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
  70 #include "log.h"
  71 #include "mem.h"
  72 #include "configuration.h"
  73 #include "addr.h"
  74 #include "syscalls.h"
  75 #include "rtp.h"
  76 #include "defs.h"
  77 #include "vector.h"
  78 #include "heap.h"
  79 #include "timeval.h"
  80 #include "client.h"
  81 #include "playrtp.h"
  82
  83 #define readahead linux_headers_are_borked
  84
  85 /** @brief Obsolete synonym */
  86 #ifndef IPV6_JOIN_GROUP
  87 # define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP
  88 #endif
  89
  90 /** @brief RTP socket */
  91 static int rtpfd;
  92
  93 /** @brief Log output */
  94 static FILE *logfp;
  95
  96 /** @brief Output device */
  97 const char *device;
  98
  99 /** @brief Minimum low watermark
 100  *
 101  * We'll stop playing if there's only this many samples in the buffer. */
 102 unsigned minbuffer = 2 * 44100 / 10;  /* 0.2 seconds */
 103
 104 /** @brief Buffer high watermark
 105  *
 106  * We'll only start playing when this many samples are available. */
 107 static unsigned readahead = 2 * 2 * 44100;
 108
 109 /** @brief Maximum buffer size
 110  *
 111  * We'll stop reading from the network if we have this many samples. */
 112 static unsigned maxbuffer;
 113
 114 /** @brief Received packets
 115  * Protected by @ref receive_lock
 116  *
 117  * Received packets are added to this list, and queue_thread() picks them off
 118  * it and adds them to @ref packets.  Whenever a packet is added to it, @ref
 119  * receive_cond is signalled.
 120  */
 121 struct packet *received_packets;
 122
 123 /** @brief Tail of @ref received_packets
 124  * Protected by @ref receive_lock
 125  */
 126 struct packet **received_tail = &received_packets;
 127
 128 /** @brief Lock protecting @ref received_packets
 129  *
 130  * Only listen_thread() and queue_thread() ever hold this lock.  It is vital
 131  * that queue_thread() not hold it any longer than it strictly has to. */
 132 pthread_mutex_t receive_lock = PTHREAD_MUTEX_INITIALIZER;
 133
 134 /** @brief Condition variable signalled when @ref received_packets is updated
 135  *
 136  * Used by listen_thread() to notify queue_thread() that it has added another
 137  * packet to @ref received_packets. */
 138 pthread_cond_t receive_cond = PTHREAD_COND_INITIALIZER;
 139
 140 /** @brief Length of @ref received_packets */
 141 uint32_t nreceived;
 142
 143 /** @brief Binary heap of received packets */
 144 struct pheap packets;
 145
 146 /** @brief Total number of samples available
 147  *
 148  * We make this volatile because we inspect it without a protecting lock,
 149  * so the usual pthread_* guarantees aren't available.
 150  */
 151 volatile uint32_t nsamples;
 152
 153 /** @brief Timestamp of next packet to play.
 154  *
 155  * This is set to the timestamp of the last packet, plus the number of
 156  * samples it contained.  Only valid if @ref active is nonzero.
 157  */
 158 uint32_t next_timestamp;
 159
 160 /** @brief True if actively playing
 161  *
 162  * This is true when playing and false when just buffering. */
 163 int active;
 164
 165 /** @brief Lock protecting @ref packets */
 166 pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
 167
 168 /** @brief Condition variable signalled whenever @ref packets is changed */
 169 pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
 170
 171 #if HAVE_ALSA_ASOUNDLIB_H
 172 # define DEFAULT_BACKEND playrtp_alsa
 173 #elif HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
 174 # define DEFAULT_BACKEND playrtp_oss
 175 #elif HAVE_COREAUDIO_AUDIOHARDWARE_H
 176 # define DEFAULT_BACKEND playrtp_coreaudio
 177 #else
 178 # error No known backend
 179 #endif
 180
 181 /** @brief Backend to play with */
 182 static void (*backend)(void) = &DEFAULT_BACKEND;
 183
 184 HEAP_DEFINE(pheap, struct packet *, lt_packet);
 185
 186 static const struct option options[] = {
 187   { "help", no_argument, 0, 'h' },
 188   { "version", no_argument, 0, 'V' },
 189   { "debug", no_argument, 0, 'd' },
 190   { "device", required_argument, 0, 'D' },
 191   { "min", required_argument, 0, 'm' },
 192   { "max", required_argument, 0, 'x' },
 193   { "buffer", required_argument, 0, 'b' },
 194   { "rcvbuf", required_argument, 0, 'R' },
 195   { "multicast", required_argument, 0, 'M' },
 196 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
 197   { "oss", no_argument, 0, 'o' },
 198 #endif
 199 #if HAVE_ALSA_ASOUNDLIB_H
 200   { "alsa", no_argument, 0, 'a' },
 201 #endif
 202 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
 203   { "core-audio", no_argument, 0, 'c' },
 204 #endif
 205   { "config", required_argument, 0, 'C' },
 206   { 0, 0, 0, 0 }
 207 };
 208
 209 /** @brief Drop the first packet
 210  *
 211  * Assumes that @ref lock is held.
 212  */
 213 static void drop_first_packet(void) {
 214   if(pheap_count(&packets)) {
 215     struct packet *const p = pheap_remove(&packets);
 216     nsamples -= p->nsamples;
 217     playrtp_free_packet(p);
 218     pthread_cond_broadcast(&cond);
 219   }
 220 }
 221
 222 /** @brief Background thread adding packets to heap
 223  *
 224  * This just transfers packets from @ref received_packets to @ref packets.  It
 225  * is important that it holds @ref receive_lock for as little time as possible,
 226  * in order to minimize the interval between calls to read() in
 227  * listen_thread().
 228  */
 229 static void *queue_thread(void attribute((unused)) *arg) {
 230   struct packet *p;
 231
 232   for(;;) {
 233     /* Get the next packet */
 234     pthread_mutex_lock(&receive_lock);
 235     while(!received_packets)
 236       pthread_cond_wait(&receive_cond, &receive_lock);
 237     p = received_packets;
 238     received_packets = p->next;
 239     if(!received_packets)
 240       received_tail = &received_packets;
 241     --nreceived;
 242     pthread_mutex_unlock(&receive_lock);
 243     /* Add it to the heap */
 244     pthread_mutex_lock(&lock);
 245     pheap_insert(&packets, p);
 246     nsamples += p->nsamples;
 247     pthread_cond_broadcast(&cond);
 248     pthread_mutex_unlock(&lock);
 249   }
 250 }
 251
 252 /** @brief Background thread collecting samples
 253  *
 254  * This function collects samples, perhaps converts them to the target format,
 255  * and adds them to the packet list.
 256  *
 257  * It is crucial that the gap between successive calls to read() is as small as
 258  * possible: otherwise packets will be dropped.
 259  *
 260  * We use a binary heap to ensure that the unavoidable effort is at worst
 261  * logarithmic in the total number of packets - in fact if packets are mostly
 262  * received in order then we will largely do constant work per packet since the
 263  * newest packet will always be last.
 264  *
 265  * Of more concern is that we must acquire the lock on the heap to add a packet
 266  * to it.  If this proves a problem in practice then the answer would be
 267  * (probably doubly) linked list with new packets added the end and a second
 268  * thread which reads packets off the list and adds them to the heap.
 269  *
 270  * We keep memory allocation (mostly) very fast by keeping pre-allocated
 271  * packets around; see @ref playrtp_new_packet().
 272  */
 273 static void *listen_thread(void attribute((unused)) *arg) {
 274   struct packet *p = 0;
 275   int n;
 276   struct rtp_header header;
 277   uint16_t seq;
 278   uint32_t timestamp;
 279   struct iovec iov[2];
 280
 281   for(;;) {
 282     if(!p)
 283       p = playrtp_new_packet();
 284     iov[0].iov_base = &header;
 285     iov[0].iov_len = sizeof header;
 286     iov[1].iov_base = p->samples_raw;
 287     iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw;
 288     n = readv(rtpfd, iov, 2);
 289     if(n < 0) {
 290       switch(errno) {
 291       case EINTR:
 292         continue;
 293       default:
 294         fatal(errno, "error reading from socket");
 295       }
 296     }
 297     /* Ignore too-short packets */
 298     if((size_t)n <= sizeof (struct rtp_header)) {
 299       info("ignored a short packet");
 300       continue;
 301     }
 302     timestamp = htonl(header.timestamp);
 303     seq = htons(header.seq);
 304     /* Ignore packets in the past */
 305     if(active && lt(timestamp, next_timestamp)) {
 306       info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
 307            timestamp, next_timestamp);
 308       continue;
 309     }
 310     p->next = 0;
 311     p->flags = 0;
 312     p->timestamp = timestamp;
 313     /* Convert to target format */
 314     if(header.mpt & 0x80)
 315       p->flags |= IDLE;
 316     switch(header.mpt & 0x7F) {
 317     case 10:
 318       p->nsamples = (n - sizeof header) / sizeof(uint16_t);
 319       break;
 320       /* TODO support other RFC3551 media types (when the speaker does) */
 321     default:
 322       fatal(0, "unsupported RTP payload type %d",
 323             header.mpt & 0x7F);
 324     }
 325     if(logfp)
 326       fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
 327               seq, timestamp, p->nsamples, timestamp + p->nsamples);
 328     /* Stop reading if we've reached the maximum.
 329      *
 330      * This is rather unsatisfactory: it means that if packets get heavily
 331      * out of order then we guarantee dropouts.  But for now... */
 332     if(nsamples >= maxbuffer) {
 333       pthread_mutex_lock(&lock);
 334       while(nsamples >= maxbuffer)
 335         pthread_cond_wait(&cond, &lock);
 336       pthread_mutex_unlock(&lock);
 337     }
 338     /* Add the packet to the receive queue */
 339     pthread_mutex_lock(&receive_lock);
 340     *received_tail = p;
 341     received_tail = &p->next;
 342     ++nreceived;
 343     pthread_cond_signal(&receive_cond);
 344     pthread_mutex_unlock(&receive_lock);
 345     /* We'll need a new packet */
 346     p = 0;
 347   }
 348 }
 349
 350 /** @brief Wait until the buffer is adequately full
 351  *
 352  * Must be called with @ref lock held.
 353  */
 354 void playrtp_fill_buffer(void) {
 355   while(nsamples)
 356     drop_first_packet();
 357   info("Buffering...");
 358   while(nsamples < readahead)
 359     pthread_cond_wait(&cond, &lock);
 360   next_timestamp = pheap_first(&packets)->timestamp;
 361   active = 1;
 362 }
 363
 364 /** @brief Find next packet
 365  * @return Packet to play or NULL if none found
 366  *
 367  * The return packet is merely guaranteed not to be in the past: it might be
 368  * the first packet in the future rather than one that is actually suitable to
 369  * play.
 370  *
 371  * Must be called with @ref lock held.
 372  */
 373 struct packet *playrtp_next_packet(void) {
 374   while(pheap_count(&packets)) {
 375     struct packet *const p = pheap_first(&packets);
 376     if(le(p->timestamp + p->nsamples, next_timestamp)) {
 377       /* This packet is in the past.  Drop it and try another one. */
 378       drop_first_packet();
 379     } else
 380       /* This packet is NOT in the past.  (It might be in the future
 381        * however.) */
 382       return p;
 383   }
 384   return 0;
 385 }
 386
 387 /** @brief Play an RTP stream
 388  *
 389  * This is the guts of the program.  It is responsible for:
 390  * - starting the listening thread
 391  * - opening the audio device
 392  * - reading ahead to build up a buffer
 393  * - arranging for audio to be played
 394  * - detecting when the buffer has got too small and re-buffering
 395  */
 396 static void play_rtp(void) {
 397   pthread_t ltid;
 398
 399   /* We receive and convert audio data in a background thread */
 400   pthread_create(&ltid, 0, listen_thread, 0);
 401   /* We have a second thread to add received packets to the queue */
 402   pthread_create(&ltid, 0, queue_thread, 0);
 403   /* The rest of the work is backend-specific */
 404   backend();
 405 }
 406
 407 /* display usage message and terminate */
 408 static void help(void) {
 409   xprintf("Usage:\n"
 410           "  disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
 411           "Options:\n"
 412           "  --device, -D DEVICE     Output device\n"
 413           "  --min, -m FRAMES        Buffer low water mark\n"
 414           "  --buffer, -b FRAMES     Buffer high water mark\n"
 415           "  --max, -x FRAMES        Buffer maximum size\n"
 416           "  --rcvbuf, -R BYTES      Socket receive buffer size\n"
 417           "  --multicast, -M GROUP   Join multicast group\n"
 418           "  --config, -C PATH       Set configuration file\n"
 419 #if HAVE_ALSA_ASOUNDLIB_H
 420           "  --alsa, -a              Use ALSA to play audio\n"
 421 #endif
 422 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
 423           "  --oss, -o               Use OSS to play audio\n"
 424 #endif
 425 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
 426           "  --core-audio, -c        Use Core Audio to play audio\n"
 427 #endif
 428           "  --help, -h              Display usage message\n"
 429           "  --version, -V           Display version number\n"
 430           );
 431   xfclose(stdout);
 432   exit(0);
 433 }
 434
 435 /* display version number and terminate */
 436 static void version(void) {
 437   xprintf("disorder-playrtp version %s\n", disorder_version_string);
 438   xfclose(stdout);
 439   exit(0);
 440 }
 441
 442 int main(int argc, char **argv) {
 443   int n;
 444   struct addrinfo *res;
 445   struct stringlist sl;
 446   char *sockname;
 447   int rcvbuf, target_rcvbuf = 131072;
 448   socklen_t len;
 449   char *multicast_group = 0;
 450   struct ip_mreq mreq;
 451   struct ipv6_mreq mreq6;
 452   disorder_client *c;
 453   char *address, *port;
 454
 455   static const struct addrinfo prefs = {
 456     AI_PASSIVE,
 457     PF_INET,
 458     SOCK_DGRAM,
 459     IPPROTO_UDP,
 460     0,
 461     0,
 462     0,
 463     0
 464   };
 465
 466   mem_init();
 467   if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
 468   while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:aocC:", options, 0)) >= 0) {
 469     switch(n) {
 470     case 'h': help();
 471     case 'V': version();
 472     case 'd': debugging = 1; break;
 473     case 'D': device = optarg; break;
 474     case 'm': minbuffer = 2 * atol(optarg); break;
 475     case 'b': readahead = 2 * atol(optarg); break;
 476     case 'x': maxbuffer = 2 * atol(optarg); break;
 477     case 'L': logfp = fopen(optarg, "w"); break;
 478     case 'R': target_rcvbuf = atoi(optarg); break;
 479     case 'M': multicast_group = optarg; break;
 480 #if HAVE_ALSA_ASOUNDLIB_H
 481     case 'a': backend = playrtp_alsa; break;
 482 #endif
 483 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
 484     case 'o': backend = playrtp_oss; break;
 485 #endif
 486 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
 487     case 'c': backend = playrtp_coreaudio; break;
 488 #endif
 489     case 'C': configfile = optarg; break;
 490     default: fatal(0, "invalid option");
 491     }
 492   }
 493   if(config_read(0)) fatal(0, "cannot read configuration");
 494   if(!maxbuffer)
 495     maxbuffer = 4 * readahead;
 496   argc -= optind;
 497   argv += optind;
 498   switch(argc) {
 499   case 0:
 500   case 1:
 501     if(!(c = disorder_new(1))) exit(EXIT_FAILURE);
 502     if(disorder_connect(c)) exit(EXIT_FAILURE);
 503     if(disorder_rtp_address(c, &address, &port)) exit(EXIT_FAILURE);
 504     sl.n = 1;
 505     sl.s = &port;
 506     /* set multicast_group if address is a multicast address */
 507     break;
 508   case 2:
 509     sl.n = argc;
 510     sl.s = argv;
 511     break;
 512   default:
 513     fatal(0, "usage: disorder-playrtp [OPTIONS] [ADDRESS [PORT]]");
 514   }
 515   /* Listen for inbound audio data */
 516   if(!(res = get_address(&sl, &prefs, &sockname)))
 517     exit(1);
 518   info("listening on %s", sockname);
 519   if((rtpfd = socket(res->ai_family,
 520                      res->ai_socktype,
 521                      res->ai_protocol)) < 0)
 522     fatal(errno, "error creating socket");
 523   if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
 524     fatal(errno, "error binding socket to %s", sockname);
 525   if(multicast_group) {
 526     if((n = getaddrinfo(multicast_group, 0, &prefs, &res)))
 527       fatal(0, "getaddrinfo %s: %s", multicast_group, gai_strerror(n));
 528     switch(res->ai_family) {
 529     case PF_INET:
 530       mreq.imr_multiaddr = ((struct sockaddr_in *)res->ai_addr)->sin_addr;
 531       mreq.imr_interface.s_addr = 0;      /* use primary interface */
 532       if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP,
 533                     &mreq, sizeof mreq) < 0)
 534         fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP");
 535       break;
 536     case PF_INET6:
 537       mreq6.ipv6mr_multiaddr = ((struct sockaddr_in6 *)res->ai_addr)->sin6_addr;
 538       memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface);
 539       if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP,
 540                     &mreq6, sizeof mreq6) < 0)
 541         fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP");
 542       break;
 543     default:
 544       fatal(0, "unsupported address family %d", res->ai_family);
 545     }
 546   }
 547   len = sizeof rcvbuf;
 548   if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0)
 549     fatal(errno, "error calling getsockopt SO_RCVBUF");
 550   if(target_rcvbuf > rcvbuf) {
 551     if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF,
 552                   &target_rcvbuf, sizeof target_rcvbuf) < 0)
 553       error(errno, "error calling setsockopt SO_RCVBUF %d",
 554             target_rcvbuf);
 555       /* We try to carry on anyway */
 556     else
 557       info("changed socket receive buffer from %d to %d",
 558            rcvbuf, target_rcvbuf);
 559   } else
 560     info("default socket receive buffer %d", rcvbuf);
 561   if(logfp)
 562     info("WARNING: -L option can impact performance");
 563   play_rtp();
 564   return 0;
 565 }
 566
 567 /*
 568 Local Variables:
 569 c-basic-offset:2
 570 comment-column:40
 571 fill-column:79
 572 indent-tabs-mode:nil
 573 End:
 574 */