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"
81 #define readahead linux_headers_are_borked
83 /** @brief RTP socket */
86 /** @brief Log output */
89 /** @brief Output device */
92 /** @brief Minimum low watermark
94 * We'll stop playing if there's only this many samples in the buffer. */
95 unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */
97 /** @brief Buffer high watermark
99 * We'll only start playing when this many samples are available. */
100 static unsigned readahead = 2 * 2 * 44100;
102 /** @brief Maximum buffer size
104 * We'll stop reading from the network if we have this many samples. */
105 static unsigned maxbuffer;
107 /** @brief Received packets
108 * Protected by @ref receive_lock
110 * Received packets are added to this list, and queue_thread() picks them off
111 * it and adds them to @ref packets. Whenever a packet is added to it, @ref
112 * receive_cond is signalled.
114 struct packet *received_packets;
116 /** @brief Tail of @ref received_packets
117 * Protected by @ref receive_lock
119 struct packet **received_tail = &received_packets;
121 /** @brief Lock protecting @ref received_packets
123 * Only listen_thread() and queue_thread() ever hold this lock. It is vital
124 * that queue_thread() not hold it any longer than it strictly has to. */
125 pthread_mutex_t receive_lock = PTHREAD_MUTEX_INITIALIZER;
127 /** @brief Condition variable signalled when @ref received_packets is updated
129 * Used by listen_thread() to notify queue_thread() that it has added another
130 * packet to @ref received_packets. */
131 pthread_cond_t receive_cond = PTHREAD_COND_INITIALIZER;
133 /** @brief Length of @ref received_packets */
136 /** @brief Binary heap of received packets */
137 struct pheap packets;
139 /** @brief Total number of samples available
141 * We make this volatile because we inspect it without a protecting lock,
142 * so the usual pthread_* guarantees aren't available.
144 volatile uint32_t nsamples;
146 /** @brief Timestamp of next packet to play.
148 * This is set to the timestamp of the last packet, plus the number of
149 * samples it contained. Only valid if @ref active is nonzero.
151 uint32_t next_timestamp;
153 /** @brief True if actively playing
155 * This is true when playing and false when just buffering. */
158 /** @brief Lock protecting @ref packets */
159 pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
161 /** @brief Condition variable signalled whenever @ref packets is changed */
162 pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
164 #if HAVE_ALSA_ASOUNDLIB_H
165 # define DEFAULT_BACKEND playrtp_alsa
166 #elif HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
167 # define DEFAULT_BACKEND playrtp_oss
168 #elif HAVE_COREAUDIO_AUDIOHARDWARE_H
169 # define DEFAULT_BACKEND playrtp_coreaudio
171 # error No known backend
174 /** @brief Backend to play with */
175 static void (*backend)(void) = &DEFAULT_BACKEND;
177 HEAP_DEFINE(pheap, struct packet *, lt_packet);
179 static const struct option options[] = {
180 { "help", no_argument, 0, 'h' },
181 { "version", no_argument, 0, 'V' },
182 { "debug", no_argument, 0, 'd' },
183 { "device", required_argument, 0, 'D' },
184 { "min", required_argument, 0, 'm' },
185 { "max", required_argument, 0, 'x' },
186 { "buffer", required_argument, 0, 'b' },
187 { "rcvbuf", required_argument, 0, 'R' },
188 { "multicast", required_argument, 0, 'M' },
189 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
190 { "oss", no_argument, 0, 'o' },
192 #if HAVE_ALSA_ASOUNDLIB_H
193 { "alsa", no_argument, 0, 'a' },
195 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
196 { "core-audio", no_argument, 0, 'c' },
201 /** @brief Drop the first packet
203 * Assumes that @ref lock is held.
205 static void drop_first_packet(void) {
206 if(pheap_count(&packets)) {
207 struct packet *const p = pheap_remove(&packets);
208 nsamples -= p->nsamples;
209 playrtp_free_packet(p);
210 pthread_cond_broadcast(&cond);
214 /** @brief Background thread adding packets to heap
216 * This just transfers packets from @ref received_packets to @ref packets. It
217 * is important that it holds @ref receive_lock for as little time as possible,
218 * in order to minimize the interval between calls to read() in
221 static void *queue_thread(void attribute((unused)) *arg) {
225 /* Get the next packet */
226 pthread_mutex_lock(&receive_lock);
227 while(!received_packets)
228 pthread_cond_wait(&receive_cond, &receive_lock);
229 p = received_packets;
230 received_packets = p->next;
231 if(!received_packets)
232 received_tail = &received_packets;
234 pthread_mutex_unlock(&receive_lock);
235 /* Add it to the heap */
236 pthread_mutex_lock(&lock);
237 pheap_insert(&packets, p);
238 nsamples += p->nsamples;
239 pthread_cond_broadcast(&cond);
240 pthread_mutex_unlock(&lock);
244 /** @brief Background thread collecting samples
246 * This function collects samples, perhaps converts them to the target format,
247 * and adds them to the packet list.
249 * It is crucial that the gap between successive calls to read() is as small as
250 * possible: otherwise packets will be dropped.
252 * We use a binary heap to ensure that the unavoidable effort is at worst
253 * logarithmic in the total number of packets - in fact if packets are mostly
254 * received in order then we will largely do constant work per packet since the
255 * newest packet will always be last.
257 * Of more concern is that we must acquire the lock on the heap to add a packet
258 * to it. If this proves a problem in practice then the answer would be
259 * (probably doubly) linked list with new packets added the end and a second
260 * thread which reads packets off the list and adds them to the heap.
262 * We keep memory allocation (mostly) very fast by keeping pre-allocated
263 * packets around; see @ref playrtp_new_packet().
265 static void *listen_thread(void attribute((unused)) *arg) {
266 struct packet *p = 0;
268 struct rtp_header header;
275 p = playrtp_new_packet();
276 iov[0].iov_base = &header;
277 iov[0].iov_len = sizeof header;
278 iov[1].iov_base = p->samples_raw;
279 iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw;
280 n = readv(rtpfd, iov, 2);
286 fatal(errno, "error reading from socket");
289 /* Ignore too-short packets */
290 if((size_t)n <= sizeof (struct rtp_header)) {
291 info("ignored a short packet");
294 timestamp = htonl(header.timestamp);
295 seq = htons(header.seq);
296 /* Ignore packets in the past */
297 if(active && lt(timestamp, next_timestamp)) {
298 info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
299 timestamp, next_timestamp);
304 p->timestamp = timestamp;
305 /* Convert to target format */
306 if(header.mpt & 0x80)
308 switch(header.mpt & 0x7F) {
310 p->nsamples = (n - sizeof header) / sizeof(uint16_t);
312 /* TODO support other RFC3551 media types (when the speaker does) */
314 fatal(0, "unsupported RTP payload type %d",
318 fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
319 seq, timestamp, p->nsamples, timestamp + p->nsamples);
320 /* Stop reading if we've reached the maximum.
322 * This is rather unsatisfactory: it means that if packets get heavily
323 * out of order then we guarantee dropouts. But for now... */
324 if(nsamples >= maxbuffer) {
325 pthread_mutex_lock(&lock);
326 while(nsamples >= maxbuffer)
327 pthread_cond_wait(&cond, &lock);
328 pthread_mutex_unlock(&lock);
330 /* Add the packet to the receive queue */
331 pthread_mutex_lock(&receive_lock);
333 received_tail = &p->next;
335 pthread_cond_signal(&receive_cond);
336 pthread_mutex_unlock(&receive_lock);
337 /* We'll need a new packet */
342 /** @brief Wait until the buffer is adequately full
344 * Must be called with @ref lock held.
346 void playrtp_fill_buffer(void) {
349 info("Buffering...");
350 while(nsamples < readahead)
351 pthread_cond_wait(&cond, &lock);
352 next_timestamp = pheap_first(&packets)->timestamp;
356 /** @brief Find next packet
357 * @return Packet to play or NULL if none found
359 * The return packet is merely guaranteed not to be in the past: it might be
360 * the first packet in the future rather than one that is actually suitable to
363 * Must be called with @ref lock held.
365 struct packet *playrtp_next_packet(void) {
366 while(pheap_count(&packets)) {
367 struct packet *const p = pheap_first(&packets);
368 if(le(p->timestamp + p->nsamples, next_timestamp)) {
369 /* This packet is in the past. Drop it and try another one. */
372 /* This packet is NOT in the past. (It might be in the future
379 /** @brief Play an RTP stream
381 * This is the guts of the program. It is responsible for:
382 * - starting the listening thread
383 * - opening the audio device
384 * - reading ahead to build up a buffer
385 * - arranging for audio to be played
386 * - detecting when the buffer has got too small and re-buffering
388 static void play_rtp(void) {
391 /* We receive and convert audio data in a background thread */
392 pthread_create(<id, 0, listen_thread, 0);
393 /* We have a second thread to add received packets to the queue */
394 pthread_create(<id, 0, queue_thread, 0);
395 /* The rest of the work is backend-specific */
399 /* display usage message and terminate */
400 static void help(void) {
402 " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n"
404 " --device, -D DEVICE Output device\n"
405 " --min, -m FRAMES Buffer low water mark\n"
406 " --buffer, -b FRAMES Buffer high water mark\n"
407 " --max, -x FRAMES Buffer maximum size\n"
408 " --rcvbuf, -R BYTES Socket receive buffer size\n"
409 " --multicast, -M GROUP Join multicast group\n"
410 #if HAVE_ALSA_ASOUNDLIB_H
411 " --alsa, -a Use ALSA to play audio\n"
413 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
414 " --oss, -o Use OSS to play audio\n"
416 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
417 " --core-audio, -c Use Core Audio to play audio\n"
419 " --help, -h Display usage message\n"
420 " --version, -V Display version number\n"
426 /* display version number and terminate */
427 static void version(void) {
428 xprintf("disorder-playrtp version %s\n", disorder_version_string);
433 int main(int argc, char **argv) {
435 struct addrinfo *res;
436 struct stringlist sl;
438 int rcvbuf, target_rcvbuf = 131072;
440 char *multicast_group = 0;
442 struct ipv6_mreq mreq6;
444 static const struct addrinfo prefs = {
456 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
457 while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:aoc", options, 0)) >= 0) {
461 case 'd': debugging = 1; break;
462 case 'D': device = optarg; break;
463 case 'm': minbuffer = 2 * atol(optarg); break;
464 case 'b': readahead = 2 * atol(optarg); break;
465 case 'x': maxbuffer = 2 * atol(optarg); break;
466 case 'L': logfp = fopen(optarg, "w"); break;
467 case 'R': target_rcvbuf = atoi(optarg); break;
468 case 'M': multicast_group = optarg; break;
469 #if HAVE_ALSA_ASOUNDLIB_H
470 case 'a': backend = playrtp_alsa; break;
472 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
473 case 'o': backend = playrtp_oss; break;
475 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
476 case 'c': backend = playrtp_coreaudio; break;
478 default: fatal(0, "invalid option");
482 maxbuffer = 4 * readahead;
485 if(argc < 1 || argc > 2)
486 fatal(0, "usage: disorder-playrtp [OPTIONS] ADDRESS [PORT]");
489 /* Listen for inbound audio data */
490 if(!(res = get_address(&sl, &prefs, &sockname)))
492 if((rtpfd = socket(res->ai_family,
494 res->ai_protocol)) < 0)
495 fatal(errno, "error creating socket");
496 if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
497 fatal(errno, "error binding socket to %s", sockname);
498 if(multicast_group) {
499 if((n = getaddrinfo(multicast_group, 0, &prefs, &res)))
500 fatal(0, "getaddrinfo %s: %s", multicast_group, gai_strerror(n));
501 switch(res->ai_family) {
503 mreq.imr_multiaddr = ((struct sockaddr_in *)res->ai_addr)->sin_addr;
504 mreq.imr_interface.s_addr = 0; /* use primary interface */
505 if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP,
506 &mreq, sizeof mreq) < 0)
507 fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP");
510 mreq6.ipv6mr_multiaddr = ((struct sockaddr_in6 *)res->ai_addr)->sin6_addr;
511 memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface);
512 if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP,
513 &mreq6, sizeof mreq6) < 0)
514 fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP");
517 fatal(0, "unsupported address family %d", res->ai_family);
521 if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0)
522 fatal(errno, "error calling getsockopt SO_RCVBUF");
523 if(target_rcvbuf > rcvbuf) {
524 if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF,
525 &target_rcvbuf, sizeof target_rcvbuf) < 0)
526 error(errno, "error calling setsockopt SO_RCVBUF %d",
528 /* We try to carry on anyway */
530 info("changed socket receive buffer from %d to %d",
531 rcvbuf, target_rcvbuf);
533 info("default socket receive buffer %d", rcvbuf);
535 info("WARNING: -L option can impact performance");