git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
198 lines
4.4 KiB
C
Executable file
198 lines
4.4 KiB
C
Executable file
/*
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* Asterisk -- A telephony toolkit for Linux.
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*
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* Use /dev/dsp as an intercom.
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*
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* Copyright (C) 1999, Mark Spencer
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*
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* Mark Spencer <markster@linux-support.net>
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*
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* This program is free software, distributed under the terms of
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* the GNU General Public License
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*/
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#include <asterisk/lock.h>
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#include <asterisk/file.h>
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#include <asterisk/frame.h>
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#include <asterisk/logger.h>
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#include <asterisk/channel.h>
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#include <asterisk/pbx.h>
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#include <asterisk/module.h>
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#include <asterisk/translate.h>
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#include <unistd.h>
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#include <errno.h>
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#include <sys/ioctl.h>
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#include <string.h>
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#include <stdlib.h>
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#include <pthread.h>
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#include <sys/time.h>
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#include <linux/soundcard.h>
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#include <netinet/in.h>
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#define DEV_DSP "/dev/dsp"
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/* Number of 32 byte buffers -- each buffer is 2 ms */
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#define BUFFER_SIZE 32
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static char *tdesc = "Intercom using /dev/dsp for output";
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static char *app = "Intercom";
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static char *synopsis = "(Obsolete) Send to Intercom";
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static char *descrip =
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" Intercom(): Sends the user to the intercom (i.e. /dev/dsp). This program\n"
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"is generally considered obselete by the chan_oss module. Returns 0 if the\n"
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"user exits with a DTMF tone, or -1 if they hangup.\n";
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STANDARD_LOCAL_USER;
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LOCAL_USER_DECL;
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static pthread_mutex_t sound_lock = AST_MUTEX_INITIALIZER;
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static int sound = -1;
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static int write_audio(short *data, int len)
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{
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int res;
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struct audio_buf_info info;
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ast_pthread_mutex_lock(&sound_lock);
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if (sound < 0) {
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ast_log(LOG_WARNING, "Sound device closed?\n");
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ast_pthread_mutex_unlock(&sound_lock);
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return -1;
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}
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if (ioctl(sound, SNDCTL_DSP_GETOSPACE, &info)) {
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ast_log(LOG_WARNING, "Unable to read output space\n");
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ast_pthread_mutex_unlock(&sound_lock);
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return -1;
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}
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res = write(sound, data, len);
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ast_pthread_mutex_unlock(&sound_lock);
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return res;
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}
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static int create_audio(void)
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{
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int fmt, desired, res, fd;
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fd = open(DEV_DSP, O_WRONLY);
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if (fd < 0) {
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ast_log(LOG_WARNING, "Unable to open %s: %s\n", DEV_DSP, strerror(errno));
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close(fd);
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return -1;
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}
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fmt = AFMT_S16_LE;
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res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
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if (res < 0) {
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ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n");
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close(fd);
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return -1;
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}
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fmt = 0;
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res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
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if (res < 0) {
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ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
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close(fd);
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return -1;
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}
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/* 8000 Hz desired */
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desired = 8000;
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fmt = desired;
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res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt);
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if (res < 0) {
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ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
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close(fd);
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return -1;
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}
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if (fmt != desired) {
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ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n", desired, fmt);
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}
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#if 1
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/* 2 bytes * 15 units of 2^5 = 32 bytes per buffer */
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fmt = ((BUFFER_SIZE) << 16) | (0x0005);
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res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
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if (res < 0) {
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ast_log(LOG_WARNING, "Unable to set fragment size -- sound may be choppy\n");
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}
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#endif
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sound = fd;
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return 0;
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}
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static int intercom_exec(struct ast_channel *chan, void *data)
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{
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int res = 0;
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struct localuser *u;
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struct ast_frame *f;
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int oreadformat;
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LOCAL_USER_ADD(u);
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/* Remember original read format */
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oreadformat = chan->readformat;
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/* Set mode to signed linear */
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res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
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if (res < 0) {
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ast_log(LOG_WARNING, "Unable to set format to signed linear on channel %s\n", chan->name);
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return -1;
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}
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/* Read packets from the channel */
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while(!res) {
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res = ast_waitfor(chan, -1);
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if (res > 0) {
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res = 0;
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f = ast_read(chan);
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if (f) {
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if (f->frametype == AST_FRAME_DTMF) {
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ast_frfree(f);
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break;
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} else {
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if (f->frametype == AST_FRAME_VOICE) {
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if (f->subclass == AST_FORMAT_SLINEAR) {
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res = write_audio(f->data, f->datalen);
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if (res > 0)
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res = 0;
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} else
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ast_log(LOG_DEBUG, "Unable to handle non-signed linear frame (%d)\n", f->subclass);
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}
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}
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ast_frfree(f);
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} else
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res = -1;
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}
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}
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LOCAL_USER_REMOVE(u);
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if (!res)
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ast_set_read_format(chan, oreadformat);
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return res;
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}
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int unload_module(void)
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{
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STANDARD_HANGUP_LOCALUSERS;
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if (sound > -1)
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close(sound);
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return ast_unregister_application(app);
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}
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int load_module(void)
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{
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if (create_audio())
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return -1;
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return ast_register_application(app, intercom_exec, synopsis, descrip);
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}
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char *description(void)
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{
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return tdesc;
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}
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int usecount(void)
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{
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int res;
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STANDARD_USECOUNT(res);
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return res;
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}
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char *key()
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{
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return ASTERISK_GPL_KEY;
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}
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