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
;
165 # define DEFAULT_BACKEND playrtp_alsa
166 #elif HAVE_SYS_SOUNDCARD_H
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
190 { "oss", no_argument
, 0, 'o' },
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"
411 " --alsa, -a Use ALSA to play audio\n"
413 #if HAVE_SYS_SOUNDCARD_H
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;
470 case 'a': backend
= playrtp_alsa
; break;
473 #if HAVE_SYS_SOUNDCARD_H
474 case 'o': backend
= playrtp_oss
; break;
476 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
477 case 'c': backend
= playrtp_coreaudio
; break;
480 default: fatal(0, "invalid option");
484 maxbuffer
= 4 * readahead
;
487 if(argc
< 1 || argc
> 2)
488 fatal(0, "usage: disorder-playrtp [OPTIONS] ADDRESS [PORT]");
491 /* Listen for inbound audio data */
492 if(!(res
= get_address(&sl
, &prefs
, &sockname
)))
494 if((rtpfd
= socket(res
->ai_family
,
496 res
->ai_protocol
)) < 0)
497 fatal(errno
, "error creating socket");
498 if(bind(rtpfd
, res
->ai_addr
, res
->ai_addrlen
) < 0)
499 fatal(errno
, "error binding socket to %s", sockname
);
500 if(multicast_group
) {
501 if((n
= getaddrinfo(multicast_group
, 0, &prefs
, &res
)))
502 fatal(0, "getaddrinfo %s: %s", multicast_group
, gai_strerror(n
));
503 switch(res
->ai_family
) {
505 mreq
.imr_multiaddr
= ((struct sockaddr_in
*)res
->ai_addr
)->sin_addr
;
506 mreq
.imr_interface
.s_addr
= 0; /* use primary interface */
507 if(setsockopt(rtpfd
, IPPROTO_IP
, IP_ADD_MEMBERSHIP
,
508 &mreq
, sizeof mreq
) < 0)
509 fatal(errno
, "error calling setsockopt IP_ADD_MEMBERSHIP");
512 mreq6
.ipv6mr_multiaddr
= ((struct sockaddr_in6
*)res
->ai_addr
)->sin6_addr
;
513 memset(&mreq6
.ipv6mr_interface
, 0, sizeof mreq6
.ipv6mr_interface
);
514 if(setsockopt(rtpfd
, IPPROTO_IPV6
, IPV6_JOIN_GROUP
,
515 &mreq6
, sizeof mreq6
) < 0)
516 fatal(errno
, "error calling setsockopt IPV6_JOIN_GROUP");
519 fatal(0, "unsupported address family %d", res
->ai_family
);
523 if(getsockopt(rtpfd
, SOL_SOCKET
, SO_RCVBUF
, &rcvbuf
, &len
) < 0)
524 fatal(errno
, "error calling getsockopt SO_RCVBUF");
525 if(target_rcvbuf
> rcvbuf
) {
526 if(setsockopt(rtpfd
, SOL_SOCKET
, SO_RCVBUF
,
527 &target_rcvbuf
, sizeof target_rcvbuf
) < 0)
528 error(errno
, "error calling setsockopt SO_RCVBUF %d",
530 /* We try to carry on anyway */
532 info("changed socket receive buffer from %d to %d",
533 rcvbuf
, target_rcvbuf
);
535 info("default socket receive buffer %d", rcvbuf
);
537 info("WARNING: -L option can impact performance");