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>
74 #include "configuration.h"
84 #include "inputline.h"
86 #define readahead linux_headers_are_borked
88 /** @brief Obsolete synonym */
89 #ifndef IPV6_JOIN_GROUP
90 # define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP
93 /** @brief RTP socket */
96 /** @brief Log output */
99 /** @brief Output device */
102 /** @brief Minimum low watermark
104 * We'll stop playing if there's only this many samples in the buffer. */
105 unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */
107 /** @brief Buffer high watermark
109 * We'll only start playing when this many samples are available. */
110 static unsigned readahead = 2 * 2 * 44100;
112 /** @brief Maximum buffer size
114 * We'll stop reading from the network if we have this many samples. */
115 static unsigned maxbuffer;
117 /** @brief Received packets
118 * Protected by @ref receive_lock
120 * Received packets are added to this list, and queue_thread() picks them off
121 * it and adds them to @ref packets. Whenever a packet is added to it, @ref
122 * receive_cond is signalled.
124 struct packet *received_packets;
126 /** @brief Tail of @ref received_packets
127 * Protected by @ref receive_lock
129 struct packet **received_tail = &received_packets;
131 /** @brief Lock protecting @ref received_packets
133 * Only listen_thread() and queue_thread() ever hold this lock. It is vital
134 * that queue_thread() not hold it any longer than it strictly has to. */
135 pthread_mutex_t receive_lock = PTHREAD_MUTEX_INITIALIZER;
137 /** @brief Condition variable signalled when @ref received_packets is updated
139 * Used by listen_thread() to notify queue_thread() that it has added another
140 * packet to @ref received_packets. */
141 pthread_cond_t receive_cond = PTHREAD_COND_INITIALIZER;
143 /** @brief Length of @ref received_packets */
146 /** @brief Binary heap of received packets */
147 struct pheap packets;
149 /** @brief Total number of samples available
151 * We make this volatile because we inspect it without a protecting lock,
152 * so the usual pthread_* guarantees aren't available.
154 volatile uint32_t nsamples;
156 /** @brief Timestamp of next packet to play.
158 * This is set to the timestamp of the last packet, plus the number of
159 * samples it contained. Only valid if @ref active is nonzero.
161 uint32_t next_timestamp;
163 /** @brief True if actively playing
165 * This is true when playing and false when just buffering. */
168 /** @brief Lock protecting @ref packets */
169 pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
171 /** @brief Condition variable signalled whenever @ref packets is changed */
172 pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
174 #if HAVE_ALSA_ASOUNDLIB_H
175 # define DEFAULT_BACKEND playrtp_alsa
176 #elif HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
177 # define DEFAULT_BACKEND playrtp_oss
178 #elif HAVE_COREAUDIO_AUDIOHARDWARE_H
179 # define DEFAULT_BACKEND playrtp_coreaudio
181 # error No known backend
184 /** @brief Backend to play with */
185 static void (*backend)(void) = &DEFAULT_BACKEND;
187 HEAP_DEFINE(pheap, struct packet *, lt_packet);
189 /** @brief Control socket or NULL */
190 const char *control_socket;
192 static const struct option options[] = {
193 { "help", no_argument, 0, 'h' },
194 { "version", no_argument, 0, 'V' },
195 { "debug", no_argument, 0, 'd' },
196 { "device", required_argument, 0, 'D' },
197 { "min", required_argument, 0, 'm' },
198 { "max", required_argument, 0, 'x' },
199 { "buffer", required_argument, 0, 'b' },
200 { "rcvbuf", required_argument, 0, 'R' },
201 { "multicast", required_argument, 0, 'M' },
202 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
203 { "oss", no_argument, 0, 'o' },
205 #if HAVE_ALSA_ASOUNDLIB_H
206 { "alsa", no_argument, 0, 'a' },
208 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
209 { "core-audio", no_argument, 0, 'c' },
211 { "socket", required_argument, 0, 's' },
212 { "config", required_argument, 0, 'C' },
216 /** @brief Control thread
218 * This thread is responsible for accepting control commands from Disobedience
219 * (or other controllers) over an AF_UNIX stream socket with a path specified
220 * by the @c --socket option. The protocol uses simple string commands and
223 * - @c stop will shut the player down
224 * - @c query will send back the reply @c running
225 * - anything else is ignored
227 * Commands and response strings terminated by shutting down the connection or
228 * by a newline. No attempt is made to multiplex multiple clients so it is
229 * important that the command be sent as soon as the connection is made - it is
230 * assumed that both parties to the protocol are entirely cooperating with one
233 static void *control_thread(void attribute((unused)) *arg) {
234 struct sockaddr_un sa;
240 assert(control_socket);
241 unlink(control_socket);
242 memset(&sa, 0, sizeof sa);
243 sa.sun_family = AF_UNIX;
244 strcpy(sa.sun_path, control_socket);
245 sfd = xsocket(PF_UNIX, SOCK_STREAM, 0);
246 if(bind(sfd, (const struct sockaddr *)&sa, sizeof sa) < 0)
247 fatal(errno, "error binding to %s", control_socket);
248 if(listen(sfd, 128) < 0)
249 fatal(errno, "error calling listen on %s", control_socket);
250 info("listening on %s", control_socket);
253 cfd = accept(sfd, (struct sockaddr *)&sa, &salen);
260 fatal(errno, "error calling accept on %s", control_socket);
263 if(!(fp = fdopen(cfd, "r+"))) {
264 error(errno, "error calling fdopen for %s connection", control_socket);
268 if(!inputline(control_socket, fp, &line, '\n')) {
269 if(!strcmp(line, "stop")) {
270 info("stopped via %s", control_socket);
271 exit(0); /* terminate immediately */
273 if(!strcmp(line, "query"))
274 fprintf(fp, "running");
278 error(errno, "error closing %s connection", control_socket);
282 /** @brief Drop the first packet
284 * Assumes that @ref lock is held.
286 static void drop_first_packet(void) {
287 if(pheap_count(&packets)) {
288 struct packet *const p = pheap_remove(&packets);
289 nsamples -= p->nsamples;
290 playrtp_free_packet(p);
291 pthread_cond_broadcast(&cond);
295 /** @brief Background thread adding packets to heap
297 * This just transfers packets from @ref received_packets to @ref packets. It
298 * is important that it holds @ref receive_lock for as little time as possible,
299 * in order to minimize the interval between calls to read() in
302 static void *queue_thread(void attribute((unused)) *arg) {
306 /* Get the next packet */
307 pthread_mutex_lock(&receive_lock);
308 while(!received_packets)
309 pthread_cond_wait(&receive_cond, &receive_lock);
310 p = received_packets;
311 received_packets = p->next;
312 if(!received_packets)
313 received_tail = &received_packets;
315 pthread_mutex_unlock(&receive_lock);
316 /* Add it to the heap */
317 pthread_mutex_lock(&lock);
318 pheap_insert(&packets, p);
319 nsamples += p->nsamples;
320 pthread_cond_broadcast(&cond);
321 pthread_mutex_unlock(&lock);
325 /** @brief Background thread collecting samples
327 * This function collects samples, perhaps converts them to the target format,
328 * and adds them to the packet list.
330 * It is crucial that the gap between successive calls to read() is as small as
331 * possible: otherwise packets will be dropped.
333 * We use a binary heap to ensure that the unavoidable effort is at worst
334 * logarithmic in the total number of packets - in fact if packets are mostly
335 * received in order then we will largely do constant work per packet since the
336 * newest packet will always be last.
338 * Of more concern is that we must acquire the lock on the heap to add a packet
339 * to it. If this proves a problem in practice then the answer would be
340 * (probably doubly) linked list with new packets added the end and a second
341 * thread which reads packets off the list and adds them to the heap.
343 * We keep memory allocation (mostly) very fast by keeping pre-allocated
344 * packets around; see @ref playrtp_new_packet().
346 static void *listen_thread(void attribute((unused)) *arg) {
347 struct packet *p = 0;
349 struct rtp_header header;
356 p = playrtp_new_packet();
357 iov[0].iov_base = &header;
358 iov[0].iov_len = sizeof header;
359 iov[1].iov_base = p->samples_raw;
360 iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw;
361 n = readv(rtpfd, iov, 2);
367 fatal(errno, "error reading from socket");
370 /* Ignore too-short packets */
371 if((size_t)n <= sizeof (struct rtp_header)) {
372 info("ignored a short packet");
375 timestamp = htonl(header.timestamp);
376 seq = htons(header.seq);
377 /* Ignore packets in the past */
378 if(active && lt(timestamp, next_timestamp)) {
379 info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
380 timestamp, next_timestamp);
385 p->timestamp = timestamp;
386 /* Convert to target format */
387 if(header.mpt & 0x80)
389 switch(header.mpt & 0x7F) {
391 p->nsamples = (n - sizeof header) / sizeof(uint16_t);
393 /* TODO support other RFC3551 media types (when the speaker does) */
395 fatal(0, "unsupported RTP payload type %d",
399 fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
400 seq, timestamp, p->nsamples, timestamp + p->nsamples);
401 /* Stop reading if we've reached the maximum.
403 * This is rather unsatisfactory: it means that if packets get heavily
404 * out of order then we guarantee dropouts. But for now... */
405 if(nsamples >= maxbuffer) {
406 pthread_mutex_lock(&lock);
407 while(nsamples >= maxbuffer)
408 pthread_cond_wait(&cond, &lock);
409 pthread_mutex_unlock(&lock);
411 /* Add the packet to the receive queue */
412 pthread_mutex_lock(&receive_lock);
414 received_tail = &p->next;
416 pthread_cond_signal(&receive_cond);
417 pthread_mutex_unlock(&receive_lock);
418 /* We'll need a new packet */
423 /** @brief Wait until the buffer is adequately full
425 * Must be called with @ref lock held.
427 void playrtp_fill_buffer(void) {
430 info("Buffering...");
431 while(nsamples < readahead)
432 pthread_cond_wait(&cond, &lock);
433 next_timestamp = pheap_first(&packets)->timestamp;
437 /** @brief Find next packet
438 * @return Packet to play or NULL if none found
440 * The return packet is merely guaranteed not to be in the past: it might be
441 * the first packet in the future rather than one that is actually suitable to
444 * Must be called with @ref lock held.
446 struct packet *playrtp_next_packet(void) {
447 while(pheap_count(&packets)) {
448 struct packet *const p = pheap_first(&packets);
449 if(le(p->timestamp + p->nsamples, next_timestamp)) {
450 /* This packet is in the past. Drop it and try another one. */
453 /* This packet is NOT in the past. (It might be in the future
460 /** @brief Play an RTP stream
462 * This is the guts of the program. It is responsible for:
463 * - starting the listening thread
464 * - opening the audio device
465 * - reading ahead to build up a buffer
466 * - arranging for audio to be played
467 * - detecting when the buffer has got too small and re-buffering
469 static void play_rtp(void) {
473 /* We receive and convert audio data in a background thread */
474 if((err = pthread_create(<id, 0, listen_thread, 0)))
475 fatal(err, "pthread_create listen_thread");
476 /* We have a second thread to add received packets to the queue */
477 if((err = pthread_create(<id, 0, queue_thread, 0)))
478 fatal(err, "pthread_create queue_thread");
479 /* The rest of the work is backend-specific */
483 /* display usage message and terminate */
484 static void help(void) {
486 " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
488 " --device, -D DEVICE Output device\n"
489 " --min, -m FRAMES Buffer low water mark\n"
490 " --buffer, -b FRAMES Buffer high water mark\n"
491 " --max, -x FRAMES Buffer maximum size\n"
492 " --rcvbuf, -R BYTES Socket receive buffer size\n"
493 " --multicast, -M GROUP Join multicast group\n"
494 " --config, -C PATH Set configuration file\n"
495 #if HAVE_ALSA_ASOUNDLIB_H
496 " --alsa, -a Use ALSA to play audio\n"
498 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
499 " --oss, -o Use OSS to play audio\n"
501 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
502 " --core-audio, -c Use Core Audio to play audio\n"
504 " --help, -h Display usage message\n"
505 " --version, -V Display version number\n"
511 /* display version number and terminate */
512 static void version(void) {
513 xprintf("disorder-playrtp version %s\n", disorder_version_string);
518 int main(int argc, char **argv) {
520 struct addrinfo *res;
521 struct stringlist sl;
523 int rcvbuf, target_rcvbuf = 131072;
525 char *multicast_group = 0;
527 struct ipv6_mreq mreq6;
529 char *address, *port;
531 static const struct addrinfo prefs = {
543 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
544 while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:aocC:", options, 0)) >= 0) {
548 case 'd': debugging = 1; break;
549 case 'D': device = optarg; break;
550 case 'm': minbuffer = 2 * atol(optarg); break;
551 case 'b': readahead = 2 * atol(optarg); break;
552 case 'x': maxbuffer = 2 * atol(optarg); break;
553 case 'L': logfp = fopen(optarg, "w"); break;
554 case 'R': target_rcvbuf = atoi(optarg); break;
555 case 'M': multicast_group = optarg; break;
556 #if HAVE_ALSA_ASOUNDLIB_H
557 case 'a': backend = playrtp_alsa; break;
559 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
560 case 'o': backend = playrtp_oss; break;
562 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
563 case 'c': backend = playrtp_coreaudio; break;
565 case 'C': configfile = optarg; break;
566 case 's': control_socket = optarg; break;
567 default: fatal(0, "invalid option");
570 if(config_read(0)) fatal(0, "cannot read configuration");
572 maxbuffer = 4 * readahead;
578 if(!(c = disorder_new(1))) exit(EXIT_FAILURE);
579 if(disorder_connect(c)) exit(EXIT_FAILURE);
580 if(disorder_rtp_address(c, &address, &port)) exit(EXIT_FAILURE);
583 /* set multicast_group if address is a multicast address */
590 fatal(0, "usage: disorder-playrtp [OPTIONS] [ADDRESS [PORT]]");
592 /* Listen for inbound audio data */
593 if(!(res = get_address(&sl, &prefs, &sockname)))
595 info("listening on %s", sockname);
596 if((rtpfd = socket(res->ai_family,
598 res->ai_protocol)) < 0)
599 fatal(errno, "error creating socket");
600 if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
601 fatal(errno, "error binding socket to %s", sockname);
602 if(multicast_group) {
603 if((n = getaddrinfo(multicast_group, 0, &prefs, &res)))
604 fatal(0, "getaddrinfo %s: %s", multicast_group, gai_strerror(n));
605 switch(res->ai_family) {
607 mreq.imr_multiaddr = ((struct sockaddr_in *)res->ai_addr)->sin_addr;
608 mreq.imr_interface.s_addr = 0; /* use primary interface */
609 if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP,
610 &mreq, sizeof mreq) < 0)
611 fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP");
614 mreq6.ipv6mr_multiaddr = ((struct sockaddr_in6 *)res->ai_addr)->sin6_addr;
615 memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface);
616 if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP,
617 &mreq6, sizeof mreq6) < 0)
618 fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP");
621 fatal(0, "unsupported address family %d", res->ai_family);
625 if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0)
626 fatal(errno, "error calling getsockopt SO_RCVBUF");
627 if(target_rcvbuf > rcvbuf) {
628 if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF,
629 &target_rcvbuf, sizeof target_rcvbuf) < 0)
630 error(errno, "error calling setsockopt SO_RCVBUF %d",
632 /* We try to carry on anyway */
634 info("changed socket receive buffer from %d to %d",
635 rcvbuf, target_rcvbuf);
637 info("default socket receive buffer %d", rcvbuf);
639 info("WARNING: -L option can impact performance");
643 if((err = pthread_create(&tid, 0, control_thread, 0)))
644 fatal(err, "pthread_create control_thread");