Version 0.1.8 from FTP

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2001-05-03 04:32:56 +00:00
parent 132d84be6c
commit 0bffff6a4d

View File

@@ -33,6 +33,10 @@
#include <stdlib.h>
#include <stdio.h>
#include <linux/soundcard.h>
#include "busy.h"
#include "ringtone.h"
#include "ring10.h"
#include "answer.h"
/* Which device to use */
#define DEV_DSP "/dev/dsp"
@@ -43,7 +47,7 @@
/* When you set the frame size, you have to come up with
the right buffer format as well. */
/* 5 64-byte frames = one frame */
#define BUFFER_FMT ((buffersize * 5) << 16) | (0x0006);
#define BUFFER_FMT ((buffersize * 10) << 16) | (0x0006);
/* Don't switch between read/write modes faster than every 300 ms */
#define MIN_SWITCH_TIME 600
@@ -70,10 +74,32 @@ static char context[AST_MAX_EXTENSION] = "default";
static char language[MAX_LANGUAGE] = "";
static char exten[AST_MAX_EXTENSION] = "s";
/* Some pipes to prevent overflow */
static int funnel[2];
static pthread_mutex_t sound_lock = PTHREAD_MUTEX_INITIALIZER;
static pthread_t silly;
/* Command pipe */
static int cmd[2];
int hookstate=0;
static short silence[FRAME_SIZE] = {0, };
struct sound {
int ind;
short *data;
int datalen;
int samplen;
int silencelen;
int repeat;
};
static struct sound sounds[] = {
{ AST_CONTROL_RINGING, ringtone, sizeof(ringtone)/2, 16000, 32000, 1 },
{ AST_CONTROL_BUSY, busy, sizeof(busy)/2, 4000, 4000, 1 },
{ AST_CONTROL_CONGESTION, busy, sizeof(busy)/2, 2000, 2000, 1 },
{ AST_CONTROL_RING, ring10, sizeof(ring10)/2, 16000, 32000, 1 },
{ AST_CONTROL_ANSWER, answer, sizeof(answer)/2, 2200, 0, 0 },
};
/* Sound command pipe */
static int sndcmd[2];
static struct chan_oss_pvt {
/* We only have one OSS structure -- near sighted perhaps, but it
@@ -99,6 +125,7 @@ static int time_has_passed()
with 160 sample frames, and a buffer size of 3, we have a 60ms buffer,
usually plenty. */
pthread_t sthread;
#define MAX_BUFFER_SIZE 100
static int buffersize = 3;
@@ -127,6 +154,108 @@ static int calc_loudness(short *frame)
return sum;
}
static int cursound = -1;
static int sampsent = 0;
static int silencelen=0;
static int offset=0;
static int nosound=0;
static int send_sound(void)
{
short myframe[FRAME_SIZE];
int total = FRAME_SIZE;
short *frame = NULL;
int amt=0;
int res;
int myoff;
audio_buf_info abi;
if (cursound > -1) {
res = ioctl(sounddev, SNDCTL_DSP_GETOSPACE ,&abi);
if (res) {
ast_log(LOG_WARNING, "Unable to read output space\n");
return -1;
}
/* Calculate how many samples we can send, max */
if (total > (abi.fragments * abi.fragsize / 2))
total = abi.fragments * abi.fragsize / 2;
res = total;
if (sampsent < sounds[cursound].samplen) {
myoff=0;
while(total) {
amt = total;
if (amt > (sounds[cursound].datalen - offset))
amt = sounds[cursound].datalen - offset;
memcpy(myframe + myoff, sounds[cursound].data + offset, amt * 2);
total -= amt;
offset += amt;
sampsent += amt;
myoff += amt;
if (offset >= sounds[cursound].datalen)
offset = 0;
}
/* Set it up for silence */
if (sampsent >= sounds[cursound].samplen)
silencelen = sounds[cursound].silencelen;
frame = myframe;
} else {
if (silencelen > 0) {
frame = silence;
silencelen -= res;
} else {
if (sounds[cursound].repeat) {
/* Start over */
sampsent = 0;
offset = 0;
} else {
cursound = -1;
nosound = 0;
}
}
}
res = write(sounddev, frame, res * 2);
if (res > 0)
return 0;
return res;
}
return 0;
}
static void *sound_thread(void *unused)
{
fd_set rfds;
fd_set wfds;
int max;
int res;
for(;;) {
FD_ZERO(&rfds);
FD_ZERO(&wfds);
max = sndcmd[0];
FD_SET(sndcmd[0], &rfds);
if (cursound > -1) {
FD_SET(sounddev, &wfds);
if (sounddev > max)
max = sounddev;
}
res = select(max + 1, &rfds, &wfds, NULL, NULL);
if (res < 1) {
ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno));
continue;
}
if (FD_ISSET(sndcmd[0], &rfds)) {
read(sndcmd[0], &cursound, sizeof(cursound));
silencelen = 0;
offset = 0;
sampsent = 0;
}
if (FD_ISSET(sounddev, &wfds))
if (send_sound())
ast_log(LOG_WARNING, "Failed to write sound\n");
}
/* Never reached */
return NULL;
}
#if 0
static int silence_suppress(short *buf)
{
#define SILBUF 3
@@ -159,57 +288,23 @@ static int silence_suppress(short *buf)
/* Write any buffered silence we have, it may have something
important */
if (silbufcnt) {
write(funnel[1], silbuf, silbufcnt * FRAME_SIZE);
write(sounddev, silbuf, silbufcnt * FRAME_SIZE);
silbufcnt = 0;
}
}
return 0;
}
static void *silly_thread(void *ignore)
{
char buf[FRAME_SIZE * 2];
int pos=0;
int res=0;
/* Read from the sound device, and write to the pipe. */
for (;;) {
/* Give the writer a better shot at the lock */
#if 0
usleep(1000);
#endif
pthread_testcancel();
pthread_mutex_lock(&sound_lock);
res = read(sounddev, buf + pos, FRAME_SIZE * 2 - pos);
pthread_mutex_unlock(&sound_lock);
if (res > 0) {
pos += res;
if (pos == FRAME_SIZE * 2) {
if (needhangup || needanswer || strlen(digits) ||
!silence_suppress((short *)buf)) {
res = write(funnel[1], buf, sizeof(buf));
}
pos = 0;
}
} else {
close(funnel[1]);
break;
}
pthread_testcancel();
}
return NULL;
}
#endif
static int setformat(void)
{
int fmt, desired, res, fd = sounddev;
static int warnedalready = 0;
static int warnedalready2 = 0;
pthread_mutex_lock(&sound_lock);
fmt = AFMT_S16_LE;
res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n");
pthread_mutex_unlock(&sound_lock);
return -1;
}
res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
@@ -222,7 +317,6 @@ static int setformat(void)
res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
if (res < 0) {
ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
pthread_mutex_unlock(&sound_lock);
return -1;
}
/* 8000 Hz desired */
@@ -231,7 +325,6 @@ static int setformat(void)
res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt);
if (res < 0) {
ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
pthread_mutex_unlock(&sound_lock);
return -1;
}
if (fmt != desired) {
@@ -246,7 +339,6 @@ static int setformat(void)
ast_log(LOG_WARNING, "Unable to set fragment size -- sound may be choppy\n");
}
#endif
pthread_mutex_unlock(&sound_lock);
return 0;
}
@@ -256,7 +348,6 @@ static int soundcard_setoutput(int force)
int fd = sounddev;
if (full_duplex || (!readmode && !force))
return 0;
pthread_mutex_lock(&sound_lock);
readmode = 0;
if (force || time_has_passed()) {
ioctl(sounddev, SNDCTL_DSP_RESET);
@@ -264,26 +355,21 @@ static int soundcard_setoutput(int force)
time. */
/* dup2(0, sound); */
close(sounddev);
fd = open(DEV_DSP, O_WRONLY);
fd = open(DEV_DSP, O_WRONLY |O_NONBLOCK);
if (fd < 0) {
ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
pthread_mutex_unlock(&sound_lock);
return -1;
}
/* dup2 will close the original and make fd be sound */
if (dup2(fd, sounddev) < 0) {
ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
pthread_mutex_unlock(&sound_lock);
return -1;
}
if (setformat()) {
pthread_mutex_unlock(&sound_lock);
return -1;
}
pthread_mutex_unlock(&sound_lock);
return 0;
}
pthread_mutex_unlock(&sound_lock);
return 1;
}
@@ -292,41 +378,35 @@ static int soundcard_setinput(int force)
int fd = sounddev;
if (full_duplex || (readmode && !force))
return 0;
pthread_mutex_lock(&sound_lock);
readmode = -1;
if (force || time_has_passed()) {
ioctl(sounddev, SNDCTL_DSP_RESET);
close(sounddev);
/* dup2(0, sound); */
fd = open(DEV_DSP, O_RDONLY);
fd = open(DEV_DSP, O_RDONLY | O_NONBLOCK);
if (fd < 0) {
ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
pthread_mutex_unlock(&sound_lock);
return -1;
}
/* dup2 will close the original and make fd be sound */
if (dup2(fd, sounddev) < 0) {
ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
pthread_mutex_unlock(&sound_lock);
return -1;
}
if (setformat()) {
pthread_mutex_unlock(&sound_lock);
return -1;
}
pthread_mutex_unlock(&sound_lock);
return 0;
}
pthread_mutex_unlock(&sound_lock);
return 1;
}
static int soundcard_init()
{
/* Assume it's full duplex for starters */
int fd = open(DEV_DSP, O_RDWR);
int fd = open(DEV_DSP, O_RDWR | O_NONBLOCK);
if (fd < 0) {
ast_log(LOG_ERROR, "Unable to open %s: %s\n", DEV_DSP, strerror(errno));
ast_log(LOG_WARNING, "Unable to open %s: %s\n", DEV_DSP, strerror(errno));
return fd;
}
gettimeofday(&lasttime, NULL);
@@ -351,33 +431,52 @@ static int oss_text(struct ast_channel *c, char *text)
static int oss_call(struct ast_channel *c, char *dest, int timeout)
{
int res = 3;
ast_verbose( " << Call placed to '%s' on console >> \n", dest);
if (autoanswer) {
ast_verbose( " << Auto-answered >> \n" );
needanswer = 1;
} else {
ast_verbose( " << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
write(sndcmd[1], &res, sizeof(res));
}
return 0;
}
static void answer_sound(void)
{
int res;
nosound = 1;
res = 4;
write(sndcmd[1], &res, sizeof(res));
}
static int oss_answer(struct ast_channel *c)
{
ast_verbose( " << Console call has been answered >> \n");
answer_sound();
c->state = AST_STATE_UP;
cursound = -1;
return 0;
}
static int oss_hangup(struct ast_channel *c)
{
int res;
cursound = -1;
c->pvt->pvt = NULL;
oss.owner = NULL;
ast_verbose( " << Hangup on console >> \n");
pthread_mutex_lock(&usecnt_lock);
ast_pthread_mutex_lock(&usecnt_lock);
usecnt--;
pthread_mutex_unlock(&usecnt_lock);
ast_pthread_mutex_unlock(&usecnt_lock);
needhangup = 0;
needanswer = 0;
if (hookstate) {
res = 2;
write(sndcmd[1], &res, sizeof(res));
}
return 0;
}
@@ -390,7 +489,6 @@ static int soundcard_writeframe(short *data)
int res;
int fd = sounddev;
static int warned=0;
pthread_mutex_lock(&sound_lock);
if (ioctl(fd, SNDCTL_DSP_GETOSPACE, &info)) {
if (!warned)
ast_log(LOG_WARNING, "Error reading output space\n");
@@ -413,7 +511,6 @@ static int soundcard_writeframe(short *data)
res = write(fd, ((void *)buffer), FRAME_SIZE * 2 * buffersize);
}
}
pthread_mutex_unlock(&sound_lock);
return res;
}
@@ -425,6 +522,11 @@ static int oss_write(struct ast_channel *chan, struct ast_frame *f)
static int sizpos = 0;
int len = sizpos;
int pos;
/* Immediately return if no sound is enabled */
if (nosound)
return 0;
/* Stop any currently playing sound */
cursound = -1;
if (!full_duplex && (strlen(digits) || needhangup || needanswer)) {
/* If we're half duplex, we have to switch to read mode
to honor immediate needs if necessary */
@@ -468,11 +570,18 @@ static struct ast_frame *oss_read(struct ast_channel *chan)
static char buf[FRAME_SIZE * 2 + AST_FRIENDLY_OFFSET];
static int readpos = 0;
int res;
int b;
int nonull=0;
#if 0
ast_log(LOG_DEBUG, "oss_read()\n");
#endif
/* Acknowledge any pending cmd */
res = read(cmd[0], &b, sizeof(b));
if (res > 0)
nonull = 1;
f.frametype = AST_FRAME_NULL;
f.subclass = 0;
f.timelen = 0;
@@ -509,6 +618,9 @@ static struct ast_frame *oss_read(struct ast_channel *chan)
return &f;
}
if (nonull)
return &f;
res = soundcard_setinput(0);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to set input mode\n");
@@ -518,14 +630,15 @@ static struct ast_frame *oss_read(struct ast_channel *chan)
/* Theoretically shouldn't happen, but anyway, return a NULL frame */
return &f;
}
res = read(funnel[0], buf + AST_FRIENDLY_OFFSET + readpos, FRAME_SIZE * 2 - readpos);
res = read(sounddev, buf + AST_FRIENDLY_OFFSET + readpos, FRAME_SIZE * 2 - readpos);
if (res < 0) {
ast_log(LOG_WARNING, "Error reading from sound device: %s\n", strerror(errno));
CRASH;
return NULL;
}
readpos += res;
if (readpos == FRAME_SIZE * 2) {
if (readpos >= FRAME_SIZE * 2) {
/* A real frame */
readpos = 0;
f.frametype = AST_FRAME_VOICE;
@@ -536,10 +649,47 @@ static struct ast_frame *oss_read(struct ast_channel *chan)
f.offset = AST_FRIENDLY_OFFSET;
f.src = type;
f.mallocd = 0;
#if 0
{ static int fd = -1;
if (fd < 0)
fd = open("output.raw", O_RDWR | O_TRUNC | O_CREAT);
write(fd, f.data, f.datalen);
}
#endif
}
return &f;
}
static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
{
struct chan_oss_pvt *p = newchan->pvt->pvt;
p->owner = newchan;
return 0;
}
static int oss_indicate(struct ast_channel *chan, int cond)
{
int res;
switch(cond) {
case AST_CONTROL_BUSY:
res = 1;
break;
case AST_CONTROL_CONGESTION:
res = 2;
break;
case AST_CONTROL_RINGING:
res = 0;
break;
default:
ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, chan->name);
return -1;
}
if (res > -1) {
write(sndcmd[1], &res, sizeof(res));
}
return 0;
}
static struct ast_channel *oss_new(struct chan_oss_pvt *p, int state)
{
struct ast_channel *tmp;
@@ -547,7 +697,8 @@ static struct ast_channel *oss_new(struct chan_oss_pvt *p, int state)
if (tmp) {
snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", DEV_DSP + 5);
tmp->type = type;
tmp->fd = funnel[0];
tmp->fds[0] = sounddev;
tmp->fds[1] = cmd[0];
tmp->nativeformats = AST_FORMAT_SLINEAR;
tmp->pvt->pvt = p;
tmp->pvt->send_digit = oss_digit;
@@ -557,6 +708,8 @@ static struct ast_channel *oss_new(struct chan_oss_pvt *p, int state)
tmp->pvt->read = oss_read;
tmp->pvt->call = oss_call;
tmp->pvt->write = oss_write;
tmp->pvt->indicate = oss_indicate;
tmp->pvt->fixup = oss_fixup;
if (strlen(p->context))
strncpy(tmp->context, p->context, sizeof(tmp->context));
if (strlen(p->exten))
@@ -565,9 +718,9 @@ static struct ast_channel *oss_new(struct chan_oss_pvt *p, int state)
strncpy(tmp->language, language, sizeof(tmp->language));
p->owner = tmp;
tmp->state = state;
pthread_mutex_lock(&usecnt_lock);
ast_pthread_mutex_lock(&usecnt_lock);
usecnt++;
pthread_mutex_unlock(&usecnt_lock);
ast_pthread_mutex_unlock(&usecnt_lock);
ast_update_use_count();
if (state != AST_STATE_DOWN) {
if (ast_pbx_start(tmp)) {
@@ -650,7 +803,10 @@ static int console_answer(int fd, int argc, char *argv[])
ast_cli(fd, "No one is calling us\n");
return RESULT_FAILURE;
}
hookstate = 1;
cursound = -1;
needanswer++;
answer_sound();
return RESULT_SUCCESS;
}
@@ -686,11 +842,14 @@ static int console_hangup(int fd, int argc, char *argv[])
{
if (argc != 1)
return RESULT_SHOWUSAGE;
if (!oss.owner) {
cursound = -1;
if (!oss.owner && !hookstate) {
ast_cli(fd, "No call to hangup up\n");
return RESULT_FAILURE;
}
needhangup++;
hookstate = 0;
if (oss.owner)
needhangup++;
return RESULT_SUCCESS;
}
@@ -703,12 +862,15 @@ static int console_dial(int fd, int argc, char *argv[])
{
char tmp[256], *tmp2;
char *mye, *myc;
int b = 0;
if ((argc != 1) && (argc != 2))
return RESULT_SHOWUSAGE;
if (oss.owner) {
if (argc == 2)
if (argc == 2) {
strncat(digits, argv[1], sizeof(digits) - strlen(digits));
else {
/* Wake up the polling thread */
write(cmd[1], &b, sizeof(b));
} else {
ast_cli(fd, "You're already in a call. You can use this only to dial digits until you hangup\n");
return RESULT_FAILURE;
}
@@ -728,6 +890,7 @@ static int console_dial(int fd, int argc, char *argv[])
if (ast_exists_extension(NULL, myc, mye, 1)) {
strncpy(oss.exten, mye, sizeof(oss.exten));
strncpy(oss.context, myc, sizeof(oss.context));
hookstate = 1;
oss_new(&oss, AST_STATE_UP);
} else
ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc);
@@ -754,28 +917,28 @@ int load_module()
int flags;
struct ast_config *cfg = ast_load(config);
struct ast_variable *v;
res = pipe(funnel);
res = pipe(cmd);
res = pipe(sndcmd);
if (res) {
ast_log(LOG_ERROR, "Unable to create pipe\n");
return -1;
}
/* We make the funnel so that writes to the funnel don't block...
Our "silly" thread can read to its heart content, preventing
recording overruns */
flags = fcntl(funnel[1], F_GETFL);
#if 0
fcntl(funnel[0], F_SETFL, flags | O_NONBLOCK);
#endif
fcntl(funnel[1], F_SETFL, flags | O_NONBLOCK);
flags = fcntl(cmd[0], F_GETFL);
fcntl(cmd[0], F_SETFL, flags | O_NONBLOCK);
flags = fcntl(cmd[1], F_GETFL);
fcntl(cmd[1], F_SETFL, flags | O_NONBLOCK);
res = soundcard_init();
if (res < 0) {
close(funnel[1]);
close(funnel[0]);
return -1;
close(cmd[1]);
close(cmd[0]);
if (option_verbose > 1) {
ast_verbose(VERBOSE_PREFIX_2 "No sound card detected -- console channel will be unavailable\n");
ast_verbose(VERBOSE_PREFIX_2 "Turn off OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf\n");
}
return 0;
}
if (!full_duplex)
ast_log(LOG_WARNING, "XXX I don't work right with non-full duplex sound cards XXX\n");
pthread_create(&silly, NULL, silly_thread, NULL);
res = ast_channel_register(type, tdesc, AST_FORMAT_SLINEAR, oss_request);
if (res < 0) {
ast_log(LOG_ERROR, "Unable to register channel class '%s'\n", type);
@@ -802,6 +965,7 @@ int load_module()
}
ast_destroy(cfg);
}
pthread_create(&sthread, NULL, sound_thread, NULL);
return 0;
}
@@ -813,13 +977,13 @@ int unload_module()
for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
ast_cli_unregister(myclis + x);
close(sounddev);
if (funnel[0] > 0) {
close(funnel[0]);
close(funnel[1]);
if (cmd[0] > 0) {
close(cmd[0]);
close(cmd[1]);
}
if (silly) {
pthread_cancel(silly);
pthread_join(silly, NULL);
if (sndcmd[0] > 0) {
close(sndcmd[0]);
close(sndcmd[1]);
}
if (oss.owner)
ast_softhangup(oss.owner);
@@ -836,9 +1000,9 @@ char *description()
int usecount()
{
int res;
pthread_mutex_lock(&usecnt_lock);
ast_pthread_mutex_lock(&usecnt_lock);
res = usecnt;
pthread_mutex_unlock(&usecnt_lock);
ast_pthread_mutex_unlock(&usecnt_lock);
return res;
}