadMeasurer( &stream->cpuLoadMeasurer );
+
+    if (stream->inputDevice != kAudioDeviceUnknown) {
+        if (stream->outputDevice == kAudioDeviceUnknown || stream->outputDevice == stream->inputDevice) {
+            err = conv_err(AudioDeviceRemoveIOProc(stream->inputDevice, AudioIOProc));
+        }
+        else {
+            err = conv_err(AudioDeviceRemoveIOProc(stream->inputDevice, AudioInputProc));
+            err = conv_err(AudioDeviceRemoveIOProc(stream->outputDevice, AudioOutputProc));
+        }
+    }
+    else {
+        err = conv_err(AudioDeviceRemoveIOProc(stream->outputDevice, AudioIOProc));
+    }
+    
+    return err;
+}
+
+
+static PaError StartStream( PaStream *s )
+{
+    PaError err = paNoError;
+    PaMacCoreStream *stream = (PaMacCoreStream*)s;
+
+    if (stream->inputDevice != kAudioDeviceUnknown) {
+        if (stream->outputDevice == kAudioDeviceUnknown || stream->outputDevice == stream->inputDevice) {
+            err = conv_err(AudioDeviceStart(stream->inputDevice, AudioIOProc));
+        }
+        else {
+            err = conv_err(AudioDeviceStart(stream->inputDevice, AudioInputProc));
+            err = conv_err(AudioDeviceStart(stream->outputDevice, AudioOutputProc));
+        }
+    }
+    else {
+        err = conv_err(AudioDeviceStart(stream->outputDevice, AudioIOProc));
+    }
+    
+    stream->isActive = 1;
+    stream->isStopped = 0;
+    return err;
+}
+
+static PaError AbortStream( PaStream *s )
+{
+    PaError err = paNoError;
+    PaMacCoreStream *stream = (PaMacCoreStream*)s;
+    
+    if (stream->inputDevice != kAudioDeviceUnknown) {
+        if (stream->outputDevice == kAudioDeviceUnknown || stream->outputDevice == stream->inputDevice) {
+            err = conv_err(AudioDeviceStop(stream->inputDevice, AudioIOProc));
+        }
+        else {
+            err = conv_err(AudioDeviceStop(stream->inputDevice, AudioInputProc));
+            err = conv_err(AudioDeviceStop(stream->outputDevice, AudioOutputProc));
+        }
+    }
+    else {
+        err = conv_err(AudioDeviceStop(stream->outputDevice, AudioIOProc));
+    }
+    
+    stream->isActive = 0;
+    stream->isStopped = 1;
+    return err;
+}    
+
+static PaError StopStream( PaStream *s )
+{
+    // TODO: this should be nicer than abort
+    return AbortStream(s);
+}
+
+static PaError IsStreamStopped( PaStream *s )
+{
+    PaMacCoreStream *stream = (PaMacCoreStream*)s;
+    
+    return stream->isStopped;
+}
+
+
+static PaError IsStreamActive( PaStream *s )
+{
+    PaMacCoreStream *stream = (PaMacCoreStream*)s;
+
+    return stream->isActive;
+}
+
+
+static PaTime GetStreamTime( PaStream *s )
+{
+    OSStatus err;
+    PaTime result;
+    PaMacCoreStream *stream = (PaMacCoreStream*)s;
+
+    AudioTimeStamp *timeStamp = PaUtil_AllocateMemory(sizeof(AudioTimeStamp));
+    if (stream->inputDevice != kAudioDeviceUnknown) {
+        err = AudioDeviceGetCurrentTime(stream->inputDevice, timeStamp);
+    }
+    else {
+        err = AudioDeviceGetCurrentTime(stream->outputDevice, timeStamp);
+    }
+    
+    result = err ? 0 : timeStamp->mSampleTime;
+    PaUtil_FreeMemory(timeStamp);
+
+    return result;
+}
+
+
+static double GetStreamCpuLoad( PaStream* s )
+{
+    PaMacCoreStream *stream = (PaMacCoreStream*)s;
+    
+    return PaUtil_GetCpuLoad( &stream->cpuLoadMeasurer );
+}
+
+
+// As separate stream interfaces are used for blocking and callback streams, the following functions can be guaranteed to only be called for blocking streams.
+
+static PaError ReadStream( PaStream* s,
+                           void *buffer,
+                           unsigned long frames )
+{
+    return paInternalError;
+}
+
+
+static PaError WriteStream( PaStream* s,
+                            const void *buffer,
+                            unsigned long frames )
+{
+    return paInternalError;
+}
+
+
+static signed long GetStreamReadAvailable( PaStream* s )
+{
+    return paInternalError;
+}
+
+
+static signed long GetStreamWriteAvailable( PaStream* s )
+{
+    return paInternalError;
+}
+
+// HostAPI-specific initialization function
+PaError PaMacCore_Initialize( PaUtilHostApiRepresentation **hostApi, PaHostApiIndex hostApiIndex )
+{
+    PaError result = paNoError;
+    PaMacCoreHostApiRepresentation *macCoreHostApi = (PaMacCoreHostApiRepresentation *)PaUtil_AllocateMemory( sizeof(PaMacCoreHostApiRepresentation) );
+    if( !macCoreHostApi )
+    {
+        result = paInsufficientMemory;
+        goto error;
+    }
+    
+    macCoreHostApi->allocations = PaUtil_CreateAllocationGroup();
+    if( !macCoreHostApi->allocations )
+    {
+        result = paInsufficientMemory;
+        goto error;
+    }
+    
+    *hostApi = &macCoreHostApi->inheritedHostApiRep;
+    (*hostApi)->info.structVersion = 1;
+    (*hostApi)->info.type = paCoreAudio;
+    (*hostApi)->info.name = "CoreAudio";
+
+    result = InitializeDeviceInfos(macCoreHostApi, hostApiIndex);
+    if (result != paNoError) {
+        goto error;
+    }
+    
+    // Set up the proper callbacks to this HostApi's functions
+    (*hostApi)->Terminate = Terminate;
+    (*hostApi)->OpenStream = OpenStream;
+    (*hostApi)->IsFormatSupported = IsFormatSupported;
+    
+    PaUtil_InitializeStreamInterface( &macCoreHostApi->callbackStreamInterface, CloseStream, StartStream,
+                                      StopStream, AbortStream, IsStreamStopped, IsStreamActive,
+                                      GetStreamTime, GetStreamCpuLoad,
+                                      PaUtil_DummyRead, PaUtil_DummyWrite,
+                                      PaUtil_DummyGetReadAvailable, PaUtil_DummyGetWriteAvailable );
+    
+    PaUtil_InitializeStreamInterface( &macCoreHostApi->blockingStreamInterface, CloseStream, StartStream,
+                                      StopStream, AbortStream, IsStreamStopped, IsStreamActive,
+                                      GetStreamTime, PaUtil_DummyGetCpuLoad,
+                                      ReadStream, WriteStream, GetStreamReadAvailable, GetStreamWriteAvailable );
+    
+    return result;
+    
+error:
+        if( macCoreHostApi ) {
+            CleanUp(macCoreHostApi);
+        }
+    
+    return result;
+}
diff --git a/pjmedia/src/pjmedia/portaudio/pa_mac_core_utilities.c b/pjmedia/src/pjmedia/portaudio/pa_mac_core_utilities.c
new file mode 100644
index 00000000..5a0f8b8d
--- /dev/null
+++ b/pjmedia/src/pjmedia/portaudio/pa_mac_core_utilities.c
@@ -0,0 +1,612 @@
+/*
+ * Helper and utility functions for pa_mac_core.c (Apple AUHAL implementation)
+ *
+ * PortAudio Portable Real-Time Audio Library
+ * Latest Version at: http://www.portaudio.com
+ *
+ * Written by Bjorn Roche of XO Audio LLC, from PA skeleton code.
+ * Portions copied from code by Dominic Mazzoni (who wrote a HAL implementation)
+ *
+ * Dominic's code was based on code by Phil Burk, Darren Gibbs,
+ * Gord Peters, Stephane Letz, and Greg Pfiel.
+ *
+ * The following people also deserve acknowledgements:
+ *
+ * Olivier Tristan for feedback and testing
+ * Glenn Zelniker and Z-Systems engineering for sponsoring the Blocking I/O
+ * interface.
+ * 
+ *
+ * Based on the Open Source API proposed by Ross Bencina
+ * Copyright (c) 1999-2002 Ross Bencina, Phil Burk
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files
+ * (the "Software"), to deal in the Software without restriction,
+ * including without limitation the rights to use, copy, modify, merge,
+ * publish, distribute, sublicense, and/or sell copies of the Software,
+ * and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+/*
+ * The text above constitutes the entire PortAudio license; however, 
+ * the PortAudio community also makes the following non-binding requests:
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version. It is also 
+ * requested that these non-binding requests be included along with the 
+ * license above.
+ */
+
+/**
+ @file
+ @ingroup hostapi_src
+*/
+
+#include "pa_mac_core_utilities.h"
+
+
+PaError PaMacCore_SetUnixError( int err, int line )
+{
+   PaError ret;
+   const char *errorText;
+
+   if( err == 0 )
+   {
+      return paNoError;
+   }
+
+   ret = paNoError;
+   errorText = strerror( err );
+
+   /** Map Unix error to PaError. Pretty much the only one that maps
+       is ENOMEM. */
+   if( err == ENOMEM )
+      ret = paInsufficientMemory;
+   else
+      ret = paInternalError;
+
+   DBUG(("%d on line %d: msg='%s'\n", err, line, errorText));
+   PaUtil_SetLastHostErrorInfo( paCoreAudio, err, errorText );
+
+   return ret;
+}
+
+/*
+ * Translates MacOS generated errors into PaErrors
+ */
+PaError PaMacCore_SetError(OSStatus error, int line, int isError)
+{
+    /*FIXME: still need to handle possible ComponentResult values.*/
+    /*       unfortunately, they don't seem to be documented anywhere.*/
+    PaError result;
+    const char *errorType; 
+    const char *errorText;
+    
+    switch (error) {
+    case kAudioHardwareNoError:
+        return paNoError;
+    case kAudioHardwareNotRunningError:
+        errorText = "Audio Hardware Not Running";
+        result = paInternalError; break;
+    case kAudioHardwareUnspecifiedError: 
+        errorText = "Unspecified Audio Hardware Error";
+        result = paInternalError; break;
+    case kAudioHardwareUnknownPropertyError:
+        errorText = "Audio Hardware: Unknown Property";
+        result = paInternalError; break;
+    case kAudioHardwareBadPropertySizeError:
+        errorText = "Audio Hardware: Bad Property Size";
+        result = paInternalError; break;
+    case kAudioHardwareIllegalOperationError: 
+        errorText = "Audio Hardware: Illegal Operation";
+        result = paInternalError; break;
+    case kAudioHardwareBadDeviceError:
+        errorText = "Audio Hardware: Bad Device";
+        result = paInvalidDevice; break;
+    case kAudioHardwareBadStreamError:
+        errorText = "Audio Hardware: BadStream";
+        result = paBadStreamPtr; break;
+    case kAudioHardwareUnsupportedOperationError:
+        errorText = "Audio Hardware: Unsupported Operation";
+        result = paInternalError; break;
+    case kAudioDeviceUnsupportedFormatError:
+        errorText = "Audio Device: Unsupported Format";
+        result = paSampleFormatNotSupported; break;
+    case kAudioDevicePermissionsError:
+        errorText = "Audio Device: Permissions Error";
+        result = paDeviceUnavailable; break;
+    /* Audio Unit Errors: http://developer.apple.com/documentation/MusicAudio/Reference/CoreAudio/audio_units/chapter_5_section_3.html */
+    case kAudioUnitErr_InvalidProperty:
+        errorText = "Audio Unit: Invalid Property";
+        result = paInternalError; break;
+    case kAudioUnitErr_InvalidParameter:
+        errorText = "Audio Unit: Invalid Parameter";
+        result = paInternalError; break;
+    case kAudioUnitErr_NoConnection:
+        errorText = "Audio Unit: No Connection";
+        result = paInternalError; break;
+    case kAudioUnitErr_FailedInitialization:
+        errorText = "Audio Unit: Initialization Failed";
+        result = paInternalError; break;
+    case kAudioUnitErr_TooManyFramesToProcess:
+        errorText = "Audio Unit: Too Many Frames";
+        result = paInternalError; break;
+    case kAudioUnitErr_IllegalInstrument:
+        errorText = "Audio Unit: Illegal Instrument";
+        result = paInternalError; break;
+    case kAudioUnitErr_InstrumentTypeNotFound:
+        errorText = "Audio Unit: Instrument Type Not Found";
+        result = paInternalError; break;
+    case kAudioUnitErr_InvalidFile:
+        errorText = "Audio Unit: Invalid File";
+        result = paInternalError; break;
+    case kAudioUnitErr_UnknownFileType:
+        errorText = "Audio Unit: Unknown File Type";
+        result = paInternalError; break;
+    case kAudioUnitErr_FileNotSpecified:
+        errorText = "Audio Unit: File Not Specified";
+        result = paInternalError; break;
+    case kAudioUnitErr_FormatNotSupported:
+        errorText = "Audio Unit: Format Not Supported";
+        result = paInternalError; break;
+    case kAudioUnitErr_Uninitialized:
+        errorText = "Audio Unit: Unitialized";
+        result = paInternalError; break;
+    case kAudioUnitErr_InvalidScope:
+        errorText = "Audio Unit: Invalid Scope";
+        result = paInternalError; break;
+    case kAudioUnitErr_PropertyNotWritable:
+        errorText = "Audio Unit: PropertyNotWritable";
+        result = paInternalError; break;
+    case kAudioUnitErr_InvalidPropertyValue:
+        errorText = "Audio Unit: Invalid Property Value";
+        result = paInternalError; break;
+    case kAudioUnitErr_PropertyNotInUse:
+        errorText = "Audio Unit: Property Not In Use";
+        result = paInternalError; break;
+    case kAudioUnitErr_Initialized:
+        errorText = "Audio Unit: Initialized";
+        result = paInternalError; break;
+    case kAudioUnitErr_InvalidOfflineRender:
+        errorText = "Audio Unit: Invalid Offline Render";
+        result = paInternalError; break;
+    case kAudioUnitErr_Unauthorized:
+        errorText = "Audio Unit: Unauthorized";
+        result = paInternalError; break;
+    case kAudioUnitErr_CannotDoInCurrentContext:
+        errorText = "Audio Unit: cannot do in current context";
+        result = paInternalError; break;
+    default:
+        errorText = "Unknown Error";
+        result = paInternalError;
+    }
+
+    if (isError)
+        errorType = "Error";
+    else
+        errorType = "Warning";
+
+    if ((int)error < -99999 || (int)error > 99999)
+        DBUG(("%s on line %d: err='%4s', msg='%s'\n", errorType, line, (const char *)&error, errorText));
+    else
+        DBUG(("%s on line %d: err=%d, 0x%x, msg='%s'\n", errorType, line, (int)error, (unsigned)error, errorText));
+
+    PaUtil_SetLastHostErrorInfo( paCoreAudio, error, errorText );
+
+    return result;
+}
+
+/*
+ * This function computes an appropriate ring buffer size given
+ * a requested latency (in seconds), sample rate and framesPerBuffer.
+ *
+ * The returned ringBufferSize is computed using the following
+ * constraints:
+ *   - it must be at least 4.
+ *   - it must be at least 3x framesPerBuffer.
+ *   - it must be at least 2x the suggestedLatency.
+ *   - it must be a power of 2.
+ * This function attempts to compute the minimum such size.
+ *
+ * FEEDBACK: too liberal/conservative/another way?
+ */
+long computeRingBufferSize( const PaStreamParameters *inputParameters,
+                                   const PaStreamParameters *outputParameters,
+                                   long inputFramesPerBuffer,
+                                   long outputFramesPerBuffer,
+                                   double sampleRate )
+{
+   long ringSize;
+   int index;
+   int i;
+   double latencyTimesChannelCount ;
+   long framesPerBufferTimesChannelCount ;
+
+   VVDBUG(( "computeRingBufferSize()\n" ));
+
+   assert( inputParameters || outputParameters );
+
+   if( outputParameters && inputParameters )
+   {
+      latencyTimesChannelCount = MAX(
+           inputParameters->suggestedLatency * inputParameters->channelCount,
+           outputParameters->suggestedLatency * outputParameters->channelCount );
+      framesPerBufferTimesChannelCount = MAX(
+           inputFramesPerBuffer * inputParameters->channelCount,
+           outputFramesPerBuffer * outputParameters->channelCount );
+   } 
+   else if( outputParameters )
+   {
+      latencyTimesChannelCount
+                = outputParameters->suggestedLatency * outputParameters->channelCount;
+      framesPerBufferTimesChannelCount
+                = outputFramesPerBuffer * outputParameters->channelCount;
+   }
+   else /* we have inputParameters  */
+   {
+      latencyTimesChannelCount
+                = inputParameters->suggestedLatency * inputParameters->channelCount;
+      framesPerBufferTimesChannelCount
+                = inputFramesPerBuffer * inputParameters->channelCount;
+   }
+
+   ringSize = (long) ( latencyTimesChannelCount * sampleRate * 2 + .5);
+   VDBUG( ( "suggested latency * channelCount: %d\n", (int) (latencyTimesChannelCount*sampleRate) ) );
+   if( ringSize < framesPerBufferTimesChannelCount * 3 )
+      ringSize = framesPerBufferTimesChannelCount * 3 ;
+   VDBUG(("framesPerBuffer*channelCount:%d\n",(int)framesPerBufferTimesChannelCount));
+   VDBUG(("Ringbuffer size (1): %d\n", (int)ringSize ));
+
+   /* make sure it's at least 4 */
+   ringSize = MAX( ringSize, 4 );
+
+   /* round up to the next power of 2 */
+   index = -1;
+   for( i=0; i<sizeof(long)*8; ++i )
+      if( ringSize >> i & 0x01 )
+         index = i;
+   assert( index > 0 );
+   if( ringSize <= ( 0x01 << index ) )
+      ringSize = 0x01 << index ;
+   else
+      ringSize = 0x01 << ( index + 1 );
+
+   VDBUG(( "Final Ringbuffer size (2): %d\n", (int)ringSize ));
+   return ringSize;
+}
+
+
+/*
+ * Durring testing of core audio, I found that serious crashes could occur
+ * if properties such as sample rate were changed multiple times in rapid
+ * succession. The function below has some fancy logic to make sure that changes
+ * are acknowledged before another is requested. That seems to help a lot.
+ */
+
+OSStatus propertyProc(
+    AudioDeviceID inDevice, 
+    UInt32 inChannel, 
+    Boolean isInput, 
+    AudioDevicePropertyID inPropertyID, 
+    void* inClientData )
+{
+   MutexAndBool *mab = (MutexAndBool *) inClientData;
+   mab->once = TRUE;
+   pthread_mutex_unlock( &(mab->mutex) );
+   return noErr;
+}
+
+/* sets the value of the given property and waits for the change to 
+   be acknowledged, and returns the final value, which is not guaranteed
+   by this function to be the same as the desired value. Obviously, this
+   function can only be used for data whose input and output are the
+   same size and format, and their size and format are known in advance.*/
+PaError AudioDeviceSetPropertyNowAndWaitForChange(
+    AudioDeviceID inDevice,
+    UInt32 inChannel, 
+    Boolean isInput, 
+    AudioDevicePropertyID inPropertyID,
+    UInt32 inPropertyDataSize, 
+    const void *inPropertyData,
+    void *outPropertyData )
+{
+   OSStatus macErr;
+   int unixErr;
+   MutexAndBool mab;
+   UInt32 outPropertyDataSize = inPropertyDataSize;
+
+   /* First, see if it already has that value. If so, return. */
+   macErr = AudioDeviceGetProperty( inDevice, inChannel,
+                                 isInput, inPropertyID, 
+                                 &outPropertyDataSize, outPropertyData );
+   if( macErr )
+      goto failMac2;
+   if( inPropertyDataSize!=outPropertyDataSize )
+      return paInternalError;
+   if( 0==memcmp( outPropertyData, inPropertyData, outPropertyDataSize ) )
+      return paNoError;
+
+   /* setup and lock mutex */
+   mab.once = FALSE;
+   unixErr = pthread_mutex_init( &mab.mutex, NULL );
+   if( unixErr )
+      goto failUnix2;
+   unixErr = pthread_mutex_lock( &mab.mutex );
+   if( unixErr )
+      goto failUnix;
+
+   /* add property listener */
+   macErr = AudioDeviceAddPropertyListener( inDevice, inChannel, isInput,
+                                   inPropertyID, propertyProc,
+                                   &mab ); 
+   if( macErr )
+      goto failMac;
+   /* set property */
+   macErr  = AudioDeviceSetProperty( inDevice, NULL, inChannel,
+                                 isInput, inPropertyID,
+                                 inPropertyDataSize, inPropertyData );
+   if( macErr ) {
+      /* we couldn't set the property, so we'll just unlock the mutex
+         and move on. */
+      pthread_mutex_unlock( &mab.mutex );
+   }
+
+   /* wait for property to change */                      
+   unixErr = pthread_mutex_lock( &mab.mutex );
+   if( unixErr )
+      goto failUnix;
+
+   /* now read the property back out */
+   macErr = AudioDeviceGetProperty( inDevice, inChannel,
+                                 isInput, inPropertyID, 
+                                 &outPropertyDataSize, outPropertyData );
+   if( macErr )
+      goto failMac;
+   /* cleanup */
+   AudioDeviceRemovePropertyListener( inDevice, inChannel, isInput,
+                                      inPropertyID, propertyProc );
+   unixErr = pthread_mutex_unlock( &mab.mutex );
+   if( unixErr )
+      goto failUnix2;
+   unixErr = pthread_mutex_destroy( &mab.mutex );
+   if( unixErr )
+      goto failUnix2;
+
+   return paNoError;
+
+ failUnix:
+   pthread_mutex_destroy( &mab.mutex );
+   AudioDeviceRemovePropertyListener( inDevice, inChannel, isInput,
+                                      inPropertyID, propertyProc );
+
+ failUnix2:
+   DBUG( ("Error #%d while setting a device property: %s\n", unixErr, strerror( unixErr ) ) );
+   return paUnanticipatedHostError;
+
+ failMac:
+   pthread_mutex_destroy( &mab.mutex );
+   AudioDeviceRemovePropertyListener( inDevice, inChannel, isInput,
+                                      inPropertyID, propertyProc );
+ failMac2:
+   return ERR( macErr );
+}
+
+/*
+ * Sets the sample rate the HAL device.
+ * if requireExact: set the sample rate or fail.
+ *
+ * otherwise      : set the exact sample rate.
+ *             If that fails, check for available sample rates, and choose one
+ *             higher than the requested rate. If there isn't a higher one,
+ *             just use the highest available.
+ */
+PaError setBestSampleRateForDevice( const AudioDeviceID device,
+                                    const bool isOutput,
+                                    const bool requireExact,
+                                    const Float64 desiredSrate )
+{
+   /*FIXME: changing the sample rate is disruptive to other programs using the
+            device, so it might be good to offer a custom flag to not change the
+            sample rate and just do conversion. (in my casual tests, there is
+            no disruption unless the sample rate really does need to change) */
+   const bool isInput = isOutput ? 0 : 1;
+   Float64 srate;
+   UInt32 propsize = sizeof( Float64 );
+   OSErr err;
+   AudioValueRange *ranges;
+   int i=0;
+   Float64 max  = -1; /*the maximum rate available*/
+   Float64 best = -1; /*the lowest sample rate still greater than desired rate*/
+   VDBUG(("Setting sample rate for device %ld to %g.\n",device,(float)desiredSrate));
+
+   /* -- try setting the sample rate -- */
+   err = AudioDeviceSetPropertyNowAndWaitForChange(
+                                 device, 0, isInput,
+                                 kAudioDevicePropertyNominalSampleRate,
+                                 propsize, &desiredSrate, &srate );
+   if( err )
+      return err;
+
+   /* -- if the rate agrees, and we got no errors, we are done -- */
+   if( !err && srate == desiredSrate )
+      return paNoError;
+   /* -- we've failed if the rates disagree and we are setting input -- */
+   if( requireExact )
+      return paInvalidSampleRate;
+
+   /* -- generate a list of available sample rates -- */
+   err = AudioDeviceGetPropertyInfo( device, 0, isInput,
+                                kAudioDevicePropertyAvailableNominalSampleRates,
+                                &propsize, NULL );
+   if( err )
+      return ERR( err );
+   ranges = (AudioValueRange *)calloc( 1, propsize );
+   if( !ranges )
+      return paInsufficientMemory;
+   err = AudioDeviceGetProperty( device, 0, isInput,
+                                kAudioDevicePropertyAvailableNominalSampleRates,
+                                &propsize, ranges );
+   if( err )
+   {
+      free( ranges );
+      return ERR( err );
+   }
+   VDBUG(("Requested sample rate of %g was not available.\n", (float)desiredSrate));
+   VDBUG(("%lu Available Sample Rates are:\n",propsize/sizeof(AudioValueRange)));
+#ifdef MAC_CORE_VERBOSE_DEBUG
+   for( i=0; i<propsize/sizeof(AudioValueRange); ++i )
+      VDBUG( ("\t%g-%g\n",
+              (float) ranges[i].mMinimum,
+              (float) ranges[i].mMaximum ) );
+#endif
+   VDBUG(("-----\n"));
+   
+   /* -- now pick the best available sample rate -- */
+   for( i=0; i<propsize/sizeof(AudioValueRange); ++i )
+   {
+      if( ranges[i].mMaximum > max ) max = ranges[i].mMaximum;
+      if( ranges[i].mMinimum > desiredSrate ) {
+         if( best < 0 )
+            best = ranges[i].mMinimum;
+         else if( ranges[i].mMinimum < best )
+            best = ranges[i].mMinimum;
+      }
+   }
+   if( best < 0 )
+      best = max;
+   VDBUG( ("Maximum Rate %g. best is %g.\n", max, best ) );
+   free( ranges );
+
+   /* -- set the sample rate -- */
+   propsize = sizeof( best );
+   err = AudioDeviceSetPropertyNowAndWaitForChange(
+                                 device, 0, isInput,
+                                 kAudioDevicePropertyNominalSampleRate,
+                                 propsize, &best, &srate );
+   if( err )
+      return err;
+
+   if( err )
+      return ERR( err );
+   /* -- if the set rate matches, we are done -- */
+   if( srate == best )
+      return paNoError;
+
+   /* -- otherwise, something wierd happened: we didn't set the rate, and we got no errors. Just bail. */
+   return paInternalError;
+}
+
+
+/*
+   Attempts to set the requestedFramesPerBuffer. If it can't set the exact
+   value, it settles for something smaller if available. If nothing smaller
+   is available, it uses the smallest available size.
+   actualFramesPerBuffer will be set to the actual value on successful return.
+   OK to pass NULL to actualFramesPerBuffer.
+   The logic is very simmilar too setBestSampleRate only failure here is
+   not usually catastrophic.
+*/
+PaError setBestFramesPerBuffer( const AudioDeviceID device,
+                                       const bool isOutput,
+                                       unsigned long requestedFramesPerBuffer, 
+                                       unsigned long *actualFramesPerBuffer )
+{
+   UInt32 afpb;
+   const bool isInput = !isOutput;
+   UInt32 propsize = sizeof(UInt32);
+   OSErr err;
+   Float64 min  = -1; /*the min blocksize*/
+   Float64 best = -1; /*the best blocksize*/
+   int i=0;
+   AudioValueRange *ranges;
+
+   if( actualFramesPerBuffer == NULL )
+      actualFramesPerBuffer = &afpb;
+
+
+   /* -- try and set exact FPB -- */
+   err = AudioDeviceSetProperty( device, NULL, 0, isInput,
+                                 kAudioDevicePropertyBufferFrameSize,
+                                 propsize, &requestedFramesPerBuffer);
+   err = AudioDeviceGetProperty( device, 0, isInput,
+                           kAudioDevicePropertyBufferFrameSize,
+                           &propsize, actualFramesPerBuffer);
+   if( err )
+      return ERR( err );
+   if( *actualFramesPerBuffer == requestedFramesPerBuffer )
+      return paNoError; /* we are done */
+
+   /* -- fetch available block sizes -- */
+   err = AudioDeviceGetPropertyInfo( device, 0, isInput,
+                           kAudioDevicePropertyBufferSizeRange,
+                           &propsize, NULL );
+   if( err )
+      return ERR( err );
+   ranges = (AudioValueRange *)calloc( 1, propsize );
+   if( !ranges )
+      return paInsufficientMemory;
+   err = AudioDeviceGetProperty( device, 0, isInput,
+                                kAudioDevicePropertyBufferSizeRange,
+                                &propsize, ranges );
+   if( err )
+   {
+      free( ranges );
+      return ERR( err );
+   }
+   VDBUG(("Requested block size of %lu was not available.\n",
+          requestedFramesPerBuffer ));
+   VDBUG(("%lu Available block sizes are:\n",propsize/sizeof(AudioValueRange)));
+#ifdef MAC_CORE_VERBOSE_DEBUG
+   for( i=0; i<propsize/sizeof(AudioValueRange); ++i )
+      VDBUG( ("\t%g-%g\n",
+              (float) ranges[i].mMinimum,
+              (float) ranges[i].mMaximum ) );
+#endif
+   VDBUG(("-----\n"));
+   
+   /* --- now pick the best available framesPerBuffer -- */
+   for( i=0; i<propsize/sizeof(AudioValueRange); ++i )
+   {
+      if( min == -1 || ranges[i].mMinimum < min ) min = ranges[i].mMinimum;
+      if( ranges[i].mMaximum < requestedFramesPerBuffer ) {
+         if( best < 0 )
+            best = ranges[i].mMaximum;
+         else if( ranges[i].mMaximum > best )
+            best = ranges[i].mMaximum;
+      }
+   }
+   if( best == -1 )
+      best = min;
+   VDBUG( ("Minimum FPB  %g. best is %g.\n", min, best ) );
+   free( ranges );
+
+   /* --- set the buffer size (ignore errors) -- */
+   requestedFramesPerBuffer = (UInt32) best ;
+   propsize = sizeof( UInt32 );
+   err = AudioDeviceSetProperty( device, NULL, 0, isInput,
+                                 kAudioDevicePropertyBufferSize,
+                                 propsize, &requestedFramesPerBuffer );
+   /* --- read the property to check that it was set -- */
+   err = AudioDeviceGetProperty( device, 0, isInput,
+                                 kAudioDevicePropertyBufferSize,
+                                 &propsize, actualFramesPerBuffer );
+
+   if( err )
+      return ERR( err );
+
+   return paNoError;
+}
diff --git a/pjmedia/src/pjmedia/portaudio/pa_mac_core_utilities.h b/pjmedia/src/pjmedia/portaudio/pa_mac_core_utilities.h
new file mode 100644
index 00000000..63ef22a9
--- /dev/null
+++ b/pjmedia/src/pjmedia/portaudio/pa_mac_core_utilities.h
@@ -0,0 +1,215 @@
+/*
+ * Helper and utility functions for pa_mac_core.c (Apple AUHAL implementation)
+ *
+ * PortAudio Portable Real-Time Audio Library
+ * Latest Version at: http://www.portaudio.com
+ *
+ * Written by Bjorn Roche of XO Audio LLC, from PA skeleton code.
+ * Portions copied from code by Dominic Mazzoni (who wrote a HAL implementation)
+ *
+ * Dominic's code was based on code by Phil Burk, Darren Gibbs,
+ * Gord Peters, Stephane Letz, and Greg Pfiel.
+ *
+ * The following people also deserve acknowledgements:
+ *
+ * Olivier Tristan for feedback and testing
+ * Glenn Zelniker and Z-Systems engineering for sponsoring the Blocking I/O
+ * interface.
+ * 
+ *
+ * Based on the Open Source API proposed by Ross Bencina
+ * Copyright (c) 1999-2002 Ross Bencina, Phil Burk
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining
+ * a copy of this software and associated documentation files
+ * (the "Software"), to deal in the Software without restriction,
+ * including without limitation the rights to use, copy, modify, merge,
+ * publish, distribute, sublicense, and/or sell copies of the Software,
+ * and to permit persons to whom the Software is furnished to do so,
+ * subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be
+ * included in all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
+ * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
+ * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
+ * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
+ * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
+ * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
+ * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+/*
+ * The text above constitutes the entire PortAudio license; however, 
+ * the PortAudio community also makes the following non-binding requests:
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version. It is also 
+ * requested that these non-binding requests be included along with the 
+ * license above.
+ */
+
+/**
+ @file
+ @ingroup hostapi_src
+*/
+
+#ifndef PA_MAC_CORE_UTILITIES_H__
+#define PA_MAC_CORE_UTILITIES_H__
+
+#include <pthread.h>
+#include "portaudio.h"
+#include "pa_util.h"
+#include <AudioUnit/AudioUnit.h>
+#include <AudioToolbox/AudioToolbox.h>
+
+#ifndef MIN
+#define MIN(a, b)  (((a)<(b))?(a):(b))
+#endif
+
+#ifndef MAX
+#define MAX(a, b)  (((a)<(b))?(b):(a))
+#endif
+
+#define ERR(mac_error) PaMacCore_SetError(mac_error, __LINE__, 1 ) 
+#define WARNING(mac_error) PaMacCore_SetError(mac_error, __LINE__, 0 )
+
+
+/* Help keep track of AUHAL element numbers */
+#define INPUT_ELEMENT  (1)
+#define OUTPUT_ELEMENT (0)
+
+/* Normal level of debugging: fine for most apps that don't mind the occational warning being printf'ed */
+/*
+ */
+#define MAC_CORE_DEBUG
+#ifdef MAC_CORE_DEBUG
+# define DBUG(MSG) do { printf("||PaMacCore (AUHAL)|| "); printf MSG ; fflush(stdout); } while(0)
+#else
+# define DBUG(MSG)
+#endif
+
+/* Verbose Debugging: useful for developement */
+/*
+#define MAC_CORE_VERBOSE_DEBUG
+*/
+#ifdef MAC_CORE_VERBOSE_DEBUG
+# define VDBUG(MSG) do { printf("||PaMacCore (v )|| "); printf MSG ; fflush(stdout); } while(0)
+#else
+# define VDBUG(MSG)
+#endif
+
+/* Very Verbose Debugging: Traces every call. */
+/*
+#define MAC_CORE_VERY_VERBOSE_DEBUG
+ */
+#ifdef MAC_CORE_VERY_VERBOSE_DEBUG
+# define VVDBUG(MSG) do { printf("||PaMacCore (vv)|| "); printf MSG ; fflush(stdout); } while(0)
+#else
+# define VVDBUG(MSG)
+#endif
+
+
+
+
+
+#define UNIX_ERR(err) PaMacCore_SetUnixError( err, __LINE__ )
+
+PaError PaMacCore_SetUnixError( int err, int line );
+
+/*
+ * Translates MacOS generated errors into PaErrors
+ */
+PaError PaMacCore_SetError(OSStatus error, int line, int isError);
+
+/*
+ * This function computes an appropriate ring buffer size given
+ * a requested latency (in seconds), sample rate and framesPerBuffer.
+ *
+ * The returned ringBufferSize is computed using the following
+ * constraints:
+ *   - it must be at least 4.
+ *   - it must be at least 3x framesPerBuffer.
+ *   - it must be at least 2x the suggestedLatency.
+ *   - it must be a power of 2.
+ * This function attempts to compute the minimum such size.
+ *
+ */
+long computeRingBufferSize( const PaStreamParameters *inputParameters,
+                                   const PaStreamParameters *outputParameters,
+                                   long inputFramesPerBuffer,
+                                   long outputFramesPerBuffer,
+                                   double sampleRate );
+
+/*
+ * Durring testing of core audio, I found that serious crashes could occur
+ * if properties such as sample rate were changed multiple times in rapid
+ * succession. The function below has some fancy logic to make sure that changes
+ * are acknowledged before another is requested. That seems to help a lot.
+ */
+
+typedef struct {
+   bool once; /* I didn't end up using this. bdr */
+   pthread_mutex_t mutex;
+} MutexAndBool ;
+
+OSStatus propertyProc(
+    AudioDeviceID inDevice, 
+    UInt32 inChannel, 
+    Boolean isInput, 
+    AudioDevicePropertyID inPropertyID, 
+    void* inClientData );
+
+/* sets the value of the given property and waits for the change to 
+   be acknowledged, and returns the final value, which is not guaranteed
+   by this function to be the same as the desired value. Obviously, this
+   function can only be used for data whose input and output are the
+   same size and format, and their size and format are known in advance.*/
+PaError AudioDeviceSetPropertyNowAndWaitForChange(
+    AudioDeviceID inDevice,
+    UInt32 inChannel, 
+    Boolean isInput, 
+    AudioDevicePropertyID inPropertyID,
+    UInt32 inPropertyDataSize, 
+    const void *inPropertyData,
+    void *outPropertyData );
+
+/*
+ * Sets the sample rate the HAL device.
+ * if requireExact: set the sample rate or fail.
+ *
+ * otherwise      : set the exact sample rate.
+ *             If that fails, check for available sample rates, and choose one
+ *             higher than the requested rate. If there isn't a higher one,
+ *             just use the highest available.
+ */
+PaError setBestSampleRateForDevice( const AudioDeviceID device,
+                                    const bool isOutput,
+                                    const bool requireExact,
+                                    const Float64 desiredSrate );
+/*
+   Attempts to set the requestedFramesPerBuffer. If it can't set the exact
+   value, it settles for something smaller if available. If nothing smaller
+   is available, it uses the smallest available size.
+   actualFramesPerBuffer will be set to the actual value on successful return.
+   OK to pass NULL to actualFramesPerBuffer.
+   The logic is very simmilar too setBestSampleRate only failure here is
+   not usually catastrophic.
+*/
+PaError setBestFramesPerBuffer( const AudioDeviceID device,
+                                       const bool isOutput,
+                                       unsigned long requestedFramesPerBuffer, 
+                                       unsigned long *actualFramesPerBuffer );
+
+#ifdef PA_OLD_CORE_AUDIO
+#	define kAudioUnitErr_PropertyNotInUse			10850
+#	define kAudioHardwareUnsupportedOperationError		1
+#	define kAudioUnitErr_Initialized			10849
+#	define kAudioUnitErr_InvalidOfflineRender		10848
+#	define kAudioUnitErr_Unauthorized			10847
+#endif
+
+#endif /* PA_MAC_CORE_UTILITIES_H__*/
+
diff --git a/pjmedia/src/pjmedia/portaudio/pa_mac_hostapis.c b/pjmedia/src/pjmedia/portaudio/pa_mac_hostapis.c
index f8306c30..89622d6b 100644
--- a/pjmedia/src/pjmedia/portaudio/pa_mac_hostapis.c
+++ b/pjmedia/src/pjmedia/portaudio/pa_mac_hostapis.c
@@ -1,6 +1,6 @@
 /*
  * $Id$
- * Portable Audio I/O Library Windows initialization table
+ * Portable Audio I/O Library Macintosh initialization table
  *
  * Based on the Open Source API proposed by Ross Bencina
  * Copyright (c) 1999-2002 Ross Bencina, Phil Burk
@@ -16,10 +16,6 @@
  * The above copyright notice and this permission notice shall be
  * included in all copies or substantial portions of the Software.
  *
- * Any person wishing to distribute modifications to the Software is
- * requested to send the modifications to the original developer so that
- * they can be incorporated into the canonical version.
- *
  * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
  * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
  * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
@@ -29,7 +25,19 @@
  * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
  */
 
+/*
+ * The text above constitutes the entire PortAudio license; however, 
+ * the PortAudio community also makes the following non-binding requests:
+ *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version. It is also 
+ * requested that these non-binding requests be included along with the 
+ * license above.
+ */
+
 /** @file
+ @ingroup macosx_src
 Mac OS host API initialization function table.
 */
 
diff --git a/pjmedia/src/pjmedia/portaudio/ringbuffer.c b/pjmedia/src/pjmedia/portaudio/ringbuffer.c
index 16031fef..c66ec613 100644
--- a/pjmedia/src/pjmedia/portaudio/ringbuffer.c
+++ b/pjmedia/src/pjmedia/portaudio/ringbuffer.c
@@ -4,9 +4,13 @@
  * Ring Buffer utility..
  *
  * Author: Phil Burk, http://www.softsynth.com
+ * modified for SMP safety on Mac OS X by Bjorn Roche
+ * also, alowed for const where possible
+ * Note that this is safe only for a single-thread reader and a
+ * single-thread writer.
  *
  * This program uses the PortAudio Portable Audio Library.
- * For more information see: http://www.audiomulch.com/portaudio/
+ * For more information see: http://www.portaudio.com
  * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
  *
  * Permission is hereby granted, free of charge, to any person obtaining
@@ -20,10 +24,6 @@
  * The above copyright notice and this permission notice shall be
  * included in all copies or substantial portions of the Software.
  *
- * Any person wishing to distribute modifications to the Software is
- * requested to send the modifications to the original developer so that
- * they can be incorporated into the canonical version.
- *
  * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
  * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
  * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
@@ -31,14 +31,69 @@
  * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
  * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
  * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+/*
+ * The text above constitutes the entire PortAudio license; however, 
+ * the PortAudio community also makes the following non-binding requests:
  *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version. It is also 
+ * requested that these non-binding requests be included along with the 
+ * license above.
  */
+
+/**
+ @file
+ @ingroup hostapi_src
+*/
+
 #include <stdio.h>
 #include <stdlib.h>
 #include <math.h>
 #include "ringbuffer.h"
 #include <string.h>
 
+/*
+ * We can undefine this, to turn off memory barriers, but that
+ * is only useful if we know we don't need to be MP safe or
+ * we are interested in doing some kind of tests.
+ */
+#define MPSAFE
+
+/****************
+ * First, we'll define some memory barrier primitives based on the system.
+ * right now only OS X and FreeBSD are supported. In addition to providing
+ * memory barriers, these functions should ensure that data cached in registers
+ * is written out to cache where it can be snooped by other CPUs. (ie, the volatile
+ * keyword should not be required)
+ *
+ * the primitives that must be defined are:
+ *
+ * FullMemoryBarrier()
+ * ReadMemoryBarrier()
+ * WriteMemoryBarrier()
+ *
+ ****************/
+
+#if 1
+#   define FullMemoryBarrier()
+#   define ReadMemoryBarrier()
+#   define WriteMemoryBarrier()
+#elif defined(__APPLE__) || defined(__FreeBSD__)
+#   include <libkern/OSAtomic.h>
+    /* Here are the memory barrier functions. Mac OS X and FreeBSD only provide
+       full memory barriers, so the three types of barriers are the same.
+       The asm volatile may be redundant with the memory barrier, but
+       until I have proof of that, I'm leaving it. */
+#   define FullMemoryBarrier()  do{ asm volatile("":::"memory"); OSMemoryBarrier(); }while(false)
+#   define ReadMemoryBarrier()  do{ asm volatile("":::"memory"); OSMemoryBarrier(); }while(false)
+#   define WriteMemoryBarrier() do{ asm volatile("":::"memory"); OSMemoryBarrier(); }while(false)
+#else
+#   error Memory Barriers not defined on this system or system unknown
+#endif
+
 /***************************************************************************
  * Initialize FIFO.
  * numBytes must be power of 2, returns -1 if not.
@@ -57,12 +112,16 @@ long RingBuffer_Init( RingBuffer *rbuf, long numBytes, void *dataPtr )
 ** Return number of bytes available for reading. */
 long RingBuffer_GetReadAvailable( RingBuffer *rbuf )
 {
+#ifdef MPSAFE
+    ReadMemoryBarrier();
+#endif
     return ( (rbuf->writeIndex - rbuf->readIndex) & rbuf->bigMask );
 }
 /***************************************************************************
 ** Return number of bytes available for writing. */
 long RingBuffer_GetWriteAvailable( RingBuffer *rbuf )
 {
+    /* Since we are calling RingBuffer_GetReadAvailable, we don't need an aditional MB */
     return ( rbuf->bufferSize - RingBuffer_GetReadAvailable(rbuf));
 }
 
@@ -112,7 +171,13 @@ long RingBuffer_GetWriteRegions( RingBuffer *rbuf, long numBytes,
 */
 long RingBuffer_AdvanceWriteIndex( RingBuffer *rbuf, long numBytes )
 {
+#ifdef MPSAFE
+    /* we need to ensure that previous writes are seen before we update the write index */
+    WriteMemoryBarrier();
     return rbuf->writeIndex = (rbuf->writeIndex + numBytes) & rbuf->bigMask;
+#else
+    return rbuf->writeIndex = (rbuf->writeIndex + numBytes) & rbuf->bigMask;
+#endif
 }
 
 /***************************************************************************
@@ -152,12 +217,18 @@ long RingBuffer_GetReadRegions( RingBuffer *rbuf, long numBytes,
 */
 long RingBuffer_AdvanceReadIndex( RingBuffer *rbuf, long numBytes )
 {
+#ifdef MPSAFE
+    /* we need to ensure that previous writes are always seen before updating the index. */
+    WriteMemoryBarrier();
+    return rbuf->readIndex = (rbuf->readIndex + numBytes) & rbuf->bigMask;
+#else
     return rbuf->readIndex = (rbuf->readIndex + numBytes) & rbuf->bigMask;
+#endif
 }
 
 /***************************************************************************
 ** Return bytes written. */
-long RingBuffer_Write( RingBuffer *rbuf, void *data, long numBytes )
+long RingBuffer_Write( RingBuffer *rbuf, const void *data, long numBytes )
 {
     long size1, size2, numWritten;
     void *data1, *data2;
diff --git a/pjmedia/src/pjmedia/portaudio/ringbuffer.h b/pjmedia/src/pjmedia/portaudio/ringbuffer.h
index dd769bee..816520e3 100644
--- a/pjmedia/src/pjmedia/portaudio/ringbuffer.h
+++ b/pjmedia/src/pjmedia/portaudio/ringbuffer.h
@@ -11,9 +11,13 @@ extern "C"
  * Ring Buffer utility..
  *
  * Author: Phil Burk, http://www.softsynth.com
+ * modified for SMP safety on OS X by Bjorn Roche.
+ * also allowed for const where possible.
+ * Note that this is safe only for a single-thread reader
+ * and a single-thread writer.
  *
  * This program is distributed with the PortAudio Portable Audio Library.
- * For more information see: http://www.audiomulch.com/portaudio/
+ * For more information see: http://www.portaudio.com
  * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
  *
  * Permission is hereby granted, free of charge, to any person obtaining
@@ -27,10 +31,6 @@ extern "C"
  * The above copyright notice and this permission notice shall be
  * included in all copies or substantial portions of the Software.
  *
- * Any person wishing to distribute modifications to the Software is
- * requested to send the modifications to the original developer so that
- * they can be incorporated into the canonical version.
- *
  * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
  * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
  * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
@@ -38,8 +38,24 @@ extern "C"
  * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
  * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
  * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
+ */
+
+/*
+ * The text above constitutes the entire PortAudio license; however, 
+ * the PortAudio community also makes the following non-binding requests:
  *
+ * Any person wishing to distribute modifications to the Software is
+ * requested to send the modifications to the original developer so that
+ * they can be incorporated into the canonical version. It is also 
+ * requested that these non-binding requests be included along with the 
+ * license above.
  */
+
+/**
+ @file
+ @ingroup hostapi_src
+*/
+
 #include <stdio.h>
 #include <stdlib.h>
 #include <math.h>
@@ -53,7 +69,7 @@ typedef struct
     long   readIndex;  /* Index of next readable byte. Set by RingBuffer_AdvanceReadIndex. */
     long   bigMask;    /* Used for wrapping indices with extra bit to distinguish full/empty. */
     long   smallMask;  /* Used for fitting indices to buffer. */
-    char *buffer;
+    char * buffer;
 }
 RingBuffer;
 /*
@@ -70,7 +86,7 @@ long RingBuffer_GetWriteAvailable( RingBuffer *rbuf );
 /* Return number of bytes available for read. */
 long RingBuffer_GetReadAvailable( RingBuffer *rbuf );
 /* Return bytes written. */
-long RingBuffer_Write( RingBuffer *rbuf, void *data, long numBytes );
+long RingBuffer_Write( RingBuffer *rbuf, const void *data, long numBytes );
 /* Return bytes read. */
 long RingBuffer_Read( RingBuffer *rbuf, void *data, long numBytes );