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);