mdw@git.distorted.org.uk Git - disorder/blob - clients/playrtp.c

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