Do our own resampling

1 files changed, 0 insertions, 1125 deletions

diff --git a/lib/libsamplerate/libsamplerate-0.2.2/examples/audio_out.c b/lib/libsamplerate/libsamplerate-0.2.2/examples/audio_out.c
deleted file mode 100755
index e7c28cc5..00000000
--- a/lib/libsamplerate/libsamplerate-0.2.2/examples/audio_out.c
+++ /dev/null
@@ -1,1125 +0,0 @@
-/*
-** Copyright (c) 1999-2016, Erik de Castro Lopo <erikd@mega-nerd.com>
-** All rights reserved.
-**
-** This code is released under 2-clause BSD license. Please see the
-** file at : https://github.com/libsndfile/libsamplerate/blob/master/COPYING
-*/
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#ifdef HAVE_UNISTD_H
-#include <unistd.h>
-#endif
-
-#ifdef _WIN32
-#ifndef WIN32_LEAN_AN_MEAN
-#define WIN32_LEAN_AN_MEAN
-#endif
-#include <windows.h>
-#include <mmsystem.h>
-#endif
-
-#include "audio_out.h"
-
-#if (HAVE_SNDFILE)
-
-#include <math.h>
-
-#include <sndfile.h>
-
-#define	BUFFER_LEN		(2048)
-
-#define MAKE_MAGIC(a,b,c,d,e,f,g,h)		\
-			((a) + ((b) << 1) + ((c) << 2) + ((d) << 3) + ((e) << 4) + ((f) << 5) + ((g) << 6) + ((h) << 7))
-
-/*------------------------------------------------------------------------------
-**	Linux (ALSA and OSS) functions for playing a sound.
-*/
-
-#if defined (__linux__)
-
-#if HAVE_ALSA
-
-#define ALSA_PCM_NEW_HW_PARAMS_API
-#define ALSA_PCM_NEW_SW_PARAMS_API
-
-#include <alsa/asoundlib.h>
-#include <sys/time.h>
-
-#define	ALSA_MAGIC		MAKE_MAGIC ('L', 'n', 'x', '-', 'A', 'L', 'S', 'A')
-
-typedef struct AUDIO_OUT
-{	int magic ;
-	snd_pcm_t * dev ;
-	int channels ;
-} ALSA_AUDIO_OUT ;
-
-static int alsa_write_float (snd_pcm_t *alsa_dev, float *data, int frames, int channels) ;
-
-static AUDIO_OUT *
-alsa_open (int channels, unsigned samplerate)
-{	ALSA_AUDIO_OUT *alsa_out ;
-	const char * device = "default" ;
-	snd_pcm_hw_params_t *hw_params ;
-	snd_pcm_uframes_t buffer_size ;
-	snd_pcm_uframes_t alsa_period_size, alsa_buffer_frames ;
-	snd_pcm_sw_params_t *sw_params ;
-
-	int err ;
-
-	alsa_period_size = 1024 ;
-	alsa_buffer_frames = 4 * alsa_period_size ;
-
-	if ((alsa_out = calloc (1, sizeof (ALSA_AUDIO_OUT))) == NULL)
-	{	perror ("alsa_open : malloc ") ;
-		exit (1) ;
-		} ;
-
-	alsa_out->magic	= ALSA_MAGIC ;
-	alsa_out->channels = channels ;
-
-	if ((err = snd_pcm_open (&alsa_out->dev, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0)
-	{	fprintf (stderr, "cannot open audio device \"%s\" (%s)\n", device, snd_strerror (err)) ;
-		goto catch_error ;
-		} ;
-
-	snd_pcm_nonblock (alsa_out->dev, 0) ;
-
-	if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0)
-	{	fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n", snd_strerror (err)) ;
-		goto catch_error ;
-		} ;
-
-	if ((err = snd_pcm_hw_params_any (alsa_out->dev, hw_params)) < 0)
-	{	fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n", snd_strerror (err)) ;
-		goto catch_error ;
-		} ;
-
-	if ((err = snd_pcm_hw_params_set_access (alsa_out->dev, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
-	{	fprintf (stderr, "cannot set access type (%s)\n", snd_strerror (err)) ;
-		goto catch_error ;
-		} ;
-
-	if ((err = snd_pcm_hw_params_set_format (alsa_out->dev, hw_params, SND_PCM_FORMAT_FLOAT)) < 0)
-	{	fprintf (stderr, "cannot set sample format (%s)\n", snd_strerror (err)) ;
-		goto catch_error ;
-		} ;
-
-	if ((err = snd_pcm_hw_params_set_rate_near (alsa_out->dev, hw_params, &samplerate, 0)) < 0)
-	{	fprintf (stderr, "cannot set sample rate (%s)\n", snd_strerror (err)) ;
-		goto catch_error ;
-		} ;
-
-	if ((err = snd_pcm_hw_params_set_channels (alsa_out->dev, hw_params, channels)) < 0)
-	{	fprintf (stderr, "cannot set channel count (%s)\n", snd_strerror (err)) ;
-		goto catch_error ;
-		} ;
-
-	if ((err = snd_pcm_hw_params_set_buffer_size_near (alsa_out->dev, hw_params, &alsa_buffer_frames)) < 0)
-	{	fprintf (stderr, "cannot set buffer size (%s)\n", snd_strerror (err)) ;
-		goto catch_error ;
-		} ;
-
-	if ((err = snd_pcm_hw_params_set_period_size_near (alsa_out->dev, hw_params, &alsa_period_size, 0)) < 0)
-	{	fprintf (stderr, "cannot set period size (%s)\n", snd_strerror (err)) ;
-		goto catch_error ;
-		} ;
-
-	if ((err = snd_pcm_hw_params (alsa_out->dev, hw_params)) < 0)
-	{	fprintf (stderr, "cannot set parameters (%s)\n", snd_strerror (err)) ;
-		goto catch_error ;
-		} ;
-
-	/* extra check: if we have only one period, this code won't work */
-	snd_pcm_hw_params_get_period_size (hw_params, &alsa_period_size, 0) ;
-	snd_pcm_hw_params_get_buffer_size (hw_params, &buffer_size) ;
-	if (alsa_period_size == buffer_size)
-	{	fprintf (stderr, "Can't use period equal to buffer size (%lu == %lu)", alsa_period_size, buffer_size) ;
-		goto catch_error ;
-		} ;
-
-	snd_pcm_hw_params_free (hw_params) ;
-
-	if ((err = snd_pcm_sw_params_malloc (&sw_params)) != 0)
-	{	fprintf (stderr, "%s: snd_pcm_sw_params_malloc: %s", __func__, snd_strerror (err)) ;
-		goto catch_error ;
-		} ;
-
-	if ((err = snd_pcm_sw_params_current (alsa_out->dev, sw_params)) != 0)
-	{	fprintf (stderr, "%s: snd_pcm_sw_params_current: %s", __func__, snd_strerror (err)) ;
-		goto catch_error ;
-		} ;
-
-	/* note: set start threshold to delay start until the ring buffer is full */
-	snd_pcm_sw_params_current (alsa_out->dev, sw_params) ;
-
-	if ((err = snd_pcm_sw_params_set_start_threshold (alsa_out->dev, sw_params, buffer_size)) < 0)
-	{	fprintf (stderr, "cannot set start threshold (%s)\n", snd_strerror (err)) ;
-		goto catch_error ;
-		} ;
-
-	if ((err = snd_pcm_sw_params (alsa_out->dev, sw_params)) != 0)
-	{	fprintf (stderr, "%s: snd_pcm_sw_params: %s", __func__, snd_strerror (err)) ;
-		goto catch_error ;
-		} ;
-
-	snd_pcm_sw_params_free (sw_params) ;
-
-	snd_pcm_reset (alsa_out->dev) ;
-
-catch_error :
-
-	if (err < 0 && alsa_out->dev != NULL)
-	{	snd_pcm_close (alsa_out->dev) ;
-		return NULL ;
-		} ;
-
-	return (AUDIO_OUT *) alsa_out ;
-} /* alsa_open */
-
-static void
-alsa_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data)
-{	static float buffer [BUFFER_LEN] ;
-	ALSA_AUDIO_OUT *alsa_out ;
-	int	read_frames ;
-
-	if ((alsa_out = (ALSA_AUDIO_OUT*) audio_out) == NULL)
-	{	printf ("alsa_close : AUDIO_OUT is NULL.\n") ;
-		return ;
-		} ;
-
-	if (alsa_out->magic != ALSA_MAGIC)
-	{	printf ("alsa_close : Bad magic number.\n") ;
-		return ;
-		} ;
-
-	while ((read_frames = callback (callback_data, buffer, BUFFER_LEN / alsa_out->channels)))
-		alsa_write_float (alsa_out->dev, buffer, read_frames, alsa_out->channels) ;
-
-	return ;
-} /* alsa_play */
-
-static int
-alsa_write_float (snd_pcm_t *alsa_dev, float *data, int frames, int channels)
-{	static	int epipe_count = 0 ;
-
-	int total = 0 ;
-	int retval ;
-
-	if (epipe_count > 0)
-		epipe_count -- ;
-
-	while (total < frames)
-	{	retval = snd_pcm_writei (alsa_dev, data + total * channels, frames - total) ;
-
-		if (retval >= 0)
-		{	total += retval ;
-			if (total == frames)
-				return total ;
-
-			continue ;
-			} ;
-
-		switch (retval)
-		{	case -EAGAIN :
-					puts ("alsa_write_float: EAGAIN") ;
-					continue ;
-					break ;
-
-			case -EPIPE :
-					if (epipe_count > 0)
-					{	printf ("alsa_write_float: EPIPE %d\n", epipe_count) ;
-						if (epipe_count > 140)
-							return retval ;
-						} ;
-					epipe_count += 100 ;
-
-#if 0
-					if (0)
-					{	snd_pcm_status_t *status ;
-
-						snd_pcm_status_alloca (&status) ;
-						if ((retval = snd_pcm_status (alsa_dev, status)) < 0)
-							fprintf (stderr, "alsa_out: xrun. can't determine length\n") ;
-						else if (snd_pcm_status_get_state (status) == SND_PCM_STATE_XRUN)
-						{	struct timeval now, diff, tstamp ;
-
-							gettimeofday (&now, 0) ;
-							snd_pcm_status_get_trigger_tstamp (status, &tstamp) ;
-							timersub (&now, &tstamp, &diff) ;
-
-							fprintf (stderr, "alsa_write_float xrun: of at least %.3f msecs. resetting stream\n",
-									diff.tv_sec * 1000 + diff.tv_usec / 1000.0) ;
-							}
-						else
-							fprintf (stderr, "alsa_write_float: xrun. can't determine length\n") ;
-						} ;
-#endif
-
-					snd_pcm_prepare (alsa_dev) ;
-					break ;
-
-			case -EBADFD :
-					fprintf (stderr, "alsa_write_float: Bad PCM state.n") ;
-					return 0 ;
-					break ;
-
-			case -ESTRPIPE :
-					fprintf (stderr, "alsa_write_float: Suspend event.n") ;
-					return 0 ;
-					break ;
-
-			case -EIO :
-					puts ("alsa_write_float: EIO") ;
-					return 0 ;
-
-			default :
-					fprintf (stderr, "alsa_write_float: retval = %d\n", retval) ;
-					return 0 ;
-					break ;
-			} ; /* switch */
-		} ; /* while */
-
-	return total ;
-} /* alsa_write_float */
-
-static void
-alsa_close (AUDIO_OUT *audio_out)
-{	ALSA_AUDIO_OUT *alsa_out ;
-
-	if ((alsa_out = (ALSA_AUDIO_OUT*) audio_out) == NULL)
-	{	printf ("alsa_close : AUDIO_OUT is NULL.\n") ;
-		return ;
-		} ;
-
-	if (alsa_out->magic != ALSA_MAGIC)
-	{	printf ("alsa_close : Bad magic number.\n") ;
-		return ;
-		} ;
-
-	memset (alsa_out, 0, sizeof (ALSA_AUDIO_OUT)) ;
-
-	free (alsa_out) ;
-
-	return ;
-} /* alsa_close */
-
-#endif /* HAVE_ALSA */
-
-#include <fcntl.h>
-#include <sys/ioctl.h>
-#include <sys/soundcard.h>
-
-#define	OSS_MAGIC		MAKE_MAGIC ('L', 'i', 'n', 'u', 'x', 'O', 'S', 'S')
-
-typedef struct
-{	int magic ;
-	int fd ;
-	int channels ;
-} OSS_AUDIO_OUT ;
-
-static AUDIO_OUT *opensoundsys_open (int channels, int samplerate) ;
-static void opensoundsys_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) ;
-static void opensoundsys_close (AUDIO_OUT *audio_out) ;
-
-
-static AUDIO_OUT *
-opensoundsys_open (int channels, int samplerate)
-{	OSS_AUDIO_OUT	*opensoundsys_out ;
-	int stereo, fmt, error ;
-
-	if ((opensoundsys_out = calloc (1, sizeof (OSS_AUDIO_OUT))) == NULL)
-	{	perror ("opensoundsys_open : malloc ") ;
-		exit (1) ;
-		} ;
-
-	opensoundsys_out->magic	= OSS_MAGIC ;
-	opensoundsys_out->channels = channels ;
-
-	if ((opensoundsys_out->fd = open ("/dev/dsp", O_WRONLY, 0)) == -1)
-	{	perror ("opensoundsys_open : open ") ;
-		exit (1) ;
-		} ;
-
-	stereo = 0 ;
-	if (ioctl (opensoundsys_out->fd, SNDCTL_DSP_STEREO, &stereo) == -1)
-	{ 	/* Fatal error */
-		perror ("opensoundsys_open : stereo ") ;
-		exit (1) ;
-		} ;
-
-	if (ioctl (opensoundsys_out->fd, SNDCTL_DSP_RESET, 0))
-	{	perror ("opensoundsys_open : reset ") ;
-		exit (1) ;
-		} ;
-
-	fmt = CPU_IS_BIG_ENDIAN ? AFMT_S16_BE : AFMT_S16_LE ;
-	if (ioctl (opensoundsys_out->fd, SNDCTL_DSP_SETFMT, &fmt) != 0)
-	{	perror ("opensoundsys_open_dsp_device : set format ") ;
-	    exit (1) ;
-  		} ;
-
-	if ((error = ioctl (opensoundsys_out->fd, SNDCTL_DSP_CHANNELS, &channels)) != 0)
-	{	perror ("opensoundsys_open : channels ") ;
-		exit (1) ;
-		} ;
-
-	if ((error = ioctl (opensoundsys_out->fd, SNDCTL_DSP_SPEED, &samplerate)) != 0)
-	{	perror ("opensoundsys_open : sample rate ") ;
-		exit (1) ;
-		} ;
-
-	if ((error = ioctl (opensoundsys_out->fd, SNDCTL_DSP_SYNC, 0)) != 0)
-	{	perror ("opensoundsys_open : sync ") ;
-		exit (1) ;
-		} ;
-
-	return 	(AUDIO_OUT*) opensoundsys_out ;
-} /* opensoundsys_open */
-
-static void
-opensoundsys_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data)
-{	OSS_AUDIO_OUT *opensoundsys_out ;
-	static float float_buffer [BUFFER_LEN] ;
-	static short buffer [BUFFER_LEN] ;
-	int		k, read_frames ;
-
-	if ((opensoundsys_out = (OSS_AUDIO_OUT*) audio_out) == NULL)
-	{	printf ("opensoundsys_play : AUDIO_OUT is NULL.\n") ;
-		return ;
-		} ;
-
-	if (opensoundsys_out->magic != OSS_MAGIC)
-	{	printf ("opensoundsys_play : Bad magic number.\n") ;
-		return ;
-		} ;
-
-	while ((read_frames = callback (callback_data, float_buffer, BUFFER_LEN / opensoundsys_out->channels)))
-	{	for (k = 0 ; k < read_frames * opensoundsys_out->channels ; k++)
-			buffer [k] = lrint (32767.0 * float_buffer [k]) ;
-		if (write (opensoundsys_out->fd, buffer, read_frames * opensoundsys_out->channels * sizeof (short))) {}
-		} ;
-
-	return ;
-} /* opensoundsys_play */
-
-static void
-opensoundsys_close (AUDIO_OUT *audio_out)
-{	OSS_AUDIO_OUT *opensoundsys_out ;
-
-	if ((opensoundsys_out = (OSS_AUDIO_OUT*) audio_out) == NULL)
-	{	printf ("opensoundsys_close : AUDIO_OUT is NULL.\n") ;
-		return ;
-		} ;
-
-	if (opensoundsys_out->magic != OSS_MAGIC)
-	{	printf ("opensoundsys_close : Bad magic number.\n") ;
-		return ;
-		} ;
-
-	memset (opensoundsys_out, 0, sizeof (OSS_AUDIO_OUT)) ;
-
-	free (opensoundsys_out) ;
-
-	return ;
-} /* opensoundsys_close */
-
-#endif /* __linux__ */
-
-/*------------------------------------------------------------------------------
-**	Mac OS X functions for playing a sound.
-*/
-
-#if (defined (__MACH__) && defined (__APPLE__)) /* MacOSX */
-
-#include <AvailabilityMacros.h>
-#include <CoreAudio/AudioHardware.h>
-#ifndef MAC_OS_VERSION_12_0
-#define kAudioObjectPropertyElementMain kAudioObjectPropertyElementMaster
-#endif
-
-#define	MACOSX_MAGIC	MAKE_MAGIC ('M', 'a', 'c', ' ', 'O', 'S', ' ', 'X')
-
-typedef struct
-{	int magic ;
-	AudioStreamBasicDescription	format ;
-
-	UInt32 			buf_size ;
-	AudioDeviceID 	device ;
-
-	int		channels ;
-	int 	samplerate ;
-	int		buffer_size ;
-	int		done_playing ;
-
-	get_audio_callback_t	callback ;
-
-	void 	*callback_data ;
-
-	AudioDeviceIOProcID ioprocid;
-
-} MACOSX_AUDIO_OUT ;
-
-static AUDIO_OUT *macosx_open (int channels, int samplerate) ;
-static void macosx_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) ;
-static void macosx_close (AUDIO_OUT *audio_out) ;
-
-static OSStatus
-macosx_audio_out_callback (AudioDeviceID device, const AudioTimeStamp* current_time,
-	const AudioBufferList* data_in, const AudioTimeStamp* time_in,
-	AudioBufferList* data_out, const AudioTimeStamp* time_out, void* client_data) ;
-
-
-static AUDIO_OUT *
-macosx_open (int channels, int samplerate)
-{	MACOSX_AUDIO_OUT *macosx_out ;
-	OSStatus	err ;
-	UInt32 		count ;
-	AudioObjectPropertyAddress  propertyAddress ;
-
-	if ((macosx_out = calloc (1, sizeof (MACOSX_AUDIO_OUT))) == NULL)
-	{	perror ("macosx_open : malloc ") ;
-		exit (1) ;
-		} ;
-
-	macosx_out->magic = MACOSX_MAGIC ;
-	macosx_out->channels = channels ;
-	macosx_out->samplerate = samplerate ;
-
-	macosx_out->device = kAudioDeviceUnknown ;
-
-	/*  get the default output device for the HAL */
-	propertyAddress.mSelector = kAudioHardwarePropertyDefaultOutputDevice;
-	propertyAddress.mScope = kAudioDevicePropertyScopeOutput;
-	propertyAddress.mElement = kAudioObjectPropertyElementMain;
-
-	count = sizeof (AudioDeviceID) ;
-	if ((err = AudioObjectGetPropertyData(kAudioObjectSystemObject, &propertyAddress, 0, NULL,
-			&count,  &(macosx_out->device))) != noErr)
-	{	printf ("AudioObjectGetPropertyData (kAudioHardwarePropertyDefaultOutputDevice) failed.\n") ;
-		free (macosx_out) ;
-		return NULL ;
-		} ;
-
-	/*  get the buffersize that the default device uses for IO */
-	count = sizeof (UInt32) ;
-	propertyAddress.mSelector = kAudioDevicePropertyBufferSize ;
-	if ((err = AudioObjectGetPropertyData (macosx_out->device, &propertyAddress, 0, NULL,
-				&count, &(macosx_out->buffer_size))) != noErr)
-	{	printf ("AudioObjectGetPropertyData (kAudioDevicePropertyBufferSize) (AudioDeviceGetProperty) failed.\n") ;
-		free (macosx_out) ;
-		return NULL ;
-		} ;
-
-	/*  get a description of the data format used by the default device */
-	count = sizeof (AudioStreamBasicDescription) ;
-	propertyAddress.mSelector = kAudioDevicePropertyStreamFormat ;
-	if ((err = AudioObjectGetPropertyData (macosx_out->device, &propertyAddress, 0, NULL,
-				&count, &(macosx_out->format))) != noErr)
-	{	printf ("AudioObjectGetPropertyData (kAudioDevicePropertyStreamFormat) failed.\n") ;
-		free (macosx_out) ;
-		return NULL ;
-		} ;
-
-	macosx_out->format.mSampleRate = samplerate ;
-	macosx_out->format.mChannelsPerFrame = channels ;
-	propertyAddress.mSelector = kAudioDevicePropertyStreamFormat ;
-	count = sizeof (AudioStreamBasicDescription) ;
-	if ((err = AudioObjectGetPropertyData (macosx_out->device, &propertyAddress, 0, NULL,
-				&count, &(macosx_out->format))) != noErr)
-	{	printf ("AudioObjectGetPropertyData (kAudioDevicePropertyStreamFormat) failed.\n") ;
-		free (macosx_out) ;
-		return NULL ;
-		} ;
-
-	/*  we want linear pcm */
-	if (macosx_out->format.mFormatID != kAudioFormatLinearPCM)
-	{	free (macosx_out) ;
-		return NULL ;
-		} ;
-
-	macosx_out->done_playing = 0 ;
-
-	/* Fire off the device. */
-	if ((err = AudioDeviceCreateIOProcID (macosx_out->device, macosx_audio_out_callback,
-			(void *) macosx_out, &macosx_out->ioprocid)) != noErr)
-	{	printf ("AudioDeviceAddIOProc failed.\n") ;
-		free (macosx_out) ;
-		return NULL ;
-		} ;
-
-	return (AUDIO_OUT *) macosx_out ;
-} /* macosx_open */
-
-static void
-macosx_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data)
-{	MACOSX_AUDIO_OUT	*macosx_out ;
-	OSStatus	err ;
-
-	if ((macosx_out = (MACOSX_AUDIO_OUT*) audio_out) == NULL)
-	{	printf ("macosx_play : AUDIO_OUT is NULL.\n") ;
-		return ;
-		} ;
-
-	if (macosx_out->magic != MACOSX_MAGIC)
-	{	printf ("macosx_play : Bad magic number.\n") ;
-		return ;
-		} ;
-
-	/* Set the callback function and callback data. */
-	macosx_out->callback = callback ;
-	macosx_out->callback_data = callback_data ;
-
-	err = AudioDeviceStart (macosx_out->device, macosx_audio_out_callback) ;
-	if (err != noErr)
-		printf ("AudioDeviceStart failed.\n") ;
-
-	while (macosx_out->done_playing == SF_FALSE)
-		usleep (10 * 1000) ; /* 10 000 milliseconds. */
-
-	return ;
-} /* macosx_play */
-
-static void
-macosx_close (AUDIO_OUT *audio_out)
-{	MACOSX_AUDIO_OUT	*macosx_out ;
-	OSStatus	err ;
-
-	if ((macosx_out = (MACOSX_AUDIO_OUT*) audio_out) == NULL)
-	{	printf ("macosx_close : AUDIO_OUT is NULL.\n") ;
-		return ;
-		} ;
-
-	if (macosx_out->magic != MACOSX_MAGIC)
-	{	printf ("macosx_close : Bad magic number.\n") ;
-		return ;
-		} ;
-
-
-	if ((err = AudioDeviceStop (macosx_out->device, macosx_audio_out_callback)) != noErr)
-	{	printf ("AudioDeviceStop failed.\n") ;
-		return ;
-		} ;
-
-	err = AudioDeviceDestroyIOProcID(macosx_out->device,
-									 macosx_out->ioprocid);
-	if (err != noErr)
-	{	printf ("AudioDeviceRemoveIOProc failed.\n") ;
-		return ;
-		} ;
-
-} /* macosx_close */
-
-static OSStatus
-macosx_audio_out_callback (AudioDeviceID device, const AudioTimeStamp* current_time,
-	const AudioBufferList* data_in, const AudioTimeStamp* time_in,
-	AudioBufferList* data_out, const AudioTimeStamp* time_out, void* client_data)
-{	MACOSX_AUDIO_OUT	*macosx_out ;
-	int		size, frame_count, read_count ;
-	float	*buffer ;
-
-	/* unused params: */
-	(void) device;
-	(void) current_time;
-	(void) data_in;
-	(void) time_in;
-	(void) time_out;
-
-	if ((macosx_out = (MACOSX_AUDIO_OUT*) client_data) == NULL)
-	{	printf ("macosx_play : AUDIO_OUT is NULL.\n") ;
-		return 42 ;
-		} ;
-
-	if (macosx_out->magic != MACOSX_MAGIC)
-	{	printf ("macosx_play : Bad magic number.\n") ;
-		return 42 ;
-		} ;
-
-	size = data_out->mBuffers [0].mDataByteSize ;
-	frame_count = size / sizeof (float) / macosx_out->channels ;
-
-	buffer = (float*) data_out->mBuffers [0].mData ;
-
-	read_count = macosx_out->callback (macosx_out->callback_data, buffer, frame_count) ;
-
-	if (read_count < frame_count)
-	{	memset (&(buffer [read_count]), 0, (frame_count - read_count) * sizeof (float)) ;
-		macosx_out->done_playing = 1 ;
-		} ;
-
-	return noErr ;
-} /* macosx_audio_out_callback */
-
-#endif /* MacOSX */
-
-
-/*------------------------------------------------------------------------------
-**	Win32 functions for playing a sound.
-**
-**	This API sucks. Its needlessly complicated and is *WAY* too loose with
-**	passing pointers arounf in integers and and using char* pointers to
-**  point to data instead of short*. It plain sucks!
-*/
-
-#if (defined (_WIN32) || defined (WIN32))
-
-#define	WIN32_BUFFER_LEN	(1<<15)
-#define	WIN32_MAGIC			MAKE_MAGIC ('W', 'i', 'n', '3', '2', 's', 'u', 'x')
-
-typedef struct
-{	int 			magic ;
-
-	HWAVEOUT		hwave ;
-	WAVEHDR			whdr [2] ;
-
-	HANDLE			Event ;
-
-	short			short_buffer [WIN32_BUFFER_LEN / sizeof (short)] ;
-	float			float_buffer [WIN32_BUFFER_LEN / sizeof (short) / 2] ;
-
-	int				bufferlen, current ;
-
-	int				channels ;
-
-	get_audio_callback_t	callback ;
-
-	void 			*callback_data ;
-} WIN32_AUDIO_OUT ;
-
-static AUDIO_OUT *win32_open (int channels, int samplerate) ;
-static void win32_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) ;
-static void win32_close (AUDIO_OUT *audio_out) ;
-
-static DWORD CALLBACK
-	win32_audio_out_callback (HWAVEOUT hwave, UINT msg, DWORD_PTR data, DWORD_PTR param1, DWORD_PTR param2) ;
-
-static AUDIO_OUT*
-win32_open (int channels, int samplerate)
-{	WIN32_AUDIO_OUT *win32_out ;
-
-	WAVEFORMATEX wf ;
-	int error ;
-
-	if ((win32_out = calloc (1, sizeof (WIN32_AUDIO_OUT))) == NULL)
-	{	perror ("win32_open : malloc ") ;
-		exit (1) ;
-		} ;
-
-	win32_out->magic	= WIN32_MAGIC ;
-	win32_out->channels = channels ;
-
-	win32_out->current = 0 ;
-
-	win32_out->Event = CreateEvent (0, FALSE, FALSE, 0) ;
-
-	wf.nChannels = channels ;
-	wf.nSamplesPerSec = samplerate ;
-	wf.nBlockAlign = (WORD) (channels * sizeof (short)) ;
-
-	wf.wFormatTag = WAVE_FORMAT_PCM ;
-	wf.cbSize = 0 ;
-	wf.wBitsPerSample = 16 ;
-	wf.nAvgBytesPerSec = wf.nBlockAlign * wf.nSamplesPerSec ;
-
-	error = waveOutOpen (&(win32_out->hwave), WAVE_MAPPER, &wf, (DWORD_PTR) win32_audio_out_callback,
-							(DWORD_PTR) win32_out, CALLBACK_FUNCTION) ;
-	if (error)
-	{	puts ("waveOutOpen failed.") ;
-		free (win32_out) ;
-		return NULL ;
-		} ;
-
-	waveOutPause (win32_out->hwave) ;
-
-	return (AUDIO_OUT *) win32_out ;
-} /* win32_open */
-
-static void
-win32_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data)
-{	WIN32_AUDIO_OUT	*win32_out ;
-	int		error ;
-
-	if ((win32_out = (WIN32_AUDIO_OUT*) audio_out) == NULL)
-	{	printf ("win32_play : AUDIO_OUT is NULL.\n") ;
-		return ;
-		} ;
-
-	if (win32_out->magic != WIN32_MAGIC)
-	{	printf ("win32_play : Bad magic number (%d %d).\n", win32_out->magic, WIN32_MAGIC) ;
-		return ;
-		} ;
-
-	/* Set the callback function and callback data. */
-	win32_out->callback = callback ;
-	win32_out->callback_data = callback_data ;
-
-	win32_out->whdr [0].lpData = (char*) win32_out->short_buffer ;
-	win32_out->whdr [1].lpData = ((char*) win32_out->short_buffer) + sizeof (win32_out->short_buffer) / 2 ;
-
-	win32_out->whdr [0].dwBufferLength = sizeof (win32_out->short_buffer) / 2 ;
-	win32_out->whdr [1].dwBufferLength = sizeof (win32_out->short_buffer) / 2 ;
-
-	win32_out->bufferlen = sizeof (win32_out->short_buffer) / 2 / sizeof (short) ;
-
-	/* Prepare the WAVEHDRs */
-	if ((error = waveOutPrepareHeader (win32_out->hwave, &(win32_out->whdr [0]), sizeof (WAVEHDR))))
-	{	printf ("waveOutPrepareHeader [0] failed : %08X\n", error) ;
-		waveOutClose (win32_out->hwave) ;
-		return ;
-		} ;
-
-	if ((error = waveOutPrepareHeader (win32_out->hwave, &(win32_out->whdr [1]), sizeof (WAVEHDR))))
-	{	printf ("waveOutPrepareHeader [1] failed : %08X\n", error) ;
-		waveOutUnprepareHeader (win32_out->hwave, &(win32_out->whdr [0]), sizeof (WAVEHDR)) ;
-		waveOutClose (win32_out->hwave) ;
-		return ;
-		} ;
-
-	waveOutRestart (win32_out->hwave) ;
-
-	/* Fake 2 calls to the callback function to queue up enough audio. */
-	win32_audio_out_callback (0, MM_WOM_DONE, (DWORD_PTR) win32_out, 0, 0) ;
-	win32_audio_out_callback (0, MM_WOM_DONE, (DWORD_PTR) win32_out, 0, 0) ;
-
-	/* Wait for playback to finish. The callback notifies us when all
-	** wave data has been played.
-	*/
-	WaitForSingleObject (win32_out->Event, INFINITE) ;
-
-	waveOutPause (win32_out->hwave) ;
-	waveOutReset (win32_out->hwave) ;
-
-	waveOutUnprepareHeader (win32_out->hwave, &(win32_out->whdr [0]), sizeof (WAVEHDR)) ;
-	waveOutUnprepareHeader (win32_out->hwave, &(win32_out->whdr [1]), sizeof (WAVEHDR)) ;
-
-	waveOutClose (win32_out->hwave) ;
-	win32_out->hwave = 0 ;
-
-	return ;
-} /* win32_play */
-
-static void
-win32_close (AUDIO_OUT *audio_out)
-{	WIN32_AUDIO_OUT *win32_out ;
-
-	if ((win32_out = (WIN32_AUDIO_OUT*) audio_out) == NULL)
-	{	printf ("win32_close : AUDIO_OUT is NULL.\n") ;
-		return ;
-		} ;
-
-	if (win32_out->magic != WIN32_MAGIC)
-	{	printf ("win32_close : Bad magic number.\n") ;
-		return ;
-		} ;
-
-	memset (win32_out, 0, sizeof (WIN32_AUDIO_OUT)) ;
-
-	free (win32_out) ;
-} /* win32_close */
-
-static DWORD CALLBACK
-win32_audio_out_callback (HWAVEOUT hwave, UINT msg, DWORD_PTR data, DWORD_PTR param1, DWORD_PTR param2)
-{
-	UNREFERENCED_PARAMETER (hwave) ;
-	UNREFERENCED_PARAMETER (param1) ;
-	UNREFERENCED_PARAMETER (param2) ;
-	WIN32_AUDIO_OUT	*win32_out ;
-	int		read_count, frame_count, k ;
-	short	*sptr ;
-
-	/*
-	** I consider this technique of passing a pointer via an integer as
-	** fundamentally broken but thats the way microsoft has defined the
-	** interface.
-	*/
-	if ((win32_out = (WIN32_AUDIO_OUT*) data) == NULL)
-	{	printf ("win32_audio_out_callback : AUDIO_OUT is NULL.\n") ;
-		return 1 ;
-		} ;
-
-	if (win32_out->magic != WIN32_MAGIC)
-	{	printf ("win32_audio_out_callback : Bad magic number (%d %d).\n", win32_out->magic, WIN32_MAGIC) ;
-		return 1 ;
-		} ;
-
-	if (msg != MM_WOM_DONE)
-		return 0 ;
-
-	/* Do the actual audio. */
-	frame_count = win32_out->bufferlen / win32_out->channels ;
-
-	read_count = win32_out->callback (win32_out->callback_data, win32_out->float_buffer, frame_count) ;
-
-	sptr = (short*) win32_out->whdr [win32_out->current].lpData ;
-
-	for (k = 0 ; k < read_count ; k++)
-		sptr [k] = (short) lrint (32767.0 * win32_out->float_buffer [k]) ;
-
-	if (read_count > 0)
-	{	/* Fix buffer length is only a partial block. */
-		if (read_count * (int) sizeof (short) < win32_out->bufferlen)
-			win32_out->whdr [win32_out->current].dwBufferLength = read_count * sizeof (short) ;
-
-		/* Queue the WAVEHDR */
-		waveOutWrite (win32_out->hwave, (LPWAVEHDR) &(win32_out->whdr [win32_out->current]), sizeof (WAVEHDR)) ;
-		}
-	else
-	{	/* Stop playback */
-		waveOutPause (win32_out->hwave) ;
-
-		SetEvent (win32_out->Event) ;
-		} ;
-
-	win32_out->current = (win32_out->current + 1) % 2 ;
-
-	return 0 ;
-} /* win32_audio_out_callback */
-
-#endif /* Win32 */
-
-/*------------------------------------------------------------------------------
-**	Solaris.
-*/
-
-#if (defined (sun) && defined (unix)) /* ie Solaris */
-
-#include <fcntl.h>
-#include <sys/ioctl.h>
-#include <sys/audioio.h>
-
-#define	SOLARIS_MAGIC	MAKE_MAGIC ('S', 'o', 'l', 'a', 'r', 'i', 's', ' ')
-
-typedef struct
-{	int magic ;
-	int fd ;
-	int channels ;
-	int samplerate ;
-} SOLARIS_AUDIO_OUT ;
-
-static AUDIO_OUT *solaris_open (int channels, int samplerate) ;
-static void solaris_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data) ;
-static void solaris_close (AUDIO_OUT *audio_out) ;
-
-static AUDIO_OUT *
-solaris_open (int channels, int samplerate)
-{	SOLARIS_AUDIO_OUT	*solaris_out ;
-	audio_info_t		audio_info ;
-	int					error ;
-
-	if ((solaris_out = calloc (1, sizeof (SOLARIS_AUDIO_OUT))) == NULL)
-	{	perror ("solaris_open : malloc ") ;
-		exit (1) ;
-		} ;
-
-	solaris_out->magic		= SOLARIS_MAGIC ;
-	solaris_out->channels	= channels ;
-	solaris_out->samplerate	= channels ;
-
-	/* open the audio device - write only, non-blocking */
-	if ((solaris_out->fd = open ("/dev/audio", O_WRONLY | O_NONBLOCK)) < 0)
-	{	perror ("open (/dev/audio) failed") ;
-		exit (1) ;
-		} ;
-
-	/*	Retrive standard values. */
-	AUDIO_INITINFO (&audio_info) ;
-
-	audio_info.play.sample_rate = samplerate ;
-	audio_info.play.channels = channels ;
-	audio_info.play.precision = 16 ;
-	audio_info.play.encoding = AUDIO_ENCODING_LINEAR ;
-	audio_info.play.gain = AUDIO_MAX_GAIN ;
-	audio_info.play.balance = AUDIO_MID_BALANCE ;
-
-	if ((error = ioctl (solaris_out->fd, AUDIO_SETINFO, &audio_info)))
-	{	perror ("ioctl (AUDIO_SETINFO) failed") ;
-		exit (1) ;
-		} ;
-
-	return 	(AUDIO_OUT*) solaris_out ;
-} /* solaris_open */
-
-static void
-solaris_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data)
-{	SOLARIS_AUDIO_OUT *solaris_out ;
-	static float float_buffer [BUFFER_LEN] ;
-	static short buffer [BUFFER_LEN] ;
-	int		k, read_frames ;
-
-	if ((solaris_out = (SOLARIS_AUDIO_OUT*) audio_out) == NULL)
-	{	printf ("solaris_play : AUDIO_OUT is NULL.\n") ;
-		return ;
-		} ;
-
-	if (solaris_out->magic != SOLARIS_MAGIC)
-	{	printf ("solaris_play : Bad magic number.\n") ;
-		return ;
-		} ;
-
-	while ((read_frames = callback (callback_data, float_buffer, BUFFER_LEN / solaris_out->channels)))
-	{	for (k = 0 ; k < read_frames * solaris_out->channels ; k++)
-			buffer [k] = psf_lrint (32767.0 * float_buffer [k]) ;
-		write (solaris_out->fd, buffer, read_frames * solaris_out->channels * sizeof (short)) ;
-		} ;
-
-	return ;
-} /* solaris_play */
-
-static void
-solaris_close (AUDIO_OUT *audio_out)
-{	SOLARIS_AUDIO_OUT *solaris_out ;
-
-	if ((solaris_out = (SOLARIS_AUDIO_OUT*) audio_out) == NULL)
-	{	printf ("solaris_close : AUDIO_OUT is NULL.\n") ;
-		return ;
-		} ;
-
-	if (solaris_out->magic != SOLARIS_MAGIC)
-	{	printf ("solaris_close : Bad magic number.\n") ;
-		return ;
-		} ;
-
-	memset (solaris_out, 0, sizeof (SOLARIS_AUDIO_OUT)) ;
-
-	free (solaris_out) ;
-
-	return ;
-} /* solaris_close */
-
-#endif /* Solaris */
-
-/*==============================================================================
-**	Main function.
-*/
-
-AUDIO_OUT *
-audio_open (int channels, int samplerate)
-{
-#if defined (__linux__)
-	#if HAVE_ALSA
-		if (access ("/proc/asound/cards", R_OK) == 0)
-			return alsa_open (channels, samplerate) ;
-	#endif
-		return opensoundsys_open (channels, samplerate) ;
-#elif (defined (__MACH__) && defined (__APPLE__))
-	return macosx_open (channels, samplerate) ;
-#elif (defined (sun) && defined (unix))
-	return solaris_open (channels, samplerate) ;
-#elif (defined (_WIN32) || defined (WIN32))
-	return win32_open (channels, samplerate) ;
-#else
-	#warning "*** Playing sound not yet supported on this platform."
-	#warning "*** Please feel free to submit a patch."
-	printf ("Error : Playing sound not yet supported on this platform.\n") ;
-	return NULL ;
-#endif
-
-
-	return NULL ;
-} /* audio_open */
-
-void
-audio_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data)
-{
-
-	if (callback == NULL)
-	{	printf ("Error : bad callback pointer.\n") ;
-		return ;
-		} ;
-
-	if (audio_out == NULL)
-	{	printf ("Error : bad audio_out pointer.\n") ;
-		return ;
-		} ;
-
-	if (callback_data == NULL)
-	{	printf ("Error : bad callback_data pointer.\n") ;
-		return ;
-		} ;
-
-#if defined (__linux__)
-	#if HAVE_ALSA
-		if (audio_out->magic == ALSA_MAGIC)
-			alsa_play (callback, audio_out, callback_data) ;
-	#endif
-		opensoundsys_play (callback, audio_out, callback_data) ;
-#elif (defined (__MACH__) && defined (__APPLE__))
-	macosx_play (callback, audio_out, callback_data) ;
-#elif (defined (sun) && defined (unix))
-	solaris_play (callback, audio_out, callback_data) ;
-#elif (defined (_WIN32) || defined (WIN32))
-	win32_play (callback, audio_out, callback_data) ;
-#else
-	#warning "*** Playing sound not yet supported on this platform."
-	#warning "*** Please feel free to submit a patch."
-	printf ("Error : Playing sound not yet supported on this platform.\n") ;
-	return ;
-#endif
-
-	return ;
-} /* audio_play */
-
-void
-audio_close (AUDIO_OUT *audio_out)
-{
-#if defined (__linux__)
-	#if HAVE_ALSA
-		if (audio_out->magic == ALSA_MAGIC)
-			alsa_close (audio_out) ;
-	#endif
-	opensoundsys_close (audio_out) ;
-#elif (defined (__MACH__) && defined (__APPLE__))
-	macosx_close (audio_out) ;
-#elif (defined (sun) && defined (unix))
-	solaris_close (audio_out) ;
-#elif (defined (_WIN32) || defined (WIN32))
-	win32_close (audio_out) ;
-#else
-	#warning "*** Playing sound not yet supported on this platform."
-	#warning "*** Please feel free to submit a patch."
-	printf ("Error : Playing sound not yet supported on this platform.\n") ;
-	return ;
-#endif
-
-	return ;
-} /* audio_close */
-
-#else /* (HAVE_SNDFILE == 0) */
-
-/* Do not have libsndfile installed so just return. */
-
-AUDIO_OUT *
-audio_open (int channels, int samplerate)
-{
-	(void) channels ;
-	(void) samplerate ;
-
-	return NULL ;
-} /* audio_open */
-
-void
-audio_play (get_audio_callback_t callback, AUDIO_OUT *audio_out, void *callback_data)
-{
-	(void) callback ;
-	(void) audio_out ;
-	(void) callback_data ;
-
-	return ;
-} /* audio_play */
-
-void
-audio_close (AUDIO_OUT *audio_out)
-{
-	audio_out = audio_out ;
-
-	return ;
-} /* audio_close */
-
-#endif /* HAVE_SNDFILE */
-