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"
85 #include "inputline.h"
89 /** @brief Obsolete synonym */
90 #ifndef IPV6_JOIN_GROUP
91 # define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP
94 /** @brief RTP socket */
97 /** @brief Log output */
100 /** @brief Output device */
102 /** @brief Buffer low watermark in samples */
105 /** @brief Maximum buffer size in samples
107 * We'll stop reading from the network if we have this many samples.
109 static unsigned maxbuffer;
111 /** @brief Received packets
112 * Protected by @ref receive_lock
114 * Received packets are added to this list, and queue_thread() picks them off
115 * it and adds them to @ref packets. Whenever a packet is added to it, @ref
116 * receive_cond is signalled.
118 struct packet *received_packets;
120 /** @brief Tail of @ref received_packets
121 * Protected by @ref receive_lock
123 struct packet **received_tail = &received_packets;
125 /** @brief Lock protecting @ref received_packets
127 * Only listen_thread() and queue_thread() ever hold this lock. It is vital
128 * that queue_thread() not hold it any longer than it strictly has to. */
129 pthread_mutex_t receive_lock = PTHREAD_MUTEX_INITIALIZER;
131 /** @brief Condition variable signalled when @ref received_packets is updated
133 * Used by listen_thread() to notify queue_thread() that it has added another
134 * packet to @ref received_packets. */
135 pthread_cond_t receive_cond = PTHREAD_COND_INITIALIZER;
137 /** @brief Length of @ref received_packets */
140 /** @brief Binary heap of received packets */
141 struct pheap packets;
143 /** @brief Total number of samples available
145 * We make this volatile because we inspect it without a protecting lock,
146 * so the usual pthread_* guarantees aren't available.
148 volatile uint32_t nsamples;
150 /** @brief Timestamp of next packet to play.
152 * This is set to the timestamp of the last packet, plus the number of
153 * samples it contained. Only valid if @ref active is nonzero.
155 uint32_t next_timestamp;
157 /** @brief True if actively playing
159 * This is true when playing and false when just buffering. */
162 /** @brief Lock protecting @ref packets */
163 pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
165 /** @brief Condition variable signalled whenever @ref packets is changed */
166 pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
168 /** @brief Backend to play with */
169 static const struct uaudio *backend;
171 HEAP_DEFINE(pheap, struct packet *, lt_packet);
173 /** @brief Control socket or NULL */
174 const char *control_socket;
176 /** @brief Buffer for debugging dump
178 * The debug dump is enabled by the @c --dump option. It records the last 20s
179 * of audio to the specified file (which will be about 3.5Mbytes). The file is
180 * written as as ring buffer, so the start point will progress through it.
182 * Use clients/dump2wav to convert this to a WAV file, which can then be loaded
183 * into (e.g.) Audacity for further inspection.
185 * All three backends (ALSA, OSS, Core Audio) now support this option.
187 * The idea is to allow the user a few seconds to react to an audible artefact.
189 int16_t *dump_buffer;
191 /** @brief Current index within debugging dump */
194 /** @brief Size of debugging dump in samples */
195 size_t dump_size = 44100/*Hz*/ * 2/*channels*/ * 20/*seconds*/;
197 static const struct option options[] = {
198 { "help", no_argument, 0, 'h' },
199 { "version", no_argument, 0, 'V' },
200 { "debug", no_argument, 0, 'd' },
201 { "device", required_argument, 0, 'D' },
202 { "min", required_argument, 0, 'm' },
203 { "max", required_argument, 0, 'x' },
204 { "rcvbuf", required_argument, 0, 'R' },
205 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
206 { "oss", no_argument, 0, 'o' },
208 #if HAVE_ALSA_ASOUNDLIB_H
209 { "alsa", no_argument, 0, 'a' },
211 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
212 { "core-audio", no_argument, 0, 'c' },
214 { "api", required_argument, 0, 'A' },
215 { "dump", required_argument, 0, 'r' },
216 { "command", required_argument, 0, 'e' },
217 { "pause-mode", required_argument, 0, 'P' },
218 { "socket", required_argument, 0, 's' },
219 { "config", required_argument, 0, 'C' },
220 { "user-config", required_argument, 0, 'u' },
221 { "monitor", no_argument, 0, 'M' },
225 /** @brief Control thread
227 * This thread is responsible for accepting control commands from Disobedience
228 * (or other controllers) over an AF_UNIX stream socket with a path specified
229 * by the @c --socket option. The protocol uses simple string commands and
232 * - @c stop will shut the player down
233 * - @c query will send back the reply @c running
234 * - anything else is ignored
236 * Commands and response strings terminated by shutting down the connection or
237 * by a newline. No attempt is made to multiplex multiple clients so it is
238 * important that the command be sent as soon as the connection is made - it is
239 * assumed that both parties to the protocol are entirely cooperating with one
242 static void *control_thread(void attribute((unused)) *arg) {
243 struct sockaddr_un sa;
250 assert(control_socket);
251 unlink(control_socket);
252 memset(&sa, 0, sizeof sa);
253 sa.sun_family = AF_UNIX;
254 strcpy(sa.sun_path, control_socket);
255 sfd = xsocket(PF_UNIX, SOCK_STREAM, 0);
256 if(bind(sfd, (const struct sockaddr *)&sa, sizeof sa) < 0)
257 disorder_fatal(errno, "error binding to %s", control_socket);
258 if(listen(sfd, 128) < 0)
259 disorder_fatal(errno, "error calling listen on %s", control_socket);
260 disorder_info("listening on %s", control_socket);
263 cfd = accept(sfd, (struct sockaddr *)&sa, &salen);
270 disorder_fatal(errno, "error calling accept on %s", control_socket);
273 if(!(fp = fdopen(cfd, "r+"))) {
274 disorder_error(errno, "error calling fdopen for %s connection", control_socket);
278 if(!inputline(control_socket, fp, &line, '\n')) {
279 if(!strcmp(line, "stop")) {
280 disorder_info("stopped via %s", control_socket);
281 exit(0); /* terminate immediately */
282 } else if(!strcmp(line, "query"))
283 fprintf(fp, "running");
284 else if(!strcmp(line, "getvol")) {
285 if(backend->get_volume) backend->get_volume(&vl, &vr);
287 fprintf(fp, "%d %d\n", vl, vr);
288 } else if(!strncmp(line, "setvol ", 7)) {
289 if(!backend->set_volume)
291 else if(sscanf(line + 7, "%d %d", &vl, &vr) == 2)
292 backend->set_volume(&vl, &vr);
294 backend->get_volume(&vl, &vr);
295 fprintf(fp, "%d %d\n", vl, vr);
300 disorder_error(errno, "error closing %s connection", control_socket);
304 /** @brief Drop the first packet
306 * Assumes that @ref lock is held.
308 static void drop_first_packet(void) {
309 if(pheap_count(&packets)) {
310 struct packet *const p = pheap_remove(&packets);
311 nsamples -= p->nsamples;
312 playrtp_free_packet(p);
313 pthread_cond_broadcast(&cond);
317 /** @brief Background thread adding packets to heap
319 * This just transfers packets from @ref received_packets to @ref packets. It
320 * is important that it holds @ref receive_lock for as little time as possible,
321 * in order to minimize the interval between calls to read() in
324 static void *queue_thread(void attribute((unused)) *arg) {
328 /* Get the next packet */
329 pthread_mutex_lock(&receive_lock);
330 while(!received_packets) {
331 pthread_cond_wait(&receive_cond, &receive_lock);
333 p = received_packets;
334 received_packets = p->next;
335 if(!received_packets)
336 received_tail = &received_packets;
338 pthread_mutex_unlock(&receive_lock);
339 /* Add it to the heap */
340 pthread_mutex_lock(&lock);
341 pheap_insert(&packets, p);
342 nsamples += p->nsamples;
343 pthread_cond_broadcast(&cond);
344 pthread_mutex_unlock(&lock);
346 #if HAVE_STUPID_GCC44
351 /** @brief Background thread collecting samples
353 * This function collects samples, perhaps converts them to the target format,
354 * and adds them to the packet list.
356 * It is crucial that the gap between successive calls to read() is as small as
357 * possible: otherwise packets will be dropped.
359 * We use a binary heap to ensure that the unavoidable effort is at worst
360 * logarithmic in the total number of packets - in fact if packets are mostly
361 * received in order then we will largely do constant work per packet since the
362 * newest packet will always be last.
364 * Of more concern is that we must acquire the lock on the heap to add a packet
365 * to it. If this proves a problem in practice then the answer would be
366 * (probably doubly) linked list with new packets added the end and a second
367 * thread which reads packets off the list and adds them to the heap.
369 * We keep memory allocation (mostly) very fast by keeping pre-allocated
370 * packets around; see @ref playrtp_new_packet().
372 static void *listen_thread(void attribute((unused)) *arg) {
373 struct packet *p = 0;
375 struct rtp_header header;
382 p = playrtp_new_packet();
383 iov[0].iov_base = &header;
384 iov[0].iov_len = sizeof header;
385 iov[1].iov_base = p->samples_raw;
386 iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw;
387 n = readv(rtpfd, iov, 2);
393 disorder_fatal(errno, "error reading from socket");
396 /* Ignore too-short packets */
397 if((size_t)n <= sizeof (struct rtp_header)) {
398 disorder_info("ignored a short packet");
401 timestamp = htonl(header.timestamp);
402 seq = htons(header.seq);
403 /* Ignore packets in the past */
404 if(active && lt(timestamp, next_timestamp)) {
405 disorder_info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
406 timestamp, next_timestamp);
409 /* Ignore packets with the extension bit set. */
410 if(header.vpxcc & 0x10)
414 p->timestamp = timestamp;
415 /* Convert to target format */
416 if(header.mpt & 0x80)
418 switch(header.mpt & 0x7F) {
420 p->nsamples = (n - sizeof header) / sizeof(uint16_t);
422 /* TODO support other RFC3551 media types (when the speaker does) */
424 disorder_fatal(0, "unsupported RTP payload type %d", header.mpt & 0x7F);
426 /* See if packet is silent */
427 const uint16_t *s = p->samples_raw;
435 fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
436 seq, timestamp, p->nsamples, timestamp + p->nsamples);
437 /* Stop reading if we've reached the maximum.
439 * This is rather unsatisfactory: it means that if packets get heavily
440 * out of order then we guarantee dropouts. But for now... */
441 if(nsamples >= maxbuffer) {
442 pthread_mutex_lock(&lock);
443 while(nsamples >= maxbuffer) {
444 pthread_cond_wait(&cond, &lock);
446 pthread_mutex_unlock(&lock);
448 /* Add the packet to the receive queue */
449 pthread_mutex_lock(&receive_lock);
451 received_tail = &p->next;
453 pthread_cond_signal(&receive_cond);
454 pthread_mutex_unlock(&receive_lock);
455 /* We'll need a new packet */
460 /** @brief Wait until the buffer is adequately full
462 * Must be called with @ref lock held.
464 void playrtp_fill_buffer(void) {
465 /* Discard current buffer contents */
467 //fprintf(stderr, "%8u/%u (%u) DROPPING\n", nsamples, maxbuffer, minbuffer);
470 disorder_info("Buffering...");
471 /* Wait until there's at least minbuffer samples available */
472 while(nsamples < minbuffer) {
473 //fprintf(stderr, "%8u/%u (%u) FILLING\n", nsamples, maxbuffer, minbuffer);
474 pthread_cond_wait(&cond, &lock);
476 /* Start from whatever is earliest */
477 next_timestamp = pheap_first(&packets)->timestamp;
481 /** @brief Find next packet
482 * @return Packet to play or NULL if none found
484 * The return packet is merely guaranteed not to be in the past: it might be
485 * the first packet in the future rather than one that is actually suitable to
488 * Must be called with @ref lock held.
490 struct packet *playrtp_next_packet(void) {
491 while(pheap_count(&packets)) {
492 struct packet *const p = pheap_first(&packets);
493 if(le(p->timestamp + p->nsamples, next_timestamp)) {
494 /* This packet is in the past. Drop it and try another one. */
497 /* This packet is NOT in the past. (It might be in the future
504 /* display usage message and terminate */
505 static void attribute((noreturn)) help(void) {
507 " disorder-playrtp [OPTIONS] [[ADDRESS] PORT]\n"
509 " --device, -D DEVICE Output device\n"
510 " --min, -m FRAMES Buffer low water mark\n"
511 " --max, -x FRAMES Buffer maximum size\n"
512 " --rcvbuf, -R BYTES Socket receive buffer size\n"
513 " --config, -C PATH Set system configuration file\n"
514 " --user-config, -u PATH Set user configuration file\n"
515 " --api, -A API Select audio API. Possibilities:\n"
518 for(int n = 0; uaudio_apis[n]; ++n) {
519 if(uaudio_apis[n]->flags & UAUDIO_API_CLIENT) {
524 xprintf("%s", uaudio_apis[n]->name);
528 " --command, -e COMMAND Pipe audio to command.\n"
529 " --pause-mode, -P silence For -e: pauses send silence (default)\n"
530 " --pause-mode, -P suspend For -e: pauses suspend writes\n"
531 " --help, -h Display usage message\n"
532 " --version, -V Display version number\n"
538 static size_t playrtp_callback(void *buffer,
540 void attribute((unused)) *userdata) {
544 pthread_mutex_lock(&lock);
545 /* Get the next packet, junking any that are now in the past */
546 const struct packet *p = playrtp_next_packet();
547 if(p && contains(p, next_timestamp)) {
548 /* This packet is ready to play; the desired next timestamp points
549 * somewhere into it. */
551 /* Timestamp of end of packet */
552 const uint32_t packet_end = p->timestamp + p->nsamples;
554 /* Offset of desired next timestamp into current packet */
555 const uint32_t offset = next_timestamp - p->timestamp;
557 /* Pointer to audio data */
558 const uint16_t *ptr = (void *)(p->samples_raw + offset);
560 /* Compute number of samples left in packet, limited to output buffer
562 samples = packet_end - next_timestamp;
563 if(samples > max_samples)
564 samples = max_samples;
566 /* Copy into buffer, converting to native endianness */
568 int16_t *bufptr = buffer;
570 *bufptr++ = (int16_t)ntohs(*ptr++);
573 silent = !!(p->flags & SILENT);
575 /* There is no suitable packet. We introduce 0s up to the next packet, or
576 * to fill the buffer if there's no next packet or that's too many. The
577 * comparison with max_samples deals with the otherwise troubling overflow
579 samples = p ? p->timestamp - next_timestamp : max_samples;
580 if(samples > max_samples)
581 samples = max_samples;
582 //info("infill by %zu", samples);
583 memset(buffer, 0, samples * uaudio_sample_size);
588 for(size_t i = 0; i < samples; ++i) {
589 dump_buffer[dump_index++] = ((int16_t *)buffer)[i];
590 dump_index %= dump_size;
593 /* Advance timestamp */
594 next_timestamp += samples;
595 /* If we're getting behind then try to drop just silent packets
597 * In theory this shouldn't be necessary. The server is supposed to send
598 * packets at the right rate and compares the number of samples sent with the
599 * time in order to ensure this.
601 * However, various things could throw this off:
603 * - the server's clock could advance at the wrong rate. This would cause it
604 * to mis-estimate the right number of samples to have sent and
605 * inappropriately throttle or speed up.
607 * - playback could happen at the wrong rate. If the playback host's sound
608 * card has a slightly incorrect clock then eventually it will get out
611 * So if we play back slightly slower than the server sends for either of
612 * these reasons then eventually our buffer, and the socket's buffer, will
613 * fill, and the kernel will start dropping packets. The result is audible
616 * Therefore if we're getting behind, we pre-emptively drop silent packets,
617 * since a change in the duration of a silence is less noticeable than a
618 * dropped packet from the middle of continuous music.
620 * (If things go wrong the other way then eventually we run out of packets to
621 * play and are forced to play silence. This doesn't seem to happen in
622 * practice but if it does then in the same way we can artificially extend
623 * silent packets to compensate.)
625 * Dropped packets are always logged; use 'disorder-playrtp --monitor' to
626 * track how close to target buffer occupancy we are on a once-a-minute
629 if(nsamples > minbuffer && silent) {
630 disorder_info("dropping %zu samples (%"PRIu32" > %"PRIu32")",
631 samples, nsamples, minbuffer);
634 /* Junk obsolete packets */
635 playrtp_next_packet();
636 pthread_mutex_unlock(&lock);
640 int main(int argc, char **argv) {
642 struct addrinfo *res;
643 struct stringlist sl;
645 int rcvbuf, target_rcvbuf = -1;
648 struct ipv6_mreq mreq6;
649 disorder_client *c = NULL;
650 char *address, *port;
654 struct sockaddr_in in;
655 struct sockaddr_in6 in6;
657 union any_sockaddr mgroup;
658 const char *dumpfile = 0;
661 static const int one = 1;
663 struct addrinfo prefs = {
664 .ai_flags = AI_PASSIVE,
665 .ai_family = PF_INET,
666 .ai_socktype = SOCK_DGRAM,
667 .ai_protocol = IPPROTO_UDP
670 /* Timing information is often important to debugging playrtp, so we include
671 * timestamps in the logs */
674 if(!setlocale(LC_CTYPE, "")) disorder_fatal(errno, "error calling setlocale");
675 while((n = getopt_long(argc, argv, "hVdD:m:x:L:R:aocC:u:re:P:MA:", options, 0)) >= 0) {
678 case 'V': version("disorder-playrtp");
679 case 'd': debugging = 1; break;
680 case 'D': uaudio_set("device", optarg); break;
681 case 'm': minbuffer = 2 * atol(optarg); break;
682 case 'x': maxbuffer = 2 * atol(optarg); break;
683 case 'L': logfp = fopen(optarg, "w"); break;
684 case 'R': target_rcvbuf = atoi(optarg); break;
685 #if HAVE_ALSA_ASOUNDLIB_H
687 disorder_error(0, "deprecated option; use --api alsa instead");
688 backend = &uaudio_alsa; break;
690 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
692 disorder_error(0, "deprecated option; use --api oss instead");
693 backend = &uaudio_oss;
696 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
698 disorder_error(0, "deprecated option; use --api coreaudio instead");
699 backend = &uaudio_coreaudio;
702 case 'A': backend = uaudio_find(optarg); break;
703 case 'C': configfile = optarg; break;
704 case 'u': userconfigfile = optarg; break;
705 case 's': control_socket = optarg; break;
706 case 'r': dumpfile = optarg; break;
707 case 'e': backend = &uaudio_command; uaudio_set("command", optarg); break;
708 case 'P': uaudio_set("pause-mode", optarg); break;
709 case 'M': monitor = 1; break;
710 default: disorder_fatal(0, "invalid option");
713 if(config_read(0, NULL)) disorder_fatal(0, "cannot read configuration");
714 /* Choose a sensible default audio backend */
716 backend = uaudio_default(uaudio_apis, UAUDIO_API_CLIENT);
718 disorder_fatal(0, "no default uaudio API found");
719 disorder_info("default audio API %s", backend->name);
721 if(backend == &uaudio_rtp) {
722 /* This means that you have NO local sound output. This can happen if you
723 * use a non-Apple GCC on a Mac (because it doesn't know how to compile
724 * CoreAudio/AudioHardware.h). */
725 disorder_fatal(0, "cannot play RTP through RTP");
727 /* Set buffering parameters if not overridden */
729 minbuffer = config->rtp_minbuffer;
730 if(!minbuffer) minbuffer = (2*44100)*4/10;
733 maxbuffer = config->rtp_maxbuffer;
734 if(!maxbuffer) maxbuffer = 2 * minbuffer;
736 if(target_rcvbuf < 0) target_rcvbuf = config->rtp_rcvbuf;
741 sl.s = xcalloc(3, sizeof *sl.s);
742 if(config->rtp_always_request) {
743 sl.s[0] = sl.s[1] = (/*unconst*/ char *)"-";
746 /* Get configuration from server */
747 if(!(c = disorder_new(1))) exit(EXIT_FAILURE);
748 if(disorder_connect(c)) exit(EXIT_FAILURE);
749 if(disorder_rtp_address(c, &address, &port)) exit(EXIT_FAILURE);
754 /* If we're requesting a new stream then apply the local network address
757 if(!strcmp(sl.s[0], "-")) {
758 if(config->rtp_request_address.port)
759 byte_xasprintf(&sl.s[1], "%d", config->rtp_request_address.port);
760 if(config->rtp_request_address.address) {
762 sl.s[1] = config->rtp_request_address.address;
767 case 1: case 2: case 3:
768 /* Use command-line ADDRESS+PORT or just PORT */
773 disorder_fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]");
775 disorder_info("version "VERSION" process ID %lu",
776 (unsigned long)getpid());
777 struct sockaddr *addr;
779 if(!strcmp(sl.s[0], "-")) {
780 /* Syntax: - [[ADDRESS] PORT]. Here, the PORT may be `-' to get the local
781 * kernel to choose. The ADDRESS may be omitted or `-' to pick something
783 const char *node, *svc;
784 struct sockaddr *sa = 0;
786 #define NULLDASH(s) (strcmp((s), "-") ? (s) : 0)
787 case 1: node = 0; svc = 0; break;
788 case 2: node = 0; svc = NULLDASH(sl.s[1]); break;
789 case 3: node = NULLDASH(sl.s[1]); svc = NULLDASH(sl.s[2]); break;
790 default: disorder_fatal(0, "too many listening-address compoennts");
793 /* We'll need a connection to request the incoming stream, so open one if
794 * we don't have one already */
796 if(!(c = disorder_new(1))) exit(EXIT_FAILURE);
797 if(disorder_connect(c)) exit(EXIT_FAILURE);
799 /* If no address was given, we need to pick one. But we already have a
800 * connection to the server, so we can probably use the address from that.
802 struct sockaddr_storage ss;
804 addr_len = sizeof ss;
805 if(disorder_client_sockname(c, (struct sockaddr *)&ss, &addr_len))
807 if(ss.ss_family != AF_INET && ss.ss_family != AF_INET6) {
808 /* We're using a Unix-domain socket, so use a loopback address. I'm
809 * cowardly using IPv4 here. */
810 struct sockaddr_in *sin = (struct sockaddr_in *)&ss;
811 sin->sin_family = AF_INET;
812 sin->sin_addr.s_addr = htonl(INADDR_LOOPBACK);
814 sa = (struct sockaddr *)&ss;
815 prefs.ai_family = sa->sa_family;
817 /* If we have an address or port to resolve then do that now */
822 if((rc = getaddrinfo(node, svc, &prefs, &ai)))
823 disorder_fatal(0, "failed to resolve address `%s' and service `%s': %s",
824 node ? node : "-", svc ? svc : "-",
825 format_error(ec_getaddrinfo, rc,
826 errbuf, sizeof(errbuf)));
830 assert(sa->sa_family == ai->ai_addr->sa_family);
831 switch(sa->sa_family) {
833 ((struct sockaddr_in *)sa)->sin_port =
834 ((struct sockaddr_in *)ai->ai_addr)->sin_port;
837 ((struct sockaddr_in6 *)sa)->sin6_port =
838 ((struct sockaddr_in6 *)ai->ai_addr)->sin6_port;
841 assert(!"unexpected address family");
845 if((rtpfd = socket(sa->sa_family, SOCK_DGRAM, IPPROTO_UDP)) < 0)
846 disorder_fatal(errno, "error creating socket (family %d)",
848 /* Bind the address */
850 sa->sa_family == AF_INET
851 ? sizeof (struct sockaddr_in) : sizeof (struct sockaddr_in6)) < 0)
852 disorder_fatal(errno, "error binding socket");
853 static struct sockaddr_storage bound_address;
854 addr = (struct sockaddr *)&bound_address;
855 addr_len = sizeof bound_address;
856 if(getsockname(rtpfd, addr, &addr_len) < 0)
857 disorder_fatal(errno, "error getting socket address");
858 /* Convert to string */
859 char addrname[128], portname[32];
860 if(getnameinfo(addr, addr_len,
861 addrname, sizeof addrname,
862 portname, sizeof portname,
863 NI_NUMERICHOST|NI_NUMERICSERV) < 0)
864 disorder_fatal(errno, "getnameinfo");
865 /* Ask for audio data */
866 if(disorder_rtp_request(c, addrname, portname)) exit(EXIT_FAILURE);
867 /* Report what we did */
868 disorder_info("listening on %s (stream requested)",
869 format_sockaddr(addr));
871 if(sl.n > 2) disorder_fatal(0, "too many address components");
872 /* Look up address and port */
873 if(!(res = get_address(&sl, &prefs, &sockname)))
876 addr_len = res->ai_addrlen;
877 /* Create the socket */
878 if((rtpfd = socket(res->ai_family,
880 res->ai_protocol)) < 0)
881 disorder_fatal(errno, "error creating socket");
882 /* Allow multiple listeners */
883 xsetsockopt(rtpfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one);
884 is_multicast = multicast(addr);
885 /* The multicast and unicast/broadcast cases are different enough that they
886 * are totally split. Trying to find commonality between them causes more
887 * trouble that it's worth. */
889 /* Stash the multicast group address */
890 memcpy(&mgroup, addr, addr_len);
891 switch(res->ai_addr->sa_family) {
893 mgroup.in.sin_port = 0;
896 mgroup.in6.sin6_port = 0;
899 disorder_fatal(0, "unsupported address family %d",
900 (int)addr->sa_family);
902 /* Bind to to the multicast group address */
903 if(bind(rtpfd, addr, addr_len) < 0)
904 disorder_fatal(errno, "error binding socket to %s",
905 format_sockaddr(addr));
906 /* Add multicast group membership */
907 switch(mgroup.sa.sa_family) {
909 mreq.imr_multiaddr = mgroup.in.sin_addr;
910 mreq.imr_interface.s_addr = 0; /* use primary interface */
911 if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP,
912 &mreq, sizeof mreq) < 0)
913 disorder_fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP");
916 mreq6.ipv6mr_multiaddr = mgroup.in6.sin6_addr;
917 memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface);
918 if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP,
919 &mreq6, sizeof mreq6) < 0)
920 disorder_fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP");
923 disorder_fatal(0, "unsupported address family %d", res->ai_family);
925 /* Report what we did */
926 disorder_info("listening on %s multicast group %s",
927 format_sockaddr(addr), format_sockaddr(&mgroup.sa));
930 switch(addr->sa_family) {
932 struct sockaddr_in *in = (struct sockaddr_in *)addr;
934 memset(&in->sin_addr, 0, sizeof (struct in_addr));
938 struct sockaddr_in6 *in6 = (struct sockaddr_in6 *)addr;
940 memset(&in6->sin6_addr, 0, sizeof (struct in6_addr));
944 disorder_fatal(0, "unsupported family %d", (int)addr->sa_family);
946 if(bind(rtpfd, addr, addr_len) < 0)
947 disorder_fatal(errno, "error binding socket to %s",
948 format_sockaddr(addr));
949 /* Report what we did */
950 disorder_info("listening on %s", format_sockaddr(addr));
954 if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0)
955 disorder_fatal(errno, "error calling getsockopt SO_RCVBUF");
956 if(target_rcvbuf > rcvbuf) {
957 if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF,
958 &target_rcvbuf, sizeof target_rcvbuf) < 0)
959 disorder_error(errno, "error calling setsockopt SO_RCVBUF %d",
961 /* We try to carry on anyway */
963 disorder_info("changed socket receive buffer from %d to %d",
964 rcvbuf, target_rcvbuf);
966 disorder_info("default socket receive buffer %d", rcvbuf);
967 //info("minbuffer %u maxbuffer %u", minbuffer, maxbuffer);
969 disorder_info("WARNING: -L option can impact performance");
973 if((err = pthread_create(&tid, 0, control_thread, 0)))
974 disorder_fatal(err, "pthread_create control_thread");
978 unsigned char buffer[65536];
981 if((fd = open(dumpfile, O_RDWR|O_TRUNC|O_CREAT, 0666)) < 0)
982 disorder_fatal(errno, "opening %s", dumpfile);
983 /* Fill with 0s to a suitable size */
984 memset(buffer, 0, sizeof buffer);
985 for(written = 0; written < dump_size * sizeof(int16_t);
986 written += sizeof buffer) {
987 if(write(fd, buffer, sizeof buffer) < 0)
988 disorder_fatal(errno, "clearing %s", dumpfile);
990 /* Map the buffer into memory for convenience */
991 dump_buffer = mmap(0, dump_size * sizeof(int16_t), PROT_READ|PROT_WRITE,
993 if(dump_buffer == (void *)-1)
994 disorder_fatal(errno, "mapping %s", dumpfile);
995 disorder_info("dumping to %s", dumpfile);
997 /* Set up output. Currently we only support L16 so there's no harm setting
998 * the format before we know what it is! */
999 uaudio_set_format(44100/*Hz*/, 2/*channels*/,
1000 16/*bits/channel*/, 1/*signed*/);
1001 uaudio_set("application", "disorder-playrtp");
1002 backend->configure();
1003 backend->start(playrtp_callback, NULL);
1004 if(backend->open_mixer) backend->open_mixer();
1005 /* We receive and convert audio data in a background thread */
1006 if((err = pthread_create(<id, 0, listen_thread, 0)))
1007 disorder_fatal(err, "pthread_create listen_thread");
1008 /* We have a second thread to add received packets to the queue */
1009 if((err = pthread_create(<id, 0, queue_thread, 0)))
1010 disorder_fatal(err, "pthread_create queue_thread");
1011 pthread_mutex_lock(&lock);
1014 /* Wait for the buffer to fill up a bit */
1015 playrtp_fill_buffer();
1016 /* Start playing now */
1017 disorder_info("Playing...");
1018 next_timestamp = pheap_first(&packets)->timestamp;
1020 pthread_mutex_unlock(&lock);
1021 backend->activate();
1022 pthread_mutex_lock(&lock);
1023 /* Wait until the buffer empties out
1025 * If there's a packet that we can play right now then we definitely
1028 * Also if there's at least minbuffer samples we carry on regardless and
1029 * insert silence. The assumption is there's been a pause but more data
1032 while(nsamples >= minbuffer
1034 && contains(pheap_first(&packets), next_timestamp))) {
1036 time_t now = xtime(0);
1038 if(now >= lastlog + 60) {
1039 int offset = nsamples - minbuffer;
1040 double offtime = (double)offset / (uaudio_rate * uaudio_channels);
1041 disorder_info("%+d samples off (%d.%02ds, %d bytes)",
1043 (int)fabs(offtime) * (offtime < 0 ? -1 : 1),
1044 (int)(fabs(offtime) * 100) % 100,
1045 offset * uaudio_bits / CHAR_BIT);
1049 //fprintf(stderr, "%8u/%u (%u) PLAYING\n", nsamples, maxbuffer, minbuffer);
1050 pthread_cond_wait(&cond, &lock);
1054 struct packet *p = pheap_first(&packets);
1055 fprintf(stderr, "nsamples=%u (%u) next_timestamp=%"PRIx32", first packet is [%"PRIx32",%"PRIx32")\n",
1056 nsamples, minbuffer, next_timestamp,p->timestamp,p->timestamp+p->nsamples);
1059 /* Stop playing for a bit until the buffer re-fills */
1060 pthread_mutex_unlock(&lock);
1061 backend->deactivate();
1062 pthread_mutex_lock(&lock);
1074 indent-tabs-mode:nil