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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 *
367 * @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 */
79struct track *tracks;
80
81/** @brief Playing track, or NULL */
82struct track *playing;
83
84/** @brief Number of bytes pre frame */
85size_t bpf;
86
87/** @brief Array of file descriptors for poll() */
88struct pollfd fds[NFDS];
89
90/** @brief Next free slot in @ref fds */
91int fdno;
92
93/** @brief Listen socket */
94static int listenfd;
95
96static time_t last_report; /* when we last reported */
97static int paused; /* pause status */
98
99/** @brief The current device state */
100enum 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 */
106int idled;
107
108/** @brief Selected backend */
109static const struct speaker_backend *backend;
110
111static 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. */
123static 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 */
140static 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 */
145static 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) */
162static 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 */
174static 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 */
188static 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 */
227static 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 */
237void 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 */
255static 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 */
269static 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 */
281static 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 */
300static 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. */
353static 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
366static 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
376int 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 */
386static 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 */
402static 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 write(fd, "", 1); /* write an ack */
494 if(t->fd != -1) {
495 error(0, "%s: already got a connection", id);
496 xclose(fd);
497 } else {
498 nonblock(fd);
499 t->fd = fd; /* yay */
500 }
501 }
502 } else
503 error(errno, "accept");
504 }
505 /* Perhaps we have a command to process */
506 if(fds[stdin_slot].revents & POLLIN) {
507 /* There might (in theory) be several commands queued up, but in general
508 * this won't be the case, so we don't bother looping around to pick them
509 * all up. */
510 n = speaker_recv(0, &sm);
511 /* TODO */
512 if(n > 0)
513 switch(sm.type) {
514 case SM_PLAY:
515 if(playing) fatal(0, "got SM_PLAY but already playing something");
516 t = findtrack(sm.id, 1);
517 D(("SM_PLAY %s fd %d", t->id, t->fd));
518 if(t->fd == -1)
519 error(0, "cannot play track because no connection arrived");
520 playing = t;
521 /* We attempt to play straight away rather than going round the loop.
522 * speaker_play() is clever enough to perform any activation that is
523 * required. */
524 speaker_play(3 * FRAMES);
525 report();
526 break;
527 case SM_PAUSE:
528 D(("SM_PAUSE"));
529 paused = 1;
530 report();
531 break;
532 case SM_RESUME:
533 D(("SM_RESUME"));
534 if(paused) {
535 paused = 0;
536 /* As for SM_PLAY we attempt to play straight away. */
537 if(playing)
538 speaker_play(3 * FRAMES);
539 }
540 report();
541 break;
542 case SM_CANCEL:
543 D(("SM_CANCEL %s", sm.id));
544 t = removetrack(sm.id);
545 if(t) {
546 if(t == playing) {
547 /* scratching the playing track */
548 sm.type = SM_FINISHED;
549 playing = 0;
550 } else {
551 /* Could be scratching the playing track before it's quite got
552 * going, or could be just removing a track from the queue. We
553 * log more because there's been a bug here recently than because
554 * it's particularly interesting; the log message will be removed
555 * if no further problems show up. */
556 info("SM_CANCEL for nonplaying track %s", sm.id);
557 sm.type = SM_STILLBORN;
558 }
559 strcpy(sm.id, t->id);
560 destroy(t);
561 } else {
562 /* Probably scratching the playing track well before it's got
563 * going, but could indicate a bug, so we log this as an error. */
564 sm.type = SM_UNKNOWN;
565 error(0, "SM_CANCEL for unknown track %s", sm.id);
566 }
567 speaker_send(1, &sm);
568 report();
569 break;
570 case SM_RELOAD:
571 D(("SM_RELOAD"));
572 if(config_read(1)) error(0, "cannot read configuration");
573 info("reloaded configuration");
574 break;
575 default:
576 error(0, "unknown message type %d", sm.type);
577 }
578 }
579 /* Read in any buffered data */
580 for(t = tracks; t; t = t->next)
581 if(t->fd != -1
582 && t->slot != -1
583 && (fds[t->slot].revents & (POLLIN | POLLHUP)))
584 speaker_fill(t);
585 /* Maybe we finished playing a track somewhere in the above */
586 maybe_finished();
587 /* If we don't need the sound device for now then close it for the benefit
588 * of anyone else who wants it. */
589 if((!playing || paused) && device_state == device_open)
590 idle();
591 /* If we've not reported out state for a second do so now. */
592 if(time(0) > last_report)
593 report();
594 }
595}
596
597int main(int argc, char **argv) {
598 int n, logsyslog = !isatty(2);
599 struct sockaddr_un addr;
600 static const int one = 1;
601 struct speaker_message sm;
602 const char *d;
603 char *dir;
604
605 set_progname(argv);
606 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
607 while((n = getopt_long(argc, argv, "hVc:dDSs", options, 0)) >= 0) {
608 switch(n) {
609 case 'h': help();
610 case 'V': version("disorder-speaker");
611 case 'c': configfile = optarg; break;
612 case 'd': debugging = 1; break;
613 case 'D': debugging = 0; break;
614 case 'S': logsyslog = 0; break;
615 case 's': logsyslog = 1; break;
616 default: fatal(0, "invalid option");
617 }
618 }
619 if((d = getenv("DISORDER_DEBUG_SPEAKER"))) debugging = atoi(d);
620 if(logsyslog) {
621 openlog(progname, LOG_PID, LOG_DAEMON);
622 log_default = &log_syslog;
623 }
624 if(config_read(1)) fatal(0, "cannot read configuration");
625 bpf = bytes_per_frame(&config->sample_format);
626 /* ignore SIGPIPE */
627 signal(SIGPIPE, SIG_IGN);
628 /* reap kids */
629 signal(SIGCHLD, reap);
630 /* set nice value */
631 xnice(config->nice_speaker);
632 /* change user */
633 become_mortal();
634 /* make sure we're not root, whatever the config says */
635 if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root");
636 /* identify the backend used to play */
637 for(n = 0; backends[n]; ++n)
638 if(backends[n]->backend == config->api)
639 break;
640 if(!backends[n])
641 fatal(0, "unsupported api %d", config->api);
642 backend = backends[n];
643 /* backend-specific initialization */
644 backend->init();
645 /* create the socket directory */
646 byte_xasprintf(&dir, "%s/speaker", config->home);
647 unlink(dir); /* might be a leftover socket */
648 if(mkdir(dir, 0700) < 0 && errno != EEXIST)
649 fatal(errno, "error creating %s", dir);
650 /* set up the listen socket */
651 listenfd = xsocket(PF_UNIX, SOCK_STREAM, 0);
652 memset(&addr, 0, sizeof addr);
653 addr.sun_family = AF_UNIX;
654 snprintf(addr.sun_path, sizeof addr.sun_path, "%s/speaker/socket",
655 config->home);
656 if(unlink(addr.sun_path) < 0 && errno != ENOENT)
657 error(errno, "removing %s", addr.sun_path);
658 xsetsockopt(listenfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one);
659 if(bind(listenfd, (const struct sockaddr *)&addr, sizeof addr) < 0)
660 fatal(errno, "error binding socket to %s", addr.sun_path);
661 xlisten(listenfd, 128);
662 nonblock(listenfd);
663 info("listening on %s", addr.sun_path);
664 memset(&sm, 0, sizeof sm);
665 sm.type = SM_READY;
666 speaker_send(1, &sm);
667 mainloop();
668 info("stopped (parent terminated)");
669 exit(0);
670}
671
672/*
673Local Variables:
674c-basic-offset:2
675comment-column:40
676fill-column:79
677indent-tabs-mode:nil
678End:
679*/