--- chan_oss.c.orig Thu Apr 21 06:30:23 2005 +++ chan_oss.c Mon May 2 22:53:19 2005 @@ -1,18 +1,15 @@ /* * Asterisk -- A telephony toolkit for Linux. * - * Use /dev/dsp as a channel, and the console to command it :). - * - * The full-duplex "simulation" is pretty weak. This is generally a - * VERY BADLY WRITTEN DRIVER so please don't use it as a model for - * writing a driver. - * * Copyright (C) 1999 - 2005, Digium, Inc. * * Mark Spencer * * This program is free software, distributed under the terms of * the GNU General Public License + * + * FreeBSD changes and multiple device support by Luigi Rizzo, 2005.05.02 + * note-this code best seen with ts=8 (8-spaces tabs) in the editor */ #include "asterisk/lock.h" @@ -23,6 +20,28 @@ #include "asterisk/options.h" #include "asterisk/pbx.h" #include "asterisk/config.h" + +/* + * 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, WITHOUT ; on each line. + * + * { + * 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) + */ +#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, strncpy(dst, __val, sizeof(dst) - 1) ) + #include "asterisk/cli.h" #include "asterisk/utils.h" #include "asterisk/causes.h" @@ -35,6 +54,7 @@ #include #include #include +#include /* for isalnum */ #ifdef __linux #include @@ -55,42 +75,75 @@ #define DEV_DSP "/dev/dsp" #endif -/* Lets use 160 sample frames, just like GSM. */ +/* + * 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 -/* When you set the frame size, you have to come up with - the right buffer format as well. */ -/* 5 64-byte frames = one frame */ -#define BUFFER_FMT ((buffersize * 10) << 16) | (0x0006); /* Don't switch between read/write modes faster than every 300 ms */ -#define MIN_SWITCH_TIME 600 +#define MIN_SWITCH_TIME 300 -static struct timeval lasttime; static int usecnt; -static int silencesuppression = 0; -static int silencethreshold = 1000; -static int playbackonly = 0; - - AST_MUTEX_DEFINE_STATIC(usecnt_lock); -static const char type[] = "Console"; -static const char desc[] = "OSS Console Channel Driver"; -static const char tdesc[] = "OSS Console Channel Driver"; -static const char config[] = "oss.conf"; - -static char context[AST_MAX_EXTENSION] = "default"; -static char language[MAX_LANGUAGE] = ""; -static char exten[AST_MAX_EXTENSION] = "s"; +static char *config = "oss.conf"; /* default config file */ -static int hookstate=0; - -static short silence[FRAME_SIZE] = {0, }; +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; @@ -99,25 +152,98 @@ }; static struct sound sounds[] = { - { AST_CONTROL_RINGING, ringtone, sizeof(ringtone)/2, 16000, 32000, 1 }, - { AST_CONTROL_BUSY, busy, sizeof(busy)/2, 4000, 4000, 1 }, - { AST_CONTROL_CONGESTION, busy, sizeof(busy)/2, 2000, 2000, 1 }, - { AST_CONTROL_RING, ring10, sizeof(ring10)/2, 16000, 32000, 1 }, - { AST_CONTROL_ANSWER, answer, sizeof(answer)/2, 2200, 0, 0 }, + { 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 */ }; -/* Sound command pipe */ -static int sndcmd[2]; -static struct chan_oss_pvt { - /* We only have one OSS structure -- near sighted perhaps, but it - keeps this driver as simple as possible -- as it should be. */ +/* + * 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; + 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; + int playbackonly; + char device[64]; /* device to open */ + + pthread_t sthread; + struct ast_channel *owner; - char exten[AST_MAX_EXTENSION]; - char context[AST_MAX_EXTENSION]; -} oss; + 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 */ -static struct ast_channel *oss_request(const char *type, int format, void *data, int *cause); +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); @@ -129,8 +255,8 @@ static int oss_fixup(struct ast_channel *oldchan, struct ast_channel *newchan); static const struct ast_channel_tech oss_tech = { - .type = type, - .description = tdesc, + .type = "Console", + .description = "OSS Console Channel Driver", .capabilities = AST_FORMAT_SLINEAR, .requester = oss_request, .send_digit = oss_digit, @@ -144,169 +270,202 @@ .fixup = oss_fixup, }; -static int time_has_passed(void) +/* + * 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 - lasttime.tv_sec) * 1000 + - (tv.tv_usec - lasttime.tv_usec) / 1000; - if (ms > MIN_SWITCH_TIME) + 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; } -/* Number of buffers... Each is FRAMESIZE/8 ms long. For example - with 160 sample frames, and a buffer size of 3, we have a 60ms buffer, - usually plenty. */ - -static pthread_t sthread; - -#define MAX_BUFFER_SIZE 100 -static int buffersize = 3; - -static int full_duplex = 0; - -/* Are we reading or writing (simulated full duplex) */ -static int readmode = 1; - -/* File descriptor for sound device */ -static int sounddev = -1; +/* + * 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; -static int autoanswer = 1; + 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; +} -#if 0 -static int calc_loudness(short *frame) +static int soundcard_writeframe(struct chan_oss_pvt *o, short *data) { - int sum = 0; - int x; - for (x=0;x 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; } - sum = sum/FRAME_SIZE; - return sum; + o->w_errors = 0; + res = write(o->sounddev, ((void *)data), FRAME_SIZE * 2); + return res; } -#endif -static int cursound = -1; -static int sampsent = 0; -static int silencelen=0; -static int offset=0; -static int nosound=0; - -static int send_sound(void) +/* + * 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 total = FRAME_SIZE; - short *frame = NULL; - int amt=0; - int res; - int myoff; - audio_buf_info abi; - if (cursound > -1) { - res = ioctl(sounddev, SNDCTL_DSP_GETOSPACE ,&abi); - if (res) { - ast_log(LOG_WARNING, "Unable to read output space\n"); - return -1; - } - /* Calculate how many samples we can send, max */ - if (total > (abi.fragments * abi.fragsize / 2)) - total = abi.fragments * abi.fragsize / 2; - res = total; - if (sampsent < sounds[cursound].samplen) { - myoff=0; - while(total) { - amt = total; - if (amt > (sounds[cursound].datalen - offset)) - amt = sounds[cursound].datalen - offset; - memcpy(myframe + myoff, sounds[cursound].data + offset, amt * 2); - total -= amt; - offset += amt; - sampsent += amt; - myoff += amt; - if (offset >= sounds[cursound].datalen) - offset = 0; - } - /* Set it up for silence */ - if (sampsent >= sounds[cursound].samplen) - silencelen = sounds[cursound].silencelen; - frame = myframe; - } else { - if (silencelen > 0) { - frame = silence; - silencelen -= res; - } else { - if (sounds[cursound].repeat) { - /* Start over */ - sampsent = 0; - offset = 0; - } else { - cursound = -1; - nosound = 0; + 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; } } - if (frame) - res = write(sounddev, frame, res * 2); - if (res > 0) - return 0; - return res; } - return 0; + l = soundcard_writeframe(o, myframe); + if (l > 0) + o->sampsent = l_sampsent; /* update status */ } -static void *sound_thread(void *unused) +static void *sound_thread(void *arg) { - fd_set rfds; - fd_set wfds; - int max; - int res; char ign[4096]; - if (read(sounddev, ign, sizeof(sounddev)) < 0) - ast_log(LOG_WARNING, "Read error on sound device: %s\n", strerror(errno)); + 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); - max = sndcmd[0]; - FD_SET(sndcmd[0], &rfds); - if (!oss.owner) { - FD_SET(sounddev, &rfds); - if (sounddev > max) - max = sounddev; - } - if (cursound > -1) { - FD_SET(sounddev, &wfds); - if (sounddev > max) - max = sounddev; + 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; } - res = ast_select(max + 1, &rfds, &wfds, NULL, NULL); + /* 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)); + ast_log(LOG_WARNING, "select failed: %s\n", + strerror(errno)); continue; } - if (FD_ISSET(sndcmd[0], &rfds)) { - read(sndcmd[0], &cursound, sizeof(cursound)); - silencelen = 0; - offset = 0; - sampsent = 0; - } - if (FD_ISSET(sounddev, &rfds)) { - /* Ignore read */ - if (read(sounddev, ign, sizeof(ign)) < 0) - ast_log(LOG_WARNING, "Read error on sound device: %s\n", strerror(errno)); - } - if (FD_ISSET(sounddev, &wfds)) - if (send_sound()) - ast_log(LOG_WARNING, "Failed to write sound\n"); + 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;xsounddev >= 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); #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 ((fmt & DSP_CAP_DUPLEX) && !res) { + if (res == 0 && (fmt & DSP_CAP_DUPLEX)) { if (option_verbose > 1) ast_verbose(VERBOSE_PREFIX_2 "Console is full duplex\n"); - full_duplex = -1; + 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) { @@ -384,98 +567,66 @@ 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 (!warnedalready++) + 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; + } } -#if 1 - fmt = BUFFER_FMT; + /* + * 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 (!warnedalready2++) + if (!(o->warned & WARN_frag)) { ast_log(LOG_WARNING, "Unable to set fragment size -- sound may be choppy\n"); + o->warned |= WARN_frag; } -#endif - return 0; } - -static int soundcard_setoutput(int force) -{ - /* Make sure the soundcard is in output mode. */ - int fd = sounddev; - if (full_duplex || (!readmode && !force)) - return 0; - readmode = 0; - if (force || time_has_passed()) { - ioctl(sounddev, SNDCTL_DSP_RESET, 0); - /* Keep the same fd reserved by closing the sound device and copying stdin at the same - time. */ - /* dup2(0, sound); */ - close(sounddev); - fd = open(DEV_DSP, O_WRONLY |O_NONBLOCK); - if (fd < 0) { - ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno)); - return -1; - } - /* dup2 will close the original and make fd be sound */ - if (dup2(fd, sounddev) < 0) { - ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno)); - return -1; - } - if (setformat()) { - return -1; } + /* 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; } - return 1; -} -static int soundcard_setinput(int force) +/* + * 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) { - int fd = sounddev; - if (full_duplex || (readmode && !force)) + if (o->duplex == M_FULL || (o->duplex == M_WRITE && !force)) return 0; - readmode = -1; - if (force || time_has_passed()) { - ioctl(sounddev, SNDCTL_DSP_RESET, 0); - close(sounddev); - /* dup2(0, sound); */ - fd = open(DEV_DSP, O_RDONLY | O_NONBLOCK); - if (fd < 0) { - ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno)); - return -1; - } - /* dup2 will close the original and make fd be sound */ - if (dup2(fd, sounddev) < 0) { - ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno)); - return -1; - } - if (setformat()) { + if (!force && too_early(o)) + return 1; + if (setformat(o, O_WRONLY)) return -1; - } return 0; } - return 1; -} -static int soundcard_init(void) +/* + * 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) { - /* Assume it's full duplex for starters */ - int fd = open(DEV_DSP, O_RDWR | O_NONBLOCK); - if (fd < 0) { - ast_log(LOG_WARNING, "Unable to open %s: %s\n", DEV_DSP, strerror(errno)); - return fd; - } - gettimeofday(&lasttime, NULL); - sounddev = fd; - setformat(); - if (!full_duplex) - soundcard_setinput(1); - return sounddev; + 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) @@ -490,129 +641,94 @@ 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) { - int res = 3; + struct chan_oss_pvt *o = c->tech_pvt; struct ast_frame f = { 0, }; + ast_verbose( " << Call placed to '%s' on console >> \n", dest); - if (autoanswer) { + if (o->autoanswer) { ast_verbose( " << Auto-answered >> \n" ); f.frametype = AST_FRAME_CONTROL; f.subclass = AST_CONTROL_ANSWER; ast_queue_frame(c, &f); } else { - nosound = 1; 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); - write(sndcmd[1], &res, sizeof(res)); + RING(o, AST_CONTROL_RING); } return 0; } -static void answer_sound(void) +static void answer_sound(struct chan_oss_pvt *o) { - int res; - nosound = 1; - res = 4; - write(sndcmd[1], &res, sizeof(res)); - + RING(o, AST_CONTROL_ANSWER); } static int oss_answer(struct ast_channel *c) { + struct chan_oss_pvt *o = c->tech_pvt; + ast_verbose( " << Console call has been answered >> \n"); - answer_sound(); + answer_sound(o); /* XXX do we really need it ? considering we shut down immediately... */ ast_setstate(c, AST_STATE_UP); - cursound = -1; - nosound=0; + o->cursound = -1; + o->nosound=0; return 0; } static int oss_hangup(struct ast_channel *c) { - int res = 0; - cursound = -1; + struct chan_oss_pvt *o = c->tech_pvt; + + o->cursound = -1; c->tech_pvt = NULL; - oss.owner = NULL; + o->owner = NULL; ast_verbose( " << Hangup on console >> \n"); - ast_mutex_lock(&usecnt_lock); + ast_mutex_lock(&usecnt_lock); /* XXX not sure why */ usecnt--; ast_mutex_unlock(&usecnt_lock); - if (hookstate) { - if (autoanswer) { + if (o->hookstate) { + if (o->autoanswer || o->autohangup) { /* Assume auto-hangup too */ - hookstate = 0; + o->hookstate = 0; } else { /* Make congestion noise */ - res = 2; - write(sndcmd[1], &res, sizeof(res)); + RING(o, AST_CONTROL_CONGESTION); } } return 0; } -static int soundcard_writeframe(short *data) +/* used for data coming from the network */ +static int oss_write(struct ast_channel *c, struct ast_frame *f) { - /* Write an exactly FRAME_SIZE sized of frame */ - static int bufcnt = 0; - static short buffer[FRAME_SIZE * MAX_BUFFER_SIZE * 5]; - struct audio_buf_info info; int res; - int fd = sounddev; - static int warned=0; - if (ioctl(fd, SNDCTL_DSP_GETOSPACE, &info)) { - if (!warned) - ast_log(LOG_WARNING, "Error reading output space\n"); - bufcnt = buffersize; - warned++; - } - if ((info.fragments >= buffersize * 5) && (bufcnt == buffersize)) { - /* We've run out of stuff, buffer again */ - bufcnt = 0; - } - if (bufcnt == buffersize) { - /* Write sample immediately */ - res = write(fd, ((void *)data), FRAME_SIZE * 2); - } else { - /* Copy the data into our buffer */ - res = FRAME_SIZE * 2; - memcpy(buffer + (bufcnt * FRAME_SIZE), data, FRAME_SIZE * 2); - bufcnt++; - if (bufcnt == buffersize) { - res = write(fd, ((void *)buffer), FRAME_SIZE * 2 * buffersize); - } - } - return res; -} - + int src; + struct chan_oss_pvt *o = c->tech_pvt; -static int oss_write(struct ast_channel *chan, struct ast_frame *f) -{ - int res; - static char sizbuf[8000]; - static int sizpos = 0; - int len = sizpos; - int pos; /* Immediately return if no sound is enabled */ - if (nosound) + if (o->nosound) return 0; /* Stop any currently playing sound */ - cursound = -1; - if (!full_duplex && !playbackonly) { + o->cursound = -1; + if (o->duplex != M_FULL && !o->playbackonly) { + /* XXX check this, looks weird! */ /* If we're half duplex, we have to switch to read mode - to honor immediate needs if necessary. But if we are in play - back only mode, then we don't switch because the console - is only being used one way -- just to playback something. */ - res = soundcard_setinput(1); + 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(0); + res = soundcard_setoutput(o, 0); if (res < 0) { ast_log(LOG_WARNING, "Unable to set output device\n"); return -1; @@ -621,198 +737,213 @@ so just pretend we wrote it */ return 0; } - /* We have to digest the frame in 160-byte portions */ - if (f->datalen > sizeof(sizbuf) - sizpos) { - ast_log(LOG_WARNING, "Frame too large\n"); - return -1; + /* + * 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; } - memcpy(sizbuf + sizpos, f->data, f->datalen); - len += f->datalen; - pos = 0; - while(len - pos > FRAME_SIZE * 2) { - soundcard_writeframe((short *)(sizbuf + pos)); - pos += FRAME_SIZE * 2; } - if (len - pos) - memmove(sizbuf, sizbuf + pos, len - pos); - sizpos = len - pos; return 0; } -static struct ast_frame *oss_read(struct ast_channel *chan) +static struct ast_frame *oss_read(struct ast_channel *c) { - static struct ast_frame f; - static char buf[FRAME_SIZE * 2 + AST_FRIENDLY_OFFSET]; - static int readpos = 0; int res; + struct chan_oss_pvt *o = c->tech_pvt; + struct ast_frame *f = &o->read_f; -#if 0 - ast_log(LOG_DEBUG, "oss_read()\n"); -#endif + /* 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; - f.frametype = AST_FRAME_NULL; - f.subclass = 0; - f.samples = 0; - f.datalen = 0; - f.data = NULL; - f.offset = 0; - f.src = type; - f.mallocd = 0; - f.delivery.tv_sec = 0; - f.delivery.tv_usec = 0; - - res = soundcard_setinput(0); + res = soundcard_setinput(o, 0); if (res < 0) { ast_log(LOG_WARNING, "Unable to set input mode\n"); return NULL; - } - if (res > 0) { + } else if (res > 0) { /* too early to switch ? */ /* Theoretically shouldn't happen, but anyway, return a NULL frame */ - return &f; - } - res = read(sounddev, buf + AST_FRIENDLY_OFFSET + readpos, FRAME_SIZE * 2 - readpos); - if (res < 0) { - ast_log(LOG_WARNING, "Error reading from sound device (If you're running 'artsd' then kill it): %s\n", strerror(errno)); -#if 0 - CRASH; -#endif - return NULL; + return f; } - readpos += res; - if (readpos >= FRAME_SIZE * 2) { - /* A real frame */ - readpos = 0; - if (chan->_state != AST_STATE_UP) { - /* Don't transmit unless it's up */ - return &f; - } - f.frametype = AST_FRAME_VOICE; - f.subclass = AST_FORMAT_SLINEAR; - f.samples = FRAME_SIZE; - f.datalen = FRAME_SIZE * 2; - f.data = buf + AST_FRIENDLY_OFFSET; - f.offset = AST_FRIENDLY_OFFSET; - f.src = type; - f.mallocd = 0; - f.delivery.tv_sec = 0; - f.delivery.tv_usec = 0; -#if 0 - { static int fd = -1; - if (fd < 0) - fd = open("output.raw", O_RDWR | O_TRUNC | O_CREAT); - write(fd, f.data, f.datalen); - } -#endif - } - 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 *p = newchan->tech_pvt; - p->owner = newchan; + struct chan_oss_pvt *o = newchan->tech_pvt; + o->owner = newchan; return 0; } -static int oss_indicate(struct ast_channel *chan, int cond) +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: - res = 1; - break; case AST_CONTROL_CONGESTION: - res = 2; - break; case AST_CONTROL_RINGING: - res = 0; + res = cond; break; case -1: - cursound = -1; + o->cursound = -1; return 0; default: - ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, chan->name); + ast_log(LOG_WARNING, + "Don't know how to display condition %d on %s\n", + cond, c->name); return -1; } - if (res > -1) { - write(sndcmd[1], &res, sizeof(res)); - } + if (res > -1) + RING(o, res); return 0; } -static struct ast_channel *oss_new(struct chan_oss_pvt *p, int state) +static struct ast_channel *oss_new(struct chan_oss_pvt *o, + char *ext, char *ctx, int state) { - struct ast_channel *tmp; - tmp = ast_channel_alloc(1); - if (tmp) { - tmp->tech = &oss_tech; - snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", DEV_DSP + 5); - tmp->type = type; - tmp->fds[0] = sounddev; - tmp->nativeformats = AST_FORMAT_SLINEAR; - tmp->readformat = AST_FORMAT_SLINEAR; - tmp->writeformat = AST_FORMAT_SLINEAR; - tmp->tech_pvt = p; - if (strlen(p->context)) - strncpy(tmp->context, p->context, sizeof(tmp->context)-1); - if (strlen(p->exten)) - strncpy(tmp->exten, p->exten, sizeof(tmp->exten)-1); - if (strlen(language)) - strncpy(tmp->language, language, sizeof(tmp->language)-1); - p->owner = tmp; - ast_setstate(tmp, 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; + c->nativeformats = AST_FORMAT_SLINEAR; + c->readformat = AST_FORMAT_SLINEAR; + c->writeformat = AST_FORMAT_SLINEAR; + c->tech_pvt = o; + + 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(tmp)) { - ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name); - ast_hangup(tmp); - tmp = NULL; + 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; } - return tmp; + +/* + * 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(const char *type, int format, void *data, int *cause) +static struct ast_channel *oss_request(const char *type, + int format, void *data, int *cause) { - int oldformat = format; - struct ast_channel *tmp; - format &= AST_FORMAT_SLINEAR; - if (!format) { - ast_log(LOG_NOTICE, "Asked to get a channel of format '%d'\n", oldformat); + 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 (oss.owner) { + if (o->owner) { ast_log(LOG_NOTICE, "Already have a call on the OSS channel\n"); *cause = AST_CAUSE_BUSY; return NULL; } - tmp= oss_new(&oss, AST_STATE_DOWN); - if (!tmp) { + 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 tmp; + 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", autoanswer ? "on" : "off"); + ast_cli(fd, "Auto answer is %s.\n", o->autoanswer ? "on" : "off"); return RESULT_SUCCESS; - } else { + } if (!strcasecmp(argv[1], "on")) - autoanswer = -1; + o->autoanswer = -1; else if (!strcasecmp(argv[1], "off")) - autoanswer = 0; + o->autoanswer = 0; else return RESULT_SHOWUSAGE; - } return RESULT_SUCCESS; } @@ -821,12 +952,14 @@ #ifndef MIN #define MIN(a,b) ((a) < (b) ? (a) : (b)) #endif + int l = strlen(word); + switch(state) { case 0: - if (strlen(word) && !strncasecmp(word, "on", MIN(strlen(word), 2))) + if (l && !strncasecmp(word, "on", MIN(l, 2))) return strdup("on"); case 1: - if (strlen(word) && !strncasecmp(word, "off", MIN(strlen(word), 3))) + if (l && !strncasecmp(word, "off", MIN(l, 3))) return strdup("off"); default: return NULL; @@ -842,17 +975,19 @@ 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 (!oss.owner) { + if (!o->owner) { ast_cli(fd, "No one is calling us\n"); return RESULT_FAILURE; } - hookstate = 1; - cursound = -1; - ast_queue_frame(oss.owner, &f); - answer_sound(); + o->hookstate = 1; + o->cursound = -1; + ast_queue_frame(o->owner, &f); + answer_sound(o); return RESULT_SUCCESS; } @@ -862,12 +997,14 @@ 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 (!oss.owner) { + if (!o->owner) { ast_cli(fd, "No one is calling us\n"); return RESULT_FAILURE; } @@ -883,7 +1020,7 @@ f.subclass = 0; f.data = text2send; f.datalen = strlen(text2send); - ast_queue_frame(oss.owner, &f); + ast_queue_frame(o->owner, &f); } return RESULT_SUCCESS; } @@ -894,84 +1031,89 @@ 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; - cursound = -1; - if (!oss.owner && !hookstate) { + o->cursound = -1; + if (!o->owner && !o->hookstate) { /* XXX maybe only one ? */ ast_cli(fd, "No call to hangup up\n"); return RESULT_FAILURE; } - hookstate = 0; - if (oss.owner) { - ast_queue_hangup(oss.owner); + 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_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; - cursound = -1; - if (!oss.owner) { + o->cursound = -1; + if (!o->owner) { /* XXX maybe !o->hookstate too ? */ ast_cli(fd, "No call to flash\n"); return RESULT_FAILURE; } - hookstate = 0; - if (oss.owner) { - ast_queue_frame(oss.owner, &f); + o->hookstate = 0; + if (o->owner) { /* XXX must be true, right ? */ + ast_queue_frame(o->owner, &f); } return RESULT_SUCCESS; } -static char hangup_usage[] = -"Usage: hangup\n" -" Hangs up any call currently placed on the console.\n"; - 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 tmp[256], *tmp2; - char *mye, *myc; - int x; + 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 (oss.owner) { - if (argc == 2) { - for (x=0;xowner) { /* 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; iowner, &f); + } return RESULT_SUCCESS; } - mye = exten; - myc = context; + /* if we have an argument split it into extension and context */ if (argc == 2) { - char *stringp=NULL; - strncpy(tmp, argv[1], sizeof(tmp)-1); - stringp=tmp; - strsep(&stringp, "@"); - tmp2 = strsep(&stringp, "@"); - if (strlen(tmp)) - mye = tmp; - if (tmp2 && strlen(tmp2)) - myc = tmp2; - } + 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)) { - strncpy(oss.exten, mye, sizeof(oss.exten)-1); - strncpy(oss.context, myc, sizeof(oss.context)-1); - hookstate = 1; - oss_new(&oss, AST_STATE_RINGING); + 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; @@ -983,29 +1125,33 @@ static int console_transfer(int fd, int argc, char *argv[]) { - char tmp[256]; - char *context; + struct chan_oss_pvt *o = find_desc(oss_active); + struct ast_channel *b = NULL; + char *ext, *ctx; + if (argc != 2) return RESULT_SHOWUSAGE; - if (oss.owner && ast_bridged_channel(oss.owner)) { - strncpy(tmp, argv[1], sizeof(tmp) - 1); - context = strchr(tmp, '@'); - if (context) { - *context = '\0'; - context++; - } else - context = oss.owner->context; - if (ast_exists_extension(ast_bridged_channel(oss.owner), context, tmp, 1, ast_bridged_channel(oss.owner)->cid.cid_num)) { - ast_cli(fd, "Whee, transferring %s to %s@%s.\n", - ast_bridged_channel(oss.owner)->name, tmp, context); - if (ast_async_goto(ast_bridged_channel(oss.owner), context, tmp, 1)) - ast_cli(fd, "Failed to transfer :(\n"); - } else { - ast_cli(fd, "No such extension exists\n"); + 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; } + + 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->cid.cid_num)) { + ast_cli(fd, "No such extension exists\n"); } else { - ast_cli(fd, "There is no call to transfer\n"); + 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; } @@ -1014,6 +1160,28 @@ " 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 }, @@ -1021,89 +1189,187 @@ { { "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 } + { { "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"} }; -int load_module() +/* + * 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 res; - int x; - struct ast_config *cfg; - struct ast_variable *v; - res = pipe(sndcmd); - if (res) { - ast_log(LOG_ERROR, "Unable to create pipe\n"); - return -1; + 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; } - res = soundcard_init(); - if (res < 0) { - if (option_verbose > 1) { - ast_verbose(VERBOSE_PREFIX_2 "No sound card detected -- console channel will be unavailable\n"); - ast_verbose(VERBOSE_PREFIX_2 "Turn off OSS support by adding 'noload=chan_oss.so' in /etc/asterisk/modules.conf\n"); } - return 0; + if (o->mixer_cmd) + free(o->mixer_cmd); + o->mixer_cmd = strdup(s); + ast_log(LOG_WARNING, "setting mixer %s\n", s); } - if (!full_duplex) - ast_log(LOG_WARNING, "XXX I don't work right with non-full duplex sound cards XXX\n"); - res = ast_channel_register(&oss_tech); - if (res < 0) { - ast_log(LOG_ERROR, "Unable to register channel class '%s'\n", type); - return -1; + +/* + * 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; } - for (x=0;xname = 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")) - autoanswer = ast_true(v->value); - else if (!strcasecmp(v->name, "silencesuppression")) - silencesuppression = ast_true(v->value); - else if (!strcasecmp(v->name, "silencethreshold")) - silencethreshold = atoi(v->value); - else if (!strcasecmp(v->name, "context")) - strncpy(context, v->value, sizeof(context)-1); - else if (!strcasecmp(v->name, "language")) - strncpy(language, v->value, sizeof(language)-1); - else if (!strcasecmp(v->name, "extension")) - strncpy(exten, v->value, sizeof(exten)-1); - else if (!strcasecmp(v->name, "playbackonly")) - playbackonly = ast_true(v->value); + M_START(v->name, v->value); + + M_BOOL("autoanswer", o->autoanswer) + M_BOOL("autohangup", o->autohangup) + M_BOOL("playbackonly", o->playbackonly) + M_BOOL("silencesuppression", o->silencesuppression) + M_UINT("silencethreshold", o->silencethreshold ) + 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(;); 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_config_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_config_destroy(cfg); } - ast_pthread_create(&sthread, NULL, sound_thread, NULL); + i = ast_channel_register(&oss_tech); + if (i < 0) { + ast_log(LOG_ERROR, "Unable to register channel class '%s'\n", + oss_default.type); + return NULL; + } + for (i=0; i 0) { - close(sndcmd[0]); - close(sndcmd[1]); - } - if (oss.owner) - ast_softhangup(oss.owner, AST_SOFTHANGUP_APPUNLOAD); - if (oss.owner) + + 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 *) desc; + return (char *)oss_tech.description; } -int usecount() +int usecount() /* XXX is this per-device or global for the module ? */ { int res; ast_mutex_lock(&usecnt_lock);