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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 | */ | |
1674096e | 20 | /** @file server/speaker.c |
cf714d85 | 21 | * @brief Speaker process |
1674096e | 22 | * |
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 via file descriptor | |
26 | * passing from the main server and plays them in the right order. | |
27 | * | |
795192f4 | 28 | * @b Encodings. For the <a href="http://www.alsa-project.org/">ALSA</a> API, |
29 | * 8- and 16- bit stereo and mono are supported, with any sample rate (within | |
30 | * the limits that ALSA can deal with.) | |
1674096e | 31 | * |
32 | * When communicating with a subprocess, <a | |
33 | * href="http://sox.sourceforge.net/">sox</a> is invoked to convert the inbound | |
34 | * data to a single consistent format. The same applies for network (RTP) | |
35 | * play, though in that case currently only 44.1KHz 16-bit stereo is supported. | |
36 | * | |
37 | * The inbound data starts with a structure defining the data format. Note | |
38 | * that this is NOT portable between different platforms or even necessarily | |
39 | * between versions; the speaker is assumed to be built from the same source | |
40 | * and run on the same host as the main server. | |
41 | * | |
795192f4 | 42 | * @b Garbage @b Collection. This program deliberately does not use the |
43 | * garbage collector even though it might be convenient to do so. This is for | |
44 | * two reasons. Firstly some sound APIs use thread threads and we do not want | |
45 | * to have to deal with potential interactions between threading and garbage | |
46 | * collection. Secondly this process needs to be able to respond quickly and | |
47 | * this is not compatible with the collector hanging the program even | |
48 | * relatively briefly. | |
49 | * | |
50 | * @b Units. This program thinks at various times in three different units. | |
51 | * Bytes are obvious. A sample is a single sample on a single channel. A | |
52 | * frame is several samples on different channels at the same point in time. | |
53 | * So (for instance) a 16-bit stereo frame is 4 bytes and consists of a pair of | |
54 | * 2-byte samples. | |
1674096e | 55 | */ |
460b9539 | 56 | |
57 | #include <config.h> | |
58 | #include "types.h" | |
59 | ||
60 | #include <getopt.h> | |
61 | #include <stdio.h> | |
62 | #include <stdlib.h> | |
63 | #include <locale.h> | |
64 | #include <syslog.h> | |
65 | #include <unistd.h> | |
66 | #include <errno.h> | |
67 | #include <ao/ao.h> | |
68 | #include <string.h> | |
69 | #include <assert.h> | |
70 | #include <sys/select.h> | |
9d5da576 | 71 | #include <sys/wait.h> |
460b9539 | 72 | #include <time.h> |
8023f60b | 73 | #include <fcntl.h> |
74 | #include <poll.h> | |
e83d0967 RK |
75 | #include <sys/socket.h> |
76 | #include <netdb.h> | |
77 | #include <gcrypt.h> | |
78 | #include <sys/uio.h> | |
460b9539 | 79 | |
80 | #include "configuration.h" | |
81 | #include "syscalls.h" | |
82 | #include "log.h" | |
83 | #include "defs.h" | |
84 | #include "mem.h" | |
ea410ba1 | 85 | #include "speaker-protocol.h" |
460b9539 | 86 | #include "user.h" |
e83d0967 RK |
87 | #include "addr.h" |
88 | #include "timeval.h" | |
89 | #include "rtp.h" | |
cf714d85 | 90 | #include "speaker.h" |
460b9539 | 91 | |
8023f60b | 92 | #if API_ALSA |
dea8f8aa | 93 | #include <alsa/asoundlib.h> |
8023f60b | 94 | #endif |
dea8f8aa | 95 | |
cf714d85 | 96 | /** @brief Linked list of all prepared tracks */ |
97 | struct track *tracks; | |
e83d0967 | 98 | |
cf714d85 | 99 | /** @brief Playing track, or NULL */ |
100 | struct track *playing; | |
460b9539 | 101 | |
102 | static time_t last_report; /* when we last reported */ | |
103 | static int paused; /* pause status */ | |
460b9539 | 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 | #if API_ALSA |
50ae38dd | 108 | /** @brief The current PCM handle */ |
109 | static snd_pcm_t *pcm; | |
0c207c37 | 110 | static snd_pcm_uframes_t last_pcm_bufsize; /* last seen buffer size */ |
8023f60b | 111 | #endif |
50ae38dd | 112 | |
5a7c42a8 | 113 | /** @brief The current device state */ |
114 | enum device_states device_state; | |
50ae38dd | 115 | |
5a7c42a8 | 116 | /** @brief The current device sample format |
55f35f2d | 117 | * |
5a7c42a8 | 118 | * Only meaningful if @ref device_state = @ref device_open or perhaps @ref |
119 | * device_error. For @ref FIXED_FORMAT backends, this should always match @c | |
120 | * config->sample_format. | |
55f35f2d | 121 | */ |
5a7c42a8 | 122 | ao_sample_format device_format; |
55f35f2d | 123 | |
124 | /** @brief Pipe to subprocess | |
125 | * | |
126 | * This is the file descriptor to write to for @ref BACKEND_COMMAND. | |
127 | */ | |
128 | static int cmdfd = -1; | |
129 | ||
130 | /** @brief Network socket | |
131 | * | |
132 | * This is the file descriptor to write to for @ref BACKEND_NETWORK. | |
133 | */ | |
134 | static int bfd = -1; | |
7aa087a7 RK |
135 | |
136 | /** @brief RTP timestamp | |
137 | * | |
138 | * This counts the number of samples played (NB not the number of frames | |
139 | * played). | |
140 | * | |
141 | * The timestamp in the packet header is only 32 bits wide. With 44100Hz | |
142 | * stereo, that only gives about half a day before wrapping, which is not | |
143 | * particularly convenient for certain debugging purposes. Therefore the | |
144 | * timestamp is maintained as a 64-bit integer, giving around six million years | |
145 | * before wrapping, and truncated to 32 bits when transmitting. | |
146 | */ | |
147 | static uint64_t rtp_time; | |
148 | ||
149 | /** @brief RTP base timestamp | |
150 | * | |
151 | * This is the real time correspoding to an @ref rtp_time of 0. It is used | |
152 | * to recalculate the timestamp after idle periods. | |
153 | */ | |
154 | static struct timeval rtp_time_0; | |
155 | ||
55f35f2d | 156 | /** @brief RTP packet sequence number */ |
157 | static uint16_t rtp_seq; | |
158 | ||
159 | /** @brief RTP SSRC */ | |
160 | static uint32_t rtp_id; | |
161 | ||
162 | /** @brief Set when idled | |
163 | * | |
164 | * This is set when the sound device is deliberately closed by idle(). | |
55f35f2d | 165 | */ |
e83d0967 | 166 | static int idled; /* set when idled */ |
55f35f2d | 167 | |
168 | /** @brief Error counter */ | |
169 | static int audio_errors; | |
460b9539 | 170 | |
29601377 | 171 | /** @brief Selected backend */ |
172 | static const struct speaker_backend *backend; | |
173 | ||
460b9539 | 174 | static const struct option options[] = { |
175 | { "help", no_argument, 0, 'h' }, | |
176 | { "version", no_argument, 0, 'V' }, | |
177 | { "config", required_argument, 0, 'c' }, | |
178 | { "debug", no_argument, 0, 'd' }, | |
179 | { "no-debug", no_argument, 0, 'D' }, | |
180 | { 0, 0, 0, 0 } | |
181 | }; | |
182 | ||
183 | /* Display usage message and terminate. */ | |
184 | static void help(void) { | |
185 | xprintf("Usage:\n" | |
186 | " disorder-speaker [OPTIONS]\n" | |
187 | "Options:\n" | |
188 | " --help, -h Display usage message\n" | |
189 | " --version, -V Display version number\n" | |
190 | " --config PATH, -c PATH Set configuration file\n" | |
191 | " --debug, -d Turn on debugging\n" | |
192 | "\n" | |
193 | "Speaker process for DisOrder. Not intended to be run\n" | |
194 | "directly.\n"); | |
195 | xfclose(stdout); | |
196 | exit(0); | |
197 | } | |
198 | ||
199 | /* Display version number and terminate. */ | |
200 | static void version(void) { | |
201 | xprintf("disorder-speaker version %s\n", disorder_version_string); | |
202 | xfclose(stdout); | |
203 | exit(0); | |
204 | } | |
205 | ||
1674096e | 206 | /** @brief Return the number of bytes per frame in @p format */ |
460b9539 | 207 | static size_t bytes_per_frame(const ao_sample_format *format) { |
208 | return format->channels * format->bits / 8; | |
209 | } | |
210 | ||
1674096e | 211 | /** @brief Find track @p id, maybe creating it if not found */ |
460b9539 | 212 | static struct track *findtrack(const char *id, int create) { |
213 | struct track *t; | |
214 | ||
215 | D(("findtrack %s %d", id, create)); | |
216 | for(t = tracks; t && strcmp(id, t->id); t = t->next) | |
217 | ; | |
218 | if(!t && create) { | |
219 | t = xmalloc(sizeof *t); | |
220 | t->next = tracks; | |
221 | strcpy(t->id, id); | |
222 | t->fd = -1; | |
223 | tracks = t; | |
224 | /* The initial input buffer will be the sample format. */ | |
225 | t->buffer = (void *)&t->format; | |
226 | t->size = sizeof t->format; | |
227 | } | |
228 | return t; | |
229 | } | |
230 | ||
1674096e | 231 | /** @brief Remove track @p id (but do not destroy it) */ |
460b9539 | 232 | static struct track *removetrack(const char *id) { |
233 | struct track *t, **tt; | |
234 | ||
235 | D(("removetrack %s", id)); | |
236 | for(tt = &tracks; (t = *tt) && strcmp(id, t->id); tt = &t->next) | |
237 | ; | |
238 | if(t) | |
239 | *tt = t->next; | |
240 | return t; | |
241 | } | |
242 | ||
1674096e | 243 | /** @brief Destroy a track */ |
460b9539 | 244 | static void destroy(struct track *t) { |
245 | D(("destroy %s", t->id)); | |
246 | if(t->fd != -1) xclose(t->fd); | |
247 | if(t->buffer != (void *)&t->format) free(t->buffer); | |
248 | free(t); | |
249 | } | |
250 | ||
1674096e | 251 | /** @brief Notice a new connection */ |
460b9539 | 252 | static void acquire(struct track *t, int fd) { |
253 | D(("acquire %s %d", t->id, fd)); | |
254 | if(t->fd != -1) | |
255 | xclose(t->fd); | |
256 | t->fd = fd; | |
257 | nonblock(fd); | |
258 | } | |
259 | ||
1674096e | 260 | /** @brief Return true if A and B denote identical libao formats, else false */ |
261 | static int formats_equal(const ao_sample_format *a, | |
262 | const ao_sample_format *b) { | |
263 | return (a->bits == b->bits | |
264 | && a->rate == b->rate | |
265 | && a->channels == b->channels | |
266 | && a->byte_format == b->byte_format); | |
267 | } | |
268 | ||
269 | /** @brief Compute arguments to sox */ | |
270 | static void soxargs(const char ***pp, char **qq, ao_sample_format *ao) { | |
271 | int n; | |
272 | ||
273 | *(*pp)++ = "-t.raw"; | |
274 | *(*pp)++ = "-s"; | |
275 | *(*pp)++ = *qq; n = sprintf(*qq, "-r%d", ao->rate); *qq += n + 1; | |
276 | *(*pp)++ = *qq; n = sprintf(*qq, "-c%d", ao->channels); *qq += n + 1; | |
277 | /* sox 12.17.9 insists on -b etc; CVS sox insists on -<n> etc; both are | |
278 | * deployed! */ | |
279 | switch(config->sox_generation) { | |
280 | case 0: | |
281 | if(ao->bits != 8 | |
282 | && ao->byte_format != AO_FMT_NATIVE | |
283 | && ao->byte_format != MACHINE_AO_FMT) { | |
284 | *(*pp)++ = "-x"; | |
285 | } | |
286 | switch(ao->bits) { | |
287 | case 8: *(*pp)++ = "-b"; break; | |
288 | case 16: *(*pp)++ = "-w"; break; | |
289 | case 32: *(*pp)++ = "-l"; break; | |
290 | case 64: *(*pp)++ = "-d"; break; | |
291 | default: fatal(0, "cannot handle sample size %d", (int)ao->bits); | |
292 | } | |
293 | break; | |
294 | case 1: | |
295 | switch(ao->byte_format) { | |
296 | case AO_FMT_NATIVE: break; | |
297 | case AO_FMT_BIG: *(*pp)++ = "-B"; break; | |
298 | case AO_FMT_LITTLE: *(*pp)++ = "-L"; break; | |
299 | } | |
300 | *(*pp)++ = *qq; n = sprintf(*qq, "-%d", ao->bits/8); *qq += n + 1; | |
301 | break; | |
302 | } | |
303 | } | |
304 | ||
305 | /** @brief Enable format translation | |
306 | * | |
307 | * If necessary, replaces a tracks inbound file descriptor with one connected | |
308 | * to a sox invocation, which performs the required translation. | |
309 | */ | |
310 | static void enable_translation(struct track *t) { | |
0763e1f4 | 311 | if((backend->flags & FIXED_FORMAT) |
312 | && !formats_equal(&t->format, &config->sample_format)) { | |
1674096e | 313 | char argbuf[1024], *q = argbuf; |
314 | const char *av[18], **pp = av; | |
315 | int soxpipe[2]; | |
316 | pid_t soxkid; | |
317 | ||
318 | *pp++ = "sox"; | |
319 | soxargs(&pp, &q, &t->format); | |
320 | *pp++ = "-"; | |
321 | soxargs(&pp, &q, &config->sample_format); | |
322 | *pp++ = "-"; | |
323 | *pp++ = 0; | |
324 | if(debugging) { | |
325 | for(pp = av; *pp; pp++) | |
326 | D(("sox arg[%d] = %s", pp - av, *pp)); | |
327 | D(("end args")); | |
328 | } | |
329 | xpipe(soxpipe); | |
330 | soxkid = xfork(); | |
331 | if(soxkid == 0) { | |
332 | signal(SIGPIPE, SIG_DFL); | |
333 | xdup2(t->fd, 0); | |
334 | xdup2(soxpipe[1], 1); | |
335 | fcntl(0, F_SETFL, fcntl(0, F_GETFL) & ~O_NONBLOCK); | |
336 | close(soxpipe[0]); | |
337 | close(soxpipe[1]); | |
338 | close(t->fd); | |
339 | execvp("sox", (char **)av); | |
340 | _exit(1); | |
341 | } | |
342 | D(("forking sox for format conversion (kid = %d)", soxkid)); | |
343 | close(t->fd); | |
344 | close(soxpipe[1]); | |
345 | t->fd = soxpipe[0]; | |
346 | t->format = config->sample_format; | |
1674096e | 347 | } |
348 | } | |
349 | ||
350 | /** @brief Read data into a sample buffer | |
351 | * @param t Pointer to track | |
352 | * @return 0 on success, -1 on EOF | |
353 | * | |
55f35f2d | 354 | * This is effectively the read callback on @c t->fd. It is called from the |
355 | * main loop whenever the track's file descriptor is readable, assuming the | |
356 | * buffer has not reached the maximum allowed occupancy. | |
1674096e | 357 | */ |
460b9539 | 358 | static int fill(struct track *t) { |
359 | size_t where, left; | |
360 | int n; | |
361 | ||
362 | D(("fill %s: eof=%d used=%zu size=%zu got_format=%d", | |
363 | t->id, t->eof, t->used, t->size, t->got_format)); | |
364 | if(t->eof) return -1; | |
365 | if(t->used < t->size) { | |
366 | /* there is room left in the buffer */ | |
367 | where = (t->start + t->used) % t->size; | |
368 | if(t->got_format) { | |
369 | /* We are reading audio data, get as much as we can */ | |
370 | if(where >= t->start) left = t->size - where; | |
371 | else left = t->start - where; | |
372 | } else | |
373 | /* We are still waiting for the format, only get that */ | |
374 | left = sizeof (ao_sample_format) - t->used; | |
375 | do { | |
376 | n = read(t->fd, t->buffer + where, left); | |
377 | } while(n < 0 && errno == EINTR); | |
378 | if(n < 0) { | |
379 | if(errno != EAGAIN) fatal(errno, "error reading sample stream"); | |
380 | return 0; | |
381 | } | |
382 | if(n == 0) { | |
383 | D(("fill %s: eof detected", t->id)); | |
384 | t->eof = 1; | |
385 | return -1; | |
386 | } | |
387 | t->used += n; | |
388 | if(!t->got_format && t->used >= sizeof (ao_sample_format)) { | |
389 | assert(t->used == sizeof (ao_sample_format)); | |
390 | /* Check that our assumptions are met. */ | |
391 | if(t->format.bits & 7) | |
392 | fatal(0, "bits per sample not a multiple of 8"); | |
1674096e | 393 | /* If the input format is unsuitable, arrange to translate it */ |
394 | enable_translation(t); | |
460b9539 | 395 | /* Make a new buffer for audio data. */ |
396 | t->size = bytes_per_frame(&t->format) * t->format.rate * BUFFER_SECONDS; | |
397 | t->buffer = xmalloc(t->size); | |
398 | t->used = 0; | |
399 | t->got_format = 1; | |
400 | D(("got format for %s", t->id)); | |
401 | } | |
402 | } | |
403 | return 0; | |
404 | } | |
405 | ||
55f35f2d | 406 | /** @brief Close the sound device |
407 | * | |
408 | * This is called to deactivate the output device when pausing, and also by the | |
409 | * ALSA backend when changing encoding (in which case the sound device will be | |
410 | * immediately reactivated). | |
411 | */ | |
460b9539 | 412 | static void idle(void) { |
460b9539 | 413 | D(("idle")); |
5a7c42a8 | 414 | if(backend->deactivate) |
b5a99ad0 | 415 | backend->deactivate(); |
5a7c42a8 | 416 | else |
417 | device_state = device_closed; | |
e83d0967 | 418 | idled = 1; |
460b9539 | 419 | } |
420 | ||
1674096e | 421 | /** @brief Abandon the current track */ |
460b9539 | 422 | static void abandon(void) { |
423 | struct speaker_message sm; | |
424 | ||
425 | D(("abandon")); | |
426 | memset(&sm, 0, sizeof sm); | |
427 | sm.type = SM_FINISHED; | |
428 | strcpy(sm.id, playing->id); | |
429 | speaker_send(1, &sm, 0); | |
430 | removetrack(playing->id); | |
431 | destroy(playing); | |
432 | playing = 0; | |
1c6e6a61 | 433 | } |
434 | ||
1674096e | 435 | /** @brief Enable sound output |
436 | * | |
437 | * Makes sure the sound device is open and has the right sample format. Return | |
438 | * 0 on success and -1 on error. | |
439 | */ | |
5a7c42a8 | 440 | static void activate(void) { |
460b9539 | 441 | /* If we don't know the format yet we cannot start. */ |
442 | if(!playing->got_format) { | |
443 | D((" - not got format for %s", playing->id)); | |
5a7c42a8 | 444 | return; |
460b9539 | 445 | } |
5a7c42a8 | 446 | if(backend->flags & FIXED_FORMAT) |
447 | device_format = config->sample_format; | |
448 | if(backend->activate) { | |
449 | backend->activate(); | |
450 | } else { | |
451 | assert(backend->flags & FIXED_FORMAT); | |
452 | /* ...otherwise device_format not set */ | |
453 | device_state = device_open; | |
454 | } | |
455 | if(device_state == device_open) | |
456 | bpf = bytes_per_frame(&device_format); | |
460b9539 | 457 | } |
458 | ||
55f35f2d | 459 | /** @brief Check whether the current track has finished |
460 | * | |
461 | * The current track is determined to have finished either if the input stream | |
462 | * eded before the format could be determined (i.e. it is malformed) or the | |
463 | * input is at end of file and there is less than a frame left unplayed. (So | |
464 | * it copes with decoders that crash mid-frame.) | |
465 | */ | |
460b9539 | 466 | static void maybe_finished(void) { |
467 | if(playing | |
468 | && playing->eof | |
469 | && (!playing->got_format | |
470 | || playing->used < bytes_per_frame(&playing->format))) | |
471 | abandon(); | |
472 | } | |
473 | ||
5a7c42a8 | 474 | /** @brief Play up to @p frames frames of audio |
475 | * | |
476 | * It is always safe to call this function. | |
477 | * - If @ref playing is 0 then it will just return | |
478 | * - If @ref paused is non-0 then it will just return | |
479 | * - If @ref device_state != @ref device_open then it will call activate() and | |
480 | * return if it it fails. | |
481 | * - If there is not enough audio to play then it play what is available. | |
482 | * | |
483 | * If there are not enough frames to play then whatever is available is played | |
484 | * instead. It is up to mainloop() to ensure that play() is not called when | |
485 | * unreasonably only an small amounts of data is available to play. | |
486 | */ | |
460b9539 | 487 | static void play(size_t frames) { |
3c68b773 | 488 | size_t avail_frames, avail_bytes, written_frames; |
9d5da576 | 489 | ssize_t written_bytes; |
460b9539 | 490 | |
5a7c42a8 | 491 | /* Make sure there's a track to play and it is not pasued */ |
492 | if(!playing || paused) | |
460b9539 | 493 | return; |
5a7c42a8 | 494 | /* Make sure the output device is open and has the right sample format */ |
495 | if(device_state != device_open | |
496 | || !formats_equal(&device_format, &playing->format)) { | |
497 | activate(); | |
498 | if(device_state != device_open) | |
499 | return; | |
460b9539 | 500 | } |
501 | D(("play: play %zu/%zu%s %dHz %db %dc", frames, playing->used / bpf, | |
502 | playing->eof ? " EOF" : "", | |
503 | playing->format.rate, | |
504 | playing->format.bits, | |
505 | playing->format.channels)); | |
460b9539 | 506 | /* Figure out how many frames there are available to write */ |
507 | if(playing->start + playing->used > playing->size) | |
7f9d5847 | 508 | /* The ring buffer is currently wrapped, only play up to the wrap point */ |
460b9539 | 509 | avail_bytes = playing->size - playing->start; |
510 | else | |
7f9d5847 | 511 | /* The ring buffer is not wrapped, can play the lot */ |
460b9539 | 512 | avail_bytes = playing->used; |
7f9d5847 | 513 | avail_frames = avail_bytes / bpf; |
514 | /* Only play up to the requested amount */ | |
515 | if(avail_frames > frames) | |
516 | avail_frames = frames; | |
517 | if(!avail_frames) | |
518 | return; | |
3c68b773 | 519 | /* Play it, Sam */ |
520 | written_frames = backend->play(avail_frames); | |
544a9ec1 | 521 | written_bytes = written_frames * bpf; |
e83d0967 RK |
522 | /* written_bytes and written_frames had better both be set and correct by |
523 | * this point */ | |
460b9539 | 524 | playing->start += written_bytes; |
525 | playing->used -= written_bytes; | |
526 | playing->played += written_frames; | |
527 | /* If the pointer is at the end of the buffer (or the buffer is completely | |
528 | * empty) wrap it back to the start. */ | |
529 | if(!playing->used || playing->start == playing->size) | |
530 | playing->start = 0; | |
531 | frames -= written_frames; | |
5a7c42a8 | 532 | return; |
460b9539 | 533 | } |
534 | ||
535 | /* Notify the server what we're up to. */ | |
536 | static void report(void) { | |
537 | struct speaker_message sm; | |
538 | ||
539 | if(playing && playing->buffer != (void *)&playing->format) { | |
540 | memset(&sm, 0, sizeof sm); | |
541 | sm.type = paused ? SM_PAUSED : SM_PLAYING; | |
542 | strcpy(sm.id, playing->id); | |
543 | sm.data = playing->played / playing->format.rate; | |
544 | speaker_send(1, &sm, 0); | |
545 | } | |
546 | time(&last_report); | |
547 | } | |
548 | ||
9d5da576 | 549 | static void reap(int __attribute__((unused)) sig) { |
e83d0967 | 550 | pid_t cmdpid; |
9d5da576 | 551 | int st; |
552 | ||
553 | do | |
e83d0967 RK |
554 | cmdpid = waitpid(-1, &st, WNOHANG); |
555 | while(cmdpid > 0); | |
9d5da576 | 556 | signal(SIGCHLD, reap); |
557 | } | |
558 | ||
460b9539 | 559 | static int addfd(int fd, int events) { |
560 | if(fdno < NFDS) { | |
561 | fds[fdno].fd = fd; | |
562 | fds[fdno].events = events; | |
563 | return fdno++; | |
564 | } else | |
565 | return -1; | |
566 | } | |
567 | ||
572d74ba | 568 | #if API_ALSA |
569 | /** @brief ALSA backend initialization */ | |
570 | static void alsa_init(void) { | |
571 | info("selected ALSA backend"); | |
572 | } | |
29601377 | 573 | |
5a7c42a8 | 574 | /** @brief Log ALSA parameters */ |
575 | static void log_params(snd_pcm_hw_params_t *hwparams, | |
576 | snd_pcm_sw_params_t *swparams) { | |
577 | snd_pcm_uframes_t f; | |
578 | unsigned u; | |
579 | ||
580 | return; /* too verbose */ | |
581 | if(hwparams) { | |
582 | /* TODO */ | |
583 | } | |
584 | if(swparams) { | |
585 | snd_pcm_sw_params_get_silence_size(swparams, &f); | |
586 | info("sw silence_size=%lu", (unsigned long)f); | |
587 | snd_pcm_sw_params_get_silence_threshold(swparams, &f); | |
588 | info("sw silence_threshold=%lu", (unsigned long)f); | |
589 | snd_pcm_sw_params_get_sleep_min(swparams, &u); | |
590 | info("sw sleep_min=%lu", (unsigned long)u); | |
591 | snd_pcm_sw_params_get_start_threshold(swparams, &f); | |
592 | info("sw start_threshold=%lu", (unsigned long)f); | |
593 | snd_pcm_sw_params_get_stop_threshold(swparams, &f); | |
594 | info("sw stop_threshold=%lu", (unsigned long)f); | |
595 | snd_pcm_sw_params_get_xfer_align(swparams, &f); | |
596 | info("sw xfer_align=%lu", (unsigned long)f); | |
597 | } | |
598 | } | |
599 | ||
600 | /** @brief ALSA deactivation */ | |
601 | static void alsa_deactivate(void) { | |
602 | if(pcm) { | |
603 | int err; | |
604 | ||
605 | if((err = snd_pcm_nonblock(pcm, 0)) < 0) | |
606 | fatal(0, "error calling snd_pcm_nonblock: %d", err); | |
607 | D(("draining pcm")); | |
608 | snd_pcm_drain(pcm); | |
609 | D(("closing pcm")); | |
610 | snd_pcm_close(pcm); | |
611 | pcm = 0; | |
612 | device_state = device_closed; | |
613 | D(("released audio device")); | |
614 | } | |
615 | } | |
616 | ||
29601377 | 617 | /** @brief ALSA backend activation */ |
5a7c42a8 | 618 | static void alsa_activate(void) { |
29601377 | 619 | /* If we need to change format then close the current device. */ |
5a7c42a8 | 620 | if(pcm && !formats_equal(&playing->format, &device_format)) |
621 | alsa_deactivate(); | |
622 | /* Now if the sound device is open it must have the right format */ | |
29601377 | 623 | if(!pcm) { |
624 | snd_pcm_hw_params_t *hwparams; | |
625 | snd_pcm_sw_params_t *swparams; | |
626 | snd_pcm_uframes_t pcm_bufsize; | |
627 | int err; | |
628 | int sample_format = 0; | |
629 | unsigned rate; | |
630 | ||
631 | D(("snd_pcm_open")); | |
632 | if((err = snd_pcm_open(&pcm, | |
633 | config->device, | |
634 | SND_PCM_STREAM_PLAYBACK, | |
635 | SND_PCM_NONBLOCK))) { | |
636 | error(0, "error from snd_pcm_open: %d", err); | |
637 | goto error; | |
638 | } | |
639 | snd_pcm_hw_params_alloca(&hwparams); | |
640 | D(("set up hw params")); | |
641 | if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0) | |
642 | fatal(0, "error from snd_pcm_hw_params_any: %d", err); | |
643 | if((err = snd_pcm_hw_params_set_access(pcm, hwparams, | |
644 | SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) | |
645 | fatal(0, "error from snd_pcm_hw_params_set_access: %d", err); | |
646 | switch(playing->format.bits) { | |
647 | case 8: | |
648 | sample_format = SND_PCM_FORMAT_S8; | |
649 | break; | |
650 | case 16: | |
651 | switch(playing->format.byte_format) { | |
652 | case AO_FMT_NATIVE: sample_format = SND_PCM_FORMAT_S16; break; | |
653 | case AO_FMT_LITTLE: sample_format = SND_PCM_FORMAT_S16_LE; break; | |
654 | case AO_FMT_BIG: sample_format = SND_PCM_FORMAT_S16_BE; break; | |
655 | error(0, "unrecognized byte format %d", playing->format.byte_format); | |
656 | goto fatal; | |
657 | } | |
658 | break; | |
659 | default: | |
660 | error(0, "unsupported sample size %d", playing->format.bits); | |
661 | goto fatal; | |
662 | } | |
663 | if((err = snd_pcm_hw_params_set_format(pcm, hwparams, | |
664 | sample_format)) < 0) { | |
665 | error(0, "error from snd_pcm_hw_params_set_format (%d): %d", | |
666 | sample_format, err); | |
667 | goto fatal; | |
668 | } | |
669 | rate = playing->format.rate; | |
670 | if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0) { | |
671 | error(0, "error from snd_pcm_hw_params_set_rate (%d): %d", | |
672 | playing->format.rate, err); | |
673 | goto fatal; | |
674 | } | |
675 | if(rate != (unsigned)playing->format.rate) | |
676 | info("want rate %d, got %u", playing->format.rate, rate); | |
677 | if((err = snd_pcm_hw_params_set_channels(pcm, hwparams, | |
678 | playing->format.channels)) < 0) { | |
679 | error(0, "error from snd_pcm_hw_params_set_channels (%d): %d", | |
680 | playing->format.channels, err); | |
681 | goto fatal; | |
682 | } | |
5a7c42a8 | 683 | pcm_bufsize = 3 * FRAMES; |
29601377 | 684 | if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams, |
685 | &pcm_bufsize)) < 0) | |
686 | fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d", | |
687 | 3 * FRAMES, err); | |
688 | if(pcm_bufsize != 3 * FRAMES && pcm_bufsize != last_pcm_bufsize) | |
689 | info("asked for PCM buffer of %d frames, got %d", | |
690 | 3 * FRAMES, (int)pcm_bufsize); | |
691 | last_pcm_bufsize = pcm_bufsize; | |
692 | if((err = snd_pcm_hw_params(pcm, hwparams)) < 0) | |
693 | fatal(0, "error calling snd_pcm_hw_params: %d", err); | |
694 | D(("set up sw params")); | |
695 | snd_pcm_sw_params_alloca(&swparams); | |
696 | if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0) | |
697 | fatal(0, "error calling snd_pcm_sw_params_current: %d", err); | |
698 | if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, FRAMES)) < 0) | |
699 | fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d", | |
700 | FRAMES, err); | |
701 | if((err = snd_pcm_sw_params(pcm, swparams)) < 0) | |
702 | fatal(0, "error calling snd_pcm_sw_params: %d", err); | |
5a7c42a8 | 703 | device_format = playing->format; |
29601377 | 704 | D(("acquired audio device")); |
705 | log_params(hwparams, swparams); | |
5a7c42a8 | 706 | device_state = device_open; |
29601377 | 707 | } |
5a7c42a8 | 708 | return; |
29601377 | 709 | fatal: |
710 | abandon(); | |
711 | error: | |
712 | /* We assume the error is temporary and that we'll retry in a bit. */ | |
713 | if(pcm) { | |
714 | snd_pcm_close(pcm); | |
715 | pcm = 0; | |
5a7c42a8 | 716 | device_state = device_error; |
29601377 | 717 | } |
5a7c42a8 | 718 | return; |
29601377 | 719 | } |
b5a99ad0 | 720 | |
7f9d5847 | 721 | /** @brief Play via ALSA */ |
722 | static size_t alsa_play(size_t frames) { | |
544a9ec1 | 723 | snd_pcm_sframes_t pcm_written_frames; |
724 | int err; | |
725 | ||
726 | pcm_written_frames = snd_pcm_writei(pcm, | |
727 | playing->buffer + playing->start, | |
728 | frames); | |
729 | D(("actually play %zu frames, wrote %d", | |
730 | frames, (int)pcm_written_frames)); | |
731 | if(pcm_written_frames < 0) { | |
732 | switch(pcm_written_frames) { | |
733 | case -EPIPE: /* underrun */ | |
734 | error(0, "snd_pcm_writei reports underrun"); | |
735 | if((err = snd_pcm_prepare(pcm)) < 0) | |
736 | fatal(0, "error calling snd_pcm_prepare: %d", err); | |
737 | return 0; | |
738 | case -EAGAIN: | |
739 | return 0; | |
740 | default: | |
741 | fatal(0, "error calling snd_pcm_writei: %d", | |
742 | (int)pcm_written_frames); | |
743 | } | |
744 | } else | |
745 | return pcm_written_frames; | |
7f9d5847 | 746 | } |
747 | ||
6ba5f1ea | 748 | static int alsa_slots, alsa_nslots = -1; |
749 | ||
750 | /** @brief Fill in poll fd array for ALSA */ | |
751 | static void alsa_beforepoll(void) { | |
752 | /* We send sample data to ALSA as fast as it can accept it, relying on | |
753 | * the fact that it has a relatively small buffer to minimize pause | |
754 | * latency. */ | |
755 | int retry = 3, err; | |
756 | ||
757 | alsa_slots = fdno; | |
758 | do { | |
759 | retry = 0; | |
760 | alsa_nslots = snd_pcm_poll_descriptors(pcm, &fds[fdno], NFDS - fdno); | |
761 | if((alsa_nslots <= 0 | |
762 | || !(fds[alsa_slots].events & POLLOUT)) | |
763 | && snd_pcm_state(pcm) == SND_PCM_STATE_XRUN) { | |
764 | error(0, "underrun detected after call to snd_pcm_poll_descriptors()"); | |
765 | if((err = snd_pcm_prepare(pcm))) | |
766 | fatal(0, "error calling snd_pcm_prepare: %d", err); | |
767 | } else | |
768 | break; | |
769 | } while(retry-- > 0); | |
770 | if(alsa_nslots >= 0) | |
771 | fdno += alsa_nslots; | |
772 | } | |
773 | ||
d62d6873 | 774 | /** @brief Process poll() results for ALSA */ |
5a7c42a8 | 775 | static int alsa_ready(void) { |
d62d6873 | 776 | int err; |
777 | ||
5a7c42a8 | 778 | unsigned short alsa_revents; |
779 | ||
780 | if((err = snd_pcm_poll_descriptors_revents(pcm, | |
781 | &fds[alsa_slots], | |
782 | alsa_nslots, | |
783 | &alsa_revents)) < 0) | |
784 | fatal(0, "error calling snd_pcm_poll_descriptors_revents: %d", err); | |
785 | if(alsa_revents & (POLLOUT | POLLERR)) | |
d62d6873 | 786 | return 1; |
5a7c42a8 | 787 | else |
788 | return 0; | |
d62d6873 | 789 | } |
5a7c42a8 | 790 | #endif |
d62d6873 | 791 | |
5a7c42a8 | 792 | /** @brief Start the subprocess for @ref BACKEND_COMMAND */ |
793 | static void fork_cmd(void) { | |
794 | pid_t cmdpid; | |
795 | int pfd[2]; | |
796 | if(cmdfd != -1) close(cmdfd); | |
797 | xpipe(pfd); | |
798 | cmdpid = xfork(); | |
799 | if(!cmdpid) { | |
800 | signal(SIGPIPE, SIG_DFL); | |
801 | xdup2(pfd[0], 0); | |
802 | close(pfd[0]); | |
803 | close(pfd[1]); | |
804 | execl("/bin/sh", "sh", "-c", config->speaker_command, (char *)0); | |
805 | fatal(errno, "error execing /bin/sh"); | |
b5a99ad0 | 806 | } |
5a7c42a8 | 807 | close(pfd[0]); |
808 | cmdfd = pfd[1]; | |
809 | D(("forked cmd %d, fd = %d", cmdpid, cmdfd)); | |
b5a99ad0 | 810 | } |
572d74ba | 811 | |
812 | /** @brief Command backend initialization */ | |
813 | static void command_init(void) { | |
814 | info("selected command backend"); | |
815 | fork_cmd(); | |
816 | } | |
817 | ||
7f9d5847 | 818 | /** @brief Play to a subprocess */ |
819 | static size_t command_play(size_t frames) { | |
3c68b773 | 820 | size_t bytes = frames * bpf; |
821 | int written_bytes; | |
822 | ||
823 | written_bytes = write(cmdfd, playing->buffer + playing->start, bytes); | |
824 | D(("actually play %zu bytes, wrote %d", | |
825 | bytes, written_bytes)); | |
826 | if(written_bytes < 0) { | |
827 | switch(errno) { | |
828 | case EPIPE: | |
829 | error(0, "hmm, command died; trying another"); | |
830 | fork_cmd(); | |
831 | return 0; | |
832 | case EAGAIN: | |
833 | return 0; | |
834 | default: | |
835 | fatal(errno, "error writing to subprocess"); | |
836 | } | |
837 | } else | |
838 | return written_bytes / bpf; | |
7f9d5847 | 839 | } |
840 | ||
6ba5f1ea | 841 | static int cmdfd_slot; |
842 | ||
843 | /** @brief Update poll array for writing to subprocess */ | |
844 | static void command_beforepoll(void) { | |
845 | /* We send sample data to the subprocess as fast as it can accept it. | |
846 | * This isn't ideal as pause latency can be very high as a result. */ | |
847 | if(cmdfd >= 0) | |
848 | cmdfd_slot = addfd(cmdfd, POLLOUT); | |
849 | } | |
850 | ||
d62d6873 | 851 | /** @brief Process poll() results for subprocess play */ |
5a7c42a8 | 852 | static int command_ready(void) { |
853 | if(fds[cmdfd_slot].revents & (POLLOUT | POLLERR)) | |
854 | return 1; | |
855 | else | |
d62d6873 | 856 | return 0; |
29601377 | 857 | } |
858 | ||
572d74ba | 859 | /** @brief Network backend initialization */ |
860 | static void network_init(void) { | |
e83d0967 RK |
861 | struct addrinfo *res, *sres; |
862 | static const struct addrinfo pref = { | |
863 | 0, | |
864 | PF_INET, | |
865 | SOCK_DGRAM, | |
866 | IPPROTO_UDP, | |
867 | 0, | |
868 | 0, | |
869 | 0, | |
870 | 0 | |
871 | }; | |
872 | static const struct addrinfo prefbind = { | |
873 | AI_PASSIVE, | |
874 | PF_INET, | |
875 | SOCK_DGRAM, | |
876 | IPPROTO_UDP, | |
877 | 0, | |
878 | 0, | |
879 | 0, | |
880 | 0 | |
881 | }; | |
882 | static const int one = 1; | |
24d0936b RK |
883 | int sndbuf, target_sndbuf = 131072; |
884 | socklen_t len; | |
e83d0967 | 885 | char *sockname, *ssockname; |
572d74ba | 886 | |
887 | res = get_address(&config->broadcast, &pref, &sockname); | |
888 | if(!res) exit(-1); | |
889 | if(config->broadcast_from.n) { | |
890 | sres = get_address(&config->broadcast_from, &prefbind, &ssockname); | |
891 | if(!sres) exit(-1); | |
892 | } else | |
893 | sres = 0; | |
894 | if((bfd = socket(res->ai_family, | |
895 | res->ai_socktype, | |
896 | res->ai_protocol)) < 0) | |
897 | fatal(errno, "error creating broadcast socket"); | |
898 | if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0) | |
899 | fatal(errno, "error setting SO_BROADCAST on broadcast socket"); | |
900 | len = sizeof sndbuf; | |
901 | if(getsockopt(bfd, SOL_SOCKET, SO_SNDBUF, | |
902 | &sndbuf, &len) < 0) | |
903 | fatal(errno, "error getting SO_SNDBUF"); | |
904 | if(target_sndbuf > sndbuf) { | |
905 | if(setsockopt(bfd, SOL_SOCKET, SO_SNDBUF, | |
906 | &target_sndbuf, sizeof target_sndbuf) < 0) | |
907 | error(errno, "error setting SO_SNDBUF to %d", target_sndbuf); | |
908 | else | |
909 | info("changed socket send buffer size from %d to %d", | |
910 | sndbuf, target_sndbuf); | |
911 | } else | |
912 | info("default socket send buffer is %d", | |
913 | sndbuf); | |
914 | /* We might well want to set additional broadcast- or multicast-related | |
915 | * options here */ | |
916 | if(sres && bind(bfd, sres->ai_addr, sres->ai_addrlen) < 0) | |
917 | fatal(errno, "error binding broadcast socket to %s", ssockname); | |
918 | if(connect(bfd, res->ai_addr, res->ai_addrlen) < 0) | |
919 | fatal(errno, "error connecting broadcast socket to %s", sockname); | |
920 | /* Select an SSRC */ | |
921 | gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM); | |
922 | info("selected network backend, sending to %s", sockname); | |
923 | if(config->sample_format.byte_format != AO_FMT_BIG) { | |
924 | info("forcing big-endian sample format"); | |
925 | config->sample_format.byte_format = AO_FMT_BIG; | |
926 | } | |
927 | } | |
928 | ||
7f9d5847 | 929 | /** @brief Play over the network */ |
930 | static size_t network_play(size_t frames) { | |
3c68b773 | 931 | struct rtp_header header; |
932 | struct iovec vec[2]; | |
933 | size_t bytes = frames * bpf, written_frames; | |
934 | int written_bytes; | |
935 | /* We transmit using RTP (RFC3550) and attempt to conform to the internet | |
936 | * AVT profile (RFC3551). */ | |
937 | ||
938 | if(idled) { | |
939 | /* There may have been a gap. Fix up the RTP time accordingly. */ | |
940 | struct timeval now; | |
941 | uint64_t delta; | |
942 | uint64_t target_rtp_time; | |
943 | ||
944 | /* Find the current time */ | |
945 | xgettimeofday(&now, 0); | |
946 | /* Find the number of microseconds elapsed since rtp_time=0 */ | |
947 | delta = tvsub_us(now, rtp_time_0); | |
948 | assert(delta <= UINT64_MAX / 88200); | |
949 | target_rtp_time = (delta * playing->format.rate | |
950 | * playing->format.channels) / 1000000; | |
951 | /* Overflows at ~6 years uptime with 44100Hz stereo */ | |
952 | ||
953 | /* rtp_time is the number of samples we've played. NB that we play | |
954 | * RTP_AHEAD_MS ahead of ourselves, so it may legitimately be ahead of | |
955 | * the value we deduce from time comparison. | |
956 | * | |
957 | * Suppose we have 1s track started at t=0, and another track begins to | |
958 | * play at t=2s. Suppose RTP_AHEAD_MS=1000 and 44100Hz stereo. In that | |
959 | * case we'll send 1s of audio as fast as we can, giving rtp_time=88200. | |
960 | * rtp_time stops at this point. | |
961 | * | |
962 | * At t=2s we'll have calculated target_rtp_time=176400. In this case we | |
963 | * set rtp_time=176400 and the player can correctly conclude that it | |
964 | * should leave 1s between the tracks. | |
965 | * | |
966 | * Suppose instead that the second track arrives at t=0.5s, and that | |
967 | * we've managed to transmit the whole of the first track already. We'll | |
968 | * have target_rtp_time=44100. | |
969 | * | |
970 | * The desired behaviour is to play the second track back to back with | |
971 | * first. In this case therefore we do not modify rtp_time. | |
972 | * | |
973 | * Is it ever right to reduce rtp_time? No; for that would imply | |
974 | * transmitting packets with overlapping timestamp ranges, which does not | |
975 | * make sense. | |
976 | */ | |
3a23a6a5 | 977 | target_rtp_time &= ~(uint64_t)1; /* stereo! */ |
3c68b773 | 978 | if(target_rtp_time > rtp_time) { |
979 | /* More time has elapsed than we've transmitted samples. That implies | |
980 | * we've been 'sending' silence. */ | |
981 | info("advancing rtp_time by %"PRIu64" samples", | |
982 | target_rtp_time - rtp_time); | |
983 | rtp_time = target_rtp_time; | |
984 | } else if(target_rtp_time < rtp_time) { | |
985 | const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS | |
986 | * config->sample_format.rate | |
987 | * config->sample_format.channels | |
988 | / 1000); | |
989 | ||
990 | if(target_rtp_time + samples_ahead < rtp_time) { | |
991 | info("reversing rtp_time by %"PRIu64" samples", | |
992 | rtp_time - target_rtp_time); | |
993 | } | |
994 | } | |
995 | } | |
996 | header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */ | |
997 | header.seq = htons(rtp_seq++); | |
998 | header.timestamp = htonl((uint32_t)rtp_time); | |
999 | header.ssrc = rtp_id; | |
1000 | header.mpt = (idled ? 0x80 : 0x00) | 10; | |
1001 | /* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from | |
1002 | * the sample rate (in a library somewhere so that configuration.c can rule | |
1003 | * out invalid rates). | |
1004 | */ | |
1005 | idled = 0; | |
1006 | if(bytes > NETWORK_BYTES - sizeof header) { | |
1007 | bytes = NETWORK_BYTES - sizeof header; | |
1008 | /* Always send a whole number of frames */ | |
1009 | bytes -= bytes % bpf; | |
1010 | } | |
1011 | /* "The RTP clock rate used for generating the RTP timestamp is independent | |
1012 | * of the number of channels and the encoding; it equals the number of | |
1013 | * sampling periods per second. For N-channel encodings, each sampling | |
1014 | * period (say, 1/8000 of a second) generates N samples. (This terminology | |
1015 | * is standard, but somewhat confusing, as the total number of samples | |
1016 | * generated per second is then the sampling rate times the channel | |
1017 | * count.)" | |
1018 | */ | |
1019 | vec[0].iov_base = (void *)&header; | |
1020 | vec[0].iov_len = sizeof header; | |
1021 | vec[1].iov_base = playing->buffer + playing->start; | |
1022 | vec[1].iov_len = bytes; | |
1023 | do { | |
1024 | written_bytes = writev(bfd, vec, 2); | |
1025 | } while(written_bytes < 0 && errno == EINTR); | |
1026 | if(written_bytes < 0) { | |
1027 | error(errno, "error transmitting audio data"); | |
1028 | ++audio_errors; | |
1029 | if(audio_errors == 10) | |
1030 | fatal(0, "too many audio errors"); | |
1031 | return 0; | |
1032 | } else | |
1033 | audio_errors /= 2; | |
1034 | written_bytes -= sizeof (struct rtp_header); | |
1035 | written_frames = written_bytes / bpf; | |
1036 | /* Advance RTP's notion of the time */ | |
1037 | rtp_time += written_frames * playing->format.channels; | |
1038 | return written_frames; | |
7f9d5847 | 1039 | } |
1040 | ||
6ba5f1ea | 1041 | static int bfd_slot; |
1042 | ||
1043 | /** @brief Set up poll array for network play */ | |
1044 | static void network_beforepoll(void) { | |
1045 | struct timeval now; | |
1046 | uint64_t target_us; | |
1047 | uint64_t target_rtp_time; | |
1048 | const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS | |
1049 | * config->sample_format.rate | |
1050 | * config->sample_format.channels | |
1051 | / 1000); | |
1052 | ||
1053 | /* If we're starting then initialize the base time */ | |
1054 | if(!rtp_time) | |
1055 | xgettimeofday(&rtp_time_0, 0); | |
1056 | /* We send audio data whenever we get RTP_AHEAD seconds or more | |
1057 | * behind */ | |
1058 | xgettimeofday(&now, 0); | |
1059 | target_us = tvsub_us(now, rtp_time_0); | |
1060 | assert(target_us <= UINT64_MAX / 88200); | |
1061 | target_rtp_time = (target_us * config->sample_format.rate | |
1062 | * config->sample_format.channels) | |
1063 | / 1000000; | |
1064 | if((int64_t)(rtp_time - target_rtp_time) < samples_ahead) | |
1065 | bfd_slot = addfd(bfd, POLLOUT); | |
1066 | } | |
1067 | ||
d62d6873 | 1068 | /** @brief Process poll() results for network play */ |
5a7c42a8 | 1069 | static int network_ready(void) { |
1070 | if(fds[bfd_slot].revents & (POLLOUT | POLLERR)) | |
d62d6873 | 1071 | return 1; |
5a7c42a8 | 1072 | else |
1073 | return 0; | |
d62d6873 | 1074 | } |
1075 | ||
572d74ba | 1076 | /** @brief Table of speaker backends */ |
1077 | static const struct speaker_backend backends[] = { | |
1078 | #if API_ALSA | |
1079 | { | |
1080 | BACKEND_ALSA, | |
0763e1f4 | 1081 | 0, |
29601377 | 1082 | alsa_init, |
b5a99ad0 | 1083 | alsa_activate, |
7f9d5847 | 1084 | alsa_play, |
6ba5f1ea | 1085 | alsa_deactivate, |
d62d6873 | 1086 | alsa_beforepoll, |
5a7c42a8 | 1087 | alsa_ready |
572d74ba | 1088 | }, |
1089 | #endif | |
1090 | { | |
1091 | BACKEND_COMMAND, | |
0763e1f4 | 1092 | FIXED_FORMAT, |
29601377 | 1093 | command_init, |
5a7c42a8 | 1094 | 0, /* activate */ |
7f9d5847 | 1095 | command_play, |
6ba5f1ea | 1096 | 0, /* deactivate */ |
d62d6873 | 1097 | command_beforepoll, |
5a7c42a8 | 1098 | command_ready |
572d74ba | 1099 | }, |
1100 | { | |
1101 | BACKEND_NETWORK, | |
0763e1f4 | 1102 | FIXED_FORMAT, |
29601377 | 1103 | network_init, |
5a7c42a8 | 1104 | 0, /* activate */ |
7f9d5847 | 1105 | network_play, |
6ba5f1ea | 1106 | 0, /* deactivate */ |
d62d6873 | 1107 | network_beforepoll, |
5a7c42a8 | 1108 | network_ready |
572d74ba | 1109 | }, |
d62d6873 | 1110 | { -1, 0, 0, 0, 0, 0, 0, 0 } /* end of list */ |
572d74ba | 1111 | }; |
1112 | ||
5a7c42a8 | 1113 | /** @brief Return nonzero if we want to play some audio |
55f35f2d | 1114 | * |
5a7c42a8 | 1115 | * We want to play audio if there is a current track; and it is not paused; and |
1116 | * there are at least @ref FRAMES frames of audio to play, or we are in sight | |
1117 | * of the end of the current track. | |
55f35f2d | 1118 | */ |
5a7c42a8 | 1119 | static int playable(void) { |
1120 | return playing | |
1121 | && !paused | |
1122 | && (playing->used >= FRAMES || playing->eof); | |
1123 | } | |
1124 | ||
1125 | /** @brief Main event loop */ | |
55f35f2d | 1126 | static void mainloop(void) { |
572d74ba | 1127 | struct track *t; |
1128 | struct speaker_message sm; | |
5a7c42a8 | 1129 | int n, fd, stdin_slot, timeout; |
460b9539 | 1130 | |
460b9539 | 1131 | while(getppid() != 1) { |
1132 | fdno = 0; | |
5a7c42a8 | 1133 | /* By default we will wait up to a second before thinking about current |
1134 | * state. */ | |
1135 | timeout = 1000; | |
460b9539 | 1136 | /* Always ready for commands from the main server. */ |
1137 | stdin_slot = addfd(0, POLLIN); | |
1138 | /* Try to read sample data for the currently playing track if there is | |
1139 | * buffer space. */ | |
5a7c42a8 | 1140 | if(playing && !playing->eof && playing->used < playing->size) |
460b9539 | 1141 | playing->slot = addfd(playing->fd, POLLIN); |
5a7c42a8 | 1142 | else if(playing) |
460b9539 | 1143 | playing->slot = -1; |
5a7c42a8 | 1144 | if(playable()) { |
1145 | /* We want to play some audio. If the device is closed then we attempt | |
1146 | * to open it. */ | |
1147 | if(device_state == device_closed) | |
1148 | activate(); | |
1149 | /* If the device is (now) open then we will wait up until it is ready for | |
1150 | * more. If something went wrong then we should have device_error | |
1151 | * instead, but the post-poll code will cope even if it's | |
1152 | * device_closed. */ | |
1153 | if(device_state == device_open) | |
1154 | backend->beforepoll(); | |
1155 | } | |
460b9539 | 1156 | /* If any other tracks don't have a full buffer, try to read sample data |
5a7c42a8 | 1157 | * from them. We do this last of all, so that if we run out of slots, |
1158 | * nothing important can't be monitored. */ | |
460b9539 | 1159 | for(t = tracks; t; t = t->next) |
1160 | if(t != playing) { | |
1161 | if(!t->eof && t->used < t->size) { | |
9d5da576 | 1162 | t->slot = addfd(t->fd, POLLIN | POLLHUP); |
460b9539 | 1163 | } else |
1164 | t->slot = -1; | |
1165 | } | |
e83d0967 RK |
1166 | /* Wait for something interesting to happen */ |
1167 | n = poll(fds, fdno, timeout); | |
460b9539 | 1168 | if(n < 0) { |
1169 | if(errno == EINTR) continue; | |
1170 | fatal(errno, "error calling poll"); | |
1171 | } | |
1172 | /* Play some sound before doing anything else */ | |
5a7c42a8 | 1173 | if(playable()) { |
1174 | /* We want to play some audio */ | |
1175 | if(device_state == device_open) { | |
1176 | if(backend->ready()) | |
1177 | play(3 * FRAMES); | |
1178 | } else { | |
1179 | /* We must be in _closed or _error, and it should be the latter, but we | |
1180 | * cope with either. | |
1181 | * | |
1182 | * We most likely timed out, so now is a good time to retry. play() | |
1183 | * knows to re-activate the device if necessary. | |
1184 | */ | |
1185 | play(3 * FRAMES); | |
1186 | } | |
460b9539 | 1187 | } |
1188 | /* Perhaps we have a command to process */ | |
1189 | if(fds[stdin_slot].revents & POLLIN) { | |
5a7c42a8 | 1190 | /* There might (in theory) be several commands queued up, but in general |
1191 | * this won't be the case, so we don't bother looping around to pick them | |
1192 | * all up. */ | |
460b9539 | 1193 | n = speaker_recv(0, &sm, &fd); |
1194 | if(n > 0) | |
1195 | switch(sm.type) { | |
1196 | case SM_PREPARE: | |
1197 | D(("SM_PREPARE %s %d", sm.id, fd)); | |
1198 | if(fd == -1) fatal(0, "got SM_PREPARE but no file descriptor"); | |
1199 | t = findtrack(sm.id, 1); | |
1200 | acquire(t, fd); | |
1201 | break; | |
1202 | case SM_PLAY: | |
1203 | D(("SM_PLAY %s %d", sm.id, fd)); | |
1204 | if(playing) fatal(0, "got SM_PLAY but already playing something"); | |
1205 | t = findtrack(sm.id, 1); | |
1206 | if(fd != -1) acquire(t, fd); | |
1207 | playing = t; | |
5a7c42a8 | 1208 | /* We attempt to play straight away rather than going round the loop. |
1209 | * play() is clever enough to perform any activation that is | |
1210 | * required. */ | |
1211 | play(3 * FRAMES); | |
460b9539 | 1212 | report(); |
1213 | break; | |
1214 | case SM_PAUSE: | |
1215 | D(("SM_PAUSE")); | |
1216 | paused = 1; | |
1217 | report(); | |
1218 | break; | |
1219 | case SM_RESUME: | |
1220 | D(("SM_RESUME")); | |
1221 | if(paused) { | |
1222 | paused = 0; | |
5a7c42a8 | 1223 | /* As for SM_PLAY we attempt to play straight away. */ |
460b9539 | 1224 | if(playing) |
5a7c42a8 | 1225 | play(3 * FRAMES); |
460b9539 | 1226 | } |
1227 | report(); | |
1228 | break; | |
1229 | case SM_CANCEL: | |
1230 | D(("SM_CANCEL %s", sm.id)); | |
1231 | t = removetrack(sm.id); | |
1232 | if(t) { | |
1233 | if(t == playing) { | |
1234 | sm.type = SM_FINISHED; | |
1235 | strcpy(sm.id, playing->id); | |
1236 | speaker_send(1, &sm, 0); | |
1237 | playing = 0; | |
1238 | } | |
1239 | destroy(t); | |
1240 | } else | |
1241 | error(0, "SM_CANCEL for unknown track %s", sm.id); | |
1242 | report(); | |
1243 | break; | |
1244 | case SM_RELOAD: | |
1245 | D(("SM_RELOAD")); | |
1246 | if(config_read()) error(0, "cannot read configuration"); | |
1247 | info("reloaded configuration"); | |
1248 | break; | |
1249 | default: | |
1250 | error(0, "unknown message type %d", sm.type); | |
1251 | } | |
1252 | } | |
1253 | /* Read in any buffered data */ | |
1254 | for(t = tracks; t; t = t->next) | |
9d5da576 | 1255 | if(t->slot != -1 && (fds[t->slot].revents & (POLLIN | POLLHUP))) |
460b9539 | 1256 | fill(t); |
460b9539 | 1257 | /* Maybe we finished playing a track somewhere in the above */ |
1258 | maybe_finished(); | |
1259 | /* If we don't need the sound device for now then close it for the benefit | |
1260 | * of anyone else who wants it. */ | |
5a7c42a8 | 1261 | if((!playing || paused) && device_state == device_open) |
460b9539 | 1262 | idle(); |
1263 | /* If we've not reported out state for a second do so now. */ | |
1264 | if(time(0) > last_report) | |
1265 | report(); | |
1266 | } | |
55f35f2d | 1267 | } |
1268 | ||
1269 | int main(int argc, char **argv) { | |
1270 | int n; | |
1271 | ||
1272 | set_progname(argv); | |
1273 | if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); | |
1274 | while((n = getopt_long(argc, argv, "hVc:dD", options, 0)) >= 0) { | |
1275 | switch(n) { | |
1276 | case 'h': help(); | |
1277 | case 'V': version(); | |
1278 | case 'c': configfile = optarg; break; | |
1279 | case 'd': debugging = 1; break; | |
1280 | case 'D': debugging = 0; break; | |
1281 | default: fatal(0, "invalid option"); | |
1282 | } | |
1283 | } | |
1284 | if(getenv("DISORDER_DEBUG_SPEAKER")) debugging = 1; | |
1285 | /* If stderr is a TTY then log there, otherwise to syslog. */ | |
1286 | if(!isatty(2)) { | |
1287 | openlog(progname, LOG_PID, LOG_DAEMON); | |
1288 | log_default = &log_syslog; | |
1289 | } | |
1290 | if(config_read()) fatal(0, "cannot read configuration"); | |
1291 | /* ignore SIGPIPE */ | |
1292 | signal(SIGPIPE, SIG_IGN); | |
1293 | /* reap kids */ | |
1294 | signal(SIGCHLD, reap); | |
1295 | /* set nice value */ | |
1296 | xnice(config->nice_speaker); | |
1297 | /* change user */ | |
1298 | become_mortal(); | |
1299 | /* make sure we're not root, whatever the config says */ | |
1300 | if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root"); | |
1301 | /* identify the backend used to play */ | |
1302 | for(n = 0; backends[n].backend != -1; ++n) | |
1303 | if(backends[n].backend == config->speaker_backend) | |
1304 | break; | |
1305 | if(backends[n].backend == -1) | |
1306 | fatal(0, "unsupported backend %d", config->speaker_backend); | |
1307 | backend = &backends[n]; | |
1308 | /* backend-specific initialization */ | |
1309 | backend->init(); | |
1310 | mainloop(); | |
460b9539 | 1311 | info("stopped (parent terminated)"); |
1312 | exit(0); | |
1313 | } | |
1314 | ||
1315 | /* | |
1316 | Local Variables: | |
1317 | c-basic-offset:2 | |
1318 | comment-column:40 | |
1319 | fill-column:79 | |
1320 | indent-tabs-mode:nil | |
1321 | End: | |
1322 | */ |