mdw@git.distorted.org.uk Git - disorder/blob - lib/uaudio-rtp.c

   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 <arpa/inet.h>
  27 #include <netinet/in.h>
  28 #include <gcrypt.h>
  29 #include <unistd.h>
  30 #include <time.h>
  31 #include <sys/uio.h>
  32
  33 #include "uaudio.h"
  34 #include "mem.h"
  35 #include "log.h"
  36 #include "syscalls.h"
  37 #include "rtp.h"
  38 #include "addr.h"
  39 #include "ifreq.h"
  40 #include "timeval.h"
  41
  42 /** @brief Bytes to send per network packet
  43  *
  44  * This is the maximum number of bytes we pass to write(2); to determine actual
  45  * packet sizes, add a UDP header and an IP header (and a link layer header if
  46  * it's the link layer size you care about).
  47  *
  48  * Don't make this too big or arithmetic will start to overflow.
  49  */
  50 #define NETWORK_BYTES (1500-8/*UDP*/-40/*IP*/-8/*conservatism*/)
  51
  52 /** @brief RTP payload type */
  53 static int rtp_payload;
  54
  55 /** @brief RTP output socket */
  56 static int rtp_fd;
  57
  58 /** @brief RTP SSRC */
  59 static uint32_t rtp_id;
  60
  61 /** @brief RTP sequence number */
  62 static uint16_t rtp_sequence;
  63
  64 /** @brief Network error count
  65  *
  66  * If too many errors occur in too short a time, we give up.
  67  */
  68 static int rtp_errors;
  69
  70 /** @brief Delay threshold in microseconds
  71  *
  72  * rtp_play() never attempts to introduce a delay shorter than this.
  73  */
  74 static int64_t rtp_delay_threshold;
  75
  76 static const char *const rtp_options[] = {
  77   "rtp-destination",
  78   "rtp-destination-port",
  79   "rtp-source",
  80   "rtp-source-port",
  81   "multicast-ttl",
  82   "multicast-loop",
  83   "delay-threshold",
  84   NULL
  85 };
  86
  87 static size_t rtp_play(void *buffer, size_t nsamples) {
  88   struct rtp_header header;
  89   struct iovec vec[2];
  90
  91   /* We do as much work as possible before checking what time it is */
  92   /* Fill out header */
  93   header.vpxcc = 2 << 6;              /* V=2, P=0, X=0, CC=0 */
  94   header.seq = htons(rtp_sequence++);
  95   header.ssrc = rtp_id;
  96   header.mpt = (uaudio_schedule_reactivated ? 0x80 : 0x00) | rtp_payload;
  97 #if !WORDS_BIGENDIAN
  98   /* Convert samples to network byte order */
  99   uint16_t *u = buffer, *const limit = u + nsamples;
 100   while(u < limit) {
 101     *u = htons(*u);
 102     ++u;
 103   }
 104 #endif
 105   vec[0].iov_base = (void *)&header;
 106   vec[0].iov_len = sizeof header;
 107   vec[1].iov_base = buffer;
 108   vec[1].iov_len = nsamples * uaudio_sample_size;
 109   uaudio_schedule_synchronize();
 110   header.timestamp = htonl((uint32_t)uaudio_schedule_timestamp);
 111   int written_bytes;
 112   do {
 113     written_bytes = writev(rtp_fd, vec, 2);
 114   } while(written_bytes < 0 && errno == EINTR);
 115   if(written_bytes < 0) {
 116     error(errno, "error transmitting audio data");
 117     ++rtp_errors;
 118     if(rtp_errors == 10)
 119       fatal(0, "too many audio tranmission errors");
 120     return 0;
 121   } else
 122     rtp_errors /= 2;                    /* gradual decay */
 123   written_bytes -= sizeof (struct rtp_header);
 124   const size_t written_samples = written_bytes / uaudio_sample_size;
 125   uaudio_schedule_update(written_samples);
 126   return written_samples;
 127 }
 128
 129 static void rtp_open(void) {
 130   struct addrinfo *res, *sres;
 131   static const struct addrinfo pref = {
 132     .ai_flags = 0,
 133     .ai_family = PF_INET,
 134     .ai_socktype = SOCK_DGRAM,
 135     .ai_protocol = IPPROTO_UDP,
 136   };
 137   static const struct addrinfo prefbind = {
 138     .ai_flags = AI_PASSIVE,
 139     .ai_family = PF_INET,
 140     .ai_socktype = SOCK_DGRAM,
 141     .ai_protocol = IPPROTO_UDP,
 142   };
 143   static const int one = 1;
 144   int sndbuf, target_sndbuf = 131072;
 145   socklen_t len;
 146   char *sockname, *ssockname;
 147   struct stringlist dst, src;
 148
 149   /* Get configuration */
 150   dst.n = 2;
 151   dst.s = xcalloc(2, sizeof *dst.s);
 152   dst.s[0] = uaudio_get("rtp-destination", NULL);
 153   dst.s[1] = uaudio_get("rtp-destination-port", NULL);
 154   src.n = 2;
 155   src.s = xcalloc(2, sizeof *dst.s);
 156   src.s[0] = uaudio_get("rtp-source", NULL);
 157   src.s[1] = uaudio_get("rtp-source-port", NULL);
 158   if(!dst.s[0])
 159     fatal(0, "'rtp-destination' not set");
 160   if(!dst.s[1])
 161     fatal(0, "'rtp-destination-port' not set");
 162   if(src.s[0]) {
 163     if(!src.s[1])
 164       fatal(0, "'rtp-source-port' not set");
 165     src.n = 2;
 166   } else
 167     src.n = 0;
 168   rtp_delay_threshold = atoi(uaudio_get("rtp-delay-threshold", "1000"));
 169   /* ...microseconds */
 170
 171   /* Resolve addresses */
 172   res = get_address(&dst, &pref, &sockname);
 173   if(!res) exit(-1);
 174   if(src.n) {
 175     sres = get_address(&src, &prefbind, &ssockname);
 176     if(!sres) exit(-1);
 177   } else
 178     sres = 0;
 179   /* Create the socket */
 180   if((rtp_fd = socket(res->ai_family,
 181                       res->ai_socktype,
 182                       res->ai_protocol)) < 0)
 183     fatal(errno, "error creating broadcast socket");
 184   if(multicast(res->ai_addr)) {
 185     /* Enable multicast options */
 186     const int ttl = atoi(uaudio_get("multicast-ttl", "1"));
 187     const int loop = !strcmp(uaudio_get("multicast-loop", "yes"), "yes");
 188     switch(res->ai_family) {
 189     case PF_INET: {
 190       if(setsockopt(rtp_fd, IPPROTO_IP, IP_MULTICAST_TTL,
 191                     &ttl, sizeof ttl) < 0)
 192         fatal(errno, "error setting IP_MULTICAST_TTL on multicast socket");
 193       if(setsockopt(rtp_fd, IPPROTO_IP, IP_MULTICAST_LOOP,
 194                     &loop, sizeof loop) < 0)
 195         fatal(errno, "error setting IP_MULTICAST_LOOP on multicast socket");
 196       break;
 197     }
 198     case PF_INET6: {
 199       if(setsockopt(rtp_fd, IPPROTO_IPV6, IPV6_MULTICAST_HOPS,
 200                     &ttl, sizeof ttl) < 0)
 201         fatal(errno, "error setting IPV6_MULTICAST_HOPS on multicast socket");
 202       if(setsockopt(rtp_fd, IPPROTO_IP, IPV6_MULTICAST_LOOP,
 203                     &loop, sizeof loop) < 0)
 204         fatal(errno, "error setting IPV6_MULTICAST_LOOP on multicast socket");
 205       break;
 206     }
 207     default:
 208       fatal(0, "unsupported address family %d", res->ai_family);
 209     }
 210     info("multicasting on %s TTL=%d loop=%s",
 211          sockname, ttl, loop ? "yes" : "no");
 212   } else {
 213     struct ifaddrs *ifs;
 214
 215     if(getifaddrs(&ifs) < 0)
 216       fatal(errno, "error calling getifaddrs");
 217     while(ifs) {
 218       /* (At least on Darwin) IFF_BROADCAST might be set but ifa_broadaddr
 219        * still a null pointer.  It turns out that there's a subsequent entry
 220        * for he same interface which _does_ have ifa_broadaddr though... */
 221       if((ifs->ifa_flags & IFF_BROADCAST)
 222          && ifs->ifa_broadaddr
 223          && sockaddr_equal(ifs->ifa_broadaddr, res->ai_addr))
 224         break;
 225       ifs = ifs->ifa_next;
 226     }
 227     if(ifs) {
 228       if(setsockopt(rtp_fd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0)
 229         fatal(errno, "error setting SO_BROADCAST on broadcast socket");
 230       info("broadcasting on %s (%s)", sockname, ifs->ifa_name);
 231     } else
 232       info("unicasting on %s", sockname);
 233   }
 234   /* Enlarge the socket buffer */
 235   len = sizeof sndbuf;
 236   if(getsockopt(rtp_fd, SOL_SOCKET, SO_SNDBUF,
 237                 &sndbuf, &len) < 0)
 238     fatal(errno, "error getting SO_SNDBUF");
 239   if(target_sndbuf > sndbuf) {
 240     if(setsockopt(rtp_fd, SOL_SOCKET, SO_SNDBUF,
 241                   &target_sndbuf, sizeof target_sndbuf) < 0)
 242       error(errno, "error setting SO_SNDBUF to %d", target_sndbuf);
 243     else
 244       info("changed socket send buffer size from %d to %d",
 245            sndbuf, target_sndbuf);
 246   } else
 247     info("default socket send buffer is %d",
 248          sndbuf);
 249   /* We might well want to set additional broadcast- or multicast-related
 250    * options here */
 251   if(sres && bind(rtp_fd, sres->ai_addr, sres->ai_addrlen) < 0)
 252     fatal(errno, "error binding broadcast socket to %s", ssockname);
 253   if(connect(rtp_fd, res->ai_addr, res->ai_addrlen) < 0)
 254     fatal(errno, "error connecting broadcast socket to %s", sockname);
 255 }
 256
 257 static void rtp_start(uaudio_callback *callback,
 258                       void *userdata) {
 259   /* We only support L16 (but we do stereo and mono and will convert sign) */
 260   if(uaudio_channels == 2
 261      && uaudio_bits == 16
 262      && uaudio_rate == 44100)
 263     rtp_payload = 10;
 264   else if(uaudio_channels == 1
 265      && uaudio_bits == 16
 266      && uaudio_rate == 44100)
 267     rtp_payload = 11;
 268   else
 269     fatal(0, "asked for %d/%d/%d 16/44100/1 and 16/44100/2",
 270           uaudio_bits, uaudio_rate, uaudio_channels);
 271   /* Various fields are required to have random initial values by RFC3550.  The
 272    * packet contents are highly public so there's no point asking for very
 273    * strong randomness. */
 274   gcry_create_nonce(&rtp_id, sizeof rtp_id);
 275   gcry_create_nonce(&rtp_sequence, sizeof rtp_sequence);
 276   rtp_open();
 277   uaudio_schedule_init();
 278   uaudio_thread_start(callback,
 279                       userdata,
 280                       rtp_play,
 281                       256 / uaudio_sample_size,
 282                       (NETWORK_BYTES - sizeof(struct rtp_header))
 283                       / uaudio_sample_size);
 284 }
 285
 286 static void rtp_stop(void) {
 287   uaudio_thread_stop();
 288   close(rtp_fd);
 289   rtp_fd = -1;
 290 }
 291
 292 static void rtp_activate(void) {
 293   uaudio_schedule_reactivated = 1;
 294   uaudio_thread_activate();
 295 }
 296
 297 static void rtp_deactivate(void) {
 298   uaudio_thread_deactivate();
 299 }
 300
 301 const struct uaudio uaudio_rtp = {
 302   .name = "rtp",
 303   .options = rtp_options,
 304   .start = rtp_start,
 305   .stop = rtp_stop,
 306   .activate = rtp_activate,
 307   .deactivate = rtp_deactivate
 308 };
 309
 310 /*
 311 Local Variables:
 312 c-basic-offset:2
 313 comment-column:40
 314 fill-column:79
 315 indent-tabs-mode:nil
 316 End:
 317 */