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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 |
21 | * @brief Speaker processs | |
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 | * | |
28 | * For the <a href="http://www.alsa-project.org/">ALSA</a> API, 8- and 16- bit | |
29 | * stereo and mono are supported, with any sample rate (within the limits that | |
30 | * ALSA can deal with.) | |
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 | * | |
42 | * This program deliberately does not use the garbage collector even though it | |
43 | * might be convenient to do so. This is for two reasons. Firstly some sound | |
44 | * APIs use thread threads and we do not want to have to deal with potential | |
45 | * interactions between threading and garbage collection. Secondly this | |
46 | * process needs to be able to respond quickly and this is not compatible with | |
47 | * the collector hanging the program even relatively briefly. | |
48 | */ | |
460b9539 | 49 | |
50 | #include <config.h> | |
51 | #include "types.h" | |
52 | ||
53 | #include <getopt.h> | |
54 | #include <stdio.h> | |
55 | #include <stdlib.h> | |
56 | #include <locale.h> | |
57 | #include <syslog.h> | |
58 | #include <unistd.h> | |
59 | #include <errno.h> | |
60 | #include <ao/ao.h> | |
61 | #include <string.h> | |
62 | #include <assert.h> | |
63 | #include <sys/select.h> | |
9d5da576 | 64 | #include <sys/wait.h> |
460b9539 | 65 | #include <time.h> |
8023f60b | 66 | #include <fcntl.h> |
67 | #include <poll.h> | |
e83d0967 RK |
68 | #include <sys/socket.h> |
69 | #include <netdb.h> | |
70 | #include <gcrypt.h> | |
71 | #include <sys/uio.h> | |
460b9539 | 72 | |
73 | #include "configuration.h" | |
74 | #include "syscalls.h" | |
75 | #include "log.h" | |
76 | #include "defs.h" | |
77 | #include "mem.h" | |
78 | #include "speaker.h" | |
79 | #include "user.h" | |
e83d0967 RK |
80 | #include "addr.h" |
81 | #include "timeval.h" | |
82 | #include "rtp.h" | |
460b9539 | 83 | |
8023f60b | 84 | #if API_ALSA |
dea8f8aa | 85 | #include <alsa/asoundlib.h> |
8023f60b | 86 | #endif |
dea8f8aa | 87 | |
5330d674 | 88 | #ifdef WORDS_BIGENDIAN |
89 | # define MACHINE_AO_FMT AO_FMT_BIG | |
90 | #else | |
91 | # define MACHINE_AO_FMT AO_FMT_LITTLE | |
92 | #endif | |
93 | ||
1674096e | 94 | /** @brief How many seconds of input to buffer |
95 | * | |
96 | * While any given connection has this much audio buffered, no more reads will | |
97 | * be issued for that connection. The decoder will have to wait. | |
98 | */ | |
99 | #define BUFFER_SECONDS 5 | |
460b9539 | 100 | |
101 | #define FRAMES 4096 /* Frame batch size */ | |
102 | ||
1674096e | 103 | /** @brief Bytes to send per network packet |
104 | * | |
105 | * Don't make this too big or arithmetic will start to overflow. | |
106 | */ | |
8d2482ec | 107 | #define NETWORK_BYTES (1024+sizeof(struct rtp_header)) |
e83d0967 | 108 | |
508acf7a RK |
109 | /** @brief Maximum RTP playahead (ms) */ |
110 | #define RTP_AHEAD_MS 1000 | |
e83d0967 | 111 | |
1674096e | 112 | /** @brief Maximum number of FDs to poll for */ |
113 | #define NFDS 256 | |
460b9539 | 114 | |
1674096e | 115 | /** @brief Track structure |
116 | * | |
117 | * Known tracks are kept in a linked list. Usually there will be at most two | |
118 | * of these but rearranging the queue can cause there to be more. | |
119 | */ | |
460b9539 | 120 | static struct track { |
121 | struct track *next; /* next track */ | |
122 | int fd; /* input FD */ | |
123 | char id[24]; /* ID */ | |
124 | size_t start, used; /* start + bytes used */ | |
125 | int eof; /* input is at EOF */ | |
126 | int got_format; /* got format yet? */ | |
127 | ao_sample_format format; /* sample format */ | |
128 | unsigned long long played; /* number of frames played */ | |
129 | char *buffer; /* sample buffer */ | |
130 | size_t size; /* sample buffer size */ | |
131 | int slot; /* poll array slot */ | |
132 | } *tracks, *playing; /* all tracks + playing track */ | |
133 | ||
134 | static time_t last_report; /* when we last reported */ | |
135 | static int paused; /* pause status */ | |
460b9539 | 136 | static ao_sample_format pcm_format; /* current format if aodev != 0 */ |
137 | static size_t bpf; /* bytes per frame */ | |
138 | static struct pollfd fds[NFDS]; /* if we need more than that */ | |
139 | static int fdno; /* fd number */ | |
8023f60b | 140 | static size_t bufsize; /* buffer size */ |
141 | #if API_ALSA | |
142 | static snd_pcm_t *pcm; /* current pcm handle */ | |
0c207c37 | 143 | static snd_pcm_uframes_t last_pcm_bufsize; /* last seen buffer size */ |
8023f60b | 144 | #endif |
9d5da576 | 145 | static int ready; /* ready to send audio */ |
460b9539 | 146 | static int forceplay; /* frames to force play */ |
e83d0967 RK |
147 | static int cmdfd = -1; /* child process input */ |
148 | static int bfd = -1; /* broadcast FD */ | |
7aa087a7 RK |
149 | |
150 | /** @brief RTP timestamp | |
151 | * | |
152 | * This counts the number of samples played (NB not the number of frames | |
153 | * played). | |
154 | * | |
155 | * The timestamp in the packet header is only 32 bits wide. With 44100Hz | |
156 | * stereo, that only gives about half a day before wrapping, which is not | |
157 | * particularly convenient for certain debugging purposes. Therefore the | |
158 | * timestamp is maintained as a 64-bit integer, giving around six million years | |
159 | * before wrapping, and truncated to 32 bits when transmitting. | |
160 | */ | |
161 | static uint64_t rtp_time; | |
162 | ||
163 | /** @brief RTP base timestamp | |
164 | * | |
165 | * This is the real time correspoding to an @ref rtp_time of 0. It is used | |
166 | * to recalculate the timestamp after idle periods. | |
167 | */ | |
168 | static struct timeval rtp_time_0; | |
169 | ||
e83d0967 RK |
170 | static uint16_t rtp_seq; /* frame sequence number */ |
171 | static uint32_t rtp_id; /* RTP SSRC */ | |
172 | static int idled; /* set when idled */ | |
173 | static int audio_errors; /* audio error counter */ | |
460b9539 | 174 | |
29601377 | 175 | /** @brief Structure of a backend */ |
176 | struct speaker_backend { | |
177 | /** @brief Which backend this is | |
178 | * | |
179 | * @c -1 terminates the list. | |
180 | */ | |
181 | int backend; | |
182 | ||
183 | /** @brief Initialization | |
184 | * | |
185 | * Called once at startup. | |
186 | */ | |
187 | void (*init)(void); | |
188 | ||
189 | /** @brief Activation | |
190 | * @return 0 on success, non-0 on error | |
191 | * | |
192 | * Called to activate the output device. | |
193 | */ | |
194 | int (*activate)(void); | |
195 | }; | |
196 | ||
197 | /** @brief Selected backend */ | |
198 | static const struct speaker_backend *backend; | |
199 | ||
460b9539 | 200 | static const struct option options[] = { |
201 | { "help", no_argument, 0, 'h' }, | |
202 | { "version", no_argument, 0, 'V' }, | |
203 | { "config", required_argument, 0, 'c' }, | |
204 | { "debug", no_argument, 0, 'd' }, | |
205 | { "no-debug", no_argument, 0, 'D' }, | |
206 | { 0, 0, 0, 0 } | |
207 | }; | |
208 | ||
209 | /* Display usage message and terminate. */ | |
210 | static void help(void) { | |
211 | xprintf("Usage:\n" | |
212 | " disorder-speaker [OPTIONS]\n" | |
213 | "Options:\n" | |
214 | " --help, -h Display usage message\n" | |
215 | " --version, -V Display version number\n" | |
216 | " --config PATH, -c PATH Set configuration file\n" | |
217 | " --debug, -d Turn on debugging\n" | |
218 | "\n" | |
219 | "Speaker process for DisOrder. Not intended to be run\n" | |
220 | "directly.\n"); | |
221 | xfclose(stdout); | |
222 | exit(0); | |
223 | } | |
224 | ||
225 | /* Display version number and terminate. */ | |
226 | static void version(void) { | |
227 | xprintf("disorder-speaker version %s\n", disorder_version_string); | |
228 | xfclose(stdout); | |
229 | exit(0); | |
230 | } | |
231 | ||
1674096e | 232 | /** @brief Return the number of bytes per frame in @p format */ |
460b9539 | 233 | static size_t bytes_per_frame(const ao_sample_format *format) { |
234 | return format->channels * format->bits / 8; | |
235 | } | |
236 | ||
1674096e | 237 | /** @brief Find track @p id, maybe creating it if not found */ |
460b9539 | 238 | static struct track *findtrack(const char *id, int create) { |
239 | struct track *t; | |
240 | ||
241 | D(("findtrack %s %d", id, create)); | |
242 | for(t = tracks; t && strcmp(id, t->id); t = t->next) | |
243 | ; | |
244 | if(!t && create) { | |
245 | t = xmalloc(sizeof *t); | |
246 | t->next = tracks; | |
247 | strcpy(t->id, id); | |
248 | t->fd = -1; | |
249 | tracks = t; | |
250 | /* The initial input buffer will be the sample format. */ | |
251 | t->buffer = (void *)&t->format; | |
252 | t->size = sizeof t->format; | |
253 | } | |
254 | return t; | |
255 | } | |
256 | ||
1674096e | 257 | /** @brief Remove track @p id (but do not destroy it) */ |
460b9539 | 258 | static struct track *removetrack(const char *id) { |
259 | struct track *t, **tt; | |
260 | ||
261 | D(("removetrack %s", id)); | |
262 | for(tt = &tracks; (t = *tt) && strcmp(id, t->id); tt = &t->next) | |
263 | ; | |
264 | if(t) | |
265 | *tt = t->next; | |
266 | return t; | |
267 | } | |
268 | ||
1674096e | 269 | /** @brief Destroy a track */ |
460b9539 | 270 | static void destroy(struct track *t) { |
271 | D(("destroy %s", t->id)); | |
272 | if(t->fd != -1) xclose(t->fd); | |
273 | if(t->buffer != (void *)&t->format) free(t->buffer); | |
274 | free(t); | |
275 | } | |
276 | ||
1674096e | 277 | /** @brief Notice a new connection */ |
460b9539 | 278 | static void acquire(struct track *t, int fd) { |
279 | D(("acquire %s %d", t->id, fd)); | |
280 | if(t->fd != -1) | |
281 | xclose(t->fd); | |
282 | t->fd = fd; | |
283 | nonblock(fd); | |
284 | } | |
285 | ||
1674096e | 286 | /** @brief Return true if A and B denote identical libao formats, else false */ |
287 | static int formats_equal(const ao_sample_format *a, | |
288 | const ao_sample_format *b) { | |
289 | return (a->bits == b->bits | |
290 | && a->rate == b->rate | |
291 | && a->channels == b->channels | |
292 | && a->byte_format == b->byte_format); | |
293 | } | |
294 | ||
295 | /** @brief Compute arguments to sox */ | |
296 | static void soxargs(const char ***pp, char **qq, ao_sample_format *ao) { | |
297 | int n; | |
298 | ||
299 | *(*pp)++ = "-t.raw"; | |
300 | *(*pp)++ = "-s"; | |
301 | *(*pp)++ = *qq; n = sprintf(*qq, "-r%d", ao->rate); *qq += n + 1; | |
302 | *(*pp)++ = *qq; n = sprintf(*qq, "-c%d", ao->channels); *qq += n + 1; | |
303 | /* sox 12.17.9 insists on -b etc; CVS sox insists on -<n> etc; both are | |
304 | * deployed! */ | |
305 | switch(config->sox_generation) { | |
306 | case 0: | |
307 | if(ao->bits != 8 | |
308 | && ao->byte_format != AO_FMT_NATIVE | |
309 | && ao->byte_format != MACHINE_AO_FMT) { | |
310 | *(*pp)++ = "-x"; | |
311 | } | |
312 | switch(ao->bits) { | |
313 | case 8: *(*pp)++ = "-b"; break; | |
314 | case 16: *(*pp)++ = "-w"; break; | |
315 | case 32: *(*pp)++ = "-l"; break; | |
316 | case 64: *(*pp)++ = "-d"; break; | |
317 | default: fatal(0, "cannot handle sample size %d", (int)ao->bits); | |
318 | } | |
319 | break; | |
320 | case 1: | |
321 | switch(ao->byte_format) { | |
322 | case AO_FMT_NATIVE: break; | |
323 | case AO_FMT_BIG: *(*pp)++ = "-B"; break; | |
324 | case AO_FMT_LITTLE: *(*pp)++ = "-L"; break; | |
325 | } | |
326 | *(*pp)++ = *qq; n = sprintf(*qq, "-%d", ao->bits/8); *qq += n + 1; | |
327 | break; | |
328 | } | |
329 | } | |
330 | ||
331 | /** @brief Enable format translation | |
332 | * | |
333 | * If necessary, replaces a tracks inbound file descriptor with one connected | |
334 | * to a sox invocation, which performs the required translation. | |
335 | */ | |
336 | static void enable_translation(struct track *t) { | |
337 | switch(config->speaker_backend) { | |
338 | case BACKEND_COMMAND: | |
339 | case BACKEND_NETWORK: | |
340 | /* These backends need a specific sample format */ | |
341 | break; | |
342 | case BACKEND_ALSA: | |
343 | /* ALSA can cope */ | |
344 | return; | |
345 | } | |
346 | if(!formats_equal(&t->format, &config->sample_format)) { | |
347 | char argbuf[1024], *q = argbuf; | |
348 | const char *av[18], **pp = av; | |
349 | int soxpipe[2]; | |
350 | pid_t soxkid; | |
351 | ||
352 | *pp++ = "sox"; | |
353 | soxargs(&pp, &q, &t->format); | |
354 | *pp++ = "-"; | |
355 | soxargs(&pp, &q, &config->sample_format); | |
356 | *pp++ = "-"; | |
357 | *pp++ = 0; | |
358 | if(debugging) { | |
359 | for(pp = av; *pp; pp++) | |
360 | D(("sox arg[%d] = %s", pp - av, *pp)); | |
361 | D(("end args")); | |
362 | } | |
363 | xpipe(soxpipe); | |
364 | soxkid = xfork(); | |
365 | if(soxkid == 0) { | |
366 | signal(SIGPIPE, SIG_DFL); | |
367 | xdup2(t->fd, 0); | |
368 | xdup2(soxpipe[1], 1); | |
369 | fcntl(0, F_SETFL, fcntl(0, F_GETFL) & ~O_NONBLOCK); | |
370 | close(soxpipe[0]); | |
371 | close(soxpipe[1]); | |
372 | close(t->fd); | |
373 | execvp("sox", (char **)av); | |
374 | _exit(1); | |
375 | } | |
376 | D(("forking sox for format conversion (kid = %d)", soxkid)); | |
377 | close(t->fd); | |
378 | close(soxpipe[1]); | |
379 | t->fd = soxpipe[0]; | |
380 | t->format = config->sample_format; | |
1674096e | 381 | } |
382 | } | |
383 | ||
384 | /** @brief Read data into a sample buffer | |
385 | * @param t Pointer to track | |
386 | * @return 0 on success, -1 on EOF | |
387 | * | |
388 | * This is effectively the read callback on @c t->fd. | |
389 | */ | |
460b9539 | 390 | static int fill(struct track *t) { |
391 | size_t where, left; | |
392 | int n; | |
393 | ||
394 | D(("fill %s: eof=%d used=%zu size=%zu got_format=%d", | |
395 | t->id, t->eof, t->used, t->size, t->got_format)); | |
396 | if(t->eof) return -1; | |
397 | if(t->used < t->size) { | |
398 | /* there is room left in the buffer */ | |
399 | where = (t->start + t->used) % t->size; | |
400 | if(t->got_format) { | |
401 | /* We are reading audio data, get as much as we can */ | |
402 | if(where >= t->start) left = t->size - where; | |
403 | else left = t->start - where; | |
404 | } else | |
405 | /* We are still waiting for the format, only get that */ | |
406 | left = sizeof (ao_sample_format) - t->used; | |
407 | do { | |
408 | n = read(t->fd, t->buffer + where, left); | |
409 | } while(n < 0 && errno == EINTR); | |
410 | if(n < 0) { | |
411 | if(errno != EAGAIN) fatal(errno, "error reading sample stream"); | |
412 | return 0; | |
413 | } | |
414 | if(n == 0) { | |
415 | D(("fill %s: eof detected", t->id)); | |
416 | t->eof = 1; | |
417 | return -1; | |
418 | } | |
419 | t->used += n; | |
420 | if(!t->got_format && t->used >= sizeof (ao_sample_format)) { | |
421 | assert(t->used == sizeof (ao_sample_format)); | |
422 | /* Check that our assumptions are met. */ | |
423 | if(t->format.bits & 7) | |
424 | fatal(0, "bits per sample not a multiple of 8"); | |
1674096e | 425 | /* If the input format is unsuitable, arrange to translate it */ |
426 | enable_translation(t); | |
460b9539 | 427 | /* Make a new buffer for audio data. */ |
428 | t->size = bytes_per_frame(&t->format) * t->format.rate * BUFFER_SECONDS; | |
429 | t->buffer = xmalloc(t->size); | |
430 | t->used = 0; | |
431 | t->got_format = 1; | |
432 | D(("got format for %s", t->id)); | |
433 | } | |
434 | } | |
435 | return 0; | |
436 | } | |
437 | ||
1674096e | 438 | /** @brief Close the sound device */ |
460b9539 | 439 | static void idle(void) { |
460b9539 | 440 | D(("idle")); |
8023f60b | 441 | #if API_ALSA |
e83d0967 | 442 | if(config->speaker_backend == BACKEND_ALSA && pcm) { |
8023f60b | 443 | int err; |
444 | ||
460b9539 | 445 | if((err = snd_pcm_nonblock(pcm, 0)) < 0) |
446 | fatal(0, "error calling snd_pcm_nonblock: %d", err); | |
447 | D(("draining pcm")); | |
448 | snd_pcm_drain(pcm); | |
449 | D(("closing pcm")); | |
450 | snd_pcm_close(pcm); | |
451 | pcm = 0; | |
452 | forceplay = 0; | |
453 | D(("released audio device")); | |
454 | } | |
8023f60b | 455 | #endif |
e83d0967 | 456 | idled = 1; |
9d5da576 | 457 | ready = 0; |
460b9539 | 458 | } |
459 | ||
1674096e | 460 | /** @brief Abandon the current track */ |
460b9539 | 461 | static void abandon(void) { |
462 | struct speaker_message sm; | |
463 | ||
464 | D(("abandon")); | |
465 | memset(&sm, 0, sizeof sm); | |
466 | sm.type = SM_FINISHED; | |
467 | strcpy(sm.id, playing->id); | |
468 | speaker_send(1, &sm, 0); | |
469 | removetrack(playing->id); | |
470 | destroy(playing); | |
471 | playing = 0; | |
472 | forceplay = 0; | |
473 | } | |
474 | ||
8023f60b | 475 | #if API_ALSA |
1674096e | 476 | /** @brief Log ALSA parameters */ |
1c6e6a61 | 477 | static void log_params(snd_pcm_hw_params_t *hwparams, |
478 | snd_pcm_sw_params_t *swparams) { | |
479 | snd_pcm_uframes_t f; | |
480 | unsigned u; | |
481 | ||
0c207c37 | 482 | return; /* too verbose */ |
1c6e6a61 | 483 | if(hwparams) { |
484 | /* TODO */ | |
485 | } | |
486 | if(swparams) { | |
487 | snd_pcm_sw_params_get_silence_size(swparams, &f); | |
488 | info("sw silence_size=%lu", (unsigned long)f); | |
489 | snd_pcm_sw_params_get_silence_threshold(swparams, &f); | |
490 | info("sw silence_threshold=%lu", (unsigned long)f); | |
491 | snd_pcm_sw_params_get_sleep_min(swparams, &u); | |
492 | info("sw sleep_min=%lu", (unsigned long)u); | |
493 | snd_pcm_sw_params_get_start_threshold(swparams, &f); | |
494 | info("sw start_threshold=%lu", (unsigned long)f); | |
495 | snd_pcm_sw_params_get_stop_threshold(swparams, &f); | |
496 | info("sw stop_threshold=%lu", (unsigned long)f); | |
497 | snd_pcm_sw_params_get_xfer_align(swparams, &f); | |
498 | info("sw xfer_align=%lu", (unsigned long)f); | |
499 | } | |
500 | } | |
8023f60b | 501 | #endif |
1c6e6a61 | 502 | |
1674096e | 503 | /** @brief Enable sound output |
504 | * | |
505 | * Makes sure the sound device is open and has the right sample format. Return | |
506 | * 0 on success and -1 on error. | |
507 | */ | |
460b9539 | 508 | static int activate(void) { |
460b9539 | 509 | /* If we don't know the format yet we cannot start. */ |
510 | if(!playing->got_format) { | |
511 | D((" - not got format for %s", playing->id)); | |
512 | return -1; | |
513 | } | |
29601377 | 514 | return backend->activate(); |
460b9539 | 515 | } |
516 | ||
517 | /* Check to see whether the current track has finished playing */ | |
518 | static void maybe_finished(void) { | |
519 | if(playing | |
520 | && playing->eof | |
521 | && (!playing->got_format | |
522 | || playing->used < bytes_per_frame(&playing->format))) | |
523 | abandon(); | |
524 | } | |
525 | ||
e83d0967 RK |
526 | static void fork_cmd(void) { |
527 | pid_t cmdpid; | |
9d5da576 | 528 | int pfd[2]; |
e83d0967 | 529 | if(cmdfd != -1) close(cmdfd); |
9d5da576 | 530 | xpipe(pfd); |
e83d0967 RK |
531 | cmdpid = xfork(); |
532 | if(!cmdpid) { | |
1674096e | 533 | signal(SIGPIPE, SIG_DFL); |
9d5da576 | 534 | xdup2(pfd[0], 0); |
535 | close(pfd[0]); | |
536 | close(pfd[1]); | |
537 | execl("/bin/sh", "sh", "-c", config->speaker_command, (char *)0); | |
538 | fatal(errno, "error execing /bin/sh"); | |
539 | } | |
540 | close(pfd[0]); | |
e83d0967 RK |
541 | cmdfd = pfd[1]; |
542 | D(("forked cmd %d, fd = %d", cmdpid, cmdfd)); | |
9d5da576 | 543 | } |
544 | ||
460b9539 | 545 | static void play(size_t frames) { |
ceb044f4 | 546 | size_t avail_bytes, write_bytes, written_frames; |
9d5da576 | 547 | ssize_t written_bytes; |
0b75463f | 548 | struct rtp_header header; |
e83d0967 | 549 | struct iovec vec[2]; |
460b9539 | 550 | |
551 | if(activate()) { | |
552 | if(playing) | |
553 | forceplay = frames; | |
554 | else | |
555 | forceplay = 0; /* Must have called abandon() */ | |
556 | return; | |
557 | } | |
558 | D(("play: play %zu/%zu%s %dHz %db %dc", frames, playing->used / bpf, | |
559 | playing->eof ? " EOF" : "", | |
560 | playing->format.rate, | |
561 | playing->format.bits, | |
562 | playing->format.channels)); | |
563 | /* If we haven't got enough bytes yet wait until we have. Exception: when | |
564 | * we are at eof. */ | |
565 | if(playing->used < frames * bpf && !playing->eof) { | |
566 | forceplay = frames; | |
567 | return; | |
568 | } | |
569 | /* We have got enough data so don't force play again */ | |
570 | forceplay = 0; | |
571 | /* Figure out how many frames there are available to write */ | |
572 | if(playing->start + playing->used > playing->size) | |
573 | avail_bytes = playing->size - playing->start; | |
574 | else | |
575 | avail_bytes = playing->used; | |
9d5da576 | 576 | |
e83d0967 | 577 | switch(config->speaker_backend) { |
8023f60b | 578 | #if API_ALSA |
3a3c7bb9 | 579 | case BACKEND_ALSA: { |
8023f60b | 580 | snd_pcm_sframes_t pcm_written_frames; |
581 | size_t avail_frames; | |
582 | int err; | |
583 | ||
9d5da576 | 584 | avail_frames = avail_bytes / bpf; |
585 | if(avail_frames > frames) | |
586 | avail_frames = frames; | |
587 | if(!avail_frames) | |
460b9539 | 588 | return; |
8023f60b | 589 | pcm_written_frames = snd_pcm_writei(pcm, |
590 | playing->buffer + playing->start, | |
591 | avail_frames); | |
9d5da576 | 592 | D(("actually play %zu frames, wrote %d", |
8023f60b | 593 | avail_frames, (int)pcm_written_frames)); |
594 | if(pcm_written_frames < 0) { | |
595 | switch(pcm_written_frames) { | |
9d5da576 | 596 | case -EPIPE: /* underrun */ |
597 | error(0, "snd_pcm_writei reports underrun"); | |
598 | if((err = snd_pcm_prepare(pcm)) < 0) | |
599 | fatal(0, "error calling snd_pcm_prepare: %d", err); | |
600 | return; | |
601 | case -EAGAIN: | |
602 | return; | |
603 | default: | |
8023f60b | 604 | fatal(0, "error calling snd_pcm_writei: %d", |
605 | (int)pcm_written_frames); | |
9d5da576 | 606 | } |
607 | } | |
8023f60b | 608 | written_frames = pcm_written_frames; |
9d5da576 | 609 | written_bytes = written_frames * bpf; |
e83d0967 | 610 | break; |
3a3c7bb9 | 611 | } |
8023f60b | 612 | #endif |
e83d0967 | 613 | case BACKEND_COMMAND: |
9d5da576 | 614 | if(avail_bytes > frames * bpf) |
615 | avail_bytes = frames * bpf; | |
e83d0967 | 616 | written_bytes = write(cmdfd, playing->buffer + playing->start, |
9d5da576 | 617 | avail_bytes); |
618 | D(("actually play %zu bytes, wrote %d", | |
619 | avail_bytes, (int)written_bytes)); | |
620 | if(written_bytes < 0) { | |
621 | switch(errno) { | |
622 | case EPIPE: | |
e83d0967 RK |
623 | error(0, "hmm, command died; trying another"); |
624 | fork_cmd(); | |
9d5da576 | 625 | return; |
626 | case EAGAIN: | |
627 | return; | |
628 | } | |
460b9539 | 629 | } |
9d5da576 | 630 | written_frames = written_bytes / bpf; /* good enough */ |
e83d0967 RK |
631 | break; |
632 | case BACKEND_NETWORK: | |
633 | /* We transmit using RTP (RFC3550) and attempt to conform to the internet | |
634 | * AVT profile (RFC3551). */ | |
7aa087a7 | 635 | |
e83d0967 | 636 | if(idled) { |
451169fc | 637 | /* There may have been a gap. Fix up the RTP time accordingly. */ |
e83d0967 | 638 | struct timeval now; |
7aa087a7 RK |
639 | uint64_t delta; |
640 | uint64_t target_rtp_time; | |
641 | ||
642 | /* Find the current time */ | |
e83d0967 | 643 | xgettimeofday(&now, 0); |
7aa087a7 RK |
644 | /* Find the number of microseconds elapsed since rtp_time=0 */ |
645 | delta = tvsub_us(now, rtp_time_0); | |
646 | assert(delta <= UINT64_MAX / 88200); | |
647 | target_rtp_time = (delta * playing->format.rate | |
648 | * playing->format.channels) / 1000000; | |
649 | /* Overflows at ~6 years uptime with 44100Hz stereo */ | |
451169fc RK |
650 | |
651 | /* rtp_time is the number of samples we've played. NB that we play | |
652 | * RTP_AHEAD_MS ahead of ourselves, so it may legitimately be ahead of | |
653 | * the value we deduce from time comparison. | |
654 | * | |
655 | * Suppose we have 1s track started at t=0, and another track begins to | |
656 | * play at t=2s. Suppose RTP_AHEAD_MS=1000 and 44100Hz stereo. In that | |
657 | * case we'll send 1s of audio as fast as we can, giving rtp_time=88200. | |
658 | * rtp_time stops at this point. | |
659 | * | |
660 | * At t=2s we'll have calculated target_rtp_time=176400. In this case we | |
661 | * set rtp_time=176400 and the player can correctly conclude that it | |
662 | * should leave 1s between the tracks. | |
663 | * | |
664 | * Suppose instead that the second track arrives at t=0.5s, and that | |
665 | * we've managed to transmit the whole of the first track already. We'll | |
666 | * have target_rtp_time=44100. | |
667 | * | |
668 | * The desired behaviour is to play the second track back to back with | |
669 | * first. In this case therefore we do not modify rtp_time. | |
670 | * | |
671 | * Is it ever right to reduce rtp_time? No; for that would imply | |
672 | * transmitting packets with overlapping timestamp ranges, which does not | |
673 | * make sense. | |
674 | */ | |
675 | if(target_rtp_time > rtp_time) { | |
676 | /* More time has elapsed than we've transmitted samples. That implies | |
677 | * we've been 'sending' silence. */ | |
7aa087a7 RK |
678 | info("advancing rtp_time by %"PRIu64" samples", |
679 | target_rtp_time - rtp_time); | |
451169fc RK |
680 | rtp_time = target_rtp_time; |
681 | } else if(target_rtp_time < rtp_time) { | |
682 | const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS | |
683 | * config->sample_format.rate | |
684 | * config->sample_format.channels | |
685 | / 1000); | |
686 | ||
687 | if(target_rtp_time + samples_ahead < rtp_time) { | |
688 | info("reversing rtp_time by %"PRIu64" samples", | |
689 | rtp_time - target_rtp_time); | |
690 | } | |
691 | } | |
e83d0967 RK |
692 | } |
693 | header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */ | |
694 | header.seq = htons(rtp_seq++); | |
7aa087a7 | 695 | header.timestamp = htonl((uint32_t)rtp_time); |
e83d0967 RK |
696 | header.ssrc = rtp_id; |
697 | header.mpt = (idled ? 0x80 : 0x00) | 10; | |
698 | /* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from | |
699 | * the sample rate (in a library somewhere so that configuration.c can rule | |
700 | * out invalid rates). | |
701 | */ | |
702 | idled = 0; | |
703 | if(avail_bytes > NETWORK_BYTES - sizeof header) { | |
704 | avail_bytes = NETWORK_BYTES - sizeof header; | |
7aa087a7 | 705 | /* Always send a whole number of frames */ |
e83d0967 RK |
706 | avail_bytes -= avail_bytes % bpf; |
707 | } | |
708 | /* "The RTP clock rate used for generating the RTP timestamp is independent | |
709 | * of the number of channels and the encoding; it equals the number of | |
710 | * sampling periods per second. For N-channel encodings, each sampling | |
711 | * period (say, 1/8000 of a second) generates N samples. (This terminology | |
712 | * is standard, but somewhat confusing, as the total number of samples | |
713 | * generated per second is then the sampling rate times the channel | |
714 | * count.)" | |
715 | */ | |
ceb044f4 | 716 | write_bytes = avail_bytes; |
ceb044f4 RK |
717 | if(write_bytes) { |
718 | vec[0].iov_base = (void *)&header; | |
719 | vec[0].iov_len = sizeof header; | |
720 | vec[1].iov_base = playing->buffer + playing->start; | |
721 | vec[1].iov_len = avail_bytes; | |
ceb044f4 RK |
722 | do { |
723 | written_bytes = writev(bfd, | |
724 | vec, | |
725 | 2); | |
726 | } while(written_bytes < 0 && errno == EINTR); | |
727 | if(written_bytes < 0) { | |
728 | error(errno, "error transmitting audio data"); | |
729 | ++audio_errors; | |
730 | if(audio_errors == 10) | |
731 | fatal(0, "too many audio errors"); | |
e83d0967 | 732 | return; |
ceb044f4 RK |
733 | } |
734 | } else | |
e83d0967 RK |
735 | audio_errors /= 2; |
736 | written_bytes = avail_bytes; | |
737 | written_frames = written_bytes / bpf; | |
738 | /* Advance RTP's notion of the time */ | |
739 | rtp_time += written_frames * playing->format.channels; | |
e83d0967 RK |
740 | break; |
741 | default: | |
742 | assert(!"reached"); | |
460b9539 | 743 | } |
e83d0967 RK |
744 | /* written_bytes and written_frames had better both be set and correct by |
745 | * this point */ | |
460b9539 | 746 | playing->start += written_bytes; |
747 | playing->used -= written_bytes; | |
748 | playing->played += written_frames; | |
749 | /* If the pointer is at the end of the buffer (or the buffer is completely | |
750 | * empty) wrap it back to the start. */ | |
751 | if(!playing->used || playing->start == playing->size) | |
752 | playing->start = 0; | |
753 | frames -= written_frames; | |
754 | } | |
755 | ||
756 | /* Notify the server what we're up to. */ | |
757 | static void report(void) { | |
758 | struct speaker_message sm; | |
759 | ||
760 | if(playing && playing->buffer != (void *)&playing->format) { | |
761 | memset(&sm, 0, sizeof sm); | |
762 | sm.type = paused ? SM_PAUSED : SM_PLAYING; | |
763 | strcpy(sm.id, playing->id); | |
764 | sm.data = playing->played / playing->format.rate; | |
765 | speaker_send(1, &sm, 0); | |
766 | } | |
767 | time(&last_report); | |
768 | } | |
769 | ||
9d5da576 | 770 | static void reap(int __attribute__((unused)) sig) { |
e83d0967 | 771 | pid_t cmdpid; |
9d5da576 | 772 | int st; |
773 | ||
774 | do | |
e83d0967 RK |
775 | cmdpid = waitpid(-1, &st, WNOHANG); |
776 | while(cmdpid > 0); | |
9d5da576 | 777 | signal(SIGCHLD, reap); |
778 | } | |
779 | ||
460b9539 | 780 | static int addfd(int fd, int events) { |
781 | if(fdno < NFDS) { | |
782 | fds[fdno].fd = fd; | |
783 | fds[fdno].events = events; | |
784 | return fdno++; | |
785 | } else | |
786 | return -1; | |
787 | } | |
788 | ||
572d74ba | 789 | #if API_ALSA |
790 | /** @brief ALSA backend initialization */ | |
791 | static void alsa_init(void) { | |
792 | info("selected ALSA backend"); | |
793 | } | |
29601377 | 794 | |
795 | /** @brief ALSA backend activation */ | |
796 | static int alsa_activate(void) { | |
797 | /* If we need to change format then close the current device. */ | |
798 | if(pcm && !formats_equal(&playing->format, &pcm_format)) | |
799 | idle(); | |
800 | if(!pcm) { | |
801 | snd_pcm_hw_params_t *hwparams; | |
802 | snd_pcm_sw_params_t *swparams; | |
803 | snd_pcm_uframes_t pcm_bufsize; | |
804 | int err; | |
805 | int sample_format = 0; | |
806 | unsigned rate; | |
807 | ||
808 | D(("snd_pcm_open")); | |
809 | if((err = snd_pcm_open(&pcm, | |
810 | config->device, | |
811 | SND_PCM_STREAM_PLAYBACK, | |
812 | SND_PCM_NONBLOCK))) { | |
813 | error(0, "error from snd_pcm_open: %d", err); | |
814 | goto error; | |
815 | } | |
816 | snd_pcm_hw_params_alloca(&hwparams); | |
817 | D(("set up hw params")); | |
818 | if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0) | |
819 | fatal(0, "error from snd_pcm_hw_params_any: %d", err); | |
820 | if((err = snd_pcm_hw_params_set_access(pcm, hwparams, | |
821 | SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) | |
822 | fatal(0, "error from snd_pcm_hw_params_set_access: %d", err); | |
823 | switch(playing->format.bits) { | |
824 | case 8: | |
825 | sample_format = SND_PCM_FORMAT_S8; | |
826 | break; | |
827 | case 16: | |
828 | switch(playing->format.byte_format) { | |
829 | case AO_FMT_NATIVE: sample_format = SND_PCM_FORMAT_S16; break; | |
830 | case AO_FMT_LITTLE: sample_format = SND_PCM_FORMAT_S16_LE; break; | |
831 | case AO_FMT_BIG: sample_format = SND_PCM_FORMAT_S16_BE; break; | |
832 | error(0, "unrecognized byte format %d", playing->format.byte_format); | |
833 | goto fatal; | |
834 | } | |
835 | break; | |
836 | default: | |
837 | error(0, "unsupported sample size %d", playing->format.bits); | |
838 | goto fatal; | |
839 | } | |
840 | if((err = snd_pcm_hw_params_set_format(pcm, hwparams, | |
841 | sample_format)) < 0) { | |
842 | error(0, "error from snd_pcm_hw_params_set_format (%d): %d", | |
843 | sample_format, err); | |
844 | goto fatal; | |
845 | } | |
846 | rate = playing->format.rate; | |
847 | if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0) { | |
848 | error(0, "error from snd_pcm_hw_params_set_rate (%d): %d", | |
849 | playing->format.rate, err); | |
850 | goto fatal; | |
851 | } | |
852 | if(rate != (unsigned)playing->format.rate) | |
853 | info("want rate %d, got %u", playing->format.rate, rate); | |
854 | if((err = snd_pcm_hw_params_set_channels(pcm, hwparams, | |
855 | playing->format.channels)) < 0) { | |
856 | error(0, "error from snd_pcm_hw_params_set_channels (%d): %d", | |
857 | playing->format.channels, err); | |
858 | goto fatal; | |
859 | } | |
860 | bufsize = 3 * FRAMES; | |
861 | pcm_bufsize = bufsize; | |
862 | if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams, | |
863 | &pcm_bufsize)) < 0) | |
864 | fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d", | |
865 | 3 * FRAMES, err); | |
866 | if(pcm_bufsize != 3 * FRAMES && pcm_bufsize != last_pcm_bufsize) | |
867 | info("asked for PCM buffer of %d frames, got %d", | |
868 | 3 * FRAMES, (int)pcm_bufsize); | |
869 | last_pcm_bufsize = pcm_bufsize; | |
870 | if((err = snd_pcm_hw_params(pcm, hwparams)) < 0) | |
871 | fatal(0, "error calling snd_pcm_hw_params: %d", err); | |
872 | D(("set up sw params")); | |
873 | snd_pcm_sw_params_alloca(&swparams); | |
874 | if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0) | |
875 | fatal(0, "error calling snd_pcm_sw_params_current: %d", err); | |
876 | if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, FRAMES)) < 0) | |
877 | fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d", | |
878 | FRAMES, err); | |
879 | if((err = snd_pcm_sw_params(pcm, swparams)) < 0) | |
880 | fatal(0, "error calling snd_pcm_sw_params: %d", err); | |
881 | pcm_format = playing->format; | |
882 | bpf = bytes_per_frame(&pcm_format); | |
883 | D(("acquired audio device")); | |
884 | log_params(hwparams, swparams); | |
885 | ready = 1; | |
886 | } | |
887 | return 0; | |
888 | fatal: | |
889 | abandon(); | |
890 | error: | |
891 | /* We assume the error is temporary and that we'll retry in a bit. */ | |
892 | if(pcm) { | |
893 | snd_pcm_close(pcm); | |
894 | pcm = 0; | |
895 | } | |
896 | return -1; | |
897 | } | |
572d74ba | 898 | #endif |
899 | ||
900 | /** @brief Command backend initialization */ | |
901 | static void command_init(void) { | |
902 | info("selected command backend"); | |
903 | fork_cmd(); | |
904 | } | |
905 | ||
29601377 | 906 | /** @brief Command backend activation */ |
907 | static int command_activate(void) { | |
908 | if(!ready) { | |
909 | pcm_format = config->sample_format; | |
910 | bufsize = 3 * FRAMES; | |
911 | bpf = bytes_per_frame(&config->sample_format); | |
912 | D(("acquired audio device")); | |
913 | ready = 1; | |
914 | } | |
915 | return 0; | |
916 | } | |
917 | ||
572d74ba | 918 | /** @brief Network backend initialization */ |
919 | static void network_init(void) { | |
e83d0967 RK |
920 | struct addrinfo *res, *sres; |
921 | static const struct addrinfo pref = { | |
922 | 0, | |
923 | PF_INET, | |
924 | SOCK_DGRAM, | |
925 | IPPROTO_UDP, | |
926 | 0, | |
927 | 0, | |
928 | 0, | |
929 | 0 | |
930 | }; | |
931 | static const struct addrinfo prefbind = { | |
932 | AI_PASSIVE, | |
933 | PF_INET, | |
934 | SOCK_DGRAM, | |
935 | IPPROTO_UDP, | |
936 | 0, | |
937 | 0, | |
938 | 0, | |
939 | 0 | |
940 | }; | |
941 | static const int one = 1; | |
24d0936b RK |
942 | int sndbuf, target_sndbuf = 131072; |
943 | socklen_t len; | |
e83d0967 | 944 | char *sockname, *ssockname; |
572d74ba | 945 | |
946 | res = get_address(&config->broadcast, &pref, &sockname); | |
947 | if(!res) exit(-1); | |
948 | if(config->broadcast_from.n) { | |
949 | sres = get_address(&config->broadcast_from, &prefbind, &ssockname); | |
950 | if(!sres) exit(-1); | |
951 | } else | |
952 | sres = 0; | |
953 | if((bfd = socket(res->ai_family, | |
954 | res->ai_socktype, | |
955 | res->ai_protocol)) < 0) | |
956 | fatal(errno, "error creating broadcast socket"); | |
957 | if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0) | |
958 | fatal(errno, "error setting SO_BROADCAST on broadcast socket"); | |
959 | len = sizeof sndbuf; | |
960 | if(getsockopt(bfd, SOL_SOCKET, SO_SNDBUF, | |
961 | &sndbuf, &len) < 0) | |
962 | fatal(errno, "error getting SO_SNDBUF"); | |
963 | if(target_sndbuf > sndbuf) { | |
964 | if(setsockopt(bfd, SOL_SOCKET, SO_SNDBUF, | |
965 | &target_sndbuf, sizeof target_sndbuf) < 0) | |
966 | error(errno, "error setting SO_SNDBUF to %d", target_sndbuf); | |
967 | else | |
968 | info("changed socket send buffer size from %d to %d", | |
969 | sndbuf, target_sndbuf); | |
970 | } else | |
971 | info("default socket send buffer is %d", | |
972 | sndbuf); | |
973 | /* We might well want to set additional broadcast- or multicast-related | |
974 | * options here */ | |
975 | if(sres && bind(bfd, sres->ai_addr, sres->ai_addrlen) < 0) | |
976 | fatal(errno, "error binding broadcast socket to %s", ssockname); | |
977 | if(connect(bfd, res->ai_addr, res->ai_addrlen) < 0) | |
978 | fatal(errno, "error connecting broadcast socket to %s", sockname); | |
979 | /* Select an SSRC */ | |
980 | gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM); | |
981 | info("selected network backend, sending to %s", sockname); | |
982 | if(config->sample_format.byte_format != AO_FMT_BIG) { | |
983 | info("forcing big-endian sample format"); | |
984 | config->sample_format.byte_format = AO_FMT_BIG; | |
985 | } | |
986 | } | |
987 | ||
29601377 | 988 | /** @brief Network backend activation */ |
989 | static int network_activate(void) { | |
990 | if(!ready) { | |
991 | pcm_format = config->sample_format; | |
992 | bufsize = 3 * FRAMES; | |
993 | bpf = bytes_per_frame(&config->sample_format); | |
994 | D(("acquired audio device")); | |
995 | ready = 1; | |
996 | } | |
997 | return 0; | |
998 | } | |
572d74ba | 999 | |
1000 | /** @brief Table of speaker backends */ | |
1001 | static const struct speaker_backend backends[] = { | |
1002 | #if API_ALSA | |
1003 | { | |
1004 | BACKEND_ALSA, | |
29601377 | 1005 | alsa_init, |
1006 | alsa_activate | |
572d74ba | 1007 | }, |
1008 | #endif | |
1009 | { | |
1010 | BACKEND_COMMAND, | |
29601377 | 1011 | command_init, |
1012 | command_activate | |
572d74ba | 1013 | }, |
1014 | { | |
1015 | BACKEND_NETWORK, | |
29601377 | 1016 | network_init, |
1017 | network_activate | |
572d74ba | 1018 | }, |
29601377 | 1019 | { -1, 0, 0 } |
572d74ba | 1020 | }; |
1021 | ||
1022 | int main(int argc, char **argv) { | |
1023 | int n, fd, stdin_slot, alsa_slots, cmdfd_slot, bfd_slot, poke, timeout; | |
1024 | struct track *t; | |
1025 | struct speaker_message sm; | |
8023f60b | 1026 | #if API_ALSA |
1027 | int alsa_nslots = -1, err; | |
1028 | #endif | |
460b9539 | 1029 | |
1030 | set_progname(argv); | |
460b9539 | 1031 | if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); |
1032 | while((n = getopt_long(argc, argv, "hVc:dD", options, 0)) >= 0) { | |
1033 | switch(n) { | |
1034 | case 'h': help(); | |
1035 | case 'V': version(); | |
1036 | case 'c': configfile = optarg; break; | |
1037 | case 'd': debugging = 1; break; | |
1038 | case 'D': debugging = 0; break; | |
1039 | default: fatal(0, "invalid option"); | |
1040 | } | |
1041 | } | |
1042 | if(getenv("DISORDER_DEBUG_SPEAKER")) debugging = 1; | |
1043 | /* If stderr is a TTY then log there, otherwise to syslog. */ | |
1044 | if(!isatty(2)) { | |
1045 | openlog(progname, LOG_PID, LOG_DAEMON); | |
1046 | log_default = &log_syslog; | |
1047 | } | |
1048 | if(config_read()) fatal(0, "cannot read configuration"); | |
1049 | /* ignore SIGPIPE */ | |
1050 | signal(SIGPIPE, SIG_IGN); | |
9d5da576 | 1051 | /* reap kids */ |
1052 | signal(SIGCHLD, reap); | |
460b9539 | 1053 | /* set nice value */ |
1054 | xnice(config->nice_speaker); | |
1055 | /* change user */ | |
1056 | become_mortal(); | |
1057 | /* make sure we're not root, whatever the config says */ | |
1058 | if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root"); | |
572d74ba | 1059 | /* identify the backend used to play */ |
1060 | for(n = 0; backends[n].backend != -1; ++n) | |
1061 | if(backends[n].backend == config->speaker_backend) | |
1062 | break; | |
1063 | if(backends[n].backend == -1) | |
1064 | fatal(0, "unsupported backend %d", config->speaker_backend); | |
1065 | backend = &backends[n]; | |
1066 | /* backend-specific initialization */ | |
1067 | backend->init(); | |
460b9539 | 1068 | while(getppid() != 1) { |
1069 | fdno = 0; | |
1070 | /* Always ready for commands from the main server. */ | |
1071 | stdin_slot = addfd(0, POLLIN); | |
1072 | /* Try to read sample data for the currently playing track if there is | |
1073 | * buffer space. */ | |
1074 | if(playing && !playing->eof && playing->used < playing->size) { | |
1075 | playing->slot = addfd(playing->fd, POLLIN); | |
1076 | } else if(playing) | |
1077 | playing->slot = -1; | |
1078 | /* If forceplay is set then wait until it succeeds before waiting on the | |
1079 | * sound device. */ | |
9d5da576 | 1080 | alsa_slots = -1; |
e83d0967 RK |
1081 | cmdfd_slot = -1; |
1082 | bfd_slot = -1; | |
1083 | /* By default we will wait up to a second before thinking about current | |
1084 | * state. */ | |
1085 | timeout = 1000; | |
8023f60b | 1086 | if(ready && !forceplay) { |
e83d0967 RK |
1087 | switch(config->speaker_backend) { |
1088 | case BACKEND_COMMAND: | |
1089 | /* We send sample data to the subprocess as fast as it can accept it. | |
1090 | * This isn't ideal as pause latency can be very high as a result. */ | |
1091 | if(cmdfd >= 0) | |
1092 | cmdfd_slot = addfd(cmdfd, POLLOUT); | |
1093 | break; | |
7aa087a7 RK |
1094 | case BACKEND_NETWORK: { |
1095 | struct timeval now; | |
1096 | uint64_t target_us; | |
1097 | uint64_t target_rtp_time; | |
508acf7a RK |
1098 | const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS |
1099 | * config->sample_format.rate | |
1100 | * config->sample_format.channels | |
1101 | / 1000); | |
ae5b28b9 | 1102 | #if 0 |
7aa087a7 | 1103 | static unsigned logit; |
ae5b28b9 | 1104 | #endif |
7aa087a7 RK |
1105 | |
1106 | /* If we're starting then initialize the base time */ | |
1107 | if(!rtp_time) | |
1108 | xgettimeofday(&rtp_time_0, 0); | |
1109 | /* We send audio data whenever we get RTP_AHEAD seconds or more | |
1110 | * behind */ | |
e83d0967 | 1111 | xgettimeofday(&now, 0); |
7aa087a7 RK |
1112 | target_us = tvsub_us(now, rtp_time_0); |
1113 | assert(target_us <= UINT64_MAX / 88200); | |
1114 | target_rtp_time = (target_us * config->sample_format.rate | |
1115 | * config->sample_format.channels) | |
1116 | ||
1117 | / 1000000; | |
ae5b28b9 | 1118 | #if 0 |
7aa087a7 RK |
1119 | /* TODO remove logging guff */ |
1120 | if(!(logit++ & 1023)) | |
1121 | info("rtp_time %llu target %llu difference %lld [%lld]", | |
1122 | rtp_time, target_rtp_time, | |
1123 | rtp_time - target_rtp_time, | |
189e9830 RK |
1124 | samples_ahead); |
1125 | #endif | |
508acf7a | 1126 | if((int64_t)(rtp_time - target_rtp_time) < samples_ahead) |
e83d0967 | 1127 | bfd_slot = addfd(bfd, POLLOUT); |
e83d0967 | 1128 | break; |
7aa087a7 | 1129 | } |
8023f60b | 1130 | #if API_ALSA |
3a3c7bb9 | 1131 | case BACKEND_ALSA: { |
e83d0967 RK |
1132 | /* We send sample data to ALSA as fast as it can accept it, relying on |
1133 | * the fact that it has a relatively small buffer to minimize pause | |
1134 | * latency. */ | |
9d5da576 | 1135 | int retry = 3; |
1136 | ||
1137 | alsa_slots = fdno; | |
1138 | do { | |
1139 | retry = 0; | |
1140 | alsa_nslots = snd_pcm_poll_descriptors(pcm, &fds[fdno], NFDS - fdno); | |
1141 | if((alsa_nslots <= 0 | |
1142 | || !(fds[alsa_slots].events & POLLOUT)) | |
1143 | && snd_pcm_state(pcm) == SND_PCM_STATE_XRUN) { | |
1144 | error(0, "underrun detected after call to snd_pcm_poll_descriptors()"); | |
1145 | if((err = snd_pcm_prepare(pcm))) | |
1146 | fatal(0, "error calling snd_pcm_prepare: %d", err); | |
1147 | } else | |
1148 | break; | |
1149 | } while(retry-- > 0); | |
1150 | if(alsa_nslots >= 0) | |
1151 | fdno += alsa_nslots; | |
e83d0967 | 1152 | break; |
3a3c7bb9 | 1153 | } |
8023f60b | 1154 | #endif |
e83d0967 RK |
1155 | default: |
1156 | assert(!"unknown backend"); | |
9d5da576 | 1157 | } |
1158 | } | |
460b9539 | 1159 | /* If any other tracks don't have a full buffer, try to read sample data |
1160 | * from them. */ | |
1161 | for(t = tracks; t; t = t->next) | |
1162 | if(t != playing) { | |
1163 | if(!t->eof && t->used < t->size) { | |
9d5da576 | 1164 | t->slot = addfd(t->fd, POLLIN | POLLHUP); |
460b9539 | 1165 | } else |
1166 | t->slot = -1; | |
1167 | } | |
e83d0967 RK |
1168 | /* Wait for something interesting to happen */ |
1169 | n = poll(fds, fdno, timeout); | |
460b9539 | 1170 | if(n < 0) { |
1171 | if(errno == EINTR) continue; | |
1172 | fatal(errno, "error calling poll"); | |
1173 | } | |
1174 | /* Play some sound before doing anything else */ | |
e83d0967 RK |
1175 | poke = 0; |
1176 | switch(config->speaker_backend) { | |
8023f60b | 1177 | #if API_ALSA |
e83d0967 RK |
1178 | case BACKEND_ALSA: |
1179 | if(alsa_slots != -1) { | |
1180 | unsigned short alsa_revents; | |
1181 | ||
1182 | if((err = snd_pcm_poll_descriptors_revents(pcm, | |
1183 | &fds[alsa_slots], | |
1184 | alsa_nslots, | |
1185 | &alsa_revents)) < 0) | |
1186 | fatal(0, "error calling snd_pcm_poll_descriptors_revents: %d", err); | |
1187 | if(alsa_revents & (POLLOUT | POLLERR)) | |
1188 | play(3 * FRAMES); | |
1189 | } else | |
1190 | poke = 1; | |
1191 | break; | |
8023f60b | 1192 | #endif |
e83d0967 RK |
1193 | case BACKEND_COMMAND: |
1194 | if(cmdfd_slot != -1) { | |
1195 | if(fds[cmdfd_slot].revents & (POLLOUT | POLLERR)) | |
1196 | play(3 * FRAMES); | |
1197 | } else | |
1198 | poke = 1; | |
1199 | break; | |
1200 | case BACKEND_NETWORK: | |
1201 | if(bfd_slot != -1) { | |
1202 | if(fds[bfd_slot].revents & (POLLOUT | POLLERR)) | |
1203 | play(3 * FRAMES); | |
1204 | } else | |
1205 | poke = 1; | |
1206 | break; | |
1207 | } | |
1208 | if(poke) { | |
460b9539 | 1209 | /* Some attempt to play must have failed */ |
1210 | if(playing && !paused) | |
1211 | play(forceplay); | |
1212 | else | |
1213 | forceplay = 0; /* just in case */ | |
1214 | } | |
1215 | /* Perhaps we have a command to process */ | |
1216 | if(fds[stdin_slot].revents & POLLIN) { | |
1217 | n = speaker_recv(0, &sm, &fd); | |
1218 | if(n > 0) | |
1219 | switch(sm.type) { | |
1220 | case SM_PREPARE: | |
1221 | D(("SM_PREPARE %s %d", sm.id, fd)); | |
1222 | if(fd == -1) fatal(0, "got SM_PREPARE but no file descriptor"); | |
1223 | t = findtrack(sm.id, 1); | |
1224 | acquire(t, fd); | |
1225 | break; | |
1226 | case SM_PLAY: | |
1227 | D(("SM_PLAY %s %d", sm.id, fd)); | |
1228 | if(playing) fatal(0, "got SM_PLAY but already playing something"); | |
1229 | t = findtrack(sm.id, 1); | |
1230 | if(fd != -1) acquire(t, fd); | |
1231 | playing = t; | |
8023f60b | 1232 | play(bufsize); |
460b9539 | 1233 | report(); |
1234 | break; | |
1235 | case SM_PAUSE: | |
1236 | D(("SM_PAUSE")); | |
1237 | paused = 1; | |
1238 | report(); | |
1239 | break; | |
1240 | case SM_RESUME: | |
1241 | D(("SM_RESUME")); | |
1242 | if(paused) { | |
1243 | paused = 0; | |
1244 | if(playing) | |
8023f60b | 1245 | play(bufsize); |
460b9539 | 1246 | } |
1247 | report(); | |
1248 | break; | |
1249 | case SM_CANCEL: | |
1250 | D(("SM_CANCEL %s", sm.id)); | |
1251 | t = removetrack(sm.id); | |
1252 | if(t) { | |
1253 | if(t == playing) { | |
1254 | sm.type = SM_FINISHED; | |
1255 | strcpy(sm.id, playing->id); | |
1256 | speaker_send(1, &sm, 0); | |
1257 | playing = 0; | |
1258 | } | |
1259 | destroy(t); | |
1260 | } else | |
1261 | error(0, "SM_CANCEL for unknown track %s", sm.id); | |
1262 | report(); | |
1263 | break; | |
1264 | case SM_RELOAD: | |
1265 | D(("SM_RELOAD")); | |
1266 | if(config_read()) error(0, "cannot read configuration"); | |
1267 | info("reloaded configuration"); | |
1268 | break; | |
1269 | default: | |
1270 | error(0, "unknown message type %d", sm.type); | |
1271 | } | |
1272 | } | |
1273 | /* Read in any buffered data */ | |
1274 | for(t = tracks; t; t = t->next) | |
9d5da576 | 1275 | if(t->slot != -1 && (fds[t->slot].revents & (POLLIN | POLLHUP))) |
460b9539 | 1276 | fill(t); |
1277 | /* We might be able to play now */ | |
9d5da576 | 1278 | if(ready && forceplay && playing && !paused) |
460b9539 | 1279 | play(forceplay); |
1280 | /* Maybe we finished playing a track somewhere in the above */ | |
1281 | maybe_finished(); | |
1282 | /* If we don't need the sound device for now then close it for the benefit | |
1283 | * of anyone else who wants it. */ | |
9d5da576 | 1284 | if((!playing || paused) && ready) |
460b9539 | 1285 | idle(); |
1286 | /* If we've not reported out state for a second do so now. */ | |
1287 | if(time(0) > last_report) | |
1288 | report(); | |
1289 | } | |
1290 | info("stopped (parent terminated)"); | |
1291 | exit(0); | |
1292 | } | |
1293 | ||
1294 | /* | |
1295 | Local Variables: | |
1296 | c-basic-offset:2 | |
1297 | comment-column:40 | |
1298 | fill-column:79 | |
1299 | indent-tabs-mode:nil | |
1300 | End: | |
1301 | */ |