| 1 | /* |
| 2 | * This file is part of DisOrder. |
| 3 | * Copyright (C) 2007-2009, 2011, 2013 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: |
| 28 | * |
| 29 | * listen_thread() is responsible for reading RTP packets off the wire and |
| 30 | * adding them to the linked list @ref received_packets, assuming they are |
| 31 | * basically sound. |
| 32 | * |
| 33 | * queue_thread() takes packets off this linked list and adds them to @ref |
| 34 | * packets (an operation which might be much slower due to contention for @ref |
| 35 | * lock). |
| 36 | * |
| 37 | * control_thread() accepts commands from Disobedience (or anything else). |
| 38 | * |
| 39 | * The main thread activates and deactivates audio playing via the @ref |
| 40 | * lib/uaudio.h API (which probably implies at least one further thread). |
| 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 | #include <math.h> |
| 68 | #include <arpa/inet.h> |
| 69 | #include <ifaddrs.h> |
| 70 | #include <net/if.h> |
| 71 | |
| 72 | #include "log.h" |
| 73 | #include "mem.h" |
| 74 | #include "configuration.h" |
| 75 | #include "addr.h" |
| 76 | #include "syscalls.h" |
| 77 | #include "rtp.h" |
| 78 | #include "defs.h" |
| 79 | #include "vector.h" |
| 80 | #include "heap.h" |
| 81 | #include "timeval.h" |
| 82 | #include "client.h" |
| 83 | #include "playrtp.h" |
| 84 | #include "inputline.h" |
| 85 | #include "version.h" |
| 86 | #include "uaudio.h" |
| 87 | |
| 88 | /** @brief Obsolete synonym */ |
| 89 | #ifndef IPV6_JOIN_GROUP |
| 90 | # define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP |
| 91 | #endif |
| 92 | |
| 93 | /** @brief RTP socket */ |
| 94 | static int rtpfd; |
| 95 | |
| 96 | /** @brief Log output */ |
| 97 | static FILE *logfp; |
| 98 | |
| 99 | /** @brief Output device */ |
| 100 | |
| 101 | /** @brief Buffer low watermark in samples */ |
| 102 | unsigned minbuffer = 4 * (2 * 44100) / 10; /* 0.4 seconds */ |
| 103 | |
| 104 | /** @brief Maximum buffer size in samples |
| 105 | * |
| 106 | * We'll stop reading from the network if we have this many samples. |
| 107 | */ |
| 108 | static unsigned maxbuffer; |
| 109 | |
| 110 | /** @brief Received packets |
| 111 | * Protected by @ref receive_lock |
| 112 | * |
| 113 | * Received packets are added to this list, and queue_thread() picks them off |
| 114 | * it and adds them to @ref packets. Whenever a packet is added to it, @ref |
| 115 | * receive_cond is signalled. |
| 116 | */ |
| 117 | struct packet *received_packets; |
| 118 | |
| 119 | /** @brief Tail of @ref received_packets |
| 120 | * Protected by @ref receive_lock |
| 121 | */ |
| 122 | struct packet **received_tail = &received_packets; |
| 123 | |
| 124 | /** @brief Lock protecting @ref received_packets |
| 125 | * |
| 126 | * Only listen_thread() and queue_thread() ever hold this lock. It is vital |
| 127 | * that queue_thread() not hold it any longer than it strictly has to. */ |
| 128 | pthread_mutex_t receive_lock = PTHREAD_MUTEX_INITIALIZER; |
| 129 | |
| 130 | /** @brief Condition variable signalled when @ref received_packets is updated |
| 131 | * |
| 132 | * Used by listen_thread() to notify queue_thread() that it has added another |
| 133 | * packet to @ref received_packets. */ |
| 134 | pthread_cond_t receive_cond = PTHREAD_COND_INITIALIZER; |
| 135 | |
| 136 | /** @brief Length of @ref received_packets */ |
| 137 | uint32_t nreceived; |
| 138 | |
| 139 | /** @brief Binary heap of received packets */ |
| 140 | struct pheap packets; |
| 141 | |
| 142 | /** @brief Total number of samples available |
| 143 | * |
| 144 | * We make this volatile because we inspect it without a protecting lock, |
| 145 | * so the usual pthread_* guarantees aren't available. |
| 146 | */ |
| 147 | volatile uint32_t nsamples; |
| 148 | |
| 149 | /** @brief Timestamp of next packet to play. |
| 150 | * |
| 151 | * This is set to the timestamp of the last packet, plus the number of |
| 152 | * samples it contained. Only valid if @ref active is nonzero. |
| 153 | */ |
| 154 | uint32_t next_timestamp; |
| 155 | |
| 156 | /** @brief True if actively playing |
| 157 | * |
| 158 | * This is true when playing and false when just buffering. */ |
| 159 | int active; |
| 160 | |
| 161 | /** @brief Lock protecting @ref packets */ |
| 162 | pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER; |
| 163 | |
| 164 | /** @brief Condition variable signalled whenever @ref packets is changed */ |
| 165 | pthread_cond_t cond = PTHREAD_COND_INITIALIZER; |
| 166 | |
| 167 | /** @brief Backend to play with */ |
| 168 | static const struct uaudio *backend; |
| 169 | |
| 170 | HEAP_DEFINE(pheap, struct packet *, lt_packet); |
| 171 | |
| 172 | /** @brief Control socket or NULL */ |
| 173 | const char *control_socket; |
| 174 | |
| 175 | /** @brief Buffer for debugging dump |
| 176 | * |
| 177 | * The debug dump is enabled by the @c --dump option. It records the last 20s |
| 178 | * of audio to the specified file (which will be about 3.5Mbytes). The file is |
| 179 | * written as as ring buffer, so the start point will progress through it. |
| 180 | * |
| 181 | * Use clients/dump2wav to convert this to a WAV file, which can then be loaded |
| 182 | * into (e.g.) Audacity for further inspection. |
| 183 | * |
| 184 | * All three backends (ALSA, OSS, Core Audio) now support this option. |
| 185 | * |
| 186 | * The idea is to allow the user a few seconds to react to an audible artefact. |
| 187 | */ |
| 188 | int16_t *dump_buffer; |
| 189 | |
| 190 | /** @brief Current index within debugging dump */ |
| 191 | size_t dump_index; |
| 192 | |
| 193 | /** @brief Size of debugging dump in samples */ |
| 194 | size_t dump_size = 44100/*Hz*/ * 2/*channels*/ * 20/*seconds*/; |
| 195 | |
| 196 | static const struct option options[] = { |
| 197 | { "help", no_argument, 0, 'h' }, |
| 198 | { "version", no_argument, 0, 'V' }, |
| 199 | { "debug", no_argument, 0, 'd' }, |
| 200 | { "device", required_argument, 0, 'D' }, |
| 201 | { "min", required_argument, 0, 'm' }, |
| 202 | { "max", required_argument, 0, 'x' }, |
| 203 | { "rcvbuf", required_argument, 0, 'R' }, |
| 204 | #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST |
| 205 | { "oss", no_argument, 0, 'o' }, |
| 206 | #endif |
| 207 | #if HAVE_ALSA_ASOUNDLIB_H |
| 208 | { "alsa", no_argument, 0, 'a' }, |
| 209 | #endif |
| 210 | #if HAVE_COREAUDIO_AUDIOHARDWARE_H |
| 211 | { "core-audio", no_argument, 0, 'c' }, |
| 212 | #endif |
| 213 | { "api", required_argument, 0, 'A' }, |
| 214 | { "dump", required_argument, 0, 'r' }, |
| 215 | { "command", required_argument, 0, 'e' }, |
| 216 | { "pause-mode", required_argument, 0, 'P' }, |
| 217 | { "socket", required_argument, 0, 's' }, |
| 218 | { "config", required_argument, 0, 'C' }, |
| 219 | { "monitor", no_argument, 0, 'M' }, |
| 220 | { 0, 0, 0, 0 } |
| 221 | }; |
| 222 | |
| 223 | /** @brief Control thread |
| 224 | * |
| 225 | * This thread is responsible for accepting control commands from Disobedience |
| 226 | * (or other controllers) over an AF_UNIX stream socket with a path specified |
| 227 | * by the @c --socket option. The protocol uses simple string commands and |
| 228 | * replies: |
| 229 | * |
| 230 | * - @c stop will shut the player down |
| 231 | * - @c query will send back the reply @c running |
| 232 | * - anything else is ignored |
| 233 | * |
| 234 | * Commands and response strings terminated by shutting down the connection or |
| 235 | * by a newline. No attempt is made to multiplex multiple clients so it is |
| 236 | * important that the command be sent as soon as the connection is made - it is |
| 237 | * assumed that both parties to the protocol are entirely cooperating with one |
| 238 | * another. |
| 239 | */ |
| 240 | static void *control_thread(void attribute((unused)) *arg) { |
| 241 | struct sockaddr_un sa; |
| 242 | int sfd, cfd; |
| 243 | char *line; |
| 244 | socklen_t salen; |
| 245 | FILE *fp; |
| 246 | |
| 247 | assert(control_socket); |
| 248 | unlink(control_socket); |
| 249 | memset(&sa, 0, sizeof sa); |
| 250 | sa.sun_family = AF_UNIX; |
| 251 | strcpy(sa.sun_path, control_socket); |
| 252 | sfd = xsocket(PF_UNIX, SOCK_STREAM, 0); |
| 253 | if(bind(sfd, (const struct sockaddr *)&sa, sizeof sa) < 0) |
| 254 | disorder_fatal(errno, "error binding to %s", control_socket); |
| 255 | if(listen(sfd, 128) < 0) |
| 256 | disorder_fatal(errno, "error calling listen on %s", control_socket); |
| 257 | disorder_info("listening on %s", control_socket); |
| 258 | for(;;) { |
| 259 | salen = sizeof sa; |
| 260 | cfd = accept(sfd, (struct sockaddr *)&sa, &salen); |
| 261 | if(cfd < 0) { |
| 262 | switch(errno) { |
| 263 | case EINTR: |
| 264 | case EAGAIN: |
| 265 | break; |
| 266 | default: |
| 267 | disorder_fatal(errno, "error calling accept on %s", control_socket); |
| 268 | } |
| 269 | } |
| 270 | if(!(fp = fdopen(cfd, "r+"))) { |
| 271 | disorder_error(errno, "error calling fdopen for %s connection", control_socket); |
| 272 | close(cfd); |
| 273 | continue; |
| 274 | } |
| 275 | if(!inputline(control_socket, fp, &line, '\n')) { |
| 276 | if(!strcmp(line, "stop")) { |
| 277 | disorder_info("stopped via %s", control_socket); |
| 278 | exit(0); /* terminate immediately */ |
| 279 | } |
| 280 | if(!strcmp(line, "query")) |
| 281 | fprintf(fp, "running"); |
| 282 | xfree(line); |
| 283 | } |
| 284 | if(fclose(fp) < 0) |
| 285 | disorder_error(errno, "error closing %s connection", control_socket); |
| 286 | } |
| 287 | } |
| 288 | |
| 289 | /** @brief Drop the first packet |
| 290 | * |
| 291 | * Assumes that @ref lock is held. |
| 292 | */ |
| 293 | static void drop_first_packet(void) { |
| 294 | if(pheap_count(&packets)) { |
| 295 | struct packet *const p = pheap_remove(&packets); |
| 296 | nsamples -= p->nsamples; |
| 297 | playrtp_free_packet(p); |
| 298 | pthread_cond_broadcast(&cond); |
| 299 | } |
| 300 | } |
| 301 | |
| 302 | /** @brief Background thread adding packets to heap |
| 303 | * |
| 304 | * This just transfers packets from @ref received_packets to @ref packets. It |
| 305 | * is important that it holds @ref receive_lock for as little time as possible, |
| 306 | * in order to minimize the interval between calls to read() in |
| 307 | * listen_thread(). |
| 308 | */ |
| 309 | static void *queue_thread(void attribute((unused)) *arg) { |
| 310 | struct packet *p; |
| 311 | |
| 312 | for(;;) { |
| 313 | /* Get the next packet */ |
| 314 | pthread_mutex_lock(&receive_lock); |
| 315 | while(!received_packets) { |
| 316 | pthread_cond_wait(&receive_cond, &receive_lock); |
| 317 | } |
| 318 | p = received_packets; |
| 319 | received_packets = p->next; |
| 320 | if(!received_packets) |
| 321 | received_tail = &received_packets; |
| 322 | --nreceived; |
| 323 | pthread_mutex_unlock(&receive_lock); |
| 324 | /* Add it to the heap */ |
| 325 | pthread_mutex_lock(&lock); |
| 326 | pheap_insert(&packets, p); |
| 327 | nsamples += p->nsamples; |
| 328 | pthread_cond_broadcast(&cond); |
| 329 | pthread_mutex_unlock(&lock); |
| 330 | } |
| 331 | #if HAVE_STUPID_GCC44 |
| 332 | return NULL; |
| 333 | #endif |
| 334 | } |
| 335 | |
| 336 | /** @brief Background thread collecting samples |
| 337 | * |
| 338 | * This function collects samples, perhaps converts them to the target format, |
| 339 | * and adds them to the packet list. |
| 340 | * |
| 341 | * It is crucial that the gap between successive calls to read() is as small as |
| 342 | * possible: otherwise packets will be dropped. |
| 343 | * |
| 344 | * We use a binary heap to ensure that the unavoidable effort is at worst |
| 345 | * logarithmic in the total number of packets - in fact if packets are mostly |
| 346 | * received in order then we will largely do constant work per packet since the |
| 347 | * newest packet will always be last. |
| 348 | * |
| 349 | * Of more concern is that we must acquire the lock on the heap to add a packet |
| 350 | * to it. If this proves a problem in practice then the answer would be |
| 351 | * (probably doubly) linked list with new packets added the end and a second |
| 352 | * thread which reads packets off the list and adds them to the heap. |
| 353 | * |
| 354 | * We keep memory allocation (mostly) very fast by keeping pre-allocated |
| 355 | * packets around; see @ref playrtp_new_packet(). |
| 356 | */ |
| 357 | static void *listen_thread(void attribute((unused)) *arg) { |
| 358 | struct packet *p = 0; |
| 359 | int n; |
| 360 | struct rtp_header header; |
| 361 | uint16_t seq; |
| 362 | uint32_t timestamp; |
| 363 | struct iovec iov[2]; |
| 364 | |
| 365 | for(;;) { |
| 366 | if(!p) |
| 367 | p = playrtp_new_packet(); |
| 368 | iov[0].iov_base = &header; |
| 369 | iov[0].iov_len = sizeof header; |
| 370 | iov[1].iov_base = p->samples_raw; |
| 371 | iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw; |
| 372 | n = readv(rtpfd, iov, 2); |
| 373 | if(n < 0) { |
| 374 | switch(errno) { |
| 375 | case EINTR: |
| 376 | continue; |
| 377 | default: |
| 378 | disorder_fatal(errno, "error reading from socket"); |
| 379 | } |
| 380 | } |
| 381 | /* Ignore too-short packets */ |
| 382 | if((size_t)n <= sizeof (struct rtp_header)) { |
| 383 | disorder_info("ignored a short packet"); |
| 384 | continue; |
| 385 | } |
| 386 | timestamp = htonl(header.timestamp); |
| 387 | seq = htons(header.seq); |
| 388 | /* Ignore packets in the past */ |
| 389 | if(active && lt(timestamp, next_timestamp)) { |
| 390 | disorder_info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32, |
| 391 | timestamp, next_timestamp); |
| 392 | continue; |
| 393 | } |
| 394 | /* Ignore packets with the extension bit set. */ |
| 395 | if(header.vpxcc & 0x10) |
| 396 | continue; |
| 397 | p->next = 0; |
| 398 | p->flags = 0; |
| 399 | p->timestamp = timestamp; |
| 400 | /* Convert to target format */ |
| 401 | if(header.mpt & 0x80) |
| 402 | p->flags |= IDLE; |
| 403 | switch(header.mpt & 0x7F) { |
| 404 | case 10: /* L16 */ |
| 405 | p->nsamples = (n - sizeof header) / sizeof(uint16_t); |
| 406 | break; |
| 407 | /* TODO support other RFC3551 media types (when the speaker does) */ |
| 408 | default: |
| 409 | disorder_fatal(0, "unsupported RTP payload type %d", header.mpt & 0x7F); |
| 410 | } |
| 411 | /* See if packet is silent */ |
| 412 | const uint16_t *s = p->samples_raw; |
| 413 | n = p->nsamples; |
| 414 | for(; n > 0; --n) |
| 415 | if(*s++) |
| 416 | break; |
| 417 | if(!n) |
| 418 | p->flags |= SILENT; |
| 419 | if(logfp) |
| 420 | fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n", |
| 421 | seq, timestamp, p->nsamples, timestamp + p->nsamples); |
| 422 | /* Stop reading if we've reached the maximum. |
| 423 | * |
| 424 | * This is rather unsatisfactory: it means that if packets get heavily |
| 425 | * out of order then we guarantee dropouts. But for now... */ |
| 426 | if(nsamples >= maxbuffer) { |
| 427 | pthread_mutex_lock(&lock); |
| 428 | while(nsamples >= maxbuffer) { |
| 429 | pthread_cond_wait(&cond, &lock); |
| 430 | } |
| 431 | pthread_mutex_unlock(&lock); |
| 432 | } |
| 433 | /* Add the packet to the receive queue */ |
| 434 | pthread_mutex_lock(&receive_lock); |
| 435 | *received_tail = p; |
| 436 | received_tail = &p->next; |
| 437 | ++nreceived; |
| 438 | pthread_cond_signal(&receive_cond); |
| 439 | pthread_mutex_unlock(&receive_lock); |
| 440 | /* We'll need a new packet */ |
| 441 | p = 0; |
| 442 | } |
| 443 | } |
| 444 | |
| 445 | /** @brief Wait until the buffer is adequately full |
| 446 | * |
| 447 | * Must be called with @ref lock held. |
| 448 | */ |
| 449 | void playrtp_fill_buffer(void) { |
| 450 | /* Discard current buffer contents */ |
| 451 | while(nsamples) { |
| 452 | //fprintf(stderr, "%8u/%u (%u) DROPPING\n", nsamples, maxbuffer, minbuffer); |
| 453 | drop_first_packet(); |
| 454 | } |
| 455 | disorder_info("Buffering..."); |
| 456 | /* Wait until there's at least minbuffer samples available */ |
| 457 | while(nsamples < minbuffer) { |
| 458 | //fprintf(stderr, "%8u/%u (%u) FILLING\n", nsamples, maxbuffer, minbuffer); |
| 459 | pthread_cond_wait(&cond, &lock); |
| 460 | } |
| 461 | /* Start from whatever is earliest */ |
| 462 | next_timestamp = pheap_first(&packets)->timestamp; |
| 463 | active = 1; |
| 464 | } |
| 465 | |
| 466 | /** @brief Find next packet |
| 467 | * @return Packet to play or NULL if none found |
| 468 | * |
| 469 | * The return packet is merely guaranteed not to be in the past: it might be |
| 470 | * the first packet in the future rather than one that is actually suitable to |
| 471 | * play. |
| 472 | * |
| 473 | * Must be called with @ref lock held. |
| 474 | */ |
| 475 | struct packet *playrtp_next_packet(void) { |
| 476 | while(pheap_count(&packets)) { |
| 477 | struct packet *const p = pheap_first(&packets); |
| 478 | if(le(p->timestamp + p->nsamples, next_timestamp)) { |
| 479 | /* This packet is in the past. Drop it and try another one. */ |
| 480 | drop_first_packet(); |
| 481 | } else |
| 482 | /* This packet is NOT in the past. (It might be in the future |
| 483 | * however.) */ |
| 484 | return p; |
| 485 | } |
| 486 | return 0; |
| 487 | } |
| 488 | |
| 489 | /* display usage message and terminate */ |
| 490 | static void help(void) { |
| 491 | xprintf("Usage:\n" |
| 492 | " disorder-playrtp [OPTIONS] [[ADDRESS] PORT]\n" |
| 493 | "Options:\n" |
| 494 | " --device, -D DEVICE Output device\n" |
| 495 | " --min, -m FRAMES Buffer low water mark\n" |
| 496 | " --max, -x FRAMES Buffer maximum size\n" |
| 497 | " --rcvbuf, -R BYTES Socket receive buffer size\n" |
| 498 | " --config, -C PATH Set configuration file\n" |
| 499 | " --api, -A API Select audio API. Possibilities:\n" |
| 500 | " "); |
| 501 | int first = 1; |
| 502 | for(int n = 0; uaudio_apis[n]; ++n) { |
| 503 | if(uaudio_apis[n]->flags & UAUDIO_API_CLIENT) { |
| 504 | if(first) |
| 505 | first = 0; |
| 506 | else |
| 507 | xprintf(", "); |
| 508 | xprintf("%s", uaudio_apis[n]->name); |
| 509 | } |
| 510 | } |
| 511 | xprintf("\n" |
| 512 | " --command, -e COMMAND Pipe audio to command.\n" |
| 513 | " --pause-mode, -P silence For -e: pauses send silence (default)\n" |
| 514 | " --pause-mode, -P suspend For -e: pauses suspend writes\n" |
| 515 | " --help, -h Display usage message\n" |
| 516 | " --version, -V Display version number\n" |
| 517 | ); |
| 518 | xfclose(stdout); |
| 519 | exit(0); |
| 520 | } |
| 521 | |
| 522 | static size_t playrtp_callback(void *buffer, |
| 523 | size_t max_samples, |
| 524 | void attribute((unused)) *userdata) { |
| 525 | size_t samples; |
| 526 | int silent = 0; |
| 527 | |
| 528 | pthread_mutex_lock(&lock); |
| 529 | /* Get the next packet, junking any that are now in the past */ |
| 530 | const struct packet *p = playrtp_next_packet(); |
| 531 | if(p && contains(p, next_timestamp)) { |
| 532 | /* This packet is ready to play; the desired next timestamp points |
| 533 | * somewhere into it. */ |
| 534 | |
| 535 | /* Timestamp of end of packet */ |
| 536 | const uint32_t packet_end = p->timestamp + p->nsamples; |
| 537 | |
| 538 | /* Offset of desired next timestamp into current packet */ |
| 539 | const uint32_t offset = next_timestamp - p->timestamp; |
| 540 | |
| 541 | /* Pointer to audio data */ |
| 542 | const uint16_t *ptr = (void *)(p->samples_raw + offset); |
| 543 | |
| 544 | /* Compute number of samples left in packet, limited to output buffer |
| 545 | * size */ |
| 546 | samples = packet_end - next_timestamp; |
| 547 | if(samples > max_samples) |
| 548 | samples = max_samples; |
| 549 | |
| 550 | /* Copy into buffer, converting to native endianness */ |
| 551 | size_t i = samples; |
| 552 | int16_t *bufptr = buffer; |
| 553 | while(i > 0) { |
| 554 | *bufptr++ = (int16_t)ntohs(*ptr++); |
| 555 | --i; |
| 556 | } |
| 557 | silent = !!(p->flags & SILENT); |
| 558 | } else { |
| 559 | /* There is no suitable packet. We introduce 0s up to the next packet, or |
| 560 | * to fill the buffer if there's no next packet or that's too many. The |
| 561 | * comparison with max_samples deals with the otherwise troubling overflow |
| 562 | * case. */ |
| 563 | samples = p ? p->timestamp - next_timestamp : max_samples; |
| 564 | if(samples > max_samples) |
| 565 | samples = max_samples; |
| 566 | //info("infill by %zu", samples); |
| 567 | memset(buffer, 0, samples * uaudio_sample_size); |
| 568 | silent = 1; |
| 569 | } |
| 570 | /* Debug dump */ |
| 571 | if(dump_buffer) { |
| 572 | for(size_t i = 0; i < samples; ++i) { |
| 573 | dump_buffer[dump_index++] = ((int16_t *)buffer)[i]; |
| 574 | dump_index %= dump_size; |
| 575 | } |
| 576 | } |
| 577 | /* Advance timestamp */ |
| 578 | next_timestamp += samples; |
| 579 | /* If we're getting behind then try to drop just silent packets |
| 580 | * |
| 581 | * In theory this shouldn't be necessary. The server is supposed to send |
| 582 | * packets at the right rate and compares the number of samples sent with the |
| 583 | * time in order to ensure this. |
| 584 | * |
| 585 | * However, various things could throw this off: |
| 586 | * |
| 587 | * - the server's clock could advance at the wrong rate. This would cause it |
| 588 | * to mis-estimate the right number of samples to have sent and |
| 589 | * inappropriately throttle or speed up. |
| 590 | * |
| 591 | * - playback could happen at the wrong rate. If the playback host's sound |
| 592 | * card has a slightly incorrect clock then eventually it will get out |
| 593 | * of step. |
| 594 | * |
| 595 | * So if we play back slightly slower than the server sends for either of |
| 596 | * these reasons then eventually our buffer, and the socket's buffer, will |
| 597 | * fill, and the kernel will start dropping packets. The result is audible |
| 598 | * and not very nice. |
| 599 | * |
| 600 | * Therefore if we're getting behind, we pre-emptively drop silent packets, |
| 601 | * since a change in the duration of a silence is less noticeable than a |
| 602 | * dropped packet from the middle of continuous music. |
| 603 | * |
| 604 | * (If things go wrong the other way then eventually we run out of packets to |
| 605 | * play and are forced to play silence. This doesn't seem to happen in |
| 606 | * practice but if it does then in the same way we can artificially extend |
| 607 | * silent packets to compensate.) |
| 608 | * |
| 609 | * Dropped packets are always logged; use 'disorder-playrtp --monitor' to |
| 610 | * track how close to target buffer occupancy we are on a once-a-minute |
| 611 | * basis. |
| 612 | */ |
| 613 | if(nsamples > minbuffer && silent) { |
| 614 | disorder_info("dropping %zu samples (%"PRIu32" > %"PRIu32")", |
| 615 | samples, nsamples, minbuffer); |
| 616 | samples = 0; |
| 617 | } |
| 618 | /* Junk obsolete packets */ |
| 619 | playrtp_next_packet(); |
| 620 | pthread_mutex_unlock(&lock); |
| 621 | return samples; |
| 622 | } |
| 623 | |
| 624 | static int compare_family(const struct ifaddrs *a, |
| 625 | const struct ifaddrs *b, |
| 626 | int family) { |
| 627 | int afamily = a->ifa_addr->sa_family; |
| 628 | int bfamily = b->ifa_addr->sa_family; |
| 629 | if(afamily != bfamily) { |
| 630 | /* Preferred family wins */ |
| 631 | if(afamily == family) return 1; |
| 632 | if(bfamily == family) return -1; |
| 633 | /* Either there's no preference or it doesn't help. Prefer IPv4 */ |
| 634 | if(afamily == AF_INET) return 1; |
| 635 | if(bfamily == AF_INET) return -1; |
| 636 | /* Failing that prefer IPv6 */ |
| 637 | if(afamily == AF_INET6) return 1; |
| 638 | if(bfamily == AF_INET6) return -1; |
| 639 | } |
| 640 | return 0; |
| 641 | } |
| 642 | |
| 643 | static int compare_flags(const struct ifaddrs *a, |
| 644 | const struct ifaddrs *b) { |
| 645 | unsigned aflags = a->ifa_flags, bflags = b->ifa_flags; |
| 646 | /* Up interfaces are better than down ones */ |
| 647 | unsigned aup = aflags & IFF_UP, bup = bflags & IFF_UP; |
| 648 | if(aup != bup) |
| 649 | return aup > bup ? 1 : -1; |
| 650 | /* Static addresses are better than dynamic */ |
| 651 | unsigned adynamic = aflags & IFF_DYNAMIC, bdynamic = bflags & IFF_DYNAMIC; |
| 652 | if(adynamic != bdynamic) |
| 653 | return adynamic < bdynamic ? 1 : -1; |
| 654 | unsigned aloopback = aflags & IFF_LOOPBACK, bloopback = bflags & IFF_LOOPBACK; |
| 655 | /* Static addresses are better than dynamic */ |
| 656 | if(aloopback != bloopback) |
| 657 | return aloopback < bloopback ? 1 : -1; |
| 658 | return 0; |
| 659 | } |
| 660 | |
| 661 | static int compare_interfaces(const struct ifaddrs *a, |
| 662 | const struct ifaddrs *b, |
| 663 | int family) { |
| 664 | int c; |
| 665 | if((c = compare_family(a, b, family))) return c; |
| 666 | return compare_flags(a, b); |
| 667 | } |
| 668 | |
| 669 | int main(int argc, char **argv) { |
| 670 | int n, err; |
| 671 | struct addrinfo *res; |
| 672 | struct stringlist sl; |
| 673 | char *sockname; |
| 674 | int rcvbuf, target_rcvbuf = 0; |
| 675 | socklen_t len; |
| 676 | struct ip_mreq mreq; |
| 677 | struct ipv6_mreq mreq6; |
| 678 | disorder_client *c = NULL; |
| 679 | char *address, *port; |
| 680 | int is_multicast; |
| 681 | union any_sockaddr { |
| 682 | struct sockaddr sa; |
| 683 | struct sockaddr_in in; |
| 684 | struct sockaddr_in6 in6; |
| 685 | }; |
| 686 | union any_sockaddr mgroup; |
| 687 | const char *dumpfile = 0; |
| 688 | pthread_t ltid; |
| 689 | int monitor = 0; |
| 690 | static const int one = 1; |
| 691 | |
| 692 | static const struct addrinfo prefs = { |
| 693 | .ai_flags = AI_PASSIVE, |
| 694 | .ai_family = PF_INET, |
| 695 | .ai_socktype = SOCK_DGRAM, |
| 696 | .ai_protocol = IPPROTO_UDP |
| 697 | }; |
| 698 | |
| 699 | /* Timing information is often important to debugging playrtp, so we include |
| 700 | * timestamps in the logs */ |
| 701 | logdate = 1; |
| 702 | mem_init(); |
| 703 | if(!setlocale(LC_CTYPE, "")) disorder_fatal(errno, "error calling setlocale"); |
| 704 | while((n = getopt_long(argc, argv, "hVdD:m:x:L:R:aocC:re:P:MA:", options, 0)) >= 0) { |
| 705 | switch(n) { |
| 706 | case 'h': help(); |
| 707 | case 'V': version("disorder-playrtp"); |
| 708 | case 'd': debugging = 1; break; |
| 709 | case 'D': uaudio_set("device", optarg); break; |
| 710 | case 'm': minbuffer = 2 * atol(optarg); break; |
| 711 | case 'x': maxbuffer = 2 * atol(optarg); break; |
| 712 | case 'L': logfp = fopen(optarg, "w"); break; |
| 713 | case 'R': target_rcvbuf = atoi(optarg); break; |
| 714 | #if HAVE_ALSA_ASOUNDLIB_H |
| 715 | case 'a': |
| 716 | disorder_error(0, "deprecated option; use --api alsa instead"); |
| 717 | backend = &uaudio_alsa; break; |
| 718 | #endif |
| 719 | #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST |
| 720 | case 'o': |
| 721 | disorder_error(0, "deprecated option; use --api oss instead"); |
| 722 | backend = &uaudio_oss; |
| 723 | break; |
| 724 | #endif |
| 725 | #if HAVE_COREAUDIO_AUDIOHARDWARE_H |
| 726 | case 'c': |
| 727 | disorder_error(0, "deprecated option; use --api coreaudio instead"); |
| 728 | backend = &uaudio_coreaudio; |
| 729 | break; |
| 730 | #endif |
| 731 | case 'A': backend = uaudio_find(optarg); break; |
| 732 | case 'C': configfile = optarg; break; |
| 733 | case 's': control_socket = optarg; break; |
| 734 | case 'r': dumpfile = optarg; break; |
| 735 | case 'e': backend = &uaudio_command; uaudio_set("command", optarg); break; |
| 736 | case 'P': uaudio_set("pause-mode", optarg); break; |
| 737 | case 'M': monitor = 1; break; |
| 738 | default: disorder_fatal(0, "invalid option"); |
| 739 | } |
| 740 | } |
| 741 | if(config_read(0, NULL)) disorder_fatal(0, "cannot read configuration"); |
| 742 | if(!backend) { |
| 743 | backend = uaudio_default(uaudio_apis, UAUDIO_API_CLIENT); |
| 744 | if(!backend) |
| 745 | disorder_fatal(0, "no default uaudio API found"); |
| 746 | disorder_info("default audio API %s", backend->name); |
| 747 | } |
| 748 | if(backend == &uaudio_rtp) { |
| 749 | /* This means that you have NO local sound output. This can happen if you |
| 750 | * use a non-Apple GCC on a Mac (because it doesn't know how to compile |
| 751 | * CoreAudio/AudioHardware.h). */ |
| 752 | disorder_fatal(0, "cannot play RTP through RTP"); |
| 753 | } |
| 754 | if(!maxbuffer) |
| 755 | maxbuffer = 2 * minbuffer; |
| 756 | argc -= optind; |
| 757 | argv += optind; |
| 758 | switch(argc) { |
| 759 | case 0: |
| 760 | /* Get configuration from server */ |
| 761 | if(!(c = disorder_new(1))) exit(EXIT_FAILURE); |
| 762 | if(disorder_connect(c)) exit(EXIT_FAILURE); |
| 763 | if(disorder_rtp_address(c, &address, &port)) exit(EXIT_FAILURE); |
| 764 | sl.n = 2; |
| 765 | sl.s = xcalloc(2, sizeof *sl.s); |
| 766 | sl.s[0] = address; |
| 767 | sl.s[1] = port; |
| 768 | break; |
| 769 | case 1: |
| 770 | case 2: |
| 771 | /* Use command-line ADDRESS+PORT or just PORT */ |
| 772 | sl.n = argc; |
| 773 | sl.s = argv; |
| 774 | break; |
| 775 | default: |
| 776 | disorder_fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]"); |
| 777 | } |
| 778 | disorder_info("version "VERSION" process ID %lu", |
| 779 | (unsigned long)getpid()); |
| 780 | struct sockaddr *addr; |
| 781 | socklen_t addr_len; |
| 782 | if(!strcmp(sl.s[0], "-")) { |
| 783 | /* Pick address family to match known-working connectivity to the server */ |
| 784 | int family = disorder_client_af(c); |
| 785 | /* Get a list of interfaces */ |
| 786 | struct ifaddrs *ifa, *bestifa = NULL; |
| 787 | if(getifaddrs(&ifa) < 0) |
| 788 | disorder_fatal(errno, "error calling getifaddrs"); |
| 789 | /* Try to pick a good one */ |
| 790 | for(; ifa; ifa = ifa->ifa_next) { |
| 791 | if(bestifa == NULL |
| 792 | || compare_interfaces(ifa, bestifa, family) > 0) |
| 793 | bestifa = ifa; |
| 794 | } |
| 795 | if(!bestifa) |
| 796 | disorder_fatal(0, "failed to select a network interface"); |
| 797 | family = bestifa->ifa_addr->sa_family; |
| 798 | if((rtpfd = socket(family, |
| 799 | SOCK_DGRAM, |
| 800 | IPPROTO_UDP)) < 0) |
| 801 | disorder_fatal(errno, "error creating socket (family %d)", family); |
| 802 | /* Bind the address */ |
| 803 | if(bind(rtpfd, bestifa->ifa_addr, |
| 804 | family == AF_INET |
| 805 | ? sizeof (struct sockaddr_in) : sizeof (struct sockaddr_in6)) < 0) |
| 806 | disorder_fatal(errno, "error binding socket"); |
| 807 | static struct sockaddr_storage bound_address; |
| 808 | addr = (struct sockaddr *)&bound_address; |
| 809 | addr_len = sizeof bound_address; |
| 810 | if(getsockname(rtpfd, addr, &addr_len) < 0) |
| 811 | disorder_fatal(errno, "error getting socket address"); |
| 812 | /* Convert to string */ |
| 813 | char addrname[128], portname[32]; |
| 814 | if(getnameinfo(addr, addr_len, |
| 815 | addrname, sizeof addrname, |
| 816 | portname, sizeof portname, |
| 817 | NI_NUMERICHOST|NI_NUMERICSERV) < 0) |
| 818 | disorder_fatal(errno, "getnameinfo"); |
| 819 | /* Ask for audio data */ |
| 820 | if(disorder_rtp_request(c, addrname, portname)) exit(EXIT_FAILURE); |
| 821 | /* Report what we did */ |
| 822 | disorder_info("listening on %s", format_sockaddr(addr)); |
| 823 | } else { |
| 824 | /* Look up address and port */ |
| 825 | if(!(res = get_address(&sl, &prefs, &sockname))) |
| 826 | exit(1); |
| 827 | addr = res->ai_addr; |
| 828 | addr_len = res->ai_addrlen; |
| 829 | /* Create the socket */ |
| 830 | if((rtpfd = socket(res->ai_family, |
| 831 | res->ai_socktype, |
| 832 | res->ai_protocol)) < 0) |
| 833 | disorder_fatal(errno, "error creating socket"); |
| 834 | /* Allow multiple listeners */ |
| 835 | xsetsockopt(rtpfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one); |
| 836 | is_multicast = multicast(addr); |
| 837 | /* The multicast and unicast/broadcast cases are different enough that they |
| 838 | * are totally split. Trying to find commonality between them causes more |
| 839 | * trouble that it's worth. */ |
| 840 | if(is_multicast) { |
| 841 | /* Stash the multicast group address */ |
| 842 | memcpy(&mgroup, addr, addr_len); |
| 843 | switch(res->ai_addr->sa_family) { |
| 844 | case AF_INET: |
| 845 | mgroup.in.sin_port = 0; |
| 846 | break; |
| 847 | case AF_INET6: |
| 848 | mgroup.in6.sin6_port = 0; |
| 849 | break; |
| 850 | default: |
| 851 | disorder_fatal(0, "unsupported address family %d", |
| 852 | (int)addr->sa_family); |
| 853 | } |
| 854 | /* Bind to to the multicast group address */ |
| 855 | if(bind(rtpfd, addr, addr_len) < 0) |
| 856 | disorder_fatal(errno, "error binding socket to %s", |
| 857 | format_sockaddr(addr)); |
| 858 | /* Add multicast group membership */ |
| 859 | switch(mgroup.sa.sa_family) { |
| 860 | case PF_INET: |
| 861 | mreq.imr_multiaddr = mgroup.in.sin_addr; |
| 862 | mreq.imr_interface.s_addr = 0; /* use primary interface */ |
| 863 | if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP, |
| 864 | &mreq, sizeof mreq) < 0) |
| 865 | disorder_fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP"); |
| 866 | break; |
| 867 | case PF_INET6: |
| 868 | mreq6.ipv6mr_multiaddr = mgroup.in6.sin6_addr; |
| 869 | memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface); |
| 870 | if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP, |
| 871 | &mreq6, sizeof mreq6) < 0) |
| 872 | disorder_fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP"); |
| 873 | break; |
| 874 | default: |
| 875 | disorder_fatal(0, "unsupported address family %d", res->ai_family); |
| 876 | } |
| 877 | /* Report what we did */ |
| 878 | disorder_info("listening on %s multicast group %s", |
| 879 | format_sockaddr(addr), format_sockaddr(&mgroup.sa)); |
| 880 | } else { |
| 881 | /* Bind to 0/port */ |
| 882 | switch(addr->sa_family) { |
| 883 | case AF_INET: { |
| 884 | struct sockaddr_in *in = (struct sockaddr_in *)addr; |
| 885 | |
| 886 | memset(&in->sin_addr, 0, sizeof (struct in_addr)); |
| 887 | break; |
| 888 | } |
| 889 | case AF_INET6: { |
| 890 | struct sockaddr_in6 *in6 = (struct sockaddr_in6 *)addr; |
| 891 | |
| 892 | memset(&in6->sin6_addr, 0, sizeof (struct in6_addr)); |
| 893 | break; |
| 894 | } |
| 895 | default: |
| 896 | disorder_fatal(0, "unsupported family %d", (int)addr->sa_family); |
| 897 | } |
| 898 | if(bind(rtpfd, addr, addr_len) < 0) |
| 899 | disorder_fatal(errno, "error binding socket to %s", |
| 900 | format_sockaddr(addr)); |
| 901 | /* Report what we did */ |
| 902 | disorder_info("listening on %s", format_sockaddr(addr)); |
| 903 | } |
| 904 | } |
| 905 | len = sizeof rcvbuf; |
| 906 | if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0) |
| 907 | disorder_fatal(errno, "error calling getsockopt SO_RCVBUF"); |
| 908 | if(target_rcvbuf > rcvbuf) { |
| 909 | if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, |
| 910 | &target_rcvbuf, sizeof target_rcvbuf) < 0) |
| 911 | disorder_error(errno, "error calling setsockopt SO_RCVBUF %d", |
| 912 | target_rcvbuf); |
| 913 | /* We try to carry on anyway */ |
| 914 | else |
| 915 | disorder_info("changed socket receive buffer from %d to %d", |
| 916 | rcvbuf, target_rcvbuf); |
| 917 | } else |
| 918 | disorder_info("default socket receive buffer %d", rcvbuf); |
| 919 | //info("minbuffer %u maxbuffer %u", minbuffer, maxbuffer); |
| 920 | if(logfp) |
| 921 | disorder_info("WARNING: -L option can impact performance"); |
| 922 | if(control_socket) { |
| 923 | pthread_t tid; |
| 924 | |
| 925 | if((err = pthread_create(&tid, 0, control_thread, 0))) |
| 926 | disorder_fatal(err, "pthread_create control_thread"); |
| 927 | } |
| 928 | if(dumpfile) { |
| 929 | int fd; |
| 930 | unsigned char buffer[65536]; |
| 931 | size_t written; |
| 932 | |
| 933 | if((fd = open(dumpfile, O_RDWR|O_TRUNC|O_CREAT, 0666)) < 0) |
| 934 | disorder_fatal(errno, "opening %s", dumpfile); |
| 935 | /* Fill with 0s to a suitable size */ |
| 936 | memset(buffer, 0, sizeof buffer); |
| 937 | for(written = 0; written < dump_size * sizeof(int16_t); |
| 938 | written += sizeof buffer) { |
| 939 | if(write(fd, buffer, sizeof buffer) < 0) |
| 940 | disorder_fatal(errno, "clearing %s", dumpfile); |
| 941 | } |
| 942 | /* Map the buffer into memory for convenience */ |
| 943 | dump_buffer = mmap(0, dump_size * sizeof(int16_t), PROT_READ|PROT_WRITE, |
| 944 | MAP_SHARED, fd, 0); |
| 945 | if(dump_buffer == (void *)-1) |
| 946 | disorder_fatal(errno, "mapping %s", dumpfile); |
| 947 | disorder_info("dumping to %s", dumpfile); |
| 948 | } |
| 949 | /* Set up output. Currently we only support L16 so there's no harm setting |
| 950 | * the format before we know what it is! */ |
| 951 | uaudio_set_format(44100/*Hz*/, 2/*channels*/, |
| 952 | 16/*bits/channel*/, 1/*signed*/); |
| 953 | uaudio_set("application", "disorder-playrtp"); |
| 954 | backend->start(playrtp_callback, NULL); |
| 955 | /* We receive and convert audio data in a background thread */ |
| 956 | if((err = pthread_create(<id, 0, listen_thread, 0))) |
| 957 | disorder_fatal(err, "pthread_create listen_thread"); |
| 958 | /* We have a second thread to add received packets to the queue */ |
| 959 | if((err = pthread_create(<id, 0, queue_thread, 0))) |
| 960 | disorder_fatal(err, "pthread_create queue_thread"); |
| 961 | pthread_mutex_lock(&lock); |
| 962 | time_t lastlog = 0; |
| 963 | for(;;) { |
| 964 | /* Wait for the buffer to fill up a bit */ |
| 965 | playrtp_fill_buffer(); |
| 966 | /* Start playing now */ |
| 967 | disorder_info("Playing..."); |
| 968 | next_timestamp = pheap_first(&packets)->timestamp; |
| 969 | active = 1; |
| 970 | pthread_mutex_unlock(&lock); |
| 971 | backend->activate(); |
| 972 | pthread_mutex_lock(&lock); |
| 973 | /* Wait until the buffer empties out |
| 974 | * |
| 975 | * If there's a packet that we can play right now then we definitely |
| 976 | * continue. |
| 977 | * |
| 978 | * Also if there's at least minbuffer samples we carry on regardless and |
| 979 | * insert silence. The assumption is there's been a pause but more data |
| 980 | * is now available. |
| 981 | */ |
| 982 | while(nsamples >= minbuffer |
| 983 | || (nsamples > 0 |
| 984 | && contains(pheap_first(&packets), next_timestamp))) { |
| 985 | if(monitor) { |
| 986 | time_t now = xtime(0); |
| 987 | |
| 988 | if(now >= lastlog + 60) { |
| 989 | int offset = nsamples - minbuffer; |
| 990 | double offtime = (double)offset / (uaudio_rate * uaudio_channels); |
| 991 | disorder_info("%+d samples off (%d.%02ds, %d bytes)", |
| 992 | offset, |
| 993 | (int)fabs(offtime) * (offtime < 0 ? -1 : 1), |
| 994 | (int)(fabs(offtime) * 100) % 100, |
| 995 | offset * uaudio_bits / CHAR_BIT); |
| 996 | lastlog = now; |
| 997 | } |
| 998 | } |
| 999 | //fprintf(stderr, "%8u/%u (%u) PLAYING\n", nsamples, maxbuffer, minbuffer); |
| 1000 | pthread_cond_wait(&cond, &lock); |
| 1001 | } |
| 1002 | #if 0 |
| 1003 | if(nsamples) { |
| 1004 | struct packet *p = pheap_first(&packets); |
| 1005 | fprintf(stderr, "nsamples=%u (%u) next_timestamp=%"PRIx32", first packet is [%"PRIx32",%"PRIx32")\n", |
| 1006 | nsamples, minbuffer, next_timestamp,p->timestamp,p->timestamp+p->nsamples); |
| 1007 | } |
| 1008 | #endif |
| 1009 | /* Stop playing for a bit until the buffer re-fills */ |
| 1010 | pthread_mutex_unlock(&lock); |
| 1011 | backend->deactivate(); |
| 1012 | pthread_mutex_lock(&lock); |
| 1013 | active = 0; |
| 1014 | /* Go back round */ |
| 1015 | } |
| 1016 | return 0; |
| 1017 | } |
| 1018 | |
| 1019 | /* |
| 1020 | Local Variables: |
| 1021 | c-basic-offset:2 |
| 1022 | comment-column:40 |
| 1023 | fill-column:79 |
| 1024 | indent-tabs-mode:nil |
| 1025 | End: |
| 1026 | */ |