Commit | Line | Data |
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460b9539 | 1 | /* |
2 | * This file is part of DisOrder | |
dea8f8aa | 3 | * Copyright (C) 2005, 2006, 2007 Richard Kettlewell |
460b9539 | 4 | * |
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. | |
9 | * | |
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. | |
14 | * | |
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 | |
18 | * USA | |
19 | */ | |
20 | ||
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 | |
27 | * briefly. */ | |
28 | ||
29 | #include <config.h> | |
30 | #include "types.h" | |
31 | ||
32 | #include <getopt.h> | |
33 | #include <stdio.h> | |
34 | #include <stdlib.h> | |
35 | #include <locale.h> | |
36 | #include <syslog.h> | |
37 | #include <unistd.h> | |
38 | #include <errno.h> | |
39 | #include <ao/ao.h> | |
40 | #include <string.h> | |
41 | #include <assert.h> | |
42 | #include <sys/select.h> | |
9d5da576 | 43 | #include <sys/wait.h> |
460b9539 | 44 | #include <time.h> |
8023f60b | 45 | #include <fcntl.h> |
46 | #include <poll.h> | |
e83d0967 RK |
47 | #include <sys/socket.h> |
48 | #include <netdb.h> | |
49 | #include <gcrypt.h> | |
50 | #include <sys/uio.h> | |
460b9539 | 51 | |
52 | #include "configuration.h" | |
53 | #include "syscalls.h" | |
54 | #include "log.h" | |
55 | #include "defs.h" | |
56 | #include "mem.h" | |
57 | #include "speaker.h" | |
58 | #include "user.h" | |
e83d0967 RK |
59 | #include "addr.h" |
60 | #include "timeval.h" | |
61 | #include "rtp.h" | |
460b9539 | 62 | |
8023f60b | 63 | #if API_ALSA |
dea8f8aa | 64 | #include <alsa/asoundlib.h> |
8023f60b | 65 | #endif |
dea8f8aa | 66 | |
5330d674 | 67 | #ifdef WORDS_BIGENDIAN |
68 | # define MACHINE_AO_FMT AO_FMT_BIG | |
69 | #else | |
70 | # define MACHINE_AO_FMT AO_FMT_LITTLE | |
71 | #endif | |
72 | ||
460b9539 | 73 | #define BUFFER_SECONDS 5 /* How many seconds of input to |
74 | * buffer. */ | |
75 | ||
76 | #define FRAMES 4096 /* Frame batch size */ | |
77 | ||
e83d0967 RK |
78 | #define NETWORK_BYTES 1024 /* Bytes to send per network packet */ |
79 | /* (don't make this too big or arithmetic will start to overflow) */ | |
80 | ||
81 | #define RTP_AHEAD 2 /* Max RTP playahead (seconds) */ | |
82 | ||
460b9539 | 83 | #define NFDS 256 /* Max FDs to poll for */ |
84 | ||
85 | /* Known tracks are kept in a linked list. We don't normally to have | |
86 | * more than two - maybe three at the outside. */ | |
87 | static struct track { | |
88 | struct track *next; /* next track */ | |
89 | int fd; /* input FD */ | |
90 | char id[24]; /* ID */ | |
91 | size_t start, used; /* start + bytes used */ | |
92 | int eof; /* input is at EOF */ | |
93 | int got_format; /* got format yet? */ | |
94 | ao_sample_format format; /* sample format */ | |
95 | unsigned long long played; /* number of frames played */ | |
96 | char *buffer; /* sample buffer */ | |
97 | size_t size; /* sample buffer size */ | |
98 | int slot; /* poll array slot */ | |
99 | } *tracks, *playing; /* all tracks + playing track */ | |
100 | ||
101 | static time_t last_report; /* when we last reported */ | |
102 | static int paused; /* pause status */ | |
460b9539 | 103 | static ao_sample_format pcm_format; /* current format if aodev != 0 */ |
104 | static size_t bpf; /* bytes per frame */ | |
105 | static struct pollfd fds[NFDS]; /* if we need more than that */ | |
106 | static int fdno; /* fd number */ | |
8023f60b | 107 | static size_t bufsize; /* buffer size */ |
108 | #if API_ALSA | |
109 | static snd_pcm_t *pcm; /* current pcm handle */ | |
0c207c37 | 110 | static snd_pcm_uframes_t last_pcm_bufsize; /* last seen buffer size */ |
8023f60b | 111 | #endif |
9d5da576 | 112 | static int ready; /* ready to send audio */ |
460b9539 | 113 | static int forceplay; /* frames to force play */ |
e83d0967 RK |
114 | static int cmdfd = -1; /* child process input */ |
115 | static int bfd = -1; /* broadcast FD */ | |
116 | static uint32_t rtp_time; /* RTP timestamp */ | |
117 | static struct timeval rtp_time_real; /* corresponding real time */ | |
118 | static uint16_t rtp_seq; /* frame sequence number */ | |
119 | static uint32_t rtp_id; /* RTP SSRC */ | |
120 | static int idled; /* set when idled */ | |
121 | static int audio_errors; /* audio error counter */ | |
460b9539 | 122 | |
123 | static const struct option options[] = { | |
124 | { "help", no_argument, 0, 'h' }, | |
125 | { "version", no_argument, 0, 'V' }, | |
126 | { "config", required_argument, 0, 'c' }, | |
127 | { "debug", no_argument, 0, 'd' }, | |
128 | { "no-debug", no_argument, 0, 'D' }, | |
129 | { 0, 0, 0, 0 } | |
130 | }; | |
131 | ||
132 | /* Display usage message and terminate. */ | |
133 | static void help(void) { | |
134 | xprintf("Usage:\n" | |
135 | " disorder-speaker [OPTIONS]\n" | |
136 | "Options:\n" | |
137 | " --help, -h Display usage message\n" | |
138 | " --version, -V Display version number\n" | |
139 | " --config PATH, -c PATH Set configuration file\n" | |
140 | " --debug, -d Turn on debugging\n" | |
141 | "\n" | |
142 | "Speaker process for DisOrder. Not intended to be run\n" | |
143 | "directly.\n"); | |
144 | xfclose(stdout); | |
145 | exit(0); | |
146 | } | |
147 | ||
148 | /* Display version number and terminate. */ | |
149 | static void version(void) { | |
150 | xprintf("disorder-speaker version %s\n", disorder_version_string); | |
151 | xfclose(stdout); | |
152 | exit(0); | |
153 | } | |
154 | ||
155 | /* Return the number of bytes per frame in FORMAT. */ | |
156 | static size_t bytes_per_frame(const ao_sample_format *format) { | |
157 | return format->channels * format->bits / 8; | |
158 | } | |
159 | ||
160 | /* Find track ID, maybe creating it if not found. */ | |
161 | static struct track *findtrack(const char *id, int create) { | |
162 | struct track *t; | |
163 | ||
164 | D(("findtrack %s %d", id, create)); | |
165 | for(t = tracks; t && strcmp(id, t->id); t = t->next) | |
166 | ; | |
167 | if(!t && create) { | |
168 | t = xmalloc(sizeof *t); | |
169 | t->next = tracks; | |
170 | strcpy(t->id, id); | |
171 | t->fd = -1; | |
172 | tracks = t; | |
173 | /* The initial input buffer will be the sample format. */ | |
174 | t->buffer = (void *)&t->format; | |
175 | t->size = sizeof t->format; | |
176 | } | |
177 | return t; | |
178 | } | |
179 | ||
180 | /* Remove track ID (but do not destroy it). */ | |
181 | static struct track *removetrack(const char *id) { | |
182 | struct track *t, **tt; | |
183 | ||
184 | D(("removetrack %s", id)); | |
185 | for(tt = &tracks; (t = *tt) && strcmp(id, t->id); tt = &t->next) | |
186 | ; | |
187 | if(t) | |
188 | *tt = t->next; | |
189 | return t; | |
190 | } | |
191 | ||
192 | /* Destroy a track. */ | |
193 | static void destroy(struct track *t) { | |
194 | D(("destroy %s", t->id)); | |
195 | if(t->fd != -1) xclose(t->fd); | |
196 | if(t->buffer != (void *)&t->format) free(t->buffer); | |
197 | free(t); | |
198 | } | |
199 | ||
200 | /* Notice a new FD. */ | |
201 | static void acquire(struct track *t, int fd) { | |
202 | D(("acquire %s %d", t->id, fd)); | |
203 | if(t->fd != -1) | |
204 | xclose(t->fd); | |
205 | t->fd = fd; | |
206 | nonblock(fd); | |
207 | } | |
208 | ||
209 | /* Read data into a sample buffer. Return 0 on success, -1 on EOF. */ | |
210 | static int fill(struct track *t) { | |
211 | size_t where, left; | |
212 | int n; | |
213 | ||
214 | D(("fill %s: eof=%d used=%zu size=%zu got_format=%d", | |
215 | t->id, t->eof, t->used, t->size, t->got_format)); | |
216 | if(t->eof) return -1; | |
217 | if(t->used < t->size) { | |
218 | /* there is room left in the buffer */ | |
219 | where = (t->start + t->used) % t->size; | |
220 | if(t->got_format) { | |
221 | /* We are reading audio data, get as much as we can */ | |
222 | if(where >= t->start) left = t->size - where; | |
223 | else left = t->start - where; | |
224 | } else | |
225 | /* We are still waiting for the format, only get that */ | |
226 | left = sizeof (ao_sample_format) - t->used; | |
227 | do { | |
228 | n = read(t->fd, t->buffer + where, left); | |
229 | } while(n < 0 && errno == EINTR); | |
230 | if(n < 0) { | |
231 | if(errno != EAGAIN) fatal(errno, "error reading sample stream"); | |
232 | return 0; | |
233 | } | |
234 | if(n == 0) { | |
235 | D(("fill %s: eof detected", t->id)); | |
236 | t->eof = 1; | |
237 | return -1; | |
238 | } | |
239 | t->used += n; | |
240 | if(!t->got_format && t->used >= sizeof (ao_sample_format)) { | |
241 | assert(t->used == sizeof (ao_sample_format)); | |
242 | /* Check that our assumptions are met. */ | |
243 | if(t->format.bits & 7) | |
244 | fatal(0, "bits per sample not a multiple of 8"); | |
245 | /* Make a new buffer for audio data. */ | |
246 | t->size = bytes_per_frame(&t->format) * t->format.rate * BUFFER_SECONDS; | |
247 | t->buffer = xmalloc(t->size); | |
248 | t->used = 0; | |
249 | t->got_format = 1; | |
250 | D(("got format for %s", t->id)); | |
251 | } | |
252 | } | |
253 | return 0; | |
254 | } | |
255 | ||
256 | /* Return true if A and B denote identical libao formats, else false. */ | |
257 | static int formats_equal(const ao_sample_format *a, | |
258 | const ao_sample_format *b) { | |
259 | return (a->bits == b->bits | |
260 | && a->rate == b->rate | |
261 | && a->channels == b->channels | |
262 | && a->byte_format == b->byte_format); | |
263 | } | |
264 | ||
265 | /* Close the sound device. */ | |
266 | static void idle(void) { | |
460b9539 | 267 | D(("idle")); |
8023f60b | 268 | #if API_ALSA |
e83d0967 | 269 | if(config->speaker_backend == BACKEND_ALSA && pcm) { |
8023f60b | 270 | int err; |
271 | ||
460b9539 | 272 | if((err = snd_pcm_nonblock(pcm, 0)) < 0) |
273 | fatal(0, "error calling snd_pcm_nonblock: %d", err); | |
274 | D(("draining pcm")); | |
275 | snd_pcm_drain(pcm); | |
276 | D(("closing pcm")); | |
277 | snd_pcm_close(pcm); | |
278 | pcm = 0; | |
279 | forceplay = 0; | |
280 | D(("released audio device")); | |
281 | } | |
8023f60b | 282 | #endif |
e83d0967 | 283 | idled = 1; |
9d5da576 | 284 | ready = 0; |
460b9539 | 285 | } |
286 | ||
287 | /* Abandon the current track */ | |
288 | static void abandon(void) { | |
289 | struct speaker_message sm; | |
290 | ||
291 | D(("abandon")); | |
292 | memset(&sm, 0, sizeof sm); | |
293 | sm.type = SM_FINISHED; | |
294 | strcpy(sm.id, playing->id); | |
295 | speaker_send(1, &sm, 0); | |
296 | removetrack(playing->id); | |
297 | destroy(playing); | |
298 | playing = 0; | |
299 | forceplay = 0; | |
300 | } | |
301 | ||
8023f60b | 302 | #if API_ALSA |
1c6e6a61 | 303 | static void log_params(snd_pcm_hw_params_t *hwparams, |
304 | snd_pcm_sw_params_t *swparams) { | |
305 | snd_pcm_uframes_t f; | |
306 | unsigned u; | |
307 | ||
0c207c37 | 308 | return; /* too verbose */ |
1c6e6a61 | 309 | if(hwparams) { |
310 | /* TODO */ | |
311 | } | |
312 | if(swparams) { | |
313 | snd_pcm_sw_params_get_silence_size(swparams, &f); | |
314 | info("sw silence_size=%lu", (unsigned long)f); | |
315 | snd_pcm_sw_params_get_silence_threshold(swparams, &f); | |
316 | info("sw silence_threshold=%lu", (unsigned long)f); | |
317 | snd_pcm_sw_params_get_sleep_min(swparams, &u); | |
318 | info("sw sleep_min=%lu", (unsigned long)u); | |
319 | snd_pcm_sw_params_get_start_threshold(swparams, &f); | |
320 | info("sw start_threshold=%lu", (unsigned long)f); | |
321 | snd_pcm_sw_params_get_stop_threshold(swparams, &f); | |
322 | info("sw stop_threshold=%lu", (unsigned long)f); | |
323 | snd_pcm_sw_params_get_xfer_align(swparams, &f); | |
324 | info("sw xfer_align=%lu", (unsigned long)f); | |
325 | } | |
326 | } | |
8023f60b | 327 | #endif |
1c6e6a61 | 328 | |
5330d674 | 329 | static void soxargs(const char ***pp, char **qq, ao_sample_format *ao) { |
9d5da576 | 330 | int n; |
331 | ||
332 | *(*pp)++ = "-t.raw"; | |
333 | *(*pp)++ = "-s"; | |
334 | *(*pp)++ = *qq; n = sprintf(*qq, "-r%d", ao->rate); *qq += n + 1; | |
5330d674 | 335 | *(*pp)++ = *qq; n = sprintf(*qq, "-c%d", ao->channels); *qq += n + 1; |
336 | /* sox 12.17.9 insists on -b etc; CVS sox insists on -<n> etc; both are | |
337 | * deployed! */ | |
338 | switch(config->sox_generation) { | |
339 | case 0: | |
340 | if(ao->bits != 8 | |
341 | && ao->byte_format != AO_FMT_NATIVE | |
342 | && ao->byte_format != MACHINE_AO_FMT) { | |
343 | *(*pp)++ = "-x"; | |
344 | } | |
345 | switch(ao->bits) { | |
346 | case 8: *(*pp)++ = "-b"; break; | |
347 | case 16: *(*pp)++ = "-w"; break; | |
348 | case 32: *(*pp)++ = "-l"; break; | |
349 | case 64: *(*pp)++ = "-d"; break; | |
350 | default: fatal(0, "cannot handle sample size %d", (int)ao->bits); | |
351 | } | |
352 | break; | |
353 | case 1: | |
354 | switch(ao->byte_format) { | |
9d5da576 | 355 | case AO_FMT_NATIVE: break; |
27801653 | 356 | case AO_FMT_BIG: *(*pp)++ = "-B"; break; |
357 | case AO_FMT_LITTLE: *(*pp)++ = "-L"; break; | |
5330d674 | 358 | } |
359 | *(*pp)++ = *qq; n = sprintf(*qq, "-%d", ao->bits/8); *qq += n + 1; | |
360 | break; | |
9d5da576 | 361 | } |
9d5da576 | 362 | } |
363 | ||
460b9539 | 364 | /* Make sure the sound device is open and has the right sample format. Return |
365 | * 0 on success and -1 on error. */ | |
366 | static int activate(void) { | |
460b9539 | 367 | /* If we don't know the format yet we cannot start. */ |
368 | if(!playing->got_format) { | |
369 | D((" - not got format for %s", playing->id)); | |
370 | return -1; | |
371 | } | |
e83d0967 RK |
372 | switch(config->speaker_backend) { |
373 | case BACKEND_COMMAND: | |
374 | case BACKEND_NETWORK: | |
375 | /* If we pass audio on to some other agent then we enforce the configured | |
376 | * sample format on the *inbound* audio data. */ | |
9d5da576 | 377 | if(!formats_equal(&playing->format, &config->sample_format)) { |
378 | char argbuf[1024], *q = argbuf; | |
379 | const char *av[18], **pp = av; | |
380 | int soxpipe[2]; | |
381 | pid_t soxkid; | |
382 | *pp++ = "sox"; | |
383 | soxargs(&pp, &q, &playing->format); | |
384 | *pp++ = "-"; | |
385 | soxargs(&pp, &q, &config->sample_format); | |
386 | *pp++ = "-"; | |
387 | *pp++ = 0; | |
388 | if(debugging) { | |
389 | for(pp = av; *pp; pp++) | |
390 | D(("sox arg[%d] = %s", pp - av, *pp)); | |
391 | D(("end args")); | |
392 | } | |
393 | xpipe(soxpipe); | |
394 | soxkid = xfork(); | |
395 | if(soxkid == 0) { | |
396 | xdup2(playing->fd, 0); | |
397 | xdup2(soxpipe[1], 1); | |
398 | fcntl(0, F_SETFL, fcntl(0, F_GETFL) & ~O_NONBLOCK); | |
399 | close(soxpipe[0]); | |
400 | close(soxpipe[1]); | |
401 | close(playing->fd); | |
402 | execvp("sox", (char **)av); | |
403 | _exit(1); | |
404 | } | |
405 | D(("forking sox for format conversion (kid = %d)", soxkid)); | |
406 | close(playing->fd); | |
407 | close(soxpipe[1]); | |
408 | playing->fd = soxpipe[0]; | |
409 | playing->format = config->sample_format; | |
410 | ready = 0; | |
411 | } | |
412 | if(!ready) { | |
413 | pcm_format = config->sample_format; | |
8023f60b | 414 | bufsize = 3 * FRAMES; |
9d5da576 | 415 | bpf = bytes_per_frame(&config->sample_format); |
416 | D(("acquired audio device")); | |
417 | ready = 1; | |
418 | } | |
419 | return 0; | |
e83d0967 | 420 | case BACKEND_ALSA: |
8023f60b | 421 | #if API_ALSA |
e83d0967 RK |
422 | /* If we need to change format then close the current device. */ |
423 | if(pcm && !formats_equal(&playing->format, &pcm_format)) | |
424 | idle(); | |
425 | if(!pcm) { | |
426 | snd_pcm_hw_params_t *hwparams; | |
427 | snd_pcm_sw_params_t *swparams; | |
428 | snd_pcm_uframes_t pcm_bufsize; | |
429 | int err; | |
430 | int sample_format = 0; | |
431 | unsigned rate; | |
432 | ||
433 | D(("snd_pcm_open")); | |
434 | if((err = snd_pcm_open(&pcm, | |
435 | config->device, | |
436 | SND_PCM_STREAM_PLAYBACK, | |
437 | SND_PCM_NONBLOCK))) { | |
438 | error(0, "error from snd_pcm_open: %d", err); | |
439 | goto error; | |
440 | } | |
441 | snd_pcm_hw_params_alloca(&hwparams); | |
442 | D(("set up hw params")); | |
443 | if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0) | |
444 | fatal(0, "error from snd_pcm_hw_params_any: %d", err); | |
445 | if((err = snd_pcm_hw_params_set_access(pcm, hwparams, | |
446 | SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) | |
447 | fatal(0, "error from snd_pcm_hw_params_set_access: %d", err); | |
448 | switch(playing->format.bits) { | |
449 | case 8: | |
450 | sample_format = SND_PCM_FORMAT_S8; | |
451 | break; | |
452 | case 16: | |
453 | switch(playing->format.byte_format) { | |
454 | case AO_FMT_NATIVE: sample_format = SND_PCM_FORMAT_S16; break; | |
455 | case AO_FMT_LITTLE: sample_format = SND_PCM_FORMAT_S16_LE; break; | |
456 | case AO_FMT_BIG: sample_format = SND_PCM_FORMAT_S16_BE; break; | |
457 | error(0, "unrecognized byte format %d", playing->format.byte_format); | |
458 | goto fatal; | |
459 | } | |
460 | break; | |
461 | default: | |
462 | error(0, "unsupported sample size %d", playing->format.bits); | |
460b9539 | 463 | goto fatal; |
464 | } | |
e83d0967 RK |
465 | if((err = snd_pcm_hw_params_set_format(pcm, hwparams, |
466 | sample_format)) < 0) { | |
467 | error(0, "error from snd_pcm_hw_params_set_format (%d): %d", | |
468 | sample_format, err); | |
469 | goto fatal; | |
470 | } | |
471 | rate = playing->format.rate; | |
472 | if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0) { | |
473 | error(0, "error from snd_pcm_hw_params_set_rate (%d): %d", | |
474 | playing->format.rate, err); | |
475 | goto fatal; | |
476 | } | |
477 | if(rate != (unsigned)playing->format.rate) | |
478 | info("want rate %d, got %u", playing->format.rate, rate); | |
479 | if((err = snd_pcm_hw_params_set_channels(pcm, hwparams, | |
480 | playing->format.channels)) < 0) { | |
481 | error(0, "error from snd_pcm_hw_params_set_channels (%d): %d", | |
482 | playing->format.channels, err); | |
483 | goto fatal; | |
484 | } | |
485 | bufsize = 3 * FRAMES; | |
486 | pcm_bufsize = bufsize; | |
487 | if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams, | |
488 | &pcm_bufsize)) < 0) | |
489 | fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d", | |
490 | 3 * FRAMES, err); | |
491 | if(pcm_bufsize != 3 * FRAMES && pcm_bufsize != last_pcm_bufsize) | |
492 | info("asked for PCM buffer of %d frames, got %d", | |
493 | 3 * FRAMES, (int)pcm_bufsize); | |
494 | last_pcm_bufsize = pcm_bufsize; | |
495 | if((err = snd_pcm_hw_params(pcm, hwparams)) < 0) | |
496 | fatal(0, "error calling snd_pcm_hw_params: %d", err); | |
497 | D(("set up sw params")); | |
498 | snd_pcm_sw_params_alloca(&swparams); | |
499 | if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0) | |
500 | fatal(0, "error calling snd_pcm_sw_params_current: %d", err); | |
501 | if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, FRAMES)) < 0) | |
502 | fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d", | |
503 | FRAMES, err); | |
504 | if((err = snd_pcm_sw_params(pcm, swparams)) < 0) | |
505 | fatal(0, "error calling snd_pcm_sw_params: %d", err); | |
506 | pcm_format = playing->format; | |
507 | bpf = bytes_per_frame(&pcm_format); | |
508 | D(("acquired audio device")); | |
509 | log_params(hwparams, swparams); | |
510 | ready = 1; | |
460b9539 | 511 | } |
e83d0967 RK |
512 | return 0; |
513 | fatal: | |
514 | abandon(); | |
515 | error: | |
516 | /* We assume the error is temporary and that we'll retry in a bit. */ | |
517 | if(pcm) { | |
518 | snd_pcm_close(pcm); | |
519 | pcm = 0; | |
460b9539 | 520 | } |
e83d0967 | 521 | return -1; |
8023f60b | 522 | #endif |
e83d0967 RK |
523 | default: |
524 | assert(!"reached"); | |
525 | } | |
460b9539 | 526 | } |
527 | ||
528 | /* Check to see whether the current track has finished playing */ | |
529 | static void maybe_finished(void) { | |
530 | if(playing | |
531 | && playing->eof | |
532 | && (!playing->got_format | |
533 | || playing->used < bytes_per_frame(&playing->format))) | |
534 | abandon(); | |
535 | } | |
536 | ||
e83d0967 RK |
537 | static void fork_cmd(void) { |
538 | pid_t cmdpid; | |
9d5da576 | 539 | int pfd[2]; |
e83d0967 | 540 | if(cmdfd != -1) close(cmdfd); |
9d5da576 | 541 | xpipe(pfd); |
e83d0967 RK |
542 | cmdpid = xfork(); |
543 | if(!cmdpid) { | |
9d5da576 | 544 | xdup2(pfd[0], 0); |
545 | close(pfd[0]); | |
546 | close(pfd[1]); | |
547 | execl("/bin/sh", "sh", "-c", config->speaker_command, (char *)0); | |
548 | fatal(errno, "error execing /bin/sh"); | |
549 | } | |
550 | close(pfd[0]); | |
e83d0967 RK |
551 | cmdfd = pfd[1]; |
552 | D(("forked cmd %d, fd = %d", cmdpid, cmdfd)); | |
9d5da576 | 553 | } |
554 | ||
460b9539 | 555 | static void play(size_t frames) { |
8023f60b | 556 | size_t avail_bytes, written_frames; |
9d5da576 | 557 | ssize_t written_bytes; |
0b75463f | 558 | struct rtp_header header; |
e83d0967 | 559 | struct iovec vec[2]; |
460b9539 | 560 | |
561 | if(activate()) { | |
562 | if(playing) | |
563 | forceplay = frames; | |
564 | else | |
565 | forceplay = 0; /* Must have called abandon() */ | |
566 | return; | |
567 | } | |
568 | D(("play: play %zu/%zu%s %dHz %db %dc", frames, playing->used / bpf, | |
569 | playing->eof ? " EOF" : "", | |
570 | playing->format.rate, | |
571 | playing->format.bits, | |
572 | playing->format.channels)); | |
573 | /* If we haven't got enough bytes yet wait until we have. Exception: when | |
574 | * we are at eof. */ | |
575 | if(playing->used < frames * bpf && !playing->eof) { | |
576 | forceplay = frames; | |
577 | return; | |
578 | } | |
579 | /* We have got enough data so don't force play again */ | |
580 | forceplay = 0; | |
581 | /* Figure out how many frames there are available to write */ | |
582 | if(playing->start + playing->used > playing->size) | |
583 | avail_bytes = playing->size - playing->start; | |
584 | else | |
585 | avail_bytes = playing->used; | |
9d5da576 | 586 | |
e83d0967 | 587 | switch(config->speaker_backend) { |
8023f60b | 588 | #if API_ALSA |
3a3c7bb9 | 589 | case BACKEND_ALSA: { |
8023f60b | 590 | snd_pcm_sframes_t pcm_written_frames; |
591 | size_t avail_frames; | |
592 | int err; | |
593 | ||
9d5da576 | 594 | avail_frames = avail_bytes / bpf; |
595 | if(avail_frames > frames) | |
596 | avail_frames = frames; | |
597 | if(!avail_frames) | |
460b9539 | 598 | return; |
8023f60b | 599 | pcm_written_frames = snd_pcm_writei(pcm, |
600 | playing->buffer + playing->start, | |
601 | avail_frames); | |
9d5da576 | 602 | D(("actually play %zu frames, wrote %d", |
8023f60b | 603 | avail_frames, (int)pcm_written_frames)); |
604 | if(pcm_written_frames < 0) { | |
605 | switch(pcm_written_frames) { | |
9d5da576 | 606 | case -EPIPE: /* underrun */ |
607 | error(0, "snd_pcm_writei reports underrun"); | |
608 | if((err = snd_pcm_prepare(pcm)) < 0) | |
609 | fatal(0, "error calling snd_pcm_prepare: %d", err); | |
610 | return; | |
611 | case -EAGAIN: | |
612 | return; | |
613 | default: | |
8023f60b | 614 | fatal(0, "error calling snd_pcm_writei: %d", |
615 | (int)pcm_written_frames); | |
9d5da576 | 616 | } |
617 | } | |
8023f60b | 618 | written_frames = pcm_written_frames; |
9d5da576 | 619 | written_bytes = written_frames * bpf; |
e83d0967 | 620 | break; |
3a3c7bb9 | 621 | } |
8023f60b | 622 | #endif |
e83d0967 | 623 | case BACKEND_COMMAND: |
9d5da576 | 624 | if(avail_bytes > frames * bpf) |
625 | avail_bytes = frames * bpf; | |
e83d0967 | 626 | written_bytes = write(cmdfd, playing->buffer + playing->start, |
9d5da576 | 627 | avail_bytes); |
628 | D(("actually play %zu bytes, wrote %d", | |
629 | avail_bytes, (int)written_bytes)); | |
630 | if(written_bytes < 0) { | |
631 | switch(errno) { | |
632 | case EPIPE: | |
e83d0967 RK |
633 | error(0, "hmm, command died; trying another"); |
634 | fork_cmd(); | |
9d5da576 | 635 | return; |
636 | case EAGAIN: | |
637 | return; | |
638 | } | |
460b9539 | 639 | } |
9d5da576 | 640 | written_frames = written_bytes / bpf; /* good enough */ |
e83d0967 RK |
641 | break; |
642 | case BACKEND_NETWORK: | |
643 | /* We transmit using RTP (RFC3550) and attempt to conform to the internet | |
644 | * AVT profile (RFC3551). */ | |
645 | if(rtp_time_real.tv_sec == 0) | |
646 | xgettimeofday(&rtp_time_real, 0); | |
647 | if(idled) { | |
648 | struct timeval now; | |
649 | xgettimeofday(&now, 0); | |
650 | /* There's been a gap. Fix up the RTP time accordingly. */ | |
651 | rtp_time += (((now.tv_sec + now.tv_usec /1000000.0) | |
652 | - (rtp_time_real.tv_sec + rtp_time_real.tv_usec / 1000000.0)) | |
653 | * playing->format.rate * playing->format.channels); | |
654 | } | |
655 | header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */ | |
656 | header.seq = htons(rtp_seq++); | |
657 | header.timestamp = htonl(rtp_time); | |
658 | header.ssrc = rtp_id; | |
659 | header.mpt = (idled ? 0x80 : 0x00) | 10; | |
660 | /* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from | |
661 | * the sample rate (in a library somewhere so that configuration.c can rule | |
662 | * out invalid rates). | |
663 | */ | |
664 | idled = 0; | |
665 | if(avail_bytes > NETWORK_BYTES - sizeof header) { | |
666 | avail_bytes = NETWORK_BYTES - sizeof header; | |
667 | avail_bytes -= avail_bytes % bpf; | |
668 | } | |
669 | /* "The RTP clock rate used for generating the RTP timestamp is independent | |
670 | * of the number of channels and the encoding; it equals the number of | |
671 | * sampling periods per second. For N-channel encodings, each sampling | |
672 | * period (say, 1/8000 of a second) generates N samples. (This terminology | |
673 | * is standard, but somewhat confusing, as the total number of samples | |
674 | * generated per second is then the sampling rate times the channel | |
675 | * count.)" | |
676 | */ | |
677 | vec[0].iov_base = (void *)&header; | |
678 | vec[0].iov_len = sizeof header; | |
679 | vec[1].iov_base = playing->buffer + playing->start; | |
680 | vec[1].iov_len = avail_bytes; | |
681 | #if 0 | |
682 | { | |
683 | char buffer[3 * sizeof header + 1]; | |
684 | size_t n; | |
685 | const uint8_t *ptr = (void *)&header; | |
686 | ||
687 | for(n = 0; n < sizeof header; ++n) | |
688 | sprintf(&buffer[3 * n], "%02x ", *ptr++); | |
689 | info(buffer); | |
690 | } | |
691 | #endif | |
692 | do { | |
693 | written_bytes = writev(bfd, | |
694 | vec, | |
695 | 2); | |
696 | } while(written_bytes < 0 && errno == EINTR); | |
697 | if(written_bytes < 0) { | |
698 | error(errno, "error transmitting audio data"); | |
699 | ++audio_errors; | |
700 | if(audio_errors == 10) | |
701 | fatal(0, "too many audio errors"); | |
702 | return; | |
703 | } | |
704 | audio_errors /= 2; | |
705 | written_bytes = avail_bytes; | |
706 | written_frames = written_bytes / bpf; | |
707 | /* Advance RTP's notion of the time */ | |
708 | rtp_time += written_frames * playing->format.channels; | |
709 | /* Advance the corresponding real time */ | |
710 | assert(NETWORK_BYTES <= 2000); /* else risk overflowing 32 bits */ | |
711 | rtp_time_real.tv_usec += written_frames * 1000000 / playing->format.rate; | |
712 | if(rtp_time_real.tv_usec >= 1000000) { | |
713 | ++rtp_time_real.tv_sec; | |
714 | rtp_time_real.tv_usec -= 1000000; | |
715 | } | |
716 | break; | |
717 | default: | |
718 | assert(!"reached"); | |
460b9539 | 719 | } |
e83d0967 RK |
720 | /* written_bytes and written_frames had better both be set and correct by |
721 | * this point */ | |
460b9539 | 722 | playing->start += written_bytes; |
723 | playing->used -= written_bytes; | |
724 | playing->played += written_frames; | |
725 | /* If the pointer is at the end of the buffer (or the buffer is completely | |
726 | * empty) wrap it back to the start. */ | |
727 | if(!playing->used || playing->start == playing->size) | |
728 | playing->start = 0; | |
729 | frames -= written_frames; | |
730 | } | |
731 | ||
732 | /* Notify the server what we're up to. */ | |
733 | static void report(void) { | |
734 | struct speaker_message sm; | |
735 | ||
736 | if(playing && playing->buffer != (void *)&playing->format) { | |
737 | memset(&sm, 0, sizeof sm); | |
738 | sm.type = paused ? SM_PAUSED : SM_PLAYING; | |
739 | strcpy(sm.id, playing->id); | |
740 | sm.data = playing->played / playing->format.rate; | |
741 | speaker_send(1, &sm, 0); | |
742 | } | |
743 | time(&last_report); | |
744 | } | |
745 | ||
9d5da576 | 746 | static void reap(int __attribute__((unused)) sig) { |
e83d0967 | 747 | pid_t cmdpid; |
9d5da576 | 748 | int st; |
749 | ||
750 | do | |
e83d0967 RK |
751 | cmdpid = waitpid(-1, &st, WNOHANG); |
752 | while(cmdpid > 0); | |
9d5da576 | 753 | signal(SIGCHLD, reap); |
754 | } | |
755 | ||
460b9539 | 756 | static int addfd(int fd, int events) { |
757 | if(fdno < NFDS) { | |
758 | fds[fdno].fd = fd; | |
759 | fds[fdno].events = events; | |
760 | return fdno++; | |
761 | } else | |
762 | return -1; | |
763 | } | |
764 | ||
765 | int main(int argc, char **argv) { | |
e83d0967 RK |
766 | int n, fd, stdin_slot, alsa_slots, cmdfd_slot, bfd_slot, poke, timeout; |
767 | struct timeval now, delta; | |
460b9539 | 768 | struct track *t; |
769 | struct speaker_message sm; | |
e83d0967 RK |
770 | struct addrinfo *res, *sres; |
771 | static const struct addrinfo pref = { | |
772 | 0, | |
773 | PF_INET, | |
774 | SOCK_DGRAM, | |
775 | IPPROTO_UDP, | |
776 | 0, | |
777 | 0, | |
778 | 0, | |
779 | 0 | |
780 | }; | |
781 | static const struct addrinfo prefbind = { | |
782 | AI_PASSIVE, | |
783 | PF_INET, | |
784 | SOCK_DGRAM, | |
785 | IPPROTO_UDP, | |
786 | 0, | |
787 | 0, | |
788 | 0, | |
789 | 0 | |
790 | }; | |
791 | static const int one = 1; | |
792 | char *sockname, *ssockname; | |
8023f60b | 793 | #if API_ALSA |
794 | int alsa_nslots = -1, err; | |
795 | #endif | |
460b9539 | 796 | |
797 | set_progname(argv); | |
460b9539 | 798 | if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); |
799 | while((n = getopt_long(argc, argv, "hVc:dD", options, 0)) >= 0) { | |
800 | switch(n) { | |
801 | case 'h': help(); | |
802 | case 'V': version(); | |
803 | case 'c': configfile = optarg; break; | |
804 | case 'd': debugging = 1; break; | |
805 | case 'D': debugging = 0; break; | |
806 | default: fatal(0, "invalid option"); | |
807 | } | |
808 | } | |
809 | if(getenv("DISORDER_DEBUG_SPEAKER")) debugging = 1; | |
810 | /* If stderr is a TTY then log there, otherwise to syslog. */ | |
811 | if(!isatty(2)) { | |
812 | openlog(progname, LOG_PID, LOG_DAEMON); | |
813 | log_default = &log_syslog; | |
814 | } | |
815 | if(config_read()) fatal(0, "cannot read configuration"); | |
816 | /* ignore SIGPIPE */ | |
817 | signal(SIGPIPE, SIG_IGN); | |
9d5da576 | 818 | /* reap kids */ |
819 | signal(SIGCHLD, reap); | |
460b9539 | 820 | /* set nice value */ |
821 | xnice(config->nice_speaker); | |
822 | /* change user */ | |
823 | become_mortal(); | |
824 | /* make sure we're not root, whatever the config says */ | |
825 | if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root"); | |
e83d0967 RK |
826 | switch(config->speaker_backend) { |
827 | case BACKEND_ALSA: | |
828 | info("selected ALSA backend"); | |
829 | case BACKEND_COMMAND: | |
830 | info("selected command backend"); | |
831 | fork_cmd(); | |
832 | break; | |
833 | case BACKEND_NETWORK: | |
834 | res = get_address(&config->broadcast, &pref, &sockname); | |
835 | if(!res) return -1; | |
836 | if(config->broadcast_from.n) { | |
837 | sres = get_address(&config->broadcast_from, &prefbind, &ssockname); | |
838 | if(!sres) return -1; | |
839 | } else | |
840 | sres = 0; | |
841 | if((bfd = socket(res->ai_family, | |
842 | res->ai_socktype, | |
843 | res->ai_protocol)) < 0) | |
844 | fatal(errno, "error creating broadcast socket"); | |
845 | if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0) | |
846 | fatal(errno, "error settting SO_BROADCAST on broadcast socket"); | |
847 | /* We might well want to set additional broadcast- or multicast-related | |
848 | * options here */ | |
849 | if(sres && bind(bfd, sres->ai_addr, sres->ai_addrlen) < 0) | |
850 | fatal(errno, "error binding broadcast socket to %s", ssockname); | |
851 | if(connect(bfd, res->ai_addr, res->ai_addrlen) < 0) | |
852 | fatal(errno, "error connecting broadcast socket to %s", sockname); | |
853 | /* Select an SSRC */ | |
854 | gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM); | |
855 | info("selected network backend, sending to %s", sockname); | |
856 | if(config->sample_format.byte_format != AO_FMT_BIG) { | |
857 | info("forcing big-endian sample format"); | |
858 | config->sample_format.byte_format = AO_FMT_BIG; | |
859 | } | |
860 | break; | |
861 | default: | |
862 | fatal(0, "unknown backend %d", config->speaker_backend); | |
8023f60b | 863 | } |
460b9539 | 864 | while(getppid() != 1) { |
865 | fdno = 0; | |
866 | /* Always ready for commands from the main server. */ | |
867 | stdin_slot = addfd(0, POLLIN); | |
868 | /* Try to read sample data for the currently playing track if there is | |
869 | * buffer space. */ | |
870 | if(playing && !playing->eof && playing->used < playing->size) { | |
871 | playing->slot = addfd(playing->fd, POLLIN); | |
872 | } else if(playing) | |
873 | playing->slot = -1; | |
874 | /* If forceplay is set then wait until it succeeds before waiting on the | |
875 | * sound device. */ | |
9d5da576 | 876 | alsa_slots = -1; |
e83d0967 RK |
877 | cmdfd_slot = -1; |
878 | bfd_slot = -1; | |
879 | /* By default we will wait up to a second before thinking about current | |
880 | * state. */ | |
881 | timeout = 1000; | |
8023f60b | 882 | if(ready && !forceplay) { |
e83d0967 RK |
883 | switch(config->speaker_backend) { |
884 | case BACKEND_COMMAND: | |
885 | /* We send sample data to the subprocess as fast as it can accept it. | |
886 | * This isn't ideal as pause latency can be very high as a result. */ | |
887 | if(cmdfd >= 0) | |
888 | cmdfd_slot = addfd(cmdfd, POLLOUT); | |
889 | break; | |
890 | case BACKEND_NETWORK: | |
891 | /* We want to keep the notional playing point somewhere in the near | |
892 | * future. If it's too near then clients that attempt even the | |
893 | * slightest amount of read-ahead will never catch up, and those that | |
894 | * don't will skip whenever there's a trivial network delay. If it's | |
895 | * too far ahead then pause latency will be too high. | |
896 | */ | |
897 | xgettimeofday(&now, 0); | |
898 | delta = tvsub(rtp_time_real, now); | |
899 | if(delta.tv_sec < RTP_AHEAD) { | |
900 | D(("delta = %ld.%06ld", (long)delta.tv_sec, (long)delta.tv_usec)); | |
901 | bfd_slot = addfd(bfd, POLLOUT); | |
902 | if(delta.tv_sec < 0) | |
903 | rtp_time_real = now; /* catch up */ | |
904 | } | |
905 | break; | |
8023f60b | 906 | #if API_ALSA |
3a3c7bb9 | 907 | case BACKEND_ALSA: { |
e83d0967 RK |
908 | /* We send sample data to ALSA as fast as it can accept it, relying on |
909 | * the fact that it has a relatively small buffer to minimize pause | |
910 | * latency. */ | |
9d5da576 | 911 | int retry = 3; |
912 | ||
913 | alsa_slots = fdno; | |
914 | do { | |
915 | retry = 0; | |
916 | alsa_nslots = snd_pcm_poll_descriptors(pcm, &fds[fdno], NFDS - fdno); | |
917 | if((alsa_nslots <= 0 | |
918 | || !(fds[alsa_slots].events & POLLOUT)) | |
919 | && snd_pcm_state(pcm) == SND_PCM_STATE_XRUN) { | |
920 | error(0, "underrun detected after call to snd_pcm_poll_descriptors()"); | |
921 | if((err = snd_pcm_prepare(pcm))) | |
922 | fatal(0, "error calling snd_pcm_prepare: %d", err); | |
923 | } else | |
924 | break; | |
925 | } while(retry-- > 0); | |
926 | if(alsa_nslots >= 0) | |
927 | fdno += alsa_nslots; | |
e83d0967 | 928 | break; |
3a3c7bb9 | 929 | } |
8023f60b | 930 | #endif |
e83d0967 RK |
931 | default: |
932 | assert(!"unknown backend"); | |
9d5da576 | 933 | } |
934 | } | |
460b9539 | 935 | /* If any other tracks don't have a full buffer, try to read sample data |
936 | * from them. */ | |
937 | for(t = tracks; t; t = t->next) | |
938 | if(t != playing) { | |
939 | if(!t->eof && t->used < t->size) { | |
9d5da576 | 940 | t->slot = addfd(t->fd, POLLIN | POLLHUP); |
460b9539 | 941 | } else |
942 | t->slot = -1; | |
943 | } | |
e83d0967 RK |
944 | /* Wait for something interesting to happen */ |
945 | n = poll(fds, fdno, timeout); | |
460b9539 | 946 | if(n < 0) { |
947 | if(errno == EINTR) continue; | |
948 | fatal(errno, "error calling poll"); | |
949 | } | |
950 | /* Play some sound before doing anything else */ | |
e83d0967 RK |
951 | poke = 0; |
952 | switch(config->speaker_backend) { | |
8023f60b | 953 | #if API_ALSA |
e83d0967 RK |
954 | case BACKEND_ALSA: |
955 | if(alsa_slots != -1) { | |
956 | unsigned short alsa_revents; | |
957 | ||
958 | if((err = snd_pcm_poll_descriptors_revents(pcm, | |
959 | &fds[alsa_slots], | |
960 | alsa_nslots, | |
961 | &alsa_revents)) < 0) | |
962 | fatal(0, "error calling snd_pcm_poll_descriptors_revents: %d", err); | |
963 | if(alsa_revents & (POLLOUT | POLLERR)) | |
964 | play(3 * FRAMES); | |
965 | } else | |
966 | poke = 1; | |
967 | break; | |
8023f60b | 968 | #endif |
e83d0967 RK |
969 | case BACKEND_COMMAND: |
970 | if(cmdfd_slot != -1) { | |
971 | if(fds[cmdfd_slot].revents & (POLLOUT | POLLERR)) | |
972 | play(3 * FRAMES); | |
973 | } else | |
974 | poke = 1; | |
975 | break; | |
976 | case BACKEND_NETWORK: | |
977 | if(bfd_slot != -1) { | |
978 | if(fds[bfd_slot].revents & (POLLOUT | POLLERR)) | |
979 | play(3 * FRAMES); | |
980 | } else | |
981 | poke = 1; | |
982 | break; | |
983 | } | |
984 | if(poke) { | |
460b9539 | 985 | /* Some attempt to play must have failed */ |
986 | if(playing && !paused) | |
987 | play(forceplay); | |
988 | else | |
989 | forceplay = 0; /* just in case */ | |
990 | } | |
991 | /* Perhaps we have a command to process */ | |
992 | if(fds[stdin_slot].revents & POLLIN) { | |
993 | n = speaker_recv(0, &sm, &fd); | |
994 | if(n > 0) | |
995 | switch(sm.type) { | |
996 | case SM_PREPARE: | |
997 | D(("SM_PREPARE %s %d", sm.id, fd)); | |
998 | if(fd == -1) fatal(0, "got SM_PREPARE but no file descriptor"); | |
999 | t = findtrack(sm.id, 1); | |
1000 | acquire(t, fd); | |
1001 | break; | |
1002 | case SM_PLAY: | |
1003 | D(("SM_PLAY %s %d", sm.id, fd)); | |
1004 | if(playing) fatal(0, "got SM_PLAY but already playing something"); | |
1005 | t = findtrack(sm.id, 1); | |
1006 | if(fd != -1) acquire(t, fd); | |
1007 | playing = t; | |
8023f60b | 1008 | play(bufsize); |
460b9539 | 1009 | report(); |
1010 | break; | |
1011 | case SM_PAUSE: | |
1012 | D(("SM_PAUSE")); | |
1013 | paused = 1; | |
1014 | report(); | |
1015 | break; | |
1016 | case SM_RESUME: | |
1017 | D(("SM_RESUME")); | |
1018 | if(paused) { | |
1019 | paused = 0; | |
1020 | if(playing) | |
8023f60b | 1021 | play(bufsize); |
460b9539 | 1022 | } |
1023 | report(); | |
1024 | break; | |
1025 | case SM_CANCEL: | |
1026 | D(("SM_CANCEL %s", sm.id)); | |
1027 | t = removetrack(sm.id); | |
1028 | if(t) { | |
1029 | if(t == playing) { | |
1030 | sm.type = SM_FINISHED; | |
1031 | strcpy(sm.id, playing->id); | |
1032 | speaker_send(1, &sm, 0); | |
1033 | playing = 0; | |
1034 | } | |
1035 | destroy(t); | |
1036 | } else | |
1037 | error(0, "SM_CANCEL for unknown track %s", sm.id); | |
1038 | report(); | |
1039 | break; | |
1040 | case SM_RELOAD: | |
1041 | D(("SM_RELOAD")); | |
1042 | if(config_read()) error(0, "cannot read configuration"); | |
1043 | info("reloaded configuration"); | |
1044 | break; | |
1045 | default: | |
1046 | error(0, "unknown message type %d", sm.type); | |
1047 | } | |
1048 | } | |
1049 | /* Read in any buffered data */ | |
1050 | for(t = tracks; t; t = t->next) | |
9d5da576 | 1051 | if(t->slot != -1 && (fds[t->slot].revents & (POLLIN | POLLHUP))) |
460b9539 | 1052 | fill(t); |
1053 | /* We might be able to play now */ | |
9d5da576 | 1054 | if(ready && forceplay && playing && !paused) |
460b9539 | 1055 | play(forceplay); |
1056 | /* Maybe we finished playing a track somewhere in the above */ | |
1057 | maybe_finished(); | |
1058 | /* If we don't need the sound device for now then close it for the benefit | |
1059 | * of anyone else who wants it. */ | |
9d5da576 | 1060 | if((!playing || paused) && ready) |
460b9539 | 1061 | idle(); |
1062 | /* If we've not reported out state for a second do so now. */ | |
1063 | if(time(0) > last_report) | |
1064 | report(); | |
1065 | } | |
1066 | info("stopped (parent terminated)"); | |
1067 | exit(0); | |
1068 | } | |
1069 | ||
1070 | /* | |
1071 | Local Variables: | |
1072 | c-basic-offset:2 | |
1073 | comment-column:40 | |
1074 | fill-column:79 | |
1075 | indent-tabs-mode:nil | |
1076 | End: | |
1077 | */ |