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 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
66 #include <sys/select.h>
74 #include "configuration.h"
79 #include "speaker-protocol.h"
85 /** @brief Linked list of all prepared tracks */
88 /** @brief Playing track, or NULL */
89 struct track
*playing
;
91 /** @brief Number of bytes pre frame */
94 /** @brief Array of file descriptors for poll() */
95 struct pollfd fds
[NFDS
];
97 /** @brief Next free slot in @ref fds */
100 /** @brief Listen socket */
103 static time_t last_report
; /* when we last reported */
104 static int paused
; /* pause status */
106 /** @brief The current device state */
107 enum device_states device_state
;
109 /** @brief Set when idled
111 * This is set when the sound device is deliberately closed by idle().
115 /** @brief Selected backend */
116 static const struct speaker_backend
*backend
;
118 static const struct option options
[] = {
119 { "help", no_argument
, 0, 'h' },
120 { "version", no_argument
, 0, 'V' },
121 { "config", required_argument
, 0, 'c' },
122 { "debug", no_argument
, 0, 'd' },
123 { "no-debug", no_argument
, 0, 'D' },
124 { "syslog", no_argument
, 0, 's' },
125 { "no-syslog", no_argument
, 0, 'S' },
129 /* Display usage message and terminate. */
130 static void help(void) {
132 " disorder-speaker [OPTIONS]\n"
134 " --help, -h Display usage message\n"
135 " --version, -V Display version number\n"
136 " --config PATH, -c PATH Set configuration file\n"
137 " --debug, -d Turn on debugging\n"
138 " --[no-]syslog Force logging\n"
140 "Speaker process for DisOrder. Not intended to be run\n"
146 /** @brief Return the number of bytes per frame in @p format */
147 static size_t bytes_per_frame(const struct stream_header
*format
) {
148 return format
->channels
* format
->bits
/ 8;
151 /** @brief Find track @p id, maybe creating it if not found */
152 static struct track
*findtrack(const char *id
, int create
) {
155 D(("findtrack %s %d", id
, create
));
156 for(t
= tracks
; t
&& strcmp(id
, t
->id
); t
= t
->next
)
159 t
= xmalloc(sizeof *t
);
168 /** @brief Remove track @p id (but do not destroy it) */
169 static struct track
*removetrack(const char *id
) {
170 struct track
*t
, **tt
;
172 D(("removetrack %s", id
));
173 for(tt
= &tracks
; (t
= *tt
) && strcmp(id
, t
->id
); tt
= &t
->next
)
180 /** @brief Destroy a track */
181 static void destroy(struct track
*t
) {
182 D(("destroy %s", t
->id
));
183 if(t
->fd
!= -1) xclose(t
->fd
);
187 /** @brief Read data into a sample buffer
188 * @param t Pointer to track
189 * @return 0 on success, -1 on EOF
191 * This is effectively the read callback on @c t->fd. It is called from the
192 * main loop whenever the track's file descriptor is readable, assuming the
193 * buffer has not reached the maximum allowed occupancy.
195 static int speaker_fill(struct track
*t
) {
199 D(("fill %s: eof=%d used=%zu",
200 t
->id
, t
->eof
, t
->used
));
201 if(t
->eof
) return -1;
202 if(t
->used
< sizeof t
->buffer
) {
203 /* there is room left in the buffer */
204 where
= (t
->start
+ t
->used
) % sizeof t
->buffer
;
205 /* Get as much data as we can */
206 if(where
>= t
->start
) left
= (sizeof t
->buffer
) - where
;
207 else left
= t
->start
- where
;
209 n
= read(t
->fd
, t
->buffer
+ where
, left
);
210 } while(n
< 0 && errno
== EINTR
);
212 if(errno
!= EAGAIN
) fatal(errno
, "error reading sample stream");
216 D(("fill %s: eof detected", t
->id
));
222 if(t
->used
== sizeof t
->buffer
)
228 /** @brief Close the sound device
230 * This is called to deactivate the output device when pausing, and also by the
231 * ALSA backend when changing encoding (in which case the sound device will be
232 * immediately reactivated).
234 static void idle(void) {
236 if(backend
->deactivate
)
237 backend
->deactivate();
239 device_state
= device_closed
;
243 /** @brief Abandon the current track */
245 struct speaker_message sm
;
248 memset(&sm
, 0, sizeof sm
);
249 sm
.type
= SM_FINISHED
;
250 strcpy(sm
.id
, playing
->id
);
251 speaker_send(1, &sm
);
252 removetrack(playing
->id
);
257 /** @brief Enable sound output
259 * Makes sure the sound device is open and has the right sample format. Return
260 * 0 on success and -1 on error.
262 static void activate(void) {
263 if(backend
->activate
)
266 device_state
= device_open
;
269 /** @brief Check whether the current track has finished
271 * The current track is determined to have finished either if the input stream
272 * eded before the format could be determined (i.e. it is malformed) or the
273 * input is at end of file and there is less than a frame left unplayed. (So
274 * it copes with decoders that crash mid-frame.)
276 static void maybe_finished(void) {
279 && playing
->used
< bytes_per_frame(&config
->sample_format
))
283 /** @brief Return nonzero if we want to play some audio
285 * We want to play audio if there is a current track; and it is not paused; and
286 * it is playable according to the rules for @ref track::playable.
288 static int playable(void) {
291 && playing
->playable
;
294 /** @brief Play up to @p frames frames of audio
296 * It is always safe to call this function.
297 * - If @ref playing is 0 then it will just return
298 * - If @ref paused is non-0 then it will just return
299 * - If @ref device_state != @ref device_open then it will call activate() and
300 * return if it it fails.
301 * - If there is not enough audio to play then it play what is available.
303 * If there are not enough frames to play then whatever is available is played
304 * instead. It is up to mainloop() to ensure that speaker_play() is not called
305 * when unreasonably only an small amounts of data is available to play.
307 static void speaker_play(size_t frames
) {
308 size_t avail_frames
, avail_bytes
, written_frames
;
309 ssize_t written_bytes
;
311 /* Make sure there's a track to play and it is not paused */
314 /* Make sure the output device is open */
315 if(device_state
!= device_open
) {
317 if(device_state
!= device_open
)
320 D(("play: play %zu/%zu%s %dHz %db %dc", frames
, playing
->used
/ bpf
,
321 playing
->eof ?
" EOF" : "",
322 config
->sample_format
.rate
,
323 config
->sample_format
.bits
,
324 config
->sample_format
.channels
));
325 /* Figure out how many frames there are available to write */
326 if(playing
->start
+ playing
->used
> sizeof playing
->buffer
)
327 /* The ring buffer is currently wrapped, only play up to the wrap point */
328 avail_bytes
= (sizeof playing
->buffer
) - playing
->start
;
330 /* The ring buffer is not wrapped, can play the lot */
331 avail_bytes
= playing
->used
;
332 avail_frames
= avail_bytes
/ bpf
;
333 /* Only play up to the requested amount */
334 if(avail_frames
> frames
)
335 avail_frames
= frames
;
339 written_frames
= backend
->play(avail_frames
);
340 written_bytes
= written_frames
* bpf
;
341 /* written_bytes and written_frames had better both be set and correct by
343 playing
->start
+= written_bytes
;
344 playing
->used
-= written_bytes
;
345 playing
->played
+= written_frames
;
346 /* If the pointer is at the end of the buffer (or the buffer is completely
347 * empty) wrap it back to the start. */
348 if(!playing
->used
|| playing
->start
== (sizeof playing
->buffer
))
350 /* If the buffer emptied out mark the track as unplayably */
351 if(!playing
->used
&& !playing
->eof
) {
352 error(0, "track buffer emptied");
353 playing
->playable
= 0;
355 frames
-= written_frames
;
359 /* Notify the server what we're up to. */
360 static void report(void) {
361 struct speaker_message sm
;
364 memset(&sm
, 0, sizeof sm
);
365 sm
.type
= paused ? SM_PAUSED
: SM_PLAYING
;
366 strcpy(sm
.id
, playing
->id
);
367 sm
.data
= playing
->played
/ config
->sample_format
.rate
;
368 speaker_send(1, &sm
);
373 static void reap(int __attribute__((unused
)) sig
) {
378 cmdpid
= waitpid(-1, &st
, WNOHANG
);
380 signal(SIGCHLD
, reap
);
383 int addfd(int fd
, int events
) {
386 fds
[fdno
].events
= events
;
392 /** @brief Table of speaker backends */
393 static const struct speaker_backend
*backends
[] = {
394 #if HAVE_ALSA_ASOUNDLIB_H
399 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
402 #if HAVE_SYS_SOUNDCARD_H
408 /** @brief Main event loop */
409 static void mainloop(void) {
411 struct speaker_message sm
;
412 int n
, fd
, stdin_slot
, timeout
, listen_slot
;
414 while(getppid() != 1) {
416 /* By default we will wait up to a second before thinking about current
419 /* Always ready for commands from the main server. */
420 stdin_slot
= addfd(0, POLLIN
);
421 /* Also always ready for inbound connections */
422 listen_slot
= addfd(listenfd
, POLLIN
);
423 /* Try to read sample data for the currently playing track if there is
428 && playing
->used
< (sizeof playing
->buffer
))
429 playing
->slot
= addfd(playing
->fd
, POLLIN
);
433 /* We want to play some audio. If the device is closed then we attempt
435 if(device_state
== device_closed
)
437 /* If the device is (now) open then we will wait up until it is ready for
438 * more. If something went wrong then we should have device_error
439 * instead, but the post-poll code will cope even if it's
441 if(device_state
== device_open
)
442 backend
->beforepoll(&timeout
);
444 /* If any other tracks don't have a full buffer, try to read sample data
445 * from them. We do this last of all, so that if we run out of slots,
446 * nothing important can't be monitored. */
447 for(t
= tracks
; t
; t
= t
->next
)
451 && t
->used
< sizeof t
->buffer
) {
452 t
->slot
= addfd(t
->fd
, POLLIN
| POLLHUP
);
456 /* Wait for something interesting to happen */
457 n
= poll(fds
, fdno
, timeout
);
459 if(errno
== EINTR
) continue;
460 fatal(errno
, "error calling poll");
462 /* Play some sound before doing anything else */
464 /* We want to play some audio */
465 if(device_state
== device_open
) {
467 speaker_play(3 * FRAMES
);
469 /* We must be in _closed or _error, and it should be the latter, but we
472 * We most likely timed out, so now is a good time to retry.
473 * speaker_play() knows to re-activate the device if necessary.
475 speaker_play(3 * FRAMES
);
478 /* Perhaps a connection has arrived */
479 if(fds
[listen_slot
].revents
& POLLIN
) {
480 struct sockaddr_un addr
;
481 socklen_t addrlen
= sizeof addr
;
485 if((fd
= accept(listenfd
, (struct sockaddr
*)&addr
, &addrlen
)) >= 0) {
487 if(read(fd
, &l
, sizeof l
) < 4) {
488 error(errno
, "reading length from inbound connection");
490 } else if(l
>= sizeof id
) {
491 error(0, "id length too long");
493 } else if(read(fd
, id
, l
) < (ssize_t
)l
) {
494 error(errno
, "reading id from inbound connection");
498 D(("id %s fd %d", id
, fd
));
499 t
= findtrack(id
, 1/*create*/);
500 write(fd
, "", 1); /* write an ack */
502 error(0, "%s: already got a connection", id
);
506 t
->fd
= fd
; /* yay */
510 error(errno
, "accept");
512 /* Perhaps we have a command to process */
513 if(fds
[stdin_slot
].revents
& POLLIN
) {
514 /* There might (in theory) be several commands queued up, but in general
515 * this won't be the case, so we don't bother looping around to pick them
517 n
= speaker_recv(0, &sm
);
522 if(playing
) fatal(0, "got SM_PLAY but already playing something");
523 t
= findtrack(sm
.id
, 1);
524 D(("SM_PLAY %s fd %d", t
->id
, t
->fd
));
526 error(0, "cannot play track because no connection arrived");
528 /* We attempt to play straight away rather than going round the loop.
529 * speaker_play() is clever enough to perform any activation that is
531 speaker_play(3 * FRAMES
);
543 /* As for SM_PLAY we attempt to play straight away. */
545 speaker_play(3 * FRAMES
);
550 D(("SM_CANCEL %s", sm
.id
));
551 t
= removetrack(sm
.id
);
554 /* scratching the playing track */
555 sm
.type
= SM_FINISHED
;
558 /* Could be scratching the playing track before it's quite got
559 * going, or could be just removing a track from the queue. We
560 * log more because there's been a bug here recently than because
561 * it's particularly interesting; the log message will be removed
562 * if no further problems show up. */
563 info("SM_CANCEL for nonplaying track %s", sm
.id
);
564 sm
.type
= SM_STILLBORN
;
566 strcpy(sm
.id
, t
->id
);
569 /* Probably scratching the playing track well before it's got
570 * going, but could indicate a bug, so we log this as an error. */
571 sm
.type
= SM_UNKNOWN
;
572 error(0, "SM_CANCEL for unknown track %s", sm
.id
);
574 speaker_send(1, &sm
);
579 if(config_read(1)) error(0, "cannot read configuration");
580 info("reloaded configuration");
583 error(0, "unknown message type %d", sm
.type
);
586 /* Read in any buffered data */
587 for(t
= tracks
; t
; t
= t
->next
)
590 && (fds
[t
->slot
].revents
& (POLLIN
| POLLHUP
)))
592 /* Maybe we finished playing a track somewhere in the above */
594 /* If we don't need the sound device for now then close it for the benefit
595 * of anyone else who wants it. */
596 if((!playing
|| paused
) && device_state
== device_open
)
598 /* If we've not reported out state for a second do so now. */
599 if(time(0) > last_report
)
604 int main(int argc
, char **argv
) {
605 int n
, logsyslog
= !isatty(2);
606 struct sockaddr_un addr
;
607 static const int one
= 1;
608 struct speaker_message sm
;
613 if(!setlocale(LC_CTYPE
, "")) fatal(errno
, "error calling setlocale");
614 while((n
= getopt_long(argc
, argv
, "hVc:dDSs", options
, 0)) >= 0) {
617 case 'V': version("disorder-speaker");
618 case 'c': configfile
= optarg
; break;
619 case 'd': debugging
= 1; break;
620 case 'D': debugging
= 0; break;
621 case 'S': logsyslog
= 0; break;
622 case 's': logsyslog
= 1; break;
623 default: fatal(0, "invalid option");
626 if((d
= getenv("DISORDER_DEBUG_SPEAKER"))) debugging
= atoi(d
);
628 openlog(progname
, LOG_PID
, LOG_DAEMON
);
629 log_default
= &log_syslog
;
631 if(config_read(1)) fatal(0, "cannot read configuration");
632 bpf
= bytes_per_frame(&config
->sample_format
);
634 signal(SIGPIPE
, SIG_IGN
);
636 signal(SIGCHLD
, reap
);
638 xnice(config
->nice_speaker
);
641 /* make sure we're not root, whatever the config says */
642 if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root");
643 /* identify the backend used to play */
644 for(n
= 0; backends
[n
]; ++n
)
645 if(backends
[n
]->backend
== config
->api
)
648 fatal(0, "unsupported api %d", config
->api
);
649 backend
= backends
[n
];
650 /* backend-specific initialization */
652 /* create the socket directory */
653 byte_xasprintf(&dir
, "%s/speaker", config
->home
);
654 unlink(dir
); /* might be a leftover socket */
655 if(mkdir(dir
, 0700) < 0 && errno
!= EEXIST
)
656 fatal(errno
, "error creating %s", dir
);
657 /* set up the listen socket */
658 listenfd
= xsocket(PF_UNIX
, SOCK_STREAM
, 0);
659 memset(&addr
, 0, sizeof addr
);
660 addr
.sun_family
= AF_UNIX
;
661 snprintf(addr
.sun_path
, sizeof addr
.sun_path
, "%s/speaker/socket",
663 if(unlink(addr
.sun_path
) < 0 && errno
!= ENOENT
)
664 error(errno
, "removing %s", addr
.sun_path
);
665 xsetsockopt(listenfd
, SOL_SOCKET
, SO_REUSEADDR
, &one
, sizeof one
);
666 if(bind(listenfd
, (const struct sockaddr
*)&addr
, sizeof addr
) < 0)
667 fatal(errno
, "error binding socket to %s", addr
.sun_path
);
668 xlisten(listenfd
, 128);
670 info("listening on %s", addr
.sun_path
);
671 memset(&sm
, 0, sizeof sm
);
673 speaker_send(1, &sm
);
675 info("stopped (parent terminated)");