mirror of
https://gitee.com/openharmony/third_party_libsnd
synced 2024-11-23 18:09:59 +00:00
901 lines
25 KiB
C
901 lines
25 KiB
C
/*
|
|
** Copyright (C) 1999-2004 Erik de Castro Lopo <erikd@mega-nerd.com>
|
|
**
|
|
** This program is free software; you can redistribute it and/or modify
|
|
** it under the terms of the GNU General Public License as published by
|
|
** the Free Software Foundation; either version 2 of the License, or
|
|
** (at your option) any later version.
|
|
**
|
|
** This program is distributed in the hope that it will be useful,
|
|
** but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
** MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
|
|
** GNU General Public License for more details.
|
|
**
|
|
** You should have received a copy of the GNU General Public License
|
|
** along with this program; if not, write to the Free Software
|
|
** Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA 02111-1307, USA.
|
|
*/
|
|
|
|
#include "config.h"
|
|
|
|
#include <stdio.h>
|
|
#include <stdlib.h>
|
|
#include <string.h>
|
|
#include <errno.h>
|
|
|
|
#if HAVE_UNISTD_H
|
|
#include <unistd.h>
|
|
#endif
|
|
|
|
#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 (__linux__)
|
|
#include <fcntl.h>
|
|
#include <sys/ioctl.h>
|
|
#include <sys/soundcard.h>
|
|
|
|
#elif (defined (__MACH__) && defined (__APPLE__))
|
|
#include <Carbon.h>
|
|
#include <CoreAudio/AudioHardware.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
|
|
|
|
#include <sndfile.h>
|
|
|
|
#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, int srate) ;
|
|
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 ;
|
|
|
|
puts (__func__) ;
|
|
|
|
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 ;
|
|
} ;
|
|
|
|
alsa_dev = alsa_open (sfinfo.channels, sfinfo.samplerate) ;
|
|
|
|
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_close (alsa_dev) ;
|
|
|
|
sf_close (sndfile) ;
|
|
} ;
|
|
|
|
return ;
|
|
} /* alsa_play */
|
|
|
|
static snd_pcm_t *
|
|
alsa_open (int channels, int samplerate)
|
|
{ const char * device = "plughw:0" ;
|
|
snd_pcm_t *alsa_dev ;
|
|
snd_pcm_hw_params_t *hw_params ;
|
|
snd_pcm_uframes_t buffer_size, xfer_align, start_threshold ;
|
|
snd_pcm_uframes_t alsa_period_size, alsa_buffer_frames ;
|
|
snd_pcm_sw_params_t *sw_params ;
|
|
|
|
int err ;
|
|
|
|
alsa_period_size = 512 ;
|
|
alsa_buffer_frames = 3 * 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)) ;
|
|
return NULL ;
|
|
} ;
|
|
|
|
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)) ;
|
|
return NULL ;
|
|
} ;
|
|
|
|
if ((err = snd_pcm_hw_params_any (alsa_dev, hw_params)) < 0)
|
|
{ fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n", snd_strerror (err)) ;
|
|
return NULL ;
|
|
} ;
|
|
|
|
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)) ;
|
|
return NULL ;
|
|
} ;
|
|
|
|
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)) ;
|
|
return NULL ;
|
|
} ;
|
|
|
|
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)) ;
|
|
return NULL ;
|
|
} ;
|
|
|
|
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)) ;
|
|
return NULL ;
|
|
} ;
|
|
|
|
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)) ;
|
|
return NULL ;
|
|
} ;
|
|
|
|
|
|
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)) ;
|
|
return NULL ;
|
|
} ;
|
|
|
|
if ((err = snd_pcm_hw_params (alsa_dev, hw_params)) < 0)
|
|
{ fprintf (stderr, "cannot set parameters (%s)\n", snd_strerror (err)) ;
|
|
return NULL ;
|
|
} ;
|
|
|
|
/* 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) ;
|
|
return NULL ;
|
|
} ;
|
|
|
|
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)) ;
|
|
return NULL ;
|
|
} ;
|
|
|
|
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)) ;
|
|
return NULL ;
|
|
} ;
|
|
|
|
/* 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_get_xfer_align (sw_params, &xfer_align)) < 0)
|
|
{ fprintf (stderr, "cannot get xfer align (%s)\n", snd_strerror (err)) ;
|
|
return NULL ;
|
|
} ;
|
|
|
|
/* round up to closest transfer boundary */
|
|
start_threshold = (buffer_size / xfer_align) * xfer_align ;
|
|
if (start_threshold < 1)
|
|
start_threshold = 1 ;
|
|
if ((err = snd_pcm_sw_params_set_start_threshold (alsa_dev, sw_params, start_threshold)) < 0)
|
|
{ fprintf (stderr, "cannot set start threshold (%s)\n", snd_strerror (err)) ;
|
|
return NULL ;
|
|
} ;
|
|
|
|
if ((err = snd_pcm_sw_params (alsa_dev, sw_params)) != 0)
|
|
{ fprintf (stderr, "%s: snd_pcm_sw_params: %s", __func__, snd_strerror (err)) ;
|
|
return NULL ;
|
|
} ;
|
|
snd_pcm_sw_params_free (sw_params) ;
|
|
|
|
snd_pcm_reset (alsa_dev) ;
|
|
|
|
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 ;
|
|
|
|
snd_pcm_status_t *status ;
|
|
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)
|
|
{ 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") ;
|
|
} ;
|
|
|
|
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 (__linux__)
|
|
|
|
static int linux_open_dsp_device (int channels, int srate) ;
|
|
|
|
static void
|
|
linux_play (int argc, char *argv [])
|
|
{ static short buffer [BUFFER_LEN] ;
|
|
SNDFILE *sndfile ;
|
|
SF_INFO sfinfo ;
|
|
int k, audio_device, 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 ;
|
|
} ;
|
|
|
|
audio_device = linux_open_dsp_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] ;
|
|
write (audio_device, buffer, readcount * sizeof (short)) ;
|
|
} ;
|
|
}
|
|
else
|
|
{ while ((readcount = sf_read_short (sndfile, buffer, BUFFER_LEN)))
|
|
write (audio_device, buffer, readcount * sizeof (short)) ;
|
|
} ;
|
|
|
|
close (audio_device) ;
|
|
|
|
sf_close (sndfile) ;
|
|
} ;
|
|
|
|
return ;
|
|
} /* linux_play */
|
|
|
|
static int
|
|
linux_open_dsp_device (int channels, int srate)
|
|
{ int fd, stereo, temp, error ;
|
|
|
|
if ((fd = open ("/dev/dsp", O_WRONLY, 0)) == -1 &&
|
|
(fd = open ("/dev/sound/dsp", O_WRONLY, 0)) == -1)
|
|
{ perror("linux_open_dsp_device : open ") ;
|
|
exit (1) ;
|
|
} ;
|
|
|
|
stereo = 0 ;
|
|
if (ioctl (fd, SNDCTL_DSP_STEREO, &stereo) == -1)
|
|
{ /* Fatal error */
|
|
perror("linux_open_dsp_device : stereo ") ;
|
|
exit (1);
|
|
} ;
|
|
|
|
if (ioctl (fd, SNDCTL_DSP_RESET, 0))
|
|
{ perror ("linux_open_dsp_device : reset ") ;
|
|
exit (1) ;
|
|
} ;
|
|
|
|
temp = 16 ;
|
|
if ((error = ioctl (fd, SOUND_PCM_WRITE_BITS, &temp)) != 0)
|
|
{ perror ("linux_open_dsp_device : bitwidth ") ;
|
|
exit (1) ;
|
|
} ;
|
|
|
|
if ((error = ioctl (fd, SOUND_PCM_WRITE_CHANNELS, &channels)) != 0)
|
|
{ perror ("linux_open_dsp_device : channels ") ;
|
|
exit (1) ;
|
|
} ;
|
|
|
|
if ((error = ioctl (fd, SOUND_PCM_WRITE_RATE, &srate)) != 0)
|
|
{ perror ("linux_open_dsp_device : sample rate ") ;
|
|
exit (1) ;
|
|
} ;
|
|
|
|
if ((error = ioctl (fd, SNDCTL_DSP_SYNC, 0)) != 0)
|
|
{ perror ("linux_open_dsp_device : sync ") ;
|
|
exit (1) ;
|
|
} ;
|
|
|
|
return fd ;
|
|
} /* linux_open_dsp_device */
|
|
|
|
#endif /* __linux__ */
|
|
|
|
/*------------------------------------------------------------------------------
|
|
** Mac OS X functions for playing a sound.
|
|
*/
|
|
|
|
#if (defined (__MACH__) && defined (__APPLE__)) /* MacOSX */
|
|
|
|
typedef struct
|
|
{ AudioStreamBasicDescription format;
|
|
|
|
UInt32 buf_size;
|
|
AudioDeviceID device;
|
|
|
|
SNDFILE *sndfile ;
|
|
SF_INFO sfinfo ;
|
|
|
|
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 ;
|
|
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 ;
|
|
|
|
read_count = sf_read_float (audio_data->sndfile, buffer, sample_count) ;
|
|
|
|
if (read_count < sample_count)
|
|
{ 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.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 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 (AudioDeviceGetProperty) 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++)
|
|
{ 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 ;
|
|
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 */
|
|
|
|
#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 (OS_IS_WIN32 == 1)
|
|
|
|
#define WIN32_BUFFER_LEN (1<<15)
|
|
|
|
typedef struct
|
|
{ HWAVEOUT hwave ;
|
|
WAVEHDR whdr [2] ;
|
|
|
|
HANDLE Event ;
|
|
|
|
short buffer [WIN32_BUFFER_LEN / sizeof (short)] ;
|
|
int current, bufferlen ;
|
|
|
|
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 ;
|
|
|
|
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 is only a partial block. */
|
|
if (thisread * SIGNED_SIZEOF (short) < 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)) ;
|
|
}
|
|
else
|
|
{ /* Stop playback */
|
|
waveOutPause (audio_data->hwave) ;
|
|
|
|
SetEvent (audio_data->Event) ;
|
|
} ;
|
|
|
|
audio_data->current = (audio_data->current + 1) % 2 ;
|
|
|
|
} /* win32_play_data */
|
|
|
|
static DWORD CALLBACK
|
|
win32_audio_out_callback (HWAVEOUT hwave, UINT msg, DWORD data, DWORD param1, DWORD param2)
|
|
{ Win32_Audio_Data *audio_data ;
|
|
|
|
if (data == 0)
|
|
return 1 ;
|
|
|
|
/* Preent compiler warnings. */
|
|
hwave = hwave ;
|
|
param1 = param2 ;
|
|
|
|
/*
|
|
** 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 ;
|
|
|
|
if (msg == MM_WOM_DONE)
|
|
win32_play_data (audio_data) ;
|
|
|
|
return 0 ;
|
|
} /* 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.current = 0 ;
|
|
|
|
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) win32_audio_out_callback,
|
|
(DWORD) &audio_data, CALLBACK_FUNCTION) ;
|
|
if (error)
|
|
{ puts ("waveOutOpen failed.") ;
|
|
audio_data.hwave = 0 ;
|
|
continue ;
|
|
} ;
|
|
|
|
waveOutPause (audio_data.hwave) ;
|
|
|
|
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.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 ;
|
|
} ;
|
|
|
|
waveOutRestart (audio_data.hwave) ;
|
|
|
|
/* Need to call this twice to queue up enough audio. */
|
|
win32_play_data (&audio_data) ;
|
|
win32_play_data (&audio_data) ;
|
|
|
|
/* Wait for playback to finish. My callback notifies me when all wave data has been played */
|
|
WaitForSingleObject (audio_data.Event, INFINITE);
|
|
|
|
waveOutPause (audio_data.hwave) ;
|
|
waveOutReset (audio_data.hwave);
|
|
|
|
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 ;
|
|
|
|
sf_close (audio_data.sndfile) ;
|
|
} ;
|
|
|
|
} /* win32_play */
|
|
|
|
#endif /* Win32 */
|
|
|
|
/*------------------------------------------------------------------------------
|
|
** 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", argv [0]) ;
|
|
#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") ;
|
|
|
|
/* This is the officially blessed by microsoft way but I can't get
|
|
** it to link.
|
|
** Sleep (15) ;
|
|
** Instead, use this:
|
|
*/
|
|
_sleep (5 * 1000) ;
|
|
#endif
|
|
return 1 ;
|
|
} ;
|
|
|
|
#if defined (__linux__)
|
|
#if HAVE_ALSA_ASOUNDLIB_H
|
|
if (access ("/proc/asound/cards", R_OK) == 0)
|
|
alsa_play (argc, argv) ;
|
|
else
|
|
#endif
|
|
linux_play (argc, argv) ;
|
|
#elif (defined (__MACH__) && defined (__APPLE__))
|
|
macosx_play (argc, argv) ;
|
|
#elif (defined (sun) && defined (unix))
|
|
solaris_play (argc, argv) ;
|
|
#elif (OS_IS_WIN32 == 1)
|
|
win32_play (argc, argv) ;
|
|
#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 */
|
|
/*
|
|
** Do not edit or modify anything in this comment block.
|
|
** The arch-tag line is a file identity tag for the GNU Arch
|
|
** revision control system.
|
|
**
|
|
** arch-tag: 8fc4110d-6cec-4e03-91df-0f384cabedac
|
|
*/
|