Torque3D/Engine/lib/libsndfile/programs/sndfile-play.c
marauder2k7 a745fc3757 Initial commit
added libraries:
opus
flac
libsndfile

updated:
libvorbis
libogg
openal

- Everything works as expected for now. Bare in mind libsndfile needed the check for whether or not it could find the xiph libraries removed in order for this to work.
2024-03-21 17:33:47 +00:00

861 lines
24 KiB
C

/*
** Copyright (C) 1999-2018 Erik de Castro Lopo <erikd@mega-nerd.com>
**
** 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>
#else
#include "sf_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__) || defined (__riscos__)
#include <fcntl.h>
#include <sys/ioctl.h>
#include <sys/soundcard.h>
#elif HAVE_SNDIO_H
#include <sndio.h>
#elif (defined (sun) && defined (unix)) || defined(__NetBSD__)
#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 > 1.0)
scale = 1.0 / scale ;
else
scale = 1.0 ;
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 ;
#if defined ESTRPIPE && ESTRPIPE != EPIPE
case -ESTRPIPE :
fprintf (stderr, "alsa_write_float: Suspend event.n") ;
return 0 ;
break ;
#endif
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__) || defined (__riscos__))
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.
*/
/* MacOSX 10.8 use a new Audio API. Someone needs to write some code for it. */
/*------------------------------------------------------------------------------
** 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 ;
short *lpData = (short *) (void *) audio_data->whdr [audio_data->current].lpData ;
thisread = (int) sf_read_short (audio_data->sndfile, 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)) || defined(__NetBSD__)
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 ;
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 or NetBSD */
/*==============================================================================
** 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__) || defined (__riscos__)
opensoundsys_play (argc, argv) ;
#elif HAVE_SNDIO_H
sndio_play (argc, argv) ;
#elif (defined (sun) && defined (unix)) || defined(__NetBSD__)
solaris_play (argc, argv) ;
#elif (OS_IS_WIN32 == 1)
win32_play (argc, argv) ;
#else
puts ("*** Playing sound not supported on this platform.") ;
puts ("*** Please feel free to submit a patch.") ;
return 1 ;
#endif
return 0 ;
} /* main */