2 * This file is part of DisOrder
3 * Copyright (C) 2005, 2006, 2007 Richard Kettlewell
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation; either version 2 of the License, or
8 * (at your option) any later version.
10 * This program is distributed in the hope that it will be useful, but
11 * WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * General Public License for more details.
15 * You should have received a copy of the GNU General Public License
16 * along with this program; if not, write to the Free Software
17 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
20 /** @file server/speaker.c
21 * @brief Speaker process
23 * This program is responsible for transmitting a single coherent audio stream
24 * to its destination (over the network, to some sound API, to some
25 * subprocess). It receives connections from decoders (or rather from the
26 * process that is about to become disorder-normalize) and plays them in the
29 * @b Encodings. For the <a href="http://www.alsa-project.org/">ALSA</a> API,
30 * 8- and 16- bit stereo and mono are supported, with any sample rate (within
31 * the limits that ALSA can deal with.)
33 * Inbound data is expected to match @c config->sample_format. In normal use
34 * this is arranged by the @c disorder-normalize program (see @ref
35 * server/normalize.c).
37 * @b Garbage @b Collection. This program deliberately does not use the
38 * garbage collector even though it might be convenient to do so. This is for
39 * two reasons. Firstly some sound APIs use thread threads and we do not want
40 * to have to deal with potential interactions between threading and garbage
41 * collection. Secondly this process needs to be able to respond quickly and
42 * this is not compatible with the collector hanging the program even
45 * @b Units. This program thinks at various times in three different units.
46 * Bytes are obvious. A sample is a single sample on a single channel. A
47 * frame is several samples on different channels at the same point in time.
48 * So (for instance) a 16-bit stereo frame is 4 bytes and consists of a pair of
65 #include <sys/select.h>
72 #include "configuration.h"
77 #include "speaker-protocol.h"
81 /** @brief Linked list of all prepared tracks */
84 /** @brief Playing track, or NULL */
85 struct track *playing;
87 /** @brief Number of bytes pre frame */
90 /** @brief Array of file descriptors for poll() */
91 struct pollfd fds[NFDS];
93 /** @brief Next free slot in @ref fds */
96 /** @brief Listen socket */
99 static time_t last_report; /* when we last reported */
100 static int paused; /* pause status */
102 /** @brief The current device state */
103 enum device_states device_state;
105 /** @brief Set when idled
107 * This is set when the sound device is deliberately closed by idle().
111 /** @brief Selected backend */
112 static const struct speaker_backend *backend;
114 static const struct option options[] = {
115 { "help", no_argument, 0, 'h' },
116 { "version", no_argument, 0, 'V' },
117 { "config", required_argument, 0, 'c' },
118 { "debug", no_argument, 0, 'd' },
119 { "no-debug", no_argument, 0, 'D' },
120 { "syslog", no_argument, 0, 's' },
121 { "no-syslog", no_argument, 0, 'S' },
125 /* Display usage message and terminate. */
126 static void help(void) {
128 " disorder-speaker [OPTIONS]\n"
130 " --help, -h Display usage message\n"
131 " --version, -V Display version number\n"
132 " --config PATH, -c PATH Set configuration file\n"
133 " --debug, -d Turn on debugging\n"
134 " --[no-]syslog Force logging\n"
136 "Speaker process for DisOrder. Not intended to be run\n"
142 /* Display version number and terminate. */
143 static void version(void) {
144 xprintf("disorder-speaker version %s\n", disorder_version_string);
149 /** @brief Return the number of bytes per frame in @p format */
150 static size_t bytes_per_frame(const struct stream_header *format) {
151 return format->channels * format->bits / 8;
154 /** @brief Find track @p id, maybe creating it if not found */
155 static struct track *findtrack(const char *id, int create) {
158 D(("findtrack %s %d", id, create));
159 for(t = tracks; t && strcmp(id, t->id); t = t->next)
162 t = xmalloc(sizeof *t);
171 /** @brief Remove track @p id (but do not destroy it) */
172 static struct track *removetrack(const char *id) {
173 struct track *t, **tt;
175 D(("removetrack %s", id));
176 for(tt = &tracks; (t = *tt) && strcmp(id, t->id); tt = &t->next)
183 /** @brief Destroy a track */
184 static void destroy(struct track *t) {
185 D(("destroy %s", t->id));
186 if(t->fd != -1) xclose(t->fd);
190 /** @brief Read data into a sample buffer
191 * @param t Pointer to track
192 * @return 0 on success, -1 on EOF
194 * This is effectively the read callback on @c t->fd. It is called from the
195 * main loop whenever the track's file descriptor is readable, assuming the
196 * buffer has not reached the maximum allowed occupancy.
198 static int speaker_fill(struct track *t) {
202 D(("fill %s: eof=%d used=%zu",
203 t->id, t->eof, t->used));
204 if(t->eof) return -1;
205 if(t->used < sizeof t->buffer) {
206 /* there is room left in the buffer */
207 where = (t->start + t->used) % sizeof t->buffer;
208 /* Get as much data as we can */
209 if(where >= t->start) left = (sizeof t->buffer) - where;
210 else left = t->start - where;
212 n = read(t->fd, t->buffer + where, left);
213 } while(n < 0 && errno == EINTR);
215 if(errno != EAGAIN) fatal(errno, "error reading sample stream");
219 D(("fill %s: eof detected", t->id));
225 if(t->used == sizeof t->buffer)
231 /** @brief Close the sound device
233 * This is called to deactivate the output device when pausing, and also by the
234 * ALSA backend when changing encoding (in which case the sound device will be
235 * immediately reactivated).
237 static void idle(void) {
239 if(backend->deactivate)
240 backend->deactivate();
242 device_state = device_closed;
246 /** @brief Abandon the current track */
248 struct speaker_message sm;
251 memset(&sm, 0, sizeof sm);
252 sm.type = SM_FINISHED;
253 strcpy(sm.id, playing->id);
254 speaker_send(1, &sm);
255 removetrack(playing->id);
260 /** @brief Enable sound output
262 * Makes sure the sound device is open and has the right sample format. Return
263 * 0 on success and -1 on error.
265 static void activate(void) {
266 if(backend->activate)
269 device_state = device_open;
272 /** @brief Check whether the current track has finished
274 * The current track is determined to have finished either if the input stream
275 * eded before the format could be determined (i.e. it is malformed) or the
276 * input is at end of file and there is less than a frame left unplayed. (So
277 * it copes with decoders that crash mid-frame.)
279 static void maybe_finished(void) {
282 && playing->used < bytes_per_frame(&config->sample_format))
286 /** @brief Return nonzero if we want to play some audio
288 * We want to play audio if there is a current track; and it is not paused; and
289 * it is playable according to the rules for @ref track::playable.
291 static int playable(void) {
294 && playing->playable;
297 /** @brief Play up to @p frames frames of audio
299 * It is always safe to call this function.
300 * - If @ref playing is 0 then it will just return
301 * - If @ref paused is non-0 then it will just return
302 * - If @ref device_state != @ref device_open then it will call activate() and
303 * return if it it fails.
304 * - If there is not enough audio to play then it play what is available.
306 * If there are not enough frames to play then whatever is available is played
307 * instead. It is up to mainloop() to ensure that speaker_play() is not called
308 * when unreasonably only an small amounts of data is available to play.
310 static void speaker_play(size_t frames) {
311 size_t avail_frames, avail_bytes, written_frames;
312 ssize_t written_bytes;
314 /* Make sure there's a track to play and it is not paused */
317 /* Make sure the output device is open */
318 if(device_state != device_open) {
320 if(device_state != device_open)
323 D(("play: play %zu/%zu%s %dHz %db %dc", frames, playing->used / bpf,
324 playing->eof ? " EOF" : "",
325 config->sample_format.rate,
326 config->sample_format.bits,
327 config->sample_format.channels));
328 /* Figure out how many frames there are available to write */
329 if(playing->start + playing->used > sizeof playing->buffer)
330 /* The ring buffer is currently wrapped, only play up to the wrap point */
331 avail_bytes = (sizeof playing->buffer) - playing->start;
333 /* The ring buffer is not wrapped, can play the lot */
334 avail_bytes = playing->used;
335 avail_frames = avail_bytes / bpf;
336 /* Only play up to the requested amount */
337 if(avail_frames > frames)
338 avail_frames = frames;
342 written_frames = backend->play(avail_frames);
343 written_bytes = written_frames * bpf;
344 /* written_bytes and written_frames had better both be set and correct by
346 playing->start += written_bytes;
347 playing->used -= written_bytes;
348 playing->played += written_frames;
349 /* If the pointer is at the end of the buffer (or the buffer is completely
350 * empty) wrap it back to the start. */
351 if(!playing->used || playing->start == (sizeof playing->buffer))
353 /* If the buffer emptied out mark the track as unplayably */
355 playing->playable = 0;
356 frames -= written_frames;
360 /* Notify the server what we're up to. */
361 static void report(void) {
362 struct speaker_message sm;
365 memset(&sm, 0, sizeof sm);
366 sm.type = paused ? SM_PAUSED : SM_PLAYING;
367 strcpy(sm.id, playing->id);
368 sm.data = playing->played / config->sample_format.rate;
369 speaker_send(1, &sm);
374 static void reap(int __attribute__((unused)) sig) {
379 cmdpid = waitpid(-1, &st, WNOHANG);
381 signal(SIGCHLD, reap);
384 int addfd(int fd, int events) {
387 fds[fdno].events = events;
393 /** @brief Table of speaker backends */
394 static const struct speaker_backend *backends[] = {
395 #if HAVE_ALSA_ASOUNDLIB_H
400 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
403 #if HAVE_SYS_SOUNDCARD_H
409 /** @brief Main event loop */
410 static void mainloop(void) {
412 struct speaker_message sm;
413 int n, fd, stdin_slot, timeout, listen_slot;
415 while(getppid() != 1) {
417 /* By default we will wait up to a second before thinking about current
420 /* Always ready for commands from the main server. */
421 stdin_slot = addfd(0, POLLIN);
422 /* Also always ready for inbound connections */
423 listen_slot = addfd(listenfd, POLLIN);
424 /* Try to read sample data for the currently playing track if there is
429 && playing->used < (sizeof playing->buffer))
430 playing->slot = addfd(playing->fd, POLLIN);
434 /* We want to play some audio. If the device is closed then we attempt
436 if(device_state == device_closed)
438 /* If the device is (now) open then we will wait up until it is ready for
439 * more. If something went wrong then we should have device_error
440 * instead, but the post-poll code will cope even if it's
442 if(device_state == device_open)
443 backend->beforepoll(&timeout);
445 /* If any other tracks don't have a full buffer, try to read sample data
446 * from them. We do this last of all, so that if we run out of slots,
447 * nothing important can't be monitored. */
448 for(t = tracks; t; t = t->next)
452 && t->used < sizeof t->buffer) {
453 t->slot = addfd(t->fd, POLLIN | POLLHUP);
457 /* Wait for something interesting to happen */
458 n = poll(fds, fdno, timeout);
460 if(errno == EINTR) continue;
461 fatal(errno, "error calling poll");
463 /* Play some sound before doing anything else */
465 /* We want to play some audio */
466 if(device_state == device_open) {
468 speaker_play(3 * FRAMES);
470 /* We must be in _closed or _error, and it should be the latter, but we
473 * We most likely timed out, so now is a good time to retry.
474 * speaker_play() knows to re-activate the device if necessary.
476 speaker_play(3 * FRAMES);
479 /* Perhaps a connection has arrived */
480 if(fds[listen_slot].revents & POLLIN) {
481 struct sockaddr_un addr;
482 socklen_t addrlen = sizeof addr;
486 if((fd = accept(listenfd, (struct sockaddr *)&addr, &addrlen)) >= 0) {
488 if(read(fd, &l, sizeof l) < 4) {
489 error(errno, "reading length from inbound connection");
491 } else if(l >= sizeof id) {
492 error(0, "id length too long");
494 } else if(read(fd, id, l) < (ssize_t)l) {
495 error(errno, "reading id from inbound connection");
499 D(("id %s fd %d", id, fd));
500 t = findtrack(id, 1/*create*/);
501 write(fd, "", 1); /* write an ack */
503 error(0, "%s: already got a connection", id);
507 t->fd = fd; /* yay */
511 error(errno, "accept");
513 /* Perhaps we have a command to process */
514 if(fds[stdin_slot].revents & POLLIN) {
515 /* There might (in theory) be several commands queued up, but in general
516 * this won't be the case, so we don't bother looping around to pick them
518 n = speaker_recv(0, &sm);
523 if(playing) fatal(0, "got SM_PLAY but already playing something");
524 t = findtrack(sm.id, 1);
525 D(("SM_PLAY %s fd %d", t->id, t->fd));
527 error(0, "cannot play track because no connection arrived");
529 /* We attempt to play straight away rather than going round the loop.
530 * speaker_play() is clever enough to perform any activation that is
532 speaker_play(3 * FRAMES);
544 /* As for SM_PLAY we attempt to play straight away. */
546 speaker_play(3 * FRAMES);
551 D(("SM_CANCEL %s", sm.id));
552 t = removetrack(sm.id);
555 sm.type = SM_FINISHED;
556 strcpy(sm.id, playing->id);
557 speaker_send(1, &sm);
562 error(0, "SM_CANCEL for unknown track %s", sm.id);
567 if(config_read(1)) error(0, "cannot read configuration");
568 info("reloaded configuration");
571 error(0, "unknown message type %d", sm.type);
574 /* Read in any buffered data */
575 for(t = tracks; t; t = t->next)
578 && (fds[t->slot].revents & (POLLIN | POLLHUP)))
580 /* Maybe we finished playing a track somewhere in the above */
582 /* If we don't need the sound device for now then close it for the benefit
583 * of anyone else who wants it. */
584 if((!playing || paused) && device_state == device_open)
586 /* If we've not reported out state for a second do so now. */
587 if(time(0) > last_report)
592 int main(int argc, char **argv) {
593 int n, logsyslog = !isatty(2);
594 struct sockaddr_un addr;
595 static const int one = 1;
596 struct speaker_message sm;
600 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
601 while((n = getopt_long(argc, argv, "hVc:dDSs", options, 0)) >= 0) {
605 case 'c': configfile = optarg; break;
606 case 'd': debugging = 1; break;
607 case 'D': debugging = 0; break;
608 case 'S': logsyslog = 0; break;
609 case 's': logsyslog = 1; break;
610 default: fatal(0, "invalid option");
613 if((d = getenv("DISORDER_DEBUG_SPEAKER"))) debugging = atoi(d);
615 openlog(progname, LOG_PID, LOG_DAEMON);
616 log_default = &log_syslog;
618 if(config_read(1)) fatal(0, "cannot read configuration");
619 bpf = bytes_per_frame(&config->sample_format);
621 signal(SIGPIPE, SIG_IGN);
623 signal(SIGCHLD, reap);
625 xnice(config->nice_speaker);
628 /* make sure we're not root, whatever the config says */
629 if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root");
630 /* identify the backend used to play */
631 for(n = 0; backends[n]; ++n)
632 if(backends[n]->backend == config->speaker_backend)
635 fatal(0, "unsupported backend %d", config->speaker_backend);
636 backend = backends[n];
637 /* backend-specific initialization */
639 /* set up the listen socket */
640 listenfd = xsocket(PF_UNIX, SOCK_STREAM, 0);
641 memset(&addr, 0, sizeof addr);
642 addr.sun_family = AF_UNIX;
643 snprintf(addr.sun_path, sizeof addr.sun_path, "%s/speaker",
645 if(unlink(addr.sun_path) < 0 && errno != ENOENT)
646 error(errno, "removing %s", addr.sun_path);
647 xsetsockopt(listenfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one);
648 if(bind(listenfd, (const struct sockaddr *)&addr, sizeof addr) < 0)
649 fatal(errno, "error binding socket to %s", addr.sun_path);
650 xlisten(listenfd, 128);
652 info("listening on %s", addr.sun_path);
653 memset(&sm, 0, sizeof sm);
655 speaker_send(1, &sm);
657 info("stopped (parent terminated)");