$FreeBSD$ --- channels/chan_oss.c +++ channels/chan_oss.c @@ -13,6 +13,8 @@ * * This program is free software, distributed under the terms of * the GNU General Public License + * + * FreeBSD changes by Luigi Rizzo, 2005.04.18 */ #include @@ -54,21 +56,30 @@ #endif /* Lets use 160 sample frames, just like GSM. */ -#define FRAME_SIZE 160 +/* this corresponds to 20ms of audio. */ +#define FRAME_SIZE 160 // was 160 -/* When you set the frame size, you have to come up with - the right buffer format as well. */ +/* + * When you set the frame size, you have to come up with + * the right buffer format as well. + * OSS lets you define a 'block' size (which should be a power of 2, + * which power is specified in the lower 16 bits) and the number of + * blocks allowed in the buffer (to avoid that the queue grows too large). + * The latter is specified in the top 16 bits. + * We use a block of 64 bytes (0x6), 5 blocks make a frame each sample + * being 2 bytes, and we make room to store two buffers. + * XXX the '10' is magic + */ + +#define N_BLOCKS (buffersize * 5 * 2) /* 5 64-byte frames = one frame */ -#define BUFFER_FMT ((buffersize * 10) << 16) | (0x0006); +#define BUFFER_FMT (N_BLOCKS << 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; AST_MUTEX_DEFINE_STATIC(usecnt_lock); @@ -78,16 +89,15 @@ static char *tdesc = "OSS Console Channel Driver"; static char *config = "oss.conf"; -static char context[AST_MAX_EXTENSION] = "default"; +static char default_context[AST_MAX_EXTENSION] = "default"; static char language[MAX_LANGUAGE] = ""; -static char exten[AST_MAX_EXTENSION] = "s"; +static char oss_exten[AST_MAX_EXTENSION] = "s"; -static int hookstate=0; -static short silence[FRAME_SIZE] = {0, }; struct sound { int ind; + char *desc; short *data; int datalen; int samplen; @@ -96,136 +106,178 @@ }; 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. */ + /* + * 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; + + 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 */ + + int silencesuppression; + int silencethreshold; + char device[64]; /* device to open */ + + pthread_t sthread; + struct ast_channel *owner; char exten[AST_MAX_EXTENSION]; char context[AST_MAX_EXTENSION]; -} oss; +} oss = { + .cursound = -1, + .sounddev = -1, + .duplex = M_UNSET, /* XXX check this */ + .autoanswer = 1, + .autohangup = 1, + .silencethreshold = 1000, +}; -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; - -static int autoanswer = 1; - -#if 0 -static int calc_loudness(short *frame) +/* + * Returns the number of blocks used in the audio output channel + */ +static int +used_blocks(struct chan_oss_pvt *o) { - int sum = 0; - int x; - for (x=0;xsounddev, SNDCTL_DSP_GETOSPACE, &info)) { + if (!warned) { + ast_log(LOG_WARNING, "Error reading output space\n"); + warned++; } - sum = sum/FRAME_SIZE; - return sum; + return 1; + } + if (o->total_blocks == 0) { + 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; } -#endif -static int cursound = -1; -static int sampsent = 0; -static int silencelen=0; -static int offset=0; -static int nosound=0; +static int soundcard_writeframe(struct chan_oss_pvt *o, short *data) +{ + /* Write an exactly FRAME_SIZE sized of frame */ + int res; + static int errors = 0; -static int send_sound(void) + /* + * nothing spectacular. + * If the buffer is full just drop the extra, otherwise write + */ + res = used_blocks(o); + if (res > 10) { /* no room to write a block */ + errors ++; + if (errors == 0) + ast_log(LOG_WARNING, "write: used %d blocks (%d)\n", res, errors); + return 0; + } + 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, + * (tone+silence) and passes it to the audio device. + */ +static int 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 = 0; + int l_sampsent = o->sampsent; + int l; + struct sound *s; + + if (o->cursound < 0) /* no sound to send */ + return 0; + s = &sounds[o->cursound]; + /* + * prepare a frame + */ + + for (ofs = 0; ofs < FRAME_SIZE; ofs += l) { + /* take chunks of sound and data until the buffer is full */ + l = s->samplen - l_sampsent; /* sound available */ + if (l > 0) { + if (l > FRAME_SIZE - ofs) + l = FRAME_SIZE - ofs; + if (0) + ast_log(LOG_WARNING, "send_sound sound %d/%d of %d into %d\n", + l_sampsent, l, s->samplen, ofs); + bcopy(s->data + l_sampsent, myframe + ofs, l*2); + 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; + if (0) + ast_log(LOG_WARNING, "send_sound silence %d/%d of %d into %d\n", + l_sampsent - s->samplen, l, s->silencelen, ofs); + bcopy(silence, myframe + ofs, l*2); + l_sampsent += l; + } else { /* silence is over, restart sound if loop */ + if (s->repeat == 0) { /* last block */ + ast_log(LOG_WARNING, "send_sound last block\n"); + 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 */ + return 0; /* fake success */ } static void *sound_thread(void *unused) @@ -235,41 +287,53 @@ int max; int res; char ign[4096]; - if (read(sounddev, ign, sizeof(sounddev)) < 0) + if (read(oss.sounddev, ign, sizeof(ign)) < 0) ast_log(LOG_WARNING, "Read error on sound device: %s\n", strerror(errno)); for(;;) { FD_ZERO(&rfds); FD_ZERO(&wfds); - max = sndcmd[0]; - FD_SET(sndcmd[0], &rfds); + max = oss.sndcmd[0]; + FD_SET(oss.sndcmd[0], &rfds); if (!oss.owner) { - FD_SET(sounddev, &rfds); - if (sounddev > max) - max = sounddev; + FD_SET(oss.sounddev, &rfds); + if (oss.sounddev > max) + max = oss.sounddev; } - if (cursound > -1) { - FD_SET(sounddev, &wfds); - if (sounddev > max) - max = sounddev; + if (oss.cursound > -1) { + FD_SET(oss.sounddev, &wfds); + if (oss.sounddev > max) + max = oss.sounddev; } res = ast_select(max + 1, &rfds, &wfds, NULL, NULL); if (res < 1) { 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(oss.sndcmd[0], &rfds)) { /* read which sound to play from the pipe */ + int i, what; + + read(oss.sndcmd[0], &what, sizeof(what)); + for (i = 0; sounds[i].ind != -1; i++) + if (sounds[i].ind == what) { + oss.cursound = i; + oss.sampsent = 0; + oss.nosound = 1; /* block other audio */ + ast_log(LOG_WARNING, "play %s\n", sounds[i].desc); + break; + } + if (sounds[i].ind == -1) + oss.cursound = -1; + ast_log(LOG_WARNING, "cursound %d samplen %d silencelen %d\n", + oss.cursound, oss.cursound >=0 ? sounds[oss.cursound].samplen : -1, + oss.cursound >=0 ? sounds[oss.cursound].silencelen : -1); } - if (FD_ISSET(sounddev, &rfds)) { + if (FD_ISSET(oss.sounddev, &rfds)) { /* Ignore read */ - if (read(sounddev, ign, sizeof(ign)) < 0) + if (read(oss.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()) + if (FD_ISSET(oss.sounddev, &wfds)) + if (send_sound(&oss) < 0) ast_log(LOG_WARNING, "Failed to write sound\n"); } /* Never reached */ @@ -277,6 +341,20 @@ } #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); 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; } - 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 (option_verbose > 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"); - 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) { @@ -348,6 +454,7 @@ 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; @@ -357,89 +464,54 @@ ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n", desired, fmt); } #if 1 - fmt = BUFFER_FMT; + /* + * on freebsd, SETFRAGMENT does not work very well on some cards. + * Better leave it out + */ + + // fmt = BUFFER_FMT; + fmt = 0x8; // 256-bytes fragment res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt); if (res < 0) { if (!warnedalready2++) ast_log(LOG_WARNING, "Unable to set fragment size -- sound may be choppy\n"); } #endif + /* 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(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; - } + if (oss.duplex == M_FULL || (oss.duplex == M_WRITE && !force)) return 0; - } - return 1; + if (!force && too_early(&oss)) + return 1; + if (setformat(&oss, 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(int force) { - int fd = sounddev; - if (full_duplex || (readmode && !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()) { - return -1; - } + if (oss.duplex == M_FULL || (oss.duplex == M_READ && !force)) return 0; - } - return 1; -} - -static int soundcard_init(void) -{ - /* 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 (!force && too_early(&oss)) + return 1; + if (setformat(&oss, O_RDONLY)) + return -1; + return 0; } static int oss_digit(struct ast_channel *c, char digit) @@ -454,120 +526,81 @@ return 0; } +/* request to play a sound on the speaker */ +#define RING(x) { int what = x; write(oss.sndcmd[1], &what, sizeof(what)); } + static int oss_call(struct ast_channel *c, char *dest, int timeout) { - int res = 3; struct ast_frame f = { 0, }; ast_verbose( " << Call placed to '%s' on console >> \n", dest); - if (autoanswer) { + if (oss.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(AST_CONTROL_RING); } return 0; } static void answer_sound(void) { - int res; - nosound = 1; - res = 4; - write(sndcmd[1], &res, sizeof(res)); - + RING(AST_CONTROL_ANSWER); } static int oss_answer(struct ast_channel *c) { ast_verbose( " << Console call has been answered >> \n"); - answer_sound(); + answer_sound(); /* XXX do we really need it ? considering we shut down immediately... */ ast_setstate(c, AST_STATE_UP); - cursound = -1; - nosound=0; + oss.cursound = -1; + oss.nosound=0; return 0; } static int oss_hangup(struct ast_channel *c) { - int res = 0; - cursound = -1; + oss.cursound = -1; c->pvt->pvt = NULL; oss.owner = NULL; ast_verbose( " << Hangup on console >> \n"); ast_mutex_lock(&usecnt_lock); usecnt--; ast_mutex_unlock(&usecnt_lock); - if (hookstate) { - if (autoanswer) { + if (oss.hookstate) { + if (oss.autoanswer || oss.autohangup) { /* Assume auto-hangup too */ - hookstate = 0; + oss.hookstate = 0; } else { /* Make congestion noise */ - res = 2; - write(sndcmd[1], &res, sizeof(res)); + RING(AST_CONTROL_CONGESTION); } } return 0; } -static int soundcard_writeframe(short *data) -{ - /* 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; -} - - +/* used for data coming from the network */ 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; + int src; + + // ast_log(LOG_WARNING, "oss_write size %d\n", f->datalen); /* Immediately return if no sound is enabled */ - if (nosound) + if (oss.nosound) return 0; /* Stop any currently playing sound */ - cursound = -1; - if (!full_duplex) { + oss.cursound = -1; + if (oss.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(1); + res = soundcard_setinput(1); /* force set if not full_duplex */ if (res < 0) { ast_log(LOG_WARNING, "Unable to set device to input mode\n"); return -1; @@ -583,21 +616,30 @@ 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; - } - 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; + /* + * 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 ) { + static char buf[FRAME_SIZE*2]; + static int dst = 0; + int l = sizeof(buf) - dst; /* how much room in the buffer */ + + if (f->datalen - src >= l) { /* enough to fill a frame */ + memcpy(buf + dst, f->data + src, l); + soundcard_writeframe(&oss, (short *)buf); + src += l; + dst = 0; + } else { /* copy residue */ + l = f->datalen - src; + memcpy(buf + dst, f->data + src, l); + src += l; /* but really, we are done */ + dst += l; + } } - if (len - pos) - memmove(sizbuf, sizbuf + pos, len - pos); - sizpos = len - pos; return 0; } @@ -628,18 +670,15 @@ ast_log(LOG_WARNING, "Unable to set input mode\n"); return NULL; } - if (res > 0) { + 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; - } + + res = read(oss.sounddev, buf + AST_FRIENDLY_OFFSET + readpos, FRAME_SIZE * 2 - readpos); + // ast_log(LOG_WARNING, "oss_read() fd %d got %d\n", oss.sounddev, res); + if (res < 0) /* audio data not ready, return a NULL frame */ + return &f; readpos += res; if (readpos >= FRAME_SIZE * 2) { @@ -682,64 +721,66 @@ 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; + oss.cursound = -1; return 0; default: ast_log(LOG_WARNING, "Don't know how to display condition %d on %s\n", cond, chan->name); return -1; } if (res > -1) { - write(sndcmd[1], &res, sizeof(res)); + RING(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 *oss, int state) { struct ast_channel *tmp; + struct ast_channel_pvt *pvt; + tmp = ast_channel_alloc(1); - if (tmp) { - snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", DEV_DSP + 5); - tmp->type = type; - tmp->fds[0] = sounddev; - tmp->nativeformats = AST_FORMAT_SLINEAR; - tmp->pvt->pvt = p; - tmp->pvt->send_digit = oss_digit; - tmp->pvt->send_text = oss_text; - tmp->pvt->hangup = oss_hangup; - tmp->pvt->answer = oss_answer; - tmp->pvt->read = oss_read; - tmp->pvt->call = oss_call; - tmp->pvt->write = oss_write; - tmp->pvt->indicate = oss_indicate; - tmp->pvt->fixup = oss_fixup; - 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); - 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 (tmp == NULL) + return NULL; + snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", oss->device + 5); + tmp->type = type; + tmp->fds[0] = oss->sounddev; + tmp->nativeformats = AST_FORMAT_SLINEAR; + pvt = tmp->pvt; + pvt->pvt = oss; +#if 1 + 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; +#endif + if (strlen(oss->context)) + strncpy(tmp->context, oss->context, sizeof(tmp->context)-1); + if (strlen(oss->exten)) + strncpy(tmp->exten, oss->exten, sizeof(tmp->exten)-1); + if (strlen(language)) + strncpy(tmp->language, language, sizeof(tmp->language)-1); + oss->owner = tmp; + ast_setstate(tmp, 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; + /* XXX what about oss->owner and the channel itself ? */ } } return tmp; @@ -770,13 +811,13 @@ if ((argc != 1) && (argc != 2)) return RESULT_SHOWUSAGE; if (argc == 1) { - ast_cli(fd, "Auto answer is %s.\n", autoanswer ? "on" : "off"); + ast_cli(fd, "Auto answer is %s.\n", oss.autoanswer ? "on" : "off"); return RESULT_SUCCESS; } else { if (!strcasecmp(argv[1], "on")) - autoanswer = -1; + oss.autoanswer = -1; else if (!strcasecmp(argv[1], "off")) - autoanswer = 0; + oss.autoanswer = 0; else return RESULT_SHOWUSAGE; } @@ -788,12 +829,14 @@ #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; @@ -816,8 +859,8 @@ ast_cli(fd, "No one is calling us\n"); return RESULT_FAILURE; } - hookstate = 1; - cursound = -1; + oss.hookstate = 1; + oss.cursound = -1; ast_queue_frame(oss.owner, &f); answer_sound(); return RESULT_SUCCESS; @@ -863,12 +906,12 @@ { if (argc != 1) return RESULT_SHOWUSAGE; - cursound = -1; - if (!oss.owner && !hookstate) { + oss.cursound = -1; + if (!oss.owner && !oss.hookstate) { ast_cli(fd, "No call to hangup up\n"); return RESULT_FAILURE; } - hookstate = 0; + oss.hookstate = 0; if (oss.owner) { ast_queue_hangup(oss.owner); } @@ -900,8 +943,8 @@ } return RESULT_SUCCESS; } - mye = exten; - myc = context; + mye = oss_exten; + myc = default_context; if (argc == 2) { char *stringp=NULL; strncpy(tmp, argv[1], sizeof(tmp)-1); @@ -916,7 +959,7 @@ 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.hookstate = 1; oss_new(&oss, AST_STATE_RINGING); } else ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc); @@ -974,21 +1017,47 @@ int res; int x; struct ast_config *cfg; - struct ast_variable *v; - res = pipe(sndcmd); + + res = pipe(oss.sndcmd); if (res) { ast_log(LOG_ERROR, "Unable to create pipe\n"); return -1; } - res = soundcard_init(); - if (res < 0) { + /* load config file */ + if ((cfg = ast_load(config))) { + struct ast_variable *v = ast_variable_browse(cfg, "general"); + while(v) { + if (!strcasecmp(v->name, "autoanswer")) + oss.autoanswer = ast_true(v->value); + else if (!strcasecmp(v->name, "autohangup")) + oss.autohangup = ast_true(v->value); + else if (!strcasecmp(v->name, "oss.silencesuppression")) + oss.silencesuppression = ast_true(v->value); + else if (!strcasecmp(v->name, "silencethreshold")) + oss.silencethreshold = atoi(v->value); + else if (!strcasecmp(v->name, "device")) + strncpy(oss.device, v->value, sizeof(oss.device)-1); + else if (!strcasecmp(v->name, "context")) + strncpy(default_context, v->value, sizeof(default_context)-1); + else if (!strcasecmp(v->name, "language")) + strncpy(language, v->value, sizeof(language)-1); + else if (!strcasecmp(v->name, "extension")) + strncpy(oss_exten, v->value, sizeof(oss_exten)-1); + v=v->next; + } + ast_destroy(cfg); + } + if (!strlen(oss.device)) + strncpy(oss.device, DEV_DSP, sizeof(oss.device)-1); + if (setformat(&oss, O_RDWR) < 0) { /* open device */ 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 (!full_duplex) + soundcard_setinput(1); /* force set if not full_duplex */ + if (oss.duplex != M_FULL) ast_log(LOG_WARNING, "XXX I don't work right with non-full duplex sound cards XXX\n"); res = ast_channel_register(type, tdesc, AST_FORMAT_SLINEAR, oss_request); if (res < 0) { @@ -997,26 +1066,7 @@ } for (x=0;xname, "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); - v=v->next; - } - ast_destroy(cfg); - } - ast_pthread_create(&sthread, NULL, sound_thread, NULL); + ast_pthread_create(&oss.sthread, NULL, sound_thread, NULL); return 0; } @@ -1027,15 +1077,16 @@ int x; for (x=0;x 0) { - close(sndcmd[0]); - close(sndcmd[1]); + close(oss.sounddev); + if (oss.sndcmd[0] > 0) { + close(oss.sndcmd[0]); + close(oss.sndcmd[1]); } if (oss.owner) ast_softhangup(oss.owner, AST_SOFTHANGUP_APPUNLOAD); if (oss.owner) return -1; + /* XXX what about the thread ? */ return 0; }