File: [Development] / inventor / apps / demos / maze / PlayClass.c++ (download)
Revision 1.2, Tue Sep 25 00:45:25 2001 UTC (16 years, 1 month ago) by jlim
Branch: MAIN
CVS Tags: release-2_1_5-9, release-2_1_5-8, release-2_1_5-10, HEAD
Changes since 1.1: +2 -2
lines
Various changes:
* Fixed Bug 63 and 64.
* Handled nonstandard sed location.
* Used for-loop in man page install.
* Included casts for 64-bit builds.
* Added placeholder for FreeBSD options.
* Included unistd.h for getopt() and stdlib.h for malloc().
* Implemented SoText[23] workaround for glibc-2.2.* iconv().
* Split long lines in SoHandleBoxDraggerGeom.h and
SoTransformerDraggerGeom.h in lib/interaction/src/draggers/geom.
* Added IV_NO_OVERLAYS/OIV_NO_OVERLAYS variables to disable overlay planes.
|
/*
*
* Copyright (C) 2000 Silicon Graphics, Inc. All Rights Reserved.
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* This library 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
* Lesser General Public License for more details.
*
* Further, this software is distributed without any warranty that it is
* free of the rightful claim of any third person regarding infringement
* or the like. Any license provided herein, whether implied or
* otherwise, applies only to this software file. Patent licenses, if
* any, provided herein do not apply to combinations of this program with
* other software, or any other product whatsoever.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
* Contact information: Silicon Graphics, Inc., 1600 Amphitheatre Pkwy,
* Mountain View, CA 94043, or:
*
* http://www.sgi.com
*
* For further information regarding this notice, see:
*
* http://oss.sgi.com/projects/GenInfo/NoticeExplan/
*
*/
#include <stdio.h>
#include <fcntl.h>
#include <errno.h>
#include <sys/types.h>
#include <sys/stat.h>
#include <sys/prctl.h>
#include <sys/wait.h>
#include <stdlib.h>
#include <math.h>
#include <string.h>
#include <unistd.h>
#include <CC/osfcn.h>
#include <signal.h>
#include <invent.h>
#include "PlayClass.h"
//#define DEBUG
/*
* local defines
*/
#ifndef FALSE
#define FALSE (0)
#define TRUE (!FALSE)
#endif
#ifndef ABS
#define ABS(a) ((a)>0.0?(a):-(a))
#endif
#ifndef MIN
#define MIN(a, b) ((a)<(b)?(a):(b))
#endif
#ifndef MAX
#define MAX(a, b) ((a)>(b)?(a):(b))
#endif
#define RANGE(a, b1, b2) \
\
((b1)<(b2)? \
((a)<(b1)? \
(b1): \
((a)>(b2)? \
(b2):(a))): \
((a)<(b2)? \
(b2): \
((a)>(b1)? \
(b1):(a))))
#ifndef INTERP
#define INTERP(x1, x2, a) ((a)*(x2)+(1.0-(a))*(x1))
#endif
PlayClass::PlayClass(char *mname)
{
myname = mname;
rude = FALSE;
quiet = FALSE;
timed = FALSE;
verbose = FALSE;
#ifdef DEBUG
verbose = TRUE;
#endif
caught_sigint = FALSE;
filename = NULL;
doneCallback = NULL;
loop = FALSE;
fading_in = FALSE;
fading_out = FALSE;
in_time = 0.0;
out_time = 0.0;
fade_level = 0.0;
max_level = 1.0;
isLive = FALSE;
isDone = TRUE;
sproced = FALSE;
signal(SIGCLD,SIG_IGN); /* ensure no zombie child processes */
}
/*
* attenuatesamps
*
* tristram, 1993
*
* put a ramp on a buffer of samples, starting at the current fade_level
* at a slope of 1 / fade_time.
*
* mode = 0 ==> fade in
* mode = 1 ==> fade out
* mode = 2 ==> don't adjust fade_level (just use max_level)
*
* this function has the side effect of setting fade_level
*
*/
void
PlayClass::attenuatesamps( void * sampbuf, long sampsize, long nsamps,
double secs_per_buf,
double max_level,
double * fade_level,
double fade_time,
int mode ) {
int i;
#ifdef DEBUG
fprintf(stderr, "attenuating max_level = %f, fade_level = %f, fade_time = %f\n", max_level, *fade_level, fade_time);
#endif
if ( verbose ) printf( "fade_level = %f\n", *fade_level );
if (( mode != 2 ) && ( fade_time == 0.0 )) {
if ( *fade_level == 0.0 ) *fade_level = 1.0;
else *fade_level = 0.0;
return;
}
for ( i = 0; i < nsamps; i++ ) {
switch ( sampsize ) {
case AL_SAMPLE_8:
((char *)sampbuf)[ i ] *= (char)(max_level * *fade_level);
break;
case AL_SAMPLE_16:
((short *)sampbuf)[ i ] *= (short)(max_level * *fade_level);
break;
case AL_SAMPLE_24:
((int32_t *)sampbuf)[ i ] *= max_level * *fade_level;
break;
}
if ( mode != 2 ) {
if ( fade_time == 0.0 ) {
if ( mode == 0 ) { /* fading in */
*fade_level = 1.0;
} else { /* fading out */
*fade_level = 0.0;
}
} else {
*fade_level += ( mode == 0 ? 1.0 : -1.0 )
* secs_per_buf / (fade_time * nsamps);
*fade_level = RANGE( *fade_level, 0.0, 1.0 );
}
}
}
}
int PlayClass::beginPlaying()
{
#ifdef DEBUG
fprintf(stderr, "beginPlaying()\n");
#endif
AFfilehandle audio_file;
ALconfig audio_port_config;
ALport audio_port;
int errseen = 0;
int result;
#ifdef DEBUG
fprintf(stderr, "checkpoint 1()\n");
#endif
if (isDone) isDone = FALSE;
else return -1;
isLive = TRUE;
fading_in = FALSE;
fading_out = FALSE;
fade_level = 0.0;
max_level = RANGE( max_level, 0.0, 1.0 );
if (in_time == 0.0) fade_level = 1.0;
#ifdef DEBUG
fprintf(stderr, "new max_level is %f\n", max_level);
#endif
if ((fd = open(filename, O_RDONLY)) < 0)
{
if (!quiet)
{
fprintf(stderr, "%s: failed to open file '%s' %s\n",
myname, filename, strerror(errno));
}
errseen = 1;
}
/*
* test the file descriptor to see whether we can attach an audio
* file handle to it
*/
else if (AFidentifyfd(fd) < 0)
{
if (!quiet)
{
fprintf(stderr,"%s: '%s' not an AIFF-C or AIFF file\n",
myname, filename);
}
errseen = 1;
}
/*
* attach an audio file handle to the file descriptor
*/
else if ((audio_file = AFopenfd(fd, "r", AF_NULL_FILESETUP))
== AF_NULL_FILEHANDLE)
{
if (!quiet)
{
fprintf(stderr, "%s: failed to open file '%s'\n",
myname, filename);
}
errseen = 1;
}
else
{
result = init_audio(audio_file, &audio_port, &audio_port_config);
if (result != -1)
{
play_audio_samps(audio_file, audio_port, audio_port_config);
} else {
// errseen = 1;
}
AFclosefile(audio_file);
ALfreeconfig(audio_port_config);
if (result != -1)
{
ALcloseport(audio_port);
}
free(sampbuf);
}
if (errseen) {
#ifdef DEBUG
fprintf(stderr, "error! returning -1\n");
#endif
isDone = 1;
if (sproced) exit(0);
return -1;
}
else {
#ifdef DEBUG
fprintf(stderr, "done\n");
#endif
if (doneCallback) {
#ifdef DEBUG
fprintf(stderr, "calling doneCallback\n");
#endif
doneCallback(userData, this);
}
isDone = 1;
if (sproced) {
#ifdef DEBUG
fprintf(stderr, "exiting zero");
#endif
exit(0);
}
#ifdef DEBUG
fprintf(stderr, "after exit(0)");
#endif
return 1;
}
}
/*
* initialize audio port and global state of IRIS Audio Processor
*/
int
PlayClass::init_audio(AFfilehandle audio_file, ALport *audio_port,
ALconfig *ap_config)
{
#ifdef DEBUG
fprintf(stderr, "init_audio(...)\n");
#endif
// long pvbuf[4];
long pvbuf[2];
long audio_rate;
long samp_wordsize;
AudioFormat_t vers;
AudioFormat_t samp_type;
samps_per_frame = (int) AFgetchannels(audio_file, AF_DEFAULT_TRACK);
file_rate = AFgetrate(audio_file, AF_DEFAULT_TRACK);
compression = AFgetcompression(audio_file, AF_DEFAULT_TRACK);
filefmt = AFgetfilefmt(audio_file, &vers);
AFgetsampfmt(audio_file, AF_DEFAULT_TRACK, &samp_type, &bits_per_samp);
/*
* need to determine whether audio is in use. if not, then we
* can just go ahead and be "rude."
*/
pvbuf[0] = AL_OUTPUT_COUNT;
// pvbuf[2] = AL_MONITOR_CTL;
// ALgetparams(AL_DEFAULT_DEVICE, pvbuf, 4);
ALgetparams(AL_DEFAULT_DEVICE, pvbuf, 2);
// if ((pvbuf[1] == 0) && (pvbuf[3] == AL_MONITOR_OFF)) {
if (pvbuf[1] == 0) {
rude = 1;
}
/*
* decide on output rate for the audio hardware
*/
switch((int32_t)file_rate)
{
case 48000: audio_rate = AL_RATE_48000; frames_per_sec = 48000; break;
case 44100: audio_rate = AL_RATE_44100; frames_per_sec = 44100; break;
case 32000: audio_rate = AL_RATE_32000; frames_per_sec = 32000; break;
case 22050: audio_rate = AL_RATE_22050; frames_per_sec = 22050; break;
case 16000: audio_rate = AL_RATE_16000; frames_per_sec = 16000; break;
case 11025: audio_rate = AL_RATE_11025; frames_per_sec = 11025; break;
case 8000: audio_rate = AL_RATE_8000; frames_per_sec = 8000; break;
default:
if (!quiet)
{
fprintf(stderr, "%s: can't play data at sample rate %f\n",
myname, file_rate);
}
frames_per_sec = 44100; /* pick some arbitrary rate */
audio_rate = AL_RATE_44100;
}
/*
* determine the current output rate
*/
pvbuf[0] = AL_OUTPUT_RATE;
ALgetparams(AL_DEFAULT_DEVICE, pvbuf, 2);
/*
* if the rates are the same, all is well. if not, then we need to proceed
* in a either a "rude" or "polite" manner.
*/
if (pvbuf[1] != audio_rate)
{
if (rude)
{
pvbuf[1] = audio_rate;
ALsetparams(AL_DEFAULT_DEVICE, pvbuf, 2);
}
else /*polite*/
{
double tmp_rate;
if (!quiet)
{
switch (pvbuf[1])
{
case AL_RATE_48000: tmp_rate = 48000; break;
case AL_RATE_44100: tmp_rate = 44100; break;
case AL_RATE_32000: tmp_rate = 32000; break;
case AL_RATE_22050: tmp_rate = 22050; break;
case AL_RATE_16000: tmp_rate = 16000; break;
case AL_RATE_11025: tmp_rate = 11025; break;
case AL_RATE_8000: tmp_rate = 8000; break;
}
fprintf(stderr,
"%s: '%s': adjust system output rate (currently %6.1f Hz)\n",
myname, filename, tmp_rate);
}
}
}
/*
* decide what size blocks of samples we should read from the
* input file and pass to ALwritesamps
*/
if (bits_per_samp <= 8)
{
bytes_per_samp = 1;
samp_wordsize = AL_SAMPLE_8;
}
else if (bits_per_samp <= 16)
{
bytes_per_samp = 2;
samp_wordsize = AL_SAMPLE_16;
}
else if (bits_per_samp <= 24)
{
bytes_per_samp = 4;
samp_wordsize = AL_SAMPLE_24;
}
else
{
if (!quiet)
{
fprintf(stderr, "%s: %s: error can't play %d bit samples\n",
myname, filename, bits_per_samp);
}
return(-1);
}
if ((samps_per_frame != 1) && (samps_per_frame!= 2))
{
if (!quiet)
{
fprintf(stderr, "%s: %s: error can't play %d channel sample data\n",
myname, filename, samps_per_frame);
}
return(-1);
}
/*
* make the buffer large enough to hold 1/2 sec of audio frames
* we add one to frames_per_sec before we divide in order to
* correctly handle the 11025 case
*/
secs_per_frame = 1.0 / ((double)frames_per_sec);
// frames_per_buf = (frames_per_sec+1)/2;
/* ok, I changed this so that now Im holding 1/4 sec of audio frames */
frames_per_buf = (int) (frames_per_sec+15)/20;
samps_per_buf = frames_per_buf * samps_per_frame;
bytes_per_buf = samps_per_buf * bytes_per_samp;
secs_per_buf = secs_per_frame * frames_per_buf;
sampbuf = (char *) malloc(bytes_per_buf);
/*
* configure and open audio port
*/
*ap_config = ALnewconfig();
ALsetwidth(*ap_config, samp_wordsize);
ALsetchannels(*ap_config, samps_per_frame);
/*
* make the ring buffer large enough to hold 1 sec of audio samples
*/
ALsetqueuesize(*ap_config, samps_per_buf*2);
*audio_port = ALopenport(myname, "w", *ap_config);
if (*audio_port != NULL) {
return (1);
}
else {
return (-1);
}
}
/*
* play audio sample data through the output port
*/
int
PlayClass::play_audio_samps(AFfilehandle audio_file, ALport audio_port,
ALconfig ap_config)
{
#ifdef DEBUG
fprintf(stderr, "play_audio_samps(...)\n");
#endif
int num_bufs;
int leftover_bytes;
int leftover_samps;
int leftover_frames;
int32_t samp_wordsize;
int samp_count;
int frame_count;
double sec_count;
int i;
int samples_read;
int frames_read;
int done;
int shortread;
int total_frames;
int total_samps;
int total_samp_bytes;
char compressionname[10];
float fileplayingtime;
sec_count = 0.0;
/*
* figure out how many reads we have to do
*/
total_frames = (int) AFgetframecnt(audio_file, AF_DEFAULT_TRACK);
total_samps = total_frames * samps_per_frame;
total_samp_bytes = total_samps * bytes_per_samp;
num_bufs = total_samp_bytes / bytes_per_buf;
leftover_bytes = total_samp_bytes % bytes_per_buf;
leftover_samps = leftover_bytes / bytes_per_samp;
leftover_frames = leftover_samps / samps_per_frame;
samp_wordsize = ALgetwidth(ap_config);
if (!quiet)
{
fileplayingtime = (float)total_frames / (float)frames_per_sec;
switch (compression)
{
default:
case AF_COMPRESSION_NONE:
strcpy(compressionname, "");
break;
case AF_COMPRESSION_G722:
strcpy(compressionname, "G.722 -->");
break;
case AF_COMPRESSION_G711_ALAW:
strcpy(compressionname, "A-law -->");
break;
case AF_COMPRESSION_G711_ULAW:
strcpy(compressionname, "u-law -->");
break;
}
}
if (verbose)
{
printf("%s: '%s' %6.3f sec %7.1f Hz %6s %s %d-bit %s\n",
myname,
filename,
fileplayingtime,
file_rate,
samps_per_frame == 1 ? "mono" : "stereo",
compressionname,
bits_per_samp,
filefmt == AF_FILE_AIFFC ? "AIFF-C" : "AIFF"
);
printf( " total sample frames = %d (%d bytes)\n",
total_frames, total_samp_bytes);
printf( " play data using blocksize = %d sample frames\n",
frames_per_buf);
printf( " total blocks = %d (%d extra frames)\n",
num_bufs, leftover_frames);
}
sec_count = 0.0;
top:
/*
* move the fileptr to the beginning of the sample data
*/
AFseekframe(audio_file, AF_DEFAULT_TRACK, 0);
/*
* note that there may be some pad bytes following the valid samples -
* for example, the sample data area may be padded so that the valid
* samples begin on a block boundary and the sample area ends on a block
* boundary (where blocksize is specified by the user)
*/
done = 0;
shortread = 0;
samp_count = 0;
frame_count = 0;
// sec_count = 0.0;
for (i=0; i<num_bufs; i++)
{
samp_count += samps_per_buf;
frame_count += frames_per_buf;
sec_count += secs_per_buf;
if (verbose)
{
printf(" %d sample frames %6.3f secs\n",
frame_count, sec_count);
}
if ((frames_read
= (int) AFreadframes(audio_file, AF_DEFAULT_TRACK,
sampbuf, frames_per_buf)) < frames_per_buf)
{
if (!quiet)
{
fprintf(stderr,
"%s: warning short read for %s: expected %d frames, got %d frames\n",
myname, filename, frames_per_buf, frames_read);
}
// done++;
shortread++;
}
samples_read = frames_read * samps_per_frame;
if ( timed && ( sec_count - in_time > play_time )) {
fading_in = FALSE;
fading_out = TRUE;
}
if ( !timed && !loop && ( sec_count > (double) fileplayingtime - out_time)) {
#ifdef DEBUG
fprintf(stderr, "fading out\n");
#endif
fading_in = FALSE;
fading_out = TRUE;
}
if (fading_in) {
attenuatesamps( sampbuf, samp_wordsize, samples_read, secs_per_buf,
max_level, &fade_level, in_time, 0 );
if ( fade_level >= 1.0 ) fading_in = FALSE;
}
if (fading_out) {
attenuatesamps( sampbuf, samp_wordsize, samples_read, secs_per_buf,
max_level, &fade_level, out_time, 1 );
if ( fade_level <= 0.0 ) {
#ifdef DEBUG
fprintf(stderr, "fade level is less than zero, exiting \n");
#endif
fading_out = FALSE;
done = TRUE;
}
}
if ( ! ( fading_out || fading_in ) && ( max_level != 1.0 ))
attenuatesamps( sampbuf, samp_wordsize, samples_read, secs_per_buf,
max_level, &fade_level, out_time, 2 );
ALwritesamps(audio_port, sampbuf, samples_read);
if (caught_sigint) {
done++;
#ifdef DEBUG
fprintf(stderr, "caught_sigint\n");
#endif
}
if (shortread)
{
/*
* allow the audio buffer to drain
*/
while(ALgetfilled(audio_port) > 81920) {
sginap(1);
}
if (loop)
goto top;
else
done = 1;
}
if (done) {
while(ALgetfilled(audio_port) > 0) {
sginap(1);
}
sginap(10);
return(0);
}
}
/*
* play the leftovers
*/
samp_count += leftover_samps;
frame_count += leftover_frames;
sec_count += ((double)leftover_frames) * secs_per_frame;
if (verbose && leftover_samps>0)
{
printf(" %d sample frames %6.3f secs\n",
frame_count, sec_count);
}
if ((frames_read =
(int) AFreadframes(audio_file, AF_DEFAULT_TRACK, sampbuf,
leftover_frames)) < leftover_frames)
{
if (!quiet)
{
fprintf(stderr,
"%s: warning short read for %s: expected %d frames, got %d frames\n",
myname, filename, leftover_frames, frames_read);
}
}
samples_read = frames_read * samps_per_frame;
if ( timed && ( sec_count - in_time > play_time )) {
fading_in = FALSE;
fading_out = TRUE;
}
if ( !timed && !loop && ( sec_count > (double) fileplayingtime - out_time)) {
#ifdef DEBUG
fprintf(stderr, "fading out 2\n");
#endif
fading_in = FALSE;
fading_out = TRUE;
}
if (fading_in) {
attenuatesamps( sampbuf, samp_wordsize, samples_read, secs_per_buf,
max_level, &fade_level, in_time, 0 );
if ( fade_level >= 1.0 ) fading_in = FALSE;
}
if (fading_out) {
attenuatesamps( sampbuf, samp_wordsize, samples_read, secs_per_buf,
max_level, &fade_level, out_time, 1 );
if ( fade_level <= 0.0 ) {
fading_out = FALSE;
done = TRUE;
}
}
if ( ! ( fading_out || fading_in ) && ( max_level != 1.0 ))
attenuatesamps( sampbuf, samp_wordsize, samples_read, secs_per_buf,
max_level, &fade_level, out_time, 2 );
ALwritesamps(audio_port, sampbuf, samples_read);
/*
* allow the audio buffer to drain
*/
while(ALgetfilled(audio_port) > 81920) {
sginap(1);
}
if (loop) goto top;
while(ALgetfilled(audio_port) > 0) {
sginap(1);
}
sginap(10);
return(0);
}
void
PlayClass::setVerbose(int i)
{
verbose = i;
}
void
PlayClass::setFilename(char *fname)
{
filename = fname;
}
void
PlayClass::setQuiet(int i)
{
quiet = i;
}
void
PlayClass::setRude(int i)
{
rude = i;
}
void
PlayClass::setLoop(short i)
{
loop = i;
}
void
PlayClass::setTime(double secs)
{
play_time = secs;
timed = TRUE;
}
void
PlayClass::setInTime(double secs)
{
if (secs == 0.0) {
fading_in = FALSE;
in_time = 0.0;
} else {
fading_in = TRUE;
in_time = secs;
}
}
void
PlayClass::setOutTime(double secs)
{
out_time = secs;
}
void PlayClass::setMaxOutputLevel(double lvl)
{
max_level = lvl;
}
short PlayClass::getIsLive()
{
return isLive;
}
short PlayClass::donep()
{
return isDone;
}
void
PlayClass::setCallback(PlayClassCB *Callback, void *d)
{
#ifdef DEBUG
fprintf(stderr, "setCallback\n");
#endif
doneCallback = Callback;
userData = d;
}
int
PlayClass::start()
{
short hasAudio = FALSE;
if (filename == NULL) return -1;
inventory_t* invry_p;
while((invry_p = getinvent()) != 0) {
if (invry_p->inv_class == INV_AUDIO) {
hasAudio = TRUE;
break;
}
}
setinvent();
endinvent();
if (!hasAudio) return -1;
// this is a quick check to see if there is an audio port available
ALconfig conf = ALnewconfig();
ALport prt;
if ( (prt = ALopenport( "noname", "w", conf )) != NULL ) {
// Close the audio port. We only opened it to check
// if it was available.
ALcloseport(prt);
} else {
return -1;
}
#ifdef DEBUG
fprintf(stderr, "start() filename = %s\n", filename);
#endif
/*
* Use the sproc(2) call to create an audio thread. The audio
* thread begins execution at the do_audio entry point. The
* second argument to sproc, PR_SALL, allows the two processes
* to share all attributes. See the manual page for the sproc(2)
* system call for more details.
*/
audio_pid =
sproc((void (*)(void *))&PlayClass::go, PR_SALL, (void *) this);
if (audio_pid < 0)
{
fprintf(stderr, "unable to create audio thread...aborting.\n");
return -1;
}
#ifdef DEBUG
fprintf(stderr, "returning 1, audio_pid = %d\n", audio_pid);
#endif
sproced = TRUE;
return 1;
}
void PlayClass::go(void *obj)
{
#ifdef DEBUG
fprintf(stderr, "go()\n");
#endif
PlayClass *objPtr = (PlayClass *) obj;
objPtr->beginPlaying();
#ifdef DEBUG
fprintf(stderr, "return from beginPlaying()\n");
fprintf(stderr, "process = %d\n", objPtr->audio_pid);
#endif
}
void
PlayClass::fadeOut()
{
if (out_time > 0.0) {
fading_out = TRUE;
fading_in = FALSE;
return;
}
}
void
PlayClass::stop()
{
#ifdef DEBUG
fprintf(stderr, "stop()\n");
#endif
caught_sigint = 1;
#ifdef DEBUG
fprintf(stderr, "waiting...");
#endif
// pid_t pid;
// int status;
// pid = wait(&status);
union wait waitStatus;
(void) waitpid(audio_pid, (int *) (&waitStatus), 0);
#ifdef DEBUG
fprintf(stderr, "done.\n");
#endif
caught_sigint = 0;
isLive = FALSE;
}
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