mirror of https://github.com/asterisk/asterisk
You can not select more than 25 topics
Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
453 lines
10 KiB
453 lines
10 KiB
/*
|
|
* Asterisk -- A telephony toolkit for Linux.
|
|
*
|
|
* Flat, binary, ADPCM vox file format.
|
|
*
|
|
* Copyright (C) 1999, Mark Spencer
|
|
*
|
|
* Mark Spencer <markster@linux-support.net>
|
|
*
|
|
* This program is free software, distributed under the terms of
|
|
* the GNU General Public License
|
|
*/
|
|
|
|
#include <asterisk/lock.h>
|
|
#include <asterisk/channel.h>
|
|
#include <asterisk/file.h>
|
|
#include <asterisk/logger.h>
|
|
#include <asterisk/sched.h>
|
|
#include <asterisk/module.h>
|
|
#include <arpa/inet.h>
|
|
#include <stdlib.h>
|
|
#include <sys/time.h>
|
|
#include <stdio.h>
|
|
#include <unistd.h>
|
|
#include <errno.h>
|
|
#include <string.h>
|
|
#include <pthread.h>
|
|
#ifdef __linux__
|
|
#include <endian.h>
|
|
#else
|
|
#include <machine/endian.h>
|
|
#endif
|
|
|
|
#define BUF_SIZE 80 /* 160 samples */
|
|
|
|
struct ast_filestream {
|
|
void *reserved[AST_RESERVED_POINTERS];
|
|
/* Believe it or not, we must decode/recode to account for the
|
|
weird MS format */
|
|
/* This is what a filestream means to us */
|
|
int fd; /* Descriptor */
|
|
struct ast_channel *owner;
|
|
struct ast_frame fr; /* Frame information */
|
|
char waste[AST_FRIENDLY_OFFSET]; /* Buffer for sending frames, etc */
|
|
char empty; /* Empty character */
|
|
unsigned char buf[BUF_SIZE + 3]; /* Output Buffer */
|
|
int lasttimeout;
|
|
struct timeval last;
|
|
int adj;
|
|
short signal; /* Signal level (file side) */
|
|
short ssindex; /* Signal ssindex (file side) */
|
|
struct ast_filestream *next;
|
|
};
|
|
|
|
|
|
static struct ast_filestream *glist = NULL;
|
|
static pthread_mutex_t vox_lock = AST_MUTEX_INITIALIZER;
|
|
static int glistcnt = 0;
|
|
|
|
static char *name = "vox";
|
|
static char *desc = "Dialogic VOX (ADPCM) File Format";
|
|
static char *exts = "vox";
|
|
|
|
/*
|
|
* Step size index shift table
|
|
*/
|
|
|
|
static short indsft[8] = { -1, -1, -1, -1, 2, 4, 6, 8 };
|
|
|
|
/*
|
|
* Step size table, where stpsz[i]=floor[16*(11/10)^i]
|
|
*/
|
|
|
|
static short stpsz[49] = {
|
|
16, 17, 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, 50, 55, 60, 66, 73,
|
|
80, 88, 97, 107, 118, 130, 143, 157, 173, 190, 209, 230, 253, 279,
|
|
307, 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, 876, 963,
|
|
1060, 1166, 1282, 1411, 1552
|
|
};
|
|
|
|
/*
|
|
* Nibble to bit map
|
|
*/
|
|
|
|
static short nbl2bit[16][4] = {
|
|
{1, 0, 0, 0}, {1, 0, 0, 1}, {1, 0, 1, 0}, {1, 0, 1, 1},
|
|
{1, 1, 0, 0}, {1, 1, 0, 1}, {1, 1, 1, 0}, {1, 1, 1, 1},
|
|
{-1, 0, 0, 0}, {-1, 0, 0, 1}, {-1, 0, 1, 0}, {-1, 0, 1, 1},
|
|
{-1, 1, 0, 0}, {-1, 1, 0, 1}, {-1, 1, 1, 0}, {-1, 1, 1, 1}
|
|
};
|
|
|
|
/*
|
|
* Decode(encoded)
|
|
* Decodes the encoded nibble from the adpcm file.
|
|
*
|
|
* Results:
|
|
* Returns the encoded difference.
|
|
*
|
|
* Side effects:
|
|
* Sets the index to the step size table for the next encode.
|
|
*/
|
|
|
|
static inline short
|
|
decode (unsigned char encoded, short *ssindex)
|
|
{
|
|
short diff, step;
|
|
step = stpsz[*ssindex];
|
|
diff = nbl2bit[encoded][0] * (step * nbl2bit[encoded][1] +
|
|
(step >> 1) * nbl2bit[encoded][2] +
|
|
(step >> 2) * nbl2bit[encoded][3] +
|
|
(step >> 3));
|
|
*ssindex = *ssindex + indsft[(encoded & 7)];
|
|
if (*ssindex < 0)
|
|
*ssindex = 0;
|
|
else if (*ssindex > 48)
|
|
*ssindex = 48;
|
|
return (diff);
|
|
}
|
|
|
|
/*
|
|
* Adpcm
|
|
* Takes a signed linear signal and encodes it as ADPCM
|
|
* For more information see http://support.dialogic.com/appnotes/adpcm.pdf
|
|
*
|
|
* Results:
|
|
* Foo.
|
|
*
|
|
* Side effects:
|
|
* signal gets updated with each pass.
|
|
*/
|
|
|
|
static inline unsigned char
|
|
adpcm (short csig, short *ssindex, short *signal)
|
|
{
|
|
short diff, step;
|
|
unsigned char encoded;
|
|
step = stpsz[*ssindex];
|
|
/*
|
|
* Clip csig if too large or too small
|
|
*/
|
|
|
|
csig >>= 4;
|
|
|
|
diff = csig - *signal;
|
|
|
|
if (diff < 0)
|
|
{
|
|
encoded = 8;
|
|
diff = -diff;
|
|
}
|
|
else
|
|
encoded = 0;
|
|
if (diff >= step)
|
|
{
|
|
encoded |= 4;
|
|
diff -= step;
|
|
}
|
|
step >>= 1;
|
|
if (diff >= step)
|
|
{
|
|
encoded |= 2;
|
|
diff -= step;
|
|
}
|
|
step >>= 1;
|
|
if (diff >= step)
|
|
encoded |= 1;
|
|
|
|
*signal += decode (encoded, ssindex);
|
|
return (encoded);
|
|
}
|
|
|
|
static struct ast_filestream *vox_open(int fd)
|
|
{
|
|
/* We don't have any header to read or anything really, but
|
|
if we did, it would go here. We also might want to check
|
|
and be sure it's a valid file. */
|
|
struct ast_filestream *tmp;
|
|
if ((tmp = malloc(sizeof(struct ast_filestream)))) {
|
|
memset(tmp, 0, sizeof(struct ast_filestream));
|
|
if (ast_pthread_mutex_lock(&vox_lock)) {
|
|
ast_log(LOG_WARNING, "Unable to lock vox list\n");
|
|
free(tmp);
|
|
return NULL;
|
|
}
|
|
tmp->next = glist;
|
|
glist = tmp;
|
|
tmp->fd = fd;
|
|
tmp->owner = NULL;
|
|
tmp->fr.data = tmp->buf;
|
|
tmp->fr.frametype = AST_FRAME_VOICE;
|
|
tmp->fr.subclass = AST_FORMAT_ADPCM;
|
|
/* datalen will vary for each frame */
|
|
tmp->fr.src = name;
|
|
tmp->fr.mallocd = 0;
|
|
tmp->lasttimeout = -1;
|
|
glistcnt++;
|
|
ast_pthread_mutex_unlock(&vox_lock);
|
|
ast_update_use_count();
|
|
}
|
|
return tmp;
|
|
}
|
|
|
|
static struct ast_filestream *vox_rewrite(int fd, char *comment)
|
|
{
|
|
/* We don't have any header to read or anything really, but
|
|
if we did, it would go here. We also might want to check
|
|
and be sure it's a valid file. */
|
|
struct ast_filestream *tmp;
|
|
if ((tmp = malloc(sizeof(struct ast_filestream)))) {
|
|
memset(tmp, 0, sizeof(struct ast_filestream));
|
|
if (ast_pthread_mutex_lock(&vox_lock)) {
|
|
ast_log(LOG_WARNING, "Unable to lock vox list\n");
|
|
free(tmp);
|
|
return NULL;
|
|
}
|
|
tmp->next = glist;
|
|
glist = tmp;
|
|
tmp->fd = fd;
|
|
tmp->owner = NULL;
|
|
tmp->lasttimeout = -1;
|
|
glistcnt++;
|
|
ast_pthread_mutex_unlock(&vox_lock);
|
|
ast_update_use_count();
|
|
} else
|
|
ast_log(LOG_WARNING, "Out of memory\n");
|
|
return tmp;
|
|
}
|
|
|
|
static struct ast_frame *vox_read(struct ast_filestream *s)
|
|
{
|
|
return NULL;
|
|
}
|
|
|
|
static void vox_close(struct ast_filestream *s)
|
|
{
|
|
struct ast_filestream *tmp, *tmpl = NULL;
|
|
if (ast_pthread_mutex_lock(&vox_lock)) {
|
|
ast_log(LOG_WARNING, "Unable to lock vox list\n");
|
|
return;
|
|
}
|
|
tmp = glist;
|
|
while(tmp) {
|
|
if (tmp == s) {
|
|
if (tmpl)
|
|
tmpl->next = tmp->next;
|
|
else
|
|
glist = tmp->next;
|
|
break;
|
|
}
|
|
tmpl = tmp;
|
|
tmp = tmp->next;
|
|
}
|
|
glistcnt--;
|
|
if (s->owner) {
|
|
s->owner->stream = NULL;
|
|
if (s->owner->streamid > -1)
|
|
ast_sched_del(s->owner->sched, s->owner->streamid);
|
|
s->owner->streamid = -1;
|
|
}
|
|
ast_pthread_mutex_unlock(&vox_lock);
|
|
ast_update_use_count();
|
|
if (!tmp)
|
|
ast_log(LOG_WARNING, "Freeing a filestream we don't seem to own\n");
|
|
close(s->fd);
|
|
free(s);
|
|
s = NULL;
|
|
}
|
|
|
|
static int ast_read_callback(void *data)
|
|
{
|
|
int retval = 0;
|
|
int res;
|
|
int delay;
|
|
struct ast_filestream *s = data;
|
|
struct timeval tv;
|
|
int x;
|
|
/* Send a frame from the file to the appropriate channel */
|
|
|
|
s->fr.frametype = AST_FRAME_VOICE;
|
|
s->fr.subclass = AST_FORMAT_ADPCM;
|
|
s->fr.offset = AST_FRIENDLY_OFFSET;
|
|
s->fr.mallocd = 0;
|
|
s->fr.data = s->buf;
|
|
if ((res = read(s->fd, s->buf + 3, BUF_SIZE)) < 1) {
|
|
if (res)
|
|
ast_log(LOG_WARNING, "Short read (%d) (%s)!\n", res, strerror(errno));
|
|
s->owner->streamid = -1;
|
|
return 0;
|
|
}
|
|
/* Store index, then signal */
|
|
s->buf[0] = s->ssindex & 0xff;
|
|
s->buf[1] = (s->signal >> 8) & 0xff;
|
|
s->buf[2] = s->signal & 0xff;
|
|
/* Do the decoder to be sure we get the right stuff in the signal and index fields. */
|
|
for (x=3;x<res+3;x++) {
|
|
s->signal += decode(s->buf[x] >> 4, &s->ssindex);
|
|
s->signal += decode(s->buf[x] & 0xf, &s->ssindex);
|
|
}
|
|
s->fr.samples = res * 2;
|
|
s->fr.datalen = res + 3;
|
|
delay = s->fr.samples / 8;
|
|
/* Lastly, process the frame */
|
|
if (ast_write(s->owner, &s->fr)) {
|
|
ast_log(LOG_WARNING, "Failed to write frame\n");
|
|
s->owner->streamid = -1;
|
|
return 0;
|
|
}
|
|
if (s->last.tv_usec || s->last.tv_usec) {
|
|
int ms;
|
|
gettimeofday(&tv, NULL);
|
|
ms = 1000 * (tv.tv_sec - s->last.tv_sec) +
|
|
(tv.tv_usec - s->last.tv_usec) / 1000;
|
|
s->last.tv_sec = tv.tv_sec;
|
|
s->last.tv_usec = tv.tv_usec;
|
|
if ((ms - delay) * (ms - delay) > 4) {
|
|
/* Compensate if we're more than 2 ms off */
|
|
s->adj -= (ms - delay);
|
|
}
|
|
#if 0
|
|
fprintf(stdout, "Delay is %d, adjustment is %d, last was %d\n", delay, s->adj, ms);
|
|
#endif
|
|
delay += s->adj;
|
|
if (delay < 1)
|
|
delay = 1;
|
|
} else
|
|
gettimeofday(&s->last, NULL);
|
|
if (s->lasttimeout != delay) {
|
|
/* We'll install the next timeout now. */
|
|
s->owner->streamid = ast_sched_add(s->owner->sched,
|
|
delay, ast_read_callback, s);
|
|
s->lasttimeout = delay;
|
|
} else {
|
|
/* Just come back again at the same time */
|
|
retval = -1;
|
|
}
|
|
return retval;
|
|
}
|
|
|
|
static int vox_apply(struct ast_channel *c, struct ast_filestream *s)
|
|
{
|
|
/* Select our owner for this stream, and get the ball rolling. */
|
|
s->owner = c;
|
|
return 0;
|
|
}
|
|
|
|
static int vox_play(struct ast_filestream *s)
|
|
{
|
|
ast_read_callback(s);
|
|
return 0;
|
|
}
|
|
|
|
static int vox_write(struct ast_filestream *fs, struct ast_frame *f)
|
|
{
|
|
int res;
|
|
if (f->frametype != AST_FRAME_VOICE) {
|
|
ast_log(LOG_WARNING, "Asked to write non-voice frame!\n");
|
|
return -1;
|
|
}
|
|
if (f->subclass != AST_FORMAT_ADPCM) {
|
|
ast_log(LOG_WARNING, "Asked to write non-ADPCM frame (%d)!\n", f->subclass);
|
|
return -1;
|
|
}
|
|
if (f->datalen < 3) {
|
|
ast_log(LOG_WARNING, "Invalid frame of data (< 3 bytes long) from %s\n", f->src);
|
|
return -1;
|
|
}
|
|
if ((res = write(fs->fd, f->data + 3, f->datalen)) != f->datalen) {
|
|
ast_log(LOG_WARNING, "Bad write (%d/%d): %s\n", res, f->datalen, strerror(errno));
|
|
return -1;
|
|
}
|
|
return 0;
|
|
}
|
|
|
|
static char *vox_getcomment(struct ast_filestream *s)
|
|
{
|
|
return NULL;
|
|
}
|
|
|
|
static int vox_seek(struct ast_filestream *fs, long sample_offset, int whence)
|
|
{
|
|
return -1;
|
|
}
|
|
|
|
static int vox_trunc(struct ast_filestream *fs)
|
|
{
|
|
return -1;
|
|
}
|
|
|
|
static long vox_tell(struct ast_filestream *fs)
|
|
{
|
|
return -1;
|
|
}
|
|
|
|
int load_module()
|
|
{
|
|
return ast_format_register(name, exts, AST_FORMAT_ADPCM,
|
|
vox_open,
|
|
vox_rewrite,
|
|
vox_apply,
|
|
vox_play,
|
|
vox_write,
|
|
vox_seek,
|
|
vox_trunc,
|
|
vox_tell,
|
|
vox_read,
|
|
vox_close,
|
|
vox_getcomment);
|
|
|
|
|
|
}
|
|
|
|
int unload_module()
|
|
{
|
|
struct ast_filestream *tmp, *tmpl;
|
|
if (ast_pthread_mutex_lock(&vox_lock)) {
|
|
ast_log(LOG_WARNING, "Unable to lock vox list\n");
|
|
return -1;
|
|
}
|
|
tmp = glist;
|
|
while(tmp) {
|
|
if (tmp->owner)
|
|
ast_softhangup(tmp->owner, AST_SOFTHANGUP_APPUNLOAD);
|
|
tmpl = tmp;
|
|
tmp = tmp->next;
|
|
free(tmpl);
|
|
}
|
|
ast_pthread_mutex_unlock(&vox_lock);
|
|
return ast_format_unregister(name);
|
|
}
|
|
|
|
int usecount()
|
|
{
|
|
int res;
|
|
if (ast_pthread_mutex_lock(&vox_lock)) {
|
|
ast_log(LOG_WARNING, "Unable to lock vox list\n");
|
|
return -1;
|
|
}
|
|
res = glistcnt;
|
|
ast_pthread_mutex_unlock(&vox_lock);
|
|
return res;
|
|
}
|
|
|
|
char *description()
|
|
{
|
|
return desc;
|
|
}
|
|
|
|
|
|
char *key()
|
|
{
|
|
return ASTERISK_GPL_KEY;
|
|
}
|