2 * This file is part of DisOrder
3 * Copyright (C) 2005-2009 Richard Kettlewell
4 * Portions (C) 2007 Mark Wooding
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.
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.
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/>.
19 /** @file server/speaker.c
20 * @brief Speaker process
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
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
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.
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 the its last bytes
40 * have been return by the callback; pause and resume is implemneted the
41 * obvious way. If the callback finds itself required to play when there is no
42 * playing track it returns dead air.
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
49 * @b Encodings. The encodings supported depend entirely on the uaudio backend
50 * chosen. See @ref uaudio.h, etc.
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).
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
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
79 #include <sys/select.h>
87 #include <sys/resource.h>
90 #include "configuration.h"
95 #include "speaker-protocol.h"
101 /** @brief Maximum number of FDs to poll for */
104 /** @brief Number of bytes before end of track to send SM_FINISHED
106 * Generally set to 1 second.
108 static size_t early_finish;
110 /** @brief Track structure
112 * Known tracks are kept in a linked list. Usually there will be at most two
113 * of these but rearranging the queue can cause there to be more.
116 /** @brief Next track */
119 /** @brief Input file descriptor */
120 int fd; /* input FD */
122 /** @brief Track ID */
125 /** @brief Start position of data in buffer */
128 /** @brief Number of bytes of data in buffer */
131 /** @brief Set @c fd is at EOF */
134 /** @brief Total number of samples played */
135 unsigned long long played;
137 /** @brief Slot in @ref fds */
140 /** @brief Set when playable
142 * A track becomes playable whenever it fills its buffer or reaches EOF; it
143 * stops being playable when it entirely empties its buffer. Tracks start
144 * out life not playable.
148 /** @brief Set when finished
150 * This is set when we've notified the server that the track is finished.
151 * Once this has happened (typically very late in the track's lifetime) the
152 * track cannot be paused or cancelled.
156 /** @brief Input buffer
158 * 1Mbyte is enough for nearly 6s of 44100Hz 16-bit stereo
160 char buffer[1048576];
163 /** @brief Lock protecting data structures
165 * This lock protects values shared between the main thread and the callback.
167 * It is held 'all' the time by the main thread, the exceptions being when
168 * called activate/deactivate callbacks and when calling (potentially) slow
169 * system calls (in particular poll(), where in fact the main thread will spend
170 * most of its time blocked).
172 * The callback holds it when it's running.
174 static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
176 /** @brief Linked list of all prepared tracks */
177 static struct track *tracks;
179 /** @brief Playing track, or NULL
181 * This means the track the speaker process intends to play. It does not
182 * reflect any other state (e.g. activation of uaudio backend).
184 static struct track *playing;
186 /** @brief Pending playing track, or NULL
188 * This means the track the server wants the speaker to play.
190 static struct track *pending_playing;
192 /** @brief Array of file descriptors for poll() */
193 static struct pollfd fds[NFDS];
195 /** @brief Next free slot in @ref fds */
198 /** @brief Listen socket */
201 /** @brief Timestamp of last potential report to server */
202 static time_t last_report;
204 /** @brief Set when paused */
207 /** @brief Set when back end activated */
208 static int activated;
210 /** @brief Signal pipe back into the poll() loop */
211 static int sigpipe[2];
213 /** @brief Selected backend */
214 static const struct uaudio *backend;
216 static const struct option options[] = {
217 { "help", no_argument, 0, 'h' },
218 { "version", no_argument, 0, 'V' },
219 { "config", required_argument, 0, 'c' },
220 { "debug", no_argument, 0, 'd' },
221 { "no-debug", no_argument, 0, 'D' },
222 { "syslog", no_argument, 0, 's' },
223 { "no-syslog", no_argument, 0, 'S' },
227 /* Display usage message and terminate. */
228 static void help(void) {
230 " disorder-speaker [OPTIONS]\n"
232 " --help, -h Display usage message\n"
233 " --version, -V Display version number\n"
234 " --config PATH, -c PATH Set configuration file\n"
235 " --debug, -d Turn on debugging\n"
236 " --[no-]syslog Force logging\n"
238 "Speaker process for DisOrder. Not intended to be run\n"
244 /** @brief Find track @p id, maybe creating it if not found
245 * @param id Track ID to find
246 * @param create If nonzero, create track structure of @p id not found
247 * @return Pointer to track structure or NULL
249 static struct track *findtrack(const char *id, int create) {
252 D(("findtrack %s %d", id, create));
253 for(t = tracks; t && strcmp(id, t->id); t = t->next)
256 t = xmalloc(sizeof *t);
265 /** @brief Remove track @p id (but do not destroy it)
266 * @param id Track ID to remove
267 * @return Track structure or NULL if not found
269 static struct track *removetrack(const char *id) {
270 struct track *t, **tt;
272 D(("removetrack %s", id));
273 for(tt = &tracks; (t = *tt) && strcmp(id, t->id); tt = &t->next)
280 /** @brief Destroy a track
281 * @param t Track structure
283 static void destroy(struct track *t) {
284 D(("destroy %s", t->id));
290 /** @brief Read data into a sample buffer
291 * @param t Pointer to track
292 * @return 0 on success, -1 on EOF
294 * This is effectively the read callback on @c t->fd. It is called from the
295 * main loop whenever the track's file descriptor is readable, assuming the
296 * buffer has not reached the maximum allowed occupancy.
298 static int speaker_fill(struct track *t) {
302 D(("fill %s: eof=%d used=%zu",
303 t->id, t->eof, t->used));
306 if(t->used < sizeof t->buffer) {
307 /* there is room left in the buffer */
308 where = (t->start + t->used) % sizeof t->buffer;
309 /* Get as much data as we can */
310 if(where >= t->start)
311 left = (sizeof t->buffer) - where;
313 left = t->start - where;
314 pthread_mutex_unlock(&lock);
316 n = read(t->fd, t->buffer + where, left);
317 } while(n < 0 && errno == EINTR);
318 pthread_mutex_lock(&lock);
321 disorder_fatal(errno, "error reading sample stream");
324 D(("fill %s: eof detected", t->id));
326 /* A track always becomes playable at EOF; we're not going to see any
332 /* A track becomes playable when it (first) fills its buffer. For
333 * 44.1KHz 16-bit stereo this is ~6s of audio data. The latency will
334 * depend how long that takes to decode (hopefuly not very!) */
335 if(t->used == sizeof t->buffer)
343 /** @brief Return nonzero if we want to play some audio
345 * We want to play audio if there is a current track; and it is not paused; and
346 * it is playable according to the rules for @ref track::playable.
348 * We don't allow tracks to be paused if we've already told the server we've
349 * finished them; that would cause such tracks to survive much longer than the
350 * few samples they're supposed to, with report() remaining silent for the
353 static int playable(void) {
355 && (!paused || playing->finished)
356 && playing->playable;
359 /** @brief Notify the server what we're up to */
360 static void report(void) {
361 struct speaker_message sm;
364 /* Had better not send a report for a track that the server thinks has
365 * finished, that would be confusing. */
366 if(playing->finished)
368 memset(&sm, 0, sizeof sm);
369 sm.type = paused ? SM_PAUSED : SM_PLAYING;
370 strcpy(sm.id, playing->id);
371 sm.data = playing->played / (uaudio_rate * uaudio_channels);
372 speaker_send(1, &sm);
377 /** @brief Add a file descriptor to the set to poll() for
378 * @param fd File descriptor
379 * @param events Events to wait for e.g. @c POLLIN
380 * @return Slot number
382 static int addfd(int fd, int events) {
385 fds[fdno].events = events;
391 /** @brief Callback to return some sampled data
392 * @param buffer Where to put sample data
393 * @param max_samples How many samples to return
394 * @param userdata User data
395 * @return Number of samples written
397 * See uaudio_callback().
399 static size_t speaker_callback(void *buffer,
401 void attribute((unused)) *userdata) {
402 const size_t max_bytes = max_samples * uaudio_sample_size;
403 size_t provided_samples = 0;
405 pthread_mutex_lock(&lock);
406 /* TODO perhaps we should immediately go silent if we've been asked to pause
407 * or cancel the playing track (maybe block in the cancel case and see what
410 if(playing->used > 0) {
412 /* Compute size of largest contiguous chunk. We get called as often as
413 * necessary so there's no need for cleverness here. */
414 if(playing->start + playing->used > sizeof playing->buffer)
415 bytes = sizeof playing->buffer - playing->start;
417 bytes = playing->used;
418 /* Limit to what we were asked for */
419 if(bytes > max_bytes)
422 memcpy(buffer, playing->buffer + playing->start, bytes);
423 playing->start += bytes;
424 playing->used -= bytes;
425 /* Wrap around to start of buffer */
426 if(playing->start == sizeof playing->buffer)
428 /* See if we've reached the end of the track */
429 if(playing->used == 0 && playing->eof) {
430 int ignored = write(sigpipe[1], "", 1);
433 provided_samples = bytes / uaudio_sample_size;
434 playing->played += provided_samples;
437 /* If we couldn't provide anything at all, play dead air */
438 /* TODO maybe it would be better to block, in some cases? */
439 if(!provided_samples) {
440 memset(buffer, 0, max_bytes);
441 provided_samples = max_samples;
443 disorder_info("%zu samples silence, playing->used=%zu",
444 provided_samples, playing->used);
446 disorder_info("%zu samples silence, playing=NULL", provided_samples);
448 pthread_mutex_unlock(&lock);
449 return provided_samples;
452 /** @brief Main event loop */
453 static void mainloop(void) {
455 struct speaker_message sm;
456 int n, fd, stdin_slot, timeout, listen_slot, sigpipe_slot;
458 /* Keep going while our parent process is alive */
459 pthread_mutex_lock(&lock);
460 while(getppid() != 1) {
461 int force_report = 0;
464 /* By default we will wait up to half a second before thinking about
467 /* Always ready for commands from the main server. */
468 stdin_slot = addfd(0, POLLIN);
469 /* Also always ready for inbound connections */
470 listen_slot = addfd(listenfd, POLLIN);
471 /* Try to read sample data for the currently playing track if there is
476 && playing->used < (sizeof playing->buffer))
477 playing->slot = addfd(playing->fd, POLLIN);
480 /* If any other tracks don't have a full buffer, try to read sample data
481 * from them. We do this last of all, so that if we run out of slots,
482 * nothing important can't be monitored. */
483 for(t = tracks; t; t = t->next)
487 && t->used < sizeof t->buffer) {
488 t->slot = addfd(t->fd, POLLIN | POLLHUP);
492 sigpipe_slot = addfd(sigpipe[0], POLLIN);
493 /* Wait for something interesting to happen */
494 pthread_mutex_unlock(&lock);
495 n = poll(fds, fdno, timeout);
496 pthread_mutex_lock(&lock);
498 if(errno == EINTR) continue;
499 disorder_fatal(errno, "error calling poll");
501 /* Perhaps a connection has arrived */
502 if(fds[listen_slot].revents & POLLIN) {
503 struct sockaddr_un addr;
504 socklen_t addrlen = sizeof addr;
508 if((fd = accept(listenfd, (struct sockaddr *)&addr, &addrlen)) >= 0) {
510 if(read(fd, &l, sizeof l) < 4) {
511 disorder_error(errno, "reading length from inbound connection");
513 } else if(l >= sizeof id) {
514 disorder_error(0, "id length too long");
516 } else if(read(fd, id, l) < (ssize_t)l) {
517 disorder_error(errno, "reading id from inbound connection");
521 D(("id %s fd %d", id, fd));
522 t = findtrack(id, 1/*create*/);
523 if (write(fd, "", 1) < 0) /* write an ack */
524 disorder_error(errno, "writing ack to inbound connection");
526 disorder_error(0, "%s: already got a connection", id);
530 t->fd = fd; /* yay */
532 /* Notify the server that the connection arrived */
533 sm.type = SM_ARRIVED;
535 speaker_send(1, &sm);
538 disorder_error(errno, "accept");
540 /* Perhaps we have a command to process */
541 if(fds[stdin_slot].revents & POLLIN) {
542 /* There might (in theory) be several commands queued up, but in general
543 * this won't be the case, so we don't bother looping around to pick them
545 n = speaker_recv(0, &sm);
547 /* As a rule we don't send success replies to most commands - we just
548 * force the regular status update to be sent immediately rather than
552 /* SM_PLAY is only allowed if the server reasonably believes that
553 * nothing is playing */
555 /* If finished isn't set then the server can't believe that this
556 * track has finished */
557 if(!playing->finished)
558 disorder_fatal(0, "got SM_PLAY but already playing something");
559 /* If pending_playing is set then the server must believe that that
562 disorder_fatal(0, "got SM_PLAY but have a pending playing track");
564 t = findtrack(sm.id, 1);
565 D(("SM_PLAY %s fd %d", t->id, t->fd));
568 "cannot play track because no connection arrived");
569 /* TODO as things stand we often report this error message but then
570 * appear to proceed successfully. Understanding why requires a look
571 * at play.c: we call prepare() which makes the connection in a child
572 * process, and then sends the SM_PLAY in the parent process. The
573 * latter may well be faster. As it happens this is harmless; we'll
574 * just sit around sending silence until the decoder connects and
575 * starts sending some sample data. But is is annoying and ought to
578 /* If nothing is currently playing then we'll switch to the pending
579 * track below so there's no point distinguishing the situations
593 D(("SM_CANCEL %s", sm.id));
594 t = removetrack(sm.id);
596 if(t == playing || t == pending_playing) {
597 /* Scratching the track that the server believes is playing,
598 * which might either be the actual playing track or a pending
600 sm.type = SM_FINISHED;
606 /* Could be scratching the playing track before it's quite got
607 * going, or could be just removing a track from the queue. We
608 * log more because there's been a bug here recently than because
609 * it's particularly interesting; the log message will be removed
610 * if no further problems show up. */
611 disorder_info("SM_CANCEL for nonplaying track %s", sm.id);
612 sm.type = SM_STILLBORN;
614 strcpy(sm.id, t->id);
617 /* Probably scratching the playing track well before it's got
618 * going, but could indicate a bug, so we log this as an error. */
619 sm.type = SM_UNKNOWN;
620 disorder_error(0, "SM_CANCEL for unknown track %s", sm.id);
622 speaker_send(1, &sm);
627 if(config_read(1, NULL))
628 disorder_error(0, "cannot read configuration");
629 disorder_info("reloaded configuration");
632 disorder_error(0, "unknown message type %d", sm.type);
635 /* Read in any buffered data */
636 for(t = tracks; t; t = t->next)
639 && (fds[t->slot].revents & (POLLIN | POLLHUP)))
641 /* Drain the signal pipe. We don't care about its contents, merely that it
642 * interrupted poll(). */
643 if(fds[sigpipe_slot].revents & POLLIN) {
645 int ignored; (void)ignored;
647 ignored = read(sigpipe[0], buffer, sizeof buffer);
649 /* Send SM_FINISHED when we're near the end of the track.
651 * This is how we implement gapless play; we hope that the SM_PLAY from the
652 * server arrives before the remaining bytes of the track play out.
656 && !playing->finished
657 && playing->used <= early_finish) {
658 memset(&sm, 0, sizeof sm);
659 sm.type = SM_FINISHED;
660 strcpy(sm.id, playing->id);
661 speaker_send(1, &sm);
662 playing->finished = 1;
664 /* When the track is actually finished, deconfigure it */
665 if(playing && playing->eof && !playing->used) {
666 removetrack(playing->id);
670 /* Act on the pending SM_PLAY */
671 if(!playing && pending_playing) {
672 playing = pending_playing;
676 /* Impose any state change required by the above */
680 pthread_mutex_unlock(&lock);
682 pthread_mutex_lock(&lock);
687 pthread_mutex_unlock(&lock);
688 backend->deactivate();
689 pthread_mutex_lock(&lock);
692 /* If we've not reported our state for a second do so now. */
693 if(force_report || xtime(0) > last_report)
698 int main(int argc, char **argv) {
699 int n, logsyslog = !isatty(2);
700 struct sockaddr_un addr;
701 static const int one = 1;
702 struct speaker_message sm;
708 if(!setlocale(LC_CTYPE, "")) disorder_fatal(errno, "error calling setlocale");
709 while((n = getopt_long(argc, argv, "hVc:dDSs", options, 0)) >= 0) {
712 case 'V': version("disorder-speaker");
713 case 'c': configfile = optarg; break;
714 case 'd': debugging = 1; break;
715 case 'D': debugging = 0; break;
716 case 'S': logsyslog = 0; break;
717 case 's': logsyslog = 1; break;
718 default: disorder_fatal(0, "invalid option");
721 if((d = getenv("DISORDER_DEBUG_SPEAKER"))) debugging = atoi(d);
723 openlog(progname, LOG_PID, LOG_DAEMON);
724 log_default = &log_syslog;
726 config_uaudio_apis = uaudio_apis;
727 if(config_read(1, NULL)) disorder_fatal(0, "cannot read configuration");
729 signal(SIGPIPE, SIG_IGN);
731 xnice(config->nice_speaker);
734 /* make sure we're not root, whatever the config says */
735 if(getuid() == 0 || geteuid() == 0)
736 disorder_fatal(0, "do not run as root");
737 /* Make sure we can't have more than NFDS files open (it would bust our
739 if(getrlimit(RLIMIT_NOFILE, rl) < 0)
740 disorder_fatal(errno, "getrlimit RLIMIT_NOFILE");
741 if(rl->rlim_cur > NFDS) {
743 if(setrlimit(RLIMIT_NOFILE, rl) < 0)
744 disorder_fatal(errno, "setrlimit to reduce RLIMIT_NOFILE to %lu",
745 (unsigned long)rl->rlim_cur);
746 disorder_info("set RLIM_NOFILE to %lu", (unsigned long)rl->rlim_cur);
748 disorder_info("RLIM_NOFILE is %lu", (unsigned long)rl->rlim_cur);
749 /* gcrypt initialization */
750 if(!gcry_check_version(NULL))
751 disorder_fatal(0, "gcry_check_version failed");
752 gcry_control(GCRYCTL_INIT_SECMEM, 0);
753 gcry_control (GCRYCTL_INITIALIZATION_FINISHED, 0);
754 /* create a pipe between the backend callback and the poll() loop */
756 nonblock(sigpipe[0]);
757 /* set up audio backend */
758 uaudio_set_format(config->sample_format.rate,
759 config->sample_format.channels,
760 config->sample_format.bits,
761 config->sample_format.bits != 8);
762 early_finish = uaudio_sample_size * uaudio_channels * uaudio_rate;
763 /* TODO other parameters! */
764 backend = uaudio_find(config->api);
765 /* backend-specific initialization */
766 if(backend->configure)
767 backend->configure();
768 backend->start(speaker_callback, NULL);
769 /* create the socket directory */
770 byte_xasprintf(&dir, "%s/speaker", config->home);
771 unlink(dir); /* might be a leftover socket */
772 if(mkdir(dir, 0700) < 0 && errno != EEXIST)
773 disorder_fatal(errno, "error creating %s", dir);
774 /* set up the listen socket */
775 listenfd = xsocket(PF_UNIX, SOCK_STREAM, 0);
776 memset(&addr, 0, sizeof addr);
777 addr.sun_family = AF_UNIX;
778 snprintf(addr.sun_path, sizeof addr.sun_path, "%s/speaker/socket",
780 if(unlink(addr.sun_path) < 0 && errno != ENOENT)
781 disorder_error(errno, "removing %s", addr.sun_path);
782 xsetsockopt(listenfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one);
783 if(bind(listenfd, (const struct sockaddr *)&addr, sizeof addr) < 0)
784 disorder_fatal(errno, "error binding socket to %s", addr.sun_path);
785 xlisten(listenfd, 128);
787 disorder_info("listening on %s", addr.sun_path);
788 memset(&sm, 0, sizeof sm);
790 speaker_send(1, &sm);
792 disorder_info("stopped (parent terminated)");