2 * This file is part of DisOrder.
3 * Copyright (C) 2007-2009, 2011, 2013 Richard Kettlewell
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.
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.
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/>.
18 /** @file clients/playrtp.c
21 * This player supports Linux (<a href="http://www.alsa-project.org/">ALSA</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.
27 * The program runs (at least) three threads:
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
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
37 * control_thread() accepts commands from Disobedience (or anything else).
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).
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.
47 * - it is safe to read uint32_t values without a lock protecting them
53 #include <sys/socket.h>
54 #include <sys/types.h>
55 #include <sys/socket.h>
61 #include <netinet/in.h>
68 #include <arpa/inet.h>
74 #include "configuration.h"
84 #include "inputline.h"
88 /** @brief Obsolete synonym */
89 #ifndef IPV6_JOIN_GROUP
90 # define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP
93 /** @brief RTP socket */
96 /** @brief Log output */
99 /** @brief Output device */
101 /** @brief Buffer low watermark in samples */
102 unsigned minbuffer = 4 * (2 * 44100) / 10; /* 0.4 seconds */
104 /** @brief Maximum buffer size in samples
106 * We'll stop reading from the network if we have this many samples.
108 static unsigned maxbuffer;
110 /** @brief Received packets
111 * Protected by @ref receive_lock
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.
117 struct packet *received_packets;
119 /** @brief Tail of @ref received_packets
120 * Protected by @ref receive_lock
122 struct packet **received_tail = &received_packets;
124 /** @brief Lock protecting @ref received_packets
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;
130 /** @brief Condition variable signalled when @ref received_packets is updated
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;
136 /** @brief Length of @ref received_packets */
139 /** @brief Binary heap of received packets */
140 struct pheap packets;
142 /** @brief Total number of samples available
144 * We make this volatile because we inspect it without a protecting lock,
145 * so the usual pthread_* guarantees aren't available.
147 volatile uint32_t nsamples;
149 /** @brief Timestamp of next packet to play.
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.
154 uint32_t next_timestamp;
156 /** @brief True if actively playing
158 * This is true when playing and false when just buffering. */
161 /** @brief Lock protecting @ref packets */
162 pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
164 /** @brief Condition variable signalled whenever @ref packets is changed */
165 pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
167 /** @brief Backend to play with */
168 static const struct uaudio *backend;
170 HEAP_DEFINE(pheap, struct packet *, lt_packet);
172 /** @brief Control socket or NULL */
173 const char *control_socket;
175 /** @brief Buffer for debugging dump
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.
181 * Use clients/dump2wav to convert this to a WAV file, which can then be loaded
182 * into (e.g.) Audacity for further inspection.
184 * All three backends (ALSA, OSS, Core Audio) now support this option.
186 * The idea is to allow the user a few seconds to react to an audible artefact.
188 int16_t *dump_buffer;
190 /** @brief Current index within debugging dump */
193 /** @brief Size of debugging dump in samples */
194 size_t dump_size = 44100/*Hz*/ * 2/*channels*/ * 20/*seconds*/;
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' },
207 #if HAVE_ALSA_ASOUNDLIB_H
208 { "alsa", no_argument, 0, 'a' },
210 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
211 { "core-audio", no_argument, 0, 'c' },
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' },
223 /** @brief Control thread
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
230 * - @c stop will shut the player down
231 * - @c query will send back the reply @c running
232 * - anything else is ignored
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
240 static void *control_thread(void attribute((unused)) *arg) {
241 struct sockaddr_un sa;
248 assert(control_socket);
249 unlink(control_socket);
250 memset(&sa, 0, sizeof sa);
251 sa.sun_family = AF_UNIX;
252 strcpy(sa.sun_path, control_socket);
253 sfd = xsocket(PF_UNIX, SOCK_STREAM, 0);
254 if(bind(sfd, (const struct sockaddr *)&sa, sizeof sa) < 0)
255 disorder_fatal(errno, "error binding to %s", control_socket);
256 if(listen(sfd, 128) < 0)
257 disorder_fatal(errno, "error calling listen on %s", control_socket);
258 disorder_info("listening on %s", control_socket);
261 cfd = accept(sfd, (struct sockaddr *)&sa, &salen);
268 disorder_fatal(errno, "error calling accept on %s", control_socket);
271 if(!(fp = fdopen(cfd, "r+"))) {
272 disorder_error(errno, "error calling fdopen for %s connection", control_socket);
276 if(!inputline(control_socket, fp, &line, '\n')) {
277 if(!strcmp(line, "stop")) {
278 disorder_info("stopped via %s", control_socket);
279 exit(0); /* terminate immediately */
280 } else if(!strcmp(line, "query"))
281 fprintf(fp, "running");
282 else if(!strcmp(line, "getvol")) {
283 if(backend->get_volume) backend->get_volume(&vl, &vr);
285 fprintf(fp, "%d %d\n", vl, vr);
286 } else if(!strncmp(line, "setvol ", 7)) {
287 if(!backend->set_volume)
289 else if(sscanf(line + 7, "%d %d", &vl, &vr) == 2)
290 backend->set_volume(&vl, &vr);
292 backend->get_volume(&vl, &vr);
293 fprintf(fp, "%d %d\n", vl, vr);
298 disorder_error(errno, "error closing %s connection", control_socket);
302 /** @brief Drop the first packet
304 * Assumes that @ref lock is held.
306 static void drop_first_packet(void) {
307 if(pheap_count(&packets)) {
308 struct packet *const p = pheap_remove(&packets);
309 nsamples -= p->nsamples;
310 playrtp_free_packet(p);
311 pthread_cond_broadcast(&cond);
315 /** @brief Background thread adding packets to heap
317 * This just transfers packets from @ref received_packets to @ref packets. It
318 * is important that it holds @ref receive_lock for as little time as possible,
319 * in order to minimize the interval between calls to read() in
322 static void *queue_thread(void attribute((unused)) *arg) {
326 /* Get the next packet */
327 pthread_mutex_lock(&receive_lock);
328 while(!received_packets) {
329 pthread_cond_wait(&receive_cond, &receive_lock);
331 p = received_packets;
332 received_packets = p->next;
333 if(!received_packets)
334 received_tail = &received_packets;
336 pthread_mutex_unlock(&receive_lock);
337 /* Add it to the heap */
338 pthread_mutex_lock(&lock);
339 pheap_insert(&packets, p);
340 nsamples += p->nsamples;
341 pthread_cond_broadcast(&cond);
342 pthread_mutex_unlock(&lock);
344 #if HAVE_STUPID_GCC44
349 /** @brief Background thread collecting samples
351 * This function collects samples, perhaps converts them to the target format,
352 * and adds them to the packet list.
354 * It is crucial that the gap between successive calls to read() is as small as
355 * possible: otherwise packets will be dropped.
357 * We use a binary heap to ensure that the unavoidable effort is at worst
358 * logarithmic in the total number of packets - in fact if packets are mostly
359 * received in order then we will largely do constant work per packet since the
360 * newest packet will always be last.
362 * Of more concern is that we must acquire the lock on the heap to add a packet
363 * to it. If this proves a problem in practice then the answer would be
364 * (probably doubly) linked list with new packets added the end and a second
365 * thread which reads packets off the list and adds them to the heap.
367 * We keep memory allocation (mostly) very fast by keeping pre-allocated
368 * packets around; see @ref playrtp_new_packet().
370 static void *listen_thread(void attribute((unused)) *arg) {
371 struct packet *p = 0;
373 struct rtp_header header;
380 p = playrtp_new_packet();
381 iov[0].iov_base = &header;
382 iov[0].iov_len = sizeof header;
383 iov[1].iov_base = p->samples_raw;
384 iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw;
385 n = readv(rtpfd, iov, 2);
391 disorder_fatal(errno, "error reading from socket");
394 /* Ignore too-short packets */
395 if((size_t)n <= sizeof (struct rtp_header)) {
396 disorder_info("ignored a short packet");
399 timestamp = htonl(header.timestamp);
400 seq = htons(header.seq);
401 /* Ignore packets in the past */
402 if(active && lt(timestamp, next_timestamp)) {
403 disorder_info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
404 timestamp, next_timestamp);
407 /* Ignore packets with the extension bit set. */
408 if(header.vpxcc & 0x10)
412 p->timestamp = timestamp;
413 /* Convert to target format */
414 if(header.mpt & 0x80)
416 switch(header.mpt & 0x7F) {
418 p->nsamples = (n - sizeof header) / sizeof(uint16_t);
420 /* TODO support other RFC3551 media types (when the speaker does) */
422 disorder_fatal(0, "unsupported RTP payload type %d", header.mpt & 0x7F);
424 /* See if packet is silent */
425 const uint16_t *s = p->samples_raw;
433 fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
434 seq, timestamp, p->nsamples, timestamp + p->nsamples);
435 /* Stop reading if we've reached the maximum.
437 * This is rather unsatisfactory: it means that if packets get heavily
438 * out of order then we guarantee dropouts. But for now... */
439 if(nsamples >= maxbuffer) {
440 pthread_mutex_lock(&lock);
441 while(nsamples >= maxbuffer) {
442 pthread_cond_wait(&cond, &lock);
444 pthread_mutex_unlock(&lock);
446 /* Add the packet to the receive queue */
447 pthread_mutex_lock(&receive_lock);
449 received_tail = &p->next;
451 pthread_cond_signal(&receive_cond);
452 pthread_mutex_unlock(&receive_lock);
453 /* We'll need a new packet */
458 /** @brief Wait until the buffer is adequately full
460 * Must be called with @ref lock held.
462 void playrtp_fill_buffer(void) {
463 /* Discard current buffer contents */
465 //fprintf(stderr, "%8u/%u (%u) DROPPING\n", nsamples, maxbuffer, minbuffer);
468 disorder_info("Buffering...");
469 /* Wait until there's at least minbuffer samples available */
470 while(nsamples < minbuffer) {
471 //fprintf(stderr, "%8u/%u (%u) FILLING\n", nsamples, maxbuffer, minbuffer);
472 pthread_cond_wait(&cond, &lock);
474 /* Start from whatever is earliest */
475 next_timestamp = pheap_first(&packets)->timestamp;
479 /** @brief Find next packet
480 * @return Packet to play or NULL if none found
482 * The return packet is merely guaranteed not to be in the past: it might be
483 * the first packet in the future rather than one that is actually suitable to
486 * Must be called with @ref lock held.
488 struct packet *playrtp_next_packet(void) {
489 while(pheap_count(&packets)) {
490 struct packet *const p = pheap_first(&packets);
491 if(le(p->timestamp + p->nsamples, next_timestamp)) {
492 /* This packet is in the past. Drop it and try another one. */
495 /* This packet is NOT in the past. (It might be in the future
502 /* display usage message and terminate */
503 static void attribute((noreturn)) help(void) {
505 " disorder-playrtp [OPTIONS] [[ADDRESS] PORT]\n"
507 " --device, -D DEVICE Output device\n"
508 " --min, -m FRAMES Buffer low water mark\n"
509 " --max, -x FRAMES Buffer maximum size\n"
510 " --rcvbuf, -R BYTES Socket receive buffer size\n"
511 " --config, -C PATH Set configuration file\n"
512 " --api, -A API Select audio API. Possibilities:\n"
515 for(int n = 0; uaudio_apis[n]; ++n) {
516 if(uaudio_apis[n]->flags & UAUDIO_API_CLIENT) {
521 xprintf("%s", uaudio_apis[n]->name);
525 " --command, -e COMMAND Pipe audio to command.\n"
526 " --pause-mode, -P silence For -e: pauses send silence (default)\n"
527 " --pause-mode, -P suspend For -e: pauses suspend writes\n"
528 " --help, -h Display usage message\n"
529 " --version, -V Display version number\n"
535 static size_t playrtp_callback(void *buffer,
537 void attribute((unused)) *userdata) {
541 pthread_mutex_lock(&lock);
542 /* Get the next packet, junking any that are now in the past */
543 const struct packet *p = playrtp_next_packet();
544 if(p && contains(p, next_timestamp)) {
545 /* This packet is ready to play; the desired next timestamp points
546 * somewhere into it. */
548 /* Timestamp of end of packet */
549 const uint32_t packet_end = p->timestamp + p->nsamples;
551 /* Offset of desired next timestamp into current packet */
552 const uint32_t offset = next_timestamp - p->timestamp;
554 /* Pointer to audio data */
555 const uint16_t *ptr = (void *)(p->samples_raw + offset);
557 /* Compute number of samples left in packet, limited to output buffer
559 samples = packet_end - next_timestamp;
560 if(samples > max_samples)
561 samples = max_samples;
563 /* Copy into buffer, converting to native endianness */
565 int16_t *bufptr = buffer;
567 *bufptr++ = (int16_t)ntohs(*ptr++);
570 silent = !!(p->flags & SILENT);
572 /* There is no suitable packet. We introduce 0s up to the next packet, or
573 * to fill the buffer if there's no next packet or that's too many. The
574 * comparison with max_samples deals with the otherwise troubling overflow
576 samples = p ? p->timestamp - next_timestamp : max_samples;
577 if(samples > max_samples)
578 samples = max_samples;
579 //info("infill by %zu", samples);
580 memset(buffer, 0, samples * uaudio_sample_size);
585 for(size_t i = 0; i < samples; ++i) {
586 dump_buffer[dump_index++] = ((int16_t *)buffer)[i];
587 dump_index %= dump_size;
590 /* Advance timestamp */
591 next_timestamp += samples;
592 /* If we're getting behind then try to drop just silent packets
594 * In theory this shouldn't be necessary. The server is supposed to send
595 * packets at the right rate and compares the number of samples sent with the
596 * time in order to ensure this.
598 * However, various things could throw this off:
600 * - the server's clock could advance at the wrong rate. This would cause it
601 * to mis-estimate the right number of samples to have sent and
602 * inappropriately throttle or speed up.
604 * - playback could happen at the wrong rate. If the playback host's sound
605 * card has a slightly incorrect clock then eventually it will get out
608 * So if we play back slightly slower than the server sends for either of
609 * these reasons then eventually our buffer, and the socket's buffer, will
610 * fill, and the kernel will start dropping packets. The result is audible
613 * Therefore if we're getting behind, we pre-emptively drop silent packets,
614 * since a change in the duration of a silence is less noticeable than a
615 * dropped packet from the middle of continuous music.
617 * (If things go wrong the other way then eventually we run out of packets to
618 * play and are forced to play silence. This doesn't seem to happen in
619 * practice but if it does then in the same way we can artificially extend
620 * silent packets to compensate.)
622 * Dropped packets are always logged; use 'disorder-playrtp --monitor' to
623 * track how close to target buffer occupancy we are on a once-a-minute
626 if(nsamples > minbuffer && silent) {
627 disorder_info("dropping %zu samples (%"PRIu32" > %"PRIu32")",
628 samples, nsamples, minbuffer);
631 /* Junk obsolete packets */
632 playrtp_next_packet();
633 pthread_mutex_unlock(&lock);
637 static int compare_family(const struct ifaddrs *a,
638 const struct ifaddrs *b,
640 int afamily = a->ifa_addr->sa_family;
641 int bfamily = b->ifa_addr->sa_family;
642 if(afamily != bfamily) {
643 /* Preferred family wins */
644 if(afamily == family) return 1;
645 if(bfamily == family) return -1;
646 /* Either there's no preference or it doesn't help. Prefer IPv4 */
647 if(afamily == AF_INET) return 1;
648 if(bfamily == AF_INET) return -1;
649 /* Failing that prefer IPv6 */
650 if(afamily == AF_INET6) return 1;
651 if(bfamily == AF_INET6) return -1;
656 static int compare_flags(const struct ifaddrs *a,
657 const struct ifaddrs *b) {
658 unsigned aflags = a->ifa_flags, bflags = b->ifa_flags;
659 /* Up interfaces are better than down ones */
660 unsigned aup = aflags & IFF_UP, bup = bflags & IFF_UP;
662 return aup > bup ? 1 : -1;
664 /* Static addresses are better than dynamic */
665 unsigned adynamic = aflags & IFF_DYNAMIC, bdynamic = bflags & IFF_DYNAMIC;
666 if(adynamic != bdynamic)
667 return adynamic < bdynamic ? 1 : -1;
669 unsigned aloopback = aflags & IFF_LOOPBACK, bloopback = bflags & IFF_LOOPBACK;
670 /* Static addresses are better than dynamic */
671 if(aloopback != bloopback)
672 return aloopback < bloopback ? 1 : -1;
676 static int compare_interfaces(const struct ifaddrs *a,
677 const struct ifaddrs *b,
680 if((c = compare_family(a, b, family))) return c;
681 return compare_flags(a, b);
684 int main(int argc, char **argv) {
686 struct addrinfo *res;
687 struct stringlist sl;
689 int rcvbuf, target_rcvbuf = 0;
692 struct ipv6_mreq mreq6;
693 disorder_client *c = NULL;
694 char *address, *port;
698 struct sockaddr_in in;
699 struct sockaddr_in6 in6;
701 union any_sockaddr mgroup;
702 const char *dumpfile = 0;
705 static const int one = 1;
707 struct addrinfo prefs = {
708 .ai_flags = AI_PASSIVE,
709 .ai_family = PF_INET,
710 .ai_socktype = SOCK_DGRAM,
711 .ai_protocol = IPPROTO_UDP
714 /* Timing information is often important to debugging playrtp, so we include
715 * timestamps in the logs */
718 if(!setlocale(LC_CTYPE, "")) disorder_fatal(errno, "error calling setlocale");
719 while((n = getopt_long(argc, argv, "hVdD:m:x:L:R:aocC:re:P:MA:", options, 0)) >= 0) {
722 case 'V': version("disorder-playrtp");
723 case 'd': debugging = 1; break;
724 case 'D': uaudio_set("device", optarg); break;
725 case 'm': minbuffer = 2 * atol(optarg); break;
726 case 'x': maxbuffer = 2 * atol(optarg); break;
727 case 'L': logfp = fopen(optarg, "w"); break;
728 case 'R': target_rcvbuf = atoi(optarg); break;
729 #if HAVE_ALSA_ASOUNDLIB_H
731 disorder_error(0, "deprecated option; use --api alsa instead");
732 backend = &uaudio_alsa; break;
734 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
736 disorder_error(0, "deprecated option; use --api oss instead");
737 backend = &uaudio_oss;
740 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
742 disorder_error(0, "deprecated option; use --api coreaudio instead");
743 backend = &uaudio_coreaudio;
746 case 'A': backend = uaudio_find(optarg); break;
747 case 'C': configfile = optarg; break;
748 case 's': control_socket = optarg; break;
749 case 'r': dumpfile = optarg; break;
750 case 'e': backend = &uaudio_command; uaudio_set("command", optarg); break;
751 case 'P': uaudio_set("pause-mode", optarg); break;
752 case 'M': monitor = 1; break;
753 default: disorder_fatal(0, "invalid option");
756 if(config_read(0, NULL)) disorder_fatal(0, "cannot read configuration");
757 /* Choose a sensible default audio backend */
759 backend = uaudio_default(uaudio_apis, UAUDIO_API_CLIENT);
761 disorder_fatal(0, "no default uaudio API found");
762 disorder_info("default audio API %s", backend->name);
764 if(backend == &uaudio_rtp) {
765 /* This means that you have NO local sound output. This can happen if you
766 * use a non-Apple GCC on a Mac (because it doesn't know how to compile
767 * CoreAudio/AudioHardware.h). */
768 disorder_fatal(0, "cannot play RTP through RTP");
771 maxbuffer = 2 * minbuffer;
776 /* Get configuration from server */
777 if(!(c = disorder_new(1))) exit(EXIT_FAILURE);
778 if(disorder_connect(c)) exit(EXIT_FAILURE);
779 if(disorder_rtp_address(c, &address, &port)) exit(EXIT_FAILURE);
781 sl.s = xcalloc(2, sizeof *sl.s);
787 /* Use command-line ADDRESS+PORT or just PORT */
792 disorder_fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]");
794 disorder_info("version "VERSION" process ID %lu",
795 (unsigned long)getpid());
796 struct sockaddr *addr;
798 if(!strcmp(sl.s[0], "-")) {
799 /* We'll need a connection to request the incoming stream, so open one if
800 * we don't have one already */
802 if(!(c = disorder_new(1))) exit(EXIT_FAILURE);
803 if(disorder_connect(c)) exit(EXIT_FAILURE);
805 /* Pick address family to match known-working connectivity to the server */
806 int family = disorder_client_af(c);
807 /* Get a list of interfaces */
808 struct ifaddrs *ifa, *bestifa = NULL;
809 if(getifaddrs(&ifa) < 0)
810 disorder_fatal(errno, "error calling getifaddrs");
811 /* Try to pick a good one */
812 for(; ifa; ifa = ifa->ifa_next) {
813 if(!ifa->ifa_addr) continue;
815 || compare_interfaces(ifa, bestifa, family) > 0)
819 disorder_fatal(0, "failed to select a network interface");
820 family = bestifa->ifa_addr->sa_family;
821 if((rtpfd = socket(family,
824 disorder_fatal(errno, "error creating socket (family %d)", family);
825 /* Bind the address */
826 if(bind(rtpfd, bestifa->ifa_addr,
828 ? sizeof (struct sockaddr_in) : sizeof (struct sockaddr_in6)) < 0)
829 disorder_fatal(errno, "error binding socket");
830 static struct sockaddr_storage bound_address;
831 addr = (struct sockaddr *)&bound_address;
832 addr_len = sizeof bound_address;
833 if(getsockname(rtpfd, addr, &addr_len) < 0)
834 disorder_fatal(errno, "error getting socket address");
835 /* Convert to string */
836 char addrname[128], portname[32];
837 if(getnameinfo(addr, addr_len,
838 addrname, sizeof addrname,
839 portname, sizeof portname,
840 NI_NUMERICHOST|NI_NUMERICSERV) < 0)
841 disorder_fatal(errno, "getnameinfo");
842 /* Ask for audio data */
843 if(disorder_rtp_request(c, addrname, portname)) exit(EXIT_FAILURE);
844 /* Report what we did */
845 disorder_info("listening on %s", format_sockaddr(addr));
847 /* Look up address and port */
848 if(!(res = get_address(&sl, &prefs, &sockname)))
851 addr_len = res->ai_addrlen;
852 /* Create the socket */
853 if((rtpfd = socket(res->ai_family,
855 res->ai_protocol)) < 0)
856 disorder_fatal(errno, "error creating socket");
857 /* Allow multiple listeners */
858 xsetsockopt(rtpfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one);
859 is_multicast = multicast(addr);
860 /* The multicast and unicast/broadcast cases are different enough that they
861 * are totally split. Trying to find commonality between them causes more
862 * trouble that it's worth. */
864 /* Stash the multicast group address */
865 memcpy(&mgroup, addr, addr_len);
866 switch(res->ai_addr->sa_family) {
868 mgroup.in.sin_port = 0;
871 mgroup.in6.sin6_port = 0;
874 disorder_fatal(0, "unsupported address family %d",
875 (int)addr->sa_family);
877 /* Bind to to the multicast group address */
878 if(bind(rtpfd, addr, addr_len) < 0)
879 disorder_fatal(errno, "error binding socket to %s",
880 format_sockaddr(addr));
881 /* Add multicast group membership */
882 switch(mgroup.sa.sa_family) {
884 mreq.imr_multiaddr = mgroup.in.sin_addr;
885 mreq.imr_interface.s_addr = 0; /* use primary interface */
886 if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP,
887 &mreq, sizeof mreq) < 0)
888 disorder_fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP");
891 mreq6.ipv6mr_multiaddr = mgroup.in6.sin6_addr;
892 memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface);
893 if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP,
894 &mreq6, sizeof mreq6) < 0)
895 disorder_fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP");
898 disorder_fatal(0, "unsupported address family %d", res->ai_family);
900 /* Report what we did */
901 disorder_info("listening on %s multicast group %s",
902 format_sockaddr(addr), format_sockaddr(&mgroup.sa));
905 switch(addr->sa_family) {
907 struct sockaddr_in *in = (struct sockaddr_in *)addr;
909 memset(&in->sin_addr, 0, sizeof (struct in_addr));
913 struct sockaddr_in6 *in6 = (struct sockaddr_in6 *)addr;
915 memset(&in6->sin6_addr, 0, sizeof (struct in6_addr));
919 disorder_fatal(0, "unsupported family %d", (int)addr->sa_family);
921 if(bind(rtpfd, addr, addr_len) < 0)
922 disorder_fatal(errno, "error binding socket to %s",
923 format_sockaddr(addr));
924 /* Report what we did */
925 disorder_info("listening on %s", format_sockaddr(addr));
929 if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0)
930 disorder_fatal(errno, "error calling getsockopt SO_RCVBUF");
931 if(target_rcvbuf > rcvbuf) {
932 if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF,
933 &target_rcvbuf, sizeof target_rcvbuf) < 0)
934 disorder_error(errno, "error calling setsockopt SO_RCVBUF %d",
936 /* We try to carry on anyway */
938 disorder_info("changed socket receive buffer from %d to %d",
939 rcvbuf, target_rcvbuf);
941 disorder_info("default socket receive buffer %d", rcvbuf);
942 //info("minbuffer %u maxbuffer %u", minbuffer, maxbuffer);
944 disorder_info("WARNING: -L option can impact performance");
948 if((err = pthread_create(&tid, 0, control_thread, 0)))
949 disorder_fatal(err, "pthread_create control_thread");
953 unsigned char buffer[65536];
956 if((fd = open(dumpfile, O_RDWR|O_TRUNC|O_CREAT, 0666)) < 0)
957 disorder_fatal(errno, "opening %s", dumpfile);
958 /* Fill with 0s to a suitable size */
959 memset(buffer, 0, sizeof buffer);
960 for(written = 0; written < dump_size * sizeof(int16_t);
961 written += sizeof buffer) {
962 if(write(fd, buffer, sizeof buffer) < 0)
963 disorder_fatal(errno, "clearing %s", dumpfile);
965 /* Map the buffer into memory for convenience */
966 dump_buffer = mmap(0, dump_size * sizeof(int16_t), PROT_READ|PROT_WRITE,
968 if(dump_buffer == (void *)-1)
969 disorder_fatal(errno, "mapping %s", dumpfile);
970 disorder_info("dumping to %s", dumpfile);
972 /* Set up output. Currently we only support L16 so there's no harm setting
973 * the format before we know what it is! */
974 uaudio_set_format(44100/*Hz*/, 2/*channels*/,
975 16/*bits/channel*/, 1/*signed*/);
976 uaudio_set("application", "disorder-playrtp");
977 backend->configure();
978 backend->start(playrtp_callback, NULL);
979 if(backend->open_mixer) backend->open_mixer();
980 /* We receive and convert audio data in a background thread */
981 if((err = pthread_create(<id, 0, listen_thread, 0)))
982 disorder_fatal(err, "pthread_create listen_thread");
983 /* We have a second thread to add received packets to the queue */
984 if((err = pthread_create(<id, 0, queue_thread, 0)))
985 disorder_fatal(err, "pthread_create queue_thread");
986 pthread_mutex_lock(&lock);
989 /* Wait for the buffer to fill up a bit */
990 playrtp_fill_buffer();
991 /* Start playing now */
992 disorder_info("Playing...");
993 next_timestamp = pheap_first(&packets)->timestamp;
995 pthread_mutex_unlock(&lock);
997 pthread_mutex_lock(&lock);
998 /* Wait until the buffer empties out
1000 * If there's a packet that we can play right now then we definitely
1003 * Also if there's at least minbuffer samples we carry on regardless and
1004 * insert silence. The assumption is there's been a pause but more data
1007 while(nsamples >= minbuffer
1009 && contains(pheap_first(&packets), next_timestamp))) {
1011 time_t now = xtime(0);
1013 if(now >= lastlog + 60) {
1014 int offset = nsamples - minbuffer;
1015 double offtime = (double)offset / (uaudio_rate * uaudio_channels);
1016 disorder_info("%+d samples off (%d.%02ds, %d bytes)",
1018 (int)fabs(offtime) * (offtime < 0 ? -1 : 1),
1019 (int)(fabs(offtime) * 100) % 100,
1020 offset * uaudio_bits / CHAR_BIT);
1024 //fprintf(stderr, "%8u/%u (%u) PLAYING\n", nsamples, maxbuffer, minbuffer);
1025 pthread_cond_wait(&cond, &lock);
1029 struct packet *p = pheap_first(&packets);
1030 fprintf(stderr, "nsamples=%u (%u) next_timestamp=%"PRIx32", first packet is [%"PRIx32",%"PRIx32")\n",
1031 nsamples, minbuffer, next_timestamp,p->timestamp,p->timestamp+p->nsamples);
1034 /* Stop playing for a bit until the buffer re-fills */
1035 pthread_mutex_unlock(&lock);
1036 backend->deactivate();
1037 pthread_mutex_lock(&lock);
1049 indent-tabs-mode:nil