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 error(0, "SM_CANCEL for unknown track %s", sm
.id
);
572 if(config_read(1)) error(0, "cannot read configuration");
573 info("reloaded configuration");
576 error(0, "unknown message type %d", sm
.type
);
579 /* Read in any buffered data */
580 for(t
= tracks
; t
; t
= t
->next
)
583 && (fds
[t
->slot
].revents
& (POLLIN
| POLLHUP
)))
585 /* Maybe we finished playing a track somewhere in the above */
587 /* If we don't need the sound device for now then close it for the benefit
588 * of anyone else who wants it. */
589 if((!playing
|| paused
) && device_state
== device_open
)
591 /* If we've not reported out state for a second do so now. */
592 if(time(0) > last_report
)
597 int main(int argc
, char **argv
) {
598 int n
, logsyslog
= !isatty(2);
599 struct sockaddr_un addr
;
600 static const int one
= 1;
601 struct speaker_message sm
;
606 if(!setlocale(LC_CTYPE
, "")) fatal(errno
, "error calling setlocale");
607 while((n
= getopt_long(argc
, argv
, "hVc:dDSs", options
, 0)) >= 0) {
611 case 'c': configfile
= optarg
; break;
612 case 'd': debugging
= 1; break;
613 case 'D': debugging
= 0; break;
614 case 'S': logsyslog
= 0; break;
615 case 's': logsyslog
= 1; break;
616 default: fatal(0, "invalid option");
619 if((d
= getenv("DISORDER_DEBUG_SPEAKER"))) debugging
= atoi(d
);
621 openlog(progname
, LOG_PID
, LOG_DAEMON
);
622 log_default
= &log_syslog
;
624 if(config_read(1)) fatal(0, "cannot read configuration");
625 bpf
= bytes_per_frame(&config
->sample_format
);
627 signal(SIGPIPE
, SIG_IGN
);
629 signal(SIGCHLD
, reap
);
631 xnice(config
->nice_speaker
);
634 /* make sure we're not root, whatever the config says */
635 if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root");
636 /* identify the backend used to play */
637 for(n
= 0; backends
[n
]; ++n
)
638 if(backends
[n
]->backend
== config
->speaker_backend
)
641 fatal(0, "unsupported backend %d", config
->speaker_backend
);
642 backend
= backends
[n
];
643 /* backend-specific initialization */
645 /* create the socket directory */
646 byte_xasprintf(&dir
, "%s/speaker", config
->home
);
647 unlink(dir
); /* might be a leftover socket */
648 if(mkdir(dir
, 0700) < 0 && errno
!= EEXIST
)
649 fatal(errno
, "error creating %s", dir
);
650 /* set up the listen socket */
651 listenfd
= xsocket(PF_UNIX
, SOCK_STREAM
, 0);
652 memset(&addr
, 0, sizeof addr
);
653 addr
.sun_family
= AF_UNIX
;
654 snprintf(addr
.sun_path
, sizeof addr
.sun_path
, "%s/speaker/socket",
656 if(unlink(addr
.sun_path
) < 0 && errno
!= ENOENT
)
657 error(errno
, "removing %s", addr
.sun_path
);
658 xsetsockopt(listenfd
, SOL_SOCKET
, SO_REUSEADDR
, &one
, sizeof one
);
659 if(bind(listenfd
, (const struct sockaddr
*)&addr
, sizeof addr
) < 0)
660 fatal(errno
, "error binding socket to %s", addr
.sun_path
);
661 xlisten(listenfd
, 128);
663 info("listening on %s", addr
.sun_path
);
664 memset(&sm
, 0, sizeof sm
);
666 speaker_send(1, &sm
);
668 info("stopped (parent terminated)");