+static size_t rtp_play(void *buffer, size_t nsamples) {
+ struct rtp_header header;
+ struct iovec vec[2];
+ struct timeval now;
+
+ /* We do as much work as possible before checking what time it is */
+ /* Fill out header */
+ header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */
+ header.seq = htons(rtp_sequence++);
+ header.ssrc = rtp_id;
+ header.mpt = (rtp_reactivated ? 0x80 : 0x00) | rtp_payload;
+#if !WORDS_BIGENDIAN
+ /* Convert samples to network byte order */
+ uint16_t *u = buffer, *const limit = u + nsamples;
+ while(u < limit) {
+ *u = htons(*u);
+ ++u;
+ }
+#endif
+ vec[0].iov_base = (void *)&header;
+ vec[0].iov_len = sizeof header;
+ vec[1].iov_base = buffer;
+ vec[1].iov_len = nsamples * uaudio_sample_size;
+retry:
+ xgettimeofday(&now, NULL);
+ if(rtp_reactivated) {
+ /* We've been deactivated for some unknown interval. We need to advance
+ * rtp_timestamp to account for the dead air. */
+ /* On the first run through we'll set the start time. */
+ if(!rtp_timeval.tv_sec)
+ rtp_timeval = now;
+ /* See how much time we missed.
+ *
+ * This will be 0 on the first run through, in which case we'll not modify
+ * anything.
+ *
+ * It'll be negative in the (rare) situation where the deactivation
+ * interval is shorter than the last packet we sent. In this case we wait
+ * for that much time and then return having sent no samples, which will
+ * cause uaudio_play_thread_fn() to retry.
+ *
+ * In the normal case it will be positive.
+ */
+ const int64_t delay = tvsub_us(now, rtp_timeval); /* microseconds */
+ if(delay < 0) {
+ usleep(-delay);
+ goto retry;
+ }
+ /* Advance the RTP timestamp to the present. With 44.1KHz stereo this will
+ * overflow the intermediate value with a delay of a bit over 6 years.
+ * This seems acceptable. */
+ uint64_t update = (delay * uaudio_rate * uaudio_channels) / 1000000;
+ /* Don't throw off channel synchronization */
+ update -= update % uaudio_channels;
+ /* We log nontrivial changes */
+ if(update)
+ info("advancing rtp_time by %"PRIu64" samples", update);
+ rtp_timestamp += update;
+ rtp_timeval = now;
+ rtp_reactivated = 0;
+ } else {
+ /* Chances are we've been called right on the heels of the previous packet.
+ * If we just sent packets as fast as we got audio data we'd get way ahead
+ * of the player and some buffer somewhere would fill (or at least become
+ * unreasonably large).
+ *
+ * First find out how far ahead of the target time we are.
+ */
+ const int64_t ahead = tvsub_us(now, rtp_timeval); /* microseconds */
+ /* Only delay at all if we are nontrivially ahead. */
+ if(ahead > rtp_delay_threshold) {
+ /* Don't delay by the full amount */
+ usleep(ahead - rtp_delay_threshold / 2);
+ /* Refetch time (so we don't get out of step with reality) */
+ xgettimeofday(&now, NULL);
+ }
+ }
+ header.timestamp = htonl((uint32_t)rtp_timestamp);
+ int written_bytes;
+ do {
+ written_bytes = writev(rtp_fd, vec, 2);
+ } while(written_bytes < 0 && errno == EINTR);
+ if(written_bytes < 0) {
+ error(errno, "error transmitting audio data");
+ ++rtp_errors;
+ if(rtp_errors == 10)
+ fatal(0, "too many audio tranmission errors");
+ return 0;
+ } else
+ rtp_errors /= 2; /* gradual decay */
+ written_bytes -= sizeof (struct rtp_header);
+ size_t written_samples = written_bytes / uaudio_sample_size;
+ /* rtp_timestamp and rtp_timestamp are supposed to refer to the first sample
+ * of the next packet */
+ rtp_timestamp += written_samples;
+ const unsigned usec = (rtp_timeval.tv_usec
+ + 1000000 * written_samples / (uaudio_rate
+ * uaudio_channels));
+ /* ...will only overflow 32 bits if one packet is more than about half an
+ * hour long, which is not plausible. */
+ rtp_timeval.tv_sec += usec / 1000000;
+ rtp_timeval.tv_usec = usec % 1000000;
+ return written_samples;
+}
+
+static void rtp_open(void) {
+ struct addrinfo *res, *sres;
+ static const struct addrinfo pref = {
+ .ai_flags = 0,
+ .ai_family = PF_INET,
+ .ai_socktype = SOCK_DGRAM,
+ .ai_protocol = IPPROTO_UDP,
+ };
+ static const struct addrinfo prefbind = {
+ .ai_flags = AI_PASSIVE,
+ .ai_family = PF_INET,
+ .ai_socktype = SOCK_DGRAM,
+ .ai_protocol = IPPROTO_UDP,
+ };
+ static const int one = 1;
+ int sndbuf, target_sndbuf = 131072;
+ socklen_t len;
+ char *sockname, *ssockname;
+ struct stringlist dst, src;
+ const char *delay;
+
+ /* Get configuration */
+ dst.n = 2;
+ dst.s = xcalloc(2, sizeof *dst.s);
+ dst.s[0] = uaudio_get("rtp-destination");
+ dst.s[1] = uaudio_get("rtp-destination-port");
+ src.n = 2;
+ src.s = xcalloc(2, sizeof *dst.s);
+ src.s[0] = uaudio_get("rtp-source");
+ src.s[1] = uaudio_get("rtp-source-port");
+ if(!dst.s[0])
+ fatal(0, "'rtp-destination' not set");
+ if(!dst.s[1])
+ fatal(0, "'rtp-destination-port' not set");
+ if(src.s[0]) {
+ if(!src.s[1])
+ fatal(0, "'rtp-source-port' not set");
+ src.n = 2;
+ } else
+ src.n = 0;
+ if((delay = uaudio_get("rtp-delay-threshold")))
+ rtp_delay_threshold = atoi(delay);
+ else
+ rtp_delay_threshold = 1000; /* microseconds */
+
+ /* Resolve addresses */
+ res = get_address(&dst, &pref, &sockname);
+ if(!res) exit(-1);
+ if(src.n) {
+ sres = get_address(&src, &prefbind, &ssockname);
+ if(!sres) exit(-1);
+ } else
+ sres = 0;
+ /* Create the socket */
+ if((rtp_fd = socket(res->ai_family,
+ res->ai_socktype,
+ res->ai_protocol)) < 0)
+ fatal(errno, "error creating broadcast socket");
+ if(multicast(res->ai_addr)) {
+ /* Enable multicast options */
+ const char *ttls = uaudio_get("multicast-ttl");
+ const int ttl = ttls ? atoi(ttls) : 1;
+ const char *loops = uaudio_get("multicast-loop");
+ const int loop = loops ? !strcmp(loops, "yes") : 1;
+ switch(res->ai_family) {
+ case PF_INET: {
+ if(setsockopt(rtp_fd, IPPROTO_IP, IP_MULTICAST_TTL,
+ &ttl, sizeof ttl) < 0)
+ fatal(errno, "error setting IP_MULTICAST_TTL on multicast socket");
+ if(setsockopt(rtp_fd, IPPROTO_IP, IP_MULTICAST_LOOP,
+ &loop, sizeof loop) < 0)
+ fatal(errno, "error setting IP_MULTICAST_LOOP on multicast socket");
+ break;
+ }
+ case PF_INET6: {
+ if(setsockopt(rtp_fd, IPPROTO_IPV6, IPV6_MULTICAST_HOPS,
+ &ttl, sizeof ttl) < 0)
+ fatal(errno, "error setting IPV6_MULTICAST_HOPS on multicast socket");
+ if(setsockopt(rtp_fd, IPPROTO_IP, IPV6_MULTICAST_LOOP,
+ &loop, sizeof loop) < 0)
+ fatal(errno, "error setting IPV6_MULTICAST_LOOP on multicast socket");
+ break;
+ }
+ default:
+ fatal(0, "unsupported address family %d", res->ai_family);
+ }
+ info("multicasting on %s TTL=%d loop=%s",
+ sockname, ttl, loop ? "yes" : "no");
+ } else {
+ struct ifaddrs *ifs;
+
+ if(getifaddrs(&ifs) < 0)
+ fatal(errno, "error calling getifaddrs");
+ while(ifs) {
+ /* (At least on Darwin) IFF_BROADCAST might be set but ifa_broadaddr
+ * still a null pointer. It turns out that there's a subsequent entry
+ * for he same interface which _does_ have ifa_broadaddr though... */
+ if((ifs->ifa_flags & IFF_BROADCAST)
+ && ifs->ifa_broadaddr
+ && sockaddr_equal(ifs->ifa_broadaddr, res->ai_addr))
+ break;
+ ifs = ifs->ifa_next;
+ }
+ if(ifs) {
+ if(setsockopt(rtp_fd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0)
+ fatal(errno, "error setting SO_BROADCAST on broadcast socket");
+ info("broadcasting on %s (%s)", sockname, ifs->ifa_name);
+ } else
+ info("unicasting on %s", sockname);
+ }
+ /* Enlarge the socket buffer */
+ len = sizeof sndbuf;
+ if(getsockopt(rtp_fd, SOL_SOCKET, SO_SNDBUF,
+ &sndbuf, &len) < 0)
+ fatal(errno, "error getting SO_SNDBUF");
+ if(target_sndbuf > sndbuf) {
+ if(setsockopt(rtp_fd, SOL_SOCKET, SO_SNDBUF,
+ &target_sndbuf, sizeof target_sndbuf) < 0)
+ error(errno, "error setting SO_SNDBUF to %d", target_sndbuf);
+ else
+ info("changed socket send buffer size from %d to %d",
+ sndbuf, target_sndbuf);
+ } else
+ info("default socket send buffer is %d",
+ sndbuf);
+ /* We might well want to set additional broadcast- or multicast-related
+ * options here */
+ if(sres && bind(rtp_fd, sres->ai_addr, sres->ai_addrlen) < 0)
+ fatal(errno, "error binding broadcast socket to %s", ssockname);
+ if(connect(rtp_fd, res->ai_addr, res->ai_addrlen) < 0)
+ fatal(errno, "error connecting broadcast socket to %s", sockname);
+ /* Various fields are required to have random initial values by RFC3550. The
+ * packet contents are highly public so there's no point asking for very
+ * strong randomness. */
+ gcry_create_nonce(&rtp_id, sizeof rtp_id);
+ gcry_create_nonce(&rtp_sequence, sizeof rtp_sequence);
+ gcry_create_nonce(&rtp_timestamp, sizeof rtp_timestamp);
+ /* rtp_play() will spot this and choose an initial value */
+ rtp_timeval.tv_sec = 0;
+}
+
~mdw / disorder / commitdiff