Fri May 26 01:45:27 2006

Asterisk developer's documentation


app_intercom.c

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00001 /*
00002  * Asterisk -- An open source telephony toolkit.
00003  *
00004  * Copyright (C) 1999 - 2005, Digium, Inc.
00005  *
00006  * Mark Spencer <markster@digium.com>
00007  *
00008  * See http://www.asterisk.org for more information about
00009  * the Asterisk project. Please do not directly contact
00010  * any of the maintainers of this project for assistance;
00011  * the project provides a web site, mailing lists and IRC
00012  * channels for your use.
00013  *
00014  * This program is free software, distributed under the terms of
00015  * the GNU General Public License Version 2. See the LICENSE file
00016  * at the top of the source tree.
00017  */
00018 
00019 /*! \file
00020  *
00021  * \brief Use /dev/dsp as an intercom.
00022  * 
00023  * \ingroup applications
00024  */
00025  
00026 #include <stdio.h>
00027 #include <unistd.h>
00028 #include <errno.h>
00029 #include <sys/ioctl.h>
00030 #include <string.h>
00031 #include <stdlib.h>
00032 #include <sys/time.h>
00033 #include <netinet/in.h>
00034 
00035 #if defined(__linux__)
00036 #include <linux/soundcard.h>
00037 #elif defined(__FreeBSD__)
00038 #include <sys/soundcard.h>
00039 #else
00040 #include <soundcard.h>
00041 #endif
00042 
00043 #include "asterisk.h"
00044 
00045 ASTERISK_FILE_VERSION(__FILE__, "$Revision: 7221 $")
00046 
00047 #include "asterisk/lock.h"
00048 #include "asterisk/file.h"
00049 #include "asterisk/frame.h"
00050 #include "asterisk/logger.h"
00051 #include "asterisk/channel.h"
00052 #include "asterisk/pbx.h"
00053 #include "asterisk/module.h"
00054 #include "asterisk/translate.h"
00055 
00056 #ifdef __OpenBSD__
00057 #define DEV_DSP "/dev/audio"
00058 #else
00059 #define DEV_DSP "/dev/dsp"
00060 #endif
00061 
00062 /* Number of 32 byte buffers -- each buffer is 2 ms */
00063 #define BUFFER_SIZE 32
00064 
00065 static char *tdesc = "Intercom using /dev/dsp for output";
00066 
00067 static char *app = "Intercom";
00068 
00069 static char *synopsis = "(Obsolete) Send to Intercom";
00070 static char *descrip = 
00071 "  Intercom(): Sends the user to the intercom (i.e. /dev/dsp).  This program\n"
00072 "is generally considered  obselete by the chan_oss module.  User can terminate\n"with a DTMF tone, or by hangup.\n";
00073 
00074 STANDARD_LOCAL_USER;
00075 
00076 LOCAL_USER_DECL;
00077 
00078 AST_MUTEX_DEFINE_STATIC(sound_lock);
00079 static int sound = -1;
00080 
00081 static int write_audio(short *data, int len)
00082 {
00083    int res;
00084    struct audio_buf_info info;
00085    ast_mutex_lock(&sound_lock);
00086    if (sound < 0) {
00087       ast_log(LOG_WARNING, "Sound device closed?\n");
00088       ast_mutex_unlock(&sound_lock);
00089       return -1;
00090    }
00091     if (ioctl(sound, SNDCTL_DSP_GETOSPACE, &info)) {
00092       ast_log(LOG_WARNING, "Unable to read output space\n");
00093       ast_mutex_unlock(&sound_lock);
00094         return -1;
00095     }
00096    res = write(sound, data, len);
00097    ast_mutex_unlock(&sound_lock);
00098    return res;
00099 }
00100 
00101 static int create_audio(void)
00102 {
00103    int fmt, desired, res, fd;
00104    fd = open(DEV_DSP, O_WRONLY);
00105    if (fd < 0) {
00106       ast_log(LOG_WARNING, "Unable to open %s: %s\n", DEV_DSP, strerror(errno));
00107       close(fd);
00108       return -1;
00109    }
00110    fmt = AFMT_S16_LE;
00111    res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
00112    if (res < 0) {
00113       ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n");
00114       close(fd);
00115       return -1;
00116    }
00117    fmt = 0;
00118    res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
00119    if (res < 0) {
00120       ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
00121       close(fd);
00122       return -1;
00123    }
00124    /* 8000 Hz desired */
00125    desired = 8000;
00126    fmt = desired;
00127    res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt);
00128    if (res < 0) {
00129       ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
00130       close(fd);
00131       return -1;
00132    }
00133    if (fmt != desired) {
00134       ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n", desired, fmt);
00135    }
00136 #if 1
00137    /* 2 bytes * 15 units of 2^5 = 32 bytes per buffer */
00138    fmt = ((BUFFER_SIZE) << 16) | (0x0005);
00139    res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
00140    if (res < 0) {
00141       ast_log(LOG_WARNING, "Unable to set fragment size -- sound may be choppy\n");
00142    }
00143 #endif
00144    sound = fd;
00145    return 0;
00146 }
00147 
00148 static int intercom_exec(struct ast_channel *chan, void *data)
00149 {
00150    int res = 0;
00151    struct localuser *u;
00152    struct ast_frame *f;
00153    int oreadformat;
00154    LOCAL_USER_ADD(u);
00155    /* Remember original read format */
00156    oreadformat = chan->readformat;
00157    /* Set mode to signed linear */
00158    res = ast_set_read_format(chan, AST_FORMAT_SLINEAR);
00159    if (res < 0) {
00160       ast_log(LOG_WARNING, "Unable to set format to signed linear on channel %s\n", chan->name);
00161       LOCAL_USER_REMOVE(u);
00162       return -1;
00163    }
00164    /* Read packets from the channel */
00165    while(!res) {
00166       res = ast_waitfor(chan, -1);
00167       if (res > 0) {
00168          res = 0;
00169          f = ast_read(chan);
00170          if (f) {
00171             if (f->frametype == AST_FRAME_DTMF) {
00172                ast_frfree(f);
00173                break;
00174             } else {
00175                if (f->frametype == AST_FRAME_VOICE) {
00176                   if (f->subclass == AST_FORMAT_SLINEAR) {
00177                      res = write_audio(f->data, f->datalen);
00178                      if (res > 0)
00179                         res = 0;
00180                   } else
00181                      ast_log(LOG_DEBUG, "Unable to handle non-signed linear frame (%d)\n", f->subclass);
00182                } 
00183             }
00184             ast_frfree(f);
00185          } else
00186             res = -1;
00187       }
00188    }
00189    
00190    if (!res)
00191       ast_set_read_format(chan, oreadformat);
00192 
00193    LOCAL_USER_REMOVE(u);
00194 
00195    return res;
00196 }
00197 
00198 int unload_module(void)
00199 {
00200    int res;
00201 
00202    if (sound > -1)
00203       close(sound);
00204 
00205    res = ast_unregister_application(app);
00206 
00207    STANDARD_HANGUP_LOCALUSERS;
00208 
00209    return res;
00210 }
00211 
00212 int load_module(void)
00213 {
00214    if (create_audio())
00215       return -1;
00216    return ast_register_application(app, intercom_exec, synopsis, descrip);
00217 }
00218 
00219 char *description(void)
00220 {
00221    return tdesc;
00222 }
00223 
00224 int usecount(void)
00225 {
00226    int res;
00227    STANDARD_USECOUNT(res);
00228    return res;
00229 }
00230 
00231 char *key()
00232 {
00233    return ASTERISK_GPL_KEY;
00234 }

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