2 * This file is part of DisOrder
3 * Copyright (C) 2005, 2006, 2007 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 2 of the License, or
9 * (at your option) any later version.
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.
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
21 /** @file server/speaker.c
22 * @brief Speaker process
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
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.)
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).
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
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
66 #include <sys/select.h>
74 #include "configuration.h"
79 #include "speaker-protocol.h"
84 /** @brief Linked list of all prepared tracks */
87 /** @brief Playing track, or NULL */
88 struct track *playing;
90 /** @brief Number of bytes pre frame */
93 /** @brief Array of file descriptors for poll() */
94 struct pollfd fds[NFDS];
96 /** @brief Next free slot in @ref fds */
99 /** @brief Listen socket */
102 static time_t last_report; /* when we last reported */
103 static int paused; /* pause status */
105 /** @brief The current device state */
106 enum device_states device_state;
108 /** @brief Set when idled
110 * This is set when the sound device is deliberately closed by idle().
114 /** @brief Selected backend */
115 static const struct speaker_backend *backend;
117 static const struct option options[] = {
118 { "help", no_argument, 0, 'h' },
119 { "version", no_argument, 0, 'V' },
120 { "config", required_argument, 0, 'c' },
121 { "debug", no_argument, 0, 'd' },
122 { "no-debug", no_argument, 0, 'D' },
123 { "syslog", no_argument, 0, 's' },
124 { "no-syslog", no_argument, 0, 'S' },
128 /* Display usage message and terminate. */
129 static void help(void) {
131 " disorder-speaker [OPTIONS]\n"
133 " --help, -h Display usage message\n"
134 " --version, -V Display version number\n"
135 " --config PATH, -c PATH Set configuration file\n"
136 " --debug, -d Turn on debugging\n"
137 " --[no-]syslog Force logging\n"
139 "Speaker process for DisOrder. Not intended to be run\n"
145 /* Display version number and terminate. */
146 static void version(void) {
147 xprintf("%s", disorder_version_string);
152 /** @brief Return the number of bytes per frame in @p format */
153 static size_t bytes_per_frame(const struct stream_header *format) {
154 return format->channels * format->bits / 8;
157 /** @brief Find track @p id, maybe creating it if not found */
158 static struct track *findtrack(const char *id, int create) {
161 D(("findtrack %s %d", id, create));
162 for(t = tracks; t && strcmp(id, t->id); t = t->next)
165 t = xmalloc(sizeof *t);
174 /** @brief Remove track @p id (but do not destroy it) */
175 static struct track *removetrack(const char *id) {
176 struct track *t, **tt;
178 D(("removetrack %s", id));
179 for(tt = &tracks; (t = *tt) && strcmp(id, t->id); tt = &t->next)
186 /** @brief Destroy a track */
187 static void destroy(struct track *t) {
188 D(("destroy %s", t->id));
189 if(t->fd != -1) xclose(t->fd);
193 /** @brief Read data into a sample buffer
194 * @param t Pointer to track
195 * @return 0 on success, -1 on EOF
197 * This is effectively the read callback on @c t->fd. It is called from the
198 * main loop whenever the track's file descriptor is readable, assuming the
199 * buffer has not reached the maximum allowed occupancy.
201 static int speaker_fill(struct track *t) {
205 D(("fill %s: eof=%d used=%zu",
206 t->id, t->eof, t->used));
207 if(t->eof) return -1;
208 if(t->used < sizeof t->buffer) {
209 /* there is room left in the buffer */
210 where = (t->start + t->used) % sizeof t->buffer;
211 /* Get as much data as we can */
212 if(where >= t->start) left = (sizeof t->buffer) - where;
213 else left = t->start - where;
215 n = read(t->fd, t->buffer + where, left);
216 } while(n < 0 && errno == EINTR);
218 if(errno != EAGAIN) fatal(errno, "error reading sample stream");
222 D(("fill %s: eof detected", t->id));
228 if(t->used == sizeof t->buffer)
234 /** @brief Close the sound device
236 * This is called to deactivate the output device when pausing, and also by the
237 * ALSA backend when changing encoding (in which case the sound device will be
238 * immediately reactivated).
240 static void idle(void) {
242 if(backend->deactivate)
243 backend->deactivate();
245 device_state = device_closed;
249 /** @brief Abandon the current track */
251 struct speaker_message sm;
254 memset(&sm, 0, sizeof sm);
255 sm.type = SM_FINISHED;
256 strcpy(sm.id, playing->id);
257 speaker_send(1, &sm);
258 removetrack(playing->id);
263 /** @brief Enable sound output
265 * Makes sure the sound device is open and has the right sample format. Return
266 * 0 on success and -1 on error.
268 static void activate(void) {
269 if(backend->activate)
272 device_state = device_open;
275 /** @brief Check whether the current track has finished
277 * The current track is determined to have finished either if the input stream
278 * eded before the format could be determined (i.e. it is malformed) or the
279 * input is at end of file and there is less than a frame left unplayed. (So
280 * it copes with decoders that crash mid-frame.)
282 static void maybe_finished(void) {
285 && playing->used < bytes_per_frame(&config->sample_format))
289 /** @brief Return nonzero if we want to play some audio
291 * We want to play audio if there is a current track; and it is not paused; and
292 * it is playable according to the rules for @ref track::playable.
294 static int playable(void) {
297 && playing->playable;
300 /** @brief Play up to @p frames frames of audio
302 * It is always safe to call this function.
303 * - If @ref playing is 0 then it will just return
304 * - If @ref paused is non-0 then it will just return
305 * - If @ref device_state != @ref device_open then it will call activate() and
306 * return if it it fails.
307 * - If there is not enough audio to play then it play what is available.
309 * If there are not enough frames to play then whatever is available is played
310 * instead. It is up to mainloop() to ensure that speaker_play() is not called
311 * when unreasonably only an small amounts of data is available to play.
313 static void speaker_play(size_t frames) {
314 size_t avail_frames, avail_bytes, written_frames;
315 ssize_t written_bytes;
317 /* Make sure there's a track to play and it is not paused */
320 /* Make sure the output device is open */
321 if(device_state != device_open) {
323 if(device_state != device_open)
326 D(("play: play %zu/%zu%s %dHz %db %dc", frames, playing->used / bpf,
327 playing->eof ? " EOF" : "",
328 config->sample_format.rate,
329 config->sample_format.bits,
330 config->sample_format.channels));
331 /* Figure out how many frames there are available to write */
332 if(playing->start + playing->used > sizeof playing->buffer)
333 /* The ring buffer is currently wrapped, only play up to the wrap point */
334 avail_bytes = (sizeof playing->buffer) - playing->start;
336 /* The ring buffer is not wrapped, can play the lot */
337 avail_bytes = playing->used;
338 avail_frames = avail_bytes / bpf;
339 /* Only play up to the requested amount */
340 if(avail_frames > frames)
341 avail_frames = frames;
345 written_frames = backend->play(avail_frames);
346 written_bytes = written_frames * bpf;
347 /* written_bytes and written_frames had better both be set and correct by
349 playing->start += written_bytes;
350 playing->used -= written_bytes;
351 playing->played += written_frames;
352 /* If the pointer is at the end of the buffer (or the buffer is completely
353 * empty) wrap it back to the start. */
354 if(!playing->used || playing->start == (sizeof playing->buffer))
356 /* If the buffer emptied out mark the track as unplayably */
357 if(!playing->used && !playing->eof) {
358 error(0, "track buffer emptied");
359 playing->playable = 0;
361 frames -= written_frames;
365 /* Notify the server what we're up to. */
366 static void report(void) {
367 struct speaker_message sm;
370 memset(&sm, 0, sizeof sm);
371 sm.type = paused ? SM_PAUSED : SM_PLAYING;
372 strcpy(sm.id, playing->id);
373 sm.data = playing->played / config->sample_format.rate;
374 speaker_send(1, &sm);
379 static void reap(int __attribute__((unused)) sig) {
384 cmdpid = waitpid(-1, &st, WNOHANG);
386 signal(SIGCHLD, reap);
389 int addfd(int fd, int events) {
392 fds[fdno].events = events;
398 /** @brief Table of speaker backends */
399 static const struct speaker_backend *backends[] = {
400 #if HAVE_ALSA_ASOUNDLIB_H
405 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
408 #if HAVE_SYS_SOUNDCARD_H
414 /** @brief Main event loop */
415 static void mainloop(void) {
417 struct speaker_message sm;
418 int n, fd, stdin_slot, timeout, listen_slot;
420 while(getppid() != 1) {
422 /* By default we will wait up to a second before thinking about current
425 /* Always ready for commands from the main server. */
426 stdin_slot = addfd(0, POLLIN);
427 /* Also always ready for inbound connections */
428 listen_slot = addfd(listenfd, POLLIN);
429 /* Try to read sample data for the currently playing track if there is
434 && playing->used < (sizeof playing->buffer))
435 playing->slot = addfd(playing->fd, POLLIN);
439 /* We want to play some audio. If the device is closed then we attempt
441 if(device_state == device_closed)
443 /* If the device is (now) open then we will wait up until it is ready for
444 * more. If something went wrong then we should have device_error
445 * instead, but the post-poll code will cope even if it's
447 if(device_state == device_open)
448 backend->beforepoll(&timeout);
450 /* If any other tracks don't have a full buffer, try to read sample data
451 * from them. We do this last of all, so that if we run out of slots,
452 * nothing important can't be monitored. */
453 for(t = tracks; t; t = t->next)
457 && t->used < sizeof t->buffer) {
458 t->slot = addfd(t->fd, POLLIN | POLLHUP);
462 /* Wait for something interesting to happen */
463 n = poll(fds, fdno, timeout);
465 if(errno == EINTR) continue;
466 fatal(errno, "error calling poll");
468 /* Play some sound before doing anything else */
470 /* We want to play some audio */
471 if(device_state == device_open) {
473 speaker_play(3 * FRAMES);
475 /* We must be in _closed or _error, and it should be the latter, but we
478 * We most likely timed out, so now is a good time to retry.
479 * speaker_play() knows to re-activate the device if necessary.
481 speaker_play(3 * FRAMES);
484 /* Perhaps a connection has arrived */
485 if(fds[listen_slot].revents & POLLIN) {
486 struct sockaddr_un addr;
487 socklen_t addrlen = sizeof addr;
491 if((fd = accept(listenfd, (struct sockaddr *)&addr, &addrlen)) >= 0) {
493 if(read(fd, &l, sizeof l) < 4) {
494 error(errno, "reading length from inbound connection");
496 } else if(l >= sizeof id) {
497 error(0, "id length too long");
499 } else if(read(fd, id, l) < (ssize_t)l) {
500 error(errno, "reading id from inbound connection");
504 D(("id %s fd %d", id, fd));
505 t = findtrack(id, 1/*create*/);
506 write(fd, "", 1); /* write an ack */
508 error(0, "%s: already got a connection", id);
512 t->fd = fd; /* yay */
516 error(errno, "accept");
518 /* Perhaps we have a command to process */
519 if(fds[stdin_slot].revents & POLLIN) {
520 /* There might (in theory) be several commands queued up, but in general
521 * this won't be the case, so we don't bother looping around to pick them
523 n = speaker_recv(0, &sm);
528 if(playing) fatal(0, "got SM_PLAY but already playing something");
529 t = findtrack(sm.id, 1);
530 D(("SM_PLAY %s fd %d", t->id, t->fd));
532 error(0, "cannot play track because no connection arrived");
534 /* We attempt to play straight away rather than going round the loop.
535 * speaker_play() is clever enough to perform any activation that is
537 speaker_play(3 * FRAMES);
549 /* As for SM_PLAY we attempt to play straight away. */
551 speaker_play(3 * FRAMES);
556 D(("SM_CANCEL %s", sm.id));
557 t = removetrack(sm.id);
560 sm.type = SM_FINISHED;
561 strcpy(sm.id, playing->id);
562 speaker_send(1, &sm);
567 sm.type = SM_UNKNOWN;
568 speaker_send(1, &sm);
569 error(0, "SM_CANCEL for unknown track %s", sm.id);
575 if(config_read(1)) error(0, "cannot read configuration");
576 info("reloaded configuration");
579 error(0, "unknown message type %d", sm.type);
582 /* Read in any buffered data */
583 for(t = tracks; t; t = t->next)
586 && (fds[t->slot].revents & (POLLIN | POLLHUP)))
588 /* Maybe we finished playing a track somewhere in the above */
590 /* If we don't need the sound device for now then close it for the benefit
591 * of anyone else who wants it. */
592 if((!playing || paused) && device_state == device_open)
594 /* If we've not reported out state for a second do so now. */
595 if(time(0) > last_report)
600 int main(int argc, char **argv) {
601 int n, logsyslog = !isatty(2);
602 struct sockaddr_un addr;
603 static const int one = 1;
604 struct speaker_message sm;
609 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
610 while((n = getopt_long(argc, argv, "hVc:dDSs", options, 0)) >= 0) {
614 case 'c': configfile = optarg; break;
615 case 'd': debugging = 1; break;
616 case 'D': debugging = 0; break;
617 case 'S': logsyslog = 0; break;
618 case 's': logsyslog = 1; break;
619 default: fatal(0, "invalid option");
622 if((d = getenv("DISORDER_DEBUG_SPEAKER"))) debugging = atoi(d);
624 openlog(progname, LOG_PID, LOG_DAEMON);
625 log_default = &log_syslog;
627 if(config_read(1)) fatal(0, "cannot read configuration");
628 bpf = bytes_per_frame(&config->sample_format);
630 signal(SIGPIPE, SIG_IGN);
632 signal(SIGCHLD, reap);
634 xnice(config->nice_speaker);
637 /* make sure we're not root, whatever the config says */
638 if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root");
639 /* identify the backend used to play */
640 for(n = 0; backends[n]; ++n)
641 if(backends[n]->backend == config->api)
644 fatal(0, "unsupported api %d", config->api);
645 backend = backends[n];
646 /* backend-specific initialization */
648 /* create the socket directory */
649 byte_xasprintf(&dir, "%s/speaker", config->home);
650 unlink(dir); /* might be a leftover socket */
651 if(mkdir(dir, 0700) < 0 && errno != EEXIST)
652 fatal(errno, "error creating %s", dir);
653 /* set up the listen socket */
654 listenfd = xsocket(PF_UNIX, SOCK_STREAM, 0);
655 memset(&addr, 0, sizeof addr);
656 addr.sun_family = AF_UNIX;
657 snprintf(addr.sun_path, sizeof addr.sun_path, "%s/speaker/socket",
659 if(unlink(addr.sun_path) < 0 && errno != ENOENT)
660 error(errno, "removing %s", addr.sun_path);
661 xsetsockopt(listenfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one);
662 if(bind(listenfd, (const struct sockaddr *)&addr, sizeof addr) < 0)
663 fatal(errno, "error binding socket to %s", addr.sun_path);
664 xlisten(listenfd, 128);
666 info("listening on %s", addr.sun_path);
667 memset(&sm, 0, sizeof sm);
669 speaker_send(1, &sm);
671 info("stopped (parent terminated)");