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 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. listen_thread() is responsible
28 * for reading RTP packets off the wire and adding them to the linked list @ref
29 * received_packets, assuming they are basically sound. queue_thread() takes
30 * packets off this linked list and adds them to @ref packets (an operation
31 * which might be much slower due to contention for @ref lock).
33 * The main thread is responsible for actually playing audio. In ALSA this
34 * means it waits until ALSA says it's ready for more audio which it then
35 * plays. See @ref clients/playrtp-alsa.c.
37 * In Core Audio the main thread is only responsible for starting and stopping
38 * play: the system does the actual playback in its own private thread, and
39 * calls adioproc() to fetch the audio data. See @ref
40 * clients/playrtp-coreaudio.c.
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>
70 #include "configuration.h"
80 #include "inputline.h"
83 #define readahead linux_headers_are_borked
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 */
99 /** @brief Minimum low watermark
101 * We'll stop playing if there's only this many samples in the buffer. */
102 unsigned minbuffer
= 2 * 44100 / 10; /* 0.2 seconds */
104 /** @brief Buffer high watermark
106 * We'll only start playing when this many samples are available. */
107 static unsigned readahead
= 2 * 2 * 44100;
109 /** @brief Maximum buffer size
111 * We'll stop reading from the network if we have this many samples. */
112 static unsigned maxbuffer
;
114 /** @brief Received packets
115 * Protected by @ref receive_lock
117 * Received packets are added to this list, and queue_thread() picks them off
118 * it and adds them to @ref packets. Whenever a packet is added to it, @ref
119 * receive_cond is signalled.
121 struct packet
*received_packets
;
123 /** @brief Tail of @ref received_packets
124 * Protected by @ref receive_lock
126 struct packet
**received_tail
= &received_packets
;
128 /** @brief Lock protecting @ref received_packets
130 * Only listen_thread() and queue_thread() ever hold this lock. It is vital
131 * that queue_thread() not hold it any longer than it strictly has to. */
132 pthread_mutex_t receive_lock
= PTHREAD_MUTEX_INITIALIZER
;
134 /** @brief Condition variable signalled when @ref received_packets is updated
136 * Used by listen_thread() to notify queue_thread() that it has added another
137 * packet to @ref received_packets. */
138 pthread_cond_t receive_cond
= PTHREAD_COND_INITIALIZER
;
140 /** @brief Length of @ref received_packets */
143 /** @brief Binary heap of received packets */
144 struct pheap packets
;
146 /** @brief Total number of samples available
148 * We make this volatile because we inspect it without a protecting lock,
149 * so the usual pthread_* guarantees aren't available.
151 volatile uint32_t nsamples
;
153 /** @brief Timestamp of next packet to play.
155 * This is set to the timestamp of the last packet, plus the number of
156 * samples it contained. Only valid if @ref active is nonzero.
158 uint32_t next_timestamp
;
160 /** @brief True if actively playing
162 * This is true when playing and false when just buffering. */
165 /** @brief Lock protecting @ref packets */
166 pthread_mutex_t lock
= PTHREAD_MUTEX_INITIALIZER
;
168 /** @brief Condition variable signalled whenever @ref packets is changed */
169 pthread_cond_t cond
= PTHREAD_COND_INITIALIZER
;
171 #if DEFAULT_BACKEND == BACKEND_ALSA
172 # define DEFAULT_PLAYRTP_BACKEND playrtp_alsa
173 #elif DEFAULT_BACKEND == BACKEND_OSS
174 # define DEFAULT_PLAYRTP_BACKEND playrtp_oss
175 #elif DEFAULT_BACKEND == BACKEND_COREAUDIO
176 # define DEFAULT_PLAYRTP_BACKEND playrtp_coreaudio
179 /** @brief Backend to play with */
180 static void (*backend
)(void) = DEFAULT_PLAYRTP_BACKEND
;
182 HEAP_DEFINE(pheap
, struct packet
*, lt_packet
);
184 /** @brief Control socket or NULL */
185 const char *control_socket
;
187 /** @brief Buffer for debugging dump
189 * The debug dump is enabled by the @c --dump option. It records the last 20s
190 * of audio to the specified file (which will be about 3.5Mbytes). The file is
191 * written as as ring buffer, so the start point will progress through it.
193 * Use clients/dump2wav to convert this to a WAV file, which can then be loaded
194 * into (e.g.) Audacity for further inspection.
196 * All three backends (ALSA, OSS, Core Audio) now support this option.
198 * The idea is to allow the user a few seconds to react to an audible artefact.
200 int16_t *dump_buffer
;
202 /** @brief Current index within debugging dump */
205 /** @brief Size of debugging dump in samples */
206 size_t dump_size
= 44100/*Hz*/ * 2/*channels*/ * 20/*seconds*/;
208 static const struct option options
[] = {
209 { "help", no_argument
, 0, 'h' },
210 { "version", no_argument
, 0, 'V' },
211 { "debug", no_argument
, 0, 'd' },
212 { "device", required_argument
, 0, 'D' },
213 { "min", required_argument
, 0, 'm' },
214 { "max", required_argument
, 0, 'x' },
215 { "buffer", required_argument
, 0, 'b' },
216 { "rcvbuf", required_argument
, 0, 'R' },
217 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
218 { "oss", no_argument
, 0, 'o' },
220 #if HAVE_ALSA_ASOUNDLIB_H
221 { "alsa", no_argument
, 0, 'a' },
223 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
224 { "core-audio", no_argument
, 0, 'c' },
226 { "dump", required_argument
, 0, 'r' },
227 { "socket", required_argument
, 0, 's' },
228 { "config", required_argument
, 0, 'C' },
232 /** @brief Control thread
234 * This thread is responsible for accepting control commands from Disobedience
235 * (or other controllers) over an AF_UNIX stream socket with a path specified
236 * by the @c --socket option. The protocol uses simple string commands and
239 * - @c stop will shut the player down
240 * - @c query will send back the reply @c running
241 * - anything else is ignored
243 * Commands and response strings terminated by shutting down the connection or
244 * by a newline. No attempt is made to multiplex multiple clients so it is
245 * important that the command be sent as soon as the connection is made - it is
246 * assumed that both parties to the protocol are entirely cooperating with one
249 static void *control_thread(void attribute((unused
)) *arg
) {
250 struct sockaddr_un sa
;
256 assert(control_socket
);
257 unlink(control_socket
);
258 memset(&sa
, 0, sizeof sa
);
259 sa
.sun_family
= AF_UNIX
;
260 strcpy(sa
.sun_path
, control_socket
);
261 sfd
= xsocket(PF_UNIX
, SOCK_STREAM
, 0);
262 if(bind(sfd
, (const struct sockaddr
*)&sa
, sizeof sa
) < 0)
263 fatal(errno
, "error binding to %s", control_socket
);
264 if(listen(sfd
, 128) < 0)
265 fatal(errno
, "error calling listen on %s", control_socket
);
266 info("listening on %s", control_socket
);
269 cfd
= accept(sfd
, (struct sockaddr
*)&sa
, &salen
);
276 fatal(errno
, "error calling accept on %s", control_socket
);
279 if(!(fp
= fdopen(cfd
, "r+"))) {
280 error(errno
, "error calling fdopen for %s connection", control_socket
);
284 if(!inputline(control_socket
, fp
, &line
, '\n')) {
285 if(!strcmp(line
, "stop")) {
286 info("stopped via %s", control_socket
);
287 exit(0); /* terminate immediately */
289 if(!strcmp(line
, "query"))
290 fprintf(fp
, "running");
294 error(errno
, "error closing %s connection", control_socket
);
298 /** @brief Drop the first packet
300 * Assumes that @ref lock is held.
302 static void drop_first_packet(void) {
303 if(pheap_count(&packets
)) {
304 struct packet
*const p
= pheap_remove(&packets
);
305 nsamples
-= p
->nsamples
;
306 playrtp_free_packet(p
);
307 pthread_cond_broadcast(&cond
);
311 /** @brief Background thread adding packets to heap
313 * This just transfers packets from @ref received_packets to @ref packets. It
314 * is important that it holds @ref receive_lock for as little time as possible,
315 * in order to minimize the interval between calls to read() in
318 static void *queue_thread(void attribute((unused
)) *arg
) {
322 /* Get the next packet */
323 pthread_mutex_lock(&receive_lock
);
324 while(!received_packets
) {
325 pthread_cond_wait(&receive_cond
, &receive_lock
);
327 p
= received_packets
;
328 received_packets
= p
->next
;
329 if(!received_packets
)
330 received_tail
= &received_packets
;
332 pthread_mutex_unlock(&receive_lock
);
333 /* Add it to the heap */
334 pthread_mutex_lock(&lock
);
335 pheap_insert(&packets
, p
);
336 nsamples
+= p
->nsamples
;
337 pthread_cond_broadcast(&cond
);
338 pthread_mutex_unlock(&lock
);
342 /** @brief Background thread collecting samples
344 * This function collects samples, perhaps converts them to the target format,
345 * and adds them to the packet list.
347 * It is crucial that the gap between successive calls to read() is as small as
348 * possible: otherwise packets will be dropped.
350 * We use a binary heap to ensure that the unavoidable effort is at worst
351 * logarithmic in the total number of packets - in fact if packets are mostly
352 * received in order then we will largely do constant work per packet since the
353 * newest packet will always be last.
355 * Of more concern is that we must acquire the lock on the heap to add a packet
356 * to it. If this proves a problem in practice then the answer would be
357 * (probably doubly) linked list with new packets added the end and a second
358 * thread which reads packets off the list and adds them to the heap.
360 * We keep memory allocation (mostly) very fast by keeping pre-allocated
361 * packets around; see @ref playrtp_new_packet().
363 static void *listen_thread(void attribute((unused
)) *arg
) {
364 struct packet
*p
= 0;
366 struct rtp_header header
;
373 p
= playrtp_new_packet();
374 iov
[0].iov_base
= &header
;
375 iov
[0].iov_len
= sizeof header
;
376 iov
[1].iov_base
= p
->samples_raw
;
377 iov
[1].iov_len
= sizeof p
->samples_raw
/ sizeof *p
->samples_raw
;
378 n
= readv(rtpfd
, iov
, 2);
384 fatal(errno
, "error reading from socket");
387 /* Ignore too-short packets */
388 if((size_t)n
<= sizeof (struct rtp_header
)) {
389 info("ignored a short packet");
392 timestamp
= htonl(header
.timestamp
);
393 seq
= htons(header
.seq
);
394 /* Ignore packets in the past */
395 if(active
&& lt(timestamp
, next_timestamp
)) {
396 info("dropping old packet, timestamp=%"PRIx32
" < %"PRIx32
,
397 timestamp
, next_timestamp
);
402 p
->timestamp
= timestamp
;
403 /* Convert to target format */
404 if(header
.mpt
& 0x80)
406 switch(header
.mpt
& 0x7F) {
408 p
->nsamples
= (n
- sizeof header
) / sizeof(uint16_t);
410 /* TODO support other RFC3551 media types (when the speaker does) */
412 fatal(0, "unsupported RTP payload type %d",
416 fprintf(logfp
, "sequence %u timestamp %"PRIx32
" length %"PRIx32
" end %"PRIx32
"\n",
417 seq
, timestamp
, p
->nsamples
, timestamp
+ p
->nsamples
);
418 /* Stop reading if we've reached the maximum.
420 * This is rather unsatisfactory: it means that if packets get heavily
421 * out of order then we guarantee dropouts. But for now... */
422 if(nsamples
>= maxbuffer
) {
423 pthread_mutex_lock(&lock
);
424 while(nsamples
>= maxbuffer
) {
425 pthread_cond_wait(&cond
, &lock
);
427 pthread_mutex_unlock(&lock
);
429 /* Add the packet to the receive queue */
430 pthread_mutex_lock(&receive_lock
);
432 received_tail
= &p
->next
;
434 pthread_cond_signal(&receive_cond
);
435 pthread_mutex_unlock(&receive_lock
);
436 /* We'll need a new packet */
441 /** @brief Wait until the buffer is adequately full
443 * Must be called with @ref lock held.
445 void playrtp_fill_buffer(void) {
448 info("Buffering...");
449 while(nsamples
< readahead
) {
450 pthread_cond_wait(&cond
, &lock
);
452 next_timestamp
= pheap_first(&packets
)->timestamp
;
456 /** @brief Find next packet
457 * @return Packet to play or NULL if none found
459 * The return packet is merely guaranteed not to be in the past: it might be
460 * the first packet in the future rather than one that is actually suitable to
463 * Must be called with @ref lock held.
465 struct packet
*playrtp_next_packet(void) {
466 while(pheap_count(&packets
)) {
467 struct packet
*const p
= pheap_first(&packets
);
468 if(le(p
->timestamp
+ p
->nsamples
, next_timestamp
)) {
469 /* This packet is in the past. Drop it and try another one. */
472 /* This packet is NOT in the past. (It might be in the future
479 /** @brief Play an RTP stream
481 * This is the guts of the program. It is responsible for:
482 * - starting the listening thread
483 * - opening the audio device
484 * - reading ahead to build up a buffer
485 * - arranging for audio to be played
486 * - detecting when the buffer has got too small and re-buffering
488 static void play_rtp(void) {
492 /* We receive and convert audio data in a background thread */
493 if((err
= pthread_create(<id
, 0, listen_thread
, 0)))
494 fatal(err
, "pthread_create listen_thread");
495 /* We have a second thread to add received packets to the queue */
496 if((err
= pthread_create(<id
, 0, queue_thread
, 0)))
497 fatal(err
, "pthread_create queue_thread");
498 /* The rest of the work is backend-specific */
502 /* display usage message and terminate */
503 static void help(void) {
505 " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
507 " --device, -D DEVICE Output device\n"
508 " --min, -m FRAMES Buffer low water mark\n"
509 " --buffer, -b FRAMES Buffer high water mark\n"
510 " --max, -x FRAMES Buffer maximum size\n"
511 " --rcvbuf, -R BYTES Socket receive buffer size\n"
512 " --config, -C PATH Set configuration file\n"
513 #if HAVE_ALSA_ASOUNDLIB_H
514 " --alsa, -a Use ALSA to play audio\n"
516 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
517 " --oss, -o Use OSS to play audio\n"
519 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
520 " --core-audio, -c Use Core Audio to play audio\n"
522 " --help, -h Display usage message\n"
523 " --version, -V Display version number\n"
529 int main(int argc
, char **argv
) {
531 struct addrinfo
*res
;
532 struct stringlist sl
;
534 int rcvbuf
, target_rcvbuf
= 131072;
537 struct ipv6_mreq mreq6
;
539 char *address
, *port
;
543 struct sockaddr_in in
;
544 struct sockaddr_in6 in6
;
546 union any_sockaddr mgroup
;
547 const char *dumpfile
= 0;
549 static const struct addrinfo prefs
= {
550 .ai_flags
= AI_PASSIVE
,
551 .ai_family
= PF_INET
,
552 .ai_socktype
= SOCK_DGRAM
,
553 .ai_protocol
= IPPROTO_UDP
557 if(!setlocale(LC_CTYPE
, "")) fatal(errno
, "error calling setlocale");
558 while((n
= getopt_long(argc
, argv
, "hVdD:m:b:x:L:R:M:aocC:r", options
, 0)) >= 0) {
561 case 'V': version("disorder-playrtp");
562 case 'd': debugging
= 1; break;
563 case 'D': device
= optarg
; break;
564 case 'm': minbuffer
= 2 * atol(optarg
); break;
565 case 'b': readahead
= 2 * atol(optarg
); break;
566 case 'x': maxbuffer
= 2 * atol(optarg
); break;
567 case 'L': logfp
= fopen(optarg
, "w"); break;
568 case 'R': target_rcvbuf
= atoi(optarg
); break;
569 #if HAVE_ALSA_ASOUNDLIB_H
570 case 'a': backend
= playrtp_alsa
; break;
572 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
573 case 'o': backend
= playrtp_oss
; break;
575 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
576 case 'c': backend
= playrtp_coreaudio
; break;
578 case 'C': configfile
= optarg
; break;
579 case 's': control_socket
= optarg
; break;
580 case 'r': dumpfile
= optarg
; break;
581 default: fatal(0, "invalid option");
584 if(config_read(0)) fatal(0, "cannot read configuration");
586 maxbuffer
= 4 * readahead
;
591 /* Get configuration from server */
592 if(!(c
= disorder_new(1))) exit(EXIT_FAILURE
);
593 if(disorder_connect(c
)) exit(EXIT_FAILURE
);
594 if(disorder_rtp_address(c
, &address
, &port
)) exit(EXIT_FAILURE
);
596 sl
.s
= xcalloc(2, sizeof *sl
.s
);
602 /* Use command-line ADDRESS+PORT or just PORT */
607 fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]");
609 /* Look up address and port */
610 if(!(res
= get_address(&sl
, &prefs
, &sockname
)))
612 /* Create the socket */
613 if((rtpfd
= socket(res
->ai_family
,
615 res
->ai_protocol
)) < 0)
616 fatal(errno
, "error creating socket");
617 /* Stash the multicast group address */
618 if((is_multicast
= multicast(res
->ai_addr
))) {
619 memcpy(&mgroup
, res
->ai_addr
, res
->ai_addrlen
);
620 switch(res
->ai_addr
->sa_family
) {
622 mgroup
.in
.sin_port
= 0;
625 mgroup
.in6
.sin6_port
= 0;
630 switch(res
->ai_addr
->sa_family
) {
632 memset(&((struct sockaddr_in
*)res
->ai_addr
)->sin_addr
, 0,
633 sizeof (struct in_addr
));
636 memset(&((struct sockaddr_in6
*)res
->ai_addr
)->sin6_addr
, 0,
637 sizeof (struct in6_addr
));
640 fatal(0, "unsupported family %d", (int)res
->ai_addr
->sa_family
);
642 if(bind(rtpfd
, res
->ai_addr
, res
->ai_addrlen
) < 0)
643 fatal(errno
, "error binding socket to %s", sockname
);
645 switch(mgroup
.sa
.sa_family
) {
647 mreq
.imr_multiaddr
= mgroup
.in
.sin_addr
;
648 mreq
.imr_interface
.s_addr
= 0; /* use primary interface */
649 if(setsockopt(rtpfd
, IPPROTO_IP
, IP_ADD_MEMBERSHIP
,
650 &mreq
, sizeof mreq
) < 0)
651 fatal(errno
, "error calling setsockopt IP_ADD_MEMBERSHIP");
654 mreq6
.ipv6mr_multiaddr
= mgroup
.in6
.sin6_addr
;
655 memset(&mreq6
.ipv6mr_interface
, 0, sizeof mreq6
.ipv6mr_interface
);
656 if(setsockopt(rtpfd
, IPPROTO_IPV6
, IPV6_JOIN_GROUP
,
657 &mreq6
, sizeof mreq6
) < 0)
658 fatal(errno
, "error calling setsockopt IPV6_JOIN_GROUP");
661 fatal(0, "unsupported address family %d", res
->ai_family
);
663 info("listening on %s multicast group %s",
664 format_sockaddr(res
->ai_addr
), format_sockaddr(&mgroup
.sa
));
666 info("listening on %s", format_sockaddr(res
->ai_addr
));
668 if(getsockopt(rtpfd
, SOL_SOCKET
, SO_RCVBUF
, &rcvbuf
, &len
) < 0)
669 fatal(errno
, "error calling getsockopt SO_RCVBUF");
670 if(target_rcvbuf
> rcvbuf
) {
671 if(setsockopt(rtpfd
, SOL_SOCKET
, SO_RCVBUF
,
672 &target_rcvbuf
, sizeof target_rcvbuf
) < 0)
673 error(errno
, "error calling setsockopt SO_RCVBUF %d",
675 /* We try to carry on anyway */
677 info("changed socket receive buffer from %d to %d",
678 rcvbuf
, target_rcvbuf
);
680 info("default socket receive buffer %d", rcvbuf
);
682 info("WARNING: -L option can impact performance");
686 if((err
= pthread_create(&tid
, 0, control_thread
, 0)))
687 fatal(err
, "pthread_create control_thread");
691 unsigned char buffer
[65536];
694 if((fd
= open(dumpfile
, O_RDWR
|O_TRUNC
|O_CREAT
, 0666)) < 0)
695 fatal(errno
, "opening %s", dumpfile
);
696 /* Fill with 0s to a suitable size */
697 memset(buffer
, 0, sizeof buffer
);
698 for(written
= 0; written
< dump_size
* sizeof(int16_t);
699 written
+= sizeof buffer
) {
700 if(write(fd
, buffer
, sizeof buffer
) < 0)
701 fatal(errno
, "clearing %s", dumpfile
);
703 /* Map the buffer into memory for convenience */
704 dump_buffer
= mmap(0, dump_size
* sizeof(int16_t), PROT_READ
|PROT_WRITE
,
706 if(dump_buffer
== (void *)-1)
707 fatal(errno
, "mapping %s", dumpfile
);
708 info("dumping to %s", dumpfile
);