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>
73 #include "configuration.h"
78 #include "speaker-protocol.h"
83 /** @brief Linked list of all prepared tracks */
86 /** @brief Playing track, or NULL */
87 struct track
*playing
;
89 /** @brief Number of bytes pre frame */
92 /** @brief Array of file descriptors for poll() */
93 struct pollfd fds
[NFDS
];
95 /** @brief Next free slot in @ref fds */
98 /** @brief Listen socket */
101 static time_t last_report
; /* when we last reported */
102 static int paused
; /* pause status */
104 /** @brief The current device state */
105 enum device_states device_state
;
107 /** @brief Set when idled
109 * This is set when the sound device is deliberately closed by idle().
113 /** @brief Selected backend */
114 static const struct speaker_backend
*backend
;
116 static const struct option options
[] = {
117 { "help", no_argument
, 0, 'h' },
118 { "version", no_argument
, 0, 'V' },
119 { "config", required_argument
, 0, 'c' },
120 { "debug", no_argument
, 0, 'd' },
121 { "no-debug", no_argument
, 0, 'D' },
122 { "syslog", no_argument
, 0, 's' },
123 { "no-syslog", no_argument
, 0, 'S' },
127 /* Display usage message and terminate. */
128 static void help(void) {
130 " disorder-speaker [OPTIONS]\n"
132 " --help, -h Display usage message\n"
133 " --version, -V Display version number\n"
134 " --config PATH, -c PATH Set configuration file\n"
135 " --debug, -d Turn on debugging\n"
136 " --[no-]syslog Force logging\n"
138 "Speaker process for DisOrder. Not intended to be run\n"
144 /* Display version number and terminate. */
145 static void version(void) {
146 xprintf("%s", disorder_version_string
);
151 /** @brief Return the number of bytes per frame in @p format */
152 static size_t bytes_per_frame(const struct stream_header
*format
) {
153 return format
->channels
* format
->bits
/ 8;
156 /** @brief Find track @p id, maybe creating it if not found */
157 static struct track
*findtrack(const char *id
, int create
) {
160 D(("findtrack %s %d", id
, create
));
161 for(t
= tracks
; t
&& strcmp(id
, t
->id
); t
= t
->next
)
164 t
= xmalloc(sizeof *t
);
173 /** @brief Remove track @p id (but do not destroy it) */
174 static struct track
*removetrack(const char *id
) {
175 struct track
*t
, **tt
;
177 D(("removetrack %s", id
));
178 for(tt
= &tracks
; (t
= *tt
) && strcmp(id
, t
->id
); tt
= &t
->next
)
185 /** @brief Destroy a track */
186 static void destroy(struct track
*t
) {
187 D(("destroy %s", t
->id
));
188 if(t
->fd
!= -1) xclose(t
->fd
);
192 /** @brief Read data into a sample buffer
193 * @param t Pointer to track
194 * @return 0 on success, -1 on EOF
196 * This is effectively the read callback on @c t->fd. It is called from the
197 * main loop whenever the track's file descriptor is readable, assuming the
198 * buffer has not reached the maximum allowed occupancy.
200 static int speaker_fill(struct track
*t
) {
204 D(("fill %s: eof=%d used=%zu",
205 t
->id
, t
->eof
, t
->used
));
206 if(t
->eof
) return -1;
207 if(t
->used
< sizeof t
->buffer
) {
208 /* there is room left in the buffer */
209 where
= (t
->start
+ t
->used
) % sizeof t
->buffer
;
210 /* Get as much data as we can */
211 if(where
>= t
->start
) left
= (sizeof t
->buffer
) - where
;
212 else left
= t
->start
- where
;
214 n
= read(t
->fd
, t
->buffer
+ where
, left
);
215 } while(n
< 0 && errno
== EINTR
);
217 if(errno
!= EAGAIN
) fatal(errno
, "error reading sample stream");
221 D(("fill %s: eof detected", t
->id
));
227 if(t
->used
== sizeof t
->buffer
)
233 /** @brief Close the sound device
235 * This is called to deactivate the output device when pausing, and also by the
236 * ALSA backend when changing encoding (in which case the sound device will be
237 * immediately reactivated).
239 static void idle(void) {
241 if(backend
->deactivate
)
242 backend
->deactivate();
244 device_state
= device_closed
;
248 /** @brief Abandon the current track */
250 struct speaker_message sm
;
253 memset(&sm
, 0, sizeof sm
);
254 sm
.type
= SM_FINISHED
;
255 strcpy(sm
.id
, playing
->id
);
256 speaker_send(1, &sm
);
257 removetrack(playing
->id
);
262 /** @brief Enable sound output
264 * Makes sure the sound device is open and has the right sample format. Return
265 * 0 on success and -1 on error.
267 static void activate(void) {
268 if(backend
->activate
)
271 device_state
= device_open
;
274 /** @brief Check whether the current track has finished
276 * The current track is determined to have finished either if the input stream
277 * eded before the format could be determined (i.e. it is malformed) or the
278 * input is at end of file and there is less than a frame left unplayed. (So
279 * it copes with decoders that crash mid-frame.)
281 static void maybe_finished(void) {
284 && playing
->used
< bytes_per_frame(&config
->sample_format
))
288 /** @brief Return nonzero if we want to play some audio
290 * We want to play audio if there is a current track; and it is not paused; and
291 * it is playable according to the rules for @ref track::playable.
293 static int playable(void) {
296 && playing
->playable
;
299 /** @brief Play up to @p frames frames of audio
301 * It is always safe to call this function.
302 * - If @ref playing is 0 then it will just return
303 * - If @ref paused is non-0 then it will just return
304 * - If @ref device_state != @ref device_open then it will call activate() and
305 * return if it it fails.
306 * - If there is not enough audio to play then it play what is available.
308 * If there are not enough frames to play then whatever is available is played
309 * instead. It is up to mainloop() to ensure that speaker_play() is not called
310 * when unreasonably only an small amounts of data is available to play.
312 static void speaker_play(size_t frames
) {
313 size_t avail_frames
, avail_bytes
, written_frames
;
314 ssize_t written_bytes
;
316 /* Make sure there's a track to play and it is not paused */
319 /* Make sure the output device is open */
320 if(device_state
!= device_open
) {
322 if(device_state
!= device_open
)
325 D(("play: play %zu/%zu%s %dHz %db %dc", frames
, playing
->used
/ bpf
,
326 playing
->eof ?
" EOF" : "",
327 config
->sample_format
.rate
,
328 config
->sample_format
.bits
,
329 config
->sample_format
.channels
));
330 /* Figure out how many frames there are available to write */
331 if(playing
->start
+ playing
->used
> sizeof playing
->buffer
)
332 /* The ring buffer is currently wrapped, only play up to the wrap point */
333 avail_bytes
= (sizeof playing
->buffer
) - playing
->start
;
335 /* The ring buffer is not wrapped, can play the lot */
336 avail_bytes
= playing
->used
;
337 avail_frames
= avail_bytes
/ bpf
;
338 /* Only play up to the requested amount */
339 if(avail_frames
> frames
)
340 avail_frames
= frames
;
344 written_frames
= backend
->play(avail_frames
);
345 written_bytes
= written_frames
* bpf
;
346 /* written_bytes and written_frames had better both be set and correct by
348 playing
->start
+= written_bytes
;
349 playing
->used
-= written_bytes
;
350 playing
->played
+= written_frames
;
351 /* If the pointer is at the end of the buffer (or the buffer is completely
352 * empty) wrap it back to the start. */
353 if(!playing
->used
|| playing
->start
== (sizeof playing
->buffer
))
355 /* If the buffer emptied out mark the track as unplayably */
356 if(!playing
->used
&& !playing
->eof
) {
357 error(0, "track buffer emptied");
358 playing
->playable
= 0;
360 frames
-= written_frames
;
364 /* Notify the server what we're up to. */
365 static void report(void) {
366 struct speaker_message sm
;
369 memset(&sm
, 0, sizeof sm
);
370 sm
.type
= paused ? SM_PAUSED
: SM_PLAYING
;
371 strcpy(sm
.id
, playing
->id
);
372 sm
.data
= playing
->played
/ config
->sample_format
.rate
;
373 speaker_send(1, &sm
);
378 static void reap(int __attribute__((unused
)) sig
) {
383 cmdpid
= waitpid(-1, &st
, WNOHANG
);
385 signal(SIGCHLD
, reap
);
388 int addfd(int fd
, int events
) {
391 fds
[fdno
].events
= events
;
397 /** @brief Table of speaker backends */
398 static const struct speaker_backend
*backends
[] = {
399 #if HAVE_ALSA_ASOUNDLIB_H
404 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
407 #if HAVE_SYS_SOUNDCARD_H
413 /** @brief Main event loop */
414 static void mainloop(void) {
416 struct speaker_message sm
;
417 int n
, fd
, stdin_slot
, timeout
, listen_slot
;
419 while(getppid() != 1) {
421 /* By default we will wait up to a second before thinking about current
424 /* Always ready for commands from the main server. */
425 stdin_slot
= addfd(0, POLLIN
);
426 /* Also always ready for inbound connections */
427 listen_slot
= addfd(listenfd
, POLLIN
);
428 /* Try to read sample data for the currently playing track if there is
433 && playing
->used
< (sizeof playing
->buffer
))
434 playing
->slot
= addfd(playing
->fd
, POLLIN
);
438 /* We want to play some audio. If the device is closed then we attempt
440 if(device_state
== device_closed
)
442 /* If the device is (now) open then we will wait up until it is ready for
443 * more. If something went wrong then we should have device_error
444 * instead, but the post-poll code will cope even if it's
446 if(device_state
== device_open
)
447 backend
->beforepoll(&timeout
);
449 /* If any other tracks don't have a full buffer, try to read sample data
450 * from them. We do this last of all, so that if we run out of slots,
451 * nothing important can't be monitored. */
452 for(t
= tracks
; t
; t
= t
->next
)
456 && t
->used
< sizeof t
->buffer
) {
457 t
->slot
= addfd(t
->fd
, POLLIN
| POLLHUP
);
461 /* Wait for something interesting to happen */
462 n
= poll(fds
, fdno
, timeout
);
464 if(errno
== EINTR
) continue;
465 fatal(errno
, "error calling poll");
467 /* Play some sound before doing anything else */
469 /* We want to play some audio */
470 if(device_state
== device_open
) {
472 speaker_play(3 * FRAMES
);
474 /* We must be in _closed or _error, and it should be the latter, but we
477 * We most likely timed out, so now is a good time to retry.
478 * speaker_play() knows to re-activate the device if necessary.
480 speaker_play(3 * FRAMES
);
483 /* Perhaps a connection has arrived */
484 if(fds
[listen_slot
].revents
& POLLIN
) {
485 struct sockaddr_un addr
;
486 socklen_t addrlen
= sizeof addr
;
490 if((fd
= accept(listenfd
, (struct sockaddr
*)&addr
, &addrlen
)) >= 0) {
492 if(read(fd
, &l
, sizeof l
) < 4) {
493 error(errno
, "reading length from inbound connection");
495 } else if(l
>= sizeof id
) {
496 error(0, "id length too long");
498 } else if(read(fd
, id
, l
) < (ssize_t
)l
) {
499 error(errno
, "reading id from inbound connection");
503 D(("id %s fd %d", id
, fd
));
504 t
= findtrack(id
, 1/*create*/);
505 write(fd
, "", 1); /* write an ack */
507 error(0, "%s: already got a connection", id
);
511 t
->fd
= fd
; /* yay */
515 error(errno
, "accept");
517 /* Perhaps we have a command to process */
518 if(fds
[stdin_slot
].revents
& POLLIN
) {
519 /* There might (in theory) be several commands queued up, but in general
520 * this won't be the case, so we don't bother looping around to pick them
522 n
= speaker_recv(0, &sm
);
527 if(playing
) fatal(0, "got SM_PLAY but already playing something");
528 t
= findtrack(sm
.id
, 1);
529 D(("SM_PLAY %s fd %d", t
->id
, t
->fd
));
531 error(0, "cannot play track because no connection arrived");
533 /* We attempt to play straight away rather than going round the loop.
534 * speaker_play() is clever enough to perform any activation that is
536 speaker_play(3 * FRAMES
);
548 /* As for SM_PLAY we attempt to play straight away. */
550 speaker_play(3 * FRAMES
);
555 D(("SM_CANCEL %s", sm
.id
));
556 t
= removetrack(sm
.id
);
559 sm
.type
= SM_FINISHED
;
560 strcpy(sm
.id
, playing
->id
);
561 speaker_send(1, &sm
);
566 error(0, "SM_CANCEL for unknown track %s", sm
.id
);
571 if(config_read(1)) error(0, "cannot read configuration");
572 info("reloaded configuration");
575 error(0, "unknown message type %d", sm
.type
);
578 /* Read in any buffered data */
579 for(t
= tracks
; t
; t
= t
->next
)
582 && (fds
[t
->slot
].revents
& (POLLIN
| POLLHUP
)))
584 /* Maybe we finished playing a track somewhere in the above */
586 /* If we don't need the sound device for now then close it for the benefit
587 * of anyone else who wants it. */
588 if((!playing
|| paused
) && device_state
== device_open
)
590 /* If we've not reported out state for a second do so now. */
591 if(time(0) > last_report
)
596 int main(int argc
, char **argv
) {
597 int n
, logsyslog
= !isatty(2);
598 struct sockaddr_un addr
;
599 static const int one
= 1;
600 struct speaker_message sm
;
605 if(!setlocale(LC_CTYPE
, "")) fatal(errno
, "error calling setlocale");
606 while((n
= getopt_long(argc
, argv
, "hVc:dDSs", options
, 0)) >= 0) {
610 case 'c': configfile
= optarg
; break;
611 case 'd': debugging
= 1; break;
612 case 'D': debugging
= 0; break;
613 case 'S': logsyslog
= 0; break;
614 case 's': logsyslog
= 1; break;
615 default: fatal(0, "invalid option");
618 if((d
= getenv("DISORDER_DEBUG_SPEAKER"))) debugging
= atoi(d
);
620 openlog(progname
, LOG_PID
, LOG_DAEMON
);
621 log_default
= &log_syslog
;
623 if(config_read(1)) fatal(0, "cannot read configuration");
624 bpf
= bytes_per_frame(&config
->sample_format
);
626 signal(SIGPIPE
, SIG_IGN
);
628 signal(SIGCHLD
, reap
);
630 xnice(config
->nice_speaker
);
633 /* make sure we're not root, whatever the config says */
634 if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root");
635 /* identify the backend used to play */
636 for(n
= 0; backends
[n
]; ++n
)
637 if(backends
[n
]->backend
== config
->speaker_backend
)
640 fatal(0, "unsupported backend %d", config
->speaker_backend
);
641 backend
= backends
[n
];
642 /* backend-specific initialization */
644 /* create the socket directory */
645 byte_xasprintf(&dir
, "%s/speaker", config
->home
);
646 unlink(dir
); /* might be a leftover socket */
647 if(mkdir(dir
, 0700) < 0)
648 fatal(errno
, "error creating %s", dir
);
649 /* set up the listen socket */
650 listenfd
= xsocket(PF_UNIX
, SOCK_STREAM
, 0);
651 memset(&addr
, 0, sizeof addr
);
652 addr
.sun_family
= AF_UNIX
;
653 snprintf(addr
.sun_path
, sizeof addr
.sun_path
, "%s/speaker/socket",
655 if(unlink(addr
.sun_path
) < 0 && errno
!= ENOENT
)
656 error(errno
, "removing %s", addr
.sun_path
);
657 xsetsockopt(listenfd
, SOL_SOCKET
, SO_REUSEADDR
, &one
, sizeof one
);
658 if(bind(listenfd
, (const struct sockaddr
*)&addr
, sizeof addr
) < 0)
659 fatal(errno
, "error binding socket to %s", addr
.sun_path
);
660 xlisten(listenfd
, 128);
662 info("listening on %s", addr
.sun_path
);
663 memset(&sm
, 0, sizeof sm
);
665 speaker_send(1, &sm
);
667 info("stopped (parent terminated)");