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Diffstat (limited to '1.2-netsec/channels/chan_oss.c')
| -rw-r--r-- | 1.2-netsec/channels/chan_oss.c | 1438 |
1 files changed, 0 insertions, 1438 deletions
diff --git a/1.2-netsec/channels/chan_oss.c b/1.2-netsec/channels/chan_oss.c deleted file mode 100644 index 70b70333e..000000000 --- a/1.2-netsec/channels/chan_oss.c +++ /dev/null @@ -1,1438 +0,0 @@ -/* - * Asterisk -- An open source telephony toolkit. - * - * Copyright (C) 1999 - 2005, Digium, Inc. - * - * Mark Spencer <markster@digium.com> - * - * FreeBSD changes and multiple device support by Luigi Rizzo, 2005.05.25 - * note-this code best seen with ts=8 (8-spaces tabs) in the editor - * - * See http://www.asterisk.org for more information about - * the Asterisk project. Please do not directly contact - * any of the maintainers of this project for assistance; - * the project provides a web site, mailing lists and IRC - * channels for your use. - * - * This program is free software, distributed under the terms of - * the GNU General Public License Version 2. See the LICENSE file - * at the top of the source tree. - */ - -/*! \file - * - * \brief Channel driver for OSS sound cards - * - * \par See also - * \arg \ref Config_oss - * - * \ingroup channel_drivers - */ - -#include <stdio.h> -#include <ctype.h> /* for isalnum */ -#include <string.h> -#include <unistd.h> -#include <sys/ioctl.h> -#include <fcntl.h> -#include <sys/time.h> -#include <stdlib.h> -#include <errno.h> - - -#ifdef __linux -#include <linux/soundcard.h> -#elif defined(__FreeBSD__) -#include <sys/soundcard.h> -#else -#include <soundcard.h> -#endif - -#include "asterisk.h" - -ASTERISK_FILE_VERSION(__FILE__, "$Revision$") - -#include "asterisk/lock.h" -#include "asterisk/frame.h" -#include "asterisk/logger.h" -#include "asterisk/channel.h" -#include "asterisk/module.h" -#include "asterisk/options.h" -#include "asterisk/pbx.h" -#include "asterisk/config.h" - -#include "asterisk/cli.h" -#include "asterisk/utils.h" -#include "asterisk/causes.h" -#include "asterisk/endian.h" - -/* ringtones we use */ -#include "busy.h" -#include "ringtone.h" -#include "ring10.h" -#include "answer.h" - -/* - * Basic mode of operation: - * - * we have one keyboard (which receives commands from the keyboard) - * and multiple headset's connected to audio cards. - * Cards/Headsets are named as the sections of oss.conf. - * The section called [general] contains the default parameters. - * - * At any time, the keyboard is attached to one card, and you - * can switch among them using the command 'console foo' - * where 'foo' is the name of the card you want. - * - * oss.conf parameters are - -[general] -; general config options, default values are shown -; all but debug can go also in the device-specific sections. -; debug=0x0 ; misc debug flags, default is 0 - -[card1] -; autoanswer = no ; no autoanswer on call -; autohangup = yes ; hangup when other party closes -; extension=s ; default extension to call -; context=default ; default context -; language="" ; default language -; overridecontext=yes ; the whole dial string is considered an extension. - ; if no, the last @ will start the context - -; device=/dev/dsp ; device to open -; mixer="-f /dev/mixer0 pcm 80 ; mixer command to run on start -; queuesize=10 ; frames in device driver -; frags=8 ; argument to SETFRAGMENT - -.. and so on for the other cards. - - */ - -/* - * Helper macros to parse config arguments. They will go in a common - * header file if their usage is globally accepted. In the meantime, - * we define them here. Typical usage is as below. - * Remember to open a block right before M_START (as it declares - * some variables) and use the M_* macros WITHOUT A SEMICOLON: - * - * { - * M_START(v->name, v->value) - * - * M_BOOL("dothis", x->flag1) - * M_STR("name", x->somestring) - * M_F("bar", some_c_code) - * M_END(some_final_statement) - * ... other code in the block - * } - * - * XXX NOTE these macros should NOT be replicated in other parts of asterisk. - * Likely we will come up with a better way of doing config file parsing. - */ -#define M_START(var, val) \ - char *__s = var; char *__val = val; -#define M_END(x) x; -#define M_F(tag, f) if (!strcasecmp((__s), tag)) { f; } else -#define M_BOOL(tag, dst) M_F(tag, (dst) = ast_true(__val) ) -#define M_UINT(tag, dst) M_F(tag, (dst) = strtoul(__val, NULL, 0) ) -#define M_STR(tag, dst) M_F(tag, ast_copy_string(dst, __val, sizeof(dst))) - -/* - * The following parameters are used in the driver: - * - * FRAME_SIZE the size of an audio frame, in samples. - * 160 is used almost universally, so you should not change it. - * - * FRAGS the argument for the SETFRAGMENT ioctl. - * Overridden by the 'frags' parameter in oss.conf - * - * Bits 0-7 are the base-2 log of the device's block size, - * bits 16-31 are the number of blocks in the driver's queue. - * There are a lot of differences in the way this parameter - * is supported by different drivers, so you may need to - * experiment a bit with the value. - * A good default for linux is 30 blocks of 64 bytes, which - * results in 6 frames of 320 bytes (160 samples). - * FreeBSD works decently with blocks of 256 or 512 bytes, - * leaving the number unspecified. - * Note that this only refers to the device buffer size, - * this module will then try to keep the lenght of audio - * buffered within small constraints. - * - * QUEUE_SIZE The max number of blocks actually allowed in the device - * driver's buffer, irrespective of the available number. - * Overridden by the 'queuesize' parameter in oss.conf - * - * Should be >=2, and at most as large as the hw queue above - * (otherwise it will never be full). - */ - -#define FRAME_SIZE 160 -#define QUEUE_SIZE 10 - -#if defined(__FreeBSD__) -#define FRAGS 0x8 -#else -#define FRAGS ( ( (6 * 5) << 16 ) | 0x6 ) -#endif - -/* - * XXX text message sizes are probably 256 chars, but i am - * not sure if there is a suitable definition anywhere. - */ -#define TEXT_SIZE 256 - -#if 0 -#define TRYOPEN 1 /* try to open on startup */ -#endif -#define O_CLOSE 0x444 /* special 'close' mode for device */ -/* Which device to use */ -#if defined( __OpenBSD__ ) || defined( __NetBSD__ ) -#define DEV_DSP "/dev/audio" -#else -#define DEV_DSP "/dev/dsp" -#endif - -#ifndef MIN -#define MIN(a,b) ((a) < (b) ? (a) : (b)) -#endif -#ifndef MAX -#define MAX(a,b) ((a) > (b) ? (a) : (b)) -#endif - - -static int usecnt; -AST_MUTEX_DEFINE_STATIC(usecnt_lock); - -static char *config = "oss.conf"; /* default config file */ - -static int oss_debug; - -/* - * Each sound is made of 'datalen' samples of sound, repeated as needed to - * generate 'samplen' samples of data, then followed by 'silencelen' samples - * of silence. The loop is repeated if 'repeat' is set. - */ -struct sound { - int ind; - char *desc; - short *data; - int datalen; - int samplen; - int silencelen; - int repeat; -}; - -static struct sound sounds[] = { - { AST_CONTROL_RINGING, "RINGING", ringtone, sizeof(ringtone)/2, 16000, 32000, 1 }, - { AST_CONTROL_BUSY, "BUSY", busy, sizeof(busy)/2, 4000, 4000, 1 }, - { AST_CONTROL_CONGESTION, "CONGESTION", busy, sizeof(busy)/2, 2000, 2000, 1 }, - { AST_CONTROL_RING, "RING10", ring10, sizeof(ring10)/2, 16000, 32000, 1 }, - { AST_CONTROL_ANSWER, "ANSWER", answer, sizeof(answer)/2, 2200, 0, 0 }, - { -1, NULL, 0, 0, 0, 0 }, /* end marker */ -}; - - -/* - * descriptor for one of our channels. - * There is one used for 'default' values (from the [general] entry in - * the configuration file), and then one instance for each device - * (the default is cloned from [general], others are only created - * if the relevant section exists). - */ -struct chan_oss_pvt { - struct chan_oss_pvt *next; - - char *type; /* XXX maybe take the one from oss_tech */ - char *name; - /* - * cursound indicates which in struct sound we play. -1 means nothing, - * any other value is a valid sound, in which case sampsent indicates - * the next sample to send in [0..samplen + silencelen] - * nosound is set to disable the audio data from the channel - * (so we can play the tones etc.). - */ - int sndcmd[2]; /* Sound command pipe */ - int cursound; /* index of sound to send */ - int sampsent; /* # of sound samples sent */ - int nosound; /* set to block audio from the PBX */ - - int total_blocks; /* total blocks in the output device */ - int sounddev; - enum { M_UNSET, M_FULL, M_READ, M_WRITE } duplex; - int autoanswer; - int autohangup; - int hookstate; - char *mixer_cmd; /* initial command to issue to the mixer */ - unsigned int queuesize; /* max fragments in queue */ - unsigned int frags; /* parameter for SETFRAGMENT */ - - int warned; /* various flags used for warnings */ -#define WARN_used_blocks 1 -#define WARN_speed 2 -#define WARN_frag 4 - int w_errors; /* overfull in the write path */ - struct timeval lastopen; - - int overridecontext; - int mute; - char device[64]; /* device to open */ - - pthread_t sthread; - - struct ast_channel *owner; - char ext[AST_MAX_EXTENSION]; - char ctx[AST_MAX_CONTEXT]; - char language[MAX_LANGUAGE]; - - /* buffers used in oss_write */ - char oss_write_buf[FRAME_SIZE*2]; - int oss_write_dst; - /* buffers used in oss_read - AST_FRIENDLY_OFFSET space for headers - * plus enough room for a full frame - */ - char oss_read_buf[FRAME_SIZE * 2 + AST_FRIENDLY_OFFSET]; - int readpos; /* read position above */ - struct ast_frame read_f; /* returned by oss_read */ -}; - -static struct chan_oss_pvt oss_default = { - .type = "Console", - .cursound = -1, - .sounddev = -1, - .duplex = M_UNSET, /* XXX check this */ - .autoanswer = 1, - .autohangup = 1, - .queuesize = QUEUE_SIZE, - .frags = FRAGS, - .ext = "s", - .ctx = "default", - .readpos = AST_FRIENDLY_OFFSET, /* start here on reads */ - .lastopen = { 0, 0 }, -}; - -static char *oss_active; /* the active device */ - -static int setformat(struct chan_oss_pvt *o, int mode); - -static struct ast_channel *oss_request(const char *type, int format, void *data -, int *cause); -static int oss_digit(struct ast_channel *c, char digit); -static int oss_text(struct ast_channel *c, const char *text); -static int oss_hangup(struct ast_channel *c); -static int oss_answer(struct ast_channel *c); -static struct ast_frame *oss_read(struct ast_channel *chan); -static int oss_call(struct ast_channel *c, char *dest, int timeout); -static int oss_write(struct ast_channel *chan, struct ast_frame *f); -static int oss_indicate(struct ast_channel *chan, int cond); -static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan); - -static const struct ast_channel_tech oss_tech = { - .type = "Console", - .description = "OSS Console Channel Driver", - .capabilities = AST_FORMAT_SLINEAR, - .requester = oss_request, - .send_digit = oss_digit, - .send_text = oss_text, - .hangup = oss_hangup, - .answer = oss_answer, - .read = oss_read, - .call = oss_call, - .write = oss_write, - .indicate = oss_indicate, - .fixup = oss_fixup, -}; - -/* - * returns a pointer to the descriptor with the given name - */ -static struct chan_oss_pvt *find_desc(char *dev) -{ - struct chan_oss_pvt *o; - - for (o = oss_default.next; o && strcmp(o->name, dev) != 0; o = o->next) - ; - if (o == NULL) - ast_log(LOG_WARNING, "could not find <%s>\n", dev); - return o; -} - -/* - * split a string in extension-context, returns pointers to malloc'ed - * strings. - * If we do not have 'overridecontext' then the last @ is considered as - * a context separator, and the context is overridden. - * This is usually not very necessary as you can play with the dialplan, - * and it is nice not to need it because you have '@' in SIP addresses. - * Return value is the buffer address. - */ -static char *ast_ext_ctx(const char *src, char **ext, char **ctx) -{ - struct chan_oss_pvt *o = find_desc(oss_active); - - if (ext == NULL || ctx == NULL) - return NULL; /* error */ - *ext = *ctx = NULL; - if (src && *src != '\0') - *ext = strdup(src); - if (*ext == NULL) - return NULL; - if (!o->overridecontext) { - /* parse from the right */ - *ctx = strrchr(*ext, '@'); - if (*ctx) - *(*ctx)++ = '\0'; - } - return *ext; -} - -/* - * Returns the number of blocks used in the audio output channel - */ -static int used_blocks(struct chan_oss_pvt *o) -{ - struct audio_buf_info info; - - if (ioctl(o->sounddev, SNDCTL_DSP_GETOSPACE, &info)) { - if (! (o->warned & WARN_used_blocks)) { - ast_log(LOG_WARNING, "Error reading output space\n"); - o->warned |= WARN_used_blocks; - } - return 1; - } - if (o->total_blocks == 0) { - if (0) /* debugging */ - ast_log(LOG_WARNING, "fragtotal %d size %d avail %d\n", - info.fragstotal, - info.fragsize, - info.fragments); - o->total_blocks = info.fragments; - } - return o->total_blocks - info.fragments; -} - -/* Write an exactly FRAME_SIZE sized frame */ -static int soundcard_writeframe(struct chan_oss_pvt *o, short *data) -{ - int res; - - if (o->sounddev < 0) - setformat(o, O_RDWR); - if (o->sounddev < 0) - return 0; /* not fatal */ - /* - * Nothing complex to manage the audio device queue. - * If the buffer is full just drop the extra, otherwise write. - * XXX in some cases it might be useful to write anyways after - * a number of failures, to restart the output chain. - */ - res = used_blocks(o); - if (res > o->queuesize) { /* no room to write a block */ - if (o->w_errors++ == 0 && (oss_debug & 0x4)) - ast_log(LOG_WARNING, "write: used %d blocks (%d)\n", - res, o->w_errors); - return 0; - } - o->w_errors = 0; - return write(o->sounddev, ((void *)data), FRAME_SIZE * 2); -} - -/* - * Handler for 'sound writable' events from the sound thread. - * Builds a frame from the high level description of the sounds, - * and passes it to the audio device. - * The actual sound is made of 1 or more sequences of sound samples - * (s->datalen, repeated to make s->samplen samples) followed by - * s->silencelen samples of silence. The position in the sequence is stored - * in o->sampsent, which goes between 0 .. s->samplen+s->silencelen. - * In case we fail to write a frame, don't update o->sampsent. - */ -static void send_sound(struct chan_oss_pvt *o) -{ - short myframe[FRAME_SIZE]; - int ofs, l, start; - int l_sampsent = o->sampsent; - struct sound *s; - - if (o->cursound < 0) /* no sound to send */ - return; - s = &sounds[o->cursound]; - for (ofs = 0; ofs < FRAME_SIZE; ofs += l) { - l = s->samplen - l_sampsent; /* # of available samples */ - if (l > 0) { - start = l_sampsent % s->datalen; /* source offset */ - if (l > FRAME_SIZE - ofs) /* don't overflow the frame */ - l = FRAME_SIZE - ofs; - if (l > s->datalen - start) /* don't overflow the source */ - l = s->datalen - start; - bcopy(s->data + start, myframe + ofs, l*2); - if (0) - ast_log(LOG_WARNING, "send_sound sound %d/%d of %d into %d\n", - l_sampsent, l, s->samplen, ofs); - l_sampsent += l; - } else { /* end of samples, maybe some silence */ - static const short silence[FRAME_SIZE] = {0, }; - - l += s->silencelen; - if (l > 0) { - if (l > FRAME_SIZE - ofs) - l = FRAME_SIZE - ofs; - bcopy(silence, myframe + ofs, l*2); - l_sampsent += l; - } else { /* silence is over, restart sound if loop */ - if (s->repeat == 0) { /* last block */ - o->cursound = -1; - o->nosound = 0; /* allow audio data */ - if (ofs < FRAME_SIZE) /* pad with silence */ - bcopy(silence, myframe + ofs, (FRAME_SIZE - ofs)*2); - } - l_sampsent = 0; - } - } - } - l = soundcard_writeframe(o, myframe); - if (l > 0) - o->sampsent = l_sampsent; /* update status */ -} - -static void *sound_thread(void *arg) -{ - char ign[4096]; - struct chan_oss_pvt *o = (struct chan_oss_pvt *)arg; - - /* - * Just in case, kick the driver by trying to read from it. - * Ignore errors - this read is almost guaranteed to fail. - */ - read(o->sounddev, ign, sizeof(ign)); - for (;;) { - fd_set rfds, wfds; - int maxfd, res; - - FD_ZERO(&rfds); - FD_ZERO(&wfds); - FD_SET(o->sndcmd[0], &rfds); - maxfd = o->sndcmd[0]; /* pipe from the main process */ - if (o->cursound > -1 && o->sounddev < 0) - setformat(o, O_RDWR); /* need the channel, try to reopen */ - else if (o->cursound == -1 && o->owner == NULL) - setformat(o, O_CLOSE); /* can close */ - if (o->sounddev > -1) { - if (!o->owner) { /* no one owns the audio, so we must drain it */ - FD_SET(o->sounddev, &rfds); - maxfd = MAX(o->sounddev, maxfd); - } - if (o->cursound > -1) { - FD_SET(o->sounddev, &wfds); - maxfd = MAX(o->sounddev, maxfd); - } - } - /* ast_select emulates linux behaviour in terms of timeout handling */ - res = ast_select(maxfd + 1, &rfds, &wfds, NULL, NULL); - if (res < 1) { - ast_log(LOG_WARNING, "select failed: %s\n", strerror(errno)); - sleep(1); - continue; - } - if (FD_ISSET(o->sndcmd[0], &rfds)) { - /* read which sound to play from the pipe */ - int i, what = -1; - - read(o->sndcmd[0], &what, sizeof(what)); - for (i = 0; sounds[i].ind != -1; i++) { - if (sounds[i].ind == what) { - o->cursound = i; - o->sampsent = 0; - o->nosound = 1; /* block audio from pbx */ - break; - } - } - if (sounds[i].ind == -1) - ast_log(LOG_WARNING, "invalid sound index: %d\n", what); - } - if (o->sounddev > -1) { - if (FD_ISSET(o->sounddev, &rfds)) /* read and ignore errors */ - read(o->sounddev, ign, sizeof(ign)); - if (FD_ISSET(o->sounddev, &wfds)) - send_sound(o); - } - } - return NULL; /* Never reached */ -} - -/* - * reset and close the device if opened, - * then open and initialize it in the desired mode, - * trigger reads and writes so we can start using it. - */ -static int setformat(struct chan_oss_pvt *o, int mode) -{ - int fmt, desired, res, fd; - - if (o->sounddev >= 0) { - ioctl(o->sounddev, SNDCTL_DSP_RESET, 0); - close(o->sounddev); - o->duplex = M_UNSET; - o->sounddev = -1; - } - if (mode == O_CLOSE) /* we are done */ - return 0; - if (ast_tvdiff_ms(ast_tvnow(), o->lastopen) < 1000) - return -1; /* don't open too often */ - o->lastopen = ast_tvnow(); - fd = o->sounddev = open(o->device, mode |O_NONBLOCK); - if (fd < 0) { - ast_log(LOG_WARNING, "Unable to re-open DSP device %s: %s\n", - o->device, strerror(errno)); - return -1; - } - if (o->owner) - o->owner->fds[0] = fd; - -#if __BYTE_ORDER == __LITTLE_ENDIAN - fmt = AFMT_S16_LE; -#else - fmt = AFMT_S16_BE; -#endif - res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt); - if (res < 0) { - ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n"); - return -1; - } - switch (mode) { - case O_RDWR: - res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0); - /* Check to see if duplex set (FreeBSD Bug)*/ - res = ioctl(fd, SNDCTL_DSP_GETCAPS, &fmt); - if (res == 0 && (fmt & DSP_CAP_DUPLEX)) { - if (option_verbose > 1) - ast_verbose(VERBOSE_PREFIX_2 "Console is full duplex\n"); - o->duplex = M_FULL; - }; - break; - case O_WRONLY: - o->duplex = M_WRITE; - break; - case O_RDONLY: - o->duplex = M_READ; - break; - } - - fmt = 0; - res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt); - if (res < 0) { - ast_log(LOG_WARNING, "Failed to set audio device to mono\n"); - return -1; - } - fmt = desired = 8000; /* 8000 Hz desired */ - res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt); - - if (res < 0) { - ast_log(LOG_WARNING, "Failed to set audio device to mono\n"); - return -1; - } - if (fmt != desired) { - if (!(o->warned & WARN_speed)) { - ast_log(LOG_WARNING, - "Requested %d Hz, got %d Hz -- sound may be choppy\n", - desired, fmt); - o->warned |= WARN_speed; - } - } - /* - * on Freebsd, SETFRAGMENT does not work very well on some cards. - * Default to use 256 bytes, let the user override - */ - if (o->frags) { - fmt = o->frags; - res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt); - if (res < 0) { - if (!(o->warned & WARN_frag)) { - ast_log(LOG_WARNING, - "Unable to set fragment size -- sound may be choppy\n"); - o->warned |= WARN_frag; - } - } - } - /* on some cards, we need SNDCTL_DSP_SETTRIGGER to start outputting */ - res = PCM_ENABLE_INPUT | PCM_ENABLE_OUTPUT; - res = ioctl(fd, SNDCTL_DSP_SETTRIGGER, &res); - /* it may fail if we are in half duplex, never mind */ - return 0; -} - -/* - * some of the standard methods supported by channels. - */ -static int oss_digit(struct ast_channel *c, char digit) -{ - /* no better use for received digits than print them */ - ast_verbose( " << Console Received digit %c >> \n", digit); - return 0; -} - -static int oss_text(struct ast_channel *c, const char *text) -{ - /* print received messages */ - ast_verbose( " << Console Received text %s >> \n", text); - return 0; -} - -/* Play ringtone 'x' on device 'o' */ -static void ring(struct chan_oss_pvt *o, int x) -{ - write(o->sndcmd[1], &x, sizeof(x)); -} - - -/* - * handler for incoming calls. Either autoanswer, or start ringing - */ -static int oss_call(struct ast_channel *c, char *dest, int timeout) -{ - struct chan_oss_pvt *o = c->tech_pvt; - struct ast_frame f = { 0, }; - - ast_verbose(" << Call to '%s' on console from <%s><%s><%s> >>\n", - dest, c->cid.cid_dnid, c->cid.cid_num, c->cid.cid_name); - if (o->autoanswer) { - ast_verbose( " << Auto-answered >> \n" ); - f.frametype = AST_FRAME_CONTROL; - f.subclass = AST_CONTROL_ANSWER; - ast_queue_frame(c, &f); - } else { - ast_verbose("<< Type 'answer' to answer, or use 'autoanswer' for future calls >> \n"); - f.frametype = AST_FRAME_CONTROL; - f.subclass = AST_CONTROL_RINGING; - ast_queue_frame(c, &f); - ring(o, AST_CONTROL_RING); - } - return 0; -} - -/* - * remote side answered the phone - */ -static int oss_answer(struct ast_channel *c) -{ - struct chan_oss_pvt *o = c->tech_pvt; - - ast_verbose( " << Console call has been answered >> \n"); -#if 0 - /* play an answer tone (XXX do we really need it ?) */ - ring(o, AST_CONTROL_ANSWER); -#endif - ast_setstate(c, AST_STATE_UP); - o->cursound = -1; - o->nosound=0; - return 0; -} - -static int oss_hangup(struct ast_channel *c) -{ - struct chan_oss_pvt *o = c->tech_pvt; - - o->cursound = -1; - o->nosound = 0; - c->tech_pvt = NULL; - o->owner = NULL; - ast_verbose( " << Hangup on console >> \n"); - ast_mutex_lock(&usecnt_lock); /* XXX not sure why */ - usecnt--; - ast_mutex_unlock(&usecnt_lock); - if (o->hookstate) { - if (o->autoanswer || o->autohangup) { - /* Assume auto-hangup too */ - o->hookstate = 0; - setformat(o, O_CLOSE); - } else { - /* Make congestion noise */ - ring(o, AST_CONTROL_CONGESTION); - } - } - return 0; -} - -/* used for data coming from the network */ -static int oss_write(struct ast_channel *c, struct ast_frame *f) -{ - int src; - struct chan_oss_pvt *o = c->tech_pvt; - - /* Immediately return if no sound is enabled */ - if (o->nosound) - return 0; - /* Stop any currently playing sound */ - o->cursound = -1; - /* - * we could receive a block which is not a multiple of our - * FRAME_SIZE, so buffer it locally and write to the device - * in FRAME_SIZE chunks. - * Keep the residue stored for future use. - */ - src = 0; /* read position into f->data */ - while ( src < f->datalen ) { - /* Compute spare room in the buffer */ - int l = sizeof(o->oss_write_buf) - o->oss_write_dst; - - if (f->datalen - src >= l) { /* enough to fill a frame */ - memcpy(o->oss_write_buf + o->oss_write_dst, - f->data + src, l); - soundcard_writeframe(o, (short *)o->oss_write_buf); - src += l; - o->oss_write_dst = 0; - } else { /* copy residue */ - l = f->datalen - src; - memcpy(o->oss_write_buf + o->oss_write_dst, - f->data + src, l); - src += l; /* but really, we are done */ - o->oss_write_dst += l; - } - } - return 0; -} - -static struct ast_frame *oss_read(struct ast_channel *c) -{ - int res; - struct chan_oss_pvt *o = c->tech_pvt; - struct ast_frame *f = &o->read_f; - - /* prepare a NULL frame in case we don't have enough data to return */ - bzero(f, sizeof(struct ast_frame)); - f->frametype = AST_FRAME_NULL; - f->src = o->type; - - res = read(o->sounddev, o->oss_read_buf + o->readpos, - sizeof(o->oss_read_buf) - o->readpos); - if (res < 0) /* audio data not ready, return a NULL frame */ - return f; - - o->readpos += res; - if (o->readpos < sizeof(o->oss_read_buf)) /* not enough samples */ - return f; - - if (o->mute) - return f; - - o->readpos = AST_FRIENDLY_OFFSET; /* reset read pointer for next frame */ - if (c->_state != AST_STATE_UP) /* drop data if frame is not up */ - return f; - /* ok we can build and deliver the frame to the caller */ - f->frametype = AST_FRAME_VOICE; - f->subclass = AST_FORMAT_SLINEAR; - f->samples = FRAME_SIZE; - f->datalen = FRAME_SIZE * 2; - f->data = o->oss_read_buf + AST_FRIENDLY_OFFSET; - f->offset = AST_FRIENDLY_OFFSET; - return f; -} - -static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan) -{ - struct chan_oss_pvt *o = newchan->tech_pvt; - o->owner = newchan; - return 0; -} - -static int oss_indicate(struct ast_channel *c, int cond) -{ - struct chan_oss_pvt *o = c->tech_pvt; - int res; - - switch(cond) { - case AST_CONTROL_BUSY: - case AST_CONTROL_CONGESTION: - case AST_CONTROL_RINGING: - res = cond; - break; - - case -1: - o->cursound = -1; - o->nosound = 0; /* when cursound is -1 nosound must be 0 */ - return 0; - - case AST_CONTROL_VIDUPDATE: - res = -1; - break; - default: - ast_log(LOG_WARNING, - "Don't know how to display condition %d on %s\n", - cond, c->name); - return -1; - } - if (res > -1) - ring(o, res); - return 0; -} - -/* - * allocate a new channel. - */ -static struct ast_channel *oss_new(struct chan_oss_pvt *o, - char *ext, char *ctx, int state) -{ - struct ast_channel *c; - - c = ast_channel_alloc(1); - if (c == NULL) - return NULL; - c->tech = &oss_tech; - snprintf(c->name, sizeof(c->name), "OSS/%s", o->device + 5); - c->type = o->type; - c->fds[0] = o->sounddev; /* -1 if device closed, override later */ - c->nativeformats = AST_FORMAT_SLINEAR; - c->readformat = AST_FORMAT_SLINEAR; - c->writeformat = AST_FORMAT_SLINEAR; - c->tech_pvt = o; - - if (!ast_strlen_zero(ctx)) - ast_copy_string(c->context, ctx, sizeof(c->context)); - if (!ast_strlen_zero(ext)) - ast_copy_string(c->exten, ext, sizeof(c->exten)); - if (!ast_strlen_zero(o->language)) - ast_copy_string(c->language, o->language, sizeof(c->language)); - - o->owner = c; - ast_setstate(c, state); - ast_mutex_lock(&usecnt_lock); - usecnt++; - ast_mutex_unlock(&usecnt_lock); - ast_update_use_count(); - if (state != AST_STATE_DOWN) { - if (ast_pbx_start(c)) { - ast_log(LOG_WARNING, "Unable to start PBX on %s\n", c->name); - ast_hangup(c); - o->owner = c = NULL; - /* XXX what about the channel itself ? */ - /* XXX what about usecnt ? */ - } - } - return c; -} - -static struct ast_channel *oss_request(const char *type, - int format, void *data, int *cause) -{ - struct ast_channel *c; - struct chan_oss_pvt *o = find_desc(data); - - ast_log(LOG_WARNING, "oss_request ty <%s> data 0x%p <%s>\n", - type, data, (char *)data); - if (o == NULL) { - ast_log(LOG_NOTICE, "Device %s not found\n", (char *)data); - /* XXX we could default to 'dsp' perhaps ? */ - return NULL; - } - if ((format & AST_FORMAT_SLINEAR) == 0) { - ast_log(LOG_NOTICE, "Format 0x%x unsupported\n", format); - return NULL; - } - if (o->owner) { - ast_log(LOG_NOTICE, "Already have a call (chan %p) on the OSS channel\n", o->owner); - *cause = AST_CAUSE_BUSY; - return NULL; - } - c= oss_new(o, NULL, NULL, AST_STATE_DOWN); - if (c == NULL) { - ast_log(LOG_WARNING, "Unable to create new OSS channel\n"); - return NULL; - } - return c; -} - -static int console_autoanswer(int fd, int argc, char *argv[]) -{ - struct chan_oss_pvt *o = find_desc(oss_active); - - if (argc == 1) { - ast_cli(fd, "Auto answer is %s.\n", o->autoanswer ? "on" : "off"); - return RESULT_SUCCESS; - } - if (argc != 2) - return RESULT_SHOWUSAGE; - if (o == NULL) { - ast_log(LOG_WARNING, "Cannot find device %s (should not happen!)\n", - oss_active); - return RESULT_FAILURE; - } - if (!strcasecmp(argv[1], "on")) - o->autoanswer = -1; - else if (!strcasecmp(argv[1], "off")) - o->autoanswer = 0; - else - return RESULT_SHOWUSAGE; - return RESULT_SUCCESS; -} - -static char *autoanswer_complete(char *line, char *word, int pos, int state) -{ - int l = strlen(word); - - switch(state) { - case 0: - if (l && !strncasecmp(word, "on", MIN(l, 2))) - return strdup("on"); - case 1: - if (l && !strncasecmp(word, "off", MIN(l, 3))) - return strdup("off"); - default: - return NULL; - } - return NULL; -} - -static char autoanswer_usage[] = -"Usage: autoanswer [on|off]\n" -" Enables or disables autoanswer feature. If used without\n" -" argument, displays the current on/off status of autoanswer.\n" -" The default value of autoanswer is in 'oss.conf'.\n"; - -/* - * answer command from the console - */ -static int console_answer(int fd, int argc, char *argv[]) -{ - struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER }; - struct chan_oss_pvt *o = find_desc(oss_active); - - if (argc != 1) - return RESULT_SHOWUSAGE; - if (!o->owner) { - ast_cli(fd, "No one is calling us\n"); - return RESULT_FAILURE; - } - o->hookstate = 1; - o->cursound = -1; - o->nosound = 0; - ast_queue_frame(o->owner, &f); -#if 0 - /* XXX do we really need it ? considering we shut down immediately... */ - ring(o, AST_CONTROL_ANSWER); -#endif - return RESULT_SUCCESS; -} - -static char sendtext_usage[] = -"Usage: send text <message>\n" -" Sends a text message for display on the remote terminal.\n"; - -/* - * concatenate all arguments into a single string - */ -static int console_sendtext(int fd, int argc, char *argv[]) -{ - struct chan_oss_pvt *o = find_desc(oss_active); - int tmparg = 2; - char text2send[TEXT_SIZE] = ""; - struct ast_frame f = { 0, }; - - if (argc < 2) - return RESULT_SHOWUSAGE; - if (!o->owner) { - ast_cli(fd, "Not in a call\n"); - return RESULT_FAILURE; - } - while (tmparg < argc) { - strncat(text2send, argv[tmparg++], - sizeof(text2send) - strlen(text2send) - 1); - strncat(text2send, " ", - sizeof(text2send) - strlen(text2send) - 1); - } - if (!ast_strlen_zero(text2send)) { - text2send[strlen(text2send) - 1] = '\n'; - f.frametype = AST_FRAME_TEXT; - f.subclass = 0; - f.data = text2send; - f.datalen = strlen(text2send); - ast_queue_frame(o->owner, &f); - } - return RESULT_SUCCESS; -} - -static char answer_usage[] = -"Usage: answer\n" -" Answers an incoming call on the console (OSS) channel.\n"; - -static int console_hangup(int fd, int argc, char *argv[]) -{ - struct chan_oss_pvt *o = find_desc(oss_active); - - if (argc != 1) - return RESULT_SHOWUSAGE; - o->cursound = -1; - o->nosound = 0; - if (!o->owner && !o->hookstate) { /* XXX maybe only one ? */ - ast_cli(fd, "No call to hang up\n"); - return RESULT_FAILURE; - } - o->hookstate = 0; - if (o->owner) - ast_queue_hangup(o->owner); - setformat(o, O_CLOSE); - return RESULT_SUCCESS; -} - -static char hangup_usage[] = -"Usage: hangup\n" -" Hangs up any call currently placed on the console.\n"; - - -static int console_flash(int fd, int argc, char *argv[]) -{ - struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH }; - struct chan_oss_pvt *o = find_desc(oss_active); - - if (argc != 1) - return RESULT_SHOWUSAGE; - o->cursound = -1; - o->nosound = 0; /* when cursound is -1 nosound must be 0 */ - if (!o->owner) { /* XXX maybe !o->hookstate too ? */ - ast_cli(fd, "No call to flash\n"); - return RESULT_FAILURE; - } - o->hookstate = 0; - if (o->owner) /* XXX must be true, right ? */ - ast_queue_frame(o->owner, &f); - return RESULT_SUCCESS; -} - - -static char flash_usage[] = -"Usage: flash\n" -" Flashes the call currently placed on the console.\n"; - - - -static int console_dial(int fd, int argc, char *argv[]) -{ - char *s = NULL, *mye = NULL, *myc = NULL; - struct chan_oss_pvt *o = find_desc(oss_active); - - if (argc != 1 && argc != 2) - return RESULT_SHOWUSAGE; - if (o->owner) { /* already in a call */ - int i; - struct ast_frame f = { AST_FRAME_DTMF, 0 }; - - if (argc == 1) { /* argument is mandatory here */ - ast_cli(fd, "Already in a call. You can only dial digits until you hangup.\n"); - return RESULT_FAILURE; - } - s = argv[1]; - /* send the string one char at a time */ - for (i=0; i<strlen(s); i++) { - f.subclass = s[i]; - ast_queue_frame(o->owner, &f); - } - return RESULT_SUCCESS; - } - /* if we have an argument split it into extension and context */ - if (argc == 2) - s = ast_ext_ctx(argv[1], &mye, &myc); - /* supply default values if needed */ - if (mye == NULL) - mye = o->ext; - if (myc == NULL) - myc = o->ctx; - if (ast_exists_extension(NULL, myc, mye, 1, NULL)) { - o->hookstate = 1; - oss_new(o, mye, myc, AST_STATE_RINGING); - } else - ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc); - if (s) - free(s); - return RESULT_SUCCESS; -} - -static char dial_usage[] = -"Usage: dial [extension[@context]]\n" -" Dials a given extensison (and context if specified)\n"; - -static char mute_usage[] = -"Usage: mute\nMutes the microphone\n"; - -static char unmute_usage[] = -"Usage: unmute\nUnmutes the microphone\n"; - -static int console_mute(int fd, int argc, char *argv[]) -{ - struct chan_oss_pvt *o = find_desc(oss_active); - - if (argc != 1) - return RESULT_SHOWUSAGE; - o->mute = 1; - return RESULT_SUCCESS; -} - -static int console_unmute(int fd, int argc, char *argv[]) -{ - struct chan_oss_pvt *o = find_desc(oss_active); - - if (argc != 1) - return RESULT_SHOWUSAGE; - o->mute = 0; - return RESULT_SUCCESS; -} - -static int console_transfer(int fd, int argc, char *argv[]) -{ - struct chan_oss_pvt *o = find_desc(oss_active); - struct ast_channel *b = NULL; - char *tmp, *ext, *ctx; - - if (argc != 2) - return RESULT_SHOWUSAGE; - if (o == NULL) - return RESULT_FAILURE; - if (o->owner ==NULL || (b = ast_bridged_channel(o->owner)) == NULL) { - ast_cli(fd, "There is no call to transfer\n"); - return RESULT_SUCCESS; - } - - tmp = ast_ext_ctx(argv[1], &ext, &ctx); - if (ctx == NULL) /* supply default context if needed */ - ctx = o->owner->context; - if (!ast_exists_extension(b, ctx, ext, 1, b->cid.cid_num)) - ast_cli(fd, "No such extension exists\n"); - else { - ast_cli(fd, "Whee, transferring %s to %s@%s.\n", - b->name, ext, ctx); - if (ast_async_goto(b, ctx, ext, 1)) - ast_cli(fd, "Failed to transfer :(\n"); - } - if (tmp) - free(tmp); - return RESULT_SUCCESS; -} - -static char transfer_usage[] = -"Usage: transfer <extension>[@context]\n" -" Transfers the currently connected call to the given extension (and\n" -"context if specified)\n"; - -static char console_usage[] = -"Usage: console [device]\n" -" If used without a parameter, displays which device is the current\n" -"console. If a device is specified, the console sound device is changed to\n" -"the device specified.\n"; - -static int console_active(int fd, int argc, char *argv[]) -{ - if (argc == 1) - ast_cli(fd, "active console is [%s]\n", oss_active); - else if (argc != 2) - return RESULT_SHOWUSAGE; - else { - struct chan_oss_pvt *o; - if (strcmp(argv[1], "show") == 0) { - for (o = oss_default.next; o ; o = o->next) - ast_cli(fd, "device [%s] exists\n", o->name); - return RESULT_SUCCESS; - } - o = find_desc(argv[1]); - if (o == NULL) - ast_cli(fd, "No device [%s] exists\n", argv[1]); - else - oss_active = o->name; - } - return RESULT_SUCCESS; -} - -static struct ast_cli_entry myclis[] = { - { { "answer", NULL }, console_answer, "Answer an incoming console call", answer_usage }, - { { "hangup", NULL }, console_hangup, "Hangup a call on the console", hangup_usage }, - { { "flash", NULL }, console_flash, "Flash a call on the console", flash_usage }, - { { "dial", NULL }, console_dial, "Dial an extension on the console", dial_usage }, - { { "mute", NULL }, console_mute, "Disable mic input", mute_usage }, - { { "unmute", NULL }, console_unmute, "Enable mic input", unmute_usage }, - { { "transfer", NULL }, console_transfer, "Transfer a call to a different extension", transfer_usage }, - { { "send", "text", NULL }, console_sendtext, "Send text to the remote device", sendtext_usage }, - { { "autoanswer", NULL }, console_autoanswer, "Sets/displays autoanswer", autoanswer_usage, autoanswer_complete }, - { { "console", NULL }, console_active, "Sets/displays active console", console_usage }, -}; - -/* - * store the mixer argument from the config file, filtering possibly - * invalid or dangerous values (the string is used as argument for - * system("mixer %s") - */ -static void store_mixer(struct chan_oss_pvt *o, char *s) -{ - int i; - - for (i=0; i < strlen(s); i++) { - if (!isalnum(s[i]) && index(" \t-/", s[i]) == NULL) { - ast_log(LOG_WARNING, - "Suspect char %c in mixer cmd, ignoring:\n\t%s\n", s[i], s); - return; - } - } - if (o->mixer_cmd) - free(o->mixer_cmd); - o->mixer_cmd = strdup(s); - ast_log(LOG_WARNING, "setting mixer %s\n", s); -} - -/* - * grab fields from the config file, init the descriptor and open the device. - */ -static struct chan_oss_pvt * store_config(struct ast_config *cfg, char *ctg) -{ - struct ast_variable *v; - struct chan_oss_pvt *o; - - if (ctg == NULL) { - o = &oss_default; - ctg = "general"; - } else { - o = (struct chan_oss_pvt *)malloc(sizeof *o); - if (o == NULL) /* fail */ - return NULL; - *o = oss_default; - /* "general" is also the default thing */ - if (strcmp(ctg, "general") == 0) { - o->name = strdup("dsp"); - oss_active = o->name; - goto openit; - } - o->name = strdup(ctg); - } - - o->lastopen = ast_tvnow(); /* don't leave it 0 or tvdiff may wrap */ - /* fill other fields from configuration */ - for (v = ast_variable_browse(cfg, ctg);v; v=v->next) { - M_START(v->name, v->value); - - M_BOOL("autoanswer", o->autoanswer) - M_BOOL("autohangup", o->autohangup) - M_BOOL("overridecontext", o->overridecontext) - M_STR("device", o->device) - M_UINT("frags", o->frags) - M_UINT("debug", oss_debug) - M_UINT("queuesize", o->queuesize) - M_STR("context", o->ctx) - M_STR("language", o->language) - M_STR("extension", o->ext) - M_F("mixer", store_mixer(o, v->value)) - M_END(;); - } - if (ast_strlen_zero(o->device)) - ast_copy_string(o->device, DEV_DSP, sizeof(o->device)); - if (o->mixer_cmd) { - char *cmd; - - asprintf(&cmd, "mixer %s", o->mixer_cmd); - ast_log(LOG_WARNING, "running [%s]\n", cmd); - system(cmd); - free(cmd); - } - if (o == &oss_default) /* we are done with the default */ - return NULL; - -openit: -#if TRYOPEN - if (setformat(o, O_RDWR) < 0) { /* open device */ - if (option_verbose > 0) { - ast_verbose(VERBOSE_PREFIX_2 "Device %s not detected\n", ctg); - ast_verbose(VERBOSE_PREFIX_2 "Turn off OSS support by adding " - "'noload=chan_oss.so' in /etc/asterisk/modules.conf\n"); - } - goto error; - } - if (o->duplex != M_FULL) - ast_log(LOG_WARNING, "XXX I don't work right with non " - "full-duplex sound cards XXX\n"); -#endif /* TRYOPEN */ - if (pipe(o->sndcmd) != 0) { - ast_log(LOG_ERROR, "Unable to create pipe\n"); - goto error; - } - ast_pthread_create(&o->sthread, NULL, sound_thread, o); - /* link into list of devices */ - if (o != &oss_default) { - o->next = oss_default.next; - oss_default.next = o; - } - return o; - -error: - if (o != &oss_default) - free(o); - return NULL; -} - -int load_module(void) -{ - int i; - struct ast_config *cfg; - - /* load config file */ - cfg = ast_config_load(config); - if (cfg != NULL) { - char *ctg = NULL; /* first pass is 'general' */ - - do { - store_config(cfg, ctg); - } while ( (ctg = ast_category_browse(cfg, ctg)) != NULL); - ast_config_destroy(cfg); - } else { - ast_log(LOG_NOTICE, "Unable to load config oss.conf\n"); - return -1; - } - if (find_desc(oss_active) == NULL) { - ast_log(LOG_NOTICE, "Device %s not found\n", oss_active); - /* XXX we could default to 'dsp' perhaps ? */ - /* XXX should cleanup allocated memory etc. */ - return -1; - } - i = ast_channel_register(&oss_tech); - if (i < 0) { - ast_log(LOG_ERROR, "Unable to register channel class '%s'\n", - oss_default.type); - /* XXX should cleanup allocated memory etc. */ - return -1; - } - ast_cli_register_multiple(myclis, sizeof(myclis)/sizeof(struct ast_cli_entry)); - return 0; -} - - -int unload_module() -{ - struct chan_oss_pvt *o; - - ast_channel_unregister(&oss_tech); - ast_cli_unregister_multiple(myclis, - sizeof(myclis)/sizeof(struct ast_cli_entry)); - - for (o = oss_default.next; o ; o = o->next) { - close(o->sounddev); - if (o->sndcmd[0] > 0) { - close(o->sndcmd[0]); - close(o->sndcmd[1]); - } - if (o->owner) - ast_softhangup(o->owner, AST_SOFTHANGUP_APPUNLOAD); - if (o->owner) /* XXX how ??? */ - return -1; - /* XXX what about the thread ? */ - /* XXX what about the memory allocated ? */ - } - return 0; -} - -char *description() -{ - return (char *)oss_tech.description; -} - -int usecount() -{ - return usecnt; -} - -char *key() -{ - return ASTERISK_GPL_KEY; -} |