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Version 0.1.7 from FTP

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2001-03-10 19:12:11 +00:00
parent 22318b627d
commit 7157261be4
11 changed files with 272 additions and 207 deletions
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73
apps/app_intercom.c
View File

@@ -116,48 +116,48 @@ static int intercom_exec(struct ast_channel *chan, void *data)
int res = 0;
struct localuser *u;
struct ast_frame *f;
struct ast_channel *trans;
int oreadformat;
if (!data) {
ast_log(LOG_WARNING, "Playback requires an argument (filename)\n");
return -1;
}
LOCAL_USER_ADD(u);
/* See if we need a translator */
if (!(chan->format & AST_FORMAT_SLINEAR))
trans = ast_translator_create(chan, AST_FORMAT_SLINEAR, AST_DIRECTION_IN);
else
trans = chan;
if (trans) {
/* Read packets from the channel */
while(!res) {
res = ast_waitfor(trans, -1);
if (res > 0) {
res = 0;
f = ast_read(trans);
if (f) {
if (f->frametype == AST_FRAME_DTMF) {
ast_frfree(f);
break;
} else {
if (f->frametype == AST_FRAME_VOICE) {
if (f->subclass == AST_FORMAT_SLINEAR) {
res = write_audio(f->data, f->datalen);
if (res > 0)
res = 0;
} else
ast_log(LOG_DEBUG, "Unable to handle non-signed linear frame (%d)\n", f->subclass);
}
}
/* Remember original read format */
oreadformat = chan->readformat;
/* Set mode to signed linear */
res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set format to signed linear on channel %s\n", chan->name);
return -1;
}
/* Read packets from the channel */
while(!res) {
res = ast_waitfor(chan, -1);
if (res > 0) {
res = 0;
f = ast_read(chan);
if (f) {
if (f->frametype == AST_FRAME_DTMF) {
ast_frfree(f);
} else
res = -1;
}
break;
} else {
if (f->frametype == AST_FRAME_VOICE) {
if (f->subclass == AST_FORMAT_SLINEAR) {
res = write_audio(f->data, f->datalen);
if (res > 0)
res = 0;
} else
ast_log(LOG_DEBUG, "Unable to handle non-signed linear frame (%d)\n", f->subclass);
}
}
ast_frfree(f);
} else
res = -1;
}
if (trans != chan)
ast_translator_destroy(trans);
} else
ast_log(LOG_WARNING, "Unable to build translator to signed linear format on '%s'\n", chan->name);
}
LOCAL_USER_REMOVE(u);
if (!res)
ast_set_read_format(chan, oreadformat);
return res;
}
@@ -187,3 +187,8 @@ int usecount(void)
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

157
apps/app_mp3.c
View File

@@ -59,12 +59,13 @@ static int mp3_exec(struct ast_channel *chan, void *data)
{
int res=0;
struct localuser *u;
struct ast_channel *trans;
int fds[2];
int rfds[2];
int ms = -1;
int pid;
int us;
int exception;
int owriteformat;
struct timeval tv;
struct timeval last;
struct ast_frame *f;
@@ -85,87 +86,90 @@ static int mp3_exec(struct ast_channel *chan, void *data)
}
LOCAL_USER_ADD(u);
ast_stopstream(chan);
if (chan->format & AST_FORMAT_SLINEAR)
trans = chan;
else
trans = ast_translator_create(chan, AST_FORMAT_SLINEAR, AST_DIRECTION_OUT);
if (trans) {
res = mp3play((char *)data, fds[1]);
if (res >= 0) {
pid = res;
/* Order is important -- there's almost always going to be mp3... we want to prioritize the
user */
rfds[0] = trans->fd;
rfds[1] = fds[0];
for (;;) {
CHECK_BLOCKING(trans);
res = ast_waitfor_n_fd(rfds, 2, &ms);
trans->blocking = 0;
if (res < 1) {
ast_log(LOG_DEBUG, "Hangup detected\n");
res = -1;
break;
} else if (res == trans->fd) {
f = ast_read(trans);
if (!f) {
ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
res = -1;
break;
}
if (f->frametype == AST_FRAME_DTMF) {
ast_log(LOG_DEBUG, "User pressed a key\n");
ast_frfree(f);
res = 0;
break;
}
ast_frfree(f);
} else if (res == fds[0]) {
gettimeofday(&tv, NULL);
if (last.tv_sec || last.tv_usec) {
/* We should wait at least a frame length */
us = sizeof(myf.frdata) / 16 * 1000;
/* Subtract 1,000,000 us for each second late we've passed */
us -= (tv.tv_sec - last.tv_sec) * 1000000;
/* And one for each us late we've passed */
us -= (tv.tv_usec - last.tv_usec);
/* Sleep that long if needed */
if (us > 0)
usleep(us);
}
last = tv;
res = read(fds[0], myf.frdata, sizeof(myf.frdata));
if (res > 0) {
myf.f.frametype = AST_FRAME_VOICE;
myf.f.subclass = AST_FORMAT_SLINEAR;
myf.f.datalen = res;
myf.f.timelen = res / 16;
myf.f.mallocd = 0;
myf.f.offset = AST_FRIENDLY_OFFSET;
myf.f.src = __PRETTY_FUNCTION__;
myf.f.data = myf.frdata;
if (ast_write(trans, &myf.f) < 0) {
res = -1;
break;
}
} else {
ast_log(LOG_DEBUG, "No more mp3\n");
res = 0;
}
} else {
ast_log(LOG_DEBUG, "HuhHHH?\n");
owriteformat = chan->writeformat;
res = ast_set_write_format(chan, AST_FORMAT_SLINEAR);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set write format to signed linear\n");
return -1;
}
res = mp3play((char *)data, fds[1]);
if (res >= 0) {
pid = res;
/* Order is important -- there's almost always going to be mp3... we want to prioritize the
user */
rfds[0] = chan->fd;
rfds[1] = fds[0];
for (;;) {
CHECK_BLOCKING(chan);
res = ast_waitfor_n_fd(rfds, 2, &ms, &exception);
chan->blocking = 0;
if (res < 1) {
ast_log(LOG_DEBUG, "Hangup detected\n");
res = -1;
break;
} else if (res == chan->fd) {
if (exception)
chan->exception = 1;
f = ast_read(chan);
if (!f) {
ast_log(LOG_DEBUG, "Null frame == hangup() detected\n");
res = -1;
break;
}
if (f->frametype == AST_FRAME_DTMF) {
ast_log(LOG_DEBUG, "User pressed a key\n");
ast_frfree(f);
res = 0;
break;
}
ast_frfree(f);
} else if (res == fds[0]) {
gettimeofday(&tv, NULL);
if (last.tv_sec || last.tv_usec) {
/* We should wait at least a frame length */
us = sizeof(myf.frdata) / 16 * 1000;
/* Subtract 1,000,000 us for each second late we've passed */
us -= (tv.tv_sec - last.tv_sec) * 1000000;
/* And one for each us late we've passed */
us -= (tv.tv_usec - last.tv_usec);
/* Sleep that long if needed */
if (us > 0)
usleep(us);
}
last = tv;
res = read(fds[0], myf.frdata, sizeof(myf.frdata));
if (res > 0) {
myf.f.frametype = AST_FRAME_VOICE;
myf.f.subclass = AST_FORMAT_SLINEAR;
myf.f.datalen = res;
myf.f.timelen = res / 16;
myf.f.mallocd = 0;
myf.f.offset = AST_FRIENDLY_OFFSET;
myf.f.src = __PRETTY_FUNCTION__;
myf.f.data = myf.frdata;
if (ast_write(chan, &myf.f) < 0) {
res = -1;
break;
}
} else {
ast_log(LOG_DEBUG, "No more mp3\n");
res = 0;
}
} else {
ast_log(LOG_DEBUG, "HuhHHH?\n");
res = -1;
break;
}
kill(pid, SIGTERM);
}
if (trans != chan)
ast_translator_destroy(trans);
} else
ast_log(LOG_WARNING, "No translator channel available\n");
kill(pid, SIGTERM);
}
close(fds[0]);
close(fds[1]);
LOCAL_USER_REMOVE(u);
if (!res)
ast_set_write_format(chan, owriteformat);
return res;
}
@@ -191,3 +195,8 @@ int usecount(void)
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}

7
apps/app_voicemail.c
View File

@@ -418,7 +418,7 @@ static int vm_execmain(struct ast_channel *chan, void *data)
ast_verbose( VERBOSE_PREFIX_3 "User '%s' logged in on channel %s with %d messages\n", username, chan->name, maxmsg);
if (!ast_streamfile(chan, "vm-instructions", chan->language)) {
for(;;) {
if (chan->stream || (chan->trans && chan->trans->stream)) {
if (chan->stream) {
d = ast_waitstream(chan, AST_DIGIT_ANY);
ast_stopstream(chan);
if (!d && (state == STATE_MESSAGE_PLAYING)) {
@@ -616,3 +616,8 @@ int usecount(void)
STANDARD_USECOUNT(res);
return res;
}
char *key()
{
return ASTERISK_GPL_KEY;
}