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Diffstat (limited to 'net/asterisk12/files/chan_oss.c')
| -rw-r--r-- | net/asterisk12/files/chan_oss.c | 1320 |
1 files changed, 1320 insertions, 0 deletions
diff --git a/net/asterisk12/files/chan_oss.c b/net/asterisk12/files/chan_oss.c new file mode 100644 index 000000000000..aef0db9dca85 --- /dev/null +++ b/net/asterisk12/files/chan_oss.c @@ -0,0 +1,1320 @@ +/* + * Asterisk -- A telephony toolkit for Linux. + * + * Copyright (C) 1999, Mark Spencer + * + * Mark Spencer <markster@linux-support.net> + * + * This program is free software, distributed under the terms of + * the GNU General Public License + * + * FreeBSD changes and multiple device support by Luigi Rizzo, 2005.04.26 + * note-this code best seen with ts=8 (8-spaces tabs) in the editor + */ + +#include <asterisk/lock.h> +#include <asterisk/frame.h> +#include <asterisk/logger.h> +#include <asterisk/channel.h> +#include <asterisk/module.h> +#include <asterisk/channel_pvt.h> +#include <asterisk/options.h> +#include <asterisk/pbx.h> +#include <asterisk/config.h> +#include <asterisk/cli.h> +#include <asterisk/utils.h> +#include <unistd.h> +#include <fcntl.h> +#include <errno.h> +#include <sys/ioctl.h> +#include <sys/time.h> +#include <string.h> +#include <stdlib.h> +#include <stdio.h> +#include <ctype.h> /* for isalnum */ +#ifdef __linux +#include <linux/soundcard.h> +#elif defined(__FreeBSD__) +#include <sys/soundcard.h> +#else +#include <soundcard.h> +#endif +#include "busy.h" +#include "ringtone.h" +#include "ring10.h" +#include "answer.h" + +/* Which device to use */ +#if defined( __OpenBSD__ ) || defined( __NetBSD__ ) +#define DEV_DSP "/dev/audio" +#else +#define DEV_DSP "/dev/dsp" +#endif + +/* + * Basic mode of operation: + * + * we have one keyboard (which receives commands from the keyboard) + * and multiple headset's connected to audio cards. Headsets are named as + * the sections of oss.conf + * + * At any time, the keyboard is attached to one headset, and you + * can switch among them using the 'console' command. + * + * The following parameters are important for the configuration of + * the device: + * + * 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 + + +/* Don't switch between read/write modes faster than every 300 ms */ +#define MIN_SWITCH_TIME 300 + + +static int usecnt; +AST_MUTEX_DEFINE_STATIC(usecnt_lock); + +static char *desc = "OSS Console Channel Driver"; +static char *tdesc = "OSS Console Channel Driver"; +static char *config = "oss.conf"; /* default config file */ + + +/* + * 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; + 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; + struct timeval lasttime; /* last setformat */ + 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 */ + + int silencesuppression; + int silencethreshold; + char device[64]; /* device to open */ + + pthread_t sthread; + + struct ast_channel *owner; + char ext[AST_MAX_EXTENSION]; + char ctx[AST_MAX_EXTENSION]; + 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, + .silencethreshold = 1000, /* currently unused */ + .ext = "s", + .ctx = "default", + .readpos = AST_FRIENDLY_OFFSET, /* start here on reads */ +}; + +static char *oss_active; /* the active device */ + +/* + * returns true if too early to switch + */ +static int too_early(struct chan_oss_pvt *o) +{ + struct timeval tv; + int ms; + gettimeofday(&tv, NULL); + ms = (tv.tv_sec - o->lasttime.tv_sec) * 1000 + + (tv.tv_usec - o->lasttime.tv_usec) / 1000; + if (ms < MIN_SWITCH_TIME) + return -1; + return 0; +} + +/* + * 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; +} + +static int soundcard_writeframe(struct chan_oss_pvt *o, short *data) +{ + /* Write an exactly FRAME_SIZE sized frame */ + int res; + + /* + * 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 && 0) + ast_log(LOG_WARNING, "write: used %d blocks (%d)\n", + res, o->w_errors); + return 0; + } + o->w_errors = 0; + res = write(o->sounddev, ((void *)data), FRAME_SIZE * 2); + return res; +} + +/* + * 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; /* sound available */ + 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 { /* no sound, maybe some silence */ + static 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; + + /* kick the driver by trying to read from it. Ignore errors */ + if (read(o->sounddev, ign, sizeof(ign)) < 0) + ast_log(LOG_WARNING, "Read error on sound device: %s\n", + strerror(errno)); + for(;;) { + fd_set rfds, wfds; + int maxfd, res; + + FD_ZERO(&rfds); + FD_ZERO(&wfds); + maxfd = o->sndcmd[0]; /* pipe from the main process */ + FD_SET(o->sndcmd[0], &rfds); + if (!o->owner) { /* no one owns the audio, so we must drain it */ + FD_SET(o->sounddev, &rfds); + if (o->sounddev > maxfd) + maxfd = o->sounddev; + } + if (o->cursound > -1) { + FD_SET(o->sounddev, &wfds); + if (o->sounddev > maxfd) + maxfd = o->sounddev; + } + /* 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)); + 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 (FD_ISSET(o->sounddev, &rfds)) { /* read and ignore errors */ + read(o->sounddev, ign, sizeof(ign)); + } + if (FD_ISSET(o->sounddev, &wfds)) + send_sound(o); + } + /* Never reached */ + return NULL; +} + +#if 0 +static int calc_loudness(short *frame) +{ + int sum = 0; + int x; + for (x=0;x<FRAME_SIZE;x++) { + if (frame[x] < 0) + sum -= frame[x]; + else + sum += frame[x]; + } + sum = sum/FRAME_SIZE; + return sum; +} + +static int silence_suppress(short *buf) +{ +#define SILBUF 3 + int loudness; + static int silentframes = 0; + static char silbuf[FRAME_SIZE * 2 * SILBUF]; + static int silbufcnt=0; + if (!oss.silencesuppression) + return 0; + loudness = calc_loudness((short *)(buf)); + if (option_debug) + ast_log(LOG_DEBUG, "loudness is %d\n", loudness); + if (loudness < silencethreshold) { + silentframes++; + silbufcnt++; + /* Keep track of the last few bits of silence so we can play + them as lead-in when the time is right */ + if (silbufcnt >= SILBUF) { + /* Make way for more buffer */ + memmove(silbuf, silbuf + FRAME_SIZE * 2, FRAME_SIZE * 2 * (SILBUF - 1)); + silbufcnt--; + } + memcpy(silbuf + FRAME_SIZE * 2 * silbufcnt, buf, FRAME_SIZE * 2); + if (silentframes > 10) { + /* We've had plenty of silence, so compress it now */ + return 1; + } + } else { + silentframes=0; + /* Write any buffered silence we have, it may have something + important */ + if (silbufcnt) { + write(oss.sounddev, silbuf, silbufcnt * FRAME_SIZE); + silbufcnt = 0; + } + } + return 0; +} +#endif + +/* + * 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; + } + fd = o->sounddev = open(o->device, mode |O_NONBLOCK); + if (o->sounddev < 0) { + ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", + strerror(errno)); + return -1; + } + + gettimeofday(&o->lasttime, NULL); + 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"); + 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; + } + /* 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"); + 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; + } + } + } + /* XXX 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; +} + +/* + * make sure output mode is available. Returns 0 if done, + * 1 if too early to switch, -1 if error + */ +static int soundcard_setoutput(struct chan_oss_pvt *o, int force) +{ + if (o->duplex == M_FULL || (o->duplex == M_WRITE && !force)) + return 0; + if (!force && too_early(o)) + return 1; + if (setformat(o, O_WRONLY)) + return -1; + return 0; +} + +/* + * make sure input mode is available. Returns 0 if done + * 1 if too early to switch, -1 if error + */ +static int soundcard_setinput(struct chan_oss_pvt *o, int force) +{ + if (o->duplex == M_FULL || (o->duplex == M_READ && !force)) + return 0; + if (!force && too_early(o)) + return 1; + if (setformat(o, O_RDONLY)) + return -1; + return 0; +} + +static int oss_digit(struct ast_channel *c, char digit) +{ + ast_verbose( " << Console Received digit %c >> \n", digit); + return 0; +} + +static int oss_text(struct ast_channel *c, char *text) +{ + ast_verbose( " << Console Received text %s >> \n", text); + return 0; +} + +/* request to play a sound on the speaker XXX fix oss. */ +#define RING(o, x) { int what = x; write((o)->sndcmd[1], &what, sizeof(what)); } + +static int oss_call(struct ast_channel *c, char *dest, int timeout) +{ + struct chan_oss_pvt *o = c->pvt->pvt; + struct ast_frame f = { 0, }; + + ast_verbose( " << Call placed to '%s' on console >> \n", dest); + 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; +} + +static void answer_sound(struct chan_oss_pvt *o) +{ + RING(o, AST_CONTROL_ANSWER); +} + +static int oss_answer(struct ast_channel *c) +{ + struct chan_oss_pvt *o = c->pvt->pvt; + + ast_verbose( " << Console call has been answered >> \n"); + answer_sound(o); /* XXX do we really need it ? considering we shut down immediately... */ + 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->pvt->pvt; + + o->cursound = -1; + c->pvt->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; + } 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 res; + int src; + struct chan_oss_pvt *o = c->pvt->pvt; + + /* Immediately return if no sound is enabled */ + if (o->nosound) + return 0; + /* Stop any currently playing sound */ + o->cursound = -1; + if (o->duplex != M_FULL) { + /* XXX check this, looks weird! */ + /* If we're half duplex, we have to switch to read mode + to honor immediate needs if necessary */ + res = soundcard_setinput(o, 1); /* force set if not full_duplex */ + if (res < 0) { + ast_log(LOG_WARNING, "Unable to set device to input mode\n"); + return -1; + } + return 0; + } + res = soundcard_setoutput(o, 0); + if (res < 0) { + ast_log(LOG_WARNING, "Unable to set output device\n"); + return -1; + } else if (res > 0) { + /* The device is still in read mode, and it's too soon to change it, + so just pretend we wrote it */ + return 0; + } + /* + * we could receive a sample which is not a multiple of our FRAME_SIZE, + * so we buffer it locally and write to the device in FRAME_SIZE + * chunks, keeping 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) +{ + /* XXX if we want multiple devices, should move these static vars + * into the device descriptor + */ + int res; + struct chan_oss_pvt *o = c->pvt->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 = soundcard_setinput(o, 0); + if (res < 0) { + ast_log(LOG_WARNING, "Unable to set input mode\n"); + return NULL; + } else if (res > 0) { /* too early to switch ? */ + /* Theoretically shouldn't happen, but anyway, return a NULL frame */ + return f; + } + + 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; + + 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->pvt->pvt; + o->owner = newchan; + return 0; +} + +static int oss_indicate(struct ast_channel *c, int cond) +{ + struct chan_oss_pvt *o = c->pvt->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; + return 0; + 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; +} + +static struct ast_channel *oss_new(struct chan_oss_pvt *o, + char *ext, char *ctx, int state) +{ + struct ast_channel *c; + struct ast_channel_pvt *pvt; + + c = ast_channel_alloc(1); + if (c == NULL) + return NULL; + snprintf(c->name, sizeof(c->name), "OSS/%s", o->device + 5); + c->type = o->type; + c->fds[0] = o->sounddev; + c->nativeformats = AST_FORMAT_SLINEAR; + pvt = c->pvt; + pvt->pvt = o; + + /* relevant callbacks */ + pvt->send_digit = oss_digit; + pvt->send_text = oss_text; + pvt->hangup = oss_hangup; + pvt->answer = oss_answer; + pvt->read = oss_read; + pvt->call = oss_call; + pvt->write = oss_write; + pvt->indicate = oss_indicate; + pvt->fixup = oss_fixup; + + if (strlen(ctx)) + strncpy(c->context, ctx, sizeof(o->ctx)-1); + if (strlen(ext)) + strncpy(c->exten, ext, sizeof(o->ext)-1); + if (strlen(o->language)) + strncpy(c->language, o->language, sizeof(o->language)-1); + 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; +} + +/* + * 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, "%s could not find <%s>\n", __func__, dev); + return o; +} + +static struct ast_channel *oss_request(char *type, int format, void *data) +{ + 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 on the OSS channel\n"); + 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) && (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 (argc == 1) { + ast_cli(fd, "Auto answer is %s.\n", o->autoanswer ? "on" : "off"); + return RESULT_SUCCESS; + } + 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) +{ +#ifndef MIN +#define MIN(a,b) ((a) < (b) ? (a) : (b)) +#endif + 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"; + +static int console_answer(int fd, int argc, char *argv[]) +{ + struct chan_oss_pvt *o = find_desc(oss_active); + + struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_ANSWER }; + 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; + ast_queue_frame(o->owner, &f); + answer_sound(o); + return RESULT_SUCCESS; +} + +static char sendtext_usage[] = +"Usage: send text <message>\n" +" Sends a text message for display on the remote terminal.\n"; + +static int console_sendtext(int fd, int argc, char *argv[]) +{ + struct chan_oss_pvt *o = find_desc(oss_active); + int tmparg = 2; + char text2send[256] = ""; + struct ast_frame f = { 0, }; + + if (argc < 2) + return RESULT_SHOWUSAGE; + if (!o->owner) { + ast_cli(fd, "No one is calling us\n"); + return RESULT_FAILURE; + } + if (strlen(text2send)) + ast_cli(fd, "Warning: message already waiting to be sent, overwriting\n"); + text2send[0] = '\0'; + while(tmparg < argc) { + strncat(text2send, argv[tmparg++], sizeof(text2send) - strlen(text2send) - 1); + strncat(text2send, " ", sizeof(text2send) - strlen(text2send) - 1); + } + if (strlen(text2send)) { + 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; + if (!o->owner && !o->hookstate) { + ast_cli(fd, "No call to hangup up\n"); + return RESULT_FAILURE; + } + o->hookstate = 0; + if (o->owner) { + ast_queue_hangup(o->owner); + } + return RESULT_SUCCESS; +} + +static char hangup_usage[] = +"Usage: hangup\n" +" Hangs up any call currently placed on the console.\n"; + + +static int console_dial(int fd, int argc, char *argv[]) +{ + char *tmp = NULL, *mye = NULL, *myc = NULL; + int i; + struct ast_frame f = { AST_FRAME_DTMF, 0 }; + struct chan_oss_pvt *o = find_desc(oss_active); + + if ((argc != 1) && (argc != 2)) + return RESULT_SHOWUSAGE; + if (o->owner) { /* already in a call */ + 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; + } + mye = argv[1]; + /* send the string one char at a time */ + for (i=0; i<strlen(mye); i++) { + f.subclass = mye[i]; + ast_queue_frame(o->owner, &f); + } + return RESULT_SUCCESS; + } + /* if we have an argument split it into extension and context */ + if (argc == 2) { + tmp = myc = strdup(argv[1]); /* make a writable copy */ + mye = strsep(&myc, "@"); /* set exten, advance to context */ + myc = strsep(&myc, "@"); /* set context */ + } + /* 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); + return RESULT_SUCCESS; +} + +static char dial_usage[] = +"Usage: dial [extension[@context]]\n" +" Dials a given extensison (and context if specified)\n"; + +static int console_transfer(int fd, int argc, char *argv[]) +{ + struct chan_oss_pvt *o = find_desc(oss_active); + struct ast_channel *b; + + char *ext, *ctx; + + if (argc != 2) + return RESULT_SHOWUSAGE; + if (o == NULL) + return RESULT_FAILURE; + if (! (o->owner && o->owner->bridge)) { + ast_cli(fd, "There is no call to transfer\n"); + return RESULT_SUCCESS; + } + b = o->owner->bridge; + + ext = ctx = strdup(argv[1]); /* make a writable copy */ + strsep(&ctx, "@"); /* set exten, advance to context */ + ctx = strsep(&ctx, "@"); /* strip trailing @ and the rest */ + + if (ctx == NULL) /* supply default context if needed */ + ctx = o->owner->context; + if (!ast_exists_extension(b, ctx, ext, 1, b->callerid)) { + 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"); + } + free(ext); + 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 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 }, + { { "dial", NULL }, console_dial, "Dial an extension on the console", dial_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 foo sets foo as the console"} +}; + +/* + * 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; + o->next = NULL; /* XXX needed ? */ + 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); + } + ast_log(LOG_WARNING, "found category [%s]\n", ctg); + + /* fill other fields from configuration */ + v = ast_variable_browse(cfg, ctg); + while(v) { + if (!strcasecmp(v->name, "autoanswer")) + o->autoanswer = ast_true(v->value); + else if (!strcasecmp(v->name, "autohangup")) + o->autohangup = ast_true(v->value); + else if (!strcasecmp(v->name, "silencesuppression")) + o->silencesuppression = ast_true(v->value); + else if (!strcasecmp(v->name, "silencethreshold")) + o->silencethreshold = atoi(v->value); + else if (!strcasecmp(v->name, "device")) + strncpy(o->device, v->value, sizeof(o->device)-1); + else if (!strcasecmp(v->name, "frags")) + o->frags = strtoul(v->value, NULL, 0); + else if (!strcasecmp(v->name, "queuesize")) + o->queuesize = strtoul(v->value, NULL, 0); + else if (!strcasecmp(v->name, "context")) + strncpy(o->ctx, v->value, sizeof(o->ctx)-1); + else if (!strcasecmp(v->name, "language")) + strncpy(o->language, v->value, sizeof(o->language)-1); + else if (!strcasecmp(v->name, "extension")) + strncpy(o->ext, v->value, sizeof(o->ext)-1); + else if (!strcasecmp(v->name, "mixer")) + store_mixer(o, v->value); + v=v->next; + } + if (!strlen(o->device)) + strncpy(o->device, DEV_DSP, sizeof(o->device)-1); + 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 (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; + } + soundcard_setinput(o, 1); /* force set if not full_duplex */ + if (o->duplex != M_FULL) + ast_log(LOG_WARNING, "XXX I don't work right with non " + "full-duplex sound cards XXX\n"); + 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() +{ + int i; + struct ast_config *cfg; + + /* load config file */ + cfg = ast_load(config); + if (cfg != NULL) { + char *ctg; + + store_config(cfg, NULL); /* init general category */ + ctg = ast_category_browse(cfg, NULL); /* initial category */ + while (ctg != NULL) { + store_config(cfg, ctg); + ctg = ast_category_browse(cfg, ctg); + } + ast_destroy(cfg); + } + i = ast_channel_register(oss_default.type, tdesc, + AST_FORMAT_SLINEAR, oss_request); + if (i < 0) { + ast_log(LOG_ERROR, "Unable to register channel class '%s'\n", + oss_default.type); + return NULL; + } + for (i=0; i<sizeof(myclis)/sizeof(struct ast_cli_entry); i++) + ast_cli_register(myclis + i); + return 0; +} + + +int unload_module() +{ + int x; + struct chan_oss_pvt *o; + + for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++) + ast_cli_unregister(myclis + x); + + 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 desc; +} + +int usecount() /* XXX is this per-device or global for the module ? */ +{ + int res; + ast_mutex_lock(&usecnt_lock); + res = usecnt; + ast_mutex_unlock(&usecnt_lock); + return res; +} + +char *key() +{ + return ASTERISK_GPL_KEY; +} |