comparison src/libsndfile-1.0.27/programs/sndfile-play.c @ 40:1df64224f5ac

Current libsndfile source

+/*
+** Copyright (C) 1999-2015 Erik de Castro Lopo <[e-mail]>
+**
+** All rights reserved.
+**
+** Redistribution and use in source and binary forms, with or without
+** modification, are permitted provided that the following conditions are
+** met:
+**
+**     * Redistributions of source code must retain the above copyright
+**       notice, this list of conditions and the following disclaimer.
+**     * Redistributions in binary form must reproduce the above copyright
+**       notice, this list of conditions and the following disclaimer in
+**       the documentation and/or other materials provided with the
+**       distribution.
+**     * Neither the author nor the names of any contributors may be used
+**       to endorse or promote products derived from this software without
+**       specific prior written permission.
+**
+** THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
+** "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED
+** TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR
+** PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER OR
+** CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL,
+** EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+** PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+** OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+** WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+** OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+** ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+*/
+
+#include "sfconfig.h"
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <errno.h>
+
+#if HAVE_UNISTD_H
+#include <unistd.h>
+#endif
+
+#include <sndfile.h>
+
+#include "common.h"
+
+#if HAVE_ALSA_ASOUNDLIB_H
+	#define ALSA_PCM_NEW_HW_PARAMS_API
+	#define ALSA_PCM_NEW_SW_PARAMS_API
+	#include <alsa/asoundlib.h>
+	#include <sys/time.h>
+#endif
+
+#if defined (__ANDROID__)
+
+#elif defined (__linux__) || defined (__FreeBSD_kernel__) || defined (__FreeBSD__)
+	#include 	<fcntl.h>
+	#include 	<sys/ioctl.h>
+	#include 	<sys/soundcard.h>
+
+#elif (defined (__MACH__) && defined (__APPLE__))
+	#include <AvailabilityMacros.h>
+	#include <Availability.h>
+
+	#if (OSX_DARWIN_VERSION > 11)
+		/* Includes go here. */
+	#elif (OSX_DARWIN_VERSION == 11)
+		#include <AudioToolbox/AudioToolbox.h>
+	#elif (OSX_DARWIN_VERSION > 0 && OSX_DARWIN_VERSION <= 10)
+		#include <CoreAudio/AudioHardware.h>
+	#endif
+
+#elif HAVE_SNDIO_H
+	#include <sndio.h>
+
+#elif (defined (sun) && defined (unix))
+	#include <fcntl.h>
+	#include <sys/ioctl.h>
+	#include <sys/audioio.h>
+
+#elif (OS_IS_WIN32 == 1)
+	#include <windows.h>
+	#include <mmsystem.h>
+
+#endif
+
+#define	SIGNED_SIZEOF(x)	((int) sizeof (x))
+#define	BUFFER_LEN			(2048)
+
+/*------------------------------------------------------------------------------
+**	Linux/OSS functions for playing a sound.
+*/
+
+#if HAVE_ALSA_ASOUNDLIB_H
+
+static snd_pcm_t * alsa_open (int channels, unsigned srate, int realtime) ;
+static int alsa_write_float (snd_pcm_t *alsa_dev, float *data, int frames, int channels) ;
+
+static void
+alsa_play (int argc, char *argv [])
+{	static float buffer [BUFFER_LEN] ;
+	SNDFILE *sndfile ;
+	SF_INFO sfinfo ;
+	snd_pcm_t * alsa_dev ;
+	int		k, readcount, subformat ;
+
+	for (k = 1 ; k < argc ; k++)
+	{	memset (&sfinfo, 0, sizeof (sfinfo)) ;
+
+		printf ("Playing %s\n", argv [k]) ;
+		if (! (sndfile = sf_open (argv [k], SFM_READ, &sfinfo)))
+		{	puts (sf_strerror (NULL)) ;
+			continue ;
+			} ;
+
+		if (sfinfo.channels < 1 || sfinfo.channels > 2)
+		{	printf ("Error : channels = %d.\n", sfinfo.channels) ;
+			continue ;
+			} ;
+
+		if ((alsa_dev = alsa_open (sfinfo.channels, (unsigned) sfinfo.samplerate, SF_FALSE)) == NULL)
+			continue ;
+
+		subformat = sfinfo.format & SF_FORMAT_SUBMASK ;
+
+		if (subformat == SF_FORMAT_FLOAT || subformat == SF_FORMAT_DOUBLE)
+		{	double	scale ;
+			int 	m ;
+
+			sf_command (sndfile, SFC_CALC_SIGNAL_MAX, &scale, sizeof (scale)) ;
+			if (scale < 1e-10)
+				scale = 1.0 ;
+			else
+				scale = 32700.0 / scale ;
+
+			while ((readcount = sf_read_float (sndfile, buffer, BUFFER_LEN)))
+			{	for (m = 0 ; m < readcount ; m++)
+					buffer [m] *= scale ;
+				alsa_write_float (alsa_dev, buffer, BUFFER_LEN / sfinfo.channels, sfinfo.channels) ;
+				} ;
+			}
+		else
+		{	while ((readcount = sf_read_float (sndfile, buffer, BUFFER_LEN)))
+				alsa_write_float (alsa_dev, buffer, BUFFER_LEN / sfinfo.channels, sfinfo.channels) ;
+			} ;
+
+		snd_pcm_drain (alsa_dev) ;
+		snd_pcm_close (alsa_dev) ;
+
+		sf_close (sndfile) ;
+		} ;
+
+	return ;
+} /* alsa_play */
+
+static snd_pcm_t *
+alsa_open (int channels, unsigned samplerate, int realtime)
+{	const char * device = "default" ;
+	snd_pcm_t *alsa_dev = NULL ;
+	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 ;
+
+	if (realtime)
+	{	alsa_period_size = 256 ;
+		alsa_buffer_frames = 3 * alsa_period_size ;
+		}
+	else
+	{	alsa_period_size = 1024 ;
+		alsa_buffer_frames = 4 * alsa_period_size ;
+		} ;
+
+	if ((err = snd_pcm_open (&alsa_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_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_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_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_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_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_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_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_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_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_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_dev, sw_params) ;
+
+	if ((err = snd_pcm_sw_params_set_start_threshold (alsa_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_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_dev) ;
+
+catch_error :
+
+	if (err < 0 && alsa_dev != NULL)
+	{	snd_pcm_close (alsa_dev) ;
+		return NULL ;
+		} ;
+
+	return alsa_dev ;
+} /* alsa_open */
+
+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 */
+
+#endif /* HAVE_ALSA_ASOUNDLIB_H */
+
+/*------------------------------------------------------------------------------
+**	Linux/OSS functions for playing a sound.
+*/
+
+#if !defined (__ANDROID__) && (defined (__linux__) || defined (__FreeBSD_kernel__) || defined (__FreeBSD__))
+
+static	int	opensoundsys_open_device (int channels, int srate) ;
+
+static int
+opensoundsys_play (int argc, char *argv [])
+{	static short buffer [BUFFER_LEN] ;
+	SNDFILE *sndfile ;
+	SF_INFO sfinfo ;
+	int		k, audio_device, readcount, writecount, subformat ;
+
+	for (k = 1 ; k < argc ; k++)
+	{	memset (&sfinfo, 0, sizeof (sfinfo)) ;
+
+		printf ("Playing %s\n", argv [k]) ;
+		if (! (sndfile = sf_open (argv [k], SFM_READ, &sfinfo)))
+		{	puts (sf_strerror (NULL)) ;
+			continue ;
+			} ;
+
+		if (sfinfo.channels < 1 || sfinfo.channels > 2)
+		{	printf ("Error : channels = %d.\n", sfinfo.channels) ;
+			continue ;
+			} ;
+
+		audio_device = opensoundsys_open_device (sfinfo.channels, sfinfo.samplerate) ;
+
+		subformat = sfinfo.format & SF_FORMAT_SUBMASK ;
+
+		if (subformat == SF_FORMAT_FLOAT || subformat == SF_FORMAT_DOUBLE)
+		{	static float float_buffer [BUFFER_LEN] ;
+			double	scale ;
+			int 	m ;
+
+			sf_command (sndfile, SFC_CALC_SIGNAL_MAX, &scale, sizeof (scale)) ;
+			if (scale < 1e-10)
+				scale = 1.0 ;
+			else
+				scale = 32700.0 / scale ;
+
+			while ((readcount = sf_read_float (sndfile, float_buffer, BUFFER_LEN)))
+			{	for (m = 0 ; m < readcount ; m++)
+					buffer [m] = scale * float_buffer [m] ;
+				writecount = write (audio_device, buffer, readcount * sizeof (short)) ;
+				} ;
+			}
+		else
+		{	while ((readcount = sf_read_short (sndfile, buffer, BUFFER_LEN)))
+				writecount = write (audio_device, buffer, readcount * sizeof (short)) ;
+			} ;
+
+		if (ioctl (audio_device, SNDCTL_DSP_POST, 0) == -1)
+			perror ("ioctl (SNDCTL_DSP_POST) ") ;
+
+		if (ioctl (audio_device, SNDCTL_DSP_SYNC, 0) == -1)
+			perror ("ioctl (SNDCTL_DSP_SYNC) ") ;
+
+		close (audio_device) ;
+
+		sf_close (sndfile) ;
+		} ;
+
+	return writecount ;
+} /* opensoundsys_play */
+
+static int
+opensoundsys_open_device (int channels, int srate)
+{	int fd, stereo, fmt ;
+
+	if ((fd = open ("/dev/dsp", O_WRONLY, 0)) == -1 &&
+		(fd = open ("/dev/sound/dsp", O_WRONLY, 0)) == -1)
+	{	perror ("opensoundsys_open_device : open ") ;
+		exit (1) ;
+		} ;
+
+	stereo = 0 ;
+	if (ioctl (fd, SNDCTL_DSP_STEREO, &stereo) == -1)
+	{ 	/* Fatal error */
+		perror ("opensoundsys_open_device : stereo ") ;
+		exit (1) ;
+		} ;
+
+	if (ioctl (fd, SNDCTL_DSP_RESET, 0))
+	{	perror ("opensoundsys_open_device : reset ") ;
+		exit (1) ;
+		} ;
+
+	fmt = CPU_IS_BIG_ENDIAN ? AFMT_S16_BE : AFMT_S16_LE ;
+	if (ioctl (fd, SNDCTL_DSP_SETFMT, &fmt) != 0)
+	{	perror ("opensoundsys_open_device : set format ") ;
+		exit (1) ;
+		} ;
+
+	if (ioctl (fd, SNDCTL_DSP_CHANNELS, &channels) != 0)
+	{	perror ("opensoundsys_open_device : channels ") ;
+		exit (1) ;
+		} ;
+
+	if (ioctl (fd, SNDCTL_DSP_SPEED, &srate) != 0)
+	{	perror ("opensoundsys_open_device : sample rate ") ;
+		exit (1) ;
+		} ;
+
+	if (ioctl (fd, SNDCTL_DSP_SYNC, 0) != 0)
+	{	perror ("opensoundsys_open_device : sync ") ;
+		exit (1) ;
+		} ;
+
+	return 	fd ;
+} /* opensoundsys_open_device */
+
+#endif /* __linux__ */
+
+/*------------------------------------------------------------------------------
+**	Mac OS X functions for playing a sound.
+*/
+
+#if (OSX_DARWIN_VERSION > 11)
+/* MacOSX 10.8 use a new Audio API. Someone needs to write some code for it. */
+#endif /* OSX_DARWIN_VERSION > 11 */
+
+#if (OSX_DARWIN_VERSION == 11)
+/* MacOSX 10.7 use AudioQueue API */
+
+#define kBytesPerAudioBuffer	(1024 * 8)
+#define kNumberOfAudioBuffers	4
+
+typedef struct
+{	AudioStreamBasicDescription		format ;
+
+	AudioQueueRef		queue ;
+	AudioQueueBufferRef	queueBuffer [kNumberOfAudioBuffers] ;
+
+	UInt32 			buf_size ;
+
+	SNDFILE 		*sndfile ;
+	SF_INFO 		sfinfo ;
+
+	int				done_playing ;
+} MacOSXAudioData ;
+
+
+static void
+macosx_fill_buffer (MacOSXAudioData *audio_data, AudioQueueBufferRef audio_buffer)
+{	int		size, sample_count, read_count ;
+	short	*buffer ;
+
+	size = audio_buffer->mAudioDataBytesCapacity ;
+	sample_count = size / sizeof (short) ;
+
+	buffer = (short*) audio_buffer->mAudioData ;
+
+	read_count = sf_read_short (audio_data->sndfile, buffer, sample_count) ;
+
+	if (read_count > 0)
+	{	audio_buffer->mAudioDataByteSize = read_count * sizeof (short) ;
+		AudioQueueEnqueueBuffer (audio_data->queue, audio_buffer, 0, NULL) ;
+		}
+	else
+		AudioQueueStop (audio_data->queue, false) ;
+
+} /* macosx_fill_buffer */
+
+
+static void
+macosx_audio_out_callback (void *user_data, AudioQueueRef audio_queue, AudioQueueBufferRef audio_buffer)
+{	MacOSXAudioData *audio_data = (MacOSXAudioData *) user_data ;
+
+	if (audio_data->queue == audio_queue)
+		macosx_fill_buffer (audio_data, audio_buffer) ;
+
+} /* macosx_audio_out_callback */
+
+
+static void
+macosx_audio_out_property_callback (void *user_data, AudioQueueRef audio_queue, AudioQueuePropertyID prop)
+{	MacOSXAudioData *audio_data = (MacOSXAudioData *) user_data ;
+
+	if (prop == kAudioQueueProperty_IsRunning)
+	{	UInt32 is_running = 0 ;
+		UInt32 is_running_size = sizeof (is_running) ;
+
+		AudioQueueGetProperty (audio_queue, kAudioQueueProperty_IsRunning, &is_running, &is_running_size) ;
+
+		if (!is_running)
+		{	audio_data->done_playing = SF_TRUE ;
+			CFRunLoopStop (CFRunLoopGetCurrent ()) ;
+			} ;
+		} ;
+} /* macosx_audio_out_property_callback */
+
+
+
+static void
+macosx_play (int argc, char *argv [])
+{	MacOSXAudioData 	audio_data ;
+	OSStatus	err ;
+	int 		i ;
+	int 		k ;
+
+	memset (&audio_data, 0x55, sizeof (audio_data)) ;
+
+	for (k = 1 ; k < argc ; k++)
+	{	memset (&(audio_data.sfinfo), 0, sizeof (audio_data.sfinfo)) ;
+
+		printf ("Playing %s\n", argv [k]) ;
+		if (! (audio_data.sndfile = sf_open (argv [k], SFM_READ, &(audio_data.sfinfo))))
+		{	puts (sf_strerror (NULL)) ;
+			continue ;
+			} ;
+
+		if (audio_data.sfinfo.channels < 1 || audio_data.sfinfo.channels > 2)
+		{	printf ("Error : channels = %d.\n", audio_data.sfinfo.channels) ;
+			continue ;
+			} ;
+
+		/*  fill ASBD */
+		audio_data.format.mSampleRate = audio_data.sfinfo.samplerate ;
+		audio_data.format.mChannelsPerFrame	= audio_data.sfinfo.channels ;
+		audio_data.format.mFormatID			= kAudioFormatLinearPCM ;
+		audio_data.format.mFormatFlags		= kAudioFormatFlagIsSignedInteger | kAudioFormatFlagsNativeEndian | kAudioFormatFlagIsPacked ;
+		audio_data.format.mBytesPerPacket	= audio_data.format.mChannelsPerFrame * 2 ;
+		audio_data.format.mFramesPerPacket	= 1 ;
+		audio_data.format.mBytesPerFrame	= audio_data.format.mBytesPerPacket ;
+		audio_data.format.mBitsPerChannel	= 16 ;
+		audio_data.format.mReserved			= 0 ;
+
+		/* create the queue */
+		if ((err = AudioQueueNewOutput (&(audio_data.format), macosx_audio_out_callback, &audio_data,
+					CFRunLoopGetCurrent (), kCFRunLoopCommonModes, 0, &(audio_data.queue))) != noErr)
+		{	printf ("AudioQueueNewOutput failed\n") ;
+			return ;
+			} ;
+
+		/*  add property listener */
+		if ((err = AudioQueueAddPropertyListener (audio_data.queue, kAudioQueueProperty_IsRunning, macosx_audio_out_property_callback, &audio_data)) != noErr)
+		{	printf ("AudioQueueAddPropertyListener failed\n") ;
+			return ;
+			} ;
+
+		/*  create the buffers */
+		for (i = 0 ; i < kNumberOfAudioBuffers ; i++)
+		{	if ((err = AudioQueueAllocateBuffer (audio_data.queue, kBytesPerAudioBuffer, &audio_data.queueBuffer [i])) != noErr)
+			{	printf ("AudioQueueAllocateBuffer failed\n") ;
+				return ;
+				} ;
+
+			macosx_fill_buffer (&audio_data, audio_data.queueBuffer [i]) ;
+			} ;
+
+		audio_data.done_playing = SF_FALSE ;
+
+		/* start queue */
+		if ((err = AudioQueueStart (audio_data.queue, NULL)) != noErr)
+		{	printf ("AudioQueueStart failed\n") ;
+			return ;
+		} ;
+
+		while (audio_data.done_playing == SF_FALSE)
+			CFRunLoopRun () ;
+
+		/*  free the buffers */
+		for (i = 0 ; i < kNumberOfAudioBuffers ; i++)
+		{	if ((err = AudioQueueFreeBuffer (audio_data.queue, audio_data.queueBuffer [i])) != noErr)
+			{	printf ("AudioQueueFreeBuffer failed\n") ;
+				return ;
+				} ;
+			} ;
+
+		/*  free the queue */
+		if ((err = AudioQueueDispose (audio_data.queue, true)) != noErr)
+		{	printf ("AudioQueueDispose failed\n") ;
+			return ;
+			} ;
+
+		sf_close (audio_data.sndfile) ;
+		} ;
+
+	return ;
+} /* macosx_play, AudioQueue implementation */
+
+#endif /* OSX_DARWIN_VERSION == 11 */
+
+#if (OSX_DARWIN_VERSION > 0 && OSX_DARWIN_VERSION <= 10)
+/* MacOSX 10.6 or earlier, use AudioHardware API */
+
+typedef struct
+{	AudioStreamBasicDescription		format ;
+
+	UInt32 			buf_size ;
+	AudioDeviceID 	device ;
+
+	SNDFILE 		*sndfile ;
+	SF_INFO 		sfinfo ;
+
+	int				fake_stereo ;
+	int				done_playing ;
+} MacOSXAudioData ;
+
+#include <math.h>
+
+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)
+{	MacOSXAudioData	*audio_data ;
+	int		size, sample_count, read_count, k ;
+	float	*buffer ;
+
+	/* Prevent compiler warnings. */
+	device = device ;
+	current_time = current_time ;
+	data_in = data_in ;
+	time_in = time_in ;
+	time_out = time_out ;
+
+	audio_data = (MacOSXAudioData*) client_data ;
+
+	size = data_out->mBuffers [0].mDataByteSize ;
+	sample_count = size / sizeof (float) ;
+
+	buffer = (float*) data_out->mBuffers [0].mData ;
+
+	if (audio_data->fake_stereo != 0)
+	{	read_count = sf_read_float (audio_data->sndfile, buffer, sample_count / 2) ;
+
+		for (k = read_count - 1 ; k >= 0 ; k--)
+		{	buffer [2 * k	] = buffer [k] ;
+			buffer [2 * k + 1] = buffer [k] ;
+			} ;
+		read_count *= 2 ;
+		}
+	else
+		read_count = sf_read_float (audio_data->sndfile, buffer, sample_count) ;
+
+	/* Fill the remainder with zeroes. */
+	if (read_count < sample_count)
+	{	if (audio_data->fake_stereo == 0)
+			memset (&(buffer [read_count]), 0, (sample_count - read_count) * sizeof (float)) ;
+		/* Tell the main application to terminate. */
+		audio_data->done_playing = SF_TRUE ;
+		} ;
+
+	return noErr ;
+} /* macosx_audio_out_callback */
+
+static void
+macosx_play (int argc, char *argv [])
+{	MacOSXAudioData 	audio_data ;
+	OSStatus	err ;
+	UInt32		count, buffer_size ;
+	int 		k ;
+
+	audio_data.fake_stereo = 0 ;
+	audio_data.device = kAudioDeviceUnknown ;
+
+	/*  get the default output device for the HAL */
+	count = sizeof (AudioDeviceID) ;
+	if ((err = AudioHardwareGetProperty (kAudioHardwarePropertyDefaultOutputDevice,
+				&count, (void *) &(audio_data.device))) != noErr)
+	{	printf ("AudioHardwareGetProperty (kAudioDevicePropertyDefaultOutputDevice) failed.\n") ;
+		return ;
+		} ;
+
+	/*  get the buffersize that the default device uses for IO */
+	count = sizeof (UInt32) ;
+	if ((err = AudioDeviceGetProperty (audio_data.device, 0, false, kAudioDevicePropertyBufferSize,
+				&count, &buffer_size)) != noErr)
+	{	printf ("AudioDeviceGetProperty (kAudioDevicePropertyBufferSize) failed.\n") ;
+		return ;
+		} ;
+
+	/*  get a description of the data format used by the default device */
+	count = sizeof (AudioStreamBasicDescription) ;
+	if ((err = AudioDeviceGetProperty (audio_data.device, 0, false, kAudioDevicePropertyStreamFormat,
+				&count, &(audio_data.format))) != noErr)
+	{	printf ("AudioDeviceGetProperty (kAudioDevicePropertyStreamFormat) failed.\n") ;
+		return ;
+		} ;
+
+	/* Base setup completed. Now play files. */
+	for (k = 1 ; k < argc ; k++)
+	{	memset (&(audio_data.sfinfo), 0, sizeof (audio_data.sfinfo)) ;
+
+		printf ("Playing %s\n", argv [k]) ;
+		if (! (audio_data.sndfile = sf_open (argv [k], SFM_READ, &(audio_data.sfinfo))))
+		{	puts (sf_strerror (NULL)) ;
+			continue ;
+			} ;
+
+		if (audio_data.sfinfo.channels < 1 || audio_data.sfinfo.channels > 2)
+		{	printf ("Error : channels = %d.\n", audio_data.sfinfo.channels) ;
+			continue ;
+			} ;
+
+		audio_data.format.mSampleRate = audio_data.sfinfo.samplerate ;
+
+		if (audio_data.sfinfo.channels == 1)
+		{	audio_data.format.mChannelsPerFrame = 2 ;
+			audio_data.fake_stereo = 1 ;
+			}
+		else
+		audio_data.format.mChannelsPerFrame = audio_data.sfinfo.channels ;
+
+		if ((err = AudioDeviceSetProperty (audio_data.device, NULL, 0, false, kAudioDevicePropertyStreamFormat,
+					sizeof (AudioStreamBasicDescription), &(audio_data.format))) != noErr)
+		{	printf ("AudioDeviceSetProperty (kAudioDevicePropertyStreamFormat) failed.\n") ;
+			return ;
+			} ;
+
+		/*  we want linear pcm */
+		if (audio_data.format.mFormatID != kAudioFormatLinearPCM)
+			return ;
+
+		/* Fire off the device. */
+		if ((err = AudioDeviceAddIOProc (audio_data.device, macosx_audio_out_callback,
+				(void *) &audio_data)) != noErr)
+		{	printf ("AudioDeviceAddIOProc failed.\n") ;
+			return ;
+			} ;
+
+		err = AudioDeviceStart (audio_data.device, macosx_audio_out_callback) ;
+		if	(err != noErr)
+			return ;
+
+		audio_data.done_playing = SF_FALSE ;
+
+		while (audio_data.done_playing == SF_FALSE)
+			usleep (10 * 1000) ; /* 10 000 milliseconds. */
+
+		if ((err = AudioDeviceStop (audio_data.device, macosx_audio_out_callback)) != noErr)
+		{	printf ("AudioDeviceStop failed.\n") ;
+			return ;
+			} ;
+
+		err = AudioDeviceRemoveIOProc (audio_data.device, macosx_audio_out_callback) ;
+		if (err != noErr)
+		{	printf ("AudioDeviceRemoveIOProc failed.\n") ;
+			return ;
+			} ;
+
+		sf_close (audio_data.sndfile) ;
+		} ;
+
+	return ;
+} /* macosx_play, AudioHardware implementation */
+
+#endif /* OSX_DARWIN_VERSION > 0 && OSX_DARWIN_VERSION <= 10 */
+
+/*------------------------------------------------------------------------------
+**	Win32 functions for playing a sound.
+**
+**	This API sucks. Its needlessly complicated and is *WAY* too loose with
+**	passing pointers around in integers and using char* pointers to
+**  point to data instead of short*. It plain sucks!
+*/
+
+#if (OS_IS_WIN32 == 1)
+
+#define	WIN32_BUFFER_LEN	(1 << 15)
+
+typedef struct
+{	HWAVEOUT	hwave ;
+	WAVEHDR		whdr [2] ;
+
+	CRITICAL_SECTION	mutex ;		/* to control access to BuffersInUSe */
+	HANDLE		Event ;			/* signal that a buffer is free */
+
+	short		buffer [WIN32_BUFFER_LEN / sizeof (short)] ;
+	int			current, bufferlen ;
+	int			BuffersInUse ;
+
+	SNDFILE 	*sndfile ;
+	SF_INFO 	sfinfo ;
+
+	sf_count_t	remaining ;
+} Win32_Audio_Data ;
+
+
+static void
+win32_play_data (Win32_Audio_Data *audio_data)
+{	int thisread, readcount ;
+
+	/* fill a buffer if there is more data and we can read it sucessfully */
+	readcount = (audio_data->remaining > audio_data->bufferlen) ? audio_data->bufferlen : (int) audio_data->remaining ;
+
+	thisread = (int) sf_read_short (audio_data->sndfile, (short *) (audio_data->whdr [audio_data->current].lpData), readcount) ;
+
+	audio_data->remaining -= thisread ;
+
+	if (thisread > 0)
+	{	/* Fix buffer length if this is only a partial block. */
+		if (thisread < audio_data->bufferlen)
+			audio_data->whdr [audio_data->current].dwBufferLength = thisread * sizeof (short) ;
+
+		/* Queue the WAVEHDR */
+		waveOutWrite (audio_data->hwave, (LPWAVEHDR) &(audio_data->whdr [audio_data->current]), sizeof (WAVEHDR)) ;
+
+		/* count another buffer in use */
+		EnterCriticalSection (&audio_data->mutex) ;
+		audio_data->BuffersInUse ++ ;
+		LeaveCriticalSection (&audio_data->mutex) ;
+
+		/* use the other buffer next time */
+		audio_data->current = (audio_data->current + 1) % 2 ;
+		} ;
+
+	return ;
+} /* win32_play_data */
+
+static void CALLBACK
+win32_audio_out_callback (HWAVEOUT hwave, UINT msg, DWORD_PTR data, DWORD param1, DWORD param2)
+{	Win32_Audio_Data	*audio_data ;
+
+	/* Prevent compiler warnings. */
+	(void) hwave ;
+	(void) param1 ;
+	(void) param2 ;
+
+	if (data == 0)
+		return ;
+
+	/*
+	** I consider this technique of passing a pointer via an integer as
+	** fundamentally broken but thats the way microsoft has defined the
+	** interface.
+	*/
+	audio_data = (Win32_Audio_Data*) data ;
+
+	/* let main loop know a buffer is free */
+	if (msg == MM_WOM_DONE)
+	{	EnterCriticalSection (&audio_data->mutex) ;
+		audio_data->BuffersInUse -- ;
+		LeaveCriticalSection (&audio_data->mutex) ;
+		SetEvent (audio_data->Event) ;
+		} ;
+
+	return ;
+} /* win32_audio_out_callback */
+
+static void
+win32_play (int argc, char *argv [])
+{	Win32_Audio_Data	audio_data ;
+
+	WAVEFORMATEX wf ;
+	int	k, error ;
+
+	audio_data.sndfile = NULL ;
+	audio_data.hwave = 0 ;
+
+	for (k = 1 ; k < argc ; k++)
+	{	printf ("Playing %s\n", argv [k]) ;
+
+		if (! (audio_data.sndfile = sf_open (argv [k], SFM_READ, &(audio_data.sfinfo))))
+		{	puts (sf_strerror (NULL)) ;
+			continue ;
+			} ;
+
+		audio_data.remaining = audio_data.sfinfo.frames * audio_data.sfinfo.channels ;
+		audio_data.current = 0 ;
+
+		InitializeCriticalSection (&audio_data.mutex) ;
+		audio_data.Event = CreateEvent (0, FALSE, FALSE, 0) ;
+
+		wf.nChannels = audio_data.sfinfo.channels ;
+		wf.wFormatTag = WAVE_FORMAT_PCM ;
+		wf.cbSize = 0 ;
+		wf.wBitsPerSample = 16 ;
+
+		wf.nSamplesPerSec = audio_data.sfinfo.samplerate ;
+
+		wf.nBlockAlign = audio_data.sfinfo.channels * sizeof (short) ;
+
+		wf.nAvgBytesPerSec = wf.nBlockAlign * wf.nSamplesPerSec ;
+
+		error = waveOutOpen (&(audio_data.hwave), WAVE_MAPPER, &wf, (DWORD_PTR) win32_audio_out_callback,
+							(DWORD_PTR) &audio_data, CALLBACK_FUNCTION) ;
+		if (error)
+		{	puts ("waveOutOpen failed.") ;
+			audio_data.hwave = 0 ;
+			continue ;
+			} ;
+
+		audio_data.whdr [0].lpData = (char*) audio_data.buffer ;
+		audio_data.whdr [1].lpData = ((char*) audio_data.buffer) + sizeof (audio_data.buffer) / 2 ;
+
+		audio_data.whdr [0].dwBufferLength = sizeof (audio_data.buffer) / 2 ;
+		audio_data.whdr [1].dwBufferLength = sizeof (audio_data.buffer) / 2 ;
+
+		audio_data.whdr [0].dwFlags = 0 ;
+		audio_data.whdr [1].dwFlags = 0 ;
+
+		/* length of each audio buffer in samples */
+		audio_data.bufferlen = sizeof (audio_data.buffer) / 2 / sizeof (short) ;
+
+		/* Prepare the WAVEHDRs */
+		if ((error = waveOutPrepareHeader (audio_data.hwave, &(audio_data.whdr [0]), sizeof (WAVEHDR))))
+		{	printf ("waveOutPrepareHeader [0] failed : %08X\n", error) ;
+			waveOutClose (audio_data.hwave) ;
+			continue ;
+			} ;
+
+		if ((error = waveOutPrepareHeader (audio_data.hwave, &(audio_data.whdr [1]), sizeof (WAVEHDR))))
+		{	printf ("waveOutPrepareHeader [1] failed : %08X\n", error) ;
+			waveOutUnprepareHeader (audio_data.hwave, &(audio_data.whdr [0]), sizeof (WAVEHDR)) ;
+			waveOutClose (audio_data.hwave) ;
+			continue ;
+			} ;
+
+		/* Fill up both buffers with audio data */
+		audio_data.BuffersInUse = 0 ;
+		win32_play_data (&audio_data) ;
+		win32_play_data (&audio_data) ;
+
+		/* loop until both buffers are released */
+		while (audio_data.BuffersInUse > 0)
+		{
+			/* wait for buffer to be released */
+			WaitForSingleObject (audio_data.Event, INFINITE) ;
+
+			/* refill the buffer if there is more data to play */
+			win32_play_data (&audio_data) ;
+			} ;
+
+		waveOutUnprepareHeader (audio_data.hwave, &(audio_data.whdr [0]), sizeof (WAVEHDR)) ;
+		waveOutUnprepareHeader (audio_data.hwave, &(audio_data.whdr [1]), sizeof (WAVEHDR)) ;
+
+		waveOutClose (audio_data.hwave) ;
+		audio_data.hwave = 0 ;
+
+		DeleteCriticalSection (&audio_data.mutex) ;
+
+		sf_close (audio_data.sndfile) ;
+		} ;
+
+} /* win32_play */
+
+#endif /* Win32 */
+
+/*------------------------------------------------------------------------------
+**	OpenBSD's sndio.
+*/
+
+#if HAVE_SNDIO_H
+
+static void
+sndio_play (int argc, char *argv [])
+{	struct sio_hdl	*hdl ;
+	struct sio_par	par ;
+	short buffer [BUFFER_LEN] ;
+	SNDFILE	*sndfile ;
+	SF_INFO	sfinfo ;
+	int		k, readcount ;
+
+	for (k = 1 ; k < argc ; k++)
+	{	printf ("Playing %s\n", argv [k]) ;
+		if (! (sndfile = sf_open (argv [k], SFM_READ, &sfinfo)))
+		{	puts (sf_strerror (NULL)) ;
+			continue ;
+			} ;
+
+		if (sfinfo.channels < 1 || sfinfo.channels > 2)
+		{	printf ("Error : channels = %d.\n", sfinfo.channels) ;
+			continue ;
+			} ;
+
+		if ((hdl = sio_open (NULL, SIO_PLAY, 0)) == NULL)
+		{	fprintf (stderr, "open sndio device failed") ;
+			return ;
+			} ;
+
+		sio_initpar (&par) ;
+		par.rate = sfinfo.samplerate ;
+		par.pchan = sfinfo.channels ;
+		par.bits = 16 ;
+		par.sig = 1 ;
+		par.le = SIO_LE_NATIVE ;
+
+		if (! sio_setpar (hdl, &par) || ! sio_getpar (hdl, &par))
+		{	fprintf (stderr, "set sndio params failed") ;
+			return ;
+			} ;
+
+		if (! sio_start (hdl))
+		{	fprintf (stderr, "sndio start failed") ;
+			return ;
+			} ;
+
+		while ((readcount = sf_read_short (sndfile, buffer, BUFFER_LEN)))
+			sio_write (hdl, buffer, readcount * sizeof (short)) ;
+
+		sio_close (hdl) ;
+		} ;
+
+	return ;
+} /* sndio_play */
+
+#endif /* sndio */
+
+/*------------------------------------------------------------------------------
+**	Solaris.
+*/
+
+#if (defined (sun) && defined (unix)) /* ie Solaris */
+
+static void
+solaris_play (int argc, char *argv [])
+{	static short 	buffer [BUFFER_LEN] ;
+	audio_info_t	audio_info ;
+	SNDFILE			*sndfile ;
+	SF_INFO			sfinfo ;
+	unsigned long	delay_time ;
+	long			k, start_count, output_count, write_count, read_count ;
+	int				audio_fd, error, done ;
+
+	for (k = 1 ; k < argc ; k++)
+	{	printf ("Playing %s\n", argv [k]) ;
+		if (! (sndfile = sf_open (argv [k], SFM_READ, &sfinfo)))
+		{	puts (sf_strerror (NULL)) ;
+			continue ;
+			} ;
+
+		if (sfinfo.channels < 1 || sfinfo.channels > 2)
+		{	printf ("Error : channels = %d.\n", sfinfo.channels) ;
+			continue ;
+			} ;
+
+		/* open the audio device - write only, non-blocking */
+		if ((audio_fd = open ("/dev/audio", O_WRONLY | O_NONBLOCK)) < 0)
+		{	perror ("open (/dev/audio) failed") ;
+			return ;
+			} ;
+
+		/*	Retrive standard values. */
+		AUDIO_INITINFO (&audio_info) ;
+
+		audio_info.play.sample_rate = sfinfo.samplerate ;
+		audio_info.play.channels = sfinfo.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 (audio_fd, AUDIO_SETINFO, &audio_info)))
+		{	perror ("ioctl (AUDIO_SETINFO) failed") ;
+			return ;
+			} ;
+
+		/* Delay time equal to 1/4 of a buffer in microseconds. */
+		delay_time = (BUFFER_LEN * 1000000) / (audio_info.play.sample_rate * 4) ;
+
+		done = 0 ;
+		while (! done)
+		{	read_count = sf_read_short (sndfile, buffer, BUFFER_LEN) ;
+			if (read_count < BUFFER_LEN)
+			{	memset (&(buffer [read_count]), 0, (BUFFER_LEN - read_count) * sizeof (short)) ;
+				/* Tell the main application to terminate. */
+				done = SF_TRUE ;
+				} ;
+
+			start_count = 0 ;
+			output_count = BUFFER_LEN * sizeof (short) ;
+
+			while (output_count > 0)
+			{	/* write as much data as possible */
+				write_count = write (audio_fd, &(buffer [start_count]), output_count) ;
+				if (write_count > 0)
+				{	output_count -= write_count ;
+					start_count += write_count ;
+					}
+				else
+				{	/*	Give the audio output time to catch up. */
+					usleep (delay_time) ;
+					} ;
+				} ; /* while (outpur_count > 0) */
+			} ; /* while (! done) */
+
+		close (audio_fd) ;
+		} ;
+
+	return ;
+} /* solaris_play */
+
+#endif /* Solaris */
+
+/*==============================================================================
+**	Main function.
+*/
+
+int
+main (int argc, char *argv [])
+{
+	if (argc < 2)
+	{
+		printf ("\nUsage : %s <input sound file>\n\n", program_name (argv [0])) ;
+		printf ("Using %s.\n\n", sf_version_string ()) ;
+#if (OS_IS_WIN32 == 1)
+		printf ("This is a Unix style command line application which\n"
+				"should be run in a MSDOS box or Command Shell window.\n\n") ;
+		printf ("Sleeping for 5 seconds before exiting.\n\n") ;
+
+		Sleep (5 * 1000) ;
+#endif
+		return 1 ;
+		} ;
+
+#if defined (__ANDROID__)
+	puts ("*** Playing sound not yet supported on Android.") ;
+	puts ("*** Please feel free to submit a patch.") ;
+	return 1 ;
+#elif defined (__linux__)
+	#if HAVE_ALSA_ASOUNDLIB_H
+		if (access ("/proc/asound/cards", R_OK) == 0)
+			alsa_play (argc, argv) ;
+		else
+	#endif
+		opensoundsys_play (argc, argv) ;
+#elif defined (__FreeBSD_kernel__) || defined (__FreeBSD__)
+	opensoundsys_play (argc, argv) ;
+#elif (defined (__MACH__) && defined (__APPLE__) && OSX_DARWIN_VERSION <= 11)
+	macosx_play (argc, argv) ;
+#elif HAVE_SNDIO_H
+	sndio_play (argc, argv) ;
+#elif (defined (sun) && defined (unix))
+	solaris_play (argc, argv) ;
+#elif (OS_IS_WIN32 == 1)
+	win32_play (argc, argv) ;
+#elif (defined (__MACH__) && defined (__APPLE__) && OSX_DARWIN_VERSION > 11)
+	printf ("OS X 10.8 and later have a new Audio API.\n") ;
+	printf ("Someone needs to write code to use that API.\n") ;
+	return 1 ;
+#elif defined (__BEOS__)
+	printf ("This program cannot be compiled on BeOS.\n") ;
+	printf ("Instead, compile the file sfplay_beos.cpp.\n") ;
+	return 1 ;
+#else
+	puts ("*** Playing sound not yet supported on this platform.") ;
+	puts ("*** Please feel free to submit a patch.") ;
+	return 1 ;
+#endif
+
+	return 0 ;
+} /* main */
+