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
;
247 assert(control_socket
);
248 unlink(control_socket
);
249 memset(&sa
, 0, sizeof sa
);
250 sa
.sun_family
= AF_UNIX
;
251 strcpy(sa
.sun_path
, control_socket
);
252 sfd
= xsocket(PF_UNIX
, SOCK_STREAM
, 0);
253 if(bind(sfd
, (const struct sockaddr
*)&sa
, sizeof sa
) < 0)
254 disorder_fatal(errno
, "error binding to %s", control_socket
);
255 if(listen(sfd
, 128) < 0)
256 disorder_fatal(errno
, "error calling listen on %s", control_socket
);
257 disorder_info("listening on %s", control_socket
);
260 cfd
= accept(sfd
, (struct sockaddr
*)&sa
, &salen
);
267 disorder_fatal(errno
, "error calling accept on %s", control_socket
);
270 if(!(fp
= fdopen(cfd
, "r+"))) {
271 disorder_error(errno
, "error calling fdopen for %s connection", control_socket
);
275 if(!inputline(control_socket
, fp
, &line
, '\n')) {
276 if(!strcmp(line
, "stop")) {
277 disorder_info("stopped via %s", control_socket
);
278 exit(0); /* terminate immediately */
280 if(!strcmp(line
, "query"))
281 fprintf(fp
, "running");
285 disorder_error(errno
, "error closing %s connection", control_socket
);
289 /** @brief Drop the first packet
291 * Assumes that @ref lock is held.
293 static void drop_first_packet(void) {
294 if(pheap_count(&packets
)) {
295 struct packet
*const p
= pheap_remove(&packets
);
296 nsamples
-= p
->nsamples
;
297 playrtp_free_packet(p
);
298 pthread_cond_broadcast(&cond
);
302 /** @brief Background thread adding packets to heap
304 * This just transfers packets from @ref received_packets to @ref packets. It
305 * is important that it holds @ref receive_lock for as little time as possible,
306 * in order to minimize the interval between calls to read() in
309 static void *queue_thread(void attribute((unused
)) *arg
) {
313 /* Get the next packet */
314 pthread_mutex_lock(&receive_lock
);
315 while(!received_packets
) {
316 pthread_cond_wait(&receive_cond
, &receive_lock
);
318 p
= received_packets
;
319 received_packets
= p
->next
;
320 if(!received_packets
)
321 received_tail
= &received_packets
;
323 pthread_mutex_unlock(&receive_lock
);
324 /* Add it to the heap */
325 pthread_mutex_lock(&lock
);
326 pheap_insert(&packets
, p
);
327 nsamples
+= p
->nsamples
;
328 pthread_cond_broadcast(&cond
);
329 pthread_mutex_unlock(&lock
);
331 #if HAVE_STUPID_GCC44
336 /** @brief Background thread collecting samples
338 * This function collects samples, perhaps converts them to the target format,
339 * and adds them to the packet list.
341 * It is crucial that the gap between successive calls to read() is as small as
342 * possible: otherwise packets will be dropped.
344 * We use a binary heap to ensure that the unavoidable effort is at worst
345 * logarithmic in the total number of packets - in fact if packets are mostly
346 * received in order then we will largely do constant work per packet since the
347 * newest packet will always be last.
349 * Of more concern is that we must acquire the lock on the heap to add a packet
350 * to it. If this proves a problem in practice then the answer would be
351 * (probably doubly) linked list with new packets added the end and a second
352 * thread which reads packets off the list and adds them to the heap.
354 * We keep memory allocation (mostly) very fast by keeping pre-allocated
355 * packets around; see @ref playrtp_new_packet().
357 static void *listen_thread(void attribute((unused
)) *arg
) {
358 struct packet
*p
= 0;
360 struct rtp_header header
;
367 p
= playrtp_new_packet();
368 iov
[0].iov_base
= &header
;
369 iov
[0].iov_len
= sizeof header
;
370 iov
[1].iov_base
= p
->samples_raw
;
371 iov
[1].iov_len
= sizeof p
->samples_raw
/ sizeof *p
->samples_raw
;
372 n
= readv(rtpfd
, iov
, 2);
378 disorder_fatal(errno
, "error reading from socket");
381 /* Ignore too-short packets */
382 if((size_t)n
<= sizeof (struct rtp_header
)) {
383 disorder_info("ignored a short packet");
386 timestamp
= htonl(header
.timestamp
);
387 seq
= htons(header
.seq
);
388 /* Ignore packets in the past */
389 if(active
&& lt(timestamp
, next_timestamp
)) {
390 disorder_info("dropping old packet, timestamp=%"PRIx32
" < %"PRIx32
,
391 timestamp
, next_timestamp
);
394 /* Ignore packets with the extension bit set. */
395 if(header
.vpxcc
& 0x10)
399 p
->timestamp
= timestamp
;
400 /* Convert to target format */
401 if(header
.mpt
& 0x80)
403 switch(header
.mpt
& 0x7F) {
405 p
->nsamples
= (n
- sizeof header
) / sizeof(uint16_t);
407 /* TODO support other RFC3551 media types (when the speaker does) */
409 disorder_fatal(0, "unsupported RTP payload type %d", header
.mpt
& 0x7F);
411 /* See if packet is silent */
412 const uint16_t *s
= p
->samples_raw
;
420 fprintf(logfp
, "sequence %u timestamp %"PRIx32
" length %"PRIx32
" end %"PRIx32
"\n",
421 seq
, timestamp
, p
->nsamples
, timestamp
+ p
->nsamples
);
422 /* Stop reading if we've reached the maximum.
424 * This is rather unsatisfactory: it means that if packets get heavily
425 * out of order then we guarantee dropouts. But for now... */
426 if(nsamples
>= maxbuffer
) {
427 pthread_mutex_lock(&lock
);
428 while(nsamples
>= maxbuffer
) {
429 pthread_cond_wait(&cond
, &lock
);
431 pthread_mutex_unlock(&lock
);
433 /* Add the packet to the receive queue */
434 pthread_mutex_lock(&receive_lock
);
436 received_tail
= &p
->next
;
438 pthread_cond_signal(&receive_cond
);
439 pthread_mutex_unlock(&receive_lock
);
440 /* We'll need a new packet */
445 /** @brief Wait until the buffer is adequately full
447 * Must be called with @ref lock held.
449 void playrtp_fill_buffer(void) {
450 /* Discard current buffer contents */
452 //fprintf(stderr, "%8u/%u (%u) DROPPING\n", nsamples, maxbuffer, minbuffer);
455 disorder_info("Buffering...");
456 /* Wait until there's at least minbuffer samples available */
457 while(nsamples
< minbuffer
) {
458 //fprintf(stderr, "%8u/%u (%u) FILLING\n", nsamples, maxbuffer, minbuffer);
459 pthread_cond_wait(&cond
, &lock
);
461 /* Start from whatever is earliest */
462 next_timestamp
= pheap_first(&packets
)->timestamp
;
466 /** @brief Find next packet
467 * @return Packet to play or NULL if none found
469 * The return packet is merely guaranteed not to be in the past: it might be
470 * the first packet in the future rather than one that is actually suitable to
473 * Must be called with @ref lock held.
475 struct packet
*playrtp_next_packet(void) {
476 while(pheap_count(&packets
)) {
477 struct packet
*const p
= pheap_first(&packets
);
478 if(le(p
->timestamp
+ p
->nsamples
, next_timestamp
)) {
479 /* This packet is in the past. Drop it and try another one. */
482 /* This packet is NOT in the past. (It might be in the future
489 /* display usage message and terminate */
490 static void help(void) {
492 " disorder-playrtp [OPTIONS] [[ADDRESS] PORT]\n"
494 " --device, -D DEVICE Output device\n"
495 " --min, -m FRAMES Buffer low water mark\n"
496 " --max, -x FRAMES Buffer maximum size\n"
497 " --rcvbuf, -R BYTES Socket receive buffer size\n"
498 " --config, -C PATH Set configuration file\n"
499 " --api, -A API Select audio API. Possibilities:\n"
502 for(int n
= 0; uaudio_apis
[n
]; ++n
) {
503 if(uaudio_apis
[n
]->flags
& UAUDIO_API_CLIENT
) {
508 xprintf("%s", uaudio_apis
[n
]->name
);
512 " --command, -e COMMAND Pipe audio to command.\n"
513 " --pause-mode, -P silence For -e: pauses send silence (default)\n"
514 " --pause-mode, -P suspend For -e: pauses suspend writes\n"
515 " --help, -h Display usage message\n"
516 " --version, -V Display version number\n"
522 static size_t playrtp_callback(void *buffer
,
524 void attribute((unused
)) *userdata
) {
528 pthread_mutex_lock(&lock
);
529 /* Get the next packet, junking any that are now in the past */
530 const struct packet
*p
= playrtp_next_packet();
531 if(p
&& contains(p
, next_timestamp
)) {
532 /* This packet is ready to play; the desired next timestamp points
533 * somewhere into it. */
535 /* Timestamp of end of packet */
536 const uint32_t packet_end
= p
->timestamp
+ p
->nsamples
;
538 /* Offset of desired next timestamp into current packet */
539 const uint32_t offset
= next_timestamp
- p
->timestamp
;
541 /* Pointer to audio data */
542 const uint16_t *ptr
= (void *)(p
->samples_raw
+ offset
);
544 /* Compute number of samples left in packet, limited to output buffer
546 samples
= packet_end
- next_timestamp
;
547 if(samples
> max_samples
)
548 samples
= max_samples
;
550 /* Copy into buffer, converting to native endianness */
552 int16_t *bufptr
= buffer
;
554 *bufptr
++ = (int16_t)ntohs(*ptr
++);
557 silent
= !!(p
->flags
& SILENT
);
559 /* There is no suitable packet. We introduce 0s up to the next packet, or
560 * to fill the buffer if there's no next packet or that's too many. The
561 * comparison with max_samples deals with the otherwise troubling overflow
563 samples
= p ? p
->timestamp
- next_timestamp
: max_samples
;
564 if(samples
> max_samples
)
565 samples
= max_samples
;
566 //info("infill by %zu", samples);
567 memset(buffer
, 0, samples
* uaudio_sample_size
);
572 for(size_t i
= 0; i
< samples
; ++i
) {
573 dump_buffer
[dump_index
++] = ((int16_t *)buffer
)[i
];
574 dump_index
%= dump_size
;
577 /* Advance timestamp */
578 next_timestamp
+= samples
;
579 /* If we're getting behind then try to drop just silent packets
581 * In theory this shouldn't be necessary. The server is supposed to send
582 * packets at the right rate and compares the number of samples sent with the
583 * time in order to ensure this.
585 * However, various things could throw this off:
587 * - the server's clock could advance at the wrong rate. This would cause it
588 * to mis-estimate the right number of samples to have sent and
589 * inappropriately throttle or speed up.
591 * - playback could happen at the wrong rate. If the playback host's sound
592 * card has a slightly incorrect clock then eventually it will get out
595 * So if we play back slightly slower than the server sends for either of
596 * these reasons then eventually our buffer, and the socket's buffer, will
597 * fill, and the kernel will start dropping packets. The result is audible
600 * Therefore if we're getting behind, we pre-emptively drop silent packets,
601 * since a change in the duration of a silence is less noticeable than a
602 * dropped packet from the middle of continuous music.
604 * (If things go wrong the other way then eventually we run out of packets to
605 * play and are forced to play silence. This doesn't seem to happen in
606 * practice but if it does then in the same way we can artificially extend
607 * silent packets to compensate.)
609 * Dropped packets are always logged; use 'disorder-playrtp --monitor' to
610 * track how close to target buffer occupancy we are on a once-a-minute
613 if(nsamples
> minbuffer
&& silent
) {
614 disorder_info("dropping %zu samples (%"PRIu32
" > %"PRIu32
")",
615 samples
, nsamples
, minbuffer
);
618 /* Junk obsolete packets */
619 playrtp_next_packet();
620 pthread_mutex_unlock(&lock
);
624 static int compare_family(const struct ifaddrs
*a
,
625 const struct ifaddrs
*b
,
627 int afamily
= a
->ifa_addr
->sa_family
;
628 int bfamily
= b
->ifa_addr
->sa_family
;
629 if(afamily
!= bfamily
) {
630 /* Preferred family wins */
631 if(afamily
== family
) return 1;
632 if(bfamily
== family
) return -1;
633 /* Either there's no preference or it doesn't help. Prefer IPv4 */
634 if(afamily
== AF_INET
) return 1;
635 if(bfamily
== AF_INET
) return -1;
636 /* Failing that prefer IPv6 */
637 if(afamily
== AF_INET6
) return 1;
638 if(bfamily
== AF_INET6
) return -1;
643 static int compare_flags(const struct ifaddrs
*a
,
644 const struct ifaddrs
*b
) {
645 unsigned aflags
= a
->ifa_flags
, bflags
= b
->ifa_flags
;
646 /* Up interfaces are better than down ones */
647 unsigned aup
= aflags
& IFF_UP
, bup
= bflags
& IFF_UP
;
649 return aup
> bup ?
1 : -1;
650 /* Static addresses are better than dynamic */
651 unsigned adynamic
= aflags
& IFF_DYNAMIC
, bdynamic
= bflags
& IFF_DYNAMIC
;
652 if(adynamic
!= bdynamic
)
653 return adynamic
< bdynamic ?
1 : -1;
654 unsigned aloopback
= aflags
& IFF_LOOPBACK
, bloopback
= bflags
& IFF_LOOPBACK
;
655 /* Static addresses are better than dynamic */
656 if(aloopback
!= bloopback
)
657 return aloopback
< bloopback ?
1 : -1;
661 static int compare_interfaces(const struct ifaddrs
*a
,
662 const struct ifaddrs
*b
,
665 if((c
= compare_family(a
, b
, family
))) return c
;
666 return compare_flags(a
, b
);
669 int main(int argc
, char **argv
) {
671 struct addrinfo
*res
;
672 struct stringlist sl
;
674 int rcvbuf
, target_rcvbuf
= 0;
677 struct ipv6_mreq mreq6
;
678 disorder_client
*c
= NULL
;
679 char *address
, *port
;
683 struct sockaddr_in in
;
684 struct sockaddr_in6 in6
;
686 union any_sockaddr mgroup
;
687 const char *dumpfile
= 0;
690 static const int one
= 1;
692 static const struct addrinfo prefs
= {
693 .ai_flags
= AI_PASSIVE
,
694 .ai_family
= PF_INET
,
695 .ai_socktype
= SOCK_DGRAM
,
696 .ai_protocol
= IPPROTO_UDP
699 /* Timing information is often important to debugging playrtp, so we include
700 * timestamps in the logs */
703 if(!setlocale(LC_CTYPE
, "")) disorder_fatal(errno
, "error calling setlocale");
704 while((n
= getopt_long(argc
, argv
, "hVdD:m:x:L:R:aocC:re:P:MA:", options
, 0)) >= 0) {
707 case 'V': version("disorder-playrtp");
708 case 'd': debugging
= 1; break;
709 case 'D': uaudio_set("device", optarg
); break;
710 case 'm': minbuffer
= 2 * atol(optarg
); break;
711 case 'x': maxbuffer
= 2 * atol(optarg
); break;
712 case 'L': logfp
= fopen(optarg
, "w"); break;
713 case 'R': target_rcvbuf
= atoi(optarg
); break;
714 #if HAVE_ALSA_ASOUNDLIB_H
716 disorder_error(0, "deprecated option; use --api alsa instead");
717 backend
= &uaudio_alsa
; break;
719 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
721 disorder_error(0, "deprecated option; use --api oss instead");
722 backend
= &uaudio_oss
;
725 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
727 disorder_error(0, "deprecated option; use --api coreaudio instead");
728 backend
= &uaudio_coreaudio
;
731 case 'A': backend
= uaudio_find(optarg
); break;
732 case 'C': configfile
= optarg
; break;
733 case 's': control_socket
= optarg
; break;
734 case 'r': dumpfile
= optarg
; break;
735 case 'e': backend
= &uaudio_command
; uaudio_set("command", optarg
); break;
736 case 'P': uaudio_set("pause-mode", optarg
); break;
737 case 'M': monitor
= 1; break;
738 default: disorder_fatal(0, "invalid option");
741 if(config_read(0, NULL
)) disorder_fatal(0, "cannot read configuration");
743 backend
= uaudio_default(uaudio_apis
, UAUDIO_API_CLIENT
);
745 disorder_fatal(0, "no default uaudio API found");
746 disorder_info("default audio API %s", backend
->name
);
748 if(backend
== &uaudio_rtp
) {
749 /* This means that you have NO local sound output. This can happen if you
750 * use a non-Apple GCC on a Mac (because it doesn't know how to compile
751 * CoreAudio/AudioHardware.h). */
752 disorder_fatal(0, "cannot play RTP through RTP");
755 maxbuffer
= 2 * minbuffer
;
760 /* Get configuration from server */
761 if(!(c
= disorder_new(1))) exit(EXIT_FAILURE
);
762 if(disorder_connect(c
)) exit(EXIT_FAILURE
);
763 if(disorder_rtp_address(c
, &address
, &port
)) exit(EXIT_FAILURE
);
765 sl
.s
= xcalloc(2, sizeof *sl
.s
);
771 /* Use command-line ADDRESS+PORT or just PORT */
776 disorder_fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]");
778 disorder_info("version "VERSION
" process ID %lu",
779 (unsigned long)getpid());
780 struct sockaddr
*addr
;
782 if(!strcmp(sl
.s
[0], "-")) {
783 /* Pick address family to match known-working connectivity to the server */
784 int family
= disorder_client_af(c
);
785 /* Get a list of interfaces */
786 struct ifaddrs
*ifa
, *bestifa
= NULL
;
787 if(getifaddrs(&ifa
) < 0)
788 disorder_fatal(errno
, "error calling getifaddrs");
789 /* Try to pick a good one */
790 for(; ifa
; ifa
= ifa
->ifa_next
) {
792 || compare_interfaces(ifa
, bestifa
, family
) > 0)
796 disorder_fatal(0, "failed to select a network interface");
797 family
= bestifa
->ifa_addr
->sa_family
;
798 if((rtpfd
= socket(family
,
801 disorder_fatal(errno
, "error creating socket (family %d)", family
);
802 /* Bind the address */
803 if(bind(rtpfd
, bestifa
->ifa_addr
,
805 ?
sizeof (struct sockaddr_in
) : sizeof (struct sockaddr_in6
)) < 0)
806 disorder_fatal(errno
, "error binding socket");
807 static struct sockaddr_storage bound_address
;
808 addr
= (struct sockaddr
*)&bound_address
;
809 addr_len
= sizeof bound_address
;
810 if(getsockname(rtpfd
, addr
, &addr_len
) < 0)
811 disorder_fatal(errno
, "error getting socket address");
812 /* Convert to string */
813 char addrname
[128], portname
[32];
814 if(getnameinfo(addr
, addr_len
,
815 addrname
, sizeof addrname
,
816 portname
, sizeof portname
,
817 NI_NUMERICHOST
|NI_NUMERICSERV
) < 0)
818 disorder_fatal(errno
, "getnameinfo");
819 /* Ask for audio data */
820 if(disorder_rtp_request(c
, addrname
, portname
)) exit(EXIT_FAILURE
);
821 /* Report what we did */
822 disorder_info("listening on %s", format_sockaddr(addr
));
824 /* Look up address and port */
825 if(!(res
= get_address(&sl
, &prefs
, &sockname
)))
828 addr_len
= res
->ai_addrlen
;
829 /* Create the socket */
830 if((rtpfd
= socket(res
->ai_family
,
832 res
->ai_protocol
)) < 0)
833 disorder_fatal(errno
, "error creating socket");
834 /* Allow multiple listeners */
835 xsetsockopt(rtpfd
, SOL_SOCKET
, SO_REUSEADDR
, &one
, sizeof one
);
836 is_multicast
= multicast(addr
);
837 /* The multicast and unicast/broadcast cases are different enough that they
838 * are totally split. Trying to find commonality between them causes more
839 * trouble that it's worth. */
841 /* Stash the multicast group address */
842 memcpy(&mgroup
, addr
, addr_len
);
843 switch(res
->ai_addr
->sa_family
) {
845 mgroup
.in
.sin_port
= 0;
848 mgroup
.in6
.sin6_port
= 0;
851 disorder_fatal(0, "unsupported address family %d",
852 (int)addr
->sa_family
);
854 /* Bind to to the multicast group address */
855 if(bind(rtpfd
, addr
, addr_len
) < 0)
856 disorder_fatal(errno
, "error binding socket to %s",
857 format_sockaddr(addr
));
858 /* Add multicast group membership */
859 switch(mgroup
.sa
.sa_family
) {
861 mreq
.imr_multiaddr
= mgroup
.in
.sin_addr
;
862 mreq
.imr_interface
.s_addr
= 0; /* use primary interface */
863 if(setsockopt(rtpfd
, IPPROTO_IP
, IP_ADD_MEMBERSHIP
,
864 &mreq
, sizeof mreq
) < 0)
865 disorder_fatal(errno
, "error calling setsockopt IP_ADD_MEMBERSHIP");
868 mreq6
.ipv6mr_multiaddr
= mgroup
.in6
.sin6_addr
;
869 memset(&mreq6
.ipv6mr_interface
, 0, sizeof mreq6
.ipv6mr_interface
);
870 if(setsockopt(rtpfd
, IPPROTO_IPV6
, IPV6_JOIN_GROUP
,
871 &mreq6
, sizeof mreq6
) < 0)
872 disorder_fatal(errno
, "error calling setsockopt IPV6_JOIN_GROUP");
875 disorder_fatal(0, "unsupported address family %d", res
->ai_family
);
877 /* Report what we did */
878 disorder_info("listening on %s multicast group %s",
879 format_sockaddr(addr
), format_sockaddr(&mgroup
.sa
));
882 switch(addr
->sa_family
) {
884 struct sockaddr_in
*in
= (struct sockaddr_in
*)addr
;
886 memset(&in
->sin_addr
, 0, sizeof (struct in_addr
));
890 struct sockaddr_in6
*in6
= (struct sockaddr_in6
*)addr
;
892 memset(&in6
->sin6_addr
, 0, sizeof (struct in6_addr
));
896 disorder_fatal(0, "unsupported family %d", (int)addr
->sa_family
);
898 if(bind(rtpfd
, addr
, addr_len
) < 0)
899 disorder_fatal(errno
, "error binding socket to %s",
900 format_sockaddr(addr
));
901 /* Report what we did */
902 disorder_info("listening on %s", format_sockaddr(addr
));
906 if(getsockopt(rtpfd
, SOL_SOCKET
, SO_RCVBUF
, &rcvbuf
, &len
) < 0)
907 disorder_fatal(errno
, "error calling getsockopt SO_RCVBUF");
908 if(target_rcvbuf
> rcvbuf
) {
909 if(setsockopt(rtpfd
, SOL_SOCKET
, SO_RCVBUF
,
910 &target_rcvbuf
, sizeof target_rcvbuf
) < 0)
911 disorder_error(errno
, "error calling setsockopt SO_RCVBUF %d",
913 /* We try to carry on anyway */
915 disorder_info("changed socket receive buffer from %d to %d",
916 rcvbuf
, target_rcvbuf
);
918 disorder_info("default socket receive buffer %d", rcvbuf
);
919 //info("minbuffer %u maxbuffer %u", minbuffer, maxbuffer);
921 disorder_info("WARNING: -L option can impact performance");
925 if((err
= pthread_create(&tid
, 0, control_thread
, 0)))
926 disorder_fatal(err
, "pthread_create control_thread");
930 unsigned char buffer
[65536];
933 if((fd
= open(dumpfile
, O_RDWR
|O_TRUNC
|O_CREAT
, 0666)) < 0)
934 disorder_fatal(errno
, "opening %s", dumpfile
);
935 /* Fill with 0s to a suitable size */
936 memset(buffer
, 0, sizeof buffer
);
937 for(written
= 0; written
< dump_size
* sizeof(int16_t);
938 written
+= sizeof buffer
) {
939 if(write(fd
, buffer
, sizeof buffer
) < 0)
940 disorder_fatal(errno
, "clearing %s", dumpfile
);
942 /* Map the buffer into memory for convenience */
943 dump_buffer
= mmap(0, dump_size
* sizeof(int16_t), PROT_READ
|PROT_WRITE
,
945 if(dump_buffer
== (void *)-1)
946 disorder_fatal(errno
, "mapping %s", dumpfile
);
947 disorder_info("dumping to %s", dumpfile
);
949 /* Set up output. Currently we only support L16 so there's no harm setting
950 * the format before we know what it is! */
951 uaudio_set_format(44100/*Hz*/, 2/*channels*/,
952 16/*bits/channel*/, 1/*signed*/);
953 uaudio_set("application", "disorder-playrtp");
954 backend
->start(playrtp_callback
, NULL
);
955 /* We receive and convert audio data in a background thread */
956 if((err
= pthread_create(<id
, 0, listen_thread
, 0)))
957 disorder_fatal(err
, "pthread_create listen_thread");
958 /* We have a second thread to add received packets to the queue */
959 if((err
= pthread_create(<id
, 0, queue_thread
, 0)))
960 disorder_fatal(err
, "pthread_create queue_thread");
961 pthread_mutex_lock(&lock
);
964 /* Wait for the buffer to fill up a bit */
965 playrtp_fill_buffer();
966 /* Start playing now */
967 disorder_info("Playing...");
968 next_timestamp
= pheap_first(&packets
)->timestamp
;
970 pthread_mutex_unlock(&lock
);
972 pthread_mutex_lock(&lock
);
973 /* Wait until the buffer empties out
975 * If there's a packet that we can play right now then we definitely
978 * Also if there's at least minbuffer samples we carry on regardless and
979 * insert silence. The assumption is there's been a pause but more data
982 while(nsamples
>= minbuffer
984 && contains(pheap_first(&packets
), next_timestamp
))) {
986 time_t now
= xtime(0);
988 if(now
>= lastlog
+ 60) {
989 int offset
= nsamples
- minbuffer
;
990 double offtime
= (double)offset
/ (uaudio_rate
* uaudio_channels
);
991 disorder_info("%+d samples off (%d.%02ds, %d bytes)",
993 (int)fabs(offtime
) * (offtime
< 0 ?
-1 : 1),
994 (int)(fabs(offtime
) * 100) % 100,
995 offset
* uaudio_bits
/ CHAR_BIT
);
999 //fprintf(stderr, "%8u/%u (%u) PLAYING\n", nsamples, maxbuffer, minbuffer);
1000 pthread_cond_wait(&cond
, &lock
);
1004 struct packet
*p
= pheap_first(&packets
);
1005 fprintf(stderr
, "nsamples=%u (%u) next_timestamp=%"PRIx32
", first packet is [%"PRIx32
",%"PRIx32
")\n",
1006 nsamples
, minbuffer
, next_timestamp
,p
->timestamp
,p
->timestamp
+p
->nsamples
);
1009 /* Stop playing for a bit until the buffer re-fills */
1010 pthread_mutex_unlock(&lock
);
1011 backend
->deactivate();
1012 pthread_mutex_lock(&lock
);
1024 indent-tabs-mode:nil