| 1 | /* |
| 2 | * This file is part of DisOrder. |
| 3 | * Copyright (C) 2009 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 lib/uaudio-rtp.c |
| 19 | * @brief Support for RTP network play backend */ |
| 20 | #include "common.h" |
| 21 | |
| 22 | #include <errno.h> |
| 23 | #include <sys/socket.h> |
| 24 | #include <ifaddrs.h> |
| 25 | #include <net/if.h> |
| 26 | #include <gcrypt.h> |
| 27 | #include <unistd.h> |
| 28 | #include <time.h> |
| 29 | #include <sys/uio.h> |
| 30 | |
| 31 | #include "uaudio.h" |
| 32 | #include "mem.h" |
| 33 | #include "log.h" |
| 34 | #include "syscalls.h" |
| 35 | #include "rtp.h" |
| 36 | #include "addr.h" |
| 37 | #include "ifreq.h" |
| 38 | #include "timeval.h" |
| 39 | |
| 40 | /** @brief Bytes to send per network packet |
| 41 | * |
| 42 | * This is the maximum number of bytes we pass to write(2); to determine actual |
| 43 | * packet sizes, add a UDP header and an IP header (and a link layer header if |
| 44 | * it's the link layer size you care about). |
| 45 | * |
| 46 | * Don't make this too big or arithmetic will start to overflow. |
| 47 | */ |
| 48 | #define NETWORK_BYTES (1500-8/*UDP*/-40/*IP*/-8/*conservatism*/) |
| 49 | |
| 50 | /** @brief RTP payload type */ |
| 51 | static int rtp_payload; |
| 52 | |
| 53 | /** @brief RTP output socket */ |
| 54 | static int rtp_fd; |
| 55 | |
| 56 | /** @brief RTP SSRC */ |
| 57 | static uint32_t rtp_id; |
| 58 | |
| 59 | /** @brief RTP sequence number */ |
| 60 | static uint16_t rtp_sequence; |
| 61 | |
| 62 | /** @brief RTP timestamp |
| 63 | * |
| 64 | * This is the timestamp that will be used on the next outbound packet. |
| 65 | * |
| 66 | * The timestamp in the packet header is only 32 bits wide. With 44100Hz |
| 67 | * stereo, that only gives about half a day before wrapping, which is not |
| 68 | * particularly convenient for certain debugging purposes. Therefore the |
| 69 | * timestamp is maintained as a 64-bit integer, giving around six million years |
| 70 | * before wrapping, and truncated to 32 bits when transmitting. |
| 71 | */ |
| 72 | static uint64_t rtp_timestamp; |
| 73 | |
| 74 | /** @brief Actual time corresponding to @ref rtp_timestamp |
| 75 | * |
| 76 | * This is the time, on this machine, at which the sample at @ref rtp_timestamp |
| 77 | * ought to be sent, interpreted as the time the last packet was sent plus the |
| 78 | * time length of the packet. */ |
| 79 | static struct timeval rtp_timeval; |
| 80 | |
| 81 | /** @brief Set when we (re-)activate, to provoke timestamp resync */ |
| 82 | static int rtp_reactivated; |
| 83 | |
| 84 | /** @brief Network error count |
| 85 | * |
| 86 | * If too many errors occur in too short a time, we give up. |
| 87 | */ |
| 88 | static int rtp_errors; |
| 89 | |
| 90 | /** @brief Delay threshold in microseconds |
| 91 | * |
| 92 | * rtp_play() never attempts to introduce a delay shorter than this. |
| 93 | */ |
| 94 | static int64_t rtp_delay_threshold; |
| 95 | |
| 96 | static const char *const rtp_options[] = { |
| 97 | "rtp-destination", |
| 98 | "rtp-destination-port", |
| 99 | "rtp-source", |
| 100 | "rtp-source-port", |
| 101 | "multicast-ttl", |
| 102 | "multicast-loop", |
| 103 | "rtp-delay-threshold", |
| 104 | NULL |
| 105 | }; |
| 106 | |
| 107 | static size_t rtp_play(void *buffer, size_t nsamples) { |
| 108 | struct rtp_header header; |
| 109 | struct iovec vec[2]; |
| 110 | struct timeval now; |
| 111 | |
| 112 | /* We do as much work as possible before checking what time it is */ |
| 113 | /* Fill out header */ |
| 114 | header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */ |
| 115 | header.seq = htons(rtp_sequence++); |
| 116 | header.ssrc = rtp_id; |
| 117 | header.mpt = (rtp_reactivated ? 0x80 : 0x00) | rtp_payload; |
| 118 | #if !WORDS_BIGENDIAN |
| 119 | /* Convert samples to network byte order */ |
| 120 | uint16_t *u = buffer, *const limit = u + nsamples; |
| 121 | while(u < limit) { |
| 122 | *u = htons(*u); |
| 123 | ++u; |
| 124 | } |
| 125 | #endif |
| 126 | vec[0].iov_base = (void *)&header; |
| 127 | vec[0].iov_len = sizeof header; |
| 128 | vec[1].iov_base = buffer; |
| 129 | vec[1].iov_len = nsamples * uaudio_sample_size; |
| 130 | retry: |
| 131 | xgettimeofday(&now, NULL); |
| 132 | if(rtp_reactivated) { |
| 133 | /* We've been deactivated for some unknown interval. We need to advance |
| 134 | * rtp_timestamp to account for the dead air. */ |
| 135 | /* On the first run through we'll set the start time. */ |
| 136 | if(!rtp_timeval.tv_sec) |
| 137 | rtp_timeval = now; |
| 138 | /* See how much time we missed. |
| 139 | * |
| 140 | * This will be 0 on the first run through, in which case we'll not modify |
| 141 | * anything. |
| 142 | * |
| 143 | * It'll be negative in the (rare) situation where the deactivation |
| 144 | * interval is shorter than the last packet we sent. In this case we wait |
| 145 | * for that much time and then return having sent no samples, which will |
| 146 | * cause uaudio_play_thread_fn() to retry. |
| 147 | * |
| 148 | * In the normal case it will be positive. |
| 149 | */ |
| 150 | const int64_t delay = tvsub_us(now, rtp_timeval); /* microseconds */ |
| 151 | if(delay < 0) { |
| 152 | usleep(-delay); |
| 153 | goto retry; |
| 154 | } |
| 155 | /* Advance the RTP timestamp to the present. With 44.1KHz stereo this will |
| 156 | * overflow the intermediate value with a delay of a bit over 6 years. |
| 157 | * This seems acceptable. */ |
| 158 | uint64_t update = (delay * uaudio_rate * uaudio_channels) / 1000000; |
| 159 | /* Don't throw off channel synchronization */ |
| 160 | update -= update % uaudio_channels; |
| 161 | /* We log nontrivial changes */ |
| 162 | if(update) |
| 163 | info("advancing rtp_time by %"PRIu64" samples", update); |
| 164 | rtp_timestamp += update; |
| 165 | rtp_timeval = now; |
| 166 | rtp_reactivated = 0; |
| 167 | } else { |
| 168 | /* Chances are we've been called right on the heels of the previous packet. |
| 169 | * If we just sent packets as fast as we got audio data we'd get way ahead |
| 170 | * of the player and some buffer somewhere would fill (or at least become |
| 171 | * unreasonably large). |
| 172 | * |
| 173 | * First find out how far ahead of the target time we are. |
| 174 | */ |
| 175 | const int64_t ahead = tvsub_us(now, rtp_timeval); /* microseconds */ |
| 176 | /* Only delay at all if we are nontrivially ahead. */ |
| 177 | if(ahead > rtp_delay_threshold) { |
| 178 | /* Don't delay by the full amount */ |
| 179 | usleep(ahead - rtp_delay_threshold / 2); |
| 180 | /* Refetch time (so we don't get out of step with reality) */ |
| 181 | xgettimeofday(&now, NULL); |
| 182 | } |
| 183 | } |
| 184 | header.timestamp = htonl((uint32_t)rtp_timestamp); |
| 185 | int written_bytes; |
| 186 | do { |
| 187 | written_bytes = writev(rtp_fd, vec, 2); |
| 188 | } while(written_bytes < 0 && errno == EINTR); |
| 189 | if(written_bytes < 0) { |
| 190 | error(errno, "error transmitting audio data"); |
| 191 | ++rtp_errors; |
| 192 | if(rtp_errors == 10) |
| 193 | fatal(0, "too many audio tranmission errors"); |
| 194 | return 0; |
| 195 | } else |
| 196 | rtp_errors /= 2; /* gradual decay */ |
| 197 | written_bytes -= sizeof (struct rtp_header); |
| 198 | size_t written_samples = written_bytes / uaudio_sample_size; |
| 199 | /* rtp_timestamp and rtp_timestamp are supposed to refer to the first sample |
| 200 | * of the next packet */ |
| 201 | rtp_timestamp += written_samples; |
| 202 | const unsigned usec = (rtp_timeval.tv_usec |
| 203 | + 1000000 * written_samples / (uaudio_rate |
| 204 | * uaudio_channels)); |
| 205 | /* ...will only overflow 32 bits if one packet is more than about half an |
| 206 | * hour long, which is not plausible. */ |
| 207 | rtp_timeval.tv_sec += usec / 1000000; |
| 208 | rtp_timeval.tv_usec = usec % 1000000; |
| 209 | return written_samples; |
| 210 | } |
| 211 | |
| 212 | static void rtp_open(void) { |
| 213 | struct addrinfo *res, *sres; |
| 214 | static const struct addrinfo pref = { |
| 215 | .ai_flags = 0, |
| 216 | .ai_family = PF_INET, |
| 217 | .ai_socktype = SOCK_DGRAM, |
| 218 | .ai_protocol = IPPROTO_UDP, |
| 219 | }; |
| 220 | static const struct addrinfo prefbind = { |
| 221 | .ai_flags = AI_PASSIVE, |
| 222 | .ai_family = PF_INET, |
| 223 | .ai_socktype = SOCK_DGRAM, |
| 224 | .ai_protocol = IPPROTO_UDP, |
| 225 | }; |
| 226 | static const int one = 1; |
| 227 | int sndbuf, target_sndbuf = 131072; |
| 228 | socklen_t len; |
| 229 | char *sockname, *ssockname; |
| 230 | struct stringlist dst, src; |
| 231 | const char *delay; |
| 232 | |
| 233 | /* Get configuration */ |
| 234 | dst.n = 2; |
| 235 | dst.s = xcalloc(2, sizeof *dst.s); |
| 236 | dst.s[0] = uaudio_get("rtp-destination"); |
| 237 | dst.s[1] = uaudio_get("rtp-destination-port"); |
| 238 | src.n = 2; |
| 239 | src.s = xcalloc(2, sizeof *dst.s); |
| 240 | src.s[0] = uaudio_get("rtp-source"); |
| 241 | src.s[1] = uaudio_get("rtp-source-port"); |
| 242 | if(!dst.s[0]) |
| 243 | fatal(0, "'rtp-destination' not set"); |
| 244 | if(!dst.s[1]) |
| 245 | fatal(0, "'rtp-destination-port' not set"); |
| 246 | if(src.s[0]) { |
| 247 | if(!src.s[1]) |
| 248 | fatal(0, "'rtp-source-port' not set"); |
| 249 | src.n = 2; |
| 250 | } else |
| 251 | src.n = 0; |
| 252 | if((delay = uaudio_get("rtp-delay-threshold"))) |
| 253 | rtp_delay_threshold = atoi(delay); |
| 254 | else |
| 255 | rtp_delay_threshold = 1000; /* microseconds */ |
| 256 | |
| 257 | /* Resolve addresses */ |
| 258 | res = get_address(&dst, &pref, &sockname); |
| 259 | if(!res) exit(-1); |
| 260 | if(src.n) { |
| 261 | sres = get_address(&src, &prefbind, &ssockname); |
| 262 | if(!sres) exit(-1); |
| 263 | } else |
| 264 | sres = 0; |
| 265 | /* Create the socket */ |
| 266 | if((rtp_fd = socket(res->ai_family, |
| 267 | res->ai_socktype, |
| 268 | res->ai_protocol)) < 0) |
| 269 | fatal(errno, "error creating broadcast socket"); |
| 270 | if(multicast(res->ai_addr)) { |
| 271 | /* Enable multicast options */ |
| 272 | const char *ttls = uaudio_get("multicast-ttl"); |
| 273 | const int ttl = ttls ? atoi(ttls) : 1; |
| 274 | const char *loops = uaudio_get("multicast-loop"); |
| 275 | const int loop = loops ? !strcmp(loops, "yes") : 1; |
| 276 | switch(res->ai_family) { |
| 277 | case PF_INET: { |
| 278 | if(setsockopt(rtp_fd, IPPROTO_IP, IP_MULTICAST_TTL, |
| 279 | &ttl, sizeof ttl) < 0) |
| 280 | fatal(errno, "error setting IP_MULTICAST_TTL on multicast socket"); |
| 281 | if(setsockopt(rtp_fd, IPPROTO_IP, IP_MULTICAST_LOOP, |
| 282 | &loop, sizeof loop) < 0) |
| 283 | fatal(errno, "error setting IP_MULTICAST_LOOP on multicast socket"); |
| 284 | break; |
| 285 | } |
| 286 | case PF_INET6: { |
| 287 | if(setsockopt(rtp_fd, IPPROTO_IPV6, IPV6_MULTICAST_HOPS, |
| 288 | &ttl, sizeof ttl) < 0) |
| 289 | fatal(errno, "error setting IPV6_MULTICAST_HOPS on multicast socket"); |
| 290 | if(setsockopt(rtp_fd, IPPROTO_IP, IPV6_MULTICAST_LOOP, |
| 291 | &loop, sizeof loop) < 0) |
| 292 | fatal(errno, "error setting IPV6_MULTICAST_LOOP on multicast socket"); |
| 293 | break; |
| 294 | } |
| 295 | default: |
| 296 | fatal(0, "unsupported address family %d", res->ai_family); |
| 297 | } |
| 298 | info("multicasting on %s TTL=%d loop=%s", |
| 299 | sockname, ttl, loop ? "yes" : "no"); |
| 300 | } else { |
| 301 | struct ifaddrs *ifs; |
| 302 | |
| 303 | if(getifaddrs(&ifs) < 0) |
| 304 | fatal(errno, "error calling getifaddrs"); |
| 305 | while(ifs) { |
| 306 | /* (At least on Darwin) IFF_BROADCAST might be set but ifa_broadaddr |
| 307 | * still a null pointer. It turns out that there's a subsequent entry |
| 308 | * for he same interface which _does_ have ifa_broadaddr though... */ |
| 309 | if((ifs->ifa_flags & IFF_BROADCAST) |
| 310 | && ifs->ifa_broadaddr |
| 311 | && sockaddr_equal(ifs->ifa_broadaddr, res->ai_addr)) |
| 312 | break; |
| 313 | ifs = ifs->ifa_next; |
| 314 | } |
| 315 | if(ifs) { |
| 316 | if(setsockopt(rtp_fd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0) |
| 317 | fatal(errno, "error setting SO_BROADCAST on broadcast socket"); |
| 318 | info("broadcasting on %s (%s)", sockname, ifs->ifa_name); |
| 319 | } else |
| 320 | info("unicasting on %s", sockname); |
| 321 | } |
| 322 | /* Enlarge the socket buffer */ |
| 323 | len = sizeof sndbuf; |
| 324 | if(getsockopt(rtp_fd, SOL_SOCKET, SO_SNDBUF, |
| 325 | &sndbuf, &len) < 0) |
| 326 | fatal(errno, "error getting SO_SNDBUF"); |
| 327 | if(target_sndbuf > sndbuf) { |
| 328 | if(setsockopt(rtp_fd, SOL_SOCKET, SO_SNDBUF, |
| 329 | &target_sndbuf, sizeof target_sndbuf) < 0) |
| 330 | error(errno, "error setting SO_SNDBUF to %d", target_sndbuf); |
| 331 | else |
| 332 | info("changed socket send buffer size from %d to %d", |
| 333 | sndbuf, target_sndbuf); |
| 334 | } else |
| 335 | info("default socket send buffer is %d", |
| 336 | sndbuf); |
| 337 | /* We might well want to set additional broadcast- or multicast-related |
| 338 | * options here */ |
| 339 | if(sres && bind(rtp_fd, sres->ai_addr, sres->ai_addrlen) < 0) |
| 340 | fatal(errno, "error binding broadcast socket to %s", ssockname); |
| 341 | if(connect(rtp_fd, res->ai_addr, res->ai_addrlen) < 0) |
| 342 | fatal(errno, "error connecting broadcast socket to %s", sockname); |
| 343 | /* Various fields are required to have random initial values by RFC3550. The |
| 344 | * packet contents are highly public so there's no point asking for very |
| 345 | * strong randomness. */ |
| 346 | gcry_create_nonce(&rtp_id, sizeof rtp_id); |
| 347 | gcry_create_nonce(&rtp_sequence, sizeof rtp_sequence); |
| 348 | gcry_create_nonce(&rtp_timestamp, sizeof rtp_timestamp); |
| 349 | /* rtp_play() will spot this and choose an initial value */ |
| 350 | rtp_timeval.tv_sec = 0; |
| 351 | } |
| 352 | |
| 353 | static void rtp_start(uaudio_callback *callback, |
| 354 | void *userdata) { |
| 355 | /* We only support L16 (but we do stereo and mono and will convert sign) */ |
| 356 | if(uaudio_channels == 2 |
| 357 | && uaudio_bits == 16 |
| 358 | && uaudio_rate == 44100) |
| 359 | rtp_payload = 10; |
| 360 | else if(uaudio_channels == 1 |
| 361 | && uaudio_bits == 16 |
| 362 | && uaudio_rate == 44100) |
| 363 | rtp_payload = 11; |
| 364 | else |
| 365 | fatal(0, "asked for %d/%d/%d 16/44100/1 and 16/44100/2", |
| 366 | uaudio_bits, uaudio_rate, uaudio_channels); |
| 367 | rtp_open(); |
| 368 | uaudio_thread_start(callback, |
| 369 | userdata, |
| 370 | rtp_play, |
| 371 | 256 / uaudio_sample_size, |
| 372 | (NETWORK_BYTES - sizeof(struct rtp_header)) |
| 373 | / uaudio_sample_size); |
| 374 | } |
| 375 | |
| 376 | static void rtp_stop(void) { |
| 377 | uaudio_thread_stop(); |
| 378 | close(rtp_fd); |
| 379 | rtp_fd = -1; |
| 380 | } |
| 381 | |
| 382 | static void rtp_activate(void) { |
| 383 | rtp_reactivated = 1; |
| 384 | uaudio_thread_activate(); |
| 385 | } |
| 386 | |
| 387 | static void rtp_deactivate(void) { |
| 388 | uaudio_thread_deactivate(); |
| 389 | } |
| 390 | |
| 391 | const struct uaudio uaudio_rtp = { |
| 392 | .name = "rtp", |
| 393 | .options = rtp_options, |
| 394 | .start = rtp_start, |
| 395 | .stop = rtp_stop, |
| 396 | .activate = rtp_activate, |
| 397 | .deactivate = rtp_deactivate |
| 398 | }; |
| 399 | |
| 400 | /* |
| 401 | Local Variables: |
| 402 | c-basic-offset:2 |
| 403 | comment-column:40 |
| 404 | fill-column:79 |
| 405 | indent-tabs-mode:nil |
| 406 | End: |
| 407 | */ |