2 * This file is part of DisOrder
3 * Copyright (C) 2005, 2006 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
21 /* This program deliberately does not use the garbage collector even though it
22 * might be convenient to do so. This is for two reasons. Firstly some libao
23 * drivers are implemented using threads and we do not want to have to deal
24 * with potential interactions between threading and garbage collection.
25 * Secondly this process needs to be able to respond quickly and this is not
26 * compatible with the collector hanging the program even relatively
42 #include <sys/select.h>
44 #include <alsa/asoundlib.h>
46 #include "configuration.h"
54 #define BUFFER_SECONDS 5 /* How many seconds of input to
57 #define FRAMES 4096 /* Frame batch size */
59 #define NFDS 256 /* Max FDs to poll for */
61 /* Known tracks are kept in a linked list. We don't normally to have
62 * more than two - maybe three at the outside. */
64 struct track
*next
; /* next track */
65 int fd
; /* input FD */
67 size_t start
, used
; /* start + bytes used */
68 int eof
; /* input is at EOF */
69 int got_format
; /* got format yet? */
70 ao_sample_format format
; /* sample format */
71 unsigned long long played
; /* number of frames played */
72 char *buffer
; /* sample buffer */
73 size_t size
; /* sample buffer size */
74 int slot
; /* poll array slot */
75 } *tracks
, *playing
; /* all tracks + playing track */
77 static time_t last_report
; /* when we last reported */
78 static int paused
; /* pause status */
79 static snd_pcm_t
*pcm
; /* current pcm handle */
80 static ao_sample_format pcm_format
; /* current format if aodev != 0 */
81 static size_t bpf
; /* bytes per frame */
82 static struct pollfd fds
[NFDS
]; /* if we need more than that */
83 static int fdno
; /* fd number */
84 static snd_pcm_uframes_t pcm_bufsize
; /* buffer size */
85 static int forceplay
; /* frames to force play */
87 static const struct option options
[] = {
88 { "help", no_argument
, 0, 'h' },
89 { "version", no_argument
, 0, 'V' },
90 { "config", required_argument
, 0, 'c' },
91 { "debug", no_argument
, 0, 'd' },
92 { "no-debug", no_argument
, 0, 'D' },
96 /* Display usage message and terminate. */
97 static void help(void) {
99 " disorder-speaker [OPTIONS]\n"
101 " --help, -h Display usage message\n"
102 " --version, -V Display version number\n"
103 " --config PATH, -c PATH Set configuration file\n"
104 " --debug, -d Turn on debugging\n"
106 "Speaker process for DisOrder. Not intended to be run\n"
112 /* Display version number and terminate. */
113 static void version(void) {
114 xprintf("disorder-speaker version %s\n", disorder_version_string
);
119 /* Return the number of bytes per frame in FORMAT. */
120 static size_t bytes_per_frame(const ao_sample_format
*format
) {
121 return format
->channels
* format
->bits
/ 8;
124 /* Find track ID, maybe creating it if not found. */
125 static struct track
*findtrack(const char *id
, int create
) {
128 D(("findtrack %s %d", id
, create
));
129 for(t
= tracks
; t
&& strcmp(id
, t
->id
); t
= t
->next
)
132 t
= xmalloc(sizeof *t
);
137 /* The initial input buffer will be the sample format. */
138 t
->buffer
= (void *)&t
->format
;
139 t
->size
= sizeof t
->format
;
144 /* Remove track ID (but do not destroy it). */
145 static struct track
*removetrack(const char *id
) {
146 struct track
*t
, **tt
;
148 D(("removetrack %s", id
));
149 for(tt
= &tracks
; (t
= *tt
) && strcmp(id
, t
->id
); tt
= &t
->next
)
156 /* Destroy a track. */
157 static void destroy(struct track
*t
) {
158 D(("destroy %s", t
->id
));
159 if(t
->fd
!= -1) xclose(t
->fd
);
160 if(t
->buffer
!= (void *)&t
->format
) free(t
->buffer
);
164 /* Notice a new FD. */
165 static void acquire(struct track
*t
, int fd
) {
166 D(("acquire %s %d", t
->id
, fd
));
173 /* Read data into a sample buffer. Return 0 on success, -1 on EOF. */
174 static int fill(struct track
*t
) {
178 D(("fill %s: eof=%d used=%zu size=%zu got_format=%d",
179 t
->id
, t
->eof
, t
->used
, t
->size
, t
->got_format
));
180 if(t
->eof
) return -1;
181 if(t
->used
< t
->size
) {
182 /* there is room left in the buffer */
183 where
= (t
->start
+ t
->used
) % t
->size
;
185 /* We are reading audio data, get as much as we can */
186 if(where
>= t
->start
) left
= t
->size
- where
;
187 else left
= t
->start
- where
;
189 /* We are still waiting for the format, only get that */
190 left
= sizeof (ao_sample_format
) - t
->used
;
192 n
= read(t
->fd
, t
->buffer
+ where
, left
);
193 } while(n
< 0 && errno
== EINTR
);
195 if(errno
!= EAGAIN
) fatal(errno
, "error reading sample stream");
199 D(("fill %s: eof detected", t
->id
));
204 if(!t
->got_format
&& t
->used
>= sizeof (ao_sample_format
)) {
205 assert(t
->used
== sizeof (ao_sample_format
));
206 /* Check that our assumptions are met. */
207 if(t
->format
.bits
& 7)
208 fatal(0, "bits per sample not a multiple of 8");
209 /* Make a new buffer for audio data. */
210 t
->size
= bytes_per_frame(&t
->format
) * t
->format
.rate
* BUFFER_SECONDS
;
211 t
->buffer
= xmalloc(t
->size
);
214 D(("got format for %s", t
->id
));
220 /* Return true if A and B denote identical libao formats, else false. */
221 static int formats_equal(const ao_sample_format
*a
,
222 const ao_sample_format
*b
) {
223 return (a
->bits
== b
->bits
224 && a
->rate
== b
->rate
225 && a
->channels
== b
->channels
226 && a
->byte_format
== b
->byte_format
);
229 /* Close the sound device. */
230 static void idle(void) {
235 if((err
= snd_pcm_nonblock(pcm
, 0)) < 0)
236 fatal(0, "error calling snd_pcm_nonblock: %d", err
);
243 D(("released audio device"));
247 /* Abandon the current track */
248 static void abandon(void) {
249 struct speaker_message sm
;
252 memset(&sm
, 0, sizeof sm
);
253 sm
.type
= SM_FINISHED
;
254 strcpy(sm
.id
, playing
->id
);
255 speaker_send(1, &sm
, 0);
256 removetrack(playing
->id
);
262 /* Make sure the sound device is open and has the right sample format. Return
263 * 0 on success and -1 on error. */
264 static int activate(void) {
266 snd_pcm_hw_params_t
*hwparams
;
267 snd_pcm_sw_params_t
*swparams
;
268 int sample_format
= 0;
271 /* If we don't know the format yet we cannot start. */
272 if(!playing
->got_format
) {
273 D((" - not got format for %s", playing
->id
));
276 /* If we need to change format then close the current device. */
277 if(pcm
&& !formats_equal(&playing
->format
, &pcm_format
))
281 if((err
= snd_pcm_open(&pcm
,
283 SND_PCM_STREAM_PLAYBACK
,
284 SND_PCM_NONBLOCK
))) {
285 error(0, "error from snd_pcm_open: %d", err
);
288 snd_pcm_hw_params_alloca(&hwparams
);
289 D(("set up hw params"));
290 if((err
= snd_pcm_hw_params_any(pcm
, hwparams
)) < 0)
291 fatal(0, "error from snd_pcm_hw_params_any: %d", err
);
292 if((err
= snd_pcm_hw_params_set_access(pcm
, hwparams
,
293 SND_PCM_ACCESS_RW_INTERLEAVED
)) < 0)
294 fatal(0, "error from snd_pcm_hw_params_set_access: %d", err
);
295 switch(playing
->format
.bits
) {
297 sample_format
= SND_PCM_FORMAT_S8
;
300 switch(playing
->format
.byte_format
) {
301 case AO_FMT_NATIVE
: sample_format
= SND_PCM_FORMAT_S16
; break;
302 case AO_FMT_LITTLE
: sample_format
= SND_PCM_FORMAT_S16_LE
; break;
303 case AO_FMT_BIG
: sample_format
= SND_PCM_FORMAT_S16_BE
; break;
304 error(0, "unrecognized byte format %d", playing
->format
.byte_format
);
309 error(0, "unsupported sample size %d", playing
->format
.bits
);
312 if((err
= snd_pcm_hw_params_set_format(pcm
, hwparams
,
313 sample_format
)) < 0) {
314 error(0, "error from snd_pcm_hw_params_set_format (%d): %d",
318 rate
= playing
->format
.rate
;
319 if((err
= snd_pcm_hw_params_set_rate_near(pcm
, hwparams
, &rate
, 0)) < 0) {
320 error(0, "error from snd_pcm_hw_params_set_rate (%d): %d",
321 playing
->format
.rate
, err
);
324 if(rate
!= (unsigned)playing
->format
.rate
)
325 info("want rate %d, got %u", playing
->format
.rate
, rate
);
326 if((err
= snd_pcm_hw_params_set_channels(pcm
, hwparams
,
327 playing
->format
.channels
)) < 0) {
328 error(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
329 playing
->format
.channels
, err
);
332 pcm_bufsize
= 3 * FRAMES
;
333 if((err
= snd_pcm_hw_params_set_buffer_size_near(pcm
, hwparams
,
335 fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d",
337 if(pcm_bufsize
!= 3 * FRAMES
)
338 info("asked for PCM buffer of %d frames, got %d",
339 3 * FRAMES
, (int)pcm_bufsize
);
340 if((err
= snd_pcm_hw_params(pcm
, hwparams
)) < 0)
341 fatal(0, "error calling snd_pcm_hw_params: %d", err
);
342 D(("set up sw params"));
343 snd_pcm_sw_params_alloca(&swparams
);
344 if((err
= snd_pcm_sw_params_current(pcm
, swparams
)) < 0)
345 fatal(0, "error calling snd_pcm_sw_params_current: %d", err
);
346 if((err
= snd_pcm_sw_params_set_avail_min(pcm
, swparams
, FRAMES
)) < 0)
347 fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d",
349 if((err
= snd_pcm_sw_params(pcm
, swparams
)) < 0)
350 fatal(0, "error calling snd_pcm_sw_params: %d", err
);
351 pcm_format
= playing
->format
;
352 bpf
= bytes_per_frame(&pcm_format
);
353 D(("acquired audio device"));
359 /* We assume the error is temporary and that we'll retry in a bit. */
367 /* Check to see whether the current track has finished playing */
368 static void maybe_finished(void) {
371 && (!playing
->got_format
372 || playing
->used
< bytes_per_frame(&playing
->format
)))
376 static void play(size_t frames
) {
377 snd_pcm_sframes_t written_frames
;
378 size_t avail_bytes
, avail_frames
, written_bytes
;
385 forceplay
= 0; /* Must have called abandon() */
388 D(("play: play %zu/%zu%s %dHz %db %dc", frames
, playing
->used
/ bpf
,
389 playing
->eof ?
" EOF" : "",
390 playing
->format
.rate
,
391 playing
->format
.bits
,
392 playing
->format
.channels
));
393 /* If we haven't got enough bytes yet wait until we have. Exception: when
395 if(playing
->used
< frames
* bpf
&& !playing
->eof
) {
399 /* We have got enough data so don't force play again */
401 /* Figure out how many frames there are available to write */
402 if(playing
->start
+ playing
->used
> playing
->size
)
403 avail_bytes
= playing
->size
- playing
->start
;
405 avail_bytes
= playing
->used
;
406 avail_frames
= avail_bytes
/ bpf
;
407 if(avail_frames
> frames
)
408 avail_frames
= frames
;
411 written_frames
= snd_pcm_writei(pcm
,
412 playing
->buffer
+ playing
->start
,
414 D(("actually play %zu frames, wrote %d",
415 avail_frames
, (int)written_frames
));
416 if(written_frames
< 0) {
417 switch(written_frames
) {
418 case -EPIPE
: /* underrun */
419 error(0, "snd_pcm_writei reports underrun");
420 if((err
= snd_pcm_prepare(pcm
)) < 0)
421 fatal(0, "error calling snd_pcm_prepare: %d", err
);
426 fatal(0, "error calling snd_pcm_writei: %d", (int)written_frames
);
429 written_bytes
= written_frames
* bpf
;
430 playing
->start
+= written_bytes
;
431 playing
->used
-= written_bytes
;
432 playing
->played
+= written_frames
;
433 /* If the pointer is at the end of the buffer (or the buffer is completely
434 * empty) wrap it back to the start. */
435 if(!playing
->used
|| playing
->start
== playing
->size
)
437 frames
-= written_frames
;
440 /* Notify the server what we're up to. */
441 static void report(void) {
442 struct speaker_message sm
;
444 if(playing
&& playing
->buffer
!= (void *)&playing
->format
) {
445 memset(&sm
, 0, sizeof sm
);
446 sm
.type
= paused ? SM_PAUSED
: SM_PLAYING
;
447 strcpy(sm
.id
, playing
->id
);
448 sm
.data
= playing
->played
/ playing
->format
.rate
;
449 speaker_send(1, &sm
, 0);
454 static int addfd(int fd
, int events
) {
457 fds
[fdno
].events
= events
;
463 int main(int argc
, char **argv
) {
464 int n
, fd
, stdin_slot
, alsa_slots
, alsa_nslots
= -1, err
;
465 unsigned short alsa_revents
;
467 struct speaker_message sm
;
471 if(!setlocale(LC_CTYPE
, "")) fatal(errno
, "error calling setlocale");
472 while((n
= getopt_long(argc
, argv
, "hVc:dD", options
, 0)) >= 0) {
476 case 'c': configfile
= optarg
; break;
477 case 'd': debugging
= 1; break;
478 case 'D': debugging
= 0; break;
479 default: fatal(0, "invalid option");
482 if(getenv("DISORDER_DEBUG_SPEAKER")) debugging
= 1;
483 /* If stderr is a TTY then log there, otherwise to syslog. */
485 openlog(progname
, LOG_PID
, LOG_DAEMON
);
486 log_default
= &log_syslog
;
488 if(config_read()) fatal(0, "cannot read configuration");
490 signal(SIGPIPE
, SIG_IGN
);
492 xnice(config
->nice_speaker
);
495 /* make sure we're not root, whatever the config says */
496 if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root");
498 while(getppid() != 1) {
500 /* Always ready for commands from the main server. */
501 stdin_slot
= addfd(0, POLLIN
);
502 /* Try to read sample data for the currently playing track if there is
504 if(playing
&& !playing
->eof
&& playing
->used
< playing
->size
) {
505 playing
->slot
= addfd(playing
->fd
, POLLIN
);
508 /* If forceplay is set then wait until it succeeds before waiting on the
510 if(pcm
&& !forceplay
) {
512 alsa_nslots
= snd_pcm_poll_descriptors(pcm
, &fds
[fdno
], NFDS
- fdno
);
516 /* If any other tracks don't have a full buffer, try to read sample data
518 for(t
= tracks
; t
; t
= t
->next
)
520 if(!t
->eof
&& t
->used
< t
->size
) {
521 t
->slot
= addfd(t
->fd
, POLLIN
);
525 /* Wait up to a second before thinking about current state */
526 n
= poll(fds
, fdno
, 1000);
528 if(errno
== EINTR
) continue;
529 fatal(errno
, "error calling poll");
531 /* Play some sound before doing anything else */
532 if(alsa_slots
!= -1) {
533 if((err
= snd_pcm_poll_descriptors_revents(pcm
,
537 fatal(0, "error calling snd_pcm_poll_descriptors_revents: %d", err
);
538 if(alsa_revents
& POLLOUT
)
541 /* Some attempt to play must have failed */
542 if(playing
&& !paused
)
545 forceplay
= 0; /* just in case */
547 /* Perhaps we have a command to process */
548 if(fds
[stdin_slot
].revents
& POLLIN
) {
549 n
= speaker_recv(0, &sm
, &fd
);
553 D(("SM_PREPARE %s %d", sm
.id
, fd
));
554 if(fd
== -1) fatal(0, "got SM_PREPARE but no file descriptor");
555 t
= findtrack(sm
.id
, 1);
559 D(("SM_PLAY %s %d", sm
.id
, fd
));
560 if(playing
) fatal(0, "got SM_PLAY but already playing something");
561 t
= findtrack(sm
.id
, 1);
562 if(fd
!= -1) acquire(t
, fd
);
582 D(("SM_CANCEL %s", sm
.id
));
583 t
= removetrack(sm
.id
);
586 sm
.type
= SM_FINISHED
;
587 strcpy(sm
.id
, playing
->id
);
588 speaker_send(1, &sm
, 0);
593 error(0, "SM_CANCEL for unknown track %s", sm
.id
);
598 if(config_read()) error(0, "cannot read configuration");
599 info("reloaded configuration");
602 error(0, "unknown message type %d", sm
.type
);
605 /* Read in any buffered data */
606 for(t
= tracks
; t
; t
= t
->next
)
607 if(t
->slot
!= -1 && (fds
[t
->slot
].revents
& POLLIN
))
609 /* We might be able to play now */
610 if(pcm
&& forceplay
&& playing
&& !paused
)
612 /* Maybe we finished playing a track somewhere in the above */
614 /* If we don't need the sound device for now then close it for the benefit
615 * of anyone else who wants it. */
616 if((!playing
|| paused
) && pcm
)
618 /* If we've not reported out state for a second do so now. */
619 if(time(0) > last_report
)
622 info("stopped (parent terminated)");
634 /* arch-tag:HQ4ayCGCjeBF97RuRnvcyg */