/*
** Copyright (C) 1999-2015 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>
#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>
#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.
*/
/* 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 ;
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 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__))
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 */