| 1 | /* |
| 2 | * This file is part of DisOrder. |
| 3 | * Copyright (C) 2007, 2008 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 3 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, |
| 11 | * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| 12 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
| 13 | * GNU 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, see <http://www.gnu.org/licenses/>. |
| 17 | */ |
| 18 | /** @file clients/playrtp.c |
| 19 | * @brief RTP player |
| 20 | * |
| 21 | * This player supports Linux (<a href="http://www.alsa-project.org/">ALSA</a>) |
| 22 | * and Apple Mac (<a |
| 23 | * href="http://developer.apple.com/audio/coreaudio.html">Core Audio</a>) |
| 24 | * systems. There is no support for Microsoft Windows yet, and that will in |
| 25 | * fact probably an entirely separate program. |
| 26 | * |
| 27 | * The program runs (at least) three threads. listen_thread() is responsible |
| 28 | * for reading RTP packets off the wire and adding them to the linked list @ref |
| 29 | * received_packets, assuming they are basically sound. queue_thread() takes |
| 30 | * packets off this linked list and adds them to @ref packets (an operation |
| 31 | * which might be much slower due to contention for @ref lock). |
| 32 | * |
| 33 | * The main thread is responsible for actually playing audio. In ALSA this |
| 34 | * means it waits until ALSA says it's ready for more audio which it then |
| 35 | * plays. See @ref clients/playrtp-alsa.c. |
| 36 | * |
| 37 | * In Core Audio the main thread is only responsible for starting and stopping |
| 38 | * play: the system does the actual playback in its own private thread, and |
| 39 | * calls adioproc() to fetch the audio data. See @ref |
| 40 | * clients/playrtp-coreaudio.c. |
| 41 | * |
| 42 | * Sometimes it happens that there is no audio available to play. This may |
| 43 | * because the server went away, or a packet was dropped, or the server |
| 44 | * deliberately did not send any sound because it encountered a silence. |
| 45 | * |
| 46 | * Assumptions: |
| 47 | * - it is safe to read uint32_t values without a lock protecting them |
| 48 | */ |
| 49 | |
| 50 | #include "common.h" |
| 51 | |
| 52 | #include <getopt.h> |
| 53 | #include <sys/socket.h> |
| 54 | #include <sys/types.h> |
| 55 | #include <sys/socket.h> |
| 56 | #include <netdb.h> |
| 57 | #include <pthread.h> |
| 58 | #include <locale.h> |
| 59 | #include <sys/uio.h> |
| 60 | #include <errno.h> |
| 61 | #include <netinet/in.h> |
| 62 | #include <sys/time.h> |
| 63 | #include <sys/un.h> |
| 64 | #include <unistd.h> |
| 65 | #include <sys/mman.h> |
| 66 | #include <fcntl.h> |
| 67 | |
| 68 | #include "log.h" |
| 69 | #include "mem.h" |
| 70 | #include "configuration.h" |
| 71 | #include "addr.h" |
| 72 | #include "syscalls.h" |
| 73 | #include "rtp.h" |
| 74 | #include "defs.h" |
| 75 | #include "vector.h" |
| 76 | #include "heap.h" |
| 77 | #include "timeval.h" |
| 78 | #include "client.h" |
| 79 | #include "playrtp.h" |
| 80 | #include "inputline.h" |
| 81 | #include "version.h" |
| 82 | #include "uaudio.h" |
| 83 | |
| 84 | #define readahead linux_headers_are_borked |
| 85 | |
| 86 | /** @brief Obsolete synonym */ |
| 87 | #ifndef IPV6_JOIN_GROUP |
| 88 | # define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP |
| 89 | #endif |
| 90 | |
| 91 | /** @brief RTP socket */ |
| 92 | static int rtpfd; |
| 93 | |
| 94 | /** @brief Log output */ |
| 95 | static FILE *logfp; |
| 96 | |
| 97 | /** @brief Output 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 = 44100; /* 0.5 seconds */ |
| 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 | /** @brief Backend to play with */ |
| 172 | static const struct uaudio *backend; |
| 173 | |
| 174 | HEAP_DEFINE(pheap, struct packet *, lt_packet); |
| 175 | |
| 176 | /** @brief Control socket or NULL */ |
| 177 | const char *control_socket; |
| 178 | |
| 179 | /** @brief Buffer for debugging dump |
| 180 | * |
| 181 | * The debug dump is enabled by the @c --dump option. It records the last 20s |
| 182 | * of audio to the specified file (which will be about 3.5Mbytes). The file is |
| 183 | * written as as ring buffer, so the start point will progress through it. |
| 184 | * |
| 185 | * Use clients/dump2wav to convert this to a WAV file, which can then be loaded |
| 186 | * into (e.g.) Audacity for further inspection. |
| 187 | * |
| 188 | * All three backends (ALSA, OSS, Core Audio) now support this option. |
| 189 | * |
| 190 | * The idea is to allow the user a few seconds to react to an audible artefact. |
| 191 | */ |
| 192 | int16_t *dump_buffer; |
| 193 | |
| 194 | /** @brief Current index within debugging dump */ |
| 195 | size_t dump_index; |
| 196 | |
| 197 | /** @brief Size of debugging dump in samples */ |
| 198 | size_t dump_size = 44100/*Hz*/ * 2/*channels*/ * 20/*seconds*/; |
| 199 | |
| 200 | static const struct option options[] = { |
| 201 | { "help", no_argument, 0, 'h' }, |
| 202 | { "version", no_argument, 0, 'V' }, |
| 203 | { "debug", no_argument, 0, 'd' }, |
| 204 | { "device", required_argument, 0, 'D' }, |
| 205 | { "min", required_argument, 0, 'm' }, |
| 206 | { "max", required_argument, 0, 'x' }, |
| 207 | { "buffer", required_argument, 0, 'b' }, |
| 208 | { "rcvbuf", required_argument, 0, 'R' }, |
| 209 | #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST |
| 210 | { "oss", no_argument, 0, 'o' }, |
| 211 | #endif |
| 212 | #if HAVE_ALSA_ASOUNDLIB_H |
| 213 | { "alsa", no_argument, 0, 'a' }, |
| 214 | #endif |
| 215 | #if HAVE_COREAUDIO_AUDIOHARDWARE_H |
| 216 | { "core-audio", no_argument, 0, 'c' }, |
| 217 | #endif |
| 218 | { "dump", required_argument, 0, 'r' }, |
| 219 | { "command", required_argument, 0, 'e' }, |
| 220 | { "socket", required_argument, 0, 's' }, |
| 221 | { "config", required_argument, 0, 'C' }, |
| 222 | { 0, 0, 0, 0 } |
| 223 | }; |
| 224 | |
| 225 | /** @brief Control thread |
| 226 | * |
| 227 | * This thread is responsible for accepting control commands from Disobedience |
| 228 | * (or other controllers) over an AF_UNIX stream socket with a path specified |
| 229 | * by the @c --socket option. The protocol uses simple string commands and |
| 230 | * replies: |
| 231 | * |
| 232 | * - @c stop will shut the player down |
| 233 | * - @c query will send back the reply @c running |
| 234 | * - anything else is ignored |
| 235 | * |
| 236 | * Commands and response strings terminated by shutting down the connection or |
| 237 | * by a newline. No attempt is made to multiplex multiple clients so it is |
| 238 | * important that the command be sent as soon as the connection is made - it is |
| 239 | * assumed that both parties to the protocol are entirely cooperating with one |
| 240 | * another. |
| 241 | */ |
| 242 | static void *control_thread(void attribute((unused)) *arg) { |
| 243 | struct sockaddr_un sa; |
| 244 | int sfd, cfd; |
| 245 | char *line; |
| 246 | socklen_t salen; |
| 247 | FILE *fp; |
| 248 | |
| 249 | assert(control_socket); |
| 250 | unlink(control_socket); |
| 251 | memset(&sa, 0, sizeof sa); |
| 252 | sa.sun_family = AF_UNIX; |
| 253 | strcpy(sa.sun_path, control_socket); |
| 254 | sfd = xsocket(PF_UNIX, SOCK_STREAM, 0); |
| 255 | if(bind(sfd, (const struct sockaddr *)&sa, sizeof sa) < 0) |
| 256 | fatal(errno, "error binding to %s", control_socket); |
| 257 | if(listen(sfd, 128) < 0) |
| 258 | fatal(errno, "error calling listen on %s", control_socket); |
| 259 | info("listening on %s", control_socket); |
| 260 | for(;;) { |
| 261 | salen = sizeof sa; |
| 262 | cfd = accept(sfd, (struct sockaddr *)&sa, &salen); |
| 263 | if(cfd < 0) { |
| 264 | switch(errno) { |
| 265 | case EINTR: |
| 266 | case EAGAIN: |
| 267 | break; |
| 268 | default: |
| 269 | fatal(errno, "error calling accept on %s", control_socket); |
| 270 | } |
| 271 | } |
| 272 | if(!(fp = fdopen(cfd, "r+"))) { |
| 273 | error(errno, "error calling fdopen for %s connection", control_socket); |
| 274 | close(cfd); |
| 275 | continue; |
| 276 | } |
| 277 | if(!inputline(control_socket, fp, &line, '\n')) { |
| 278 | if(!strcmp(line, "stop")) { |
| 279 | info("stopped via %s", control_socket); |
| 280 | exit(0); /* terminate immediately */ |
| 281 | } |
| 282 | if(!strcmp(line, "query")) |
| 283 | fprintf(fp, "running"); |
| 284 | xfree(line); |
| 285 | } |
| 286 | if(fclose(fp) < 0) |
| 287 | error(errno, "error closing %s connection", control_socket); |
| 288 | } |
| 289 | } |
| 290 | |
| 291 | /** @brief Drop the first packet |
| 292 | * |
| 293 | * Assumes that @ref lock is held. |
| 294 | */ |
| 295 | static void drop_first_packet(void) { |
| 296 | if(pheap_count(&packets)) { |
| 297 | struct packet *const p = pheap_remove(&packets); |
| 298 | nsamples -= p->nsamples; |
| 299 | playrtp_free_packet(p); |
| 300 | pthread_cond_broadcast(&cond); |
| 301 | } |
| 302 | } |
| 303 | |
| 304 | /** @brief Background thread adding packets to heap |
| 305 | * |
| 306 | * This just transfers packets from @ref received_packets to @ref packets. It |
| 307 | * is important that it holds @ref receive_lock for as little time as possible, |
| 308 | * in order to minimize the interval between calls to read() in |
| 309 | * listen_thread(). |
| 310 | */ |
| 311 | static void *queue_thread(void attribute((unused)) *arg) { |
| 312 | struct packet *p; |
| 313 | |
| 314 | for(;;) { |
| 315 | /* Get the next packet */ |
| 316 | pthread_mutex_lock(&receive_lock); |
| 317 | while(!received_packets) { |
| 318 | pthread_cond_wait(&receive_cond, &receive_lock); |
| 319 | } |
| 320 | p = received_packets; |
| 321 | received_packets = p->next; |
| 322 | if(!received_packets) |
| 323 | received_tail = &received_packets; |
| 324 | --nreceived; |
| 325 | pthread_mutex_unlock(&receive_lock); |
| 326 | /* Add it to the heap */ |
| 327 | pthread_mutex_lock(&lock); |
| 328 | pheap_insert(&packets, p); |
| 329 | nsamples += p->nsamples; |
| 330 | pthread_cond_broadcast(&cond); |
| 331 | pthread_mutex_unlock(&lock); |
| 332 | } |
| 333 | } |
| 334 | |
| 335 | /** @brief Background thread collecting samples |
| 336 | * |
| 337 | * This function collects samples, perhaps converts them to the target format, |
| 338 | * and adds them to the packet list. |
| 339 | * |
| 340 | * It is crucial that the gap between successive calls to read() is as small as |
| 341 | * possible: otherwise packets will be dropped. |
| 342 | * |
| 343 | * We use a binary heap to ensure that the unavoidable effort is at worst |
| 344 | * logarithmic in the total number of packets - in fact if packets are mostly |
| 345 | * received in order then we will largely do constant work per packet since the |
| 346 | * newest packet will always be last. |
| 347 | * |
| 348 | * Of more concern is that we must acquire the lock on the heap to add a packet |
| 349 | * to it. If this proves a problem in practice then the answer would be |
| 350 | * (probably doubly) linked list with new packets added the end and a second |
| 351 | * thread which reads packets off the list and adds them to the heap. |
| 352 | * |
| 353 | * We keep memory allocation (mostly) very fast by keeping pre-allocated |
| 354 | * packets around; see @ref playrtp_new_packet(). |
| 355 | */ |
| 356 | static void *listen_thread(void attribute((unused)) *arg) { |
| 357 | struct packet *p = 0; |
| 358 | int n; |
| 359 | struct rtp_header header; |
| 360 | uint16_t seq; |
| 361 | uint32_t timestamp; |
| 362 | struct iovec iov[2]; |
| 363 | |
| 364 | for(;;) { |
| 365 | if(!p) |
| 366 | p = playrtp_new_packet(); |
| 367 | iov[0].iov_base = &header; |
| 368 | iov[0].iov_len = sizeof header; |
| 369 | iov[1].iov_base = p->samples_raw; |
| 370 | iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw; |
| 371 | n = readv(rtpfd, iov, 2); |
| 372 | if(n < 0) { |
| 373 | switch(errno) { |
| 374 | case EINTR: |
| 375 | continue; |
| 376 | default: |
| 377 | fatal(errno, "error reading from socket"); |
| 378 | } |
| 379 | } |
| 380 | /* Ignore too-short packets */ |
| 381 | if((size_t)n <= sizeof (struct rtp_header)) { |
| 382 | info("ignored a short packet"); |
| 383 | continue; |
| 384 | } |
| 385 | timestamp = htonl(header.timestamp); |
| 386 | seq = htons(header.seq); |
| 387 | /* Ignore packets in the past */ |
| 388 | if(active && lt(timestamp, next_timestamp)) { |
| 389 | info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32, |
| 390 | timestamp, next_timestamp); |
| 391 | continue; |
| 392 | } |
| 393 | /* Ignore packets with the extension bit set. */ |
| 394 | if(header.vpxcc & 0x10) |
| 395 | continue; |
| 396 | p->next = 0; |
| 397 | p->flags = 0; |
| 398 | p->timestamp = timestamp; |
| 399 | /* Convert to target format */ |
| 400 | if(header.mpt & 0x80) |
| 401 | p->flags |= IDLE; |
| 402 | switch(header.mpt & 0x7F) { |
| 403 | case 10: /* L16 */ |
| 404 | p->nsamples = (n - sizeof header) / sizeof(uint16_t); |
| 405 | break; |
| 406 | /* TODO support other RFC3551 media types (when the speaker does) */ |
| 407 | default: |
| 408 | fatal(0, "unsupported RTP payload type %d", |
| 409 | header.mpt & 0x7F); |
| 410 | } |
| 411 | if(logfp) |
| 412 | fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n", |
| 413 | seq, timestamp, p->nsamples, timestamp + p->nsamples); |
| 414 | /* Stop reading if we've reached the maximum. |
| 415 | * |
| 416 | * This is rather unsatisfactory: it means that if packets get heavily |
| 417 | * out of order then we guarantee dropouts. But for now... */ |
| 418 | if(nsamples >= maxbuffer) { |
| 419 | pthread_mutex_lock(&lock); |
| 420 | while(nsamples >= maxbuffer) { |
| 421 | pthread_cond_wait(&cond, &lock); |
| 422 | } |
| 423 | pthread_mutex_unlock(&lock); |
| 424 | } |
| 425 | /* Add the packet to the receive queue */ |
| 426 | pthread_mutex_lock(&receive_lock); |
| 427 | *received_tail = p; |
| 428 | received_tail = &p->next; |
| 429 | ++nreceived; |
| 430 | pthread_cond_signal(&receive_cond); |
| 431 | pthread_mutex_unlock(&receive_lock); |
| 432 | /* We'll need a new packet */ |
| 433 | p = 0; |
| 434 | } |
| 435 | } |
| 436 | |
| 437 | /** @brief Wait until the buffer is adequately full |
| 438 | * |
| 439 | * Must be called with @ref lock held. |
| 440 | */ |
| 441 | void playrtp_fill_buffer(void) { |
| 442 | while(nsamples) |
| 443 | drop_first_packet(); |
| 444 | info("Buffering..."); |
| 445 | while(nsamples < readahead) { |
| 446 | pthread_cond_wait(&cond, &lock); |
| 447 | } |
| 448 | next_timestamp = pheap_first(&packets)->timestamp; |
| 449 | active = 1; |
| 450 | } |
| 451 | |
| 452 | /** @brief Find next packet |
| 453 | * @return Packet to play or NULL if none found |
| 454 | * |
| 455 | * The return packet is merely guaranteed not to be in the past: it might be |
| 456 | * the first packet in the future rather than one that is actually suitable to |
| 457 | * play. |
| 458 | * |
| 459 | * Must be called with @ref lock held. |
| 460 | */ |
| 461 | struct packet *playrtp_next_packet(void) { |
| 462 | while(pheap_count(&packets)) { |
| 463 | struct packet *const p = pheap_first(&packets); |
| 464 | if(le(p->timestamp + p->nsamples, next_timestamp)) { |
| 465 | /* This packet is in the past. Drop it and try another one. */ |
| 466 | drop_first_packet(); |
| 467 | } else |
| 468 | /* This packet is NOT in the past. (It might be in the future |
| 469 | * however.) */ |
| 470 | return p; |
| 471 | } |
| 472 | return 0; |
| 473 | } |
| 474 | |
| 475 | /* display usage message and terminate */ |
| 476 | static void help(void) { |
| 477 | xprintf("Usage:\n" |
| 478 | " disorder-playrtp [OPTIONS] [[ADDRESS] PORT]\n" |
| 479 | "Options:\n" |
| 480 | " --device, -D DEVICE Output device\n" |
| 481 | " --min, -m FRAMES Buffer low water mark\n" |
| 482 | " --buffer, -b FRAMES Buffer high water mark\n" |
| 483 | " --max, -x FRAMES Buffer maximum size\n" |
| 484 | " --rcvbuf, -R BYTES Socket receive buffer size\n" |
| 485 | " --config, -C PATH Set configuration file\n" |
| 486 | #if HAVE_ALSA_ASOUNDLIB_H |
| 487 | " --alsa, -a Use ALSA to play audio\n" |
| 488 | #endif |
| 489 | #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST |
| 490 | " --oss, -o Use OSS to play audio\n" |
| 491 | #endif |
| 492 | #if HAVE_COREAUDIO_AUDIOHARDWARE_H |
| 493 | " --core-audio, -c Use Core Audio to play audio\n" |
| 494 | #endif |
| 495 | " --command, -e COMMAND Pipe audio to command\n" |
| 496 | " --help, -h Display usage message\n" |
| 497 | " --version, -V Display version number\n" |
| 498 | ); |
| 499 | xfclose(stdout); |
| 500 | exit(0); |
| 501 | } |
| 502 | |
| 503 | static size_t playrtp_callback(void *buffer, |
| 504 | size_t max_samples, |
| 505 | void attribute((unused)) *userdata) { |
| 506 | size_t samples; |
| 507 | |
| 508 | pthread_mutex_lock(&lock); |
| 509 | /* Get the next packet, junking any that are now in the past */ |
| 510 | const struct packet *p = playrtp_next_packet(); |
| 511 | if(p && contains(p, next_timestamp)) { |
| 512 | /* This packet is ready to play; the desired next timestamp points |
| 513 | * somewhere into it. */ |
| 514 | |
| 515 | /* Timestamp of end of packet */ |
| 516 | const uint32_t packet_end = p->timestamp + p->nsamples; |
| 517 | |
| 518 | /* Offset of desired next timestamp into current packet */ |
| 519 | const uint32_t offset = next_timestamp - p->timestamp; |
| 520 | |
| 521 | /* Pointer to audio data */ |
| 522 | const uint16_t *ptr = (void *)(p->samples_raw + offset); |
| 523 | |
| 524 | /* Compute number of samples left in packet, limited to output buffer |
| 525 | * size */ |
| 526 | samples = packet_end - next_timestamp; |
| 527 | if(samples > max_samples) |
| 528 | samples = max_samples; |
| 529 | |
| 530 | /* Copy into buffer, converting to native endianness */ |
| 531 | size_t i = samples; |
| 532 | int16_t *bufptr = buffer; |
| 533 | while(i > 0) { |
| 534 | *bufptr++ = (int16_t)ntohs(*ptr++); |
| 535 | --i; |
| 536 | } |
| 537 | /* We don't junk the packet here; a subsequent call to |
| 538 | * playrtp_next_packet() will dispose of it (if it's actually done with). */ |
| 539 | } else { |
| 540 | /* There is no suitable packet. We introduce 0s up to the next packet, or |
| 541 | * to fill the buffer if there's no next packet or that's too many. The |
| 542 | * comparison with max_samples deals with the otherwise troubling overflow |
| 543 | * case. */ |
| 544 | samples = p ? p->timestamp - next_timestamp : max_samples; |
| 545 | if(samples > max_samples) |
| 546 | samples = max_samples; |
| 547 | //info("infill by %zu", samples); |
| 548 | memset(buffer, 0, samples * uaudio_sample_size); |
| 549 | } |
| 550 | /* Debug dump */ |
| 551 | if(dump_buffer) { |
| 552 | for(size_t i = 0; i < samples; ++i) { |
| 553 | dump_buffer[dump_index++] = ((int16_t *)buffer)[i]; |
| 554 | dump_index %= dump_size; |
| 555 | } |
| 556 | } |
| 557 | /* Advance timestamp */ |
| 558 | next_timestamp += samples; |
| 559 | pthread_mutex_unlock(&lock); |
| 560 | return samples; |
| 561 | } |
| 562 | |
| 563 | int main(int argc, char **argv) { |
| 564 | int n, err; |
| 565 | struct addrinfo *res; |
| 566 | struct stringlist sl; |
| 567 | char *sockname; |
| 568 | int rcvbuf, target_rcvbuf = 131072; |
| 569 | socklen_t len; |
| 570 | struct ip_mreq mreq; |
| 571 | struct ipv6_mreq mreq6; |
| 572 | disorder_client *c; |
| 573 | char *address, *port; |
| 574 | int is_multicast; |
| 575 | union any_sockaddr { |
| 576 | struct sockaddr sa; |
| 577 | struct sockaddr_in in; |
| 578 | struct sockaddr_in6 in6; |
| 579 | }; |
| 580 | union any_sockaddr mgroup; |
| 581 | const char *dumpfile = 0; |
| 582 | pthread_t ltid; |
| 583 | |
| 584 | static const struct addrinfo prefs = { |
| 585 | .ai_flags = AI_PASSIVE, |
| 586 | .ai_family = PF_INET, |
| 587 | .ai_socktype = SOCK_DGRAM, |
| 588 | .ai_protocol = IPPROTO_UDP |
| 589 | }; |
| 590 | |
| 591 | mem_init(); |
| 592 | if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); |
| 593 | backend = uaudio_apis[0]; |
| 594 | while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:aocC:re:", options, 0)) >= 0) { |
| 595 | switch(n) { |
| 596 | case 'h': help(); |
| 597 | case 'V': version("disorder-playrtp"); |
| 598 | case 'd': debugging = 1; break; |
| 599 | case 'D': uaudio_set("device", optarg); break; |
| 600 | case 'm': minbuffer = 2 * atol(optarg); break; |
| 601 | case 'b': readahead = 2 * atol(optarg); break; |
| 602 | case 'x': maxbuffer = 2 * atol(optarg); break; |
| 603 | case 'L': logfp = fopen(optarg, "w"); break; |
| 604 | case 'R': target_rcvbuf = atoi(optarg); break; |
| 605 | #if HAVE_ALSA_ASOUNDLIB_H |
| 606 | case 'a': backend = &uaudio_alsa; break; |
| 607 | #endif |
| 608 | #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST |
| 609 | case 'o': backend = &uaudio_oss; break; |
| 610 | #endif |
| 611 | #if HAVE_COREAUDIO_AUDIOHARDWARE_H |
| 612 | case 'c': backend = &uaudio_coreaudio; break; |
| 613 | #endif |
| 614 | case 'C': configfile = optarg; break; |
| 615 | case 's': control_socket = optarg; break; |
| 616 | case 'r': dumpfile = optarg; break; |
| 617 | case 'e': backend = &uaudio_command; uaudio_set("command", optarg); break; |
| 618 | default: fatal(0, "invalid option"); |
| 619 | } |
| 620 | } |
| 621 | if(config_read(0)) fatal(0, "cannot read configuration"); |
| 622 | if(!maxbuffer) |
| 623 | maxbuffer = 4 * readahead; |
| 624 | argc -= optind; |
| 625 | argv += optind; |
| 626 | switch(argc) { |
| 627 | case 0: |
| 628 | /* Get configuration from server */ |
| 629 | if(!(c = disorder_new(1))) exit(EXIT_FAILURE); |
| 630 | if(disorder_connect(c)) exit(EXIT_FAILURE); |
| 631 | if(disorder_rtp_address(c, &address, &port)) exit(EXIT_FAILURE); |
| 632 | sl.n = 2; |
| 633 | sl.s = xcalloc(2, sizeof *sl.s); |
| 634 | sl.s[0] = address; |
| 635 | sl.s[1] = port; |
| 636 | break; |
| 637 | case 1: |
| 638 | case 2: |
| 639 | /* Use command-line ADDRESS+PORT or just PORT */ |
| 640 | sl.n = argc; |
| 641 | sl.s = argv; |
| 642 | break; |
| 643 | default: |
| 644 | fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]"); |
| 645 | } |
| 646 | /* Look up address and port */ |
| 647 | if(!(res = get_address(&sl, &prefs, &sockname))) |
| 648 | exit(1); |
| 649 | /* Create the socket */ |
| 650 | if((rtpfd = socket(res->ai_family, |
| 651 | res->ai_socktype, |
| 652 | res->ai_protocol)) < 0) |
| 653 | fatal(errno, "error creating socket"); |
| 654 | /* Stash the multicast group address */ |
| 655 | if((is_multicast = multicast(res->ai_addr))) { |
| 656 | memcpy(&mgroup, res->ai_addr, res->ai_addrlen); |
| 657 | switch(res->ai_addr->sa_family) { |
| 658 | case AF_INET: |
| 659 | mgroup.in.sin_port = 0; |
| 660 | break; |
| 661 | case AF_INET6: |
| 662 | mgroup.in6.sin6_port = 0; |
| 663 | break; |
| 664 | } |
| 665 | } |
| 666 | /* Bind to 0/port */ |
| 667 | switch(res->ai_addr->sa_family) { |
| 668 | case AF_INET: |
| 669 | memset(&((struct sockaddr_in *)res->ai_addr)->sin_addr, 0, |
| 670 | sizeof (struct in_addr)); |
| 671 | break; |
| 672 | case AF_INET6: |
| 673 | memset(&((struct sockaddr_in6 *)res->ai_addr)->sin6_addr, 0, |
| 674 | sizeof (struct in6_addr)); |
| 675 | break; |
| 676 | default: |
| 677 | fatal(0, "unsupported family %d", (int)res->ai_addr->sa_family); |
| 678 | } |
| 679 | if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0) |
| 680 | fatal(errno, "error binding socket to %s", sockname); |
| 681 | if(is_multicast) { |
| 682 | switch(mgroup.sa.sa_family) { |
| 683 | case PF_INET: |
| 684 | mreq.imr_multiaddr = mgroup.in.sin_addr; |
| 685 | mreq.imr_interface.s_addr = 0; /* use primary interface */ |
| 686 | if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP, |
| 687 | &mreq, sizeof mreq) < 0) |
| 688 | fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP"); |
| 689 | break; |
| 690 | case PF_INET6: |
| 691 | mreq6.ipv6mr_multiaddr = mgroup.in6.sin6_addr; |
| 692 | memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface); |
| 693 | if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP, |
| 694 | &mreq6, sizeof mreq6) < 0) |
| 695 | fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP"); |
| 696 | break; |
| 697 | default: |
| 698 | fatal(0, "unsupported address family %d", res->ai_family); |
| 699 | } |
| 700 | info("listening on %s multicast group %s", |
| 701 | format_sockaddr(res->ai_addr), format_sockaddr(&mgroup.sa)); |
| 702 | } else |
| 703 | info("listening on %s", format_sockaddr(res->ai_addr)); |
| 704 | len = sizeof rcvbuf; |
| 705 | if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0) |
| 706 | fatal(errno, "error calling getsockopt SO_RCVBUF"); |
| 707 | if(target_rcvbuf > rcvbuf) { |
| 708 | if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, |
| 709 | &target_rcvbuf, sizeof target_rcvbuf) < 0) |
| 710 | error(errno, "error calling setsockopt SO_RCVBUF %d", |
| 711 | target_rcvbuf); |
| 712 | /* We try to carry on anyway */ |
| 713 | else |
| 714 | info("changed socket receive buffer from %d to %d", |
| 715 | rcvbuf, target_rcvbuf); |
| 716 | } else |
| 717 | info("default socket receive buffer %d", rcvbuf); |
| 718 | if(logfp) |
| 719 | info("WARNING: -L option can impact performance"); |
| 720 | if(control_socket) { |
| 721 | pthread_t tid; |
| 722 | |
| 723 | if((err = pthread_create(&tid, 0, control_thread, 0))) |
| 724 | fatal(err, "pthread_create control_thread"); |
| 725 | } |
| 726 | if(dumpfile) { |
| 727 | int fd; |
| 728 | unsigned char buffer[65536]; |
| 729 | size_t written; |
| 730 | |
| 731 | if((fd = open(dumpfile, O_RDWR|O_TRUNC|O_CREAT, 0666)) < 0) |
| 732 | fatal(errno, "opening %s", dumpfile); |
| 733 | /* Fill with 0s to a suitable size */ |
| 734 | memset(buffer, 0, sizeof buffer); |
| 735 | for(written = 0; written < dump_size * sizeof(int16_t); |
| 736 | written += sizeof buffer) { |
| 737 | if(write(fd, buffer, sizeof buffer) < 0) |
| 738 | fatal(errno, "clearing %s", dumpfile); |
| 739 | } |
| 740 | /* Map the buffer into memory for convenience */ |
| 741 | dump_buffer = mmap(0, dump_size * sizeof(int16_t), PROT_READ|PROT_WRITE, |
| 742 | MAP_SHARED, fd, 0); |
| 743 | if(dump_buffer == (void *)-1) |
| 744 | fatal(errno, "mapping %s", dumpfile); |
| 745 | info("dumping to %s", dumpfile); |
| 746 | } |
| 747 | /* Set up output. Currently we only support L16 so there's no harm setting |
| 748 | * the format before we know what it is! */ |
| 749 | uaudio_set_format(44100/*Hz*/, 2/*channels*/, |
| 750 | 16/*bits/channel*/, 1/*signed*/); |
| 751 | backend->start(playrtp_callback, NULL); |
| 752 | /* We receive and convert audio data in a background thread */ |
| 753 | if((err = pthread_create(<id, 0, listen_thread, 0))) |
| 754 | fatal(err, "pthread_create listen_thread"); |
| 755 | /* We have a second thread to add received packets to the queue */ |
| 756 | if((err = pthread_create(<id, 0, queue_thread, 0))) |
| 757 | fatal(err, "pthread_create queue_thread"); |
| 758 | pthread_mutex_lock(&lock); |
| 759 | for(;;) { |
| 760 | /* Wait for the buffer to fill up a bit */ |
| 761 | playrtp_fill_buffer(); |
| 762 | /* Start playing now */ |
| 763 | info("Playing..."); |
| 764 | next_timestamp = pheap_first(&packets)->timestamp; |
| 765 | active = 1; |
| 766 | pthread_mutex_unlock(&lock); |
| 767 | backend->activate(); |
| 768 | pthread_mutex_lock(&lock); |
| 769 | /* Wait until the buffer empties out */ |
| 770 | while(nsamples >= minbuffer |
| 771 | || (nsamples > 0 |
| 772 | && contains(pheap_first(&packets), next_timestamp))) { |
| 773 | pthread_cond_wait(&cond, &lock); |
| 774 | } |
| 775 | /* Stop playing for a bit until the buffer re-fills */ |
| 776 | pthread_mutex_unlock(&lock); |
| 777 | backend->deactivate(); |
| 778 | pthread_mutex_lock(&lock); |
| 779 | active = 0; |
| 780 | /* Go back round */ |
| 781 | } |
| 782 | return 0; |
| 783 | } |
| 784 | |
| 785 | /* |
| 786 | Local Variables: |
| 787 | c-basic-offset:2 |
| 788 | comment-column:40 |
| 789 | fill-column:79 |
| 790 | indent-tabs-mode:nil |
| 791 | End: |
| 792 | */ |