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>
71 #include "configuration.h"
82 #define readahead linux_headers_are_borked
84 /** @brief RTP socket */
87 /** @brief Log output */
90 /** @brief Output device */
93 /** @brief Minimum low watermark
95 * We'll stop playing if there's only this many samples in the buffer. */
96 unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */
98 /** @brief Buffer high watermark
100 * We'll only start playing when this many samples are available. */
101 static unsigned readahead = 2 * 2 * 44100;
103 /** @brief Maximum buffer size
105 * We'll stop reading from the network if we have this many samples. */
106 static unsigned maxbuffer;
108 /** @brief Received packets
109 * Protected by @ref receive_lock
111 * Received packets are added to this list, and queue_thread() picks them off
112 * it and adds them to @ref packets. Whenever a packet is added to it, @ref
113 * receive_cond is signalled.
115 struct packet *received_packets;
117 /** @brief Tail of @ref received_packets
118 * Protected by @ref receive_lock
120 struct packet **received_tail = &received_packets;
122 /** @brief Lock protecting @ref received_packets
124 * Only listen_thread() and queue_thread() ever hold this lock. It is vital
125 * that queue_thread() not hold it any longer than it strictly has to. */
126 pthread_mutex_t receive_lock = PTHREAD_MUTEX_INITIALIZER;
128 /** @brief Condition variable signalled when @ref received_packets is updated
130 * Used by listen_thread() to notify queue_thread() that it has added another
131 * packet to @ref received_packets. */
132 pthread_cond_t receive_cond = PTHREAD_COND_INITIALIZER;
134 /** @brief Length of @ref received_packets */
137 /** @brief Binary heap of received packets */
138 struct pheap packets;
140 /** @brief Total number of samples available
142 * We make this volatile because we inspect it without a protecting lock,
143 * so the usual pthread_* guarantees aren't available.
145 volatile uint32_t nsamples;
147 /** @brief Timestamp of next packet to play.
149 * This is set to the timestamp of the last packet, plus the number of
150 * samples it contained. Only valid if @ref active is nonzero.
152 uint32_t next_timestamp;
154 /** @brief True if actively playing
156 * This is true when playing and false when just buffering. */
159 /** @brief Lock protecting @ref packets */
160 pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
162 /** @brief Condition variable signalled whenever @ref packets is changed */
163 pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
165 #if HAVE_ALSA_ASOUNDLIB_H
166 # define DEFAULT_BACKEND playrtp_alsa
167 #elif HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
168 # define DEFAULT_BACKEND playrtp_oss
169 #elif HAVE_COREAUDIO_AUDIOHARDWARE_H
170 # define DEFAULT_BACKEND playrtp_coreaudio
172 # error No known backend
175 /** @brief Backend to play with */
176 static void (*backend)(void) = &DEFAULT_BACKEND;
178 HEAP_DEFINE(pheap, struct packet *, lt_packet);
180 static const struct option options[] = {
181 { "help", no_argument, 0, 'h' },
182 { "version", no_argument, 0, 'V' },
183 { "debug", no_argument, 0, 'd' },
184 { "device", required_argument, 0, 'D' },
185 { "min", required_argument, 0, 'm' },
186 { "max", required_argument, 0, 'x' },
187 { "buffer", required_argument, 0, 'b' },
188 { "rcvbuf", required_argument, 0, 'R' },
189 { "multicast", required_argument, 0, 'M' },
190 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
191 { "oss", no_argument, 0, 'o' },
193 #if HAVE_ALSA_ASOUNDLIB_H
194 { "alsa", no_argument, 0, 'a' },
196 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
197 { "core-audio", no_argument, 0, 'c' },
199 { "config", required_argument, 0, 'C' },
203 /** @brief Drop the first packet
205 * Assumes that @ref lock is held.
207 static void drop_first_packet(void) {
208 if(pheap_count(&packets)) {
209 struct packet *const p = pheap_remove(&packets);
210 nsamples -= p->nsamples;
211 playrtp_free_packet(p);
212 pthread_cond_broadcast(&cond);
216 /** @brief Background thread adding packets to heap
218 * This just transfers packets from @ref received_packets to @ref packets. It
219 * is important that it holds @ref receive_lock for as little time as possible,
220 * in order to minimize the interval between calls to read() in
223 static void *queue_thread(void attribute((unused)) *arg) {
227 /* Get the next packet */
228 pthread_mutex_lock(&receive_lock);
229 while(!received_packets)
230 pthread_cond_wait(&receive_cond, &receive_lock);
231 p = received_packets;
232 received_packets = p->next;
233 if(!received_packets)
234 received_tail = &received_packets;
236 pthread_mutex_unlock(&receive_lock);
237 /* Add it to the heap */
238 pthread_mutex_lock(&lock);
239 pheap_insert(&packets, p);
240 nsamples += p->nsamples;
241 pthread_cond_broadcast(&cond);
242 pthread_mutex_unlock(&lock);
246 /** @brief Background thread collecting samples
248 * This function collects samples, perhaps converts them to the target format,
249 * and adds them to the packet list.
251 * It is crucial that the gap between successive calls to read() is as small as
252 * possible: otherwise packets will be dropped.
254 * We use a binary heap to ensure that the unavoidable effort is at worst
255 * logarithmic in the total number of packets - in fact if packets are mostly
256 * received in order then we will largely do constant work per packet since the
257 * newest packet will always be last.
259 * Of more concern is that we must acquire the lock on the heap to add a packet
260 * to it. If this proves a problem in practice then the answer would be
261 * (probably doubly) linked list with new packets added the end and a second
262 * thread which reads packets off the list and adds them to the heap.
264 * We keep memory allocation (mostly) very fast by keeping pre-allocated
265 * packets around; see @ref playrtp_new_packet().
267 static void *listen_thread(void attribute((unused)) *arg) {
268 struct packet *p = 0;
270 struct rtp_header header;
277 p = playrtp_new_packet();
278 iov[0].iov_base = &header;
279 iov[0].iov_len = sizeof header;
280 iov[1].iov_base = p->samples_raw;
281 iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw;
282 n = readv(rtpfd, iov, 2);
288 fatal(errno, "error reading from socket");
291 /* Ignore too-short packets */
292 if((size_t)n <= sizeof (struct rtp_header)) {
293 info("ignored a short packet");
296 timestamp = htonl(header.timestamp);
297 seq = htons(header.seq);
298 /* Ignore packets in the past */
299 if(active && lt(timestamp, next_timestamp)) {
300 info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
301 timestamp, next_timestamp);
306 p->timestamp = timestamp;
307 /* Convert to target format */
308 if(header.mpt & 0x80)
310 switch(header.mpt & 0x7F) {
312 p->nsamples = (n - sizeof header) / sizeof(uint16_t);
314 /* TODO support other RFC3551 media types (when the speaker does) */
316 fatal(0, "unsupported RTP payload type %d",
320 fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
321 seq, timestamp, p->nsamples, timestamp + p->nsamples);
322 /* Stop reading if we've reached the maximum.
324 * This is rather unsatisfactory: it means that if packets get heavily
325 * out of order then we guarantee dropouts. But for now... */
326 if(nsamples >= maxbuffer) {
327 pthread_mutex_lock(&lock);
328 while(nsamples >= maxbuffer)
329 pthread_cond_wait(&cond, &lock);
330 pthread_mutex_unlock(&lock);
332 /* Add the packet to the receive queue */
333 pthread_mutex_lock(&receive_lock);
335 received_tail = &p->next;
337 pthread_cond_signal(&receive_cond);
338 pthread_mutex_unlock(&receive_lock);
339 /* We'll need a new packet */
344 /** @brief Wait until the buffer is adequately full
346 * Must be called with @ref lock held.
348 void playrtp_fill_buffer(void) {
351 info("Buffering...");
352 while(nsamples < readahead)
353 pthread_cond_wait(&cond, &lock);
354 next_timestamp = pheap_first(&packets)->timestamp;
358 /** @brief Find next packet
359 * @return Packet to play or NULL if none found
361 * The return packet is merely guaranteed not to be in the past: it might be
362 * the first packet in the future rather than one that is actually suitable to
365 * Must be called with @ref lock held.
367 struct packet *playrtp_next_packet(void) {
368 while(pheap_count(&packets)) {
369 struct packet *const p = pheap_first(&packets);
370 if(le(p->timestamp + p->nsamples, next_timestamp)) {
371 /* This packet is in the past. Drop it and try another one. */
374 /* This packet is NOT in the past. (It might be in the future
381 /** @brief Play an RTP stream
383 * This is the guts of the program. It is responsible for:
384 * - starting the listening thread
385 * - opening the audio device
386 * - reading ahead to build up a buffer
387 * - arranging for audio to be played
388 * - detecting when the buffer has got too small and re-buffering
390 static void play_rtp(void) {
393 /* We receive and convert audio data in a background thread */
394 pthread_create(<id, 0, listen_thread, 0);
395 /* We have a second thread to add received packets to the queue */
396 pthread_create(<id, 0, queue_thread, 0);
397 /* The rest of the work is backend-specific */
401 /* display usage message and terminate */
402 static void help(void) {
404 " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
406 " --device, -D DEVICE Output device\n"
407 " --min, -m FRAMES Buffer low water mark\n"
408 " --buffer, -b FRAMES Buffer high water mark\n"
409 " --max, -x FRAMES Buffer maximum size\n"
410 " --rcvbuf, -R BYTES Socket receive buffer size\n"
411 " --multicast, -M GROUP Join multicast group\n"
412 " --config, -C PATH Set configuration file\n"
413 #if HAVE_ALSA_ASOUNDLIB_H
414 " --alsa, -a Use ALSA to play audio\n"
416 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
417 " --oss, -o Use OSS to play audio\n"
419 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
420 " --core-audio, -c Use Core Audio to play audio\n"
422 " --help, -h Display usage message\n"
423 " --version, -V Display version number\n"
429 /* display version number and terminate */
430 static void version(void) {
431 xprintf("disorder-playrtp version %s\n", disorder_version_string);
436 int main(int argc, char **argv) {
438 struct addrinfo *res;
439 struct stringlist sl;
441 int rcvbuf, target_rcvbuf = 131072;
443 char *multicast_group = 0;
445 struct ipv6_mreq mreq6;
447 char *address, *port;
449 static const struct addrinfo prefs = {
461 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
462 while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:aocC:", options, 0)) >= 0) {
466 case 'd': debugging = 1; break;
467 case 'D': device = optarg; break;
468 case 'm': minbuffer = 2 * atol(optarg); break;
469 case 'b': readahead = 2 * atol(optarg); break;
470 case 'x': maxbuffer = 2 * atol(optarg); break;
471 case 'L': logfp = fopen(optarg, "w"); break;
472 case 'R': target_rcvbuf = atoi(optarg); break;
473 case 'M': multicast_group = optarg; break;
474 #if HAVE_ALSA_ASOUNDLIB_H
475 case 'a': backend = playrtp_alsa; break;
477 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
478 case 'o': backend = playrtp_oss; break;
480 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
481 case 'c': backend = playrtp_coreaudio; break;
483 case 'C': configfile = optarg; break;
484 default: fatal(0, "invalid option");
487 if(config_read(0)) fatal(0, "cannot read configuration");
489 maxbuffer = 4 * readahead;
495 if(!(c = disorder_new(1))) exit(EXIT_FAILURE);
496 if(disorder_connect(c)) exit(EXIT_FAILURE);
497 if(disorder_rtp_address(c, &address, &port)) exit(EXIT_FAILURE);
500 /* set multicast_group if address is a multicast address */
507 fatal(0, "usage: disorder-playrtp [OPTIONS] [ADDRESS [PORT]]");
509 /* Listen for inbound audio data */
510 if(!(res = get_address(&sl, &prefs, &sockname)))
512 info("listening on %s", sockname);
513 if((rtpfd = socket(res->ai_family,
515 res->ai_protocol)) < 0)
516 fatal(errno, "error creating socket");
517 if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
518 fatal(errno, "error binding socket to %s", sockname);
519 if(multicast_group) {
520 if((n = getaddrinfo(multicast_group, 0, &prefs, &res)))
521 fatal(0, "getaddrinfo %s: %s", multicast_group, gai_strerror(n));
522 switch(res->ai_family) {
524 mreq.imr_multiaddr = ((struct sockaddr_in *)res->ai_addr)->sin_addr;
525 mreq.imr_interface.s_addr = 0; /* use primary interface */
526 if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP,
527 &mreq, sizeof mreq) < 0)
528 fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP");
531 mreq6.ipv6mr_multiaddr = ((struct sockaddr_in6 *)res->ai_addr)->sin6_addr;
532 memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface);
533 if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP,
534 &mreq6, sizeof mreq6) < 0)
535 fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP");
538 fatal(0, "unsupported address family %d", res->ai_family);
542 if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0)
543 fatal(errno, "error calling getsockopt SO_RCVBUF");
544 if(target_rcvbuf > rcvbuf) {
545 if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF,
546 &target_rcvbuf, sizeof target_rcvbuf) < 0)
547 error(errno, "error calling setsockopt SO_RCVBUF %d",
549 /* We try to carry on anyway */
551 info("changed socket receive buffer from %d to %d",
552 rcvbuf, target_rcvbuf);
554 info("default socket receive buffer %d", rcvbuf);
556 info("WARNING: -L option can impact performance");