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 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 7 * @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
61 #include <sys/select.h>
69 #include "configuration.h"
74 #include "speaker-protocol.h"
80 /** @brief Linked list of all prepared tracks */
83 /** @brief Playing track, or NULL */
84 struct track *playing;
86 /** @brief Number of bytes pre frame */
89 /** @brief Array of file descriptors for poll() */
90 struct pollfd fds[NFDS];
92 /** @brief Next free slot in @ref fds */
95 /** @brief Listen socket */
98 static time_t last_report; /* when we last reported */
99 static int paused; /* pause status */
101 /** @brief The current device state */
102 enum device_states device_state;
104 /** @brief Set when idled
106 * This is set when the sound device is deliberately closed by idle().
110 /** @brief Selected backend */
111 static const struct speaker_backend *backend;
113 static const struct option options[] = {
114 { "help", no_argument, 0, 'h' },
115 { "version", no_argument, 0, 'V' },
116 { "config", required_argument, 0, 'c' },
117 { "debug", no_argument, 0, 'd' },
118 { "no-debug", no_argument, 0, 'D' },
119 { "syslog", no_argument, 0, 's' },
120 { "no-syslog", no_argument, 0, 'S' },
124 /* Display usage message and terminate. */
125 static void help(void) {
127 " disorder-speaker [OPTIONS]\n"
129 " --help, -h Display usage message\n"
130 " --version, -V Display version number\n"
131 " --config PATH, -c PATH Set configuration file\n"
132 " --debug, -d Turn on debugging\n"
133 " --[no-]syslog Force logging\n"
135 "Speaker process for DisOrder. Not intended to be run\n"
141 /** @brief Return the number of bytes per frame in @p format */
142 static size_t bytes_per_frame(const struct stream_header *format) {
143 return format->channels * format->bits / 8;
146 /** @brief Find track @p id, maybe creating it if not found */
147 static struct track *findtrack(const char *id, int create) {
150 D(("findtrack %s %d", id, create));
151 for(t = tracks; t && strcmp(id, t->id); t = t->next)
154 t = xmalloc(sizeof *t);
163 /** @brief Remove track @p id (but do not destroy it) */
164 static struct track *removetrack(const char *id) {
165 struct track *t, **tt;
167 D(("removetrack %s", id));
168 for(tt = &tracks; (t = *tt) && strcmp(id, t->id); tt = &t->next)
175 /** @brief Destroy a track */
176 static void destroy(struct track *t) {
177 D(("destroy %s", t->id));
178 if(t->fd != -1) xclose(t->fd);
182 /** @brief Read data into a sample buffer
183 * @param t Pointer to track
184 * @return 0 on success, -1 on EOF
186 * This is effectively the read callback on @c t->fd. It is called from the
187 * main loop whenever the track's file descriptor is readable, assuming the
188 * buffer has not reached the maximum allowed occupancy.
190 static int speaker_fill(struct track *t) {
194 D(("fill %s: eof=%d used=%zu",
195 t->id, t->eof, t->used));
196 if(t->eof) return -1;
197 if(t->used < sizeof t->buffer) {
198 /* there is room left in the buffer */
199 where = (t->start + t->used) % sizeof t->buffer;
200 /* Get as much data as we can */
201 if(where >= t->start) left = (sizeof t->buffer) - where;
202 else left = t->start - where;
204 n = read(t->fd, t->buffer + where, left);
205 } while(n < 0 && errno == EINTR);
207 if(errno != EAGAIN) fatal(errno, "error reading sample stream");
211 D(("fill %s: eof detected", t->id));
217 if(t->used == sizeof t->buffer)
223 /** @brief Close the sound device
225 * This is called to deactivate the output device when pausing, and also by the
226 * ALSA backend when changing encoding (in which case the sound device will be
227 * immediately reactivated).
229 static void idle(void) {
231 if(backend->deactivate)
232 backend->deactivate();
234 device_state = device_closed;
238 /** @brief Abandon the current track */
240 struct speaker_message sm;
243 memset(&sm, 0, sizeof sm);
244 sm.type = SM_FINISHED;
245 strcpy(sm.id, playing->id);
246 speaker_send(1, &sm);
247 removetrack(playing->id);
252 /** @brief Enable sound output
254 * Makes sure the sound device is open and has the right sample format. Return
255 * 0 on success and -1 on error.
257 static void activate(void) {
258 if(backend->activate)
261 device_state = device_open;
264 /** @brief Check whether the current track has finished
266 * The current track is determined to have finished either if the input stream
267 * eded before the format could be determined (i.e. it is malformed) or the
268 * input is at end of file and there is less than a frame left unplayed. (So
269 * it copes with decoders that crash mid-frame.)
271 static void maybe_finished(void) {
274 && playing->used < bytes_per_frame(&config->sample_format))
278 /** @brief Return nonzero if we want to play some audio
280 * We want to play audio if there is a current track; and it is not paused; and
281 * it is playable according to the rules for @ref track::playable.
283 static int playable(void) {
286 && playing->playable;
289 /** @brief Play up to @p frames frames of audio
291 * It is always safe to call this function.
292 * - If @ref playing is 0 then it will just return
293 * - If @ref paused is non-0 then it will just return
294 * - If @ref device_state != @ref device_open then it will call activate() and
295 * return if it it fails.
296 * - If there is not enough audio to play then it play what is available.
298 * If there are not enough frames to play then whatever is available is played
299 * instead. It is up to mainloop() to ensure that speaker_play() is not called
300 * when unreasonably only an small amounts of data is available to play.
302 static void speaker_play(size_t frames) {
303 size_t avail_frames, avail_bytes, written_frames;
304 ssize_t written_bytes;
306 /* Make sure there's a track to play and it is not paused */
309 /* Make sure the output device is open */
310 if(device_state != device_open) {
312 if(device_state != device_open)
315 D(("play: play %zu/%zu%s %dHz %db %dc", frames, playing->used / bpf,
316 playing->eof ? " EOF" : "",
317 config->sample_format.rate,
318 config->sample_format.bits,
319 config->sample_format.channels));
320 /* Figure out how many frames there are available to write */
321 if(playing->start + playing->used > sizeof playing->buffer)
322 /* The ring buffer is currently wrapped, only play up to the wrap point */
323 avail_bytes = (sizeof playing->buffer) - playing->start;
325 /* The ring buffer is not wrapped, can play the lot */
326 avail_bytes = playing->used;
327 avail_frames = avail_bytes / bpf;
328 /* Only play up to the requested amount */
329 if(avail_frames > frames)
330 avail_frames = frames;
334 written_frames = backend->play(avail_frames);
335 written_bytes = written_frames * bpf;
336 /* written_bytes and written_frames had better both be set and correct by
338 playing->start += written_bytes;
339 playing->used -= written_bytes;
340 playing->played += written_frames;
341 /* If the pointer is at the end of the buffer (or the buffer is completely
342 * empty) wrap it back to the start. */
343 if(!playing->used || playing->start == (sizeof playing->buffer))
345 /* If the buffer emptied out mark the track as unplayably */
346 if(!playing->used && !playing->eof) {
347 error(0, "track buffer emptied");
348 playing->playable = 0;
350 frames -= written_frames;
354 /* Notify the server what we're up to. */
355 static void report(void) {
356 struct speaker_message sm;
359 memset(&sm, 0, sizeof sm);
360 sm.type = paused ? SM_PAUSED : SM_PLAYING;
361 strcpy(sm.id, playing->id);
362 sm.data = playing->played / config->sample_format.rate;
363 speaker_send(1, &sm);
368 static void reap(int __attribute__((unused)) sig) {
373 cmdpid = waitpid(-1, &st, WNOHANG);
375 signal(SIGCHLD, reap);
378 int addfd(int fd, int events) {
381 fds[fdno].events = events;
387 /** @brief Table of speaker backends */
388 static const struct speaker_backend *backends[] = {
389 #if HAVE_ALSA_ASOUNDLIB_H
394 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
397 #if HAVE_SYS_SOUNDCARD_H
403 /** @brief Main event loop */
404 static void mainloop(void) {
406 struct speaker_message sm;
407 int n, fd, stdin_slot, timeout, listen_slot;
409 while(getppid() != 1) {
411 /* By default we will wait up to a second before thinking about current
414 /* Always ready for commands from the main server. */
415 stdin_slot = addfd(0, POLLIN);
416 /* Also always ready for inbound connections */
417 listen_slot = addfd(listenfd, POLLIN);
418 /* Try to read sample data for the currently playing track if there is
423 && playing->used < (sizeof playing->buffer))
424 playing->slot = addfd(playing->fd, POLLIN);
428 /* We want to play some audio. If the device is closed then we attempt
430 if(device_state == device_closed)
432 /* If the device is (now) open then we will wait up until it is ready for
433 * more. If something went wrong then we should have device_error
434 * instead, but the post-poll code will cope even if it's
436 if(device_state == device_open)
437 backend->beforepoll(&timeout);
439 /* If any other tracks don't have a full buffer, try to read sample data
440 * from them. We do this last of all, so that if we run out of slots,
441 * nothing important can't be monitored. */
442 for(t = tracks; t; t = t->next)
446 && t->used < sizeof t->buffer) {
447 t->slot = addfd(t->fd, POLLIN | POLLHUP);
451 /* Wait for something interesting to happen */
452 n = poll(fds, fdno, timeout);
454 if(errno == EINTR) continue;
455 fatal(errno, "error calling poll");
457 /* Play some sound before doing anything else */
459 /* We want to play some audio */
460 if(device_state == device_open) {
462 speaker_play(3 * FRAMES);
464 /* We must be in _closed or _error, and it should be the latter, but we
467 * We most likely timed out, so now is a good time to retry.
468 * speaker_play() knows to re-activate the device if necessary.
470 speaker_play(3 * FRAMES);
473 /* Perhaps a connection has arrived */
474 if(fds[listen_slot].revents & POLLIN) {
475 struct sockaddr_un addr;
476 socklen_t addrlen = sizeof addr;
480 if((fd = accept(listenfd, (struct sockaddr *)&addr, &addrlen)) >= 0) {
482 if(read(fd, &l, sizeof l) < 4) {
483 error(errno, "reading length from inbound connection");
485 } else if(l >= sizeof id) {
486 error(0, "id length too long");
488 } else if(read(fd, id, l) < (ssize_t)l) {
489 error(errno, "reading id from inbound connection");
493 D(("id %s fd %d", id, fd));
494 t = findtrack(id, 1/*create*/);
495 write(fd, "", 1); /* write an ack */
497 error(0, "%s: already got a connection", id);
501 t->fd = fd; /* yay */
505 error(errno, "accept");
507 /* Perhaps we have a command to process */
508 if(fds[stdin_slot].revents & POLLIN) {
509 /* There might (in theory) be several commands queued up, but in general
510 * this won't be the case, so we don't bother looping around to pick them
512 n = speaker_recv(0, &sm);
517 if(playing) fatal(0, "got SM_PLAY but already playing something");
518 t = findtrack(sm.id, 1);
519 D(("SM_PLAY %s fd %d", t->id, t->fd));
521 error(0, "cannot play track because no connection arrived");
523 /* We attempt to play straight away rather than going round the loop.
524 * speaker_play() is clever enough to perform any activation that is
526 speaker_play(3 * FRAMES);
538 /* As for SM_PLAY we attempt to play straight away. */
540 speaker_play(3 * FRAMES);
545 D(("SM_CANCEL %s", sm.id));
546 t = removetrack(sm.id);
549 /* scratching the playing track */
550 sm.type = SM_FINISHED;
553 /* Could be scratching the playing track before it's quite got
554 * going, or could be just removing a track from the queue. We
555 * log more because there's been a bug here recently than because
556 * it's particularly interesting; the log message will be removed
557 * if no further problems show up. */
558 info("SM_CANCEL for nonplaying track %s", sm.id);
559 sm.type = SM_STILLBORN;
561 strcpy(sm.id, t->id);
564 /* Probably scratching the playing track well before it's got
565 * going, but could indicate a bug, so we log this as an error. */
566 sm.type = SM_UNKNOWN;
567 error(0, "SM_CANCEL for unknown track %s", sm.id);
569 speaker_send(1, &sm);
574 if(config_read(1)) error(0, "cannot read configuration");
575 info("reloaded configuration");
578 error(0, "unknown message type %d", sm.type);
581 /* Read in any buffered data */
582 for(t = tracks; t; t = t->next)
585 && (fds[t->slot].revents & (POLLIN | POLLHUP)))
587 /* Maybe we finished playing a track somewhere in the above */
589 /* If we don't need the sound device for now then close it for the benefit
590 * of anyone else who wants it. */
591 if((!playing || paused) && device_state == device_open)
593 /* If we've not reported out state for a second do so now. */
594 if(time(0) > last_report)
599 int main(int argc, char **argv) {
600 int n, logsyslog = !isatty(2);
601 struct sockaddr_un addr;
602 static const int one = 1;
603 struct speaker_message sm;
608 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
609 while((n = getopt_long(argc, argv, "hVc:dDSs", options, 0)) >= 0) {
612 case 'V': version("disorder-speaker");
613 case 'c': configfile = optarg; break;
614 case 'd': debugging = 1; break;
615 case 'D': debugging = 0; break;
616 case 'S': logsyslog = 0; break;
617 case 's': logsyslog = 1; break;
618 default: fatal(0, "invalid option");
621 if((d = getenv("DISORDER_DEBUG_SPEAKER"))) debugging = atoi(d);
623 openlog(progname, LOG_PID, LOG_DAEMON);
624 log_default = &log_syslog;
626 if(config_read(1)) fatal(0, "cannot read configuration");
627 bpf = bytes_per_frame(&config->sample_format);
629 signal(SIGPIPE, SIG_IGN);
631 signal(SIGCHLD, reap);
633 xnice(config->nice_speaker);
636 /* make sure we're not root, whatever the config says */
637 if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root");
638 /* identify the backend used to play */
639 for(n = 0; backends[n]; ++n)
640 if(backends[n]->backend == config->api)
643 fatal(0, "unsupported api %d", config->api);
644 backend = backends[n];
645 /* backend-specific initialization */
647 /* create the socket directory */
648 byte_xasprintf(&dir, "%s/speaker", config->home);
649 unlink(dir); /* might be a leftover socket */
650 if(mkdir(dir, 0700) < 0 && errno != EEXIST)
651 fatal(errno, "error creating %s", dir);
652 /* set up the listen socket */
653 listenfd = xsocket(PF_UNIX, SOCK_STREAM, 0);
654 memset(&addr, 0, sizeof addr);
655 addr.sun_family = AF_UNIX;
656 snprintf(addr.sun_path, sizeof addr.sun_path, "%s/speaker/socket",
658 if(unlink(addr.sun_path) < 0 && errno != ENOENT)
659 error(errno, "removing %s", addr.sun_path);
660 xsetsockopt(listenfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one);
661 if(bind(listenfd, (const struct sockaddr *)&addr, sizeof addr) < 0)
662 fatal(errno, "error binding socket to %s", addr.sun_path);
663 xlisten(listenfd, 128);
665 info("listening on %s", addr.sun_path);
666 memset(&sm, 0, sizeof sm);
668 speaker_send(1, &sm);
670 info("stopped (parent terminated)");