put speaker socket in its own private directory
[disorder] / server / speaker.c
1 /*
2 * This file is part of DisOrder
3 * Copyright (C) 2005, 2006, 2007 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 2 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, but
12 * WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * 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, write to the Free Software
18 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
19 * USA
20 */
21 /** @file server/speaker.c
22 * @brief Speaker process
23 *
24 * This program is responsible for transmitting a single coherent audio stream
25 * to its destination (over the network, to some sound API, to some
26 * subprocess). It receives connections from decoders (or rather from the
27 * process that is about to become disorder-normalize) and plays them in the
28 * right order.
29 *
30 * @b Encodings. For the <a href="http://www.alsa-project.org/">ALSA</a> API,
31 * 8- and 16- bit stereo and mono are supported, with any sample rate (within
32 * the limits that ALSA can deal with.)
33 *
34 * Inbound data is expected to match @c config->sample_format. In normal use
35 * this is arranged by the @c disorder-normalize program (see @ref
36 * server/normalize.c).
37 *
38 * @b Garbage @b Collection. This program deliberately does not use the
39 * garbage collector even though it might be convenient to do so. This is for
40 * two reasons. Firstly some sound APIs use thread threads and we do not want
41 * to have to deal with potential interactions between threading and garbage
42 * collection. Secondly this process needs to be able to respond quickly and
43 * this is not compatible with the collector hanging the program even
44 * relatively briefly.
45 *
46 * @b Units. This program thinks at various times in three different units.
47 * Bytes are obvious. A sample is a single sample on a single channel. A
48 * frame is several samples on different channels at the same point in time.
49 * So (for instance) a 16-bit stereo frame is 4 bytes and consists of a pair of
50 * 2-byte samples.
51 */
52
53 #include <config.h>
54 #include "types.h"
55
56 #include <getopt.h>
57 #include <stdio.h>
58 #include <stdlib.h>
59 #include <locale.h>
60 #include <syslog.h>
61 #include <unistd.h>
62 #include <errno.h>
63 #include <ao/ao.h>
64 #include <string.h>
65 #include <assert.h>
66 #include <sys/select.h>
67 #include <sys/wait.h>
68 #include <time.h>
69 #include <fcntl.h>
70 #include <poll.h>
71 #include <sys/un.h>
72
73 #include "configuration.h"
74 #include "syscalls.h"
75 #include "log.h"
76 #include "defs.h"
77 #include "mem.h"
78 #include "speaker-protocol.h"
79 #include "user.h"
80 #include "speaker.h"
81 #include "printf.h"
82
83 /** @brief Linked list of all prepared tracks */
84 struct track *tracks;
85
86 /** @brief Playing track, or NULL */
87 struct track *playing;
88
89 /** @brief Number of bytes pre frame */
90 size_t bpf;
91
92 /** @brief Array of file descriptors for poll() */
93 struct pollfd fds[NFDS];
94
95 /** @brief Next free slot in @ref fds */
96 int fdno;
97
98 /** @brief Listen socket */
99 static int listenfd;
100
101 static time_t last_report; /* when we last reported */
102 static int paused; /* pause status */
103
104 /** @brief The current device state */
105 enum device_states device_state;
106
107 /** @brief Set when idled
108 *
109 * This is set when the sound device is deliberately closed by idle().
110 */
111 int idled;
112
113 /** @brief Selected backend */
114 static const struct speaker_backend *backend;
115
116 static const struct option options[] = {
117 { "help", no_argument, 0, 'h' },
118 { "version", no_argument, 0, 'V' },
119 { "config", required_argument, 0, 'c' },
120 { "debug", no_argument, 0, 'd' },
121 { "no-debug", no_argument, 0, 'D' },
122 { "syslog", no_argument, 0, 's' },
123 { "no-syslog", no_argument, 0, 'S' },
124 { 0, 0, 0, 0 }
125 };
126
127 /* Display usage message and terminate. */
128 static void help(void) {
129 xprintf("Usage:\n"
130 " disorder-speaker [OPTIONS]\n"
131 "Options:\n"
132 " --help, -h Display usage message\n"
133 " --version, -V Display version number\n"
134 " --config PATH, -c PATH Set configuration file\n"
135 " --debug, -d Turn on debugging\n"
136 " --[no-]syslog Force logging\n"
137 "\n"
138 "Speaker process for DisOrder. Not intended to be run\n"
139 "directly.\n");
140 xfclose(stdout);
141 exit(0);
142 }
143
144 /* Display version number and terminate. */
145 static void version(void) {
146 xprintf("%s", disorder_version_string);
147 xfclose(stdout);
148 exit(0);
149 }
150
151 /** @brief Return the number of bytes per frame in @p format */
152 static size_t bytes_per_frame(const struct stream_header *format) {
153 return format->channels * format->bits / 8;
154 }
155
156 /** @brief Find track @p id, maybe creating it if not found */
157 static struct track *findtrack(const char *id, int create) {
158 struct track *t;
159
160 D(("findtrack %s %d", id, create));
161 for(t = tracks; t && strcmp(id, t->id); t = t->next)
162 ;
163 if(!t && create) {
164 t = xmalloc(sizeof *t);
165 t->next = tracks;
166 strcpy(t->id, id);
167 t->fd = -1;
168 tracks = t;
169 }
170 return t;
171 }
172
173 /** @brief Remove track @p id (but do not destroy it) */
174 static struct track *removetrack(const char *id) {
175 struct track *t, **tt;
176
177 D(("removetrack %s", id));
178 for(tt = &tracks; (t = *tt) && strcmp(id, t->id); tt = &t->next)
179 ;
180 if(t)
181 *tt = t->next;
182 return t;
183 }
184
185 /** @brief Destroy a track */
186 static void destroy(struct track *t) {
187 D(("destroy %s", t->id));
188 if(t->fd != -1) xclose(t->fd);
189 free(t);
190 }
191
192 /** @brief Read data into a sample buffer
193 * @param t Pointer to track
194 * @return 0 on success, -1 on EOF
195 *
196 * This is effectively the read callback on @c t->fd. It is called from the
197 * main loop whenever the track's file descriptor is readable, assuming the
198 * buffer has not reached the maximum allowed occupancy.
199 */
200 static int speaker_fill(struct track *t) {
201 size_t where, left;
202 int n;
203
204 D(("fill %s: eof=%d used=%zu",
205 t->id, t->eof, t->used));
206 if(t->eof) return -1;
207 if(t->used < sizeof t->buffer) {
208 /* there is room left in the buffer */
209 where = (t->start + t->used) % sizeof t->buffer;
210 /* Get as much data as we can */
211 if(where >= t->start) left = (sizeof t->buffer) - where;
212 else left = t->start - where;
213 do {
214 n = read(t->fd, t->buffer + where, left);
215 } while(n < 0 && errno == EINTR);
216 if(n < 0) {
217 if(errno != EAGAIN) fatal(errno, "error reading sample stream");
218 return 0;
219 }
220 if(n == 0) {
221 D(("fill %s: eof detected", t->id));
222 t->eof = 1;
223 t->playable = 1;
224 return -1;
225 }
226 t->used += n;
227 if(t->used == sizeof t->buffer)
228 t->playable = 1;
229 }
230 return 0;
231 }
232
233 /** @brief Close the sound device
234 *
235 * This is called to deactivate the output device when pausing, and also by the
236 * ALSA backend when changing encoding (in which case the sound device will be
237 * immediately reactivated).
238 */
239 static void idle(void) {
240 D(("idle"));
241 if(backend->deactivate)
242 backend->deactivate();
243 else
244 device_state = device_closed;
245 idled = 1;
246 }
247
248 /** @brief Abandon the current track */
249 void abandon(void) {
250 struct speaker_message sm;
251
252 D(("abandon"));
253 memset(&sm, 0, sizeof sm);
254 sm.type = SM_FINISHED;
255 strcpy(sm.id, playing->id);
256 speaker_send(1, &sm);
257 removetrack(playing->id);
258 destroy(playing);
259 playing = 0;
260 }
261
262 /** @brief Enable sound output
263 *
264 * Makes sure the sound device is open and has the right sample format. Return
265 * 0 on success and -1 on error.
266 */
267 static void activate(void) {
268 if(backend->activate)
269 backend->activate();
270 else
271 device_state = device_open;
272 }
273
274 /** @brief Check whether the current track has finished
275 *
276 * The current track is determined to have finished either if the input stream
277 * eded before the format could be determined (i.e. it is malformed) or the
278 * input is at end of file and there is less than a frame left unplayed. (So
279 * it copes with decoders that crash mid-frame.)
280 */
281 static void maybe_finished(void) {
282 if(playing
283 && playing->eof
284 && playing->used < bytes_per_frame(&config->sample_format))
285 abandon();
286 }
287
288 /** @brief Return nonzero if we want to play some audio
289 *
290 * We want to play audio if there is a current track; and it is not paused; and
291 * it is playable according to the rules for @ref track::playable.
292 */
293 static int playable(void) {
294 return playing
295 && !paused
296 && playing->playable;
297 }
298
299 /** @brief Play up to @p frames frames of audio
300 *
301 * It is always safe to call this function.
302 * - If @ref playing is 0 then it will just return
303 * - If @ref paused is non-0 then it will just return
304 * - If @ref device_state != @ref device_open then it will call activate() and
305 * return if it it fails.
306 * - If there is not enough audio to play then it play what is available.
307 *
308 * If there are not enough frames to play then whatever is available is played
309 * instead. It is up to mainloop() to ensure that speaker_play() is not called
310 * when unreasonably only an small amounts of data is available to play.
311 */
312 static void speaker_play(size_t frames) {
313 size_t avail_frames, avail_bytes, written_frames;
314 ssize_t written_bytes;
315
316 /* Make sure there's a track to play and it is not paused */
317 if(!playable())
318 return;
319 /* Make sure the output device is open */
320 if(device_state != device_open) {
321 activate();
322 if(device_state != device_open)
323 return;
324 }
325 D(("play: play %zu/%zu%s %dHz %db %dc", frames, playing->used / bpf,
326 playing->eof ? " EOF" : "",
327 config->sample_format.rate,
328 config->sample_format.bits,
329 config->sample_format.channels));
330 /* Figure out how many frames there are available to write */
331 if(playing->start + playing->used > sizeof playing->buffer)
332 /* The ring buffer is currently wrapped, only play up to the wrap point */
333 avail_bytes = (sizeof playing->buffer) - playing->start;
334 else
335 /* The ring buffer is not wrapped, can play the lot */
336 avail_bytes = playing->used;
337 avail_frames = avail_bytes / bpf;
338 /* Only play up to the requested amount */
339 if(avail_frames > frames)
340 avail_frames = frames;
341 if(!avail_frames)
342 return;
343 /* Play it, Sam */
344 written_frames = backend->play(avail_frames);
345 written_bytes = written_frames * bpf;
346 /* written_bytes and written_frames had better both be set and correct by
347 * this point */
348 playing->start += written_bytes;
349 playing->used -= written_bytes;
350 playing->played += written_frames;
351 /* If the pointer is at the end of the buffer (or the buffer is completely
352 * empty) wrap it back to the start. */
353 if(!playing->used || playing->start == (sizeof playing->buffer))
354 playing->start = 0;
355 /* If the buffer emptied out mark the track as unplayably */
356 if(!playing->used && !playing->eof) {
357 error(0, "track buffer emptied");
358 playing->playable = 0;
359 }
360 frames -= written_frames;
361 return;
362 }
363
364 /* Notify the server what we're up to. */
365 static void report(void) {
366 struct speaker_message sm;
367
368 if(playing) {
369 memset(&sm, 0, sizeof sm);
370 sm.type = paused ? SM_PAUSED : SM_PLAYING;
371 strcpy(sm.id, playing->id);
372 sm.data = playing->played / config->sample_format.rate;
373 speaker_send(1, &sm);
374 }
375 time(&last_report);
376 }
377
378 static void reap(int __attribute__((unused)) sig) {
379 pid_t cmdpid;
380 int st;
381
382 do
383 cmdpid = waitpid(-1, &st, WNOHANG);
384 while(cmdpid > 0);
385 signal(SIGCHLD, reap);
386 }
387
388 int addfd(int fd, int events) {
389 if(fdno < NFDS) {
390 fds[fdno].fd = fd;
391 fds[fdno].events = events;
392 return fdno++;
393 } else
394 return -1;
395 }
396
397 /** @brief Table of speaker backends */
398 static const struct speaker_backend *backends[] = {
399 #if HAVE_ALSA_ASOUNDLIB_H
400 &alsa_backend,
401 #endif
402 &command_backend,
403 &network_backend,
404 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
405 &coreaudio_backend,
406 #endif
407 #if HAVE_SYS_SOUNDCARD_H
408 &oss_backend,
409 #endif
410 0
411 };
412
413 /** @brief Main event loop */
414 static void mainloop(void) {
415 struct track *t;
416 struct speaker_message sm;
417 int n, fd, stdin_slot, timeout, listen_slot;
418
419 while(getppid() != 1) {
420 fdno = 0;
421 /* By default we will wait up to a second before thinking about current
422 * state. */
423 timeout = 1000;
424 /* Always ready for commands from the main server. */
425 stdin_slot = addfd(0, POLLIN);
426 /* Also always ready for inbound connections */
427 listen_slot = addfd(listenfd, POLLIN);
428 /* Try to read sample data for the currently playing track if there is
429 * buffer space. */
430 if(playing
431 && playing->fd >= 0
432 && !playing->eof
433 && playing->used < (sizeof playing->buffer))
434 playing->slot = addfd(playing->fd, POLLIN);
435 else if(playing)
436 playing->slot = -1;
437 if(playable()) {
438 /* We want to play some audio. If the device is closed then we attempt
439 * to open it. */
440 if(device_state == device_closed)
441 activate();
442 /* If the device is (now) open then we will wait up until it is ready for
443 * more. If something went wrong then we should have device_error
444 * instead, but the post-poll code will cope even if it's
445 * device_closed. */
446 if(device_state == device_open)
447 backend->beforepoll(&timeout);
448 }
449 /* If any other tracks don't have a full buffer, try to read sample data
450 * from them. We do this last of all, so that if we run out of slots,
451 * nothing important can't be monitored. */
452 for(t = tracks; t; t = t->next)
453 if(t != playing) {
454 if(t->fd >= 0
455 && !t->eof
456 && t->used < sizeof t->buffer) {
457 t->slot = addfd(t->fd, POLLIN | POLLHUP);
458 } else
459 t->slot = -1;
460 }
461 /* Wait for something interesting to happen */
462 n = poll(fds, fdno, timeout);
463 if(n < 0) {
464 if(errno == EINTR) continue;
465 fatal(errno, "error calling poll");
466 }
467 /* Play some sound before doing anything else */
468 if(playable()) {
469 /* We want to play some audio */
470 if(device_state == device_open) {
471 if(backend->ready())
472 speaker_play(3 * FRAMES);
473 } else {
474 /* We must be in _closed or _error, and it should be the latter, but we
475 * cope with either.
476 *
477 * We most likely timed out, so now is a good time to retry.
478 * speaker_play() knows to re-activate the device if necessary.
479 */
480 speaker_play(3 * FRAMES);
481 }
482 }
483 /* Perhaps a connection has arrived */
484 if(fds[listen_slot].revents & POLLIN) {
485 struct sockaddr_un addr;
486 socklen_t addrlen = sizeof addr;
487 uint32_t l;
488 char id[24];
489
490 if((fd = accept(listenfd, (struct sockaddr *)&addr, &addrlen)) >= 0) {
491 blocking(fd);
492 if(read(fd, &l, sizeof l) < 4) {
493 error(errno, "reading length from inbound connection");
494 xclose(fd);
495 } else if(l >= sizeof id) {
496 error(0, "id length too long");
497 xclose(fd);
498 } else if(read(fd, id, l) < (ssize_t)l) {
499 error(errno, "reading id from inbound connection");
500 xclose(fd);
501 } else {
502 id[l] = 0;
503 D(("id %s fd %d", id, fd));
504 t = findtrack(id, 1/*create*/);
505 write(fd, "", 1); /* write an ack */
506 if(t->fd != -1) {
507 error(0, "%s: already got a connection", id);
508 xclose(fd);
509 } else {
510 nonblock(fd);
511 t->fd = fd; /* yay */
512 }
513 }
514 } else
515 error(errno, "accept");
516 }
517 /* Perhaps we have a command to process */
518 if(fds[stdin_slot].revents & POLLIN) {
519 /* There might (in theory) be several commands queued up, but in general
520 * this won't be the case, so we don't bother looping around to pick them
521 * all up. */
522 n = speaker_recv(0, &sm);
523 /* TODO */
524 if(n > 0)
525 switch(sm.type) {
526 case SM_PLAY:
527 if(playing) fatal(0, "got SM_PLAY but already playing something");
528 t = findtrack(sm.id, 1);
529 D(("SM_PLAY %s fd %d", t->id, t->fd));
530 if(t->fd == -1)
531 error(0, "cannot play track because no connection arrived");
532 playing = t;
533 /* We attempt to play straight away rather than going round the loop.
534 * speaker_play() is clever enough to perform any activation that is
535 * required. */
536 speaker_play(3 * FRAMES);
537 report();
538 break;
539 case SM_PAUSE:
540 D(("SM_PAUSE"));
541 paused = 1;
542 report();
543 break;
544 case SM_RESUME:
545 D(("SM_RESUME"));
546 if(paused) {
547 paused = 0;
548 /* As for SM_PLAY we attempt to play straight away. */
549 if(playing)
550 speaker_play(3 * FRAMES);
551 }
552 report();
553 break;
554 case SM_CANCEL:
555 D(("SM_CANCEL %s", sm.id));
556 t = removetrack(sm.id);
557 if(t) {
558 if(t == playing) {
559 sm.type = SM_FINISHED;
560 strcpy(sm.id, playing->id);
561 speaker_send(1, &sm);
562 playing = 0;
563 }
564 destroy(t);
565 } else
566 error(0, "SM_CANCEL for unknown track %s", sm.id);
567 report();
568 break;
569 case SM_RELOAD:
570 D(("SM_RELOAD"));
571 if(config_read(1)) error(0, "cannot read configuration");
572 info("reloaded configuration");
573 break;
574 default:
575 error(0, "unknown message type %d", sm.type);
576 }
577 }
578 /* Read in any buffered data */
579 for(t = tracks; t; t = t->next)
580 if(t->fd != -1
581 && t->slot != -1
582 && (fds[t->slot].revents & (POLLIN | POLLHUP)))
583 speaker_fill(t);
584 /* Maybe we finished playing a track somewhere in the above */
585 maybe_finished();
586 /* If we don't need the sound device for now then close it for the benefit
587 * of anyone else who wants it. */
588 if((!playing || paused) && device_state == device_open)
589 idle();
590 /* If we've not reported out state for a second do so now. */
591 if(time(0) > last_report)
592 report();
593 }
594 }
595
596 int main(int argc, char **argv) {
597 int n, logsyslog = !isatty(2);
598 struct sockaddr_un addr;
599 static const int one = 1;
600 struct speaker_message sm;
601 const char *d;
602 char *dir;
603
604 set_progname(argv);
605 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
606 while((n = getopt_long(argc, argv, "hVc:dDSs", options, 0)) >= 0) {
607 switch(n) {
608 case 'h': help();
609 case 'V': version();
610 case 'c': configfile = optarg; break;
611 case 'd': debugging = 1; break;
612 case 'D': debugging = 0; break;
613 case 'S': logsyslog = 0; break;
614 case 's': logsyslog = 1; break;
615 default: fatal(0, "invalid option");
616 }
617 }
618 if((d = getenv("DISORDER_DEBUG_SPEAKER"))) debugging = atoi(d);
619 if(logsyslog) {
620 openlog(progname, LOG_PID, LOG_DAEMON);
621 log_default = &log_syslog;
622 }
623 if(config_read(1)) fatal(0, "cannot read configuration");
624 bpf = bytes_per_frame(&config->sample_format);
625 /* ignore SIGPIPE */
626 signal(SIGPIPE, SIG_IGN);
627 /* reap kids */
628 signal(SIGCHLD, reap);
629 /* set nice value */
630 xnice(config->nice_speaker);
631 /* change user */
632 become_mortal();
633 /* make sure we're not root, whatever the config says */
634 if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root");
635 /* identify the backend used to play */
636 for(n = 0; backends[n]; ++n)
637 if(backends[n]->backend == config->speaker_backend)
638 break;
639 if(!backends[n])
640 fatal(0, "unsupported backend %d", config->speaker_backend);
641 backend = backends[n];
642 /* backend-specific initialization */
643 backend->init();
644 /* create the socket directory */
645 byte_xasprintf(&dir, "%s/speaker", config->home);
646 unlink(dir); /* might be a leftover socket */
647 if(mkdir(dir, 0700) < 0)
648 fatal(errno, "error creating %s", dir);
649 /* set up the listen socket */
650 listenfd = xsocket(PF_UNIX, SOCK_STREAM, 0);
651 memset(&addr, 0, sizeof addr);
652 addr.sun_family = AF_UNIX;
653 snprintf(addr.sun_path, sizeof addr.sun_path, "%s/speaker/socket",
654 config->home);
655 if(unlink(addr.sun_path) < 0 && errno != ENOENT)
656 error(errno, "removing %s", addr.sun_path);
657 xsetsockopt(listenfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one);
658 if(bind(listenfd, (const struct sockaddr *)&addr, sizeof addr) < 0)
659 fatal(errno, "error binding socket to %s", addr.sun_path);
660 xlisten(listenfd, 128);
661 nonblock(listenfd);
662 info("listening on %s", addr.sun_path);
663 memset(&sm, 0, sizeof sm);
664 sm.type = SM_READY;
665 speaker_send(1, &sm);
666 mainloop();
667 info("stopped (parent terminated)");
668 exit(0);
669 }
670
671 /*
672 Local Variables:
673 c-basic-offset:2
674 comment-column:40
675 fill-column:79
676 indent-tabs-mode:nil
677 End:
678 */