2 * This file is part of DisOrder
3 * Copyright (C) 2005-2008 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 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.)
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).
36 * @b Garbage @b Collection. This program deliberately does not use the
37 * garbage collector even though it might be convenient to do so. This is for
38 * two reasons. Firstly some sound APIs use thread threads and we do not want
39 * to have to deal with potential interactions between threading and garbage
40 * collection. Secondly this process needs to be able to respond quickly and
41 * this is not compatible with the collector hanging the program even
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
59 #include <sys/select.h>
67 #include "configuration.h"
72 #include "speaker-protocol.h"
78 /** @brief Linked list of all prepared tracks */
81 /** @brief Playing track, or NULL */
82 struct track *playing;
84 /** @brief Number of bytes pre frame */
87 /** @brief Array of file descriptors for poll() */
88 struct pollfd fds[NFDS];
90 /** @brief Next free slot in @ref fds */
93 /** @brief Listen socket */
96 static time_t last_report; /* when we last reported */
97 static int paused; /* pause status */
99 /** @brief The current device state */
100 enum device_states device_state;
102 /** @brief Set when idled
104 * This is set when the sound device is deliberately closed by idle().
108 /** @brief Selected backend */
109 static const struct speaker_backend *backend;
111 static const struct option options[] = {
112 { "help", no_argument, 0, 'h' },
113 { "version", no_argument, 0, 'V' },
114 { "config", required_argument, 0, 'c' },
115 { "debug", no_argument, 0, 'd' },
116 { "no-debug", no_argument, 0, 'D' },
117 { "syslog", no_argument, 0, 's' },
118 { "no-syslog", no_argument, 0, 'S' },
122 /* Display usage message and terminate. */
123 static void help(void) {
125 " disorder-speaker [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"
133 "Speaker process for DisOrder. Not intended to be run\n"
139 /** @brief Return the number of bytes per frame in @p format */
140 static size_t bytes_per_frame(const struct stream_header *format) {
141 return format->channels * format->bits / 8;
144 /** @brief Find track @p id, maybe creating it if not found */
145 static struct track *findtrack(const char *id, int create) {
148 D(("findtrack %s %d", id, create));
149 for(t = tracks; t && strcmp(id, t->id); t = t->next)
152 t = xmalloc(sizeof *t);
161 /** @brief Remove track @p id (but do not destroy it) */
162 static struct track *removetrack(const char *id) {
163 struct track *t, **tt;
165 D(("removetrack %s", id));
166 for(tt = &tracks; (t = *tt) && strcmp(id, t->id); tt = &t->next)
173 /** @brief Destroy a track */
174 static void destroy(struct track *t) {
175 D(("destroy %s", t->id));
176 if(t->fd != -1) xclose(t->fd);
180 /** @brief Read data into a sample buffer
181 * @param t Pointer to track
182 * @return 0 on success, -1 on EOF
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.
188 static int speaker_fill(struct track *t) {
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;
202 n = read(t->fd, t->buffer + where, left);
203 } while(n < 0 && errno == EINTR);
205 if(errno != EAGAIN) fatal(errno, "error reading sample stream");
209 D(("fill %s: eof detected", t->id));
215 if(t->used == sizeof t->buffer)
221 /** @brief Close the sound device
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).
227 static void idle(void) {
229 if(backend->deactivate)
230 backend->deactivate();
232 device_state = device_closed;
236 /** @brief Abandon the current track */
238 struct speaker_message sm;
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);
250 /** @brief Enable sound output
252 * Makes sure the sound device is open and has the right sample format. Return
253 * 0 on success and -1 on error.
255 static void activate(void) {
256 if(backend->activate)
259 device_state = device_open;
262 /** @brief Check whether the current track has finished
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.)
269 static void maybe_finished(void) {
272 && playing->used < bytes_per_frame(&config->sample_format))
276 /** @brief Return nonzero if we want to play some audio
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.
281 static int playable(void) {
284 && playing->playable;
287 /** @brief Play up to @p frames frames of audio
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.
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.
300 static void speaker_play(size_t frames) {
301 size_t avail_frames, avail_bytes, written_frames;
302 ssize_t written_bytes;
304 /* Make sure there's a track to play and it is not paused */
307 /* Make sure the output device is open */
308 if(device_state != device_open) {
310 if(device_state != device_open)
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;
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;
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
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))
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;
348 frames -= written_frames;
352 /* Notify the server what we're up to. */
353 static void report(void) {
354 struct speaker_message sm;
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);
366 static void reap(int __attribute__((unused)) sig) {
371 cmdpid = waitpid(-1, &st, WNOHANG);
373 signal(SIGCHLD, reap);
376 int addfd(int fd, int events) {
379 fds[fdno].events = events;
385 /** @brief Table of speaker backends */
386 static const struct speaker_backend *backends[] = {
387 #if HAVE_ALSA_ASOUNDLIB_H
392 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
395 #if HAVE_SYS_SOUNDCARD_H
401 /** @brief Main event loop */
402 static void mainloop(void) {
404 struct speaker_message sm;
405 int n, fd, stdin_slot, timeout, listen_slot;
407 while(getppid() != 1) {
409 /* By default we will wait up to a second before thinking about current
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
421 && playing->used < (sizeof playing->buffer))
422 playing->slot = addfd(playing->fd, POLLIN);
426 /* We want to play some audio. If the device is closed then we attempt
428 if(device_state == device_closed)
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
434 if(device_state == device_open)
435 backend->beforepoll(&timeout);
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)
444 && t->used < sizeof t->buffer) {
445 t->slot = addfd(t->fd, POLLIN | POLLHUP);
449 /* Wait for something interesting to happen */
450 n = poll(fds, fdno, timeout);
452 if(errno == EINTR) continue;
453 fatal(errno, "error calling poll");
455 /* Play some sound before doing anything else */
457 /* We want to play some audio */
458 if(device_state == device_open) {
460 speaker_play(3 * FRAMES);
462 /* We must be in _closed or _error, and it should be the latter, but we
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.
468 speaker_play(3 * FRAMES);
471 /* Perhaps a connection has arrived */
472 if(fds[listen_slot].revents & POLLIN) {
473 struct sockaddr_un addr;
474 socklen_t addrlen = sizeof addr;
478 if((fd = accept(listenfd, (struct sockaddr *)&addr, &addrlen)) >= 0) {
480 if(read(fd, &l, sizeof l) < 4) {
481 error(errno, "reading length from inbound connection");
483 } else if(l >= sizeof id) {
484 error(0, "id length too long");
486 } else if(read(fd, id, l) < (ssize_t)l) {
487 error(errno, "reading id from inbound connection");
491 D(("id %s fd %d", id, fd));
492 t = findtrack(id, 1/*create*/);
493 if (write(fd, "", 1) < 0) /* write an ack */
494 error(errno, "writing ack to inbound connection");
496 error(0, "%s: already got a connection", id);
500 t->fd = fd; /* yay */
504 error(errno, "accept");
506 /* Perhaps we have a command to process */
507 if(fds[stdin_slot].revents & POLLIN) {
508 /* There might (in theory) be several commands queued up, but in general
509 * this won't be the case, so we don't bother looping around to pick them
511 n = speaker_recv(0, &sm);
516 if(playing) fatal(0, "got SM_PLAY but already playing something");
517 t = findtrack(sm.id, 1);
518 D(("SM_PLAY %s fd %d", t->id, t->fd));
520 error(0, "cannot play track because no connection arrived");
522 /* We attempt to play straight away rather than going round the loop.
523 * speaker_play() is clever enough to perform any activation that is
525 speaker_play(3 * FRAMES);
537 /* As for SM_PLAY we attempt to play straight away. */
539 speaker_play(3 * FRAMES);
544 D(("SM_CANCEL %s", sm.id));
545 t = removetrack(sm.id);
548 /* scratching the playing track */
549 sm.type = SM_FINISHED;
552 /* Could be scratching the playing track before it's quite got
553 * going, or could be just removing a track from the queue. We
554 * log more because there's been a bug here recently than because
555 * it's particularly interesting; the log message will be removed
556 * if no further problems show up. */
557 info("SM_CANCEL for nonplaying track %s", sm.id);
558 sm.type = SM_STILLBORN;
560 strcpy(sm.id, t->id);
563 /* Probably scratching the playing track well before it's got
564 * going, but could indicate a bug, so we log this as an error. */
565 sm.type = SM_UNKNOWN;
566 error(0, "SM_CANCEL for unknown track %s", sm.id);
568 speaker_send(1, &sm);
573 if(config_read(1)) error(0, "cannot read configuration");
574 info("reloaded configuration");
577 error(0, "unknown message type %d", sm.type);
580 /* Read in any buffered data */
581 for(t = tracks; t; t = t->next)
584 && (fds[t->slot].revents & (POLLIN | POLLHUP)))
586 /* Maybe we finished playing a track somewhere in the above */
588 /* If we don't need the sound device for now then close it for the benefit
589 * of anyone else who wants it. */
590 if((!playing || paused) && device_state == device_open)
592 /* If we've not reported out state for a second do so now. */
593 if(time(0) > last_report)
598 int main(int argc, char **argv) {
599 int n, logsyslog = !isatty(2);
600 struct sockaddr_un addr;
601 static const int one = 1;
602 struct speaker_message sm;
607 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
608 while((n = getopt_long(argc, argv, "hVc:dDSs", options, 0)) >= 0) {
611 case 'V': version("disorder-speaker");
612 case 'c': configfile = optarg; break;
613 case 'd': debugging = 1; break;
614 case 'D': debugging = 0; break;
615 case 'S': logsyslog = 0; break;
616 case 's': logsyslog = 1; break;
617 default: fatal(0, "invalid option");
620 if((d = getenv("DISORDER_DEBUG_SPEAKER"))) debugging = atoi(d);
622 openlog(progname, LOG_PID, LOG_DAEMON);
623 log_default = &log_syslog;
625 if(config_read(1)) fatal(0, "cannot read configuration");
626 bpf = bytes_per_frame(&config->sample_format);
628 signal(SIGPIPE, SIG_IGN);
630 signal(SIGCHLD, reap);
632 xnice(config->nice_speaker);
635 /* make sure we're not root, whatever the config says */
636 if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root");
637 /* identify the backend used to play */
638 for(n = 0; backends[n]; ++n)
639 if(backends[n]->backend == config->api)
642 fatal(0, "unsupported api %d", config->api);
643 backend = backends[n];
644 /* backend-specific initialization */
646 /* create the socket directory */
647 byte_xasprintf(&dir, "%s/speaker", config->home);
648 unlink(dir); /* might be a leftover socket */
649 if(mkdir(dir, 0700) < 0 && errno != EEXIST)
650 fatal(errno, "error creating %s", dir);
651 /* set up the listen socket */
652 listenfd = xsocket(PF_UNIX, SOCK_STREAM, 0);
653 memset(&addr, 0, sizeof addr);
654 addr.sun_family = AF_UNIX;
655 snprintf(addr.sun_path, sizeof addr.sun_path, "%s/speaker/socket",
657 if(unlink(addr.sun_path) < 0 && errno != ENOENT)
658 error(errno, "removing %s", addr.sun_path);
659 xsetsockopt(listenfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one);
660 if(bind(listenfd, (const struct sockaddr *)&addr, sizeof addr) < 0)
661 fatal(errno, "error binding socket to %s", addr.sun_path);
662 xlisten(listenfd, 128);
664 info("listening on %s", addr.sun_path);
665 memset(&sm, 0, sizeof sm);
667 speaker_send(1, &sm);
669 info("stopped (parent terminated)");