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>
71 #include "configuration.h"
81 #include "inputline.h"
85 /** @brief Obsolete synonym */
86 #ifndef IPV6_JOIN_GROUP
87 # define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP
90 /** @brief RTP socket */
93 /** @brief Log output */
96 /** @brief Output device */
98 /** @brief Buffer low watermark in samples */
99 unsigned minbuffer
= 4 * (2 * 44100) / 10; /* 0.4 seconds */
101 /** @brief Maximum buffer size in samples
103 * We'll stop reading from the network if we have this many samples.
105 static unsigned maxbuffer
;
107 /** @brief Received packets
108 * Protected by @ref receive_lock
110 * Received packets are added to this list, and queue_thread() picks them off
111 * it and adds them to @ref packets. Whenever a packet is added to it, @ref
112 * receive_cond is signalled.
114 struct packet
*received_packets
;
116 /** @brief Tail of @ref received_packets
117 * Protected by @ref receive_lock
119 struct packet
**received_tail
= &received_packets
;
121 /** @brief Lock protecting @ref received_packets
123 * Only listen_thread() and queue_thread() ever hold this lock. It is vital
124 * that queue_thread() not hold it any longer than it strictly has to. */
125 pthread_mutex_t receive_lock
= PTHREAD_MUTEX_INITIALIZER
;
127 /** @brief Condition variable signalled when @ref received_packets is updated
129 * Used by listen_thread() to notify queue_thread() that it has added another
130 * packet to @ref received_packets. */
131 pthread_cond_t receive_cond
= PTHREAD_COND_INITIALIZER
;
133 /** @brief Length of @ref received_packets */
136 /** @brief Binary heap of received packets */
137 struct pheap packets
;
139 /** @brief Total number of samples available
141 * We make this volatile because we inspect it without a protecting lock,
142 * so the usual pthread_* guarantees aren't available.
144 volatile uint32_t nsamples
;
146 /** @brief Timestamp of next packet to play.
148 * This is set to the timestamp of the last packet, plus the number of
149 * samples it contained. Only valid if @ref active is nonzero.
151 uint32_t next_timestamp
;
153 /** @brief True if actively playing
155 * This is true when playing and false when just buffering. */
158 /** @brief Lock protecting @ref packets */
159 pthread_mutex_t lock
= PTHREAD_MUTEX_INITIALIZER
;
161 /** @brief Condition variable signalled whenever @ref packets is changed */
162 pthread_cond_t cond
= PTHREAD_COND_INITIALIZER
;
164 /** @brief Backend to play with */
165 static const struct uaudio
*backend
;
167 HEAP_DEFINE(pheap
, struct packet
*, lt_packet
);
169 /** @brief Control socket or NULL */
170 const char *control_socket
;
172 /** @brief Buffer for debugging dump
174 * The debug dump is enabled by the @c --dump option. It records the last 20s
175 * of audio to the specified file (which will be about 3.5Mbytes). The file is
176 * written as as ring buffer, so the start point will progress through it.
178 * Use clients/dump2wav to convert this to a WAV file, which can then be loaded
179 * into (e.g.) Audacity for further inspection.
181 * All three backends (ALSA, OSS, Core Audio) now support this option.
183 * The idea is to allow the user a few seconds to react to an audible artefact.
185 int16_t *dump_buffer
;
187 /** @brief Current index within debugging dump */
190 /** @brief Size of debugging dump in samples */
191 size_t dump_size
= 44100/*Hz*/ * 2/*channels*/ * 20/*seconds*/;
193 static const struct option options
[] = {
194 { "help", no_argument
, 0, 'h' },
195 { "version", no_argument
, 0, 'V' },
196 { "debug", no_argument
, 0, 'd' },
197 { "device", required_argument
, 0, 'D' },
198 { "min", required_argument
, 0, 'm' },
199 { "max", required_argument
, 0, 'x' },
200 { "rcvbuf", required_argument
, 0, 'R' },
201 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
202 { "oss", no_argument
, 0, 'o' },
204 #if HAVE_ALSA_ASOUNDLIB_H
205 { "alsa", no_argument
, 0, 'a' },
207 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
208 { "core-audio", no_argument
, 0, 'c' },
210 { "api", required_argument
, 0, 'A' },
211 { "dump", required_argument
, 0, 'r' },
212 { "command", required_argument
, 0, 'e' },
213 { "pause-mode", required_argument
, 0, 'P' },
214 { "socket", required_argument
, 0, 's' },
215 { "config", required_argument
, 0, 'C' },
216 { "monitor", no_argument
, 0, 'M' },
220 /** @brief Control thread
222 * This thread is responsible for accepting control commands from Disobedience
223 * (or other controllers) over an AF_UNIX stream socket with a path specified
224 * by the @c --socket option. The protocol uses simple string commands and
227 * - @c stop will shut the player down
228 * - @c query will send back the reply @c running
229 * - anything else is ignored
231 * Commands and response strings terminated by shutting down the connection or
232 * by a newline. No attempt is made to multiplex multiple clients so it is
233 * important that the command be sent as soon as the connection is made - it is
234 * assumed that both parties to the protocol are entirely cooperating with one
237 static void *control_thread(void attribute((unused
)) *arg
) {
238 struct sockaddr_un sa
;
244 assert(control_socket
);
245 unlink(control_socket
);
246 memset(&sa
, 0, sizeof sa
);
247 sa
.sun_family
= AF_UNIX
;
248 strcpy(sa
.sun_path
, control_socket
);
249 sfd
= xsocket(PF_UNIX
, SOCK_STREAM
, 0);
250 if(bind(sfd
, (const struct sockaddr
*)&sa
, sizeof sa
) < 0)
251 disorder_fatal(errno
, "error binding to %s", control_socket
);
252 if(listen(sfd
, 128) < 0)
253 disorder_fatal(errno
, "error calling listen on %s", control_socket
);
254 disorder_info("listening on %s", control_socket
);
257 cfd
= accept(sfd
, (struct sockaddr
*)&sa
, &salen
);
264 disorder_fatal(errno
, "error calling accept on %s", control_socket
);
267 if(!(fp
= fdopen(cfd
, "r+"))) {
268 disorder_error(errno
, "error calling fdopen for %s connection", control_socket
);
272 if(!inputline(control_socket
, fp
, &line
, '\n')) {
273 if(!strcmp(line
, "stop")) {
274 disorder_info("stopped via %s", control_socket
);
275 exit(0); /* terminate immediately */
277 if(!strcmp(line
, "query"))
278 fprintf(fp
, "running");
282 disorder_error(errno
, "error closing %s connection", control_socket
);
286 /** @brief Drop the first packet
288 * Assumes that @ref lock is held.
290 static void drop_first_packet(void) {
291 if(pheap_count(&packets
)) {
292 struct packet
*const p
= pheap_remove(&packets
);
293 nsamples
-= p
->nsamples
;
294 playrtp_free_packet(p
);
295 pthread_cond_broadcast(&cond
);
299 /** @brief Background thread adding packets to heap
301 * This just transfers packets from @ref received_packets to @ref packets. It
302 * is important that it holds @ref receive_lock for as little time as possible,
303 * in order to minimize the interval between calls to read() in
306 static void *queue_thread(void attribute((unused
)) *arg
) {
310 /* Get the next packet */
311 pthread_mutex_lock(&receive_lock
);
312 while(!received_packets
) {
313 pthread_cond_wait(&receive_cond
, &receive_lock
);
315 p
= received_packets
;
316 received_packets
= p
->next
;
317 if(!received_packets
)
318 received_tail
= &received_packets
;
320 pthread_mutex_unlock(&receive_lock
);
321 /* Add it to the heap */
322 pthread_mutex_lock(&lock
);
323 pheap_insert(&packets
, p
);
324 nsamples
+= p
->nsamples
;
325 pthread_cond_broadcast(&cond
);
326 pthread_mutex_unlock(&lock
);
328 #if HAVE_STUPID_GCC44
333 /** @brief Background thread collecting samples
335 * This function collects samples, perhaps converts them to the target format,
336 * and adds them to the packet list.
338 * It is crucial that the gap between successive calls to read() is as small as
339 * possible: otherwise packets will be dropped.
341 * We use a binary heap to ensure that the unavoidable effort is at worst
342 * logarithmic in the total number of packets - in fact if packets are mostly
343 * received in order then we will largely do constant work per packet since the
344 * newest packet will always be last.
346 * Of more concern is that we must acquire the lock on the heap to add a packet
347 * to it. If this proves a problem in practice then the answer would be
348 * (probably doubly) linked list with new packets added the end and a second
349 * thread which reads packets off the list and adds them to the heap.
351 * We keep memory allocation (mostly) very fast by keeping pre-allocated
352 * packets around; see @ref playrtp_new_packet().
354 static void *listen_thread(void attribute((unused
)) *arg
) {
355 struct packet
*p
= 0;
357 struct rtp_header header
;
364 p
= playrtp_new_packet();
365 iov
[0].iov_base
= &header
;
366 iov
[0].iov_len
= sizeof header
;
367 iov
[1].iov_base
= p
->samples_raw
;
368 iov
[1].iov_len
= sizeof p
->samples_raw
/ sizeof *p
->samples_raw
;
369 n
= readv(rtpfd
, iov
, 2);
375 disorder_fatal(errno
, "error reading from socket");
378 /* Ignore too-short packets */
379 if((size_t)n
<= sizeof (struct rtp_header
)) {
380 disorder_info("ignored a short packet");
383 timestamp
= htonl(header
.timestamp
);
384 seq
= htons(header
.seq
);
385 /* Ignore packets in the past */
386 if(active
&& lt(timestamp
, next_timestamp
)) {
387 disorder_info("dropping old packet, timestamp=%"PRIx32
" < %"PRIx32
,
388 timestamp
, next_timestamp
);
391 /* Ignore packets with the extension bit set. */
392 if(header
.vpxcc
& 0x10)
396 p
->timestamp
= timestamp
;
397 /* Convert to target format */
398 if(header
.mpt
& 0x80)
400 switch(header
.mpt
& 0x7F) {
402 p
->nsamples
= (n
- sizeof header
) / sizeof(uint16_t);
404 /* TODO support other RFC3551 media types (when the speaker does) */
406 disorder_fatal(0, "unsupported RTP payload type %d", header
.mpt
& 0x7F);
408 /* See if packet is silent */
409 const uint16_t *s
= p
->samples_raw
;
417 fprintf(logfp
, "sequence %u timestamp %"PRIx32
" length %"PRIx32
" end %"PRIx32
"\n",
418 seq
, timestamp
, p
->nsamples
, timestamp
+ p
->nsamples
);
419 /* Stop reading if we've reached the maximum.
421 * This is rather unsatisfactory: it means that if packets get heavily
422 * out of order then we guarantee dropouts. But for now... */
423 if(nsamples
>= maxbuffer
) {
424 pthread_mutex_lock(&lock
);
425 while(nsamples
>= maxbuffer
) {
426 pthread_cond_wait(&cond
, &lock
);
428 pthread_mutex_unlock(&lock
);
430 /* Add the packet to the receive queue */
431 pthread_mutex_lock(&receive_lock
);
433 received_tail
= &p
->next
;
435 pthread_cond_signal(&receive_cond
);
436 pthread_mutex_unlock(&receive_lock
);
437 /* We'll need a new packet */
442 /** @brief Wait until the buffer is adequately full
444 * Must be called with @ref lock held.
446 void playrtp_fill_buffer(void) {
447 /* Discard current buffer contents */
449 //fprintf(stderr, "%8u/%u (%u) DROPPING\n", nsamples, maxbuffer, minbuffer);
452 disorder_info("Buffering...");
453 /* Wait until there's at least minbuffer samples available */
454 while(nsamples
< minbuffer
) {
455 //fprintf(stderr, "%8u/%u (%u) FILLING\n", nsamples, maxbuffer, minbuffer);
456 pthread_cond_wait(&cond
, &lock
);
458 /* Start from whatever is earliest */
459 next_timestamp
= pheap_first(&packets
)->timestamp
;
463 /** @brief Find next packet
464 * @return Packet to play or NULL if none found
466 * The return packet is merely guaranteed not to be in the past: it might be
467 * the first packet in the future rather than one that is actually suitable to
470 * Must be called with @ref lock held.
472 struct packet
*playrtp_next_packet(void) {
473 while(pheap_count(&packets
)) {
474 struct packet
*const p
= pheap_first(&packets
);
475 if(le(p
->timestamp
+ p
->nsamples
, next_timestamp
)) {
476 /* This packet is in the past. Drop it and try another one. */
479 /* This packet is NOT in the past. (It might be in the future
486 /* display usage message and terminate */
487 static void help(void) {
489 " disorder-playrtp [OPTIONS] [[ADDRESS] PORT]\n"
491 " --device, -D DEVICE Output device\n"
492 " --min, -m FRAMES Buffer low water mark\n"
493 " --max, -x FRAMES Buffer maximum size\n"
494 " --rcvbuf, -R BYTES Socket receive buffer size\n"
495 " --config, -C PATH Set configuration file\n"
496 " --api, -A API Select audio API. Possibilities:\n"
499 for(int n
= 0; uaudio_apis
[n
]; ++n
) {
500 if(uaudio_apis
[n
]->flags
& UAUDIO_API_CLIENT
) {
505 xprintf("%s", uaudio_apis
[n
]->name
);
509 " --command, -e COMMAND Pipe audio to command.\n"
510 " --pause-mode, -P silence For -e: pauses send silence (default)\n"
511 " --pause-mode, -P suspend For -e: pauses suspend writes\n"
512 " --help, -h Display usage message\n"
513 " --version, -V Display version number\n"
519 static size_t playrtp_callback(void *buffer
,
521 void attribute((unused
)) *userdata
) {
525 pthread_mutex_lock(&lock
);
526 /* Get the next packet, junking any that are now in the past */
527 const struct packet
*p
= playrtp_next_packet();
528 if(p
&& contains(p
, next_timestamp
)) {
529 /* This packet is ready to play; the desired next timestamp points
530 * somewhere into it. */
532 /* Timestamp of end of packet */
533 const uint32_t packet_end
= p
->timestamp
+ p
->nsamples
;
535 /* Offset of desired next timestamp into current packet */
536 const uint32_t offset
= next_timestamp
- p
->timestamp
;
538 /* Pointer to audio data */
539 const uint16_t *ptr
= (void *)(p
->samples_raw
+ offset
);
541 /* Compute number of samples left in packet, limited to output buffer
543 samples
= packet_end
- next_timestamp
;
544 if(samples
> max_samples
)
545 samples
= max_samples
;
547 /* Copy into buffer, converting to native endianness */
549 int16_t *bufptr
= buffer
;
551 *bufptr
++ = (int16_t)ntohs(*ptr
++);
554 silent
= !!(p
->flags
& SILENT
);
556 /* There is no suitable packet. We introduce 0s up to the next packet, or
557 * to fill the buffer if there's no next packet or that's too many. The
558 * comparison with max_samples deals with the otherwise troubling overflow
560 samples
= p ? p
->timestamp
- next_timestamp
: max_samples
;
561 if(samples
> max_samples
)
562 samples
= max_samples
;
563 //info("infill by %zu", samples);
564 memset(buffer
, 0, samples
* uaudio_sample_size
);
569 for(size_t i
= 0; i
< samples
; ++i
) {
570 dump_buffer
[dump_index
++] = ((int16_t *)buffer
)[i
];
571 dump_index
%= dump_size
;
574 /* Advance timestamp */
575 next_timestamp
+= samples
;
576 /* If we're getting behind then try to drop just silent packets
578 * In theory this shouldn't be necessary. The server is supposed to send
579 * packets at the right rate and compares the number of samples sent with the
580 * time in order to ensure this.
582 * However, various things could throw this off:
584 * - the server's clock could advance at the wrong rate. This would cause it
585 * to mis-estimate the right number of samples to have sent and
586 * inappropriately throttle or speed up.
588 * - playback could happen at the wrong rate. If the playback host's sound
589 * card has a slightly incorrect clock then eventually it will get out
592 * So if we play back slightly slower than the server sends for either of
593 * these reasons then eventually our buffer, and the socket's buffer, will
594 * fill, and the kernel will start dropping packets. The result is audible
597 * Therefore if we're getting behind, we pre-emptively drop silent packets,
598 * since a change in the duration of a silence is less noticeable than a
599 * dropped packet from the middle of continuous music.
601 * (If things go wrong the other way then eventually we run out of packets to
602 * play and are forced to play silence. This doesn't seem to happen in
603 * practice but if it does then in the same way we can artificially extend
604 * silent packets to compensate.)
606 * Dropped packets are always logged; use 'disorder-playrtp --monitor' to
607 * track how close to target buffer occupancy we are on a once-a-minute
610 if(nsamples
> minbuffer
&& silent
) {
611 disorder_info("dropping %zu samples (%"PRIu32
" > %"PRIu32
")",
612 samples
, nsamples
, minbuffer
);
615 /* Junk obsolete packets */
616 playrtp_next_packet();
617 pthread_mutex_unlock(&lock
);
621 int main(int argc
, char **argv
) {
623 struct addrinfo
*res
;
624 struct stringlist sl
;
626 int rcvbuf
, target_rcvbuf
= 0;
629 struct ipv6_mreq mreq6
;
631 char *address
, *port
;
635 struct sockaddr_in in
;
636 struct sockaddr_in6 in6
;
638 union any_sockaddr mgroup
;
639 const char *dumpfile
= 0;
642 static const int one
= 1;
644 static const struct addrinfo prefs
= {
645 .ai_flags
= AI_PASSIVE
,
646 .ai_family
= PF_INET
,
647 .ai_socktype
= SOCK_DGRAM
,
648 .ai_protocol
= IPPROTO_UDP
651 /* Timing information is often important to debugging playrtp, so we include
652 * timestamps in the logs */
655 if(!setlocale(LC_CTYPE
, "")) disorder_fatal(errno
, "error calling setlocale");
656 while((n
= getopt_long(argc
, argv
, "hVdD:m:x:L:R:aocC:re:P:MA:", options
, 0)) >= 0) {
659 case 'V': version("disorder-playrtp");
660 case 'd': debugging
= 1; break;
661 case 'D': uaudio_set("device", optarg
); break;
662 case 'm': minbuffer
= 2 * atol(optarg
); break;
663 case 'x': maxbuffer
= 2 * atol(optarg
); break;
664 case 'L': logfp
= fopen(optarg
, "w"); break;
665 case 'R': target_rcvbuf
= atoi(optarg
); break;
666 #if HAVE_ALSA_ASOUNDLIB_H
668 disorder_error(0, "deprecated option; use --api alsa instead");
669 backend
= &uaudio_alsa
; break;
671 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
673 disorder_error(0, "deprecated option; use --api oss instead");
674 backend
= &uaudio_oss
;
677 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
679 disorder_error(0, "deprecated option; use --api coreaudio instead");
680 backend
= &uaudio_coreaudio
;
683 case 'A': backend
= uaudio_find(optarg
); break;
684 case 'C': configfile
= optarg
; break;
685 case 's': control_socket
= optarg
; break;
686 case 'r': dumpfile
= optarg
; break;
687 case 'e': backend
= &uaudio_command
; uaudio_set("command", optarg
); break;
688 case 'P': uaudio_set("pause-mode", optarg
); break;
689 case 'M': monitor
= 1; break;
690 default: disorder_fatal(0, "invalid option");
693 if(config_read(0, NULL
)) disorder_fatal(0, "cannot read configuration");
695 backend
= uaudio_default(uaudio_apis
, UAUDIO_API_CLIENT
);
697 disorder_fatal(0, "no default uaudio API found");
698 disorder_info("default audio API %s", backend
->name
);
700 if(backend
== &uaudio_rtp
) {
701 /* This means that you have NO local sound output. This can happen if you
702 * use a non-Apple GCC on a Mac (because it doesn't know how to compile
703 * CoreAudio/AudioHardware.h). */
704 disorder_fatal(0, "cannot play RTP through RTP");
707 maxbuffer
= 2 * minbuffer
;
712 /* Get configuration from server */
713 if(!(c
= disorder_new(1))) exit(EXIT_FAILURE
);
714 if(disorder_connect(c
)) exit(EXIT_FAILURE
);
715 if(disorder_rtp_address(c
, &address
, &port
)) exit(EXIT_FAILURE
);
717 sl
.s
= xcalloc(2, sizeof *sl
.s
);
723 /* Use command-line ADDRESS+PORT or just PORT */
728 disorder_fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]");
730 disorder_info("version "VERSION
" process ID %lu",
731 (unsigned long)getpid());
732 /* Look up address and port */
733 if(!(res
= get_address(&sl
, &prefs
, &sockname
)))
735 /* Create the socket */
736 if((rtpfd
= socket(res
->ai_family
,
738 res
->ai_protocol
)) < 0)
739 disorder_fatal(errno
, "error creating socket");
740 /* Allow multiple listeners */
741 xsetsockopt(rtpfd
, SOL_SOCKET
, SO_REUSEADDR
, &one
, sizeof one
);
742 is_multicast
= multicast(res
->ai_addr
);
743 /* The multicast and unicast/broadcast cases are different enough that they
744 * are totally split. Trying to find commonality between them causes more
745 * trouble that it's worth. */
747 /* Stash the multicast group address */
748 memcpy(&mgroup
, res
->ai_addr
, res
->ai_addrlen
);
749 switch(res
->ai_addr
->sa_family
) {
751 mgroup
.in
.sin_port
= 0;
754 mgroup
.in6
.sin6_port
= 0;
757 disorder_fatal(0, "unsupported address family %d",
758 (int)res
->ai_addr
->sa_family
);
760 /* Bind to to the multicast group address */
761 if(bind(rtpfd
, res
->ai_addr
, res
->ai_addrlen
) < 0)
762 disorder_fatal(errno
, "error binding socket to %s",
763 format_sockaddr(res
->ai_addr
));
764 /* Add multicast group membership */
765 switch(mgroup
.sa
.sa_family
) {
767 mreq
.imr_multiaddr
= mgroup
.in
.sin_addr
;
768 mreq
.imr_interface
.s_addr
= 0; /* use primary interface */
769 if(setsockopt(rtpfd
, IPPROTO_IP
, IP_ADD_MEMBERSHIP
,
770 &mreq
, sizeof mreq
) < 0)
771 disorder_fatal(errno
, "error calling setsockopt IP_ADD_MEMBERSHIP");
774 mreq6
.ipv6mr_multiaddr
= mgroup
.in6
.sin6_addr
;
775 memset(&mreq6
.ipv6mr_interface
, 0, sizeof mreq6
.ipv6mr_interface
);
776 if(setsockopt(rtpfd
, IPPROTO_IPV6
, IPV6_JOIN_GROUP
,
777 &mreq6
, sizeof mreq6
) < 0)
778 disorder_fatal(errno
, "error calling setsockopt IPV6_JOIN_GROUP");
781 disorder_fatal(0, "unsupported address family %d", res
->ai_family
);
783 /* Report what we did */
784 disorder_info("listening on %s multicast group %s",
785 format_sockaddr(res
->ai_addr
), format_sockaddr(&mgroup
.sa
));
788 switch(res
->ai_addr
->sa_family
) {
790 struct sockaddr_in
*in
= (struct sockaddr_in
*)res
->ai_addr
;
792 memset(&in
->sin_addr
, 0, sizeof (struct in_addr
));
796 struct sockaddr_in6
*in6
= (struct sockaddr_in6
*)res
->ai_addr
;
798 memset(&in6
->sin6_addr
, 0, sizeof (struct in6_addr
));
802 disorder_fatal(0, "unsupported family %d", (int)res
->ai_addr
->sa_family
);
804 if(bind(rtpfd
, res
->ai_addr
, res
->ai_addrlen
) < 0)
805 disorder_fatal(errno
, "error binding socket to %s",
806 format_sockaddr(res
->ai_addr
));
807 /* Report what we did */
808 disorder_info("listening on %s", format_sockaddr(res
->ai_addr
));
811 if(getsockopt(rtpfd
, SOL_SOCKET
, SO_RCVBUF
, &rcvbuf
, &len
) < 0)
812 disorder_fatal(errno
, "error calling getsockopt SO_RCVBUF");
813 if(target_rcvbuf
> rcvbuf
) {
814 if(setsockopt(rtpfd
, SOL_SOCKET
, SO_RCVBUF
,
815 &target_rcvbuf
, sizeof target_rcvbuf
) < 0)
816 disorder_error(errno
, "error calling setsockopt SO_RCVBUF %d",
818 /* We try to carry on anyway */
820 disorder_info("changed socket receive buffer from %d to %d",
821 rcvbuf
, target_rcvbuf
);
823 disorder_info("default socket receive buffer %d", rcvbuf
);
824 //info("minbuffer %u maxbuffer %u", minbuffer, maxbuffer);
826 disorder_info("WARNING: -L option can impact performance");
830 if((err
= pthread_create(&tid
, 0, control_thread
, 0)))
831 disorder_fatal(err
, "pthread_create control_thread");
835 unsigned char buffer
[65536];
838 if((fd
= open(dumpfile
, O_RDWR
|O_TRUNC
|O_CREAT
, 0666)) < 0)
839 disorder_fatal(errno
, "opening %s", dumpfile
);
840 /* Fill with 0s to a suitable size */
841 memset(buffer
, 0, sizeof buffer
);
842 for(written
= 0; written
< dump_size
* sizeof(int16_t);
843 written
+= sizeof buffer
) {
844 if(write(fd
, buffer
, sizeof buffer
) < 0)
845 disorder_fatal(errno
, "clearing %s", dumpfile
);
847 /* Map the buffer into memory for convenience */
848 dump_buffer
= mmap(0, dump_size
* sizeof(int16_t), PROT_READ
|PROT_WRITE
,
850 if(dump_buffer
== (void *)-1)
851 disorder_fatal(errno
, "mapping %s", dumpfile
);
852 disorder_info("dumping to %s", dumpfile
);
854 /* Set up output. Currently we only support L16 so there's no harm setting
855 * the format before we know what it is! */
856 uaudio_set_format(44100/*Hz*/, 2/*channels*/,
857 16/*bits/channel*/, 1/*signed*/);
858 backend
->start(playrtp_callback
, NULL
);
859 /* We receive and convert audio data in a background thread */
860 if((err
= pthread_create(<id
, 0, listen_thread
, 0)))
861 disorder_fatal(err
, "pthread_create listen_thread");
862 /* We have a second thread to add received packets to the queue */
863 if((err
= pthread_create(<id
, 0, queue_thread
, 0)))
864 disorder_fatal(err
, "pthread_create queue_thread");
865 pthread_mutex_lock(&lock
);
868 /* Wait for the buffer to fill up a bit */
869 playrtp_fill_buffer();
870 /* Start playing now */
871 disorder_info("Playing...");
872 next_timestamp
= pheap_first(&packets
)->timestamp
;
874 pthread_mutex_unlock(&lock
);
876 pthread_mutex_lock(&lock
);
877 /* Wait until the buffer empties out
879 * If there's a packet that we can play right now then we definitely
882 * Also if there's at least minbuffer samples we carry on regardless and
883 * insert silence. The assumption is there's been a pause but more data
886 while(nsamples
>= minbuffer
888 && contains(pheap_first(&packets
), next_timestamp
))) {
890 time_t now
= xtime(0);
892 if(now
>= lastlog
+ 60) {
893 int offset
= nsamples
- minbuffer
;
894 double offtime
= (double)offset
/ (uaudio_rate
* uaudio_channels
);
895 disorder_info("%+d samples off (%d.%02ds, %d bytes)",
897 (int)fabs(offtime
) * (offtime
< 0 ?
-1 : 1),
898 (int)(fabs(offtime
) * 100) % 100,
899 offset
* uaudio_bits
/ CHAR_BIT
);
903 //fprintf(stderr, "%8u/%u (%u) PLAYING\n", nsamples, maxbuffer, minbuffer);
904 pthread_cond_wait(&cond
, &lock
);
908 struct packet
*p
= pheap_first(&packets
);
909 fprintf(stderr
, "nsamples=%u (%u) next_timestamp=%"PRIx32
", first packet is [%"PRIx32
",%"PRIx32
")\n",
910 nsamples
, minbuffer
, next_timestamp
,p
->timestamp
,p
->timestamp
+p
->nsamples
);
913 /* Stop playing for a bit until the buffer re-fills */
914 pthread_mutex_unlock(&lock
);
915 backend
->deactivate();
916 pthread_mutex_lock(&lock
);