2 * This file is part of DisOrder
3 * Copyright (C) 2005, 2006, 2007 Richard Kettlewell
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation; either version 2 of the License, or
8 * (at your option) any later version.
10 * This program is distributed in the hope that it will be useful, but
11 * WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * General Public License for more details.
15 * You should have received a copy of the GNU General Public License
16 * along with this program; if not, write to the Free Software
17 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
20 /** @file server/speaker.c
21 * @brief Speaker process
23 * This program is responsible for transmitting a single coherent audio stream
24 * to its destination (over the network, to some sound API, to some
25 * subprocess). It receives connections from decoders (or rather from the
26 * process that is about to become disorder-normalize) and plays them in the
29 * @b Encodings. For the <a href="http://www.alsa-project.org/">ALSA</a> API,
30 * 8- and 16- bit stereo and mono are supported, with any sample rate (within
31 * the limits that ALSA can deal with.)
33 * Inbound data is expected to match @c config->sample_format. In normal use
34 * this is arranged by the @c disorder-normalize program (see @ref
35 * server/normalize.c).
37 * @b Garbage @b Collection. This program deliberately does not use the
38 * garbage collector even though it might be convenient to do so. This is for
39 * two reasons. Firstly some sound APIs use thread threads and we do not want
40 * to have to deal with potential interactions between threading and garbage
41 * collection. Secondly this process needs to be able to respond quickly and
42 * this is not compatible with the collector hanging the program even
45 * @b Units. This program thinks at various times in three different units.
46 * Bytes are obvious. A sample is a single sample on a single channel. A
47 * frame is several samples on different channels at the same point in time.
48 * So (for instance) a 16-bit stereo frame is 4 bytes and consists of a pair of
65 #include <sys/select.h>
72 #include "configuration.h"
77 #include "speaker-protocol.h"
81 /** @brief Linked list of all prepared tracks */
84 /** @brief Playing track, or NULL */
85 struct track
*playing
;
87 /** @brief Number of bytes pre frame */
90 /** @brief Array of file descriptors for poll() */
91 struct pollfd fds
[NFDS
];
93 /** @brief Next free slot in @ref fds */
96 /** @brief Listen socket */
99 static time_t last_report
; /* when we last reported */
100 static int paused
; /* pause status */
102 /** @brief The current device state */
103 enum device_states device_state
;
105 /** @brief Set when idled
107 * This is set when the sound device is deliberately closed by idle().
111 /** @brief Selected backend */
112 static const struct speaker_backend
*backend
;
114 static const struct option options
[] = {
115 { "help", no_argument
, 0, 'h' },
116 { "version", no_argument
, 0, 'V' },
117 { "config", required_argument
, 0, 'c' },
118 { "debug", no_argument
, 0, 'd' },
119 { "no-debug", no_argument
, 0, 'D' },
123 /* Display usage message and terminate. */
124 static void help(void) {
126 " disorder-speaker [OPTIONS]\n"
128 " --help, -h Display usage message\n"
129 " --version, -V Display version number\n"
130 " --config PATH, -c PATH Set configuration file\n"
131 " --debug, -d Turn on debugging\n"
133 "Speaker process for DisOrder. Not intended to be run\n"
139 /* Display version number and terminate. */
140 static void version(void) {
141 xprintf("disorder-speaker version %s\n", disorder_version_string
);
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 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
));
225 /** @brief Close the sound device
227 * This is called to deactivate the output device when pausing, and also by the
228 * ALSA backend when changing encoding (in which case the sound device will be
229 * immediately reactivated).
231 static void idle(void) {
233 if(backend
->deactivate
)
234 backend
->deactivate();
236 device_state
= device_closed
;
240 /** @brief Abandon the current track */
242 struct speaker_message sm
;
245 memset(&sm
, 0, sizeof sm
);
246 sm
.type
= SM_FINISHED
;
247 strcpy(sm
.id
, playing
->id
);
248 speaker_send(1, &sm
);
249 removetrack(playing
->id
);
254 /** @brief Enable sound output
256 * Makes sure the sound device is open and has the right sample format. Return
257 * 0 on success and -1 on error.
259 static void activate(void) {
260 if(backend
->activate
)
263 device_state
= device_open
;
266 /** @brief Check whether the current track has finished
268 * The current track is determined to have finished either if the input stream
269 * eded before the format could be determined (i.e. it is malformed) or the
270 * input is at end of file and there is less than a frame left unplayed. (So
271 * it copes with decoders that crash mid-frame.)
273 static void maybe_finished(void) {
276 && playing
->used
< bytes_per_frame(&config
->sample_format
))
280 /** @brief Play up to @p frames frames of audio
282 * It is always safe to call this function.
283 * - If @ref playing is 0 then it will just return
284 * - If @ref paused is non-0 then it will just return
285 * - If @ref device_state != @ref device_open then it will call activate() and
286 * return if it it fails.
287 * - If there is not enough audio to play then it play what is available.
289 * If there are not enough frames to play then whatever is available is played
290 * instead. It is up to mainloop() to ensure that play() is not called when
291 * unreasonably only an small amounts of data is available to play.
293 static void play(size_t frames
) {
294 size_t avail_frames
, avail_bytes
, written_frames
;
295 ssize_t written_bytes
;
297 /* Make sure there's a track to play and it is not pasued */
298 if(!playing
|| paused
)
300 /* Make sure the output device is open */
301 if(device_state
!= device_open
) {
303 if(device_state
!= device_open
)
306 D(("play: play %zu/%zu%s %dHz %db %dc", frames
, playing
->used
/ bpf
,
307 playing
->eof ?
" EOF" : "",
308 config
->sample_format
.rate
,
309 config
->sample_format
.bits
,
310 config
->sample_format
.channels
));
311 /* Figure out how many frames there are available to write */
312 if(playing
->start
+ playing
->used
> sizeof playing
->buffer
)
313 /* The ring buffer is currently wrapped, only play up to the wrap point */
314 avail_bytes
= (sizeof playing
->buffer
) - playing
->start
;
316 /* The ring buffer is not wrapped, can play the lot */
317 avail_bytes
= playing
->used
;
318 avail_frames
= avail_bytes
/ bpf
;
319 /* Only play up to the requested amount */
320 if(avail_frames
> frames
)
321 avail_frames
= frames
;
325 written_frames
= backend
->play(avail_frames
);
326 written_bytes
= written_frames
* bpf
;
327 /* written_bytes and written_frames had better both be set and correct by
329 playing
->start
+= written_bytes
;
330 playing
->used
-= written_bytes
;
331 playing
->played
+= written_frames
;
332 /* If the pointer is at the end of the buffer (or the buffer is completely
333 * empty) wrap it back to the start. */
334 if(!playing
->used
|| playing
->start
== (sizeof playing
->buffer
))
336 frames
-= written_frames
;
340 /* Notify the server what we're up to. */
341 static void report(void) {
342 struct speaker_message sm
;
345 memset(&sm
, 0, sizeof sm
);
346 sm
.type
= paused ? SM_PAUSED
: SM_PLAYING
;
347 strcpy(sm
.id
, playing
->id
);
348 sm
.data
= playing
->played
/ config
->sample_format
.rate
;
349 speaker_send(1, &sm
);
354 static void reap(int __attribute__((unused
)) sig
) {
359 cmdpid
= waitpid(-1, &st
, WNOHANG
);
361 signal(SIGCHLD
, reap
);
364 int addfd(int fd
, int events
) {
367 fds
[fdno
].events
= events
;
373 /** @brief Table of speaker backends */
374 static const struct speaker_backend
*backends
[] = {
380 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
383 #if HAVE_SYS_SOUNDCARD_H
389 /** @brief Return nonzero if we want to play some audio
391 * We want to play audio if there is a current track; and it is not paused; and
392 * there are at least @ref FRAMES frames of audio to play, or we are in sight
393 * of the end of the current track.
395 static int playable(void) {
398 && (playing
->used
>= FRAMES
|| playing
->eof
);
401 /** @brief Main event loop */
402 static void mainloop(void) {
404 struct speaker_message sm
;
405 int n
, fd
, stdin_slot
, timeout
, listen_slot
;
407 while(getppid() != 1) {
409 /* By default we will wait up to a second before thinking about current
412 /* Always ready for commands from the main server. */
413 stdin_slot
= addfd(0, POLLIN
);
414 /* Also always ready for inbound connections */
415 listen_slot
= addfd(listenfd
, POLLIN
);
416 /* Try to read sample data for the currently playing track if there is
421 && playing
->used
< (sizeof playing
->buffer
))
422 playing
->slot
= addfd(playing
->fd
, POLLIN
);
426 /* We want to play some audio. If the device is closed then we attempt
428 if(device_state
== device_closed
)
430 /* If the device is (now) open then we will wait up until it is ready for
431 * more. If something went wrong then we should have device_error
432 * instead, but the post-poll code will cope even if it's
434 if(device_state
== device_open
)
435 backend
->beforepoll();
437 /* If any other tracks don't have a full buffer, try to read sample data
438 * from them. We do this last of all, so that if we run out of slots,
439 * nothing important can't be monitored. */
440 for(t
= tracks
; t
; t
= t
->next
)
444 && t
->used
< sizeof t
->buffer
) {
445 t
->slot
= addfd(t
->fd
, POLLIN
| POLLHUP
);
449 /* Wait for something interesting to happen */
450 n
= poll(fds
, fdno
, timeout
);
452 if(errno
== EINTR
) continue;
453 fatal(errno
, "error calling poll");
455 /* Play some sound before doing anything else */
457 /* We want to play some audio */
458 if(device_state
== device_open
) {
462 /* We must be in _closed or _error, and it should be the latter, but we
465 * We most likely timed out, so now is a good time to retry. play()
466 * knows to re-activate the device if necessary.
471 /* Perhaps a connection has arrived */
472 if(fds
[listen_slot
].revents
& POLLIN
) {
473 struct sockaddr_un addr
;
474 socklen_t addrlen
= sizeof addr
;
478 if((fd
= accept(listenfd
, (struct sockaddr
*)&addr
, &addrlen
)) >= 0) {
480 if(read(fd
, &l
, sizeof l
) < 4) {
481 error(errno
, "reading length from inbound connection");
483 } else if(l
>= sizeof id
) {
484 error(0, "id length too long");
486 } else if(read(fd
, id
, l
) < (ssize_t
)l
) {
487 error(errno
, "reading id from inbound connection");
491 D(("id %s fd %d", id
, fd
));
492 t
= findtrack(id
, 1/*create*/);
493 write(fd
, "", 1); /* write an ack */
495 error(0, "got a connection for a track that already has one");
499 t
->fd
= fd
; /* yay */
503 error(errno
, "accept");
505 /* Perhaps we have a command to process */
506 if(fds
[stdin_slot
].revents
& POLLIN
) {
507 /* There might (in theory) be several commands queued up, but in general
508 * this won't be the case, so we don't bother looping around to pick them
510 n
= speaker_recv(0, &sm
);
515 if(playing
) fatal(0, "got SM_PLAY but already playing something");
516 t
= findtrack(sm
.id
, 1);
517 D(("SM_PLAY %s fd %d", t
->id
, t
->fd
));
519 error(0, "cannot play track because no connection arrived");
521 /* We attempt to play straight away rather than going round the loop.
522 * play() is clever enough to perform any activation that is
536 /* As for SM_PLAY we attempt to play straight away. */
543 D(("SM_CANCEL %s", sm
.id
));
544 t
= removetrack(sm
.id
);
547 sm
.type
= SM_FINISHED
;
548 strcpy(sm
.id
, playing
->id
);
549 speaker_send(1, &sm
);
554 error(0, "SM_CANCEL for unknown track %s", sm
.id
);
559 if(config_read(1)) error(0, "cannot read configuration");
560 info("reloaded configuration");
563 error(0, "unknown message type %d", sm
.type
);
566 /* Read in any buffered data */
567 for(t
= tracks
; t
; t
= t
->next
)
570 && (fds
[t
->slot
].revents
& (POLLIN
| POLLHUP
)))
572 /* Maybe we finished playing a track somewhere in the above */
574 /* If we don't need the sound device for now then close it for the benefit
575 * of anyone else who wants it. */
576 if((!playing
|| paused
) && device_state
== device_open
)
578 /* If we've not reported out state for a second do so now. */
579 if(time(0) > last_report
)
584 int main(int argc
, char **argv
) {
586 struct sockaddr_un addr
;
587 static const int one
= 1;
588 struct speaker_message sm
;
591 if(!setlocale(LC_CTYPE
, "")) fatal(errno
, "error calling setlocale");
592 while((n
= getopt_long(argc
, argv
, "hVc:dD", options
, 0)) >= 0) {
596 case 'c': configfile
= optarg
; break;
597 case 'd': debugging
= 1; break;
598 case 'D': debugging
= 0; break;
599 default: fatal(0, "invalid option");
602 if(getenv("DISORDER_DEBUG_SPEAKER")) debugging
= 1;
603 /* If stderr is a TTY then log there, otherwise to syslog. */
605 openlog(progname
, LOG_PID
, LOG_DAEMON
);
606 log_default
= &log_syslog
;
608 if(config_read(1)) fatal(0, "cannot read configuration");
609 bpf
= bytes_per_frame(&config
->sample_format
);
611 signal(SIGPIPE
, SIG_IGN
);
613 signal(SIGCHLD
, reap
);
615 xnice(config
->nice_speaker
);
618 /* make sure we're not root, whatever the config says */
619 if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root");
620 /* identify the backend used to play */
621 for(n
= 0; backends
[n
]; ++n
)
622 if(backends
[n
]->backend
== config
->speaker_backend
)
625 fatal(0, "unsupported backend %d", config
->speaker_backend
);
626 backend
= backends
[n
];
627 /* backend-specific initialization */
629 /* set up the listen socket */
630 listenfd
= xsocket(PF_UNIX
, SOCK_STREAM
, 0);
631 memset(&addr
, 0, sizeof addr
);
632 addr
.sun_family
= AF_UNIX
;
633 snprintf(addr
.sun_path
, sizeof addr
.sun_path
, "%s/speaker",
635 if(unlink(addr
.sun_path
) < 0 && errno
!= ENOENT
)
636 error(errno
, "removing %s", addr
.sun_path
);
637 xsetsockopt(listenfd
, SOL_SOCKET
, SO_REUSEADDR
, &one
, sizeof one
);
638 if(bind(listenfd
, (const struct sockaddr
*)&addr
, sizeof addr
) < 0)
639 fatal(errno
, "error binding socket to %s", addr
.sun_path
);
640 xlisten(listenfd
, 128);
642 info("listening on %s", addr
.sun_path
);
643 memset(&sm
, 0, sizeof sm
);
645 speaker_send(1, &sm
);
647 info("stopped (parent terminated)");