2 * This file is part of DisOrder.
3 * Copyright (C) 2007, 2008 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 2 of the License, or
8 * (at your option) any later version.
10 * This program is distributed in the hope that it will be useful, but
11 * WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * 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, write to the Free Software
17 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
20 /** @file clients/playrtp.c
23 * This player supports Linux (<a href="http://www.alsa-project.org/">ALSA</a>)
25 * href="http://developer.apple.com/audio/coreaudio.html">Core Audio</a>)
26 * systems. There is no support for Microsoft Windows yet, and that will in
27 * fact probably an entirely separate program.
29 * The program runs (at least) three threads. listen_thread() is responsible
30 * for reading RTP packets off the wire and adding them to the linked list @ref
31 * received_packets, assuming they are basically sound. queue_thread() takes
32 * packets off this linked list and adds them to @ref packets (an operation
33 * which might be much slower due to contention for @ref lock).
35 * The main thread is responsible for actually playing audio. In ALSA this
36 * means it waits until ALSA says it's ready for more audio which it then
37 * plays. See @ref clients/playrtp-alsa.c.
39 * In Core Audio the main thread is only responsible for starting and stopping
40 * play: the system does the actual playback in its own private thread, and
41 * calls adioproc() to fetch the audio data. See @ref
42 * clients/playrtp-coreaudio.c.
44 * Sometimes it happens that there is no audio available to play. This may
45 * because the server went away, or a packet was dropped, or the server
46 * deliberately did not send any sound because it encountered a silence.
49 * - it is safe to read uint32_t values without a lock protecting them
58 #include <sys/socket.h>
59 #include <sys/types.h>
60 #include <sys/socket.h>
68 #include <netinet/in.h>
77 #include "configuration.h"
87 #include "inputline.h"
90 #define readahead linux_headers_are_borked
92 /** @brief Obsolete synonym */
93 #ifndef IPV6_JOIN_GROUP
94 # define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP
97 /** @brief RTP socket */
100 /** @brief Log output */
103 /** @brief Output device */
106 /** @brief Minimum low watermark
108 * We'll stop playing if there's only this many samples in the buffer. */
109 unsigned minbuffer
= 2 * 44100 / 10; /* 0.2 seconds */
111 /** @brief Buffer high watermark
113 * We'll only start playing when this many samples are available. */
114 static unsigned readahead
= 2 * 2 * 44100;
116 /** @brief Maximum buffer size
118 * We'll stop reading from the network if we have this many samples. */
119 static unsigned maxbuffer
;
121 /** @brief Received packets
122 * Protected by @ref receive_lock
124 * Received packets are added to this list, and queue_thread() picks them off
125 * it and adds them to @ref packets. Whenever a packet is added to it, @ref
126 * receive_cond is signalled.
128 struct packet
*received_packets
;
130 /** @brief Tail of @ref received_packets
131 * Protected by @ref receive_lock
133 struct packet
**received_tail
= &received_packets
;
135 /** @brief Lock protecting @ref received_packets
137 * Only listen_thread() and queue_thread() ever hold this lock. It is vital
138 * that queue_thread() not hold it any longer than it strictly has to. */
139 pthread_mutex_t receive_lock
= PTHREAD_MUTEX_INITIALIZER
;
141 /** @brief Condition variable signalled when @ref received_packets is updated
143 * Used by listen_thread() to notify queue_thread() that it has added another
144 * packet to @ref received_packets. */
145 pthread_cond_t receive_cond
= PTHREAD_COND_INITIALIZER
;
147 /** @brief Length of @ref received_packets */
150 /** @brief Binary heap of received packets */
151 struct pheap packets
;
153 /** @brief Total number of samples available
155 * We make this volatile because we inspect it without a protecting lock,
156 * so the usual pthread_* guarantees aren't available.
158 volatile uint32_t nsamples
;
160 /** @brief Timestamp of next packet to play.
162 * This is set to the timestamp of the last packet, plus the number of
163 * samples it contained. Only valid if @ref active is nonzero.
165 uint32_t next_timestamp
;
167 /** @brief True if actively playing
169 * This is true when playing and false when just buffering. */
172 /** @brief Lock protecting @ref packets */
173 pthread_mutex_t lock
= PTHREAD_MUTEX_INITIALIZER
;
175 /** @brief Condition variable signalled whenever @ref packets is changed */
176 pthread_cond_t cond
= PTHREAD_COND_INITIALIZER
;
178 #if DEFAULT_BACKEND == BACKEND_ALSA
179 # define DEFAULT_PLAYRTP_BACKEND playrtp_alsa
180 #elif DEFAULT_BACKEND == BACKEND_OSS
181 # define DEFAULT_PLAYRTP_BACKEND playrtp_oss
182 #elif DEFAULT_BACKEND == BACKEND_COREAUDIO
183 # define DEFAULT_PLAYRTP_BACKEND playrtp_coreaudio
186 /** @brief Backend to play with */
187 static void (*backend
)(void) = DEFAULT_PLAYRTP_BACKEND
;
189 HEAP_DEFINE(pheap
, struct packet
*, lt_packet
);
191 /** @brief Control socket or NULL */
192 const char *control_socket
;
194 /** @brief Buffer for debugging dump
196 * The debug dump is enabled by the @c --dump option. It records the last 20s
197 * of audio to the specified file (which will be about 3.5Mbytes). The file is
198 * written as as ring buffer, so the start point will progress through it.
200 * Use clients/dump2wav to convert this to a WAV file, which can then be loaded
201 * into (e.g.) Audacity for further inspection.
203 * All three backends (ALSA, OSS, Core Audio) now support this option.
205 * The idea is to allow the user a few seconds to react to an audible artefact.
207 int16_t *dump_buffer
;
209 /** @brief Current index within debugging dump */
212 /** @brief Size of debugging dump in samples */
213 size_t dump_size
= 44100/*Hz*/ * 2/*channels*/ * 20/*seconds*/;
215 static const struct option options
[] = {
216 { "help", no_argument
, 0, 'h' },
217 { "version", no_argument
, 0, 'V' },
218 { "debug", no_argument
, 0, 'd' },
219 { "device", required_argument
, 0, 'D' },
220 { "min", required_argument
, 0, 'm' },
221 { "max", required_argument
, 0, 'x' },
222 { "buffer", required_argument
, 0, 'b' },
223 { "rcvbuf", required_argument
, 0, 'R' },
224 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
225 { "oss", no_argument
, 0, 'o' },
227 #if HAVE_ALSA_ASOUNDLIB_H
228 { "alsa", no_argument
, 0, 'a' },
230 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
231 { "core-audio", no_argument
, 0, 'c' },
233 { "dump", required_argument
, 0, 'r' },
234 { "socket", required_argument
, 0, 's' },
235 { "config", required_argument
, 0, 'C' },
239 /** @brief Control thread
241 * This thread is responsible for accepting control commands from Disobedience
242 * (or other controllers) over an AF_UNIX stream socket with a path specified
243 * by the @c --socket option. The protocol uses simple string commands and
246 * - @c stop will shut the player down
247 * - @c query will send back the reply @c running
248 * - anything else is ignored
250 * Commands and response strings terminated by shutting down the connection or
251 * by a newline. No attempt is made to multiplex multiple clients so it is
252 * important that the command be sent as soon as the connection is made - it is
253 * assumed that both parties to the protocol are entirely cooperating with one
256 static void *control_thread(void attribute((unused
)) *arg
) {
257 struct sockaddr_un sa
;
263 assert(control_socket
);
264 unlink(control_socket
);
265 memset(&sa
, 0, sizeof sa
);
266 sa
.sun_family
= AF_UNIX
;
267 strcpy(sa
.sun_path
, control_socket
);
268 sfd
= xsocket(PF_UNIX
, SOCK_STREAM
, 0);
269 if(bind(sfd
, (const struct sockaddr
*)&sa
, sizeof sa
) < 0)
270 fatal(errno
, "error binding to %s", control_socket
);
271 if(listen(sfd
, 128) < 0)
272 fatal(errno
, "error calling listen on %s", control_socket
);
273 info("listening on %s", control_socket
);
276 cfd
= accept(sfd
, (struct sockaddr
*)&sa
, &salen
);
283 fatal(errno
, "error calling accept on %s", control_socket
);
286 if(!(fp
= fdopen(cfd
, "r+"))) {
287 error(errno
, "error calling fdopen for %s connection", control_socket
);
291 if(!inputline(control_socket
, fp
, &line
, '\n')) {
292 if(!strcmp(line
, "stop")) {
293 info("stopped via %s", control_socket
);
294 exit(0); /* terminate immediately */
296 if(!strcmp(line
, "query"))
297 fprintf(fp
, "running");
301 error(errno
, "error closing %s connection", control_socket
);
305 /** @brief Drop the first packet
307 * Assumes that @ref lock is held.
309 static void drop_first_packet(void) {
310 if(pheap_count(&packets
)) {
311 struct packet
*const p
= pheap_remove(&packets
);
312 nsamples
-= p
->nsamples
;
313 playrtp_free_packet(p
);
314 pthread_cond_broadcast(&cond
);
318 /** @brief Background thread adding packets to heap
320 * This just transfers packets from @ref received_packets to @ref packets. It
321 * is important that it holds @ref receive_lock for as little time as possible,
322 * in order to minimize the interval between calls to read() in
325 static void *queue_thread(void attribute((unused
)) *arg
) {
329 /* Get the next packet */
330 pthread_mutex_lock(&receive_lock
);
331 while(!received_packets
) {
332 pthread_cond_wait(&receive_cond
, &receive_lock
);
334 p
= received_packets
;
335 received_packets
= p
->next
;
336 if(!received_packets
)
337 received_tail
= &received_packets
;
339 pthread_mutex_unlock(&receive_lock
);
340 /* Add it to the heap */
341 pthread_mutex_lock(&lock
);
342 pheap_insert(&packets
, p
);
343 nsamples
+= p
->nsamples
;
344 pthread_cond_broadcast(&cond
);
345 pthread_mutex_unlock(&lock
);
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 fatal(errno
, "error reading from socket");
394 /* Ignore too-short packets */
395 if((size_t)n
<= sizeof (struct rtp_header
)) {
396 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 info("dropping old packet, timestamp=%"PRIx32
" < %"PRIx32
,
404 timestamp
, next_timestamp
);
409 p
->timestamp
= timestamp
;
410 /* Convert to target format */
411 if(header
.mpt
& 0x80)
413 switch(header
.mpt
& 0x7F) {
415 p
->nsamples
= (n
- sizeof header
) / sizeof(uint16_t);
417 /* TODO support other RFC3551 media types (when the speaker does) */
419 fatal(0, "unsupported RTP payload type %d",
423 fprintf(logfp
, "sequence %u timestamp %"PRIx32
" length %"PRIx32
" end %"PRIx32
"\n",
424 seq
, timestamp
, p
->nsamples
, timestamp
+ p
->nsamples
);
425 /* Stop reading if we've reached the maximum.
427 * This is rather unsatisfactory: it means that if packets get heavily
428 * out of order then we guarantee dropouts. But for now... */
429 if(nsamples
>= maxbuffer
) {
430 pthread_mutex_lock(&lock
);
431 while(nsamples
>= maxbuffer
) {
432 pthread_cond_wait(&cond
, &lock
);
434 pthread_mutex_unlock(&lock
);
436 /* Add the packet to the receive queue */
437 pthread_mutex_lock(&receive_lock
);
439 received_tail
= &p
->next
;
441 pthread_cond_signal(&receive_cond
);
442 pthread_mutex_unlock(&receive_lock
);
443 /* We'll need a new packet */
448 /** @brief Wait until the buffer is adequately full
450 * Must be called with @ref lock held.
452 void playrtp_fill_buffer(void) {
455 info("Buffering...");
456 while(nsamples
< readahead
) {
457 pthread_cond_wait(&cond
, &lock
);
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 /** @brief Play an RTP stream
488 * This is the guts of the program. It is responsible for:
489 * - starting the listening thread
490 * - opening the audio device
491 * - reading ahead to build up a buffer
492 * - arranging for audio to be played
493 * - detecting when the buffer has got too small and re-buffering
495 static void play_rtp(void) {
499 /* We receive and convert audio data in a background thread */
500 if((err
= pthread_create(<id
, 0, listen_thread
, 0)))
501 fatal(err
, "pthread_create listen_thread");
502 /* We have a second thread to add received packets to the queue */
503 if((err
= pthread_create(<id
, 0, queue_thread
, 0)))
504 fatal(err
, "pthread_create queue_thread");
505 /* The rest of the work is backend-specific */
509 /* display usage message and terminate */
510 static void help(void) {
512 " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
514 " --device, -D DEVICE Output device\n"
515 " --min, -m FRAMES Buffer low water mark\n"
516 " --buffer, -b FRAMES Buffer high water mark\n"
517 " --max, -x FRAMES Buffer maximum size\n"
518 " --rcvbuf, -R BYTES Socket receive buffer size\n"
519 " --config, -C PATH Set configuration file\n"
520 #if HAVE_ALSA_ASOUNDLIB_H
521 " --alsa, -a Use ALSA to play audio\n"
523 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
524 " --oss, -o Use OSS to play audio\n"
526 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
527 " --core-audio, -c Use Core Audio to play audio\n"
529 " --help, -h Display usage message\n"
530 " --version, -V Display version number\n"
536 int main(int argc
, char **argv
) {
538 struct addrinfo
*res
;
539 struct stringlist sl
;
541 int rcvbuf
, target_rcvbuf
= 131072;
544 struct ipv6_mreq mreq6
;
546 char *address
, *port
;
550 struct sockaddr_in in
;
551 struct sockaddr_in6 in6
;
553 union any_sockaddr mgroup
;
554 const char *dumpfile
= 0;
556 static const struct addrinfo prefs
= {
557 .ai_flags
= AI_PASSIVE
,
558 .ai_family
= PF_INET
,
559 .ai_socktype
= SOCK_DGRAM
,
560 .ai_protocol
= IPPROTO_UDP
564 if(!setlocale(LC_CTYPE
, "")) fatal(errno
, "error calling setlocale");
565 while((n
= getopt_long(argc
, argv
, "hVdD:m:b:x:L:R:M:aocC:r", options
, 0)) >= 0) {
568 case 'V': version("disorder-playrtp");
569 case 'd': debugging
= 1; break;
570 case 'D': device
= optarg
; break;
571 case 'm': minbuffer
= 2 * atol(optarg
); break;
572 case 'b': readahead
= 2 * atol(optarg
); break;
573 case 'x': maxbuffer
= 2 * atol(optarg
); break;
574 case 'L': logfp
= fopen(optarg
, "w"); break;
575 case 'R': target_rcvbuf
= atoi(optarg
); break;
576 #if HAVE_ALSA_ASOUNDLIB_H
577 case 'a': backend
= playrtp_alsa
; break;
579 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
580 case 'o': backend
= playrtp_oss
; break;
582 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
583 case 'c': backend
= playrtp_coreaudio
; break;
585 case 'C': configfile
= optarg
; break;
586 case 's': control_socket
= optarg
; break;
587 case 'r': dumpfile
= optarg
; break;
588 default: fatal(0, "invalid option");
591 if(config_read(0)) fatal(0, "cannot read configuration");
593 maxbuffer
= 4 * readahead
;
598 /* Get configuration from server */
599 if(!(c
= disorder_new(1))) exit(EXIT_FAILURE
);
600 if(disorder_connect(c
)) exit(EXIT_FAILURE
);
601 if(disorder_rtp_address(c
, &address
, &port
)) exit(EXIT_FAILURE
);
603 sl
.s
= xcalloc(2, sizeof *sl
.s
);
609 /* Use command-line ADDRESS+PORT or just PORT */
614 fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]");
616 /* Look up address and port */
617 if(!(res
= get_address(&sl
, &prefs
, &sockname
)))
619 /* Create the socket */
620 if((rtpfd
= socket(res
->ai_family
,
622 res
->ai_protocol
)) < 0)
623 fatal(errno
, "error creating socket");
624 /* Stash the multicast group address */
625 if((is_multicast
= multicast(res
->ai_addr
))) {
626 memcpy(&mgroup
, res
->ai_addr
, res
->ai_addrlen
);
627 switch(res
->ai_addr
->sa_family
) {
629 mgroup
.in
.sin_port
= 0;
632 mgroup
.in6
.sin6_port
= 0;
637 switch(res
->ai_addr
->sa_family
) {
639 memset(&((struct sockaddr_in
*)res
->ai_addr
)->sin_addr
, 0,
640 sizeof (struct in_addr
));
643 memset(&((struct sockaddr_in6
*)res
->ai_addr
)->sin6_addr
, 0,
644 sizeof (struct in6_addr
));
647 fatal(0, "unsupported family %d", (int)res
->ai_addr
->sa_family
);
649 if(bind(rtpfd
, res
->ai_addr
, res
->ai_addrlen
) < 0)
650 fatal(errno
, "error binding socket to %s", sockname
);
652 switch(mgroup
.sa
.sa_family
) {
654 mreq
.imr_multiaddr
= mgroup
.in
.sin_addr
;
655 mreq
.imr_interface
.s_addr
= 0; /* use primary interface */
656 if(setsockopt(rtpfd
, IPPROTO_IP
, IP_ADD_MEMBERSHIP
,
657 &mreq
, sizeof mreq
) < 0)
658 fatal(errno
, "error calling setsockopt IP_ADD_MEMBERSHIP");
661 mreq6
.ipv6mr_multiaddr
= mgroup
.in6
.sin6_addr
;
662 memset(&mreq6
.ipv6mr_interface
, 0, sizeof mreq6
.ipv6mr_interface
);
663 if(setsockopt(rtpfd
, IPPROTO_IPV6
, IPV6_JOIN_GROUP
,
664 &mreq6
, sizeof mreq6
) < 0)
665 fatal(errno
, "error calling setsockopt IPV6_JOIN_GROUP");
668 fatal(0, "unsupported address family %d", res
->ai_family
);
670 info("listening on %s multicast group %s",
671 format_sockaddr(res
->ai_addr
), format_sockaddr(&mgroup
.sa
));
673 info("listening on %s", format_sockaddr(res
->ai_addr
));
675 if(getsockopt(rtpfd
, SOL_SOCKET
, SO_RCVBUF
, &rcvbuf
, &len
) < 0)
676 fatal(errno
, "error calling getsockopt SO_RCVBUF");
677 if(target_rcvbuf
> rcvbuf
) {
678 if(setsockopt(rtpfd
, SOL_SOCKET
, SO_RCVBUF
,
679 &target_rcvbuf
, sizeof target_rcvbuf
) < 0)
680 error(errno
, "error calling setsockopt SO_RCVBUF %d",
682 /* We try to carry on anyway */
684 info("changed socket receive buffer from %d to %d",
685 rcvbuf
, target_rcvbuf
);
687 info("default socket receive buffer %d", rcvbuf
);
689 info("WARNING: -L option can impact performance");
693 if((err
= pthread_create(&tid
, 0, control_thread
, 0)))
694 fatal(err
, "pthread_create control_thread");
698 unsigned char buffer
[65536];
701 if((fd
= open(dumpfile
, O_RDWR
|O_TRUNC
|O_CREAT
, 0666)) < 0)
702 fatal(errno
, "opening %s", dumpfile
);
703 /* Fill with 0s to a suitable size */
704 memset(buffer
, 0, sizeof buffer
);
705 for(written
= 0; written
< dump_size
* sizeof(int16_t);
706 written
+= sizeof buffer
) {
707 if(write(fd
, buffer
, sizeof buffer
) < 0)
708 fatal(errno
, "clearing %s", dumpfile
);
710 /* Map the buffer into memory for convenience */
711 dump_buffer
= mmap(0, dump_size
* sizeof(int16_t), PROT_READ
|PROT_WRITE
,
713 if(dump_buffer
== (void *)-1)
714 fatal(errno
, "mapping %s", dumpfile
);
715 info("dumping to %s", dumpfile
);