| 1 | /* |
| 2 | * This file is part of DisOrder |
| 3 | * Copyright (C) 2005, 2006, 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 server/speaker-network.c |
| 21 | * @brief Support for @ref BACKEND_NETWORK */ |
| 22 | |
| 23 | #include <config.h> |
| 24 | #include "types.h" |
| 25 | |
| 26 | #include <unistd.h> |
| 27 | #include <poll.h> |
| 28 | #include <netdb.h> |
| 29 | #include <gcrypt.h> |
| 30 | #include <sys/socket.h> |
| 31 | #include <sys/uio.h> |
| 32 | #include <assert.h> |
| 33 | #include <net/if.h> |
| 34 | #include <ifaddrs.h> |
| 35 | #include <errno.h> |
| 36 | |
| 37 | #include "configuration.h" |
| 38 | #include "syscalls.h" |
| 39 | #include "log.h" |
| 40 | #include "addr.h" |
| 41 | #include "timeval.h" |
| 42 | #include "rtp.h" |
| 43 | #include "ifreq.h" |
| 44 | #include "speaker-protocol.h" |
| 45 | #include "speaker.h" |
| 46 | |
| 47 | /** @brief Network socket |
| 48 | * |
| 49 | * This is the file descriptor to write to for @ref BACKEND_NETWORK. |
| 50 | */ |
| 51 | static int bfd = -1; |
| 52 | |
| 53 | /** @brief RTP timestamp |
| 54 | * |
| 55 | * This counts the number of samples played (NB not the number of frames |
| 56 | * played). |
| 57 | * |
| 58 | * The timestamp in the packet header is only 32 bits wide. With 44100Hz |
| 59 | * stereo, that only gives about half a day before wrapping, which is not |
| 60 | * particularly convenient for certain debugging purposes. Therefore the |
| 61 | * timestamp is maintained as a 64-bit integer, giving around six million years |
| 62 | * before wrapping, and truncated to 32 bits when transmitting. |
| 63 | */ |
| 64 | static uint64_t rtp_time; |
| 65 | |
| 66 | /** @brief RTP base timestamp |
| 67 | * |
| 68 | * This is the real time correspoding to an @ref rtp_time of 0. It is used |
| 69 | * to recalculate the timestamp after idle periods. |
| 70 | */ |
| 71 | static struct timeval rtp_time_0; |
| 72 | |
| 73 | /** @brief RTP packet sequence number */ |
| 74 | static uint16_t rtp_seq; |
| 75 | |
| 76 | /** @brief RTP SSRC */ |
| 77 | static uint32_t rtp_id; |
| 78 | |
| 79 | /** @brief Error counter */ |
| 80 | static int audio_errors; |
| 81 | |
| 82 | /** @brief Network backend initialization */ |
| 83 | static void network_init(void) { |
| 84 | struct addrinfo *res, *sres; |
| 85 | static const struct addrinfo pref = { |
| 86 | 0, |
| 87 | PF_INET, |
| 88 | SOCK_DGRAM, |
| 89 | IPPROTO_UDP, |
| 90 | 0, |
| 91 | 0, |
| 92 | 0, |
| 93 | 0 |
| 94 | }; |
| 95 | static const struct addrinfo prefbind = { |
| 96 | AI_PASSIVE, |
| 97 | PF_INET, |
| 98 | SOCK_DGRAM, |
| 99 | IPPROTO_UDP, |
| 100 | 0, |
| 101 | 0, |
| 102 | 0, |
| 103 | 0 |
| 104 | }; |
| 105 | static const int one = 1; |
| 106 | int sndbuf, target_sndbuf = 131072; |
| 107 | socklen_t len; |
| 108 | char *sockname, *ssockname; |
| 109 | |
| 110 | res = get_address(&config->broadcast, &pref, &sockname); |
| 111 | if(!res) exit(-1); |
| 112 | if(config->broadcast_from.n) { |
| 113 | sres = get_address(&config->broadcast_from, &prefbind, &ssockname); |
| 114 | if(!sres) exit(-1); |
| 115 | } else |
| 116 | sres = 0; |
| 117 | if((bfd = socket(res->ai_family, |
| 118 | res->ai_socktype, |
| 119 | res->ai_protocol)) < 0) |
| 120 | fatal(errno, "error creating broadcast socket"); |
| 121 | if((res->ai_family == PF_INET |
| 122 | && IN_MULTICAST( |
| 123 | ntohl(((struct sockaddr_in *)res->ai_addr)->sin_addr.s_addr) |
| 124 | )) |
| 125 | || (res->ai_family == PF_INET6 |
| 126 | && IN6_IS_ADDR_MULTICAST( |
| 127 | &((struct sockaddr_in6 *)res->ai_addr)->sin6_addr |
| 128 | ))) { |
| 129 | /* Multicasting */ |
| 130 | switch(res->ai_family) { |
| 131 | case PF_INET: { |
| 132 | const int mttl = config->multicast_ttl; |
| 133 | if(setsockopt(bfd, IPPROTO_IP, IP_MULTICAST_TTL, &mttl, sizeof mttl) < 0) |
| 134 | fatal(errno, "error setting IP_MULTICAST_TTL on multicast socket"); |
| 135 | break; |
| 136 | } |
| 137 | case PF_INET6: { |
| 138 | const int mttl = config->multicast_ttl; |
| 139 | if(setsockopt(bfd, IPPROTO_IPV6, IPV6_MULTICAST_HOPS, |
| 140 | &mttl, sizeof mttl) < 0) |
| 141 | fatal(errno, "error setting IPV6_MULTICAST_HOPS on multicast socket"); |
| 142 | break; |
| 143 | } |
| 144 | default: |
| 145 | fatal(0, "unsupported address family %d", res->ai_family); |
| 146 | } |
| 147 | info("multicasting on %s", sockname); |
| 148 | } else { |
| 149 | struct ifaddrs *ifs; |
| 150 | |
| 151 | if(getifaddrs(&ifs) < 0) |
| 152 | fatal(errno, "error calling getifaddrs"); |
| 153 | while(ifs) { |
| 154 | /* (At least on Darwin) IFF_BROADCAST might be set but ifa_broadaddr |
| 155 | * still a null pointer. It turns out that there's a subsequent entry |
| 156 | * for he same interface which _does_ have ifa_broadaddr though... */ |
| 157 | if((ifs->ifa_flags & IFF_BROADCAST) |
| 158 | && ifs->ifa_broadaddr |
| 159 | && sockaddr_equal(ifs->ifa_broadaddr, res->ai_addr)) |
| 160 | break; |
| 161 | ifs = ifs->ifa_next; |
| 162 | } |
| 163 | if(ifs) { |
| 164 | if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0) |
| 165 | fatal(errno, "error setting SO_BROADCAST on broadcast socket"); |
| 166 | info("broadcasting on %s (%s)", sockname, ifs->ifa_name); |
| 167 | } else |
| 168 | info("unicasting on %s", sockname); |
| 169 | } |
| 170 | len = sizeof sndbuf; |
| 171 | if(getsockopt(bfd, SOL_SOCKET, SO_SNDBUF, |
| 172 | &sndbuf, &len) < 0) |
| 173 | fatal(errno, "error getting SO_SNDBUF"); |
| 174 | if(target_sndbuf > sndbuf) { |
| 175 | if(setsockopt(bfd, SOL_SOCKET, SO_SNDBUF, |
| 176 | &target_sndbuf, sizeof target_sndbuf) < 0) |
| 177 | error(errno, "error setting SO_SNDBUF to %d", target_sndbuf); |
| 178 | else |
| 179 | info("changed socket send buffer size from %d to %d", |
| 180 | sndbuf, target_sndbuf); |
| 181 | } else |
| 182 | info("default socket send buffer is %d", |
| 183 | sndbuf); |
| 184 | /* We might well want to set additional broadcast- or multicast-related |
| 185 | * options here */ |
| 186 | if(sres && bind(bfd, sres->ai_addr, sres->ai_addrlen) < 0) |
| 187 | fatal(errno, "error binding broadcast socket to %s", ssockname); |
| 188 | if(connect(bfd, res->ai_addr, res->ai_addrlen) < 0) |
| 189 | fatal(errno, "error connecting broadcast socket to %s", sockname); |
| 190 | /* Select an SSRC */ |
| 191 | gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM); |
| 192 | info("selected network backend, sending to %s", sockname); |
| 193 | } |
| 194 | |
| 195 | /** @brief Play over the network */ |
| 196 | static size_t network_play(size_t frames) { |
| 197 | struct rtp_header header; |
| 198 | struct iovec vec[2]; |
| 199 | size_t bytes = frames * bpf, written_frames; |
| 200 | int written_bytes; |
| 201 | /* We transmit using RTP (RFC3550) and attempt to conform to the internet |
| 202 | * AVT profile (RFC3551). */ |
| 203 | |
| 204 | if(idled) { |
| 205 | /* There may have been a gap. Fix up the RTP time accordingly. */ |
| 206 | struct timeval now; |
| 207 | uint64_t delta; |
| 208 | uint64_t target_rtp_time; |
| 209 | |
| 210 | /* Find the current time */ |
| 211 | xgettimeofday(&now, 0); |
| 212 | /* Find the number of microseconds elapsed since rtp_time=0 */ |
| 213 | delta = tvsub_us(now, rtp_time_0); |
| 214 | assert(delta <= UINT64_MAX / 88200); |
| 215 | target_rtp_time = (delta * config->sample_format.rate |
| 216 | * config->sample_format.channels) / 1000000; |
| 217 | /* Overflows at ~6 years uptime with 44100Hz stereo */ |
| 218 | |
| 219 | /* rtp_time is the number of samples we've played. NB that we play |
| 220 | * RTP_AHEAD_MS ahead of ourselves, so it may legitimately be ahead of |
| 221 | * the value we deduce from time comparison. |
| 222 | * |
| 223 | * Suppose we have 1s track started at t=0, and another track begins to |
| 224 | * play at t=2s. Suppose RTP_AHEAD_MS=1000 and 44100Hz stereo. In that |
| 225 | * case we'll send 1s of audio as fast as we can, giving rtp_time=88200. |
| 226 | * rtp_time stops at this point. |
| 227 | * |
| 228 | * At t=2s we'll have calculated target_rtp_time=176400. In this case we |
| 229 | * set rtp_time=176400 and the player can correctly conclude that it |
| 230 | * should leave 1s between the tracks. |
| 231 | * |
| 232 | * Suppose instead that the second track arrives at t=0.5s, and that |
| 233 | * we've managed to transmit the whole of the first track already. We'll |
| 234 | * have target_rtp_time=44100. |
| 235 | * |
| 236 | * The desired behaviour is to play the second track back to back with |
| 237 | * first. In this case therefore we do not modify rtp_time. |
| 238 | * |
| 239 | * Is it ever right to reduce rtp_time? No; for that would imply |
| 240 | * transmitting packets with overlapping timestamp ranges, which does not |
| 241 | * make sense. |
| 242 | */ |
| 243 | target_rtp_time &= ~(uint64_t)1; /* stereo! */ |
| 244 | if(target_rtp_time > rtp_time) { |
| 245 | /* More time has elapsed than we've transmitted samples. That implies |
| 246 | * we've been 'sending' silence. */ |
| 247 | info("advancing rtp_time by %"PRIu64" samples", |
| 248 | target_rtp_time - rtp_time); |
| 249 | rtp_time = target_rtp_time; |
| 250 | } else if(target_rtp_time < rtp_time) { |
| 251 | const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS |
| 252 | * config->sample_format.rate |
| 253 | * config->sample_format.channels |
| 254 | / 1000); |
| 255 | |
| 256 | if(target_rtp_time + samples_ahead < rtp_time) { |
| 257 | info("reversing rtp_time by %"PRIu64" samples", |
| 258 | rtp_time - target_rtp_time); |
| 259 | } |
| 260 | } |
| 261 | } |
| 262 | header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */ |
| 263 | header.seq = htons(rtp_seq++); |
| 264 | header.timestamp = htonl((uint32_t)rtp_time); |
| 265 | header.ssrc = rtp_id; |
| 266 | header.mpt = (idled ? 0x80 : 0x00) | 10; |
| 267 | /* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from |
| 268 | * the sample rate (in a library somewhere so that configuration.c can rule |
| 269 | * out invalid rates). |
| 270 | */ |
| 271 | idled = 0; |
| 272 | if(bytes > NETWORK_BYTES - sizeof header) { |
| 273 | bytes = NETWORK_BYTES - sizeof header; |
| 274 | /* Always send a whole number of frames */ |
| 275 | bytes -= bytes % bpf; |
| 276 | } |
| 277 | /* "The RTP clock rate used for generating the RTP timestamp is independent |
| 278 | * of the number of channels and the encoding; it equals the number of |
| 279 | * sampling periods per second. For N-channel encodings, each sampling |
| 280 | * period (say, 1/8000 of a second) generates N samples. (This terminology |
| 281 | * is standard, but somewhat confusing, as the total number of samples |
| 282 | * generated per second is then the sampling rate times the channel |
| 283 | * count.)" |
| 284 | */ |
| 285 | vec[0].iov_base = (void *)&header; |
| 286 | vec[0].iov_len = sizeof header; |
| 287 | vec[1].iov_base = playing->buffer + playing->start; |
| 288 | vec[1].iov_len = bytes; |
| 289 | do { |
| 290 | written_bytes = writev(bfd, vec, 2); |
| 291 | } while(written_bytes < 0 && errno == EINTR); |
| 292 | if(written_bytes < 0) { |
| 293 | error(errno, "error transmitting audio data"); |
| 294 | ++audio_errors; |
| 295 | if(audio_errors == 10) |
| 296 | fatal(0, "too many audio errors"); |
| 297 | return 0; |
| 298 | } else |
| 299 | audio_errors /= 2; |
| 300 | written_bytes -= sizeof (struct rtp_header); |
| 301 | written_frames = written_bytes / bpf; |
| 302 | /* Advance RTP's notion of the time */ |
| 303 | rtp_time += written_frames * config->sample_format.channels; |
| 304 | return written_frames; |
| 305 | } |
| 306 | |
| 307 | static int bfd_slot; |
| 308 | |
| 309 | /** @brief Set up poll array for network play */ |
| 310 | static void network_beforepoll(void) { |
| 311 | struct timeval now; |
| 312 | uint64_t target_us; |
| 313 | uint64_t target_rtp_time; |
| 314 | const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS |
| 315 | * config->sample_format.rate |
| 316 | * config->sample_format.channels |
| 317 | / 1000); |
| 318 | |
| 319 | /* If we're starting then initialize the base time */ |
| 320 | if(!rtp_time) |
| 321 | xgettimeofday(&rtp_time_0, 0); |
| 322 | /* We send audio data whenever we get RTP_AHEAD seconds or more |
| 323 | * behind */ |
| 324 | xgettimeofday(&now, 0); |
| 325 | target_us = tvsub_us(now, rtp_time_0); |
| 326 | assert(target_us <= UINT64_MAX / 88200); |
| 327 | target_rtp_time = (target_us * config->sample_format.rate |
| 328 | * config->sample_format.channels) |
| 329 | / 1000000; |
| 330 | if((int64_t)(rtp_time - target_rtp_time) < samples_ahead) |
| 331 | bfd_slot = addfd(bfd, POLLOUT); |
| 332 | } |
| 333 | |
| 334 | /** @brief Process poll() results for network play */ |
| 335 | static int network_ready(void) { |
| 336 | if(fds[bfd_slot].revents & (POLLOUT | POLLERR)) |
| 337 | return 1; |
| 338 | else |
| 339 | return 0; |
| 340 | } |
| 341 | |
| 342 | const struct speaker_backend network_backend = { |
| 343 | BACKEND_NETWORK, |
| 344 | 0, |
| 345 | network_init, |
| 346 | 0, /* activate */ |
| 347 | network_play, |
| 348 | 0, /* deactivate */ |
| 349 | network_beforepoll, |
| 350 | network_ready |
| 351 | }; |
| 352 | |
| 353 | /* |
| 354 | Local Variables: |
| 355 | c-basic-offset:2 |
| 356 | comment-column:40 |
| 357 | fill-column:79 |
| 358 | indent-tabs-mode:nil |
| 359 | End: |
| 360 | */ |