chiark / gitweb /
OSS backend for playrtp
[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>
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68
69#include "log.h"
70#include "mem.h"
71#include "configuration.h"
72#include "addr.h"
73#include "syscalls.h"
74#include "rtp.h"
0b75463f 75#include "defs.h"
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76#include "vector.h"
77#include "heap.h"
189e9830 78#include "timeval.h"
8e3fe3d8 79#include "playrtp.h"
e83d0967 80
1153fd23 81#define readahead linux_headers_are_borked
82
0b75463f 83/** @brief RTP socket */
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84static int rtpfd;
85
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86/** @brief Log output */
87static FILE *logfp;
88
0b75463f 89/** @brief Output device */
8e3fe3d8 90const char *device;
0b75463f 91
9086a105 92/** @brief Minimum low watermark
0b75463f 93 *
94 * We'll stop playing if there's only this many samples in the buffer. */
c593cf7c 95unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */
0b75463f 96
9086a105 97/** @brief Buffer high watermark
1153fd23 98 *
99 * We'll only start playing when this many samples are available. */
8d0c14d7 100static unsigned readahead = 2 * 2 * 44100;
0b75463f 101
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102/** @brief Maximum buffer size
103 *
104 * We'll stop reading from the network if we have this many samples. */
105static unsigned maxbuffer;
106
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107/** @brief Received packets
108 * Protected by @ref receive_lock
109 *
110 * Received packets are added to this list, and queue_thread() picks them off
111 * it and adds them to @ref packets. Whenever a packet is added to it, @ref
112 * receive_cond is signalled.
113 */
8e3fe3d8 114struct packet *received_packets;
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115
116/** @brief Tail of @ref received_packets
117 * Protected by @ref receive_lock
118 */
8e3fe3d8 119struct packet **received_tail = &received_packets;
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120
121/** @brief Lock protecting @ref received_packets
122 *
123 * Only listen_thread() and queue_thread() ever hold this lock. It is vital
124 * that queue_thread() not hold it any longer than it strictly has to. */
8e3fe3d8 125pthread_mutex_t receive_lock = PTHREAD_MUTEX_INITIALIZER;
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126
127/** @brief Condition variable signalled when @ref received_packets is updated
128 *
129 * Used by listen_thread() to notify queue_thread() that it has added another
130 * packet to @ref received_packets. */
8e3fe3d8 131pthread_cond_t receive_cond = PTHREAD_COND_INITIALIZER;
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132
133/** @brief Length of @ref received_packets */
8e3fe3d8 134uint32_t nreceived;
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135
136/** @brief Binary heap of received packets */
8e3fe3d8 137struct pheap packets;
28bacdc0 138
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139/** @brief Total number of samples available
140 *
141 * We make this volatile because we inspect it without a protecting lock,
142 * so the usual pthread_* guarantees aren't available.
143 */
8e3fe3d8 144volatile uint32_t nsamples;
0b75463f 145
146/** @brief Timestamp of next packet to play.
147 *
148 * This is set to the timestamp of the last packet, plus the number of
09ee2f0d 149 * samples it contained. Only valid if @ref active is nonzero.
0b75463f 150 */
8e3fe3d8 151uint32_t next_timestamp;
e83d0967 152
09ee2f0d 153/** @brief True if actively playing
154 *
155 * This is true when playing and false when just buffering. */
8e3fe3d8 156int active;
09ee2f0d 157
189e9830 158/** @brief Lock protecting @ref packets */
8e3fe3d8 159pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
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160
161/** @brief Condition variable signalled whenever @ref packets is changed */
8e3fe3d8 162pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
2c7c9eae 163
146e86fb 164#if HAVE_ALSA_ASOUNDLIB_H
c593cf7c 165# define DEFAULT_BACKEND playrtp_alsa
166#elif HAVE_SYS_SOUNDCARD_H
167# define DEFAULT_BACKEND playrtp_oss
168#elif HAVE_COREAUDIO_AUDIOHARDWARE_H
169# define DEFAULT_BACKEND playrtp_coreaudio
170#else
171# error No known backend
172#endif
173
174/** @brief Backend to play with */
175static void (*backend)(void) = &DEFAULT_BACKEND;
176
8e3fe3d8 177HEAP_DEFINE(pheap, struct packet *, lt_packet);
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178
179static const struct option options[] = {
180 { "help", no_argument, 0, 'h' },
181 { "version", no_argument, 0, 'V' },
182 { "debug", no_argument, 0, 'd' },
0b75463f 183 { "device", required_argument, 0, 'D' },
1153fd23 184 { "min", required_argument, 0, 'm' },
9086a105 185 { "max", required_argument, 0, 'x' },
1153fd23 186 { "buffer", required_argument, 0, 'b' },
1f10f780 187 { "rcvbuf", required_argument, 0, 'R' },
23205f9c 188 { "multicast", required_argument, 0, 'M' },
c593cf7c 189#if HAVE_SYS_SOUNDCARD_H
190 { "oss", no_argument, 0, 'o' },
191#endif
146e86fb 192#if HAVE_ALSA_ASOUNDLIB_H
c593cf7c 193 { "alsa", no_argument, 0, 'a' },
194#endif
195#if HAVE_COREAUDIO_AUDIOHARDWARE_H
196 { "core-audio", no_argument, 0, 'c' },
197#endif
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198 { 0, 0, 0, 0 }
199};
200
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201/** @brief Drop the first packet
202 *
203 * Assumes that @ref lock is held.
204 */
205static void drop_first_packet(void) {
206 if(pheap_count(&packets)) {
207 struct packet *const p = pheap_remove(&packets);
208 nsamples -= p->nsamples;
c593cf7c 209 playrtp_free_packet(p);
2c7c9eae 210 pthread_cond_broadcast(&cond);
2c7c9eae 211 }
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212}
213
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214/** @brief Background thread adding packets to heap
215 *
216 * This just transfers packets from @ref received_packets to @ref packets. It
217 * is important that it holds @ref receive_lock for as little time as possible,
218 * in order to minimize the interval between calls to read() in
219 * listen_thread().
220 */
221static void *queue_thread(void attribute((unused)) *arg) {
222 struct packet *p;
223
224 for(;;) {
225 /* Get the next packet */
226 pthread_mutex_lock(&receive_lock);
227 while(!received_packets)
228 pthread_cond_wait(&receive_cond, &receive_lock);
229 p = received_packets;
230 received_packets = p->next;
231 if(!received_packets)
232 received_tail = &received_packets;
233 --nreceived;
234 pthread_mutex_unlock(&receive_lock);
235 /* Add it to the heap */
236 pthread_mutex_lock(&lock);
237 pheap_insert(&packets, p);
238 nsamples += p->nsamples;
239 pthread_cond_broadcast(&cond);
240 pthread_mutex_unlock(&lock);
241 }
242}
243
09ee2f0d 244/** @brief Background thread collecting samples
0b75463f 245 *
246 * This function collects samples, perhaps converts them to the target format,
b0fdc63d 247 * and adds them to the packet list.
248 *
249 * It is crucial that the gap between successive calls to read() is as small as
250 * possible: otherwise packets will be dropped.
251 *
252 * We use a binary heap to ensure that the unavoidable effort is at worst
253 * logarithmic in the total number of packets - in fact if packets are mostly
254 * received in order then we will largely do constant work per packet since the
255 * newest packet will always be last.
256 *
257 * Of more concern is that we must acquire the lock on the heap to add a packet
258 * to it. If this proves a problem in practice then the answer would be
259 * (probably doubly) linked list with new packets added the end and a second
260 * thread which reads packets off the list and adds them to the heap.
261 *
262 * We keep memory allocation (mostly) very fast by keeping pre-allocated
c593cf7c 263 * packets around; see @ref playrtp_new_packet().
b0fdc63d 264 */
0b75463f 265static void *listen_thread(void attribute((unused)) *arg) {
2c7c9eae 266 struct packet *p = 0;
0b75463f 267 int n;
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268 struct rtp_header header;
269 uint16_t seq;
270 uint32_t timestamp;
271 struct iovec iov[2];
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272
273 for(;;) {
189e9830 274 if(!p)
c593cf7c 275 p = playrtp_new_packet();
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276 iov[0].iov_base = &header;
277 iov[0].iov_len = sizeof header;
278 iov[1].iov_base = p->samples_raw;
b64efe7e 279 iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw;
2c7c9eae 280 n = readv(rtpfd, iov, 2);
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281 if(n < 0) {
282 switch(errno) {
283 case EINTR:
284 continue;
285 default:
286 fatal(errno, "error reading from socket");
287 }
288 }
0b75463f 289 /* Ignore too-short packets */
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290 if((size_t)n <= sizeof (struct rtp_header)) {
291 info("ignored a short packet");
0b75463f 292 continue;
345ebe66 293 }
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294 timestamp = htonl(header.timestamp);
295 seq = htons(header.seq);
09ee2f0d 296 /* Ignore packets in the past */
2c7c9eae 297 if(active && lt(timestamp, next_timestamp)) {
c0e41690 298 info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
2c7c9eae 299 timestamp, next_timestamp);
09ee2f0d 300 continue;
c0e41690 301 }
189e9830 302 p->next = 0;
58b5a68f 303 p->flags = 0;
2c7c9eae 304 p->timestamp = timestamp;
e83d0967 305 /* Convert to target format */
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306 if(header.mpt & 0x80)
307 p->flags |= IDLE;
2c7c9eae 308 switch(header.mpt & 0x7F) {
e83d0967 309 case 10:
2c7c9eae 310 p->nsamples = (n - sizeof header) / sizeof(uint16_t);
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311 break;
312 /* TODO support other RFC3551 media types (when the speaker does) */
313 default:
0b75463f 314 fatal(0, "unsupported RTP payload type %d",
2c7c9eae 315 header.mpt & 0x7F);
e83d0967 316 }
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317 if(logfp)
318 fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
2c7c9eae 319 seq, timestamp, p->nsamples, timestamp + p->nsamples);
0b75463f 320 /* Stop reading if we've reached the maximum.
321 *
322 * This is rather unsatisfactory: it means that if packets get heavily
323 * out of order then we guarantee dropouts. But for now... */
345ebe66 324 if(nsamples >= maxbuffer) {
189e9830 325 pthread_mutex_lock(&lock);
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326 while(nsamples >= maxbuffer)
327 pthread_cond_wait(&cond, &lock);
189e9830 328 pthread_mutex_unlock(&lock);
345ebe66 329 }
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330 /* Add the packet to the receive queue */
331 pthread_mutex_lock(&receive_lock);
332 *received_tail = p;
333 received_tail = &p->next;
334 ++nreceived;
335 pthread_cond_signal(&receive_cond);
336 pthread_mutex_unlock(&receive_lock);
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337 /* We'll need a new packet */
338 p = 0;
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339 }
340}
341
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342/** @brief Wait until the buffer is adequately full
343 *
344 * Must be called with @ref lock held.
345 */
c593cf7c 346void playrtp_fill_buffer(void) {
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347 while(nsamples)
348 drop_first_packet();
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349 info("Buffering...");
350 while(nsamples < readahead)
351 pthread_cond_wait(&cond, &lock);
352 next_timestamp = pheap_first(&packets)->timestamp;
353 active = 1;
354}
355
356/** @brief Find next packet
357 * @return Packet to play or NULL if none found
358 *
359 * The return packet is merely guaranteed not to be in the past: it might be
360 * the first packet in the future rather than one that is actually suitable to
361 * play.
362 *
363 * Must be called with @ref lock held.
364 */
c593cf7c 365struct packet *playrtp_next_packet(void) {
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366 while(pheap_count(&packets)) {
367 struct packet *const p = pheap_first(&packets);
368 if(le(p->timestamp + p->nsamples, next_timestamp)) {
369 /* This packet is in the past. Drop it and try another one. */
370 drop_first_packet();
371 } else
372 /* This packet is NOT in the past. (It might be in the future
373 * however.) */
374 return p;
375 }
376 return 0;
377}
378
09ee2f0d 379/** @brief Play an RTP stream
380 *
381 * This is the guts of the program. It is responsible for:
382 * - starting the listening thread
383 * - opening the audio device
384 * - reading ahead to build up a buffer
385 * - arranging for audio to be played
386 * - detecting when the buffer has got too small and re-buffering
387 */
0b75463f 388static void play_rtp(void) {
389 pthread_t ltid;
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390
391 /* We receive and convert audio data in a background thread */
0b75463f 392 pthread_create(&ltid, 0, listen_thread, 0);
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393 /* We have a second thread to add received packets to the queue */
394 pthread_create(&ltid, 0, queue_thread, 0);
c593cf7c 395 /* The rest of the work is backend-specific */
396 backend();
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397}
398
399/* display usage message and terminate */
400static void help(void) {
401 xprintf("Usage:\n"
402 " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
403 "Options:\n"
1153fd23 404 " --device, -D DEVICE Output device\n"
405 " --min, -m FRAMES Buffer low water mark\n"
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406 " --buffer, -b FRAMES Buffer high water mark\n"
407 " --max, -x FRAMES Buffer maximum size\n"
1f10f780 408 " --rcvbuf, -R BYTES Socket receive buffer size\n"
23205f9c 409 " --multicast, -M GROUP Join multicast group\n"
146e86fb 410#if HAVE_ALSA_ASOUNDLIB_H
c593cf7c 411 " --alsa, -a Use ALSA to play audio\n"
412#endif
413#if HAVE_SYS_SOUNDCARD_H
414 " --oss, -o Use OSS to play audio\n"
415#endif
416#if HAVE_COREAUDIO_AUDIOHARDWARE_H
417 " --core-audio, -c Use Core Audio to play audio\n"
418#endif
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419 " --help, -h Display usage message\n"
420 " --version, -V Display version number\n"
421 );
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422 xfclose(stdout);
423 exit(0);
424}
425
426/* display version number and terminate */
427static void version(void) {
428 xprintf("disorder-playrtp version %s\n", disorder_version_string);
429 xfclose(stdout);
430 exit(0);
431}
432
433int main(int argc, char **argv) {
434 int n;
435 struct addrinfo *res;
436 struct stringlist sl;
0b75463f 437 char *sockname;
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438 int rcvbuf, target_rcvbuf = 131072;
439 socklen_t len;
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440 char *multicast_group = 0;
441 struct ip_mreq mreq;
442 struct ipv6_mreq mreq6;
e83d0967 443
0b75463f 444 static const struct addrinfo prefs = {
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445 AI_PASSIVE,
446 PF_INET,
447 SOCK_DGRAM,
448 IPPROTO_UDP,
449 0,
450 0,
451 0,
452 0
453 };
454
455 mem_init();
456 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
c593cf7c 457 while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:aoc", options, 0)) >= 0) {
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458 switch(n) {
459 case 'h': help();
460 case 'V': version();
461 case 'd': debugging = 1; break;
0b75463f 462 case 'D': device = optarg; break;
1153fd23 463 case 'm': minbuffer = 2 * atol(optarg); break;
464 case 'b': readahead = 2 * atol(optarg); break;
9086a105 465 case 'x': maxbuffer = 2 * atol(optarg); break;
345ebe66 466 case 'L': logfp = fopen(optarg, "w"); break;
1f10f780 467 case 'R': target_rcvbuf = atoi(optarg); break;
23205f9c 468 case 'M': multicast_group = optarg; break;
146e86fb 469#if HAVE_ALSA_ASOUNDLIB_H
c593cf7c 470 case 'a': backend = playrtp_alsa; break;
471#endif
c593cf7c 472#if HAVE_SYS_SOUNDCARD_H
473 case 'o': backend = playrtp_oss; break;
474#endif
475#if HAVE_COREAUDIO_AUDIOHARDWARE_H
476 case 'c': backend = playrtp_coreaudio; break;
477#endif
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478 default: fatal(0, "invalid option");
479 }
480 }
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481 if(!maxbuffer)
482 maxbuffer = 4 * readahead;
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483 argc -= optind;
484 argv += optind;
485 if(argc < 1 || argc > 2)
486 fatal(0, "usage: disorder-playrtp [OPTIONS] ADDRESS [PORT]");
487 sl.n = argc;
488 sl.s = argv;
489 /* Listen for inbound audio data */
0b75463f 490 if(!(res = get_address(&sl, &prefs, &sockname)))
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491 exit(1);
492 if((rtpfd = socket(res->ai_family,
493 res->ai_socktype,
494 res->ai_protocol)) < 0)
495 fatal(errno, "error creating socket");
496 if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
497 fatal(errno, "error binding socket to %s", sockname);
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498 if(multicast_group) {
499 if((n = getaddrinfo(multicast_group, 0, &prefs, &res)))
500 fatal(0, "getaddrinfo %s: %s", multicast_group, gai_strerror(n));
501 switch(res->ai_family) {
502 case PF_INET:
503 mreq.imr_multiaddr = ((struct sockaddr_in *)res->ai_addr)->sin_addr;
504 mreq.imr_interface.s_addr = 0; /* use primary interface */
505 if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP,
506 &mreq, sizeof mreq) < 0)
507 fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP");
508 break;
509 case PF_INET6:
510 mreq6.ipv6mr_multiaddr = ((struct sockaddr_in6 *)res->ai_addr)->sin6_addr;
511 memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface);
512 if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP,
513 &mreq6, sizeof mreq6) < 0)
514 fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP");
515 break;
516 default:
517 fatal(0, "unsupported address family %d", res->ai_family);
518 }
519 }
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520 len = sizeof rcvbuf;
521 if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0)
522 fatal(errno, "error calling getsockopt SO_RCVBUF");
f0bae611 523 if(target_rcvbuf > rcvbuf) {
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524 if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF,
525 &target_rcvbuf, sizeof target_rcvbuf) < 0)
526 error(errno, "error calling setsockopt SO_RCVBUF %d",
527 target_rcvbuf);
528 /* We try to carry on anyway */
529 else
530 info("changed socket receive buffer from %d to %d",
531 rcvbuf, target_rcvbuf);
532 } else
533 info("default socket receive buffer %d", rcvbuf);
534 if(logfp)
535 info("WARNING: -L option can impact performance");
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536 play_rtp();
537 return 0;
538}
539
540/*
541Local Variables:
542c-basic-offset:2
543comment-column:40
544fill-column:79
545indent-tabs-mode:nil
546End:
547*/