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)");