2 * This file is part of DisOrder.
3 * Copyright (C) 2007 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>
75 #include "configuration.h"
85 #include "inputline.h"
87 #define readahead linux_headers_are_borked
89 /** @brief Obsolete synonym */
90 #ifndef IPV6_JOIN_GROUP
91 # define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP
94 /** @brief RTP socket */
97 /** @brief Log output */
100 /** @brief Output device */
103 /** @brief Minimum low watermark
105 * We'll stop playing if there's only this many samples in the buffer. */
106 unsigned minbuffer
= 2 * 44100 / 10; /* 0.2 seconds */
108 /** @brief Buffer high watermark
110 * We'll only start playing when this many samples are available. */
111 static unsigned readahead
= 2 * 2 * 44100;
113 /** @brief Maximum buffer size
115 * We'll stop reading from the network if we have this many samples. */
116 static unsigned maxbuffer
;
118 /** @brief Received packets
119 * Protected by @ref receive_lock
121 * Received packets are added to this list, and queue_thread() picks them off
122 * it and adds them to @ref packets. Whenever a packet is added to it, @ref
123 * receive_cond is signalled.
125 struct packet
*received_packets
;
127 /** @brief Tail of @ref received_packets
128 * Protected by @ref receive_lock
130 struct packet
**received_tail
= &received_packets
;
132 /** @brief Lock protecting @ref received_packets
134 * Only listen_thread() and queue_thread() ever hold this lock. It is vital
135 * that queue_thread() not hold it any longer than it strictly has to. */
136 pthread_mutex_t receive_lock
= PTHREAD_MUTEX_INITIALIZER
;
138 /** @brief Condition variable signalled when @ref received_packets is updated
140 * Used by listen_thread() to notify queue_thread() that it has added another
141 * packet to @ref received_packets. */
142 pthread_cond_t receive_cond
= PTHREAD_COND_INITIALIZER
;
144 /** @brief Length of @ref received_packets */
147 /** @brief Binary heap of received packets */
148 struct pheap packets
;
150 /** @brief Total number of samples available
152 * We make this volatile because we inspect it without a protecting lock,
153 * so the usual pthread_* guarantees aren't available.
155 volatile uint32_t nsamples
;
157 /** @brief Timestamp of next packet to play.
159 * This is set to the timestamp of the last packet, plus the number of
160 * samples it contained. Only valid if @ref active is nonzero.
162 uint32_t next_timestamp
;
164 /** @brief True if actively playing
166 * This is true when playing and false when just buffering. */
169 /** @brief Lock protecting @ref packets */
170 pthread_mutex_t lock
= PTHREAD_MUTEX_INITIALIZER
;
172 /** @brief Condition variable signalled whenever @ref packets is changed */
173 pthread_cond_t cond
= PTHREAD_COND_INITIALIZER
;
175 #if HAVE_ALSA_ASOUNDLIB_H
176 # define DEFAULT_BACKEND playrtp_alsa
177 #elif HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
178 # define DEFAULT_BACKEND playrtp_oss
179 #elif HAVE_COREAUDIO_AUDIOHARDWARE_H
180 # define DEFAULT_BACKEND playrtp_coreaudio
182 # error No known backend
185 /** @brief Backend to play with */
186 static void (*backend
)(void) = &DEFAULT_BACKEND
;
188 HEAP_DEFINE(pheap
, struct packet
*, lt_packet
);
190 /** @brief Control socket or NULL */
191 const char *control_socket
;
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 { "buffer", required_argument
, 0, 'b' },
201 { "rcvbuf", required_argument
, 0, 'R' },
202 { "multicast", required_argument
, 0, 'M' },
203 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
204 { "oss", no_argument
, 0, 'o' },
206 #if HAVE_ALSA_ASOUNDLIB_H
207 { "alsa", no_argument
, 0, 'a' },
209 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
210 { "core-audio", no_argument
, 0, 'c' },
212 { "socket", required_argument
, 0, 's' },
213 { "config", required_argument
, 0, 'C' },
217 /** @brief Control thread
219 * This thread is responsible for accepting control commands from Disobedience
220 * (or other controllers) over an AF_UNIX stream socket with a path specified
221 * by the @c --socket option. The protocol uses simple string commands and
224 * - @c stop will shut the player down
225 * - @c query will send back the reply @c running
226 * - anything else is ignored
228 * Commands and response strings terminated by shutting down the connection or
229 * by a newline. No attempt is made to multiplex multiple clients so it is
230 * important that the command be sent as soon as the connection is made - it is
231 * assumed that both parties to the protocol are entirely cooperating with one
234 static void *control_thread(void attribute((unused
)) *arg
) {
235 struct sockaddr_un sa
;
241 assert(control_socket
);
242 unlink(control_socket
);
243 memset(&sa
, 0, sizeof sa
);
244 sa
.sun_family
= AF_UNIX
;
245 strcpy(sa
.sun_path
, control_socket
);
246 sfd
= xsocket(PF_UNIX
, SOCK_STREAM
, 0);
247 if(bind(sfd
, (const struct sockaddr
*)&sa
, sizeof sa
) < 0)
248 fatal(errno
, "error binding to %s", control_socket
);
249 if(listen(sfd
, 128) < 0)
250 fatal(errno
, "error calling listen on %s", control_socket
);
251 info("listening on %s", control_socket
);
254 cfd
= accept(sfd
, (struct sockaddr
*)&sa
, &salen
);
261 fatal(errno
, "error calling accept on %s", control_socket
);
264 if(!(fp
= fdopen(cfd
, "r+"))) {
265 error(errno
, "error calling fdopen for %s connection", control_socket
);
269 if(!inputline(control_socket
, fp
, &line
, '\n')) {
270 if(!strcmp(line
, "stop")) {
271 info("stopped via %s", control_socket
);
272 exit(0); /* terminate immediately */
274 if(!strcmp(line
, "query"))
275 fprintf(fp
, "running");
279 error(errno
, "error closing %s connection", control_socket
);
283 /** @brief Drop the first packet
285 * Assumes that @ref lock is held.
287 static void drop_first_packet(void) {
288 if(pheap_count(&packets
)) {
289 struct packet
*const p
= pheap_remove(&packets
);
290 nsamples
-= p
->nsamples
;
291 playrtp_free_packet(p
);
292 pthread_cond_broadcast(&cond
);
296 /** @brief Background thread adding packets to heap
298 * This just transfers packets from @ref received_packets to @ref packets. It
299 * is important that it holds @ref receive_lock for as little time as possible,
300 * in order to minimize the interval between calls to read() in
303 static void *queue_thread(void attribute((unused
)) *arg
) {
307 /* Get the next packet */
308 pthread_mutex_lock(&receive_lock
);
309 while(!received_packets
)
310 pthread_cond_wait(&receive_cond
, &receive_lock
);
311 p
= received_packets
;
312 received_packets
= p
->next
;
313 if(!received_packets
)
314 received_tail
= &received_packets
;
316 pthread_mutex_unlock(&receive_lock
);
317 /* Add it to the heap */
318 pthread_mutex_lock(&lock
);
319 pheap_insert(&packets
, p
);
320 nsamples
+= p
->nsamples
;
321 pthread_cond_broadcast(&cond
);
322 pthread_mutex_unlock(&lock
);
326 /** @brief Background thread collecting samples
328 * This function collects samples, perhaps converts them to the target format,
329 * and adds them to the packet list.
331 * It is crucial that the gap between successive calls to read() is as small as
332 * possible: otherwise packets will be dropped.
334 * We use a binary heap to ensure that the unavoidable effort is at worst
335 * logarithmic in the total number of packets - in fact if packets are mostly
336 * received in order then we will largely do constant work per packet since the
337 * newest packet will always be last.
339 * Of more concern is that we must acquire the lock on the heap to add a packet
340 * to it. If this proves a problem in practice then the answer would be
341 * (probably doubly) linked list with new packets added the end and a second
342 * thread which reads packets off the list and adds them to the heap.
344 * We keep memory allocation (mostly) very fast by keeping pre-allocated
345 * packets around; see @ref playrtp_new_packet().
347 static void *listen_thread(void attribute((unused
)) *arg
) {
348 struct packet
*p
= 0;
350 struct rtp_header header
;
357 p
= playrtp_new_packet();
358 iov
[0].iov_base
= &header
;
359 iov
[0].iov_len
= sizeof header
;
360 iov
[1].iov_base
= p
->samples_raw
;
361 iov
[1].iov_len
= sizeof p
->samples_raw
/ sizeof *p
->samples_raw
;
362 n
= readv(rtpfd
, iov
, 2);
368 fatal(errno
, "error reading from socket");
371 /* Ignore too-short packets */
372 if((size_t)n
<= sizeof (struct rtp_header
)) {
373 info("ignored a short packet");
376 timestamp
= htonl(header
.timestamp
);
377 seq
= htons(header
.seq
);
378 /* Ignore packets in the past */
379 if(active
&& lt(timestamp
, next_timestamp
)) {
380 info("dropping old packet, timestamp=%"PRIx32
" < %"PRIx32
,
381 timestamp
, next_timestamp
);
386 p
->timestamp
= timestamp
;
387 /* Convert to target format */
388 if(header
.mpt
& 0x80)
390 switch(header
.mpt
& 0x7F) {
392 p
->nsamples
= (n
- sizeof header
) / sizeof(uint16_t);
394 /* TODO support other RFC3551 media types (when the speaker does) */
396 fatal(0, "unsupported RTP payload type %d",
400 fprintf(logfp
, "sequence %u timestamp %"PRIx32
" length %"PRIx32
" end %"PRIx32
"\n",
401 seq
, timestamp
, p
->nsamples
, timestamp
+ p
->nsamples
);
402 /* Stop reading if we've reached the maximum.
404 * This is rather unsatisfactory: it means that if packets get heavily
405 * out of order then we guarantee dropouts. But for now... */
406 if(nsamples
>= maxbuffer
) {
407 pthread_mutex_lock(&lock
);
408 while(nsamples
>= maxbuffer
)
409 pthread_cond_wait(&cond
, &lock
);
410 pthread_mutex_unlock(&lock
);
412 /* Add the packet to the receive queue */
413 pthread_mutex_lock(&receive_lock
);
415 received_tail
= &p
->next
;
417 pthread_cond_signal(&receive_cond
);
418 pthread_mutex_unlock(&receive_lock
);
419 /* We'll need a new packet */
424 /** @brief Wait until the buffer is adequately full
426 * Must be called with @ref lock held.
428 void playrtp_fill_buffer(void) {
431 info("Buffering...");
432 while(nsamples
< readahead
)
433 pthread_cond_wait(&cond
, &lock
);
434 next_timestamp
= pheap_first(&packets
)->timestamp
;
438 /** @brief Find next packet
439 * @return Packet to play or NULL if none found
441 * The return packet is merely guaranteed not to be in the past: it might be
442 * the first packet in the future rather than one that is actually suitable to
445 * Must be called with @ref lock held.
447 struct packet
*playrtp_next_packet(void) {
448 while(pheap_count(&packets
)) {
449 struct packet
*const p
= pheap_first(&packets
);
450 if(le(p
->timestamp
+ p
->nsamples
, next_timestamp
)) {
451 /* This packet is in the past. Drop it and try another one. */
454 /* This packet is NOT in the past. (It might be in the future
461 /** @brief Play an RTP stream
463 * This is the guts of the program. It is responsible for:
464 * - starting the listening thread
465 * - opening the audio device
466 * - reading ahead to build up a buffer
467 * - arranging for audio to be played
468 * - detecting when the buffer has got too small and re-buffering
470 static void play_rtp(void) {
474 /* We receive and convert audio data in a background thread */
475 if((err
= pthread_create(<id
, 0, listen_thread
, 0)))
476 fatal(err
, "pthread_create listen_thread");
477 /* We have a second thread to add received packets to the queue */
478 if((err
= pthread_create(<id
, 0, queue_thread
, 0)))
479 fatal(err
, "pthread_create queue_thread");
480 /* The rest of the work is backend-specific */
484 /* display usage message and terminate */
485 static void help(void) {
487 " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
489 " --device, -D DEVICE Output device\n"
490 " --min, -m FRAMES Buffer low water mark\n"
491 " --buffer, -b FRAMES Buffer high water mark\n"
492 " --max, -x FRAMES Buffer maximum size\n"
493 " --rcvbuf, -R BYTES Socket receive buffer size\n"
494 " --multicast, -M GROUP Join multicast group\n"
495 " --config, -C PATH Set configuration file\n"
496 #if HAVE_ALSA_ASOUNDLIB_H
497 " --alsa, -a Use ALSA to play audio\n"
499 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
500 " --oss, -o Use OSS to play audio\n"
502 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
503 " --core-audio, -c Use Core Audio to play audio\n"
505 " --help, -h Display usage message\n"
506 " --version, -V Display version number\n"
512 /* display version number and terminate */
513 static void version(void) {
514 xprintf("disorder-playrtp version %s\n", disorder_version_string
);
519 int main(int argc
, char **argv
) {
521 struct addrinfo
*res
;
522 struct stringlist sl
;
524 int rcvbuf
, target_rcvbuf
= 131072;
526 char *multicast_group
= 0;
528 struct ipv6_mreq mreq6
;
530 char *address
, *port
;
532 static const struct addrinfo prefs
= {
544 if(!setlocale(LC_CTYPE
, "")) fatal(errno
, "error calling setlocale");
545 while((n
= getopt_long(argc
, argv
, "hVdD:m:b:x:L:R:M:aocC:", options
, 0)) >= 0) {
549 case 'd': debugging
= 1; break;
550 case 'D': device
= optarg
; break;
551 case 'm': minbuffer
= 2 * atol(optarg
); break;
552 case 'b': readahead
= 2 * atol(optarg
); break;
553 case 'x': maxbuffer
= 2 * atol(optarg
); break;
554 case 'L': logfp
= fopen(optarg
, "w"); break;
555 case 'R': target_rcvbuf
= atoi(optarg
); break;
556 case 'M': multicast_group
= optarg
; break;
557 #if HAVE_ALSA_ASOUNDLIB_H
558 case 'a': backend
= playrtp_alsa
; break;
560 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
561 case 'o': backend
= playrtp_oss
; break;
563 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
564 case 'c': backend
= playrtp_coreaudio
; break;
566 case 'C': configfile
= optarg
; break;
567 case 's': control_socket
= optarg
; break;
568 default: fatal(0, "invalid option");
571 if(config_read(0)) fatal(0, "cannot read configuration");
573 maxbuffer
= 4 * readahead
;
579 if(!(c
= disorder_new(1))) exit(EXIT_FAILURE
);
580 if(disorder_connect(c
)) exit(EXIT_FAILURE
);
581 if(disorder_rtp_address(c
, &address
, &port
)) exit(EXIT_FAILURE
);
584 /* set multicast_group if address is a multicast address */
591 fatal(0, "usage: disorder-playrtp [OPTIONS] [ADDRESS [PORT]]");
593 /* Listen for inbound audio data */
594 if(!(res
= get_address(&sl
, &prefs
, &sockname
)))
596 info("listening on %s", sockname
);
597 if((rtpfd
= socket(res
->ai_family
,
599 res
->ai_protocol
)) < 0)
600 fatal(errno
, "error creating socket");
601 if(bind(rtpfd
, res
->ai_addr
, res
->ai_addrlen
) < 0)
602 fatal(errno
, "error binding socket to %s", sockname
);
603 if(multicast_group
) {
604 if((n
= getaddrinfo(multicast_group
, 0, &prefs
, &res
)))
605 fatal(0, "getaddrinfo %s: %s", multicast_group
, gai_strerror(n
));
606 switch(res
->ai_family
) {
608 mreq
.imr_multiaddr
= ((struct sockaddr_in
*)res
->ai_addr
)->sin_addr
;
609 mreq
.imr_interface
.s_addr
= 0; /* use primary interface */
610 if(setsockopt(rtpfd
, IPPROTO_IP
, IP_ADD_MEMBERSHIP
,
611 &mreq
, sizeof mreq
) < 0)
612 fatal(errno
, "error calling setsockopt IP_ADD_MEMBERSHIP");
615 mreq6
.ipv6mr_multiaddr
= ((struct sockaddr_in6
*)res
->ai_addr
)->sin6_addr
;
616 memset(&mreq6
.ipv6mr_interface
, 0, sizeof mreq6
.ipv6mr_interface
);
617 if(setsockopt(rtpfd
, IPPROTO_IPV6
, IPV6_JOIN_GROUP
,
618 &mreq6
, sizeof mreq6
) < 0)
619 fatal(errno
, "error calling setsockopt IPV6_JOIN_GROUP");
622 fatal(0, "unsupported address family %d", res
->ai_family
);
626 if(getsockopt(rtpfd
, SOL_SOCKET
, SO_RCVBUF
, &rcvbuf
, &len
) < 0)
627 fatal(errno
, "error calling getsockopt SO_RCVBUF");
628 if(target_rcvbuf
> rcvbuf
) {
629 if(setsockopt(rtpfd
, SOL_SOCKET
, SO_RCVBUF
,
630 &target_rcvbuf
, sizeof target_rcvbuf
) < 0)
631 error(errno
, "error calling setsockopt SO_RCVBUF %d",
633 /* We try to carry on anyway */
635 info("changed socket receive buffer from %d to %d",
636 rcvbuf
, target_rcvbuf
);
638 info("default socket receive buffer %d", rcvbuf
);
640 info("WARNING: -L option can impact performance");
644 if((err
= pthread_create(&tid
, 0, control_thread
, 0)))
645 fatal(err
, "pthread_create control_thread");