chiark / gitweb /
ensure tests avoid user config!
[disorder] / clients / playrtp.c
CommitLineData
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1/*
2 * This file is part of DisOrder.
3 * Copyright (C) 2007 Richard Kettlewell
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 */
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20/** @file clients/playrtp.c
21 * @brief RTP player
22 *
b0fdc63d 23 * This player supports Linux (<a href="http://www.alsa-project.org/">ALSA</a>)
24 * and Apple Mac (<a
25 * href="http://developer.apple.com/audio/coreaudio.html">Core Audio</a>)
26 * systems. There is no support for Microsoft Windows yet, and that will in
27 * fact probably an entirely separate program.
28 *
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29 * The program runs (at least) three threads. listen_thread() is responsible
30 * for reading RTP packets off the wire and adding them to the linked list @ref
31 * received_packets, assuming they are basically sound. queue_thread() takes
32 * packets off this linked list and adds them to @ref packets (an operation
33 * which might be much slower due to contention for @ref lock).
b0fdc63d 34 *
35 * The main thread is responsible for actually playing audio. In ALSA this
36 * means it waits until ALSA says it's ready for more audio which it then
c593cf7c 37 * plays. See @ref clients/playrtp-alsa.c.
b0fdc63d 38 *
8e3fe3d8 39 * In Core Audio the main thread is only responsible for starting and stopping
b0fdc63d 40 * play: the system does the actual playback in its own private thread, and
c593cf7c 41 * calls adioproc() to fetch the audio data. See @ref
42 * clients/playrtp-coreaudio.c.
b0fdc63d 43 *
44 * Sometimes it happens that there is no audio available to play. This may
45 * because the server went away, or a packet was dropped, or the server
46 * deliberately did not send any sound because it encountered a silence.
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47 *
48 * Assumptions:
49 * - it is safe to read uint32_t values without a lock protecting them
28bacdc0 50 */
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51
52#include <config.h>
53#include "types.h"
54
55#include <getopt.h>
56#include <stdio.h>
57#include <stdlib.h>
58#include <sys/socket.h>
59#include <sys/types.h>
60#include <sys/socket.h>
61#include <netdb.h>
62#include <pthread.h>
0b75463f 63#include <locale.h>
2c7c9eae 64#include <sys/uio.h>
28bacdc0 65#include <string.h>
8e3fe3d8 66#include <assert.h>
c593cf7c 67#include <errno.h>
e3426f7b 68#include <netinet/in.h>
2d2effe2 69#include <sys/time.h>
a99c4e9a 70#include <sys/un.h>
9fbe0996 71#include <unistd.h>
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72#include <sys/mman.h>
73#include <fcntl.h>
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74
75#include "log.h"
76#include "mem.h"
77#include "configuration.h"
78#include "addr.h"
79#include "syscalls.h"
80#include "rtp.h"
0b75463f 81#include "defs.h"
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82#include "vector.h"
83#include "heap.h"
189e9830 84#include "timeval.h"
a7e9570a 85#include "client.h"
8e3fe3d8 86#include "playrtp.h"
a99c4e9a 87#include "inputline.h"
e83d0967 88
1153fd23 89#define readahead linux_headers_are_borked
90
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91/** @brief Obsolete synonym */
92#ifndef IPV6_JOIN_GROUP
93# define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP
94#endif
95
0b75463f 96/** @brief RTP socket */
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97static int rtpfd;
98
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99/** @brief Log output */
100static FILE *logfp;
101
0b75463f 102/** @brief Output device */
8e3fe3d8 103const char *device;
0b75463f 104
9086a105 105/** @brief Minimum low watermark
0b75463f 106 *
107 * We'll stop playing if there's only this many samples in the buffer. */
c593cf7c 108unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */
0b75463f 109
9086a105 110/** @brief Buffer high watermark
1153fd23 111 *
112 * We'll only start playing when this many samples are available. */
8d0c14d7 113static unsigned readahead = 2 * 2 * 44100;
0b75463f 114
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115/** @brief Maximum buffer size
116 *
117 * We'll stop reading from the network if we have this many samples. */
118static unsigned maxbuffer;
119
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120/** @brief Received packets
121 * Protected by @ref receive_lock
122 *
123 * Received packets are added to this list, and queue_thread() picks them off
124 * it and adds them to @ref packets. Whenever a packet is added to it, @ref
125 * receive_cond is signalled.
126 */
8e3fe3d8 127struct packet *received_packets;
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128
129/** @brief Tail of @ref received_packets
130 * Protected by @ref receive_lock
131 */
8e3fe3d8 132struct packet **received_tail = &received_packets;
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133
134/** @brief Lock protecting @ref received_packets
135 *
136 * Only listen_thread() and queue_thread() ever hold this lock. It is vital
137 * that queue_thread() not hold it any longer than it strictly has to. */
8e3fe3d8 138pthread_mutex_t receive_lock = PTHREAD_MUTEX_INITIALIZER;
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139
140/** @brief Condition variable signalled when @ref received_packets is updated
141 *
142 * Used by listen_thread() to notify queue_thread() that it has added another
143 * packet to @ref received_packets. */
8e3fe3d8 144pthread_cond_t receive_cond = PTHREAD_COND_INITIALIZER;
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145
146/** @brief Length of @ref received_packets */
8e3fe3d8 147uint32_t nreceived;
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148
149/** @brief Binary heap of received packets */
8e3fe3d8 150struct pheap packets;
28bacdc0 151
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152/** @brief Total number of samples available
153 *
154 * We make this volatile because we inspect it without a protecting lock,
155 * so the usual pthread_* guarantees aren't available.
156 */
8e3fe3d8 157volatile uint32_t nsamples;
0b75463f 158
159/** @brief Timestamp of next packet to play.
160 *
161 * This is set to the timestamp of the last packet, plus the number of
09ee2f0d 162 * samples it contained. Only valid if @ref active is nonzero.
0b75463f 163 */
8e3fe3d8 164uint32_t next_timestamp;
e83d0967 165
09ee2f0d 166/** @brief True if actively playing
167 *
168 * This is true when playing and false when just buffering. */
8e3fe3d8 169int active;
09ee2f0d 170
189e9830 171/** @brief Lock protecting @ref packets */
8e3fe3d8 172pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
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173
174/** @brief Condition variable signalled whenever @ref packets is changed */
8e3fe3d8 175pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
2c7c9eae 176
146e86fb 177#if HAVE_ALSA_ASOUNDLIB_H
c593cf7c 178# define DEFAULT_BACKEND playrtp_alsa
a9f0ad12 179#elif HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
c593cf7c 180# define DEFAULT_BACKEND playrtp_oss
181#elif HAVE_COREAUDIO_AUDIOHARDWARE_H
182# define DEFAULT_BACKEND playrtp_coreaudio
183#else
184# error No known backend
185#endif
186
187/** @brief Backend to play with */
188static void (*backend)(void) = &DEFAULT_BACKEND;
189
8e3fe3d8 190HEAP_DEFINE(pheap, struct packet *, lt_packet);
e83d0967 191
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192/** @brief Control socket or NULL */
193const char *control_socket;
194
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195/** @brief Buffer for debugging dump
196 *
197 * The debug dump is enabled by the @c --dump option. It records the last 20s
198 * of audio to the specified file (which will be about 3.5Mbytes). The file is
199 * written as as ring buffer, so the start point will progress through it.
200 *
201 * Use clients/dump2wav to convert this to a WAV file, which can then be loaded
202 * into (e.g.) Audacity for further inspection.
203 *
204 * All three backends (ALSA, OSS, Core Audio) now support this option.
205 *
206 * The idea is to allow the user a few seconds to react to an audible artefact.
207 */
e9b635a3 208int16_t *dump_buffer;
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209
210/** @brief Current index within debugging dump */
e9b635a3 211size_t dump_index;
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212
213/** @brief Size of debugging dump in samples */
214size_t dump_size = 44100/*Hz*/ * 2/*channels*/ * 20/*seconds*/;
e9b635a3 215
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216static const struct option options[] = {
217 { "help", no_argument, 0, 'h' },
218 { "version", no_argument, 0, 'V' },
219 { "debug", no_argument, 0, 'd' },
0b75463f 220 { "device", required_argument, 0, 'D' },
1153fd23 221 { "min", required_argument, 0, 'm' },
9086a105 222 { "max", required_argument, 0, 'x' },
1153fd23 223 { "buffer", required_argument, 0, 'b' },
1f10f780 224 { "rcvbuf", required_argument, 0, 'R' },
a9f0ad12 225#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
c593cf7c 226 { "oss", no_argument, 0, 'o' },
227#endif
146e86fb 228#if HAVE_ALSA_ASOUNDLIB_H
c593cf7c 229 { "alsa", no_argument, 0, 'a' },
230#endif
231#if HAVE_COREAUDIO_AUDIOHARDWARE_H
232 { "core-audio", no_argument, 0, 'c' },
233#endif
e9b635a3 234 { "dump", required_argument, 0, 'r' },
a99c4e9a 235 { "socket", required_argument, 0, 's' },
a7e9570a 236 { "config", required_argument, 0, 'C' },
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237 { 0, 0, 0, 0 }
238};
239
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240/** @brief Control thread
241 *
242 * This thread is responsible for accepting control commands from Disobedience
243 * (or other controllers) over an AF_UNIX stream socket with a path specified
244 * by the @c --socket option. The protocol uses simple string commands and
245 * replies:
246 *
247 * - @c stop will shut the player down
248 * - @c query will send back the reply @c running
249 * - anything else is ignored
250 *
251 * Commands and response strings terminated by shutting down the connection or
252 * by a newline. No attempt is made to multiplex multiple clients so it is
253 * important that the command be sent as soon as the connection is made - it is
254 * assumed that both parties to the protocol are entirely cooperating with one
255 * another.
256 */
257static void *control_thread(void attribute((unused)) *arg) {
258 struct sockaddr_un sa;
259 int sfd, cfd;
260 char *line;
261 socklen_t salen;
262 FILE *fp;
263
264 assert(control_socket);
265 unlink(control_socket);
266 memset(&sa, 0, sizeof sa);
267 sa.sun_family = AF_UNIX;
268 strcpy(sa.sun_path, control_socket);
269 sfd = xsocket(PF_UNIX, SOCK_STREAM, 0);
270 if(bind(sfd, (const struct sockaddr *)&sa, sizeof sa) < 0)
271 fatal(errno, "error binding to %s", control_socket);
272 if(listen(sfd, 128) < 0)
273 fatal(errno, "error calling listen on %s", control_socket);
274 info("listening on %s", control_socket);
275 for(;;) {
276 salen = sizeof sa;
277 cfd = accept(sfd, (struct sockaddr *)&sa, &salen);
278 if(cfd < 0) {
279 switch(errno) {
280 case EINTR:
281 case EAGAIN:
282 break;
283 default:
284 fatal(errno, "error calling accept on %s", control_socket);
285 }
286 }
287 if(!(fp = fdopen(cfd, "r+"))) {
288 error(errno, "error calling fdopen for %s connection", control_socket);
289 close(cfd);
290 continue;
291 }
292 if(!inputline(control_socket, fp, &line, '\n')) {
293 if(!strcmp(line, "stop")) {
294 info("stopped via %s", control_socket);
295 exit(0); /* terminate immediately */
296 }
297 if(!strcmp(line, "query"))
298 fprintf(fp, "running");
299 xfree(line);
300 }
301 if(fclose(fp) < 0)
302 error(errno, "error closing %s connection", control_socket);
303 }
304}
305
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306/** @brief Drop the first packet
307 *
308 * Assumes that @ref lock is held.
309 */
310static void drop_first_packet(void) {
311 if(pheap_count(&packets)) {
312 struct packet *const p = pheap_remove(&packets);
313 nsamples -= p->nsamples;
c593cf7c 314 playrtp_free_packet(p);
2c7c9eae 315 pthread_cond_broadcast(&cond);
2c7c9eae 316 }
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317}
318
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319/** @brief Background thread adding packets to heap
320 *
321 * This just transfers packets from @ref received_packets to @ref packets. It
322 * is important that it holds @ref receive_lock for as little time as possible,
323 * in order to minimize the interval between calls to read() in
324 * listen_thread().
325 */
326static void *queue_thread(void attribute((unused)) *arg) {
327 struct packet *p;
328
329 for(;;) {
330 /* Get the next packet */
331 pthread_mutex_lock(&receive_lock);
332 while(!received_packets)
333 pthread_cond_wait(&receive_cond, &receive_lock);
334 p = received_packets;
335 received_packets = p->next;
336 if(!received_packets)
337 received_tail = &received_packets;
338 --nreceived;
339 pthread_mutex_unlock(&receive_lock);
340 /* Add it to the heap */
341 pthread_mutex_lock(&lock);
342 pheap_insert(&packets, p);
343 nsamples += p->nsamples;
344 pthread_cond_broadcast(&cond);
345 pthread_mutex_unlock(&lock);
346 }
347}
348
09ee2f0d 349/** @brief Background thread collecting samples
0b75463f 350 *
351 * This function collects samples, perhaps converts them to the target format,
b0fdc63d 352 * and adds them to the packet list.
353 *
354 * It is crucial that the gap between successive calls to read() is as small as
355 * possible: otherwise packets will be dropped.
356 *
357 * We use a binary heap to ensure that the unavoidable effort is at worst
358 * logarithmic in the total number of packets - in fact if packets are mostly
359 * received in order then we will largely do constant work per packet since the
360 * newest packet will always be last.
361 *
362 * Of more concern is that we must acquire the lock on the heap to add a packet
363 * to it. If this proves a problem in practice then the answer would be
364 * (probably doubly) linked list with new packets added the end and a second
365 * thread which reads packets off the list and adds them to the heap.
366 *
367 * We keep memory allocation (mostly) very fast by keeping pre-allocated
c593cf7c 368 * packets around; see @ref playrtp_new_packet().
b0fdc63d 369 */
0b75463f 370static void *listen_thread(void attribute((unused)) *arg) {
2c7c9eae 371 struct packet *p = 0;
0b75463f 372 int n;
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373 struct rtp_header header;
374 uint16_t seq;
375 uint32_t timestamp;
376 struct iovec iov[2];
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377
378 for(;;) {
189e9830 379 if(!p)
c593cf7c 380 p = playrtp_new_packet();
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381 iov[0].iov_base = &header;
382 iov[0].iov_len = sizeof header;
383 iov[1].iov_base = p->samples_raw;
b64efe7e 384 iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw;
2c7c9eae 385 n = readv(rtpfd, iov, 2);
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386 if(n < 0) {
387 switch(errno) {
388 case EINTR:
389 continue;
390 default:
391 fatal(errno, "error reading from socket");
392 }
393 }
0b75463f 394 /* Ignore too-short packets */
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395 if((size_t)n <= sizeof (struct rtp_header)) {
396 info("ignored a short packet");
0b75463f 397 continue;
345ebe66 398 }
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399 timestamp = htonl(header.timestamp);
400 seq = htons(header.seq);
09ee2f0d 401 /* Ignore packets in the past */
2c7c9eae 402 if(active && lt(timestamp, next_timestamp)) {
c0e41690 403 info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
2c7c9eae 404 timestamp, next_timestamp);
09ee2f0d 405 continue;
c0e41690 406 }
189e9830 407 p->next = 0;
58b5a68f 408 p->flags = 0;
2c7c9eae 409 p->timestamp = timestamp;
e83d0967 410 /* Convert to target format */
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411 if(header.mpt & 0x80)
412 p->flags |= IDLE;
2c7c9eae 413 switch(header.mpt & 0x7F) {
e83d0967 414 case 10:
2c7c9eae 415 p->nsamples = (n - sizeof header) / sizeof(uint16_t);
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416 break;
417 /* TODO support other RFC3551 media types (when the speaker does) */
418 default:
0b75463f 419 fatal(0, "unsupported RTP payload type %d",
2c7c9eae 420 header.mpt & 0x7F);
e83d0967 421 }
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422 if(logfp)
423 fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
2c7c9eae 424 seq, timestamp, p->nsamples, timestamp + p->nsamples);
0b75463f 425 /* Stop reading if we've reached the maximum.
426 *
427 * This is rather unsatisfactory: it means that if packets get heavily
428 * out of order then we guarantee dropouts. But for now... */
345ebe66 429 if(nsamples >= maxbuffer) {
189e9830 430 pthread_mutex_lock(&lock);
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431 while(nsamples >= maxbuffer)
432 pthread_cond_wait(&cond, &lock);
189e9830 433 pthread_mutex_unlock(&lock);
345ebe66 434 }
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435 /* Add the packet to the receive queue */
436 pthread_mutex_lock(&receive_lock);
437 *received_tail = p;
438 received_tail = &p->next;
439 ++nreceived;
440 pthread_cond_signal(&receive_cond);
441 pthread_mutex_unlock(&receive_lock);
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442 /* We'll need a new packet */
443 p = 0;
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444 }
445}
446
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447/** @brief Wait until the buffer is adequately full
448 *
449 * Must be called with @ref lock held.
450 */
c593cf7c 451void playrtp_fill_buffer(void) {
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452 while(nsamples)
453 drop_first_packet();
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454 info("Buffering...");
455 while(nsamples < readahead)
456 pthread_cond_wait(&cond, &lock);
457 next_timestamp = pheap_first(&packets)->timestamp;
458 active = 1;
459}
460
461/** @brief Find next packet
462 * @return Packet to play or NULL if none found
463 *
464 * The return packet is merely guaranteed not to be in the past: it might be
465 * the first packet in the future rather than one that is actually suitable to
466 * play.
467 *
468 * Must be called with @ref lock held.
469 */
c593cf7c 470struct packet *playrtp_next_packet(void) {
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471 while(pheap_count(&packets)) {
472 struct packet *const p = pheap_first(&packets);
473 if(le(p->timestamp + p->nsamples, next_timestamp)) {
474 /* This packet is in the past. Drop it and try another one. */
475 drop_first_packet();
476 } else
477 /* This packet is NOT in the past. (It might be in the future
478 * however.) */
479 return p;
480 }
481 return 0;
482}
483
09ee2f0d 484/** @brief Play an RTP stream
485 *
486 * This is the guts of the program. It is responsible for:
487 * - starting the listening thread
488 * - opening the audio device
489 * - reading ahead to build up a buffer
490 * - arranging for audio to be played
491 * - detecting when the buffer has got too small and re-buffering
492 */
0b75463f 493static void play_rtp(void) {
494 pthread_t ltid;
a99c4e9a 495 int err;
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496
497 /* We receive and convert audio data in a background thread */
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498 if((err = pthread_create(&ltid, 0, listen_thread, 0)))
499 fatal(err, "pthread_create listen_thread");
189e9830 500 /* We have a second thread to add received packets to the queue */
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501 if((err = pthread_create(&ltid, 0, queue_thread, 0)))
502 fatal(err, "pthread_create queue_thread");
c593cf7c 503 /* The rest of the work is backend-specific */
504 backend();
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505}
506
507/* display usage message and terminate */
508static void help(void) {
509 xprintf("Usage:\n"
510 " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
511 "Options:\n"
1153fd23 512 " --device, -D DEVICE Output device\n"
513 " --min, -m FRAMES Buffer low water mark\n"
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514 " --buffer, -b FRAMES Buffer high water mark\n"
515 " --max, -x FRAMES Buffer maximum size\n"
1f10f780 516 " --rcvbuf, -R BYTES Socket receive buffer size\n"
a7e9570a 517 " --config, -C PATH Set configuration file\n"
146e86fb 518#if HAVE_ALSA_ASOUNDLIB_H
c593cf7c 519 " --alsa, -a Use ALSA to play audio\n"
520#endif
a9f0ad12 521#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
c593cf7c 522 " --oss, -o Use OSS to play audio\n"
523#endif
524#if HAVE_COREAUDIO_AUDIOHARDWARE_H
525 " --core-audio, -c Use Core Audio to play audio\n"
526#endif
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527 " --help, -h Display usage message\n"
528 " --version, -V Display version number\n"
529 );
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530 xfclose(stdout);
531 exit(0);
532}
533
534/* display version number and terminate */
535static void version(void) {
536 xprintf("disorder-playrtp version %s\n", disorder_version_string);
537 xfclose(stdout);
538 exit(0);
539}
540
541int main(int argc, char **argv) {
a99c4e9a 542 int n, err;
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543 struct addrinfo *res;
544 struct stringlist sl;
0b75463f 545 char *sockname;
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546 int rcvbuf, target_rcvbuf = 131072;
547 socklen_t len;
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548 struct ip_mreq mreq;
549 struct ipv6_mreq mreq6;
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550 disorder_client *c;
551 char *address, *port;
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552 int is_multicast;
553 union any_sockaddr {
554 struct sockaddr sa;
555 struct sockaddr_in in;
556 struct sockaddr_in6 in6;
557 };
558 union any_sockaddr mgroup;
e9b635a3 559 const char *dumpfile = 0;
e83d0967 560
0b75463f 561 static const struct addrinfo prefs = {
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562 AI_PASSIVE,
563 PF_INET,
564 SOCK_DGRAM,
565 IPPROTO_UDP,
566 0,
567 0,
568 0,
569 0
570 };
571
572 mem_init();
573 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
e9b635a3 574 while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:aocC:r", options, 0)) >= 0) {
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575 switch(n) {
576 case 'h': help();
577 case 'V': version();
578 case 'd': debugging = 1; break;
0b75463f 579 case 'D': device = optarg; break;
1153fd23 580 case 'm': minbuffer = 2 * atol(optarg); break;
581 case 'b': readahead = 2 * atol(optarg); break;
9086a105 582 case 'x': maxbuffer = 2 * atol(optarg); break;
345ebe66 583 case 'L': logfp = fopen(optarg, "w"); break;
1f10f780 584 case 'R': target_rcvbuf = atoi(optarg); break;
146e86fb 585#if HAVE_ALSA_ASOUNDLIB_H
c593cf7c 586 case 'a': backend = playrtp_alsa; break;
587#endif
a9f0ad12 588#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
c593cf7c 589 case 'o': backend = playrtp_oss; break;
590#endif
591#if HAVE_COREAUDIO_AUDIOHARDWARE_H
592 case 'c': backend = playrtp_coreaudio; break;
593#endif
a7e9570a 594 case 'C': configfile = optarg; break;
a99c4e9a 595 case 's': control_socket = optarg; break;
e9b635a3 596 case 'r': dumpfile = optarg; break;
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597 default: fatal(0, "invalid option");
598 }
599 }
a7e9570a 600 if(config_read(0)) fatal(0, "cannot read configuration");
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601 if(!maxbuffer)
602 maxbuffer = 4 * readahead;
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603 argc -= optind;
604 argv += optind;
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605 switch(argc) {
606 case 0:
6fba990c 607 /* Get configuration from server */
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608 if(!(c = disorder_new(1))) exit(EXIT_FAILURE);
609 if(disorder_connect(c)) exit(EXIT_FAILURE);
610 if(disorder_rtp_address(c, &address, &port)) exit(EXIT_FAILURE);
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611 sl.n = 2;
612 sl.s = xcalloc(2, sizeof *sl.s);
613 sl.s[0] = address;
614 sl.s[1] = port;
a7e9570a 615 break;
6fba990c 616 case 1:
a7e9570a 617 case 2:
6fba990c 618 /* Use command-line ADDRESS+PORT or just PORT */
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619 sl.n = argc;
620 sl.s = argv;
621 break;
622 default:
6fba990c 623 fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]");
a7e9570a 624 }
6fba990c 625 /* Look up address and port */
0b75463f 626 if(!(res = get_address(&sl, &prefs, &sockname)))
e83d0967 627 exit(1);
6fba990c 628 /* Create the socket */
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629 if((rtpfd = socket(res->ai_family,
630 res->ai_socktype,
631 res->ai_protocol)) < 0)
632 fatal(errno, "error creating socket");
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633 /* Stash the multicast group address */
634 if((is_multicast = multicast(res->ai_addr))) {
635 memcpy(&mgroup, res->ai_addr, res->ai_addrlen);
636 switch(res->ai_addr->sa_family) {
637 case AF_INET:
638 mgroup.in.sin_port = 0;
639 break;
640 case AF_INET6:
641 mgroup.in6.sin6_port = 0;
642 break;
643 }
644 }
645 /* Bind to 0/port */
646 switch(res->ai_addr->sa_family) {
647 case AF_INET:
648 memset(&((struct sockaddr_in *)res->ai_addr)->sin_addr, 0,
649 sizeof (struct in_addr));
650 break;
651 case AF_INET6:
652 memset(&((struct sockaddr_in6 *)res->ai_addr)->sin6_addr, 0,
653 sizeof (struct in6_addr));
654 break;
655 default:
656 fatal(0, "unsupported family %d", (int)res->ai_addr->sa_family);
657 }
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658 if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
659 fatal(errno, "error binding socket to %s", sockname);
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660 if(is_multicast) {
661 switch(mgroup.sa.sa_family) {
23205f9c 662 case PF_INET:
6fba990c 663 mreq.imr_multiaddr = mgroup.in.sin_addr;
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664 mreq.imr_interface.s_addr = 0; /* use primary interface */
665 if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP,
666 &mreq, sizeof mreq) < 0)
667 fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP");
668 break;
669 case PF_INET6:
6fba990c 670 mreq6.ipv6mr_multiaddr = mgroup.in6.sin6_addr;
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671 memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface);
672 if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP,
673 &mreq6, sizeof mreq6) < 0)
674 fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP");
675 break;
676 default:
677 fatal(0, "unsupported address family %d", res->ai_family);
678 }
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679 info("listening on %s multicast group %s",
680 format_sockaddr(res->ai_addr), format_sockaddr(&mgroup.sa));
681 } else
682 info("listening on %s", format_sockaddr(res->ai_addr));
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683 len = sizeof rcvbuf;
684 if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0)
685 fatal(errno, "error calling getsockopt SO_RCVBUF");
f0bae611 686 if(target_rcvbuf > rcvbuf) {
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687 if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF,
688 &target_rcvbuf, sizeof target_rcvbuf) < 0)
689 error(errno, "error calling setsockopt SO_RCVBUF %d",
690 target_rcvbuf);
691 /* We try to carry on anyway */
692 else
693 info("changed socket receive buffer from %d to %d",
694 rcvbuf, target_rcvbuf);
695 } else
696 info("default socket receive buffer %d", rcvbuf);
697 if(logfp)
698 info("WARNING: -L option can impact performance");
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699 if(control_socket) {
700 pthread_t tid;
701
702 if((err = pthread_create(&tid, 0, control_thread, 0)))
703 fatal(err, "pthread_create control_thread");
704 }
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705 if(dumpfile) {
706 int fd;
707 unsigned char buffer[65536];
708 size_t written;
709
710 if((fd = open(dumpfile, O_RDWR|O_TRUNC|O_CREAT, 0666)) < 0)
711 fatal(errno, "opening %s", dumpfile);
712 /* Fill with 0s to a suitable size */
713 memset(buffer, 0, sizeof buffer);
714 for(written = 0; written < dump_size * sizeof(int16_t);
715 written += sizeof buffer) {
716 if(write(fd, buffer, sizeof buffer) < 0)
717 fatal(errno, "clearing %s", dumpfile);
718 }
719 /* Map the buffer into memory for convenience */
720 dump_buffer = mmap(0, dump_size * sizeof(int16_t), PROT_READ|PROT_WRITE,
721 MAP_SHARED, fd, 0);
722 if(dump_buffer == (void *)-1)
723 fatal(errno, "mapping %s", dumpfile);
724 info("dumping to %s", dumpfile);
725 }
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726 play_rtp();
727 return 0;
728}
729
730/*
731Local Variables:
732c-basic-offset:2
733comment-column:40
734fill-column:79
735indent-tabs-mode:nil
736End:
737*/