| 1 | /* |
| 2 | * This file is part of DisOrder |
| 3 | * Copyright (C) 2005-2008 Richard Kettlewell |
| 4 | * Portions (C) 2007 Mark Wooding |
| 5 | * |
| 6 | * This program is free software: you can redistribute it and/or modify |
| 7 | * it under the terms of the GNU General Public License as published by |
| 8 | * the Free Software Foundation, either version 3 of the License, or |
| 9 | * (at your option) any later version. |
| 10 | * |
| 11 | * This program is distributed in the hope that it will be useful, |
| 12 | * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| 13 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
| 14 | * GNU General Public License for more details. |
| 15 | * |
| 16 | * You should have received a copy of the GNU General Public License |
| 17 | * along with this program. If not, see <http://www.gnu.org/licenses/>. |
| 18 | */ |
| 19 | /** @file server/speaker.c |
| 20 | * @brief Speaker process |
| 21 | * |
| 22 | * This program is responsible for transmitting a single coherent audio stream |
| 23 | * to its destination (over the network, to some sound API, to some |
| 24 | * subprocess). It receives connections from decoders (or rather from the |
| 25 | * process that is about to become disorder-normalize) and plays them in the |
| 26 | * right order. |
| 27 | * |
| 28 | * @b Encodings. For the <a href="http://www.alsa-project.org/">ALSA</a> API, |
| 29 | * 8- and 16- bit stereo and mono are supported, with any sample rate (within |
| 30 | * the limits that ALSA can deal with.) |
| 31 | * |
| 32 | * Inbound data is expected to match @c config->sample_format. In normal use |
| 33 | * this is arranged by the @c disorder-normalize program (see @ref |
| 34 | * server/normalize.c). |
| 35 | * |
| 36 | * @b Garbage @b Collection. This program deliberately does not use the |
| 37 | * garbage collector even though it might be convenient to do so. This is for |
| 38 | * two reasons. Firstly some sound APIs use thread threads and we do not want |
| 39 | * to have to deal with potential interactions between threading and garbage |
| 40 | * collection. Secondly this process needs to be able to respond quickly and |
| 41 | * this is not compatible with the collector hanging the program even |
| 42 | * relatively briefly. |
| 43 | * |
| 44 | * @b Units. This program thinks at various times in three different units. |
| 45 | * Bytes are obvious. A sample is a single sample on a single channel. A |
| 46 | * frame is several samples on different channels at the same point in time. |
| 47 | * So (for instance) a 16-bit stereo frame is 4 bytes and consists of a pair of |
| 48 | * 2-byte samples. |
| 49 | */ |
| 50 | |
| 51 | #include "common.h" |
| 52 | |
| 53 | #include <getopt.h> |
| 54 | #include <locale.h> |
| 55 | #include <syslog.h> |
| 56 | #include <unistd.h> |
| 57 | #include <errno.h> |
| 58 | #include <ao/ao.h> |
| 59 | #include <sys/select.h> |
| 60 | #include <sys/wait.h> |
| 61 | #include <time.h> |
| 62 | #include <fcntl.h> |
| 63 | #include <poll.h> |
| 64 | #include <sys/un.h> |
| 65 | #include <sys/stat.h> |
| 66 | |
| 67 | #include "configuration.h" |
| 68 | #include "syscalls.h" |
| 69 | #include "log.h" |
| 70 | #include "defs.h" |
| 71 | #include "mem.h" |
| 72 | #include "speaker-protocol.h" |
| 73 | #include "user.h" |
| 74 | #include "speaker.h" |
| 75 | #include "printf.h" |
| 76 | #include "version.h" |
| 77 | |
| 78 | /** @brief Linked list of all prepared tracks */ |
| 79 | struct track *tracks; |
| 80 | |
| 81 | /** @brief Playing track, or NULL */ |
| 82 | struct track *playing; |
| 83 | |
| 84 | /** @brief Number of bytes pre frame */ |
| 85 | size_t bpf; |
| 86 | |
| 87 | /** @brief Array of file descriptors for poll() */ |
| 88 | struct pollfd fds[NFDS]; |
| 89 | |
| 90 | /** @brief Next free slot in @ref fds */ |
| 91 | int fdno; |
| 92 | |
| 93 | /** @brief Listen socket */ |
| 94 | static int listenfd; |
| 95 | |
| 96 | static time_t last_report; /* when we last reported */ |
| 97 | static int paused; /* pause status */ |
| 98 | |
| 99 | /** @brief The current device state */ |
| 100 | enum device_states device_state; |
| 101 | |
| 102 | /** @brief Set when idled |
| 103 | * |
| 104 | * This is set when the sound device is deliberately closed by idle(). |
| 105 | */ |
| 106 | int idled; |
| 107 | |
| 108 | /** @brief Selected backend */ |
| 109 | static const struct speaker_backend *backend; |
| 110 | |
| 111 | static const struct option options[] = { |
| 112 | { "help", no_argument, 0, 'h' }, |
| 113 | { "version", no_argument, 0, 'V' }, |
| 114 | { "config", required_argument, 0, 'c' }, |
| 115 | { "debug", no_argument, 0, 'd' }, |
| 116 | { "no-debug", no_argument, 0, 'D' }, |
| 117 | { "syslog", no_argument, 0, 's' }, |
| 118 | { "no-syslog", no_argument, 0, 'S' }, |
| 119 | { 0, 0, 0, 0 } |
| 120 | }; |
| 121 | |
| 122 | /* Display usage message and terminate. */ |
| 123 | static void help(void) { |
| 124 | xprintf("Usage:\n" |
| 125 | " disorder-speaker [OPTIONS]\n" |
| 126 | "Options:\n" |
| 127 | " --help, -h Display usage message\n" |
| 128 | " --version, -V Display version number\n" |
| 129 | " --config PATH, -c PATH Set configuration file\n" |
| 130 | " --debug, -d Turn on debugging\n" |
| 131 | " --[no-]syslog Force logging\n" |
| 132 | "\n" |
| 133 | "Speaker process for DisOrder. Not intended to be run\n" |
| 134 | "directly.\n"); |
| 135 | xfclose(stdout); |
| 136 | exit(0); |
| 137 | } |
| 138 | |
| 139 | /** @brief Return the number of bytes per frame in @p format */ |
| 140 | static size_t bytes_per_frame(const struct stream_header *format) { |
| 141 | return format->channels * format->bits / 8; |
| 142 | } |
| 143 | |
| 144 | /** @brief Find track @p id, maybe creating it if not found */ |
| 145 | static struct track *findtrack(const char *id, int create) { |
| 146 | struct track *t; |
| 147 | |
| 148 | D(("findtrack %s %d", id, create)); |
| 149 | for(t = tracks; t && strcmp(id, t->id); t = t->next) |
| 150 | ; |
| 151 | if(!t && create) { |
| 152 | t = xmalloc(sizeof *t); |
| 153 | t->next = tracks; |
| 154 | strcpy(t->id, id); |
| 155 | t->fd = -1; |
| 156 | tracks = t; |
| 157 | } |
| 158 | return t; |
| 159 | } |
| 160 | |
| 161 | /** @brief Remove track @p id (but do not destroy it) */ |
| 162 | static struct track *removetrack(const char *id) { |
| 163 | struct track *t, **tt; |
| 164 | |
| 165 | D(("removetrack %s", id)); |
| 166 | for(tt = &tracks; (t = *tt) && strcmp(id, t->id); tt = &t->next) |
| 167 | ; |
| 168 | if(t) |
| 169 | *tt = t->next; |
| 170 | return t; |
| 171 | } |
| 172 | |
| 173 | /** @brief Destroy a track */ |
| 174 | static void destroy(struct track *t) { |
| 175 | D(("destroy %s", t->id)); |
| 176 | if(t->fd != -1) xclose(t->fd); |
| 177 | free(t); |
| 178 | } |
| 179 | |
| 180 | /** @brief Read data into a sample buffer |
| 181 | * @param t Pointer to track |
| 182 | * @return 0 on success, -1 on EOF |
| 183 | * |
| 184 | * This is effectively the read callback on @c t->fd. It is called from the |
| 185 | * main loop whenever the track's file descriptor is readable, assuming the |
| 186 | * buffer has not reached the maximum allowed occupancy. |
| 187 | */ |
| 188 | static int speaker_fill(struct track *t) { |
| 189 | size_t where, left; |
| 190 | int n; |
| 191 | |
| 192 | D(("fill %s: eof=%d used=%zu", |
| 193 | t->id, t->eof, t->used)); |
| 194 | if(t->eof) return -1; |
| 195 | if(t->used < sizeof t->buffer) { |
| 196 | /* there is room left in the buffer */ |
| 197 | where = (t->start + t->used) % sizeof t->buffer; |
| 198 | /* Get as much data as we can */ |
| 199 | if(where >= t->start) left = (sizeof t->buffer) - where; |
| 200 | else left = t->start - where; |
| 201 | do { |
| 202 | n = read(t->fd, t->buffer + where, left); |
| 203 | } while(n < 0 && errno == EINTR); |
| 204 | if(n < 0) { |
| 205 | if(errno != EAGAIN) fatal(errno, "error reading sample stream"); |
| 206 | return 0; |
| 207 | } |
| 208 | if(n == 0) { |
| 209 | D(("fill %s: eof detected", t->id)); |
| 210 | t->eof = 1; |
| 211 | t->playable = 1; |
| 212 | return -1; |
| 213 | } |
| 214 | t->used += n; |
| 215 | if(t->used == sizeof t->buffer) |
| 216 | t->playable = 1; |
| 217 | } |
| 218 | return 0; |
| 219 | } |
| 220 | |
| 221 | /** @brief Close the sound device |
| 222 | * |
| 223 | * This is called to deactivate the output device when pausing, and also by the |
| 224 | * ALSA backend when changing encoding (in which case the sound device will be |
| 225 | * immediately reactivated). |
| 226 | */ |
| 227 | static void idle(void) { |
| 228 | D(("idle")); |
| 229 | if(backend->deactivate) |
| 230 | backend->deactivate(); |
| 231 | else |
| 232 | device_state = device_closed; |
| 233 | idled = 1; |
| 234 | } |
| 235 | |
| 236 | /** @brief Abandon the current track */ |
| 237 | void abandon(void) { |
| 238 | struct speaker_message sm; |
| 239 | |
| 240 | D(("abandon")); |
| 241 | memset(&sm, 0, sizeof sm); |
| 242 | sm.type = SM_FINISHED; |
| 243 | strcpy(sm.id, playing->id); |
| 244 | speaker_send(1, &sm); |
| 245 | removetrack(playing->id); |
| 246 | destroy(playing); |
| 247 | playing = 0; |
| 248 | } |
| 249 | |
| 250 | /** @brief Enable sound output |
| 251 | * |
| 252 | * Makes sure the sound device is open and has the right sample format. Return |
| 253 | * 0 on success and -1 on error. |
| 254 | */ |
| 255 | static void activate(void) { |
| 256 | if(backend->activate) |
| 257 | backend->activate(); |
| 258 | else |
| 259 | device_state = device_open; |
| 260 | } |
| 261 | |
| 262 | /** @brief Check whether the current track has finished |
| 263 | * |
| 264 | * The current track is determined to have finished either if the input stream |
| 265 | * eded before the format could be determined (i.e. it is malformed) or the |
| 266 | * input is at end of file and there is less than a frame left unplayed. (So |
| 267 | * it copes with decoders that crash mid-frame.) |
| 268 | */ |
| 269 | static void maybe_finished(void) { |
| 270 | if(playing |
| 271 | && playing->eof |
| 272 | && playing->used < bytes_per_frame(&config->sample_format)) |
| 273 | abandon(); |
| 274 | } |
| 275 | |
| 276 | /** @brief Return nonzero if we want to play some audio |
| 277 | * |
| 278 | * We want to play audio if there is a current track; and it is not paused; and |
| 279 | * it is playable according to the rules for @ref track::playable. |
| 280 | */ |
| 281 | static int playable(void) { |
| 282 | return playing |
| 283 | && !paused |
| 284 | && playing->playable; |
| 285 | } |
| 286 | |
| 287 | /** @brief Play up to @p frames frames of audio |
| 288 | * |
| 289 | * It is always safe to call this function. |
| 290 | * - If @ref playing is 0 then it will just return |
| 291 | * - If @ref paused is non-0 then it will just return |
| 292 | * - If @ref device_state != @ref device_open then it will call activate() and |
| 293 | * return if it it fails. |
| 294 | * - If there is not enough audio to play then it play what is available. |
| 295 | * |
| 296 | * If there are not enough frames to play then whatever is available is played |
| 297 | * instead. It is up to mainloop() to ensure that speaker_play() is not called |
| 298 | * when unreasonably only an small amounts of data is available to play. |
| 299 | */ |
| 300 | static void speaker_play(size_t frames) { |
| 301 | size_t avail_frames, avail_bytes, written_frames; |
| 302 | ssize_t written_bytes; |
| 303 | |
| 304 | /* Make sure there's a track to play and it is not paused */ |
| 305 | if(!playable()) |
| 306 | return; |
| 307 | /* Make sure the output device is open */ |
| 308 | if(device_state != device_open) { |
| 309 | activate(); |
| 310 | if(device_state != device_open) |
| 311 | return; |
| 312 | } |
| 313 | D(("play: play %zu/%zu%s %dHz %db %dc", frames, playing->used / bpf, |
| 314 | playing->eof ? " EOF" : "", |
| 315 | config->sample_format.rate, |
| 316 | config->sample_format.bits, |
| 317 | config->sample_format.channels)); |
| 318 | /* Figure out how many frames there are available to write */ |
| 319 | if(playing->start + playing->used > sizeof playing->buffer) |
| 320 | /* The ring buffer is currently wrapped, only play up to the wrap point */ |
| 321 | avail_bytes = (sizeof playing->buffer) - playing->start; |
| 322 | else |
| 323 | /* The ring buffer is not wrapped, can play the lot */ |
| 324 | avail_bytes = playing->used; |
| 325 | avail_frames = avail_bytes / bpf; |
| 326 | /* Only play up to the requested amount */ |
| 327 | if(avail_frames > frames) |
| 328 | avail_frames = frames; |
| 329 | if(!avail_frames) |
| 330 | return; |
| 331 | /* Play it, Sam */ |
| 332 | written_frames = backend->play(avail_frames); |
| 333 | written_bytes = written_frames * bpf; |
| 334 | /* written_bytes and written_frames had better both be set and correct by |
| 335 | * this point */ |
| 336 | playing->start += written_bytes; |
| 337 | playing->used -= written_bytes; |
| 338 | playing->played += written_frames; |
| 339 | /* If the pointer is at the end of the buffer (or the buffer is completely |
| 340 | * empty) wrap it back to the start. */ |
| 341 | if(!playing->used || playing->start == (sizeof playing->buffer)) |
| 342 | playing->start = 0; |
| 343 | /* If the buffer emptied out mark the track as unplayably */ |
| 344 | if(!playing->used && !playing->eof) { |
| 345 | error(0, "track buffer emptied"); |
| 346 | playing->playable = 0; |
| 347 | } |
| 348 | frames -= written_frames; |
| 349 | return; |
| 350 | } |
| 351 | |
| 352 | /* Notify the server what we're up to. */ |
| 353 | static void report(void) { |
| 354 | struct speaker_message sm; |
| 355 | |
| 356 | if(playing) { |
| 357 | memset(&sm, 0, sizeof sm); |
| 358 | sm.type = paused ? SM_PAUSED : SM_PLAYING; |
| 359 | strcpy(sm.id, playing->id); |
| 360 | sm.data = playing->played / config->sample_format.rate; |
| 361 | speaker_send(1, &sm); |
| 362 | } |
| 363 | time(&last_report); |
| 364 | } |
| 365 | |
| 366 | static void reap(int __attribute__((unused)) sig) { |
| 367 | pid_t cmdpid; |
| 368 | int st; |
| 369 | |
| 370 | do |
| 371 | cmdpid = waitpid(-1, &st, WNOHANG); |
| 372 | while(cmdpid > 0); |
| 373 | signal(SIGCHLD, reap); |
| 374 | } |
| 375 | |
| 376 | int addfd(int fd, int events) { |
| 377 | if(fdno < NFDS) { |
| 378 | fds[fdno].fd = fd; |
| 379 | fds[fdno].events = events; |
| 380 | return fdno++; |
| 381 | } else |
| 382 | return -1; |
| 383 | } |
| 384 | |
| 385 | /** @brief Table of speaker backends */ |
| 386 | static const struct speaker_backend *backends[] = { |
| 387 | #if HAVE_ALSA_ASOUNDLIB_H |
| 388 | &alsa_backend, |
| 389 | #endif |
| 390 | &command_backend, |
| 391 | &network_backend, |
| 392 | #if HAVE_COREAUDIO_AUDIOHARDWARE_H |
| 393 | &coreaudio_backend, |
| 394 | #endif |
| 395 | #if HAVE_SYS_SOUNDCARD_H |
| 396 | &oss_backend, |
| 397 | #endif |
| 398 | 0 |
| 399 | }; |
| 400 | |
| 401 | /** @brief Main event loop */ |
| 402 | static void mainloop(void) { |
| 403 | struct track *t; |
| 404 | struct speaker_message sm; |
| 405 | int n, fd, stdin_slot, timeout, listen_slot; |
| 406 | |
| 407 | while(getppid() != 1) { |
| 408 | fdno = 0; |
| 409 | /* By default we will wait up to a second before thinking about current |
| 410 | * state. */ |
| 411 | timeout = 1000; |
| 412 | /* Always ready for commands from the main server. */ |
| 413 | stdin_slot = addfd(0, POLLIN); |
| 414 | /* Also always ready for inbound connections */ |
| 415 | listen_slot = addfd(listenfd, POLLIN); |
| 416 | /* Try to read sample data for the currently playing track if there is |
| 417 | * buffer space. */ |
| 418 | if(playing |
| 419 | && playing->fd >= 0 |
| 420 | && !playing->eof |
| 421 | && playing->used < (sizeof playing->buffer)) |
| 422 | playing->slot = addfd(playing->fd, POLLIN); |
| 423 | else if(playing) |
| 424 | playing->slot = -1; |
| 425 | if(playable()) { |
| 426 | /* We want to play some audio. If the device is closed then we attempt |
| 427 | * to open it. */ |
| 428 | if(device_state == device_closed) |
| 429 | activate(); |
| 430 | /* If the device is (now) open then we will wait up until it is ready for |
| 431 | * more. If something went wrong then we should have device_error |
| 432 | * instead, but the post-poll code will cope even if it's |
| 433 | * device_closed. */ |
| 434 | if(device_state == device_open) |
| 435 | backend->beforepoll(&timeout); |
| 436 | } |
| 437 | /* If any other tracks don't have a full buffer, try to read sample data |
| 438 | * from them. We do this last of all, so that if we run out of slots, |
| 439 | * nothing important can't be monitored. */ |
| 440 | for(t = tracks; t; t = t->next) |
| 441 | if(t != playing) { |
| 442 | if(t->fd >= 0 |
| 443 | && !t->eof |
| 444 | && t->used < sizeof t->buffer) { |
| 445 | t->slot = addfd(t->fd, POLLIN | POLLHUP); |
| 446 | } else |
| 447 | t->slot = -1; |
| 448 | } |
| 449 | /* Wait for something interesting to happen */ |
| 450 | n = poll(fds, fdno, timeout); |
| 451 | if(n < 0) { |
| 452 | if(errno == EINTR) continue; |
| 453 | fatal(errno, "error calling poll"); |
| 454 | } |
| 455 | /* Play some sound before doing anything else */ |
| 456 | if(playable()) { |
| 457 | /* We want to play some audio */ |
| 458 | if(device_state == device_open) { |
| 459 | if(backend->ready()) |
| 460 | speaker_play(3 * FRAMES); |
| 461 | } else { |
| 462 | /* We must be in _closed or _error, and it should be the latter, but we |
| 463 | * cope with either. |
| 464 | * |
| 465 | * We most likely timed out, so now is a good time to retry. |
| 466 | * speaker_play() knows to re-activate the device if necessary. |
| 467 | */ |
| 468 | speaker_play(3 * FRAMES); |
| 469 | } |
| 470 | } |
| 471 | /* Perhaps a connection has arrived */ |
| 472 | if(fds[listen_slot].revents & POLLIN) { |
| 473 | struct sockaddr_un addr; |
| 474 | socklen_t addrlen = sizeof addr; |
| 475 | uint32_t l; |
| 476 | char id[24]; |
| 477 | |
| 478 | if((fd = accept(listenfd, (struct sockaddr *)&addr, &addrlen)) >= 0) { |
| 479 | blocking(fd); |
| 480 | if(read(fd, &l, sizeof l) < 4) { |
| 481 | error(errno, "reading length from inbound connection"); |
| 482 | xclose(fd); |
| 483 | } else if(l >= sizeof id) { |
| 484 | error(0, "id length too long"); |
| 485 | xclose(fd); |
| 486 | } else if(read(fd, id, l) < (ssize_t)l) { |
| 487 | error(errno, "reading id from inbound connection"); |
| 488 | xclose(fd); |
| 489 | } else { |
| 490 | id[l] = 0; |
| 491 | D(("id %s fd %d", id, fd)); |
| 492 | t = findtrack(id, 1/*create*/); |
| 493 | if (write(fd, "", 1) < 0) /* write an ack */ |
| 494 | error(errno, "writing ack to inbound connection"); |
| 495 | if(t->fd != -1) { |
| 496 | error(0, "%s: already got a connection", id); |
| 497 | xclose(fd); |
| 498 | } else { |
| 499 | nonblock(fd); |
| 500 | t->fd = fd; /* yay */ |
| 501 | } |
| 502 | } |
| 503 | } else |
| 504 | error(errno, "accept"); |
| 505 | } |
| 506 | /* Perhaps we have a command to process */ |
| 507 | if(fds[stdin_slot].revents & POLLIN) { |
| 508 | /* There might (in theory) be several commands queued up, but in general |
| 509 | * this won't be the case, so we don't bother looping around to pick them |
| 510 | * all up. */ |
| 511 | n = speaker_recv(0, &sm); |
| 512 | /* TODO */ |
| 513 | if(n > 0) |
| 514 | switch(sm.type) { |
| 515 | case SM_PLAY: |
| 516 | if(playing) fatal(0, "got SM_PLAY but already playing something"); |
| 517 | t = findtrack(sm.id, 1); |
| 518 | D(("SM_PLAY %s fd %d", t->id, t->fd)); |
| 519 | if(t->fd == -1) |
| 520 | error(0, "cannot play track because no connection arrived"); |
| 521 | playing = t; |
| 522 | /* We attempt to play straight away rather than going round the loop. |
| 523 | * speaker_play() is clever enough to perform any activation that is |
| 524 | * required. */ |
| 525 | speaker_play(3 * FRAMES); |
| 526 | report(); |
| 527 | break; |
| 528 | case SM_PAUSE: |
| 529 | D(("SM_PAUSE")); |
| 530 | paused = 1; |
| 531 | report(); |
| 532 | break; |
| 533 | case SM_RESUME: |
| 534 | D(("SM_RESUME")); |
| 535 | if(paused) { |
| 536 | paused = 0; |
| 537 | /* As for SM_PLAY we attempt to play straight away. */ |
| 538 | if(playing) |
| 539 | speaker_play(3 * FRAMES); |
| 540 | } |
| 541 | report(); |
| 542 | break; |
| 543 | case SM_CANCEL: |
| 544 | D(("SM_CANCEL %s", sm.id)); |
| 545 | t = removetrack(sm.id); |
| 546 | if(t) { |
| 547 | if(t == playing) { |
| 548 | /* scratching the playing track */ |
| 549 | sm.type = SM_FINISHED; |
| 550 | playing = 0; |
| 551 | } else { |
| 552 | /* Could be scratching the playing track before it's quite got |
| 553 | * going, or could be just removing a track from the queue. We |
| 554 | * log more because there's been a bug here recently than because |
| 555 | * it's particularly interesting; the log message will be removed |
| 556 | * if no further problems show up. */ |
| 557 | info("SM_CANCEL for nonplaying track %s", sm.id); |
| 558 | sm.type = SM_STILLBORN; |
| 559 | } |
| 560 | strcpy(sm.id, t->id); |
| 561 | destroy(t); |
| 562 | } else { |
| 563 | /* Probably scratching the playing track well before it's got |
| 564 | * going, but could indicate a bug, so we log this as an error. */ |
| 565 | sm.type = SM_UNKNOWN; |
| 566 | error(0, "SM_CANCEL for unknown track %s", sm.id); |
| 567 | } |
| 568 | speaker_send(1, &sm); |
| 569 | report(); |
| 570 | break; |
| 571 | case SM_RELOAD: |
| 572 | D(("SM_RELOAD")); |
| 573 | if(config_read(1)) error(0, "cannot read configuration"); |
| 574 | info("reloaded configuration"); |
| 575 | break; |
| 576 | default: |
| 577 | error(0, "unknown message type %d", sm.type); |
| 578 | } |
| 579 | } |
| 580 | /* Read in any buffered data */ |
| 581 | for(t = tracks; t; t = t->next) |
| 582 | if(t->fd != -1 |
| 583 | && t->slot != -1 |
| 584 | && (fds[t->slot].revents & (POLLIN | POLLHUP))) |
| 585 | speaker_fill(t); |
| 586 | /* Maybe we finished playing a track somewhere in the above */ |
| 587 | maybe_finished(); |
| 588 | /* If we don't need the sound device for now then close it for the benefit |
| 589 | * of anyone else who wants it. */ |
| 590 | if((!playing || paused) && device_state == device_open) |
| 591 | idle(); |
| 592 | /* If we've not reported out state for a second do so now. */ |
| 593 | if(time(0) > last_report) |
| 594 | report(); |
| 595 | } |
| 596 | } |
| 597 | |
| 598 | int main(int argc, char **argv) { |
| 599 | int n, logsyslog = !isatty(2); |
| 600 | struct sockaddr_un addr; |
| 601 | static const int one = 1; |
| 602 | struct speaker_message sm; |
| 603 | const char *d; |
| 604 | char *dir; |
| 605 | |
| 606 | set_progname(argv); |
| 607 | if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); |
| 608 | while((n = getopt_long(argc, argv, "hVc:dDSs", options, 0)) >= 0) { |
| 609 | switch(n) { |
| 610 | case 'h': help(); |
| 611 | case 'V': version("disorder-speaker"); |
| 612 | case 'c': configfile = optarg; break; |
| 613 | case 'd': debugging = 1; break; |
| 614 | case 'D': debugging = 0; break; |
| 615 | case 'S': logsyslog = 0; break; |
| 616 | case 's': logsyslog = 1; break; |
| 617 | default: fatal(0, "invalid option"); |
| 618 | } |
| 619 | } |
| 620 | if((d = getenv("DISORDER_DEBUG_SPEAKER"))) debugging = atoi(d); |
| 621 | if(logsyslog) { |
| 622 | openlog(progname, LOG_PID, LOG_DAEMON); |
| 623 | log_default = &log_syslog; |
| 624 | } |
| 625 | if(config_read(1)) fatal(0, "cannot read configuration"); |
| 626 | bpf = bytes_per_frame(&config->sample_format); |
| 627 | /* ignore SIGPIPE */ |
| 628 | signal(SIGPIPE, SIG_IGN); |
| 629 | /* reap kids */ |
| 630 | signal(SIGCHLD, reap); |
| 631 | /* set nice value */ |
| 632 | xnice(config->nice_speaker); |
| 633 | /* change user */ |
| 634 | become_mortal(); |
| 635 | /* make sure we're not root, whatever the config says */ |
| 636 | if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root"); |
| 637 | /* identify the backend used to play */ |
| 638 | for(n = 0; backends[n]; ++n) |
| 639 | if(backends[n]->backend == config->api) |
| 640 | break; |
| 641 | if(!backends[n]) |
| 642 | fatal(0, "unsupported api %d", config->api); |
| 643 | backend = backends[n]; |
| 644 | /* backend-specific initialization */ |
| 645 | backend->init(); |
| 646 | /* create the socket directory */ |
| 647 | byte_xasprintf(&dir, "%s/speaker", config->home); |
| 648 | unlink(dir); /* might be a leftover socket */ |
| 649 | if(mkdir(dir, 0700) < 0 && errno != EEXIST) |
| 650 | fatal(errno, "error creating %s", dir); |
| 651 | /* set up the listen socket */ |
| 652 | listenfd = xsocket(PF_UNIX, SOCK_STREAM, 0); |
| 653 | memset(&addr, 0, sizeof addr); |
| 654 | addr.sun_family = AF_UNIX; |
| 655 | snprintf(addr.sun_path, sizeof addr.sun_path, "%s/speaker/socket", |
| 656 | config->home); |
| 657 | if(unlink(addr.sun_path) < 0 && errno != ENOENT) |
| 658 | error(errno, "removing %s", addr.sun_path); |
| 659 | xsetsockopt(listenfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one); |
| 660 | if(bind(listenfd, (const struct sockaddr *)&addr, sizeof addr) < 0) |
| 661 | fatal(errno, "error binding socket to %s", addr.sun_path); |
| 662 | xlisten(listenfd, 128); |
| 663 | nonblock(listenfd); |
| 664 | info("listening on %s", addr.sun_path); |
| 665 | memset(&sm, 0, sizeof sm); |
| 666 | sm.type = SM_READY; |
| 667 | speaker_send(1, &sm); |
| 668 | mainloop(); |
| 669 | info("stopped (parent terminated)"); |
| 670 | exit(0); |
| 671 | } |
| 672 | |
| 673 | /* |
| 674 | Local Variables: |
| 675 | c-basic-offset:2 |
| 676 | comment-column:40 |
| 677 | fill-column:79 |
| 678 | indent-tabs-mode:nil |
| 679 | End: |
| 680 | */ |