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[disorder] / server / speaker.c
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460b9539 1/*
2 * This file is part of DisOrder
dea8f8aa 3 * Copyright (C) 2005, 2006, 2007 Richard Kettlewell
460b9539 4 *
5 * This program is free software; you can redistribute it and/or modify
6 * it under the terms of the GNU General Public License as published by
7 * the Free Software Foundation; either version 2 of the License, or
8 * (at your option) any later version.
9 *
10 * This program is distributed in the hope that it will be useful, but
11 * WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * General Public License for more details.
14 *
15 * You should have received a copy of the GNU General Public License
16 * along with this program; if not, write to the Free Software
17 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
18 * USA
19 */
20
21/* This program deliberately does not use the garbage collector even though it
22 * might be convenient to do so. This is for two reasons. Firstly some libao
23 * drivers are implemented using threads and we do not want to have to deal
24 * with potential interactions between threading and garbage collection.
25 * Secondly this process needs to be able to respond quickly and this is not
26 * compatible with the collector hanging the program even relatively
27 * briefly. */
28
29#include <config.h>
30#include "types.h"
31
32#include <getopt.h>
33#include <stdio.h>
34#include <stdlib.h>
35#include <locale.h>
36#include <syslog.h>
37#include <unistd.h>
38#include <errno.h>
39#include <ao/ao.h>
40#include <string.h>
41#include <assert.h>
42#include <sys/select.h>
9d5da576 43#include <sys/wait.h>
460b9539 44#include <time.h>
8023f60b 45#include <fcntl.h>
46#include <poll.h>
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47#include <sys/socket.h>
48#include <netdb.h>
49#include <gcrypt.h>
50#include <sys/uio.h>
460b9539 51
52#include "configuration.h"
53#include "syscalls.h"
54#include "log.h"
55#include "defs.h"
56#include "mem.h"
57#include "speaker.h"
58#include "user.h"
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59#include "addr.h"
60#include "timeval.h"
61#include "rtp.h"
460b9539 62
8023f60b 63#if API_ALSA
dea8f8aa 64#include <alsa/asoundlib.h>
8023f60b 65#endif
dea8f8aa 66
5330d674 67#ifdef WORDS_BIGENDIAN
68# define MACHINE_AO_FMT AO_FMT_BIG
69#else
70# define MACHINE_AO_FMT AO_FMT_LITTLE
71#endif
72
460b9539 73#define BUFFER_SECONDS 5 /* How many seconds of input to
74 * buffer. */
75
76#define FRAMES 4096 /* Frame batch size */
77
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78#define NETWORK_BYTES 1024 /* Bytes to send per network packet */
79/* (don't make this too big or arithmetic will start to overflow) */
80
81#define RTP_AHEAD 2 /* Max RTP playahead (seconds) */
82
460b9539 83#define NFDS 256 /* Max FDs to poll for */
84
85/* Known tracks are kept in a linked list. We don't normally to have
86 * more than two - maybe three at the outside. */
87static struct track {
88 struct track *next; /* next track */
89 int fd; /* input FD */
90 char id[24]; /* ID */
91 size_t start, used; /* start + bytes used */
92 int eof; /* input is at EOF */
93 int got_format; /* got format yet? */
94 ao_sample_format format; /* sample format */
95 unsigned long long played; /* number of frames played */
96 char *buffer; /* sample buffer */
97 size_t size; /* sample buffer size */
98 int slot; /* poll array slot */
99} *tracks, *playing; /* all tracks + playing track */
100
101static time_t last_report; /* when we last reported */
102static int paused; /* pause status */
460b9539 103static ao_sample_format pcm_format; /* current format if aodev != 0 */
104static size_t bpf; /* bytes per frame */
105static struct pollfd fds[NFDS]; /* if we need more than that */
106static int fdno; /* fd number */
8023f60b 107static size_t bufsize; /* buffer size */
108#if API_ALSA
109static snd_pcm_t *pcm; /* current pcm handle */
0c207c37 110static snd_pcm_uframes_t last_pcm_bufsize; /* last seen buffer size */
8023f60b 111#endif
9d5da576 112static int ready; /* ready to send audio */
460b9539 113static int forceplay; /* frames to force play */
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114static int cmdfd = -1; /* child process input */
115static int bfd = -1; /* broadcast FD */
116static uint32_t rtp_time; /* RTP timestamp */
117static struct timeval rtp_time_real; /* corresponding real time */
118static uint16_t rtp_seq; /* frame sequence number */
119static uint32_t rtp_id; /* RTP SSRC */
120static int idled; /* set when idled */
121static int audio_errors; /* audio error counter */
460b9539 122
123static const struct option options[] = {
124 { "help", no_argument, 0, 'h' },
125 { "version", no_argument, 0, 'V' },
126 { "config", required_argument, 0, 'c' },
127 { "debug", no_argument, 0, 'd' },
128 { "no-debug", no_argument, 0, 'D' },
129 { 0, 0, 0, 0 }
130};
131
132/* Display usage message and terminate. */
133static void help(void) {
134 xprintf("Usage:\n"
135 " disorder-speaker [OPTIONS]\n"
136 "Options:\n"
137 " --help, -h Display usage message\n"
138 " --version, -V Display version number\n"
139 " --config PATH, -c PATH Set configuration file\n"
140 " --debug, -d Turn on debugging\n"
141 "\n"
142 "Speaker process for DisOrder. Not intended to be run\n"
143 "directly.\n");
144 xfclose(stdout);
145 exit(0);
146}
147
148/* Display version number and terminate. */
149static void version(void) {
150 xprintf("disorder-speaker version %s\n", disorder_version_string);
151 xfclose(stdout);
152 exit(0);
153}
154
155/* Return the number of bytes per frame in FORMAT. */
156static size_t bytes_per_frame(const ao_sample_format *format) {
157 return format->channels * format->bits / 8;
158}
159
160/* Find track ID, maybe creating it if not found. */
161static struct track *findtrack(const char *id, int create) {
162 struct track *t;
163
164 D(("findtrack %s %d", id, create));
165 for(t = tracks; t && strcmp(id, t->id); t = t->next)
166 ;
167 if(!t && create) {
168 t = xmalloc(sizeof *t);
169 t->next = tracks;
170 strcpy(t->id, id);
171 t->fd = -1;
172 tracks = t;
173 /* The initial input buffer will be the sample format. */
174 t->buffer = (void *)&t->format;
175 t->size = sizeof t->format;
176 }
177 return t;
178}
179
180/* Remove track ID (but do not destroy it). */
181static struct track *removetrack(const char *id) {
182 struct track *t, **tt;
183
184 D(("removetrack %s", id));
185 for(tt = &tracks; (t = *tt) && strcmp(id, t->id); tt = &t->next)
186 ;
187 if(t)
188 *tt = t->next;
189 return t;
190}
191
192/* Destroy a track. */
193static void destroy(struct track *t) {
194 D(("destroy %s", t->id));
195 if(t->fd != -1) xclose(t->fd);
196 if(t->buffer != (void *)&t->format) free(t->buffer);
197 free(t);
198}
199
200/* Notice a new FD. */
201static void acquire(struct track *t, int fd) {
202 D(("acquire %s %d", t->id, fd));
203 if(t->fd != -1)
204 xclose(t->fd);
205 t->fd = fd;
206 nonblock(fd);
207}
208
209/* Read data into a sample buffer. Return 0 on success, -1 on EOF. */
210static int fill(struct track *t) {
211 size_t where, left;
212 int n;
213
214 D(("fill %s: eof=%d used=%zu size=%zu got_format=%d",
215 t->id, t->eof, t->used, t->size, t->got_format));
216 if(t->eof) return -1;
217 if(t->used < t->size) {
218 /* there is room left in the buffer */
219 where = (t->start + t->used) % t->size;
220 if(t->got_format) {
221 /* We are reading audio data, get as much as we can */
222 if(where >= t->start) left = t->size - where;
223 else left = t->start - where;
224 } else
225 /* We are still waiting for the format, only get that */
226 left = sizeof (ao_sample_format) - t->used;
227 do {
228 n = read(t->fd, t->buffer + where, left);
229 } while(n < 0 && errno == EINTR);
230 if(n < 0) {
231 if(errno != EAGAIN) fatal(errno, "error reading sample stream");
232 return 0;
233 }
234 if(n == 0) {
235 D(("fill %s: eof detected", t->id));
236 t->eof = 1;
237 return -1;
238 }
239 t->used += n;
240 if(!t->got_format && t->used >= sizeof (ao_sample_format)) {
241 assert(t->used == sizeof (ao_sample_format));
242 /* Check that our assumptions are met. */
243 if(t->format.bits & 7)
244 fatal(0, "bits per sample not a multiple of 8");
245 /* Make a new buffer for audio data. */
246 t->size = bytes_per_frame(&t->format) * t->format.rate * BUFFER_SECONDS;
247 t->buffer = xmalloc(t->size);
248 t->used = 0;
249 t->got_format = 1;
250 D(("got format for %s", t->id));
251 }
252 }
253 return 0;
254}
255
256/* Return true if A and B denote identical libao formats, else false. */
257static int formats_equal(const ao_sample_format *a,
258 const ao_sample_format *b) {
259 return (a->bits == b->bits
260 && a->rate == b->rate
261 && a->channels == b->channels
262 && a->byte_format == b->byte_format);
263}
264
265/* Close the sound device. */
266static void idle(void) {
460b9539 267 D(("idle"));
8023f60b 268#if API_ALSA
e83d0967 269 if(config->speaker_backend == BACKEND_ALSA && pcm) {
8023f60b 270 int err;
271
460b9539 272 if((err = snd_pcm_nonblock(pcm, 0)) < 0)
273 fatal(0, "error calling snd_pcm_nonblock: %d", err);
274 D(("draining pcm"));
275 snd_pcm_drain(pcm);
276 D(("closing pcm"));
277 snd_pcm_close(pcm);
278 pcm = 0;
279 forceplay = 0;
280 D(("released audio device"));
281 }
8023f60b 282#endif
e83d0967 283 idled = 1;
9d5da576 284 ready = 0;
460b9539 285}
286
287/* Abandon the current track */
288static void abandon(void) {
289 struct speaker_message sm;
290
291 D(("abandon"));
292 memset(&sm, 0, sizeof sm);
293 sm.type = SM_FINISHED;
294 strcpy(sm.id, playing->id);
295 speaker_send(1, &sm, 0);
296 removetrack(playing->id);
297 destroy(playing);
298 playing = 0;
299 forceplay = 0;
300}
301
8023f60b 302#if API_ALSA
1c6e6a61 303static void log_params(snd_pcm_hw_params_t *hwparams,
304 snd_pcm_sw_params_t *swparams) {
305 snd_pcm_uframes_t f;
306 unsigned u;
307
0c207c37 308 return; /* too verbose */
1c6e6a61 309 if(hwparams) {
310 /* TODO */
311 }
312 if(swparams) {
313 snd_pcm_sw_params_get_silence_size(swparams, &f);
314 info("sw silence_size=%lu", (unsigned long)f);
315 snd_pcm_sw_params_get_silence_threshold(swparams, &f);
316 info("sw silence_threshold=%lu", (unsigned long)f);
317 snd_pcm_sw_params_get_sleep_min(swparams, &u);
318 info("sw sleep_min=%lu", (unsigned long)u);
319 snd_pcm_sw_params_get_start_threshold(swparams, &f);
320 info("sw start_threshold=%lu", (unsigned long)f);
321 snd_pcm_sw_params_get_stop_threshold(swparams, &f);
322 info("sw stop_threshold=%lu", (unsigned long)f);
323 snd_pcm_sw_params_get_xfer_align(swparams, &f);
324 info("sw xfer_align=%lu", (unsigned long)f);
325 }
326}
8023f60b 327#endif
1c6e6a61 328
5330d674 329static void soxargs(const char ***pp, char **qq, ao_sample_format *ao) {
9d5da576 330 int n;
331
332 *(*pp)++ = "-t.raw";
333 *(*pp)++ = "-s";
334 *(*pp)++ = *qq; n = sprintf(*qq, "-r%d", ao->rate); *qq += n + 1;
5330d674 335 *(*pp)++ = *qq; n = sprintf(*qq, "-c%d", ao->channels); *qq += n + 1;
336 /* sox 12.17.9 insists on -b etc; CVS sox insists on -<n> etc; both are
337 * deployed! */
338 switch(config->sox_generation) {
339 case 0:
340 if(ao->bits != 8
341 && ao->byte_format != AO_FMT_NATIVE
342 && ao->byte_format != MACHINE_AO_FMT) {
343 *(*pp)++ = "-x";
344 }
345 switch(ao->bits) {
346 case 8: *(*pp)++ = "-b"; break;
347 case 16: *(*pp)++ = "-w"; break;
348 case 32: *(*pp)++ = "-l"; break;
349 case 64: *(*pp)++ = "-d"; break;
350 default: fatal(0, "cannot handle sample size %d", (int)ao->bits);
351 }
352 break;
353 case 1:
354 switch(ao->byte_format) {
9d5da576 355 case AO_FMT_NATIVE: break;
27801653 356 case AO_FMT_BIG: *(*pp)++ = "-B"; break;
357 case AO_FMT_LITTLE: *(*pp)++ = "-L"; break;
5330d674 358 }
359 *(*pp)++ = *qq; n = sprintf(*qq, "-%d", ao->bits/8); *qq += n + 1;
360 break;
9d5da576 361 }
9d5da576 362}
363
460b9539 364/* Make sure the sound device is open and has the right sample format. Return
365 * 0 on success and -1 on error. */
366static int activate(void) {
460b9539 367 /* If we don't know the format yet we cannot start. */
368 if(!playing->got_format) {
369 D((" - not got format for %s", playing->id));
370 return -1;
371 }
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RK
372 switch(config->speaker_backend) {
373 case BACKEND_COMMAND:
374 case BACKEND_NETWORK:
375 /* If we pass audio on to some other agent then we enforce the configured
376 * sample format on the *inbound* audio data. */
9d5da576 377 if(!formats_equal(&playing->format, &config->sample_format)) {
378 char argbuf[1024], *q = argbuf;
379 const char *av[18], **pp = av;
380 int soxpipe[2];
381 pid_t soxkid;
382 *pp++ = "sox";
383 soxargs(&pp, &q, &playing->format);
384 *pp++ = "-";
385 soxargs(&pp, &q, &config->sample_format);
386 *pp++ = "-";
387 *pp++ = 0;
388 if(debugging) {
389 for(pp = av; *pp; pp++)
390 D(("sox arg[%d] = %s", pp - av, *pp));
391 D(("end args"));
392 }
393 xpipe(soxpipe);
394 soxkid = xfork();
395 if(soxkid == 0) {
396 xdup2(playing->fd, 0);
397 xdup2(soxpipe[1], 1);
398 fcntl(0, F_SETFL, fcntl(0, F_GETFL) & ~O_NONBLOCK);
399 close(soxpipe[0]);
400 close(soxpipe[1]);
401 close(playing->fd);
402 execvp("sox", (char **)av);
403 _exit(1);
404 }
405 D(("forking sox for format conversion (kid = %d)", soxkid));
406 close(playing->fd);
407 close(soxpipe[1]);
408 playing->fd = soxpipe[0];
409 playing->format = config->sample_format;
410 ready = 0;
411 }
412 if(!ready) {
413 pcm_format = config->sample_format;
8023f60b 414 bufsize = 3 * FRAMES;
9d5da576 415 bpf = bytes_per_frame(&config->sample_format);
416 D(("acquired audio device"));
417 ready = 1;
418 }
419 return 0;
e83d0967 420 case BACKEND_ALSA:
8023f60b 421#if API_ALSA
e83d0967
RK
422 /* If we need to change format then close the current device. */
423 if(pcm && !formats_equal(&playing->format, &pcm_format))
424 idle();
425 if(!pcm) {
426 snd_pcm_hw_params_t *hwparams;
427 snd_pcm_sw_params_t *swparams;
428 snd_pcm_uframes_t pcm_bufsize;
429 int err;
430 int sample_format = 0;
431 unsigned rate;
432
433 D(("snd_pcm_open"));
434 if((err = snd_pcm_open(&pcm,
435 config->device,
436 SND_PCM_STREAM_PLAYBACK,
437 SND_PCM_NONBLOCK))) {
438 error(0, "error from snd_pcm_open: %d", err);
439 goto error;
440 }
441 snd_pcm_hw_params_alloca(&hwparams);
442 D(("set up hw params"));
443 if((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0)
444 fatal(0, "error from snd_pcm_hw_params_any: %d", err);
445 if((err = snd_pcm_hw_params_set_access(pcm, hwparams,
446 SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
447 fatal(0, "error from snd_pcm_hw_params_set_access: %d", err);
448 switch(playing->format.bits) {
449 case 8:
450 sample_format = SND_PCM_FORMAT_S8;
451 break;
452 case 16:
453 switch(playing->format.byte_format) {
454 case AO_FMT_NATIVE: sample_format = SND_PCM_FORMAT_S16; break;
455 case AO_FMT_LITTLE: sample_format = SND_PCM_FORMAT_S16_LE; break;
456 case AO_FMT_BIG: sample_format = SND_PCM_FORMAT_S16_BE; break;
457 error(0, "unrecognized byte format %d", playing->format.byte_format);
458 goto fatal;
459 }
460 break;
461 default:
462 error(0, "unsupported sample size %d", playing->format.bits);
460b9539 463 goto fatal;
464 }
e83d0967
RK
465 if((err = snd_pcm_hw_params_set_format(pcm, hwparams,
466 sample_format)) < 0) {
467 error(0, "error from snd_pcm_hw_params_set_format (%d): %d",
468 sample_format, err);
469 goto fatal;
470 }
471 rate = playing->format.rate;
472 if((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0) {
473 error(0, "error from snd_pcm_hw_params_set_rate (%d): %d",
474 playing->format.rate, err);
475 goto fatal;
476 }
477 if(rate != (unsigned)playing->format.rate)
478 info("want rate %d, got %u", playing->format.rate, rate);
479 if((err = snd_pcm_hw_params_set_channels(pcm, hwparams,
480 playing->format.channels)) < 0) {
481 error(0, "error from snd_pcm_hw_params_set_channels (%d): %d",
482 playing->format.channels, err);
483 goto fatal;
484 }
485 bufsize = 3 * FRAMES;
486 pcm_bufsize = bufsize;
487 if((err = snd_pcm_hw_params_set_buffer_size_near(pcm, hwparams,
488 &pcm_bufsize)) < 0)
489 fatal(0, "error from snd_pcm_hw_params_set_buffer_size (%d): %d",
490 3 * FRAMES, err);
491 if(pcm_bufsize != 3 * FRAMES && pcm_bufsize != last_pcm_bufsize)
492 info("asked for PCM buffer of %d frames, got %d",
493 3 * FRAMES, (int)pcm_bufsize);
494 last_pcm_bufsize = pcm_bufsize;
495 if((err = snd_pcm_hw_params(pcm, hwparams)) < 0)
496 fatal(0, "error calling snd_pcm_hw_params: %d", err);
497 D(("set up sw params"));
498 snd_pcm_sw_params_alloca(&swparams);
499 if((err = snd_pcm_sw_params_current(pcm, swparams)) < 0)
500 fatal(0, "error calling snd_pcm_sw_params_current: %d", err);
501 if((err = snd_pcm_sw_params_set_avail_min(pcm, swparams, FRAMES)) < 0)
502 fatal(0, "error calling snd_pcm_sw_params_set_avail_min %d: %d",
503 FRAMES, err);
504 if((err = snd_pcm_sw_params(pcm, swparams)) < 0)
505 fatal(0, "error calling snd_pcm_sw_params: %d", err);
506 pcm_format = playing->format;
507 bpf = bytes_per_frame(&pcm_format);
508 D(("acquired audio device"));
509 log_params(hwparams, swparams);
510 ready = 1;
460b9539 511 }
e83d0967
RK
512 return 0;
513 fatal:
514 abandon();
515 error:
516 /* We assume the error is temporary and that we'll retry in a bit. */
517 if(pcm) {
518 snd_pcm_close(pcm);
519 pcm = 0;
460b9539 520 }
e83d0967 521 return -1;
8023f60b 522#endif
e83d0967
RK
523 default:
524 assert(!"reached");
525 }
460b9539 526}
527
528/* Check to see whether the current track has finished playing */
529static void maybe_finished(void) {
530 if(playing
531 && playing->eof
532 && (!playing->got_format
533 || playing->used < bytes_per_frame(&playing->format)))
534 abandon();
535}
536
e83d0967
RK
537static void fork_cmd(void) {
538 pid_t cmdpid;
9d5da576 539 int pfd[2];
e83d0967 540 if(cmdfd != -1) close(cmdfd);
9d5da576 541 xpipe(pfd);
e83d0967
RK
542 cmdpid = xfork();
543 if(!cmdpid) {
9d5da576 544 xdup2(pfd[0], 0);
545 close(pfd[0]);
546 close(pfd[1]);
547 execl("/bin/sh", "sh", "-c", config->speaker_command, (char *)0);
548 fatal(errno, "error execing /bin/sh");
549 }
550 close(pfd[0]);
e83d0967
RK
551 cmdfd = pfd[1];
552 D(("forked cmd %d, fd = %d", cmdpid, cmdfd));
9d5da576 553}
554
460b9539 555static void play(size_t frames) {
8023f60b 556 size_t avail_bytes, written_frames;
9d5da576 557 ssize_t written_bytes;
e83d0967
RK
558 struct rtp header;
559 struct iovec vec[2];
460b9539 560
561 if(activate()) {
562 if(playing)
563 forceplay = frames;
564 else
565 forceplay = 0; /* Must have called abandon() */
566 return;
567 }
568 D(("play: play %zu/%zu%s %dHz %db %dc", frames, playing->used / bpf,
569 playing->eof ? " EOF" : "",
570 playing->format.rate,
571 playing->format.bits,
572 playing->format.channels));
573 /* If we haven't got enough bytes yet wait until we have. Exception: when
574 * we are at eof. */
575 if(playing->used < frames * bpf && !playing->eof) {
576 forceplay = frames;
577 return;
578 }
579 /* We have got enough data so don't force play again */
580 forceplay = 0;
581 /* Figure out how many frames there are available to write */
582 if(playing->start + playing->used > playing->size)
583 avail_bytes = playing->size - playing->start;
584 else
585 avail_bytes = playing->used;
9d5da576 586
e83d0967 587 switch(config->speaker_backend) {
8023f60b 588#if API_ALSA
3a3c7bb9 589 case BACKEND_ALSA: {
8023f60b 590 snd_pcm_sframes_t pcm_written_frames;
591 size_t avail_frames;
592 int err;
593
9d5da576 594 avail_frames = avail_bytes / bpf;
595 if(avail_frames > frames)
596 avail_frames = frames;
597 if(!avail_frames)
460b9539 598 return;
8023f60b 599 pcm_written_frames = snd_pcm_writei(pcm,
600 playing->buffer + playing->start,
601 avail_frames);
9d5da576 602 D(("actually play %zu frames, wrote %d",
8023f60b 603 avail_frames, (int)pcm_written_frames));
604 if(pcm_written_frames < 0) {
605 switch(pcm_written_frames) {
9d5da576 606 case -EPIPE: /* underrun */
607 error(0, "snd_pcm_writei reports underrun");
608 if((err = snd_pcm_prepare(pcm)) < 0)
609 fatal(0, "error calling snd_pcm_prepare: %d", err);
610 return;
611 case -EAGAIN:
612 return;
613 default:
8023f60b 614 fatal(0, "error calling snd_pcm_writei: %d",
615 (int)pcm_written_frames);
9d5da576 616 }
617 }
8023f60b 618 written_frames = pcm_written_frames;
9d5da576 619 written_bytes = written_frames * bpf;
e83d0967 620 break;
3a3c7bb9 621 }
8023f60b 622#endif
e83d0967 623 case BACKEND_COMMAND:
9d5da576 624 if(avail_bytes > frames * bpf)
625 avail_bytes = frames * bpf;
e83d0967 626 written_bytes = write(cmdfd, playing->buffer + playing->start,
9d5da576 627 avail_bytes);
628 D(("actually play %zu bytes, wrote %d",
629 avail_bytes, (int)written_bytes));
630 if(written_bytes < 0) {
631 switch(errno) {
632 case EPIPE:
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633 error(0, "hmm, command died; trying another");
634 fork_cmd();
9d5da576 635 return;
636 case EAGAIN:
637 return;
638 }
460b9539 639 }
9d5da576 640 written_frames = written_bytes / bpf; /* good enough */
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641 break;
642 case BACKEND_NETWORK:
643 /* We transmit using RTP (RFC3550) and attempt to conform to the internet
644 * AVT profile (RFC3551). */
645 if(rtp_time_real.tv_sec == 0)
646 xgettimeofday(&rtp_time_real, 0);
647 if(idled) {
648 struct timeval now;
649 xgettimeofday(&now, 0);
650 /* There's been a gap. Fix up the RTP time accordingly. */
651 rtp_time += (((now.tv_sec + now.tv_usec /1000000.0)
652 - (rtp_time_real.tv_sec + rtp_time_real.tv_usec / 1000000.0))
653 * playing->format.rate * playing->format.channels);
654 }
655 header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */
656 header.seq = htons(rtp_seq++);
657 header.timestamp = htonl(rtp_time);
658 header.ssrc = rtp_id;
659 header.mpt = (idled ? 0x80 : 0x00) | 10;
660 /* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from
661 * the sample rate (in a library somewhere so that configuration.c can rule
662 * out invalid rates).
663 */
664 idled = 0;
665 if(avail_bytes > NETWORK_BYTES - sizeof header) {
666 avail_bytes = NETWORK_BYTES - sizeof header;
667 avail_bytes -= avail_bytes % bpf;
668 }
669 /* "The RTP clock rate used for generating the RTP timestamp is independent
670 * of the number of channels and the encoding; it equals the number of
671 * sampling periods per second. For N-channel encodings, each sampling
672 * period (say, 1/8000 of a second) generates N samples. (This terminology
673 * is standard, but somewhat confusing, as the total number of samples
674 * generated per second is then the sampling rate times the channel
675 * count.)"
676 */
677 vec[0].iov_base = (void *)&header;
678 vec[0].iov_len = sizeof header;
679 vec[1].iov_base = playing->buffer + playing->start;
680 vec[1].iov_len = avail_bytes;
681#if 0
682 {
683 char buffer[3 * sizeof header + 1];
684 size_t n;
685 const uint8_t *ptr = (void *)&header;
686
687 for(n = 0; n < sizeof header; ++n)
688 sprintf(&buffer[3 * n], "%02x ", *ptr++);
689 info(buffer);
690 }
691#endif
692 do {
693 written_bytes = writev(bfd,
694 vec,
695 2);
696 } while(written_bytes < 0 && errno == EINTR);
697 if(written_bytes < 0) {
698 error(errno, "error transmitting audio data");
699 ++audio_errors;
700 if(audio_errors == 10)
701 fatal(0, "too many audio errors");
702 return;
703 }
704 audio_errors /= 2;
705 written_bytes = avail_bytes;
706 written_frames = written_bytes / bpf;
707 /* Advance RTP's notion of the time */
708 rtp_time += written_frames * playing->format.channels;
709 /* Advance the corresponding real time */
710 assert(NETWORK_BYTES <= 2000); /* else risk overflowing 32 bits */
711 rtp_time_real.tv_usec += written_frames * 1000000 / playing->format.rate;
712 if(rtp_time_real.tv_usec >= 1000000) {
713 ++rtp_time_real.tv_sec;
714 rtp_time_real.tv_usec -= 1000000;
715 }
716 break;
717 default:
718 assert(!"reached");
460b9539 719 }
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720 /* written_bytes and written_frames had better both be set and correct by
721 * this point */
460b9539 722 playing->start += written_bytes;
723 playing->used -= written_bytes;
724 playing->played += written_frames;
725 /* If the pointer is at the end of the buffer (or the buffer is completely
726 * empty) wrap it back to the start. */
727 if(!playing->used || playing->start == playing->size)
728 playing->start = 0;
729 frames -= written_frames;
730}
731
732/* Notify the server what we're up to. */
733static void report(void) {
734 struct speaker_message sm;
735
736 if(playing && playing->buffer != (void *)&playing->format) {
737 memset(&sm, 0, sizeof sm);
738 sm.type = paused ? SM_PAUSED : SM_PLAYING;
739 strcpy(sm.id, playing->id);
740 sm.data = playing->played / playing->format.rate;
741 speaker_send(1, &sm, 0);
742 }
743 time(&last_report);
744}
745
9d5da576 746static void reap(int __attribute__((unused)) sig) {
e83d0967 747 pid_t cmdpid;
9d5da576 748 int st;
749
750 do
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751 cmdpid = waitpid(-1, &st, WNOHANG);
752 while(cmdpid > 0);
9d5da576 753 signal(SIGCHLD, reap);
754}
755
460b9539 756static int addfd(int fd, int events) {
757 if(fdno < NFDS) {
758 fds[fdno].fd = fd;
759 fds[fdno].events = events;
760 return fdno++;
761 } else
762 return -1;
763}
764
765int main(int argc, char **argv) {
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766 int n, fd, stdin_slot, alsa_slots, cmdfd_slot, bfd_slot, poke, timeout;
767 struct timeval now, delta;
460b9539 768 struct track *t;
769 struct speaker_message sm;
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770 struct addrinfo *res, *sres;
771 static const struct addrinfo pref = {
772 0,
773 PF_INET,
774 SOCK_DGRAM,
775 IPPROTO_UDP,
776 0,
777 0,
778 0,
779 0
780 };
781 static const struct addrinfo prefbind = {
782 AI_PASSIVE,
783 PF_INET,
784 SOCK_DGRAM,
785 IPPROTO_UDP,
786 0,
787 0,
788 0,
789 0
790 };
791 static const int one = 1;
792 char *sockname, *ssockname;
8023f60b 793#if API_ALSA
794 int alsa_nslots = -1, err;
795#endif
460b9539 796
797 set_progname(argv);
460b9539 798 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
799 while((n = getopt_long(argc, argv, "hVc:dD", options, 0)) >= 0) {
800 switch(n) {
801 case 'h': help();
802 case 'V': version();
803 case 'c': configfile = optarg; break;
804 case 'd': debugging = 1; break;
805 case 'D': debugging = 0; break;
806 default: fatal(0, "invalid option");
807 }
808 }
809 if(getenv("DISORDER_DEBUG_SPEAKER")) debugging = 1;
810 /* If stderr is a TTY then log there, otherwise to syslog. */
811 if(!isatty(2)) {
812 openlog(progname, LOG_PID, LOG_DAEMON);
813 log_default = &log_syslog;
814 }
815 if(config_read()) fatal(0, "cannot read configuration");
816 /* ignore SIGPIPE */
817 signal(SIGPIPE, SIG_IGN);
9d5da576 818 /* reap kids */
819 signal(SIGCHLD, reap);
460b9539 820 /* set nice value */
821 xnice(config->nice_speaker);
822 /* change user */
823 become_mortal();
824 /* make sure we're not root, whatever the config says */
825 if(getuid() == 0 || geteuid() == 0) fatal(0, "do not run as root");
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826 switch(config->speaker_backend) {
827 case BACKEND_ALSA:
828 info("selected ALSA backend");
829 case BACKEND_COMMAND:
830 info("selected command backend");
831 fork_cmd();
832 break;
833 case BACKEND_NETWORK:
834 res = get_address(&config->broadcast, &pref, &sockname);
835 if(!res) return -1;
836 if(config->broadcast_from.n) {
837 sres = get_address(&config->broadcast_from, &prefbind, &ssockname);
838 if(!sres) return -1;
839 } else
840 sres = 0;
841 if((bfd = socket(res->ai_family,
842 res->ai_socktype,
843 res->ai_protocol)) < 0)
844 fatal(errno, "error creating broadcast socket");
845 if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0)
846 fatal(errno, "error settting SO_BROADCAST on broadcast socket");
847 /* We might well want to set additional broadcast- or multicast-related
848 * options here */
849 if(sres && bind(bfd, sres->ai_addr, sres->ai_addrlen) < 0)
850 fatal(errno, "error binding broadcast socket to %s", ssockname);
851 if(connect(bfd, res->ai_addr, res->ai_addrlen) < 0)
852 fatal(errno, "error connecting broadcast socket to %s", sockname);
853 /* Select an SSRC */
854 gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM);
855 info("selected network backend, sending to %s", sockname);
856 if(config->sample_format.byte_format != AO_FMT_BIG) {
857 info("forcing big-endian sample format");
858 config->sample_format.byte_format = AO_FMT_BIG;
859 }
860 break;
861 default:
862 fatal(0, "unknown backend %d", config->speaker_backend);
8023f60b 863 }
460b9539 864 while(getppid() != 1) {
865 fdno = 0;
866 /* Always ready for commands from the main server. */
867 stdin_slot = addfd(0, POLLIN);
868 /* Try to read sample data for the currently playing track if there is
869 * buffer space. */
870 if(playing && !playing->eof && playing->used < playing->size) {
871 playing->slot = addfd(playing->fd, POLLIN);
872 } else if(playing)
873 playing->slot = -1;
874 /* If forceplay is set then wait until it succeeds before waiting on the
875 * sound device. */
9d5da576 876 alsa_slots = -1;
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877 cmdfd_slot = -1;
878 bfd_slot = -1;
879 /* By default we will wait up to a second before thinking about current
880 * state. */
881 timeout = 1000;
8023f60b 882 if(ready && !forceplay) {
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883 switch(config->speaker_backend) {
884 case BACKEND_COMMAND:
885 /* We send sample data to the subprocess as fast as it can accept it.
886 * This isn't ideal as pause latency can be very high as a result. */
887 if(cmdfd >= 0)
888 cmdfd_slot = addfd(cmdfd, POLLOUT);
889 break;
890 case BACKEND_NETWORK:
891 /* We want to keep the notional playing point somewhere in the near
892 * future. If it's too near then clients that attempt even the
893 * slightest amount of read-ahead will never catch up, and those that
894 * don't will skip whenever there's a trivial network delay. If it's
895 * too far ahead then pause latency will be too high.
896 */
897 xgettimeofday(&now, 0);
898 delta = tvsub(rtp_time_real, now);
899 if(delta.tv_sec < RTP_AHEAD) {
900 D(("delta = %ld.%06ld", (long)delta.tv_sec, (long)delta.tv_usec));
901 bfd_slot = addfd(bfd, POLLOUT);
902 if(delta.tv_sec < 0)
903 rtp_time_real = now; /* catch up */
904 }
905 break;
8023f60b 906#if API_ALSA
3a3c7bb9 907 case BACKEND_ALSA: {
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908 /* We send sample data to ALSA as fast as it can accept it, relying on
909 * the fact that it has a relatively small buffer to minimize pause
910 * latency. */
9d5da576 911 int retry = 3;
912
913 alsa_slots = fdno;
914 do {
915 retry = 0;
916 alsa_nslots = snd_pcm_poll_descriptors(pcm, &fds[fdno], NFDS - fdno);
917 if((alsa_nslots <= 0
918 || !(fds[alsa_slots].events & POLLOUT))
919 && snd_pcm_state(pcm) == SND_PCM_STATE_XRUN) {
920 error(0, "underrun detected after call to snd_pcm_poll_descriptors()");
921 if((err = snd_pcm_prepare(pcm)))
922 fatal(0, "error calling snd_pcm_prepare: %d", err);
923 } else
924 break;
925 } while(retry-- > 0);
926 if(alsa_nslots >= 0)
927 fdno += alsa_nslots;
e83d0967 928 break;
3a3c7bb9 929 }
8023f60b 930#endif
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931 default:
932 assert(!"unknown backend");
9d5da576 933 }
934 }
460b9539 935 /* If any other tracks don't have a full buffer, try to read sample data
936 * from them. */
937 for(t = tracks; t; t = t->next)
938 if(t != playing) {
939 if(!t->eof && t->used < t->size) {
9d5da576 940 t->slot = addfd(t->fd, POLLIN | POLLHUP);
460b9539 941 } else
942 t->slot = -1;
943 }
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944 /* Wait for something interesting to happen */
945 n = poll(fds, fdno, timeout);
460b9539 946 if(n < 0) {
947 if(errno == EINTR) continue;
948 fatal(errno, "error calling poll");
949 }
950 /* Play some sound before doing anything else */
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951 poke = 0;
952 switch(config->speaker_backend) {
8023f60b 953#if API_ALSA
e83d0967
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954 case BACKEND_ALSA:
955 if(alsa_slots != -1) {
956 unsigned short alsa_revents;
957
958 if((err = snd_pcm_poll_descriptors_revents(pcm,
959 &fds[alsa_slots],
960 alsa_nslots,
961 &alsa_revents)) < 0)
962 fatal(0, "error calling snd_pcm_poll_descriptors_revents: %d", err);
963 if(alsa_revents & (POLLOUT | POLLERR))
964 play(3 * FRAMES);
965 } else
966 poke = 1;
967 break;
8023f60b 968#endif
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969 case BACKEND_COMMAND:
970 if(cmdfd_slot != -1) {
971 if(fds[cmdfd_slot].revents & (POLLOUT | POLLERR))
972 play(3 * FRAMES);
973 } else
974 poke = 1;
975 break;
976 case BACKEND_NETWORK:
977 if(bfd_slot != -1) {
978 if(fds[bfd_slot].revents & (POLLOUT | POLLERR))
979 play(3 * FRAMES);
980 } else
981 poke = 1;
982 break;
983 }
984 if(poke) {
460b9539 985 /* Some attempt to play must have failed */
986 if(playing && !paused)
987 play(forceplay);
988 else
989 forceplay = 0; /* just in case */
990 }
991 /* Perhaps we have a command to process */
992 if(fds[stdin_slot].revents & POLLIN) {
993 n = speaker_recv(0, &sm, &fd);
994 if(n > 0)
995 switch(sm.type) {
996 case SM_PREPARE:
997 D(("SM_PREPARE %s %d", sm.id, fd));
998 if(fd == -1) fatal(0, "got SM_PREPARE but no file descriptor");
999 t = findtrack(sm.id, 1);
1000 acquire(t, fd);
1001 break;
1002 case SM_PLAY:
1003 D(("SM_PLAY %s %d", sm.id, fd));
1004 if(playing) fatal(0, "got SM_PLAY but already playing something");
1005 t = findtrack(sm.id, 1);
1006 if(fd != -1) acquire(t, fd);
1007 playing = t;
8023f60b 1008 play(bufsize);
460b9539 1009 report();
1010 break;
1011 case SM_PAUSE:
1012 D(("SM_PAUSE"));
1013 paused = 1;
1014 report();
1015 break;
1016 case SM_RESUME:
1017 D(("SM_RESUME"));
1018 if(paused) {
1019 paused = 0;
1020 if(playing)
8023f60b 1021 play(bufsize);
460b9539 1022 }
1023 report();
1024 break;
1025 case SM_CANCEL:
1026 D(("SM_CANCEL %s", sm.id));
1027 t = removetrack(sm.id);
1028 if(t) {
1029 if(t == playing) {
1030 sm.type = SM_FINISHED;
1031 strcpy(sm.id, playing->id);
1032 speaker_send(1, &sm, 0);
1033 playing = 0;
1034 }
1035 destroy(t);
1036 } else
1037 error(0, "SM_CANCEL for unknown track %s", sm.id);
1038 report();
1039 break;
1040 case SM_RELOAD:
1041 D(("SM_RELOAD"));
1042 if(config_read()) error(0, "cannot read configuration");
1043 info("reloaded configuration");
1044 break;
1045 default:
1046 error(0, "unknown message type %d", sm.type);
1047 }
1048 }
1049 /* Read in any buffered data */
1050 for(t = tracks; t; t = t->next)
9d5da576 1051 if(t->slot != -1 && (fds[t->slot].revents & (POLLIN | POLLHUP)))
460b9539 1052 fill(t);
1053 /* We might be able to play now */
9d5da576 1054 if(ready && forceplay && playing && !paused)
460b9539 1055 play(forceplay);
1056 /* Maybe we finished playing a track somewhere in the above */
1057 maybe_finished();
1058 /* If we don't need the sound device for now then close it for the benefit
1059 * of anyone else who wants it. */
9d5da576 1060 if((!playing || paused) && ready)
460b9539 1061 idle();
1062 /* If we've not reported out state for a second do so now. */
1063 if(time(0) > last_report)
1064 report();
1065 }
1066 info("stopped (parent terminated)");
1067 exit(0);
1068}
1069
1070/*
1071Local Variables:
1072c-basic-offset:2
1073comment-column:40
1074fill-column:79
1075indent-tabs-mode:nil
1076End:
1077*/