| 1 | /* |
| 2 | * This file is part of DisOrder |
| 3 | * Copyright (C) 2005-2013 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 Model. mainloop() implements a select loop awaiting commands from the |
| 29 | * main server, new connections to the speaker socket, and audio data on those |
| 30 | * connections. Each connection starts with a queue ID (with a 32-bit |
| 31 | * native-endian length word), allowing it to be referred to in commands from |
| 32 | * the server. |
| 33 | * |
| 34 | * Data read on connections is buffered, up to a limit (currently 1Mbyte per |
| 35 | * track). No attempt is made here to limit the number of tracks, it is |
| 36 | * assumed that the main server won't start outrageously many decoders. |
| 37 | * |
| 38 | * Audio is supplied from this buffer to the uaudio play callback. Playback is |
| 39 | * enabled when a track is to be played and disabled when its last bytes |
| 40 | * have been returned by the callback; pause and resume is implemented the |
| 41 | * obvious way. If the callback finds itself required to play when there is no |
| 42 | * playing track it returns dead air. |
| 43 | * |
| 44 | * To implement gapless playback, the server is notified that a track has |
| 45 | * finished slightly early. @ref SM_PLAY is therefore allowed to arrive while |
| 46 | * the previous track is still playing provided an early @ref SM_FINISHED has |
| 47 | * been sent for it. |
| 48 | * |
| 49 | * @b Encodings. The encodings supported depend entirely on the uaudio backend |
| 50 | * chosen. See @ref uaudio.h, etc. |
| 51 | * |
| 52 | * Inbound data is expected to match @c config->sample_format. In normal use |
| 53 | * this is arranged by the @c disorder-normalize program (see @ref |
| 54 | * server/normalize.c). |
| 55 | * |
| 56 | * @b Garbage @b Collection. This program deliberately does not use the |
| 57 | * garbage collector even though it might be convenient to do so. This is for |
| 58 | * two reasons. Firstly some sound APIs use thread threads and we do not want |
| 59 | * to have to deal with potential interactions between threading and garbage |
| 60 | * collection. Secondly this process needs to be able to respond quickly and |
| 61 | * this is not compatible with the collector hanging the program even |
| 62 | * relatively briefly. |
| 63 | * |
| 64 | * @b Units. This program thinks at various times in three different units. |
| 65 | * Bytes are obvious. A sample is a single sample on a single channel. A |
| 66 | * frame is several samples on different channels at the same point in time. |
| 67 | * So (for instance) a 16-bit stereo frame is 4 bytes and consists of a pair of |
| 68 | * 2-byte samples. |
| 69 | */ |
| 70 | |
| 71 | #include "common.h" |
| 72 | |
| 73 | #include <getopt.h> |
| 74 | #include <locale.h> |
| 75 | #include <syslog.h> |
| 76 | #include <unistd.h> |
| 77 | #include <errno.h> |
| 78 | #include <sys/select.h> |
| 79 | #include <sys/wait.h> |
| 80 | #include <time.h> |
| 81 | #include <fcntl.h> |
| 82 | #include <poll.h> |
| 83 | #include <sys/un.h> |
| 84 | #include <sys/stat.h> |
| 85 | #include <pthread.h> |
| 86 | #include <sys/resource.h> |
| 87 | #include <gcrypt.h> |
| 88 | |
| 89 | #include "configuration.h" |
| 90 | #include "syscalls.h" |
| 91 | #include "log.h" |
| 92 | #include "defs.h" |
| 93 | #include "mem.h" |
| 94 | #include "speaker-protocol.h" |
| 95 | #include "user.h" |
| 96 | #include "printf.h" |
| 97 | #include "version.h" |
| 98 | #include "uaudio.h" |
| 99 | |
| 100 | /** @brief Maximum number of FDs to poll for */ |
| 101 | #define NFDS 1024 |
| 102 | |
| 103 | /** @brief Number of bytes before end of track to send SM_FINISHED |
| 104 | * |
| 105 | * Generally set to 1 second. |
| 106 | */ |
| 107 | static size_t early_finish; |
| 108 | |
| 109 | /** @brief Track structure |
| 110 | * |
| 111 | * Known tracks are kept in a linked list. Usually there will be at most two |
| 112 | * of these but rearranging the queue can cause there to be more. |
| 113 | */ |
| 114 | struct track { |
| 115 | /** @brief Next track */ |
| 116 | struct track *next; |
| 117 | |
| 118 | /** @brief Input file descriptor */ |
| 119 | int fd; |
| 120 | |
| 121 | /** @brief Track ID */ |
| 122 | char id[24]; |
| 123 | |
| 124 | /** @brief Start position of data in buffer */ |
| 125 | size_t start; |
| 126 | |
| 127 | /** @brief Number of bytes of data in buffer */ |
| 128 | size_t used; |
| 129 | |
| 130 | /** @brief Set @c fd is at EOF */ |
| 131 | int eof; |
| 132 | |
| 133 | /** @brief Total number of samples played */ |
| 134 | unsigned long long played; |
| 135 | |
| 136 | /** @brief Slot in @ref fds */ |
| 137 | int slot; |
| 138 | |
| 139 | /** @brief Set when playable |
| 140 | * |
| 141 | * A track becomes playable whenever it fills its buffer or reaches EOF; it |
| 142 | * stops being playable when it entirely empties its buffer. Tracks start |
| 143 | * out life not playable. |
| 144 | */ |
| 145 | int playable; |
| 146 | |
| 147 | /** @brief Set when finished |
| 148 | * |
| 149 | * This is set when we've notified the server that the track is finished. |
| 150 | * Once this has happened (typically very late in the track's lifetime) the |
| 151 | * track cannot be paused or cancelled. |
| 152 | */ |
| 153 | int finished; |
| 154 | |
| 155 | /** @brief Input buffer |
| 156 | * |
| 157 | * 1Mbyte is enough for nearly 6s of 44100Hz 16-bit stereo |
| 158 | */ |
| 159 | char buffer[1048576]; |
| 160 | }; |
| 161 | |
| 162 | /** @brief Lock protecting data structures |
| 163 | * |
| 164 | * This lock protects values shared between the main thread and the callback. |
| 165 | * |
| 166 | * It is held 'all' the time by the main thread, the exceptions being when |
| 167 | * called activate/deactivate callbacks and when calling (potentially) slow |
| 168 | * system calls (in particular poll(), where in fact the main thread will spend |
| 169 | * most of its time blocked). |
| 170 | * |
| 171 | * The callback holds it when it's running. |
| 172 | */ |
| 173 | static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER; |
| 174 | |
| 175 | /** @brief Linked list of all prepared tracks |
| 176 | * |
| 177 | * This includes @ref playing and @ref pending_playing. |
| 178 | */ |
| 179 | static struct track *tracks; |
| 180 | |
| 181 | /** @brief Playing track, or NULL |
| 182 | * |
| 183 | * This means the track the speaker process intends to play. It does not |
| 184 | * reflect any other state (e.g. activation of uaudio backend). |
| 185 | * |
| 186 | * This track remains on @ref track. |
| 187 | */ |
| 188 | static struct track *playing; |
| 189 | |
| 190 | /** @brief Pending playing track, or NULL |
| 191 | * |
| 192 | * This means the track the server wants the speaker to play. |
| 193 | * |
| 194 | * This track remains on @p track. |
| 195 | */ |
| 196 | static struct track *pending_playing; |
| 197 | |
| 198 | /** @brief Array of file descriptors for poll() */ |
| 199 | static struct pollfd fds[NFDS]; |
| 200 | |
| 201 | /** @brief Next free slot in @ref fds |
| 202 | * |
| 203 | * This is used when filling in the @ref fds array each iteration through the |
| 204 | * event loop. |
| 205 | */ |
| 206 | static int fdno; |
| 207 | |
| 208 | /** @brief Listen socket */ |
| 209 | static int listenfd; |
| 210 | |
| 211 | /** @brief Timestamp of last potential report to server */ |
| 212 | static time_t last_report; |
| 213 | |
| 214 | /** @brief Set when paused */ |
| 215 | static int paused; |
| 216 | |
| 217 | /** @brief Set when back end activated */ |
| 218 | static int activated; |
| 219 | |
| 220 | /** @brief Signal pipe back into the poll() loop */ |
| 221 | static int sigpipe[2]; |
| 222 | |
| 223 | /** @brief Selected backend */ |
| 224 | static const struct uaudio *backend; |
| 225 | |
| 226 | static const struct option options[] = { |
| 227 | { "help", no_argument, 0, 'h' }, |
| 228 | { "version", no_argument, 0, 'V' }, |
| 229 | { "config", required_argument, 0, 'c' }, |
| 230 | { "debug", no_argument, 0, 'd' }, |
| 231 | { "no-debug", no_argument, 0, 'D' }, |
| 232 | { "syslog", no_argument, 0, 's' }, |
| 233 | { "no-syslog", no_argument, 0, 'S' }, |
| 234 | { 0, 0, 0, 0 } |
| 235 | }; |
| 236 | |
| 237 | /* Display usage message and terminate. */ |
| 238 | static void help(void) { |
| 239 | xprintf("Usage:\n" |
| 240 | " disorder-speaker [OPTIONS]\n" |
| 241 | "Options:\n" |
| 242 | " --help, -h Display usage message\n" |
| 243 | " --version, -V Display version number\n" |
| 244 | " --config PATH, -c PATH Set configuration file\n" |
| 245 | " --debug, -d Turn on debugging\n" |
| 246 | " --[no-]syslog Force logging\n" |
| 247 | "\n" |
| 248 | "Speaker process for DisOrder. Not intended to be run\n" |
| 249 | "directly.\n"); |
| 250 | xfclose(stdout); |
| 251 | exit(0); |
| 252 | } |
| 253 | |
| 254 | /** @brief Find track @p id, maybe creating it if not found |
| 255 | * @param id Track ID to find |
| 256 | * @param create If nonzero, create track structure of @p id if not found |
| 257 | * @return Pointer to track structure or NULL |
| 258 | */ |
| 259 | static struct track *findtrack(const char *id, int create) { |
| 260 | struct track *t; |
| 261 | |
| 262 | D(("findtrack %s %d", id, create)); |
| 263 | for(t = tracks; t && strcmp(id, t->id); t = t->next) |
| 264 | ; |
| 265 | if(!t && create) { |
| 266 | t = xmalloc(sizeof *t); |
| 267 | t->next = tracks; |
| 268 | strcpy(t->id, id); |
| 269 | t->fd = -1; |
| 270 | tracks = t; |
| 271 | } |
| 272 | return t; |
| 273 | } |
| 274 | |
| 275 | /** @brief Remove track @p id (but do not destroy it) |
| 276 | * @param id Track ID to remove |
| 277 | * @return Track structure or NULL if not found |
| 278 | */ |
| 279 | static struct track *removetrack(const char *id) { |
| 280 | struct track *t, **tt; |
| 281 | |
| 282 | D(("removetrack %s", id)); |
| 283 | for(tt = &tracks; (t = *tt) && strcmp(id, t->id); tt = &t->next) |
| 284 | ; |
| 285 | if(t) |
| 286 | *tt = t->next; |
| 287 | return t; |
| 288 | } |
| 289 | |
| 290 | /** @brief Destroy a track |
| 291 | * @param t Track structure |
| 292 | */ |
| 293 | static void destroy(struct track *t) { |
| 294 | D(("destroy %s", t->id)); |
| 295 | if(t->fd != -1) |
| 296 | xclose(t->fd); |
| 297 | free(t); |
| 298 | } |
| 299 | |
| 300 | /** @brief Read data into a sample buffer |
| 301 | * @param t Pointer to track |
| 302 | * @return 0 on success, -1 on EOF |
| 303 | * |
| 304 | * This is effectively the read callback on @c t->fd. It is called from the |
| 305 | * main loop whenever the track's file descriptor is readable, assuming the |
| 306 | * buffer has not reached the maximum allowed occupancy. |
| 307 | * |
| 308 | * Errors count as EOF. |
| 309 | */ |
| 310 | static int speaker_fill(struct track *t) { |
| 311 | size_t where, left; |
| 312 | int n, rc; |
| 313 | |
| 314 | D(("fill %s: eof=%d used=%zu", |
| 315 | t->id, t->eof, t->used)); |
| 316 | if(t->eof) |
| 317 | return -1; |
| 318 | if(t->used < sizeof t->buffer) { |
| 319 | /* there is room left in the buffer */ |
| 320 | where = (t->start + t->used) % sizeof t->buffer; |
| 321 | /* Get as much data as we can */ |
| 322 | if(where >= t->start) |
| 323 | left = (sizeof t->buffer) - where; |
| 324 | else |
| 325 | left = t->start - where; |
| 326 | pthread_mutex_unlock(&lock); |
| 327 | do { |
| 328 | n = read(t->fd, t->buffer + where, left); |
| 329 | } while(n < 0 && errno == EINTR); |
| 330 | pthread_mutex_lock(&lock); |
| 331 | if(n < 0 && errno == EAGAIN) { |
| 332 | /* EAGAIN means more later */ |
| 333 | rc = 0; |
| 334 | } else if(n <= 0) { |
| 335 | /* n=0 means EOF. n<0 means some error occurred. We log the error but |
| 336 | * otherwise treat it as identical to EOF. */ |
| 337 | if(n < 0) |
| 338 | disorder_error(errno, "error reading sample stream for %s", t->id); |
| 339 | else |
| 340 | D(("fill %s: eof detected", t->id)); |
| 341 | t->eof = 1; |
| 342 | /* A track always becomes playable at EOF; we're not going to see any |
| 343 | * more data. */ |
| 344 | t->playable = 1; |
| 345 | rc = -1; |
| 346 | } else { |
| 347 | t->used += n; |
| 348 | /* A track becomes playable when it (first) fills its buffer. For |
| 349 | * 44.1KHz 16-bit stereo this is ~6s of audio data. The latency will |
| 350 | * depend how long that takes to decode (hopefuly not very!) */ |
| 351 | if(t->used == sizeof t->buffer) |
| 352 | t->playable = 1; |
| 353 | rc = 0; |
| 354 | } |
| 355 | } else |
| 356 | rc = 0; |
| 357 | return rc; |
| 358 | } |
| 359 | |
| 360 | /** @brief Return nonzero if we want to play some audio |
| 361 | * |
| 362 | * We want to play audio if there is a current track; and it is not paused; and |
| 363 | * it is playable according to the rules for @ref track::playable. |
| 364 | * |
| 365 | * We don't allow tracks to be paused if we've already told the server we've |
| 366 | * finished them; that would cause such tracks to survive much longer than the |
| 367 | * few samples they're supposed to, with report() remaining silent for the |
| 368 | * duration. The effect is that if you hit pause towards the end of a track, |
| 369 | * what should happen is that it finished but the next one is paused right at |
| 370 | * its start. |
| 371 | */ |
| 372 | static int playable(void) { |
| 373 | return playing |
| 374 | && (!paused || playing->finished) |
| 375 | && playing->playable; |
| 376 | } |
| 377 | |
| 378 | /** @brief Notify the server what we're up to */ |
| 379 | static void report(void) { |
| 380 | struct speaker_message sm; |
| 381 | |
| 382 | if(playing) { |
| 383 | /* Had better not send a report for a track that the server thinks has |
| 384 | * finished, that would be confusing. */ |
| 385 | if(playing->finished) |
| 386 | return; |
| 387 | memset(&sm, 0, sizeof sm); |
| 388 | sm.type = paused ? SM_PAUSED : SM_PLAYING; |
| 389 | strcpy(sm.u.id, playing->id); |
| 390 | sm.data = playing->played / (uaudio_rate * uaudio_channels); |
| 391 | speaker_send(1, &sm); |
| 392 | xtime(&last_report); |
| 393 | } |
| 394 | } |
| 395 | |
| 396 | /** @brief Add a file descriptor to the set to poll() for |
| 397 | * @param fd File descriptor |
| 398 | * @param events Events to wait for e.g. @c POLLIN |
| 399 | * @return Slot number |
| 400 | */ |
| 401 | static int addfd(int fd, int events) { |
| 402 | if(fdno < NFDS) { |
| 403 | fds[fdno].fd = fd; |
| 404 | fds[fdno].events = events; |
| 405 | return fdno++; |
| 406 | } else |
| 407 | return -1; |
| 408 | } |
| 409 | |
| 410 | /** @brief Callback to return some sampled data |
| 411 | * @param buffer Where to put sample data |
| 412 | * @param max_samples How many samples to return |
| 413 | * @param userdata User data |
| 414 | * @return Number of samples written |
| 415 | * |
| 416 | * See uaudio_callback(). |
| 417 | */ |
| 418 | static size_t speaker_callback(void *buffer, |
| 419 | size_t max_samples, |
| 420 | void attribute((unused)) *userdata) { |
| 421 | size_t max_bytes = max_samples * uaudio_sample_size; |
| 422 | size_t provided_samples = 0; |
| 423 | |
| 424 | /* Be sure to keep the amount of data in a buffer a whole number of frames: |
| 425 | * otherwise the playing threads can become stuck. */ |
| 426 | max_bytes -= max_bytes % (uaudio_sample_size * uaudio_channels); |
| 427 | |
| 428 | pthread_mutex_lock(&lock); |
| 429 | /* TODO perhaps we should immediately go silent if we've been asked to pause |
| 430 | * or cancel the playing track (maybe block in the cancel case and see what |
| 431 | * else turns up?) */ |
| 432 | if(playing) { |
| 433 | if(playing->used > 0) { |
| 434 | size_t bytes; |
| 435 | /* Compute size of largest contiguous chunk. We get called as often as |
| 436 | * necessary so there's no need for cleverness here. */ |
| 437 | if(playing->start + playing->used > sizeof playing->buffer) |
| 438 | bytes = sizeof playing->buffer - playing->start; |
| 439 | else |
| 440 | bytes = playing->used; |
| 441 | /* Limit to what we were asked for */ |
| 442 | if(bytes > max_bytes) |
| 443 | bytes = max_bytes; |
| 444 | /* And truncate to a whole number of frames. */ |
| 445 | bytes -= bytes % (uaudio_sample_size * uaudio_channels); |
| 446 | /* Provide it */ |
| 447 | memcpy(buffer, playing->buffer + playing->start, bytes); |
| 448 | playing->start += bytes; |
| 449 | playing->used -= bytes; |
| 450 | /* Wrap around to start of buffer */ |
| 451 | if(playing->start == sizeof playing->buffer) |
| 452 | playing->start = 0; |
| 453 | /* See if we've reached the end of the track; if so make sure the event |
| 454 | * loop wakes up. */ |
| 455 | if(playing->used == 0 && playing->eof) { |
| 456 | int ignored = write(sigpipe[1], "", 1); |
| 457 | (void) ignored; |
| 458 | } |
| 459 | provided_samples = bytes / uaudio_sample_size; |
| 460 | playing->played += provided_samples; |
| 461 | } |
| 462 | } |
| 463 | /* If we couldn't provide anything at all, play dead air */ |
| 464 | /* TODO maybe it would be better to block, in some cases? */ |
| 465 | if(!provided_samples) { |
| 466 | memset(buffer, 0, max_bytes); |
| 467 | provided_samples = max_samples; |
| 468 | if(playing) |
| 469 | disorder_info("%zu samples silence, playing->used=%zu", |
| 470 | provided_samples, playing->used); |
| 471 | else |
| 472 | disorder_info("%zu samples silence, playing=NULL", provided_samples); |
| 473 | } |
| 474 | pthread_mutex_unlock(&lock); |
| 475 | return provided_samples; |
| 476 | } |
| 477 | |
| 478 | /** @brief Main event loop */ |
| 479 | static void mainloop(void) { |
| 480 | struct track *t; |
| 481 | struct speaker_message sm; |
| 482 | int n, fd, stdin_slot, timeout, listen_slot, sigpipe_slot; |
| 483 | |
| 484 | pthread_mutex_lock(&lock); |
| 485 | /* Keep going while our parent process is alive */ |
| 486 | while(getppid() != 1) { |
| 487 | int force_report = 0; |
| 488 | |
| 489 | fdno = 0; |
| 490 | /* By default we will wait up to half a second before thinking about |
| 491 | * current state. */ |
| 492 | timeout = 500; |
| 493 | /* Always ready for commands from the main server. */ |
| 494 | stdin_slot = addfd(0, POLLIN); |
| 495 | /* Also always ready for inbound connections */ |
| 496 | listen_slot = addfd(listenfd, POLLIN); |
| 497 | /* Try to read sample data for the currently playing track if there is |
| 498 | * buffer space. */ |
| 499 | if(playing |
| 500 | && playing->fd >= 0 |
| 501 | && !playing->eof |
| 502 | && playing->used < (sizeof playing->buffer)) |
| 503 | playing->slot = addfd(playing->fd, POLLIN); |
| 504 | else if(playing) |
| 505 | playing->slot = -1; |
| 506 | /* Allow the poll() to be interrupted at the end of a track */ |
| 507 | sigpipe_slot = addfd(sigpipe[0], POLLIN); |
| 508 | /* If any other tracks don't have a full buffer, try to read sample data |
| 509 | * from them. We do this last of all, so that if we run out of slots, |
| 510 | * nothing important can't be monitored. */ |
| 511 | for(t = tracks; t; t = t->next) |
| 512 | if(t != playing) { |
| 513 | if(t->fd >= 0 |
| 514 | && !t->eof |
| 515 | && t->used < sizeof t->buffer) { |
| 516 | t->slot = addfd(t->fd, POLLIN | POLLHUP); |
| 517 | } else |
| 518 | t->slot = -1; |
| 519 | } |
| 520 | /* Wait for something interesting to happen */ |
| 521 | pthread_mutex_unlock(&lock); |
| 522 | n = poll(fds, fdno, timeout); |
| 523 | pthread_mutex_lock(&lock); |
| 524 | if(n < 0) { |
| 525 | if(errno == EINTR) continue; |
| 526 | disorder_fatal(errno, "error calling poll"); |
| 527 | } |
| 528 | /* Perhaps a connection has arrived */ |
| 529 | if(fds[listen_slot].revents & POLLIN) { |
| 530 | struct sockaddr_un addr; |
| 531 | socklen_t addrlen = sizeof addr; |
| 532 | uint32_t l; |
| 533 | char id[24]; |
| 534 | |
| 535 | if((fd = accept(listenfd, (struct sockaddr *)&addr, &addrlen)) >= 0) { |
| 536 | /* We do blocking reads for the header. In theory this means that the |
| 537 | * connecting process could wedge the speaker indefinitely. In |
| 538 | * practice that would mean that the main server was broken anyway. |
| 539 | * Still, this is ugly, and a rewrite would be nice. */ |
| 540 | blocking(fd); |
| 541 | if(read(fd, &l, sizeof l) < 4) { |
| 542 | disorder_error(errno, "reading length from inbound connection"); |
| 543 | xclose(fd); |
| 544 | } else if(l >= sizeof id) { |
| 545 | disorder_error(0, "id length too long"); |
| 546 | xclose(fd); |
| 547 | } else if(read(fd, id, l) < (ssize_t)l) { |
| 548 | disorder_error(errno, "reading id from inbound connection"); |
| 549 | xclose(fd); |
| 550 | } else { |
| 551 | id[l] = 0; |
| 552 | D(("id %s fd %d", id, fd)); |
| 553 | t = findtrack(id, 1/*create*/); |
| 554 | if (write(fd, "", 1) < 0) /* write an ack */ |
| 555 | disorder_error(errno, "writing ack to inbound connection for %s", |
| 556 | id); |
| 557 | if(t->fd != -1) { |
| 558 | disorder_error(0, "%s: already got a connection", id); |
| 559 | xclose(fd); |
| 560 | } else { |
| 561 | nonblock(fd); |
| 562 | t->fd = fd; /* yay */ |
| 563 | } |
| 564 | /* Notify the server that the connection arrived */ |
| 565 | sm.type = SM_ARRIVED; |
| 566 | strcpy(sm.u.id, id); |
| 567 | speaker_send(1, &sm); |
| 568 | } |
| 569 | } else |
| 570 | disorder_error(errno, "accept"); |
| 571 | } |
| 572 | /* Perhaps we have a command to process */ |
| 573 | if(fds[stdin_slot].revents & POLLIN) { |
| 574 | /* There might (in theory) be several commands queued up, but in general |
| 575 | * this won't be the case, so we don't bother looping around to pick them |
| 576 | * all up. */ |
| 577 | n = speaker_recv(0, &sm); |
| 578 | if(n > 0) |
| 579 | /* As a rule we don't send success replies to most commands - we just |
| 580 | * force the regular status update to be sent immediately rather than |
| 581 | * on schedule. */ |
| 582 | switch(sm.type) { |
| 583 | case SM_PLAY: |
| 584 | /* SM_PLAY is only allowed if the server reasonably believes that |
| 585 | * nothing is playing */ |
| 586 | if(playing) { |
| 587 | /* If finished isn't set then the server can't believe that this |
| 588 | * track has finished */ |
| 589 | if(!playing->finished) |
| 590 | disorder_fatal(0, "got SM_PLAY but already playing something"); |
| 591 | /* If pending_playing is set then the server must believe that that |
| 592 | * is playing */ |
| 593 | if(pending_playing) |
| 594 | disorder_fatal(0, "got SM_PLAY but have a pending playing track"); |
| 595 | } |
| 596 | t = findtrack(sm.u.id, 1); |
| 597 | D(("SM_PLAY %s fd %d", t->id, t->fd)); |
| 598 | if(t->fd == -1) |
| 599 | disorder_error(0, |
| 600 | "cannot play track because no connection arrived"); |
| 601 | /* TODO as things stand we often report this error message but then |
| 602 | * appear to proceed successfully. Understanding why requires a look |
| 603 | * at play.c: we call prepare() which makes the connection in a child |
| 604 | * process, and then sends the SM_PLAY in the parent process. The |
| 605 | * latter may well be faster. As it happens this is harmless; we'll |
| 606 | * just sit around sending silence until the decoder connects and |
| 607 | * starts sending some sample data. But is is annoying and ought to |
| 608 | * be fixed. */ |
| 609 | pending_playing = t; |
| 610 | /* If nothing is currently playing then we'll switch to the pending |
| 611 | * track below so there's no point distinguishing the situations |
| 612 | * here. */ |
| 613 | break; |
| 614 | case SM_PAUSE: |
| 615 | D(("SM_PAUSE")); |
| 616 | paused = 1; |
| 617 | force_report = 1; |
| 618 | break; |
| 619 | case SM_RESUME: |
| 620 | D(("SM_RESUME")); |
| 621 | paused = 0; |
| 622 | force_report = 1; |
| 623 | break; |
| 624 | case SM_CANCEL: |
| 625 | D(("SM_CANCEL %s", sm.u.id)); |
| 626 | t = removetrack(sm.u.id); |
| 627 | if(t) { |
| 628 | if(t == playing || t == pending_playing) { |
| 629 | /* Scratching the track that the server believes is playing, |
| 630 | * which might either be the actual playing track or a pending |
| 631 | * playing track */ |
| 632 | sm.type = SM_FINISHED; |
| 633 | if(t == playing) |
| 634 | playing = 0; |
| 635 | else |
| 636 | pending_playing = 0; |
| 637 | } else { |
| 638 | /* Could be scratching the playing track before it's quite got |
| 639 | * going, or could be just removing a track from the queue. We |
| 640 | * log more because there's been a bug here recently than because |
| 641 | * it's particularly interesting; the log message will be removed |
| 642 | * if no further problems show up. */ |
| 643 | disorder_info("SM_CANCEL for nonplaying track %s", sm.u.id); |
| 644 | sm.type = SM_STILLBORN; |
| 645 | } |
| 646 | strcpy(sm.u.id, t->id); |
| 647 | destroy(t); |
| 648 | } else { |
| 649 | /* Probably scratching the playing track well before it's got |
| 650 | * going, but could indicate a bug, so we log this as an error. */ |
| 651 | sm.type = SM_UNKNOWN; |
| 652 | disorder_error(0, "SM_CANCEL for unknown track %s", sm.u.id); |
| 653 | } |
| 654 | speaker_send(1, &sm); |
| 655 | force_report = 1; |
| 656 | break; |
| 657 | case SM_RELOAD: |
| 658 | D(("SM_RELOAD")); |
| 659 | if(config_read(1, NULL)) |
| 660 | disorder_error(0, "cannot read configuration"); |
| 661 | disorder_info("reloaded configuration"); |
| 662 | break; |
| 663 | case SM_RTP_REQUEST: |
| 664 | /* TODO the error behavior here is really unhelpful */ |
| 665 | if(rtp_add_recipient(&sm.u.address)) |
| 666 | disorder_error(0, "unacceptable RTP destination"); |
| 667 | break; |
| 668 | case SM_RTP_CANCEL: |
| 669 | if(rtp_remove_recipient(&sm.u.address)) |
| 670 | disorder_error(0, "unacceptable RTP destination for removal"); |
| 671 | break; |
| 672 | default: |
| 673 | disorder_error(0, "unknown message type %d", sm.type); |
| 674 | } |
| 675 | } |
| 676 | /* Read in any buffered data */ |
| 677 | for(t = tracks; t; t = t->next) |
| 678 | if(t->fd != -1 |
| 679 | && t->slot != -1 |
| 680 | && (fds[t->slot].revents & (POLLIN | POLLHUP))) |
| 681 | speaker_fill(t); |
| 682 | /* Drain the signal pipe. We don't care about its contents, merely that it |
| 683 | * interrupted poll(). */ |
| 684 | if(fds[sigpipe_slot].revents & POLLIN) { |
| 685 | char buffer[64]; |
| 686 | int ignored; (void)ignored; |
| 687 | |
| 688 | ignored = read(sigpipe[0], buffer, sizeof buffer); |
| 689 | } |
| 690 | /* Send SM_FINISHED when we're near the end of the track. |
| 691 | * |
| 692 | * This is how we implement gapless play; we hope that the SM_PLAY from the |
| 693 | * server arrives before the remaining bytes of the track play out. |
| 694 | */ |
| 695 | if(playing |
| 696 | && playing->eof |
| 697 | && !playing->finished |
| 698 | && playing->used <= early_finish) { |
| 699 | memset(&sm, 0, sizeof sm); |
| 700 | sm.type = SM_FINISHED; |
| 701 | strcpy(sm.u.id, playing->id); |
| 702 | speaker_send(1, &sm); |
| 703 | playing->finished = 1; |
| 704 | } |
| 705 | /* When the track is actually finished, deconfigure it */ |
| 706 | if(playing && playing->eof && !playing->used) { |
| 707 | if(!playing->finished) { |
| 708 | /* should never happen but we'd like to know if it does */ |
| 709 | disorder_fatal(0, "track finish state inconsistent"); |
| 710 | } |
| 711 | removetrack(playing->id); |
| 712 | destroy(playing); |
| 713 | playing = 0; |
| 714 | } |
| 715 | /* Act on the pending SM_PLAY */ |
| 716 | if(!playing && pending_playing) { |
| 717 | playing = pending_playing; |
| 718 | pending_playing = 0; |
| 719 | force_report = 1; |
| 720 | } |
| 721 | /* Impose any state change required by the above */ |
| 722 | if(playable()) { |
| 723 | if(!activated) { |
| 724 | activated = 1; |
| 725 | pthread_mutex_unlock(&lock); |
| 726 | backend->activate(); |
| 727 | pthread_mutex_lock(&lock); |
| 728 | } |
| 729 | } else { |
| 730 | if(activated) { |
| 731 | activated = 0; |
| 732 | pthread_mutex_unlock(&lock); |
| 733 | backend->deactivate(); |
| 734 | pthread_mutex_lock(&lock); |
| 735 | } |
| 736 | } |
| 737 | /* If we've not reported our state for a second do so now. */ |
| 738 | if(force_report || xtime(0) > last_report) |
| 739 | report(); |
| 740 | } |
| 741 | } |
| 742 | |
| 743 | int main(int argc, char **argv) { |
| 744 | int n, logsyslog = !isatty(2); |
| 745 | struct sockaddr_un addr; |
| 746 | static const int one = 1; |
| 747 | struct speaker_message sm; |
| 748 | const char *d; |
| 749 | char *dir; |
| 750 | struct rlimit rl[1]; |
| 751 | |
| 752 | set_progname(argv); |
| 753 | if(!setlocale(LC_CTYPE, "")) disorder_fatal(errno, "error calling setlocale"); |
| 754 | while((n = getopt_long(argc, argv, "hVc:dDSs", options, 0)) >= 0) { |
| 755 | switch(n) { |
| 756 | case 'h': help(); |
| 757 | case 'V': version("disorder-speaker"); |
| 758 | case 'c': configfile = optarg; break; |
| 759 | case 'd': debugging = 1; break; |
| 760 | case 'D': debugging = 0; break; |
| 761 | case 'S': logsyslog = 0; break; |
| 762 | case 's': logsyslog = 1; break; |
| 763 | default: disorder_fatal(0, "invalid option"); |
| 764 | } |
| 765 | } |
| 766 | if((d = getenv("DISORDER_DEBUG_SPEAKER"))) debugging = atoi(d); |
| 767 | if(logsyslog) { |
| 768 | openlog(progname, LOG_PID, LOG_DAEMON); |
| 769 | log_default = &log_syslog; |
| 770 | } |
| 771 | config_uaudio_apis = uaudio_apis; |
| 772 | if(config_read(1, NULL)) disorder_fatal(0, "cannot read configuration"); |
| 773 | /* ignore SIGPIPE */ |
| 774 | signal(SIGPIPE, SIG_IGN); |
| 775 | /* set nice value */ |
| 776 | xnice(config->nice_speaker); |
| 777 | /* change user */ |
| 778 | become_mortal(); |
| 779 | /* make sure we're not root, whatever the config says */ |
| 780 | if(getuid() == 0 || geteuid() == 0) |
| 781 | disorder_fatal(0, "do not run as root"); |
| 782 | /* Make sure we can't have more than NFDS files open (it would bust our |
| 783 | * poll() array) */ |
| 784 | if(getrlimit(RLIMIT_NOFILE, rl) < 0) |
| 785 | disorder_fatal(errno, "getrlimit RLIMIT_NOFILE"); |
| 786 | if(rl->rlim_cur > NFDS) { |
| 787 | rl->rlim_cur = NFDS; |
| 788 | if(setrlimit(RLIMIT_NOFILE, rl) < 0) |
| 789 | disorder_fatal(errno, "setrlimit to reduce RLIMIT_NOFILE to %lu", |
| 790 | (unsigned long)rl->rlim_cur); |
| 791 | disorder_info("set RLIM_NOFILE to %lu", (unsigned long)rl->rlim_cur); |
| 792 | } else |
| 793 | disorder_info("RLIM_NOFILE is %lu", (unsigned long)rl->rlim_cur); |
| 794 | /* gcrypt initialization */ |
| 795 | if(!gcry_check_version(NULL)) |
| 796 | disorder_fatal(0, "gcry_check_version failed"); |
| 797 | gcry_control(GCRYCTL_INIT_SECMEM, 0); |
| 798 | gcry_control (GCRYCTL_INITIALIZATION_FINISHED, 0); |
| 799 | /* create a pipe between the backend callback and the poll() loop */ |
| 800 | xpipe(sigpipe); |
| 801 | nonblock(sigpipe[0]); |
| 802 | /* set up audio backend */ |
| 803 | uaudio_set_format(config->sample_format.rate, |
| 804 | config->sample_format.channels, |
| 805 | config->sample_format.bits, |
| 806 | config->sample_format.bits != 8); |
| 807 | early_finish = uaudio_sample_size * uaudio_channels * uaudio_rate; |
| 808 | /* TODO other parameters! */ |
| 809 | backend = uaudio_find(config->api); |
| 810 | /* backend-specific initialization */ |
| 811 | if(backend->configure) |
| 812 | backend->configure(); |
| 813 | uaudio_set("application", "disorder-speaker"); |
| 814 | backend->start(speaker_callback, NULL); |
| 815 | /* create the private socket directory */ |
| 816 | byte_xasprintf(&dir, "%s/private", config->home); |
| 817 | unlink(dir); /* might be a leftover socket */ |
| 818 | if(mkdir(dir, 0700) < 0 && errno != EEXIST) |
| 819 | disorder_fatal(errno, "error creating %s", dir); |
| 820 | /* set up the listen socket */ |
| 821 | listenfd = xsocket(PF_UNIX, SOCK_STREAM, 0); |
| 822 | memset(&addr, 0, sizeof addr); |
| 823 | addr.sun_family = AF_UNIX; |
| 824 | snprintf(addr.sun_path, sizeof addr.sun_path, "%s/private/speaker", |
| 825 | config->home); |
| 826 | if(unlink(addr.sun_path) < 0 && errno != ENOENT) |
| 827 | disorder_error(errno, "removing %s", addr.sun_path); |
| 828 | xsetsockopt(listenfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one); |
| 829 | if(bind(listenfd, (const struct sockaddr *)&addr, sizeof addr) < 0) |
| 830 | disorder_fatal(errno, "error binding socket to %s", addr.sun_path); |
| 831 | xlisten(listenfd, 128); |
| 832 | nonblock(listenfd); |
| 833 | disorder_info("version "VERSION" process ID %lu", |
| 834 | (unsigned long)getpid()); |
| 835 | disorder_info("listening on %s", addr.sun_path); |
| 836 | memset(&sm, 0, sizeof sm); |
| 837 | sm.type = SM_READY; |
| 838 | speaker_send(1, &sm); |
| 839 | mainloop(); |
| 840 | disorder_info("stopped (parent terminated)"); |
| 841 | exit(0); |
| 842 | } |
| 843 | |
| 844 | /* |
| 845 | Local Variables: |
| 846 | c-basic-offset:2 |
| 847 | comment-column:40 |
| 848 | fill-column:79 |
| 849 | indent-tabs-mode:nil |
| 850 | End: |
| 851 | */ |