mirror of
https://github.com/asterisk/asterisk.git
synced 2025-10-31 02:37:10 +00:00
Version 0.1.0 from FTP
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@17 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
189
apps/app_intercom.c
Executable file
189
apps/app_intercom.c
Executable file
@@ -0,0 +1,189 @@
|
||||
/*
|
||||
* Asterisk -- A telephony toolkit for Linux.
|
||||
*
|
||||
* Use /dev/dsp as an intercom.
|
||||
*
|
||||
* Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
|
||||
*
|
||||
* Mark Spencer <markster@linux-support.net>
|
||||
*
|
||||
* This program is free software, distributed under the terms of
|
||||
* the GNU General Public License
|
||||
*/
|
||||
|
||||
#include <asterisk/file.h>
|
||||
#include <asterisk/frame.h>
|
||||
#include <asterisk/logger.h>
|
||||
#include <asterisk/channel.h>
|
||||
#include <asterisk/pbx.h>
|
||||
#include <asterisk/module.h>
|
||||
#include <asterisk/translate.h>
|
||||
#include <unistd.h>
|
||||
#include <errno.h>
|
||||
#include <sys/ioctl.h>
|
||||
#include <string.h>
|
||||
#include <stdlib.h>
|
||||
#include <pthread.h>
|
||||
#include <sys/time.h>
|
||||
#include <linux/soundcard.h>
|
||||
#include <netinet/in.h>
|
||||
|
||||
#define DEV_DSP "/dev/dsp"
|
||||
|
||||
/* Number of 32 byte buffers -- each buffer is 2 ms */
|
||||
#define BUFFER_SIZE 32
|
||||
|
||||
static char *tdesc = "Intercom using /dev/dsp for output";
|
||||
|
||||
static char *app = "Intercom";
|
||||
|
||||
STANDARD_LOCAL_USER;
|
||||
|
||||
LOCAL_USER_DECL;
|
||||
|
||||
static pthread_mutex_t sound_lock = PTHREAD_MUTEX_INITIALIZER;
|
||||
static int sound = -1;
|
||||
|
||||
static int write_audio(short *data, int len)
|
||||
{
|
||||
int res;
|
||||
struct audio_buf_info info;
|
||||
pthread_mutex_lock(&sound_lock);
|
||||
if (sound < 0) {
|
||||
ast_log(LOG_WARNING, "Sound device closed?\n");
|
||||
pthread_mutex_unlock(&sound_lock);
|
||||
return -1;
|
||||
}
|
||||
if (ioctl(sound, SNDCTL_DSP_GETOSPACE, &info)) {
|
||||
ast_log(LOG_WARNING, "Unable to read output space\n");
|
||||
pthread_mutex_unlock(&sound_lock);
|
||||
return -1;
|
||||
}
|
||||
res = write(sound, data, len);
|
||||
pthread_mutex_unlock(&sound_lock);
|
||||
return res;
|
||||
}
|
||||
|
||||
static int create_audio()
|
||||
{
|
||||
int fmt, desired, res, fd;
|
||||
fd = open(DEV_DSP, O_WRONLY);
|
||||
if (fd < 0) {
|
||||
ast_log(LOG_WARNING, "Unable to open %s: %s\n", DEV_DSP, strerror(errno));
|
||||
close(fd);
|
||||
return -1;
|
||||
}
|
||||
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");
|
||||
close(fd);
|
||||
return -1;
|
||||
}
|
||||
fmt = 0;
|
||||
res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
|
||||
if (res < 0) {
|
||||
ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
|
||||
close(fd);
|
||||
return -1;
|
||||
}
|
||||
/* 8000 Hz desired */
|
||||
desired = 8000;
|
||||
fmt = desired;
|
||||
res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt);
|
||||
if (res < 0) {
|
||||
ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
|
||||
close(fd);
|
||||
return -1;
|
||||
}
|
||||
if (fmt != desired) {
|
||||
ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n");
|
||||
}
|
||||
#if 1
|
||||
/* 2 bytes * 15 units of 2^5 = 32 bytes per buffer */
|
||||
fmt = ((BUFFER_SIZE) << 16) | (0x0005);
|
||||
res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
|
||||
if (res < 0) {
|
||||
ast_log(LOG_WARNING, "Unable to set fragment size -- sound may be choppy\n");
|
||||
}
|
||||
#endif
|
||||
sound = fd;
|
||||
return 0;
|
||||
}
|
||||
|
||||
static int intercom_exec(struct ast_channel *chan, void *data)
|
||||
{
|
||||
int res = 0;
|
||||
struct localuser *u;
|
||||
struct ast_frame *f;
|
||||
struct ast_channel *trans;
|
||||
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);
|
||||
}
|
||||
}
|
||||
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);
|
||||
return res;
|
||||
}
|
||||
|
||||
int unload_module(void)
|
||||
{
|
||||
STANDARD_HANGUP_LOCALUSERS;
|
||||
if (sound > -1)
|
||||
close(sound);
|
||||
return ast_unregister_application(app);
|
||||
}
|
||||
|
||||
int load_module(void)
|
||||
{
|
||||
if (create_audio())
|
||||
return -1;
|
||||
return ast_register_application(app, intercom_exec);
|
||||
}
|
||||
|
||||
char *description(void)
|
||||
{
|
||||
return tdesc;
|
||||
}
|
||||
|
||||
int usecount(void)
|
||||
{
|
||||
int res;
|
||||
STANDARD_USECOUNT(res);
|
||||
return res;
|
||||
}
|
||||
Reference in New Issue
Block a user