Commit | Line | Data |
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460b9539 | 1 | /* |
2 | * This file is part of DisOrder | |
dea8f8aa | 3 | * Copyright (C) 2005, 2006, 2007 Richard Kettlewell |
460b9539 | 4 | * |
5 | * This program is free software; you can redistribute it and/or modify | |
6 | * it under the terms of the GNU General Public License as published by | |
7 | * the Free Software Foundation; either version 2 of the License, or | |
8 | * (at your option) any later version. | |
9 | * | |
10 | * This program is distributed in the hope that it will be useful, but | |
11 | * WITHOUT ANY WARRANTY; without even the implied warranty of | |
12 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
13 | * General Public License for more details. | |
14 | * | |
15 | * You should have received a copy of the GNU General Public License | |
16 | * along with this program; if not, write to the Free Software | |
17 | * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 | |
18 | * USA | |
19 | */ | |
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 | */ | |
107 | #define NETWORK_BYTES 1024 | |
e83d0967 | 108 | |
1674096e | 109 | /** @brief Maximum RTP playahead (seconds) */ |
110 | #define RTP_AHEAD 2 | |
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 */ | |
149 | static uint32_t rtp_time; /* RTP timestamp */ | |
150 | static struct timeval rtp_time_real; /* corresponding real time */ | |
151 | static uint16_t rtp_seq; /* frame sequence number */ | |
152 | static uint32_t rtp_id; /* RTP SSRC */ | |
153 | static int idled; /* set when idled */ | |
154 | static int audio_errors; /* audio error counter */ | |
460b9539 | 155 | |
156 | static const struct option options[] = { | |
157 | { "help", no_argument, 0, 'h' }, | |
158 | { "version", no_argument, 0, 'V' }, | |
159 | { "config", required_argument, 0, 'c' }, | |
160 | { "debug", no_argument, 0, 'd' }, | |
161 | { "no-debug", no_argument, 0, 'D' }, | |
162 | { 0, 0, 0, 0 } | |
163 | }; | |
164 | ||
165 | /* Display usage message and terminate. */ | |
166 | static void help(void) { | |
167 | xprintf("Usage:\n" | |
168 | " disorder-speaker [OPTIONS]\n" | |
169 | "Options:\n" | |
170 | " --help, -h Display usage message\n" | |
171 | " --version, -V Display version number\n" | |
172 | " --config PATH, -c PATH Set configuration file\n" | |
173 | " --debug, -d Turn on debugging\n" | |
174 | "\n" | |
175 | "Speaker process for DisOrder. Not intended to be run\n" | |
176 | "directly.\n"); | |
177 | xfclose(stdout); | |
178 | exit(0); | |
179 | } | |
180 | ||
181 | /* Display version number and terminate. */ | |
182 | static void version(void) { | |
183 | xprintf("disorder-speaker version %s\n", disorder_version_string); | |
184 | xfclose(stdout); | |
185 | exit(0); | |
186 | } | |
187 | ||
1674096e | 188 | /** @brief Return the number of bytes per frame in @p format */ |
460b9539 | 189 | static size_t bytes_per_frame(const ao_sample_format *format) { |
190 | return format->channels * format->bits / 8; | |
191 | } | |
192 | ||
1674096e | 193 | /** @brief Find track @p id, maybe creating it if not found */ |
460b9539 | 194 | static struct track *findtrack(const char *id, int create) { |
195 | struct track *t; | |
196 | ||
197 | D(("findtrack %s %d", id, create)); | |
198 | for(t = tracks; t && strcmp(id, t->id); t = t->next) | |
199 | ; | |
200 | if(!t && create) { | |
201 | t = xmalloc(sizeof *t); | |
202 | t->next = tracks; | |
203 | strcpy(t->id, id); | |
204 | t->fd = -1; | |
205 | tracks = t; | |
206 | /* The initial input buffer will be the sample format. */ | |
207 | t->buffer = (void *)&t->format; | |
208 | t->size = sizeof t->format; | |
209 | } | |
210 | return t; | |
211 | } | |
212 | ||
1674096e | 213 | /** @brief Remove track @p id (but do not destroy it) */ |
460b9539 | 214 | static struct track *removetrack(const char *id) { |
215 | struct track *t, **tt; | |
216 | ||
217 | D(("removetrack %s", id)); | |
218 | for(tt = &tracks; (t = *tt) && strcmp(id, t->id); tt = &t->next) | |
219 | ; | |
220 | if(t) | |
221 | *tt = t->next; | |
222 | return t; | |
223 | } | |
224 | ||
1674096e | 225 | /** @brief Destroy a track */ |
460b9539 | 226 | static void destroy(struct track *t) { |
227 | D(("destroy %s", t->id)); | |
228 | if(t->fd != -1) xclose(t->fd); | |
229 | if(t->buffer != (void *)&t->format) free(t->buffer); | |
230 | free(t); | |
231 | } | |
232 | ||
1674096e | 233 | /** @brief Notice a new connection */ |
460b9539 | 234 | static void acquire(struct track *t, int fd) { |
235 | D(("acquire %s %d", t->id, fd)); | |
236 | if(t->fd != -1) | |
237 | xclose(t->fd); | |
238 | t->fd = fd; | |
239 | nonblock(fd); | |
240 | } | |
241 | ||
1674096e | 242 | /** @brief Return true if A and B denote identical libao formats, else false */ |
243 | static int formats_equal(const ao_sample_format *a, | |
244 | const ao_sample_format *b) { | |
245 | return (a->bits == b->bits | |
246 | && a->rate == b->rate | |
247 | && a->channels == b->channels | |
248 | && a->byte_format == b->byte_format); | |
249 | } | |
250 | ||
251 | /** @brief Compute arguments to sox */ | |
252 | static void soxargs(const char ***pp, char **qq, ao_sample_format *ao) { | |
253 | int n; | |
254 | ||
255 | *(*pp)++ = "-t.raw"; | |
256 | *(*pp)++ = "-s"; | |
257 | *(*pp)++ = *qq; n = sprintf(*qq, "-r%d", ao->rate); *qq += n + 1; | |
258 | *(*pp)++ = *qq; n = sprintf(*qq, "-c%d", ao->channels); *qq += n + 1; | |
259 | /* sox 12.17.9 insists on -b etc; CVS sox insists on -<n> etc; both are | |
260 | * deployed! */ | |
261 | switch(config->sox_generation) { | |
262 | case 0: | |
263 | if(ao->bits != 8 | |
264 | && ao->byte_format != AO_FMT_NATIVE | |
265 | && ao->byte_format != MACHINE_AO_FMT) { | |
266 | *(*pp)++ = "-x"; | |
267 | } | |
268 | switch(ao->bits) { | |
269 | case 8: *(*pp)++ = "-b"; break; | |
270 | case 16: *(*pp)++ = "-w"; break; | |
271 | case 32: *(*pp)++ = "-l"; break; | |
272 | case 64: *(*pp)++ = "-d"; break; | |
273 | default: fatal(0, "cannot handle sample size %d", (int)ao->bits); | |
274 | } | |
275 | break; | |
276 | case 1: | |
277 | switch(ao->byte_format) { | |
278 | case AO_FMT_NATIVE: break; | |
279 | case AO_FMT_BIG: *(*pp)++ = "-B"; break; | |
280 | case AO_FMT_LITTLE: *(*pp)++ = "-L"; break; | |
281 | } | |
282 | *(*pp)++ = *qq; n = sprintf(*qq, "-%d", ao->bits/8); *qq += n + 1; | |
283 | break; | |
284 | } | |
285 | } | |
286 | ||
287 | /** @brief Enable format translation | |
288 | * | |
289 | * If necessary, replaces a tracks inbound file descriptor with one connected | |
290 | * to a sox invocation, which performs the required translation. | |
291 | */ | |
292 | static void enable_translation(struct track *t) { | |
293 | switch(config->speaker_backend) { | |
294 | case BACKEND_COMMAND: | |
295 | case BACKEND_NETWORK: | |
296 | /* These backends need a specific sample format */ | |
297 | break; | |
298 | case BACKEND_ALSA: | |
299 | /* ALSA can cope */ | |
300 | return; | |
301 | } | |
302 | if(!formats_equal(&t->format, &config->sample_format)) { | |
303 | char argbuf[1024], *q = argbuf; | |
304 | const char *av[18], **pp = av; | |
305 | int soxpipe[2]; | |
306 | pid_t soxkid; | |
307 | ||
308 | *pp++ = "sox"; | |
309 | soxargs(&pp, &q, &t->format); | |
310 | *pp++ = "-"; | |
311 | soxargs(&pp, &q, &config->sample_format); | |
312 | *pp++ = "-"; | |
313 | *pp++ = 0; | |
314 | if(debugging) { | |
315 | for(pp = av; *pp; pp++) | |
316 | D(("sox arg[%d] = %s", pp - av, *pp)); | |
317 | D(("end args")); | |
318 | } | |
319 | xpipe(soxpipe); | |
320 | soxkid = xfork(); | |
321 | if(soxkid == 0) { | |
322 | signal(SIGPIPE, SIG_DFL); | |
323 | xdup2(t->fd, 0); | |
324 | xdup2(soxpipe[1], 1); | |
325 | fcntl(0, F_SETFL, fcntl(0, F_GETFL) & ~O_NONBLOCK); | |
326 | close(soxpipe[0]); | |
327 | close(soxpipe[1]); | |
328 | close(t->fd); | |
329 | execvp("sox", (char **)av); | |
330 | _exit(1); | |
331 | } | |
332 | D(("forking sox for format conversion (kid = %d)", soxkid)); | |
333 | close(t->fd); | |
334 | close(soxpipe[1]); | |
335 | t->fd = soxpipe[0]; | |
336 | t->format = config->sample_format; | |
337 | ready = 0; | |
338 | } | |
339 | } | |
340 | ||
341 | /** @brief Read data into a sample buffer | |
342 | * @param t Pointer to track | |
343 | * @return 0 on success, -1 on EOF | |
344 | * | |
345 | * This is effectively the read callback on @c t->fd. | |
346 | */ | |
460b9539 | 347 | static int fill(struct track *t) { |
348 | size_t where, left; | |
349 | int n; | |
350 | ||
351 | D(("fill %s: eof=%d used=%zu size=%zu got_format=%d", | |
352 | t->id, t->eof, t->used, t->size, t->got_format)); | |
353 | if(t->eof) return -1; | |
354 | if(t->used < t->size) { | |
355 | /* there is room left in the buffer */ | |
356 | where = (t->start + t->used) % t->size; | |
357 | if(t->got_format) { | |
358 | /* We are reading audio data, get as much as we can */ | |
359 | if(where >= t->start) left = t->size - where; | |
360 | else left = t->start - where; | |
361 | } else | |
362 | /* We are still waiting for the format, only get that */ | |
363 | left = sizeof (ao_sample_format) - t->used; | |
364 | do { | |
365 | n = read(t->fd, t->buffer + where, left); | |
366 | } while(n < 0 && errno == EINTR); | |
367 | if(n < 0) { | |
368 | if(errno != EAGAIN) fatal(errno, "error reading sample stream"); | |
369 | return 0; | |
370 | } | |
371 | if(n == 0) { | |
372 | D(("fill %s: eof detected", t->id)); | |
373 | t->eof = 1; | |
374 | return -1; | |
375 | } | |
376 | t->used += n; | |
377 | if(!t->got_format && t->used >= sizeof (ao_sample_format)) { | |
378 | assert(t->used == sizeof (ao_sample_format)); | |
379 | /* Check that our assumptions are met. */ | |
380 | if(t->format.bits & 7) | |
381 | fatal(0, "bits per sample not a multiple of 8"); | |
1674096e | 382 | /* If the input format is unsuitable, arrange to translate it */ |
383 | enable_translation(t); | |
460b9539 | 384 | /* Make a new buffer for audio data. */ |
385 | t->size = bytes_per_frame(&t->format) * t->format.rate * BUFFER_SECONDS; | |
386 | t->buffer = xmalloc(t->size); | |
387 | t->used = 0; | |
388 | t->got_format = 1; | |
389 | D(("got format for %s", t->id)); | |
390 | } | |
391 | } | |
392 | return 0; | |
393 | } | |
394 | ||
1674096e | 395 | /** @brief Close the sound device */ |
460b9539 | 396 | static void idle(void) { |
460b9539 | 397 | D(("idle")); |
8023f60b | 398 | #if API_ALSA |
e83d0967 | 399 | if(config->speaker_backend == BACKEND_ALSA && pcm) { |
8023f60b | 400 | int err; |
401 | ||
460b9539 | 402 | if((err = snd_pcm_nonblock(pcm, 0)) < 0) |
403 | fatal(0, "error calling snd_pcm_nonblock: %d", err); | |
404 | D(("draining pcm")); | |
405 | snd_pcm_drain(pcm); | |
406 | D(("closing pcm")); | |
407 | snd_pcm_close(pcm); | |
408 | pcm = 0; | |
409 | forceplay = 0; | |
410 | D(("released audio device")); | |
411 | } | |
8023f60b | 412 | #endif |
e83d0967 | 413 | idled = 1; |
9d5da576 | 414 | ready = 0; |
460b9539 | 415 | } |
416 | ||
1674096e | 417 | /** @brief Abandon the current track */ |
460b9539 | 418 | static void abandon(void) { |
419 | struct speaker_message sm; | |
420 | ||
421 | D(("abandon")); | |
422 | memset(&sm, 0, sizeof sm); | |
423 | sm.type = SM_FINISHED; | |
424 | strcpy(sm.id, playing->id); | |
425 | speaker_send(1, &sm, 0); | |
426 | removetrack(playing->id); | |
427 | destroy(playing); | |
428 | playing = 0; | |
429 | forceplay = 0; | |
430 | } | |
431 | ||
8023f60b | 432 | #if API_ALSA |
1674096e | 433 | /** @brief Log ALSA parameters */ |
1c6e6a61 | 434 | static void log_params(snd_pcm_hw_params_t *hwparams, |
435 | snd_pcm_sw_params_t *swparams) { | |
436 | snd_pcm_uframes_t f; | |
437 | unsigned u; | |
438 | ||
0c207c37 | 439 | return; /* too verbose */ |
1c6e6a61 | 440 | if(hwparams) { |
441 | /* TODO */ | |
442 | } | |
443 | if(swparams) { | |
444 | snd_pcm_sw_params_get_silence_size(swparams, &f); | |
445 | info("sw silence_size=%lu", (unsigned long)f); | |
446 | snd_pcm_sw_params_get_silence_threshold(swparams, &f); | |
447 | info("sw silence_threshold=%lu", (unsigned long)f); | |
448 | snd_pcm_sw_params_get_sleep_min(swparams, &u); | |
449 | info("sw sleep_min=%lu", (unsigned long)u); | |
450 | snd_pcm_sw_params_get_start_threshold(swparams, &f); | |
451 | info("sw start_threshold=%lu", (unsigned long)f); | |
452 | snd_pcm_sw_params_get_stop_threshold(swparams, &f); | |
453 | info("sw stop_threshold=%lu", (unsigned long)f); | |
454 | snd_pcm_sw_params_get_xfer_align(swparams, &f); | |
455 | info("sw xfer_align=%lu", (unsigned long)f); | |
456 | } | |
457 | } | |
8023f60b | 458 | #endif |
1c6e6a61 | 459 | |
1674096e | 460 | /** @brief Enable sound output |
461 | * | |
462 | * Makes sure the sound device is open and has the right sample format. Return | |
463 | * 0 on success and -1 on error. | |
464 | */ | |
460b9539 | 465 | static int activate(void) { |
460b9539 | 466 | /* If we don't know the format yet we cannot start. */ |
467 | if(!playing->got_format) { | |
468 | D((" - not got format for %s", playing->id)); | |
469 | return -1; | |
470 | } | |
e83d0967 RK |
471 | switch(config->speaker_backend) { |
472 | case BACKEND_COMMAND: | |
473 | case BACKEND_NETWORK: | |
9d5da576 | 474 | if(!ready) { |
475 | pcm_format = config->sample_format; | |
8023f60b | 476 | bufsize = 3 * FRAMES; |
9d5da576 | 477 | bpf = bytes_per_frame(&config->sample_format); |
478 | D(("acquired audio device")); | |
479 | ready = 1; | |
480 | } | |
481 | return 0; | |
e83d0967 | 482 | case BACKEND_ALSA: |
8023f60b | 483 | #if API_ALSA |
e83d0967 RK |
484 | /* If we need to change format then close the current device. */ |
485 | if(pcm && !formats_equal(&playing->format, &pcm_format)) | |
486 | idle(); | |
487 | if(!pcm) { | |
488 | snd_pcm_hw_params_t *hwparams; | |
489 | snd_pcm_sw_params_t *swparams; | |
490 | snd_pcm_uframes_t pcm_bufsize; | |
491 | int err; | |
492 | int sample_format = 0; | |
493 | unsigned rate; | |
494 | ||
495 | D(("snd_pcm_open")); | |
496 | if((err = snd_pcm_open(&pcm, | |
497 | config->device, | |
498 | SND_PCM_STREAM_PLAYBACK, | |
499 | SND_PCM_NONBLOCK))) { | |
500 | error(0, "error from snd_pcm_open: %d", err); | |
501 | goto error; | |
502 | } | |
503 | snd_pcm_hw_params_alloca(&hwparams); | |
504 | D(("set up hw params")); | |
505 | if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0) | |
506 | fatal(0, "error from snd_pcm_hw_params_any: %d", err); | |
507 | if((err = snd_pcm_hw_params_set_access(pcm, hwparams, | |
508 | SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) | |
509 | fatal(0, "error from snd_pcm_hw_params_set_access: %d", err); | |
510 | switch(playing->format.bits) { | |
511 | case 8: | |
512 | sample_format = SND_PCM_FORMAT_S8; | |
513 | break; | |
514 | case 16: | |
515 | switch(playing->format.byte_format) { | |
516 | case AO_FMT_NATIVE: sample_format = SND_PCM_FORMAT_S16; break; | |
517 | case AO_FMT_LITTLE: sample_format = SND_PCM_FORMAT_S16_LE; break; | |
518 | case AO_FMT_BIG: sample_format = SND_PCM_FORMAT_S16_BE; break; | |
519 | error(0, "unrecognized byte format %d", playing->format.byte_format); | |
520 | goto fatal; | |
521 | } | |
522 | break; | |
523 | default: | |
524 | error(0, "unsupported sample size %d", playing->format.bits); | |
460b9539 | 525 | goto fatal; |
526 | } | |
e83d0967 RK |
527 | if((err = snd_pcm_hw_params_set_format(pcm, hwparams, |
528 | sample_format)) < 0) { | |
529 | error(0, "error from snd_pcm_hw_params_set_format (%d): %d", | |
530 | sample_format, err); | |
531 | goto fatal; | |
532 | } | |
533 | rate = playing->format.rate; | |
534 | if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0) { | |
535 | error(0, "error from snd_pcm_hw_params_set_rate (%d): %d", | |
536 | playing->format.rate, err); | |
537 | goto fatal; | |
538 | } | |
539 | if(rate != (unsigned)playing->format.rate) | |
540 | info("want rate %d, got %u", playing->format.rate, rate); | |
541 | if((err = snd_pcm_hw_params_set_channels(pcm, hwparams, | |
542 | playing->format.channels)) < 0) { | |
543 | error(0, "error from snd_pcm_hw_params_set_channels (%d): %d", | |
544 | playing->format.channels, err); | |
545 | goto fatal; | |
546 | } | |
547 | bufsize = 3 * FRAMES; | |
548 | pcm_bufsize = bufsize; | |
549 | if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams, | |
550 | &pcm_bufsize)) < 0) | |
551 | fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d", | |
552 | 3 * FRAMES, err); | |
553 | if(pcm_bufsize != 3 * FRAMES && pcm_bufsize != last_pcm_bufsize) | |
554 | info("asked for PCM buffer of %d frames, got %d", | |
555 | 3 * FRAMES, (int)pcm_bufsize); | |
556 | last_pcm_bufsize = pcm_bufsize; | |
557 | if((err = snd_pcm_hw_params(pcm, hwparams)) < 0) | |
558 | fatal(0, "error calling snd_pcm_hw_params: %d", err); | |
559 | D(("set up sw params")); | |
560 | snd_pcm_sw_params_alloca(&swparams); | |
561 | if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0) | |
562 | fatal(0, "error calling snd_pcm_sw_params_current: %d", err); | |
563 | if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, FRAMES)) < 0) | |
564 | fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d", | |
565 | FRAMES, err); | |
566 | if((err = snd_pcm_sw_params(pcm, swparams)) < 0) | |
567 | fatal(0, "error calling snd_pcm_sw_params: %d", err); | |
568 | pcm_format = playing->format; | |
569 | bpf = bytes_per_frame(&pcm_format); | |
570 | D(("acquired audio device")); | |
571 | log_params(hwparams, swparams); | |
572 | ready = 1; | |
460b9539 | 573 | } |
e83d0967 RK |
574 | return 0; |
575 | fatal: | |
576 | abandon(); | |
577 | error: | |
578 | /* We assume the error is temporary and that we'll retry in a bit. */ | |
579 | if(pcm) { | |
580 | snd_pcm_close(pcm); | |
581 | pcm = 0; | |
460b9539 | 582 | } |
e83d0967 | 583 | return -1; |
8023f60b | 584 | #endif |
e83d0967 RK |
585 | default: |
586 | assert(!"reached"); | |
587 | } | |
460b9539 | 588 | } |
589 | ||
590 | /* Check to see whether the current track has finished playing */ | |
591 | static void maybe_finished(void) { | |
592 | if(playing | |
593 | && playing->eof | |
594 | && (!playing->got_format | |
595 | || playing->used < bytes_per_frame(&playing->format))) | |
596 | abandon(); | |
597 | } | |
598 | ||
e83d0967 RK |
599 | static void fork_cmd(void) { |
600 | pid_t cmdpid; | |
9d5da576 | 601 | int pfd[2]; |
e83d0967 | 602 | if(cmdfd != -1) close(cmdfd); |
9d5da576 | 603 | xpipe(pfd); |
e83d0967 RK |
604 | cmdpid = xfork(); |
605 | if(!cmdpid) { | |
1674096e | 606 | signal(SIGPIPE, SIG_DFL); |
9d5da576 | 607 | xdup2(pfd[0], 0); |
608 | close(pfd[0]); | |
609 | close(pfd[1]); | |
610 | execl("/bin/sh", "sh", "-c", config->speaker_command, (char *)0); | |
611 | fatal(errno, "error execing /bin/sh"); | |
612 | } | |
613 | close(pfd[0]); | |
e83d0967 RK |
614 | cmdfd = pfd[1]; |
615 | D(("forked cmd %d, fd = %d", cmdpid, cmdfd)); | |
9d5da576 | 616 | } |
617 | ||
460b9539 | 618 | static void play(size_t frames) { |
ceb044f4 | 619 | size_t avail_bytes, write_bytes, written_frames; |
9d5da576 | 620 | ssize_t written_bytes; |
0b75463f | 621 | struct rtp_header header; |
e83d0967 | 622 | struct iovec vec[2]; |
460b9539 | 623 | |
624 | if(activate()) { | |
625 | if(playing) | |
626 | forceplay = frames; | |
627 | else | |
628 | forceplay = 0; /* Must have called abandon() */ | |
629 | return; | |
630 | } | |
631 | D(("play: play %zu/%zu%s %dHz %db %dc", frames, playing->used / bpf, | |
632 | playing->eof ? " EOF" : "", | |
633 | playing->format.rate, | |
634 | playing->format.bits, | |
635 | playing->format.channels)); | |
636 | /* If we haven't got enough bytes yet wait until we have. Exception: when | |
637 | * we are at eof. */ | |
638 | if(playing->used < frames * bpf && !playing->eof) { | |
639 | forceplay = frames; | |
640 | return; | |
641 | } | |
642 | /* We have got enough data so don't force play again */ | |
643 | forceplay = 0; | |
644 | /* Figure out how many frames there are available to write */ | |
645 | if(playing->start + playing->used > playing->size) | |
646 | avail_bytes = playing->size - playing->start; | |
647 | else | |
648 | avail_bytes = playing->used; | |
9d5da576 | 649 | |
e83d0967 | 650 | switch(config->speaker_backend) { |
8023f60b | 651 | #if API_ALSA |
3a3c7bb9 | 652 | case BACKEND_ALSA: { |
8023f60b | 653 | snd_pcm_sframes_t pcm_written_frames; |
654 | size_t avail_frames; | |
655 | int err; | |
656 | ||
9d5da576 | 657 | avail_frames = avail_bytes / bpf; |
658 | if(avail_frames > frames) | |
659 | avail_frames = frames; | |
660 | if(!avail_frames) | |
460b9539 | 661 | return; |
8023f60b | 662 | pcm_written_frames = snd_pcm_writei(pcm, |
663 | playing->buffer + playing->start, | |
664 | avail_frames); | |
9d5da576 | 665 | D(("actually play %zu frames, wrote %d", |
8023f60b | 666 | avail_frames, (int)pcm_written_frames)); |
667 | if(pcm_written_frames < 0) { | |
668 | switch(pcm_written_frames) { | |
9d5da576 | 669 | case -EPIPE: /* underrun */ |
670 | error(0, "snd_pcm_writei reports underrun"); | |
671 | if((err = snd_pcm_prepare(pcm)) < 0) | |
672 | fatal(0, "error calling snd_pcm_prepare: %d", err); | |
673 | return; | |
674 | case -EAGAIN: | |
675 | return; | |
676 | default: | |
8023f60b | 677 | fatal(0, "error calling snd_pcm_writei: %d", |
678 | (int)pcm_written_frames); | |
9d5da576 | 679 | } |
680 | } | |
8023f60b | 681 | written_frames = pcm_written_frames; |
9d5da576 | 682 | written_bytes = written_frames * bpf; |
e83d0967 | 683 | break; |
3a3c7bb9 | 684 | } |
8023f60b | 685 | #endif |
e83d0967 | 686 | case BACKEND_COMMAND: |
9d5da576 | 687 | if(avail_bytes > frames * bpf) |
688 | avail_bytes = frames * bpf; | |
e83d0967 | 689 | written_bytes = write(cmdfd, playing->buffer + playing->start, |
9d5da576 | 690 | avail_bytes); |
691 | D(("actually play %zu bytes, wrote %d", | |
692 | avail_bytes, (int)written_bytes)); | |
693 | if(written_bytes < 0) { | |
694 | switch(errno) { | |
695 | case EPIPE: | |
e83d0967 RK |
696 | error(0, "hmm, command died; trying another"); |
697 | fork_cmd(); | |
9d5da576 | 698 | return; |
699 | case EAGAIN: | |
700 | return; | |
701 | } | |
460b9539 | 702 | } |
9d5da576 | 703 | written_frames = written_bytes / bpf; /* good enough */ |
e83d0967 RK |
704 | break; |
705 | case BACKEND_NETWORK: | |
706 | /* We transmit using RTP (RFC3550) and attempt to conform to the internet | |
707 | * AVT profile (RFC3551). */ | |
708 | if(rtp_time_real.tv_sec == 0) | |
709 | xgettimeofday(&rtp_time_real, 0); | |
710 | if(idled) { | |
711 | struct timeval now; | |
712 | xgettimeofday(&now, 0); | |
713 | /* There's been a gap. Fix up the RTP time accordingly. */ | |
dbf24eb4 RK |
714 | const long offset = (((now.tv_sec + now.tv_usec /1000000.0) |
715 | - (rtp_time_real.tv_sec + rtp_time_real.tv_usec / 1000000.0)) | |
716 | * playing->format.rate * playing->format.channels); | |
01ddd909 RK |
717 | if(offset >= 0) { |
718 | info("offset RTP timestamp by %ld", offset); | |
719 | rtp_time += offset; | |
9aa6b167 RK |
720 | } |
721 | rtp_time_real = now; | |
e83d0967 RK |
722 | } |
723 | header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */ | |
724 | header.seq = htons(rtp_seq++); | |
725 | header.timestamp = htonl(rtp_time); | |
726 | header.ssrc = rtp_id; | |
727 | header.mpt = (idled ? 0x80 : 0x00) | 10; | |
728 | /* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from | |
729 | * the sample rate (in a library somewhere so that configuration.c can rule | |
730 | * out invalid rates). | |
731 | */ | |
732 | idled = 0; | |
733 | if(avail_bytes > NETWORK_BYTES - sizeof header) { | |
734 | avail_bytes = NETWORK_BYTES - sizeof header; | |
735 | avail_bytes -= avail_bytes % bpf; | |
736 | } | |
737 | /* "The RTP clock rate used for generating the RTP timestamp is independent | |
738 | * of the number of channels and the encoding; it equals the number of | |
739 | * sampling periods per second. For N-channel encodings, each sampling | |
740 | * period (say, 1/8000 of a second) generates N samples. (This terminology | |
741 | * is standard, but somewhat confusing, as the total number of samples | |
742 | * generated per second is then the sampling rate times the channel | |
743 | * count.)" | |
744 | */ | |
ceb044f4 | 745 | write_bytes = avail_bytes; |
e83d0967 | 746 | #if 0 |
ceb044f4 RK |
747 | while(write_bytes > 0 && (uint32_t)(playing->buffer + playing->start + write_bytes - 4) == 0) |
748 | write_bytes -= 4; | |
e83d0967 | 749 | #endif |
ceb044f4 RK |
750 | if(write_bytes) { |
751 | vec[0].iov_base = (void *)&header; | |
752 | vec[0].iov_len = sizeof header; | |
753 | vec[1].iov_base = playing->buffer + playing->start; | |
754 | vec[1].iov_len = avail_bytes; | |
755 | #if 0 | |
756 | { | |
757 | char buffer[3 * sizeof header + 1]; | |
758 | size_t n; | |
759 | const uint8_t *ptr = (void *)&header; | |
760 | ||
761 | for(n = 0; n < sizeof header; ++n) | |
762 | sprintf(&buffer[3 * n], "%02x ", *ptr++); | |
763 | info(buffer); | |
764 | } | |
765 | #endif | |
766 | do { | |
767 | written_bytes = writev(bfd, | |
768 | vec, | |
769 | 2); | |
770 | } while(written_bytes < 0 && errno == EINTR); | |
771 | if(written_bytes < 0) { | |
772 | error(errno, "error transmitting audio data"); | |
773 | ++audio_errors; | |
774 | if(audio_errors == 10) | |
775 | fatal(0, "too many audio errors"); | |
e83d0967 | 776 | return; |
ceb044f4 RK |
777 | } |
778 | } else | |
e83d0967 RK |
779 | audio_errors /= 2; |
780 | written_bytes = avail_bytes; | |
781 | written_frames = written_bytes / bpf; | |
782 | /* Advance RTP's notion of the time */ | |
783 | rtp_time += written_frames * playing->format.channels; | |
784 | /* Advance the corresponding real time */ | |
785 | assert(NETWORK_BYTES <= 2000); /* else risk overflowing 32 bits */ | |
786 | rtp_time_real.tv_usec += written_frames * 1000000 / playing->format.rate; | |
787 | if(rtp_time_real.tv_usec >= 1000000) { | |
788 | ++rtp_time_real.tv_sec; | |
789 | rtp_time_real.tv_usec -= 1000000; | |
790 | } | |
01ddd909 | 791 | assert(rtp_time_real.tv_usec < 1000000); |
e83d0967 RK |
792 | break; |
793 | default: | |
794 | assert(!"reached"); | |
460b9539 | 795 | } |
e83d0967 RK |
796 | /* written_bytes and written_frames had better both be set and correct by |
797 | * this point */ | |
460b9539 | 798 | playing->start += written_bytes; |
799 | playing->used -= written_bytes; | |
800 | playing->played += written_frames; | |
801 | /* If the pointer is at the end of the buffer (or the buffer is completely | |
802 | * empty) wrap it back to the start. */ | |
803 | if(!playing->used || playing->start == playing->size) | |
804 | playing->start = 0; | |
805 | frames -= written_frames; | |
806 | } | |
807 | ||
808 | /* Notify the server what we're up to. */ | |
809 | static void report(void) { | |
810 | struct speaker_message sm; | |
811 | ||
812 | if(playing && playing->buffer != (void *)&playing->format) { | |
813 | memset(&sm, 0, sizeof sm); | |
814 | sm.type = paused ? SM_PAUSED : SM_PLAYING; | |
815 | strcpy(sm.id, playing->id); | |
816 | sm.data = playing->played / playing->format.rate; | |
817 | speaker_send(1, &sm, 0); | |
818 | } | |
819 | time(&last_report); | |
820 | } | |
821 | ||
9d5da576 | 822 | static void reap(int __attribute__((unused)) sig) { |
e83d0967 | 823 | pid_t cmdpid; |
9d5da576 | 824 | int st; |
825 | ||
826 | do | |
e83d0967 RK |
827 | cmdpid = waitpid(-1, &st, WNOHANG); |
828 | while(cmdpid > 0); | |
9d5da576 | 829 | signal(SIGCHLD, reap); |
830 | } | |
831 | ||
460b9539 | 832 | static int addfd(int fd, int events) { |
833 | if(fdno < NFDS) { | |
834 | fds[fdno].fd = fd; | |
835 | fds[fdno].events = events; | |
836 | return fdno++; | |
837 | } else | |
838 | return -1; | |
839 | } | |
840 | ||
841 | int main(int argc, char **argv) { | |
e83d0967 RK |
842 | int n, fd, stdin_slot, alsa_slots, cmdfd_slot, bfd_slot, poke, timeout; |
843 | struct timeval now, delta; | |
460b9539 | 844 | struct track *t; |
845 | struct speaker_message sm; | |
e83d0967 RK |
846 | struct addrinfo *res, *sres; |
847 | static const struct addrinfo pref = { | |
848 | 0, | |
849 | PF_INET, | |
850 | SOCK_DGRAM, | |
851 | IPPROTO_UDP, | |
852 | 0, | |
853 | 0, | |
854 | 0, | |
855 | 0 | |
856 | }; | |
857 | static const struct addrinfo prefbind = { | |
858 | AI_PASSIVE, | |
859 | PF_INET, | |
860 | SOCK_DGRAM, | |
861 | IPPROTO_UDP, | |
862 | 0, | |
863 | 0, | |
864 | 0, | |
865 | 0 | |
866 | }; | |
867 | static const int one = 1; | |
868 | char *sockname, *ssockname; | |
8023f60b | 869 | #if API_ALSA |
870 | int alsa_nslots = -1, err; | |
871 | #endif | |
460b9539 | 872 | |
873 | set_progname(argv); | |
460b9539 | 874 | if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); |
875 | while((n = getopt_long(argc, argv, "hVc:dD", options, 0)) >= 0) { | |
876 | switch(n) { | |
877 | case 'h': help(); | |
878 | case 'V': version(); | |
879 | case 'c': configfile = optarg; break; | |
880 | case 'd': debugging = 1; break; | |
881 | case 'D': debugging = 0; break; | |
882 | default: fatal(0, "invalid option"); | |
883 | } | |
884 | } | |
885 | if(getenv("DISORDER_DEBUG_SPEAKER")) debugging = 1; | |
886 | /* If stderr is a TTY then log there, otherwise to syslog. */ | |
887 | if(!isatty(2)) { | |
888 | openlog(progname, LOG_PID, LOG_DAEMON); | |
889 | log_default = &log_syslog; | |
890 | } | |
891 | if(config_read()) fatal(0, "cannot read configuration"); | |
892 | /* ignore SIGPIPE */ | |
893 | signal(SIGPIPE, SIG_IGN); | |
9d5da576 | 894 | /* reap kids */ |
895 | signal(SIGCHLD, reap); | |
460b9539 | 896 | /* set nice value */ |
897 | xnice(config->nice_speaker); | |
898 | /* change user */ | |
899 | become_mortal(); | |
900 | /* make sure we're not root, whatever the config says */ | |
901 | if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root"); | |
e83d0967 RK |
902 | switch(config->speaker_backend) { |
903 | case BACKEND_ALSA: | |
904 | info("selected ALSA backend"); | |
905 | case BACKEND_COMMAND: | |
906 | info("selected command backend"); | |
907 | fork_cmd(); | |
908 | break; | |
909 | case BACKEND_NETWORK: | |
910 | res = get_address(&config->broadcast, &pref, &sockname); | |
911 | if(!res) return -1; | |
912 | if(config->broadcast_from.n) { | |
913 | sres = get_address(&config->broadcast_from, &prefbind, &ssockname); | |
914 | if(!sres) return -1; | |
915 | } else | |
916 | sres = 0; | |
917 | if((bfd = socket(res->ai_family, | |
918 | res->ai_socktype, | |
919 | res->ai_protocol)) < 0) | |
920 | fatal(errno, "error creating broadcast socket"); | |
921 | if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0) | |
922 | fatal(errno, "error settting SO_BROADCAST on broadcast socket"); | |
923 | /* We might well want to set additional broadcast- or multicast-related | |
924 | * options here */ | |
925 | if(sres && bind(bfd, sres->ai_addr, sres->ai_addrlen) < 0) | |
926 | fatal(errno, "error binding broadcast socket to %s", ssockname); | |
927 | if(connect(bfd, res->ai_addr, res->ai_addrlen) < 0) | |
928 | fatal(errno, "error connecting broadcast socket to %s", sockname); | |
929 | /* Select an SSRC */ | |
930 | gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM); | |
931 | info("selected network backend, sending to %s", sockname); | |
932 | if(config->sample_format.byte_format != AO_FMT_BIG) { | |
933 | info("forcing big-endian sample format"); | |
934 | config->sample_format.byte_format = AO_FMT_BIG; | |
935 | } | |
936 | break; | |
937 | default: | |
938 | fatal(0, "unknown backend %d", config->speaker_backend); | |
8023f60b | 939 | } |
460b9539 | 940 | while(getppid() != 1) { |
941 | fdno = 0; | |
942 | /* Always ready for commands from the main server. */ | |
943 | stdin_slot = addfd(0, POLLIN); | |
944 | /* Try to read sample data for the currently playing track if there is | |
945 | * buffer space. */ | |
946 | if(playing && !playing->eof && playing->used < playing->size) { | |
947 | playing->slot = addfd(playing->fd, POLLIN); | |
948 | } else if(playing) | |
949 | playing->slot = -1; | |
950 | /* If forceplay is set then wait until it succeeds before waiting on the | |
951 | * sound device. */ | |
9d5da576 | 952 | alsa_slots = -1; |
e83d0967 RK |
953 | cmdfd_slot = -1; |
954 | bfd_slot = -1; | |
955 | /* By default we will wait up to a second before thinking about current | |
956 | * state. */ | |
957 | timeout = 1000; | |
8023f60b | 958 | if(ready && !forceplay) { |
e83d0967 RK |
959 | switch(config->speaker_backend) { |
960 | case BACKEND_COMMAND: | |
961 | /* We send sample data to the subprocess as fast as it can accept it. | |
962 | * This isn't ideal as pause latency can be very high as a result. */ | |
963 | if(cmdfd >= 0) | |
964 | cmdfd_slot = addfd(cmdfd, POLLOUT); | |
965 | break; | |
966 | case BACKEND_NETWORK: | |
967 | /* We want to keep the notional playing point somewhere in the near | |
968 | * future. If it's too near then clients that attempt even the | |
969 | * slightest amount of read-ahead will never catch up, and those that | |
970 | * don't will skip whenever there's a trivial network delay. If it's | |
971 | * too far ahead then pause latency will be too high. | |
972 | */ | |
973 | xgettimeofday(&now, 0); | |
974 | delta = tvsub(rtp_time_real, now); | |
975 | if(delta.tv_sec < RTP_AHEAD) { | |
976 | D(("delta = %ld.%06ld", (long)delta.tv_sec, (long)delta.tv_usec)); | |
977 | bfd_slot = addfd(bfd, POLLOUT); | |
978 | if(delta.tv_sec < 0) | |
979 | rtp_time_real = now; /* catch up */ | |
980 | } | |
981 | break; | |
8023f60b | 982 | #if API_ALSA |
3a3c7bb9 | 983 | case BACKEND_ALSA: { |
e83d0967 RK |
984 | /* We send sample data to ALSA as fast as it can accept it, relying on |
985 | * the fact that it has a relatively small buffer to minimize pause | |
986 | * latency. */ | |
9d5da576 | 987 | int retry = 3; |
988 | ||
989 | alsa_slots = fdno; | |
990 | do { | |
991 | retry = 0; | |
992 | alsa_nslots = snd_pcm_poll_descriptors(pcm, &fds[fdno], NFDS - fdno); | |
993 | if((alsa_nslots <= 0 | |
994 | || !(fds[alsa_slots].events & POLLOUT)) | |
995 | && snd_pcm_state(pcm) == SND_PCM_STATE_XRUN) { | |
996 | error(0, "underrun detected after call to snd_pcm_poll_descriptors()"); | |
997 | if((err = snd_pcm_prepare(pcm))) | |
998 | fatal(0, "error calling snd_pcm_prepare: %d", err); | |
999 | } else | |
1000 | break; | |
1001 | } while(retry-- > 0); | |
1002 | if(alsa_nslots >= 0) | |
1003 | fdno += alsa_nslots; | |
e83d0967 | 1004 | break; |
3a3c7bb9 | 1005 | } |
8023f60b | 1006 | #endif |
e83d0967 RK |
1007 | default: |
1008 | assert(!"unknown backend"); | |
9d5da576 | 1009 | } |
1010 | } | |
460b9539 | 1011 | /* If any other tracks don't have a full buffer, try to read sample data |
1012 | * from them. */ | |
1013 | for(t = tracks; t; t = t->next) | |
1014 | if(t != playing) { | |
1015 | if(!t->eof && t->used < t->size) { | |
9d5da576 | 1016 | t->slot = addfd(t->fd, POLLIN | POLLHUP); |
460b9539 | 1017 | } else |
1018 | t->slot = -1; | |
1019 | } | |
e83d0967 RK |
1020 | /* Wait for something interesting to happen */ |
1021 | n = poll(fds, fdno, timeout); | |
460b9539 | 1022 | if(n < 0) { |
1023 | if(errno == EINTR) continue; | |
1024 | fatal(errno, "error calling poll"); | |
1025 | } | |
1026 | /* Play some sound before doing anything else */ | |
e83d0967 RK |
1027 | poke = 0; |
1028 | switch(config->speaker_backend) { | |
8023f60b | 1029 | #if API_ALSA |
e83d0967 RK |
1030 | case BACKEND_ALSA: |
1031 | if(alsa_slots != -1) { | |
1032 | unsigned short alsa_revents; | |
1033 | ||
1034 | if((err = snd_pcm_poll_descriptors_revents(pcm, | |
1035 | &fds[alsa_slots], | |
1036 | alsa_nslots, | |
1037 | &alsa_revents)) < 0) | |
1038 | fatal(0, "error calling snd_pcm_poll_descriptors_revents: %d", err); | |
1039 | if(alsa_revents & (POLLOUT | POLLERR)) | |
1040 | play(3 * FRAMES); | |
1041 | } else | |
1042 | poke = 1; | |
1043 | break; | |
8023f60b | 1044 | #endif |
e83d0967 RK |
1045 | case BACKEND_COMMAND: |
1046 | if(cmdfd_slot != -1) { | |
1047 | if(fds[cmdfd_slot].revents & (POLLOUT | POLLERR)) | |
1048 | play(3 * FRAMES); | |
1049 | } else | |
1050 | poke = 1; | |
1051 | break; | |
1052 | case BACKEND_NETWORK: | |
1053 | if(bfd_slot != -1) { | |
1054 | if(fds[bfd_slot].revents & (POLLOUT | POLLERR)) | |
1055 | play(3 * FRAMES); | |
1056 | } else | |
1057 | poke = 1; | |
1058 | break; | |
1059 | } | |
1060 | if(poke) { | |
460b9539 | 1061 | /* Some attempt to play must have failed */ |
1062 | if(playing && !paused) | |
1063 | play(forceplay); | |
1064 | else | |
1065 | forceplay = 0; /* just in case */ | |
1066 | } | |
1067 | /* Perhaps we have a command to process */ | |
1068 | if(fds[stdin_slot].revents & POLLIN) { | |
1069 | n = speaker_recv(0, &sm, &fd); | |
1070 | if(n > 0) | |
1071 | switch(sm.type) { | |
1072 | case SM_PREPARE: | |
1073 | D(("SM_PREPARE %s %d", sm.id, fd)); | |
1074 | if(fd == -1) fatal(0, "got SM_PREPARE but no file descriptor"); | |
1075 | t = findtrack(sm.id, 1); | |
1076 | acquire(t, fd); | |
1077 | break; | |
1078 | case SM_PLAY: | |
1079 | D(("SM_PLAY %s %d", sm.id, fd)); | |
1080 | if(playing) fatal(0, "got SM_PLAY but already playing something"); | |
1081 | t = findtrack(sm.id, 1); | |
1082 | if(fd != -1) acquire(t, fd); | |
1083 | playing = t; | |
8023f60b | 1084 | play(bufsize); |
460b9539 | 1085 | report(); |
1086 | break; | |
1087 | case SM_PAUSE: | |
1088 | D(("SM_PAUSE")); | |
1089 | paused = 1; | |
1090 | report(); | |
1091 | break; | |
1092 | case SM_RESUME: | |
1093 | D(("SM_RESUME")); | |
1094 | if(paused) { | |
1095 | paused = 0; | |
1096 | if(playing) | |
8023f60b | 1097 | play(bufsize); |
460b9539 | 1098 | } |
1099 | report(); | |
1100 | break; | |
1101 | case SM_CANCEL: | |
1102 | D(("SM_CANCEL %s", sm.id)); | |
1103 | t = removetrack(sm.id); | |
1104 | if(t) { | |
1105 | if(t == playing) { | |
1106 | sm.type = SM_FINISHED; | |
1107 | strcpy(sm.id, playing->id); | |
1108 | speaker_send(1, &sm, 0); | |
1109 | playing = 0; | |
1110 | } | |
1111 | destroy(t); | |
1112 | } else | |
1113 | error(0, "SM_CANCEL for unknown track %s", sm.id); | |
1114 | report(); | |
1115 | break; | |
1116 | case SM_RELOAD: | |
1117 | D(("SM_RELOAD")); | |
1118 | if(config_read()) error(0, "cannot read configuration"); | |
1119 | info("reloaded configuration"); | |
1120 | break; | |
1121 | default: | |
1122 | error(0, "unknown message type %d", sm.type); | |
1123 | } | |
1124 | } | |
1125 | /* Read in any buffered data */ | |
1126 | for(t = tracks; t; t = t->next) | |
9d5da576 | 1127 | if(t->slot != -1 && (fds[t->slot].revents & (POLLIN | POLLHUP))) |
460b9539 | 1128 | fill(t); |
1129 | /* We might be able to play now */ | |
9d5da576 | 1130 | if(ready && forceplay && playing && !paused) |
460b9539 | 1131 | play(forceplay); |
1132 | /* Maybe we finished playing a track somewhere in the above */ | |
1133 | maybe_finished(); | |
1134 | /* If we don't need the sound device for now then close it for the benefit | |
1135 | * of anyone else who wants it. */ | |
9d5da576 | 1136 | if((!playing || paused) && ready) |
460b9539 | 1137 | idle(); |
1138 | /* If we've not reported out state for a second do so now. */ | |
1139 | if(time(0) > last_report) | |
1140 | report(); | |
1141 | } | |
1142 | info("stopped (parent terminated)"); | |
1143 | exit(0); | |
1144 | } | |
1145 | ||
1146 | /* | |
1147 | Local Variables: | |
1148 | c-basic-offset:2 | |
1149 | comment-column:40 | |
1150 | fill-column:79 | |
1151 | indent-tabs-mode:nil | |
1152 | End: | |
1153 | */ |