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 3 of the License, or
8 * (at your option) any later version.
10 * This program is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
13 * GNU 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, see <http://www.gnu.org/licenses/>.
18 /** @file clients/playrtp.c
21 * This player supports Linux (<a href="http://www.alsa-project.org/">ALSA</a>)
23 * href="http://developer.apple.com/audio/coreaudio.html">Core Audio</a>)
24 * systems. There is no support for Microsoft Windows yet, and that will in
25 * fact probably an entirely separate program.
27 * The program runs (at least) three threads. listen_thread() is responsible
28 * for reading RTP packets off the wire and adding them to the linked list @ref
29 * received_packets, assuming they are basically sound. queue_thread() takes
30 * packets off this linked list and adds them to @ref packets (an operation
31 * which might be much slower due to contention for @ref lock).
33 * The main thread is responsible for actually playing audio. In ALSA this
34 * means it waits until ALSA says it's ready for more audio which it then
35 * plays. See @ref clients/playrtp-alsa.c.
37 * In Core Audio the main thread is only responsible for starting and stopping
38 * play: the system does the actual playback in its own private thread, and
39 * calls adioproc() to fetch the audio data. See @ref
40 * clients/playrtp-coreaudio.c.
42 * Sometimes it happens that there is no audio available to play. This may
43 * because the server went away, or a packet was dropped, or the server
44 * deliberately did not send any sound because it encountered a silence.
47 * - it is safe to read uint32_t values without a lock protecting them
53 #include <sys/socket.h>
54 #include <sys/types.h>
55 #include <sys/socket.h>
61 #include <netinet/in.h>
70 #include "configuration.h"
80 #include "inputline.h"
84 #define readahead linux_headers_are_borked
86 /** @brief Obsolete synonym */
87 #ifndef IPV6_JOIN_GROUP
88 # define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP
91 /** @brief RTP socket */
94 /** @brief Log output */
97 /** @brief Output device */
99 /** @brief Minimum low watermark
101 * We'll stop playing if there's only this many samples in the buffer. */
102 unsigned minbuffer
= 2 * 44100 / 10; /* 0.2 seconds */
104 /** @brief Buffer high watermark
106 * We'll only start playing when this many samples are available. */
107 static unsigned readahead
= 2 * 2 * 44100;
109 /** @brief Maximum buffer size
111 * We'll stop reading from the network if we have this many samples. */
112 static unsigned maxbuffer
;
114 /** @brief Received packets
115 * Protected by @ref receive_lock
117 * Received packets are added to this list, and queue_thread() picks them off
118 * it and adds them to @ref packets. Whenever a packet is added to it, @ref
119 * receive_cond is signalled.
121 struct packet
*received_packets
;
123 /** @brief Tail of @ref received_packets
124 * Protected by @ref receive_lock
126 struct packet
**received_tail
= &received_packets
;
128 /** @brief Lock protecting @ref received_packets
130 * Only listen_thread() and queue_thread() ever hold this lock. It is vital
131 * that queue_thread() not hold it any longer than it strictly has to. */
132 pthread_mutex_t receive_lock
= PTHREAD_MUTEX_INITIALIZER
;
134 /** @brief Condition variable signalled when @ref received_packets is updated
136 * Used by listen_thread() to notify queue_thread() that it has added another
137 * packet to @ref received_packets. */
138 pthread_cond_t receive_cond
= PTHREAD_COND_INITIALIZER
;
140 /** @brief Length of @ref received_packets */
143 /** @brief Binary heap of received packets */
144 struct pheap packets
;
146 /** @brief Total number of samples available
148 * We make this volatile because we inspect it without a protecting lock,
149 * so the usual pthread_* guarantees aren't available.
151 volatile uint32_t nsamples
;
153 /** @brief Timestamp of next packet to play.
155 * This is set to the timestamp of the last packet, plus the number of
156 * samples it contained. Only valid if @ref active is nonzero.
158 uint32_t next_timestamp
;
160 /** @brief True if actively playing
162 * This is true when playing and false when just buffering. */
165 /** @brief Lock protecting @ref packets */
166 pthread_mutex_t lock
= PTHREAD_MUTEX_INITIALIZER
;
168 /** @brief Condition variable signalled whenever @ref packets is changed */
169 pthread_cond_t cond
= PTHREAD_COND_INITIALIZER
;
171 /** @brief Backend to play with */
172 static const struct uaudio
*backend
;
174 HEAP_DEFINE(pheap
, struct packet
*, lt_packet
);
176 /** @brief Control socket or NULL */
177 const char *control_socket
;
179 /** @brief Buffer for debugging dump
181 * The debug dump is enabled by the @c --dump option. It records the last 20s
182 * of audio to the specified file (which will be about 3.5Mbytes). The file is
183 * written as as ring buffer, so the start point will progress through it.
185 * Use clients/dump2wav to convert this to a WAV file, which can then be loaded
186 * into (e.g.) Audacity for further inspection.
188 * All three backends (ALSA, OSS, Core Audio) now support this option.
190 * The idea is to allow the user a few seconds to react to an audible artefact.
192 int16_t *dump_buffer
;
194 /** @brief Current index within debugging dump */
197 /** @brief Size of debugging dump in samples */
198 size_t dump_size
= 44100/*Hz*/ * 2/*channels*/ * 20/*seconds*/;
200 static const struct option options
[] = {
201 { "help", no_argument
, 0, 'h' },
202 { "version", no_argument
, 0, 'V' },
203 { "debug", no_argument
, 0, 'd' },
204 { "device", required_argument
, 0, 'D' },
205 { "min", required_argument
, 0, 'm' },
206 { "max", required_argument
, 0, 'x' },
207 { "buffer", required_argument
, 0, 'b' },
208 { "rcvbuf", required_argument
, 0, 'R' },
209 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
210 { "oss", no_argument
, 0, 'o' },
212 #if HAVE_ALSA_ASOUNDLIB_H
213 { "alsa", no_argument
, 0, 'a' },
215 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
216 { "core-audio", no_argument
, 0, 'c' },
218 { "dump", required_argument
, 0, 'r' },
219 { "socket", required_argument
, 0, 's' },
220 { "config", required_argument
, 0, 'C' },
224 /** @brief Control thread
226 * This thread is responsible for accepting control commands from Disobedience
227 * (or other controllers) over an AF_UNIX stream socket with a path specified
228 * by the @c --socket option. The protocol uses simple string commands and
231 * - @c stop will shut the player down
232 * - @c query will send back the reply @c running
233 * - anything else is ignored
235 * Commands and response strings terminated by shutting down the connection or
236 * by a newline. No attempt is made to multiplex multiple clients so it is
237 * important that the command be sent as soon as the connection is made - it is
238 * assumed that both parties to the protocol are entirely cooperating with one
241 static void *control_thread(void attribute((unused
)) *arg
) {
242 struct sockaddr_un sa
;
248 assert(control_socket
);
249 unlink(control_socket
);
250 memset(&sa
, 0, sizeof sa
);
251 sa
.sun_family
= AF_UNIX
;
252 strcpy(sa
.sun_path
, control_socket
);
253 sfd
= xsocket(PF_UNIX
, SOCK_STREAM
, 0);
254 if(bind(sfd
, (const struct sockaddr
*)&sa
, sizeof sa
) < 0)
255 fatal(errno
, "error binding to %s", control_socket
);
256 if(listen(sfd
, 128) < 0)
257 fatal(errno
, "error calling listen on %s", control_socket
);
258 info("listening on %s", control_socket
);
261 cfd
= accept(sfd
, (struct sockaddr
*)&sa
, &salen
);
268 fatal(errno
, "error calling accept on %s", control_socket
);
271 if(!(fp
= fdopen(cfd
, "r+"))) {
272 error(errno
, "error calling fdopen for %s connection", control_socket
);
276 if(!inputline(control_socket
, fp
, &line
, '\n')) {
277 if(!strcmp(line
, "stop")) {
278 info("stopped via %s", control_socket
);
279 exit(0); /* terminate immediately */
281 if(!strcmp(line
, "query"))
282 fprintf(fp
, "running");
286 error(errno
, "error closing %s connection", control_socket
);
290 /** @brief Drop the first packet
292 * Assumes that @ref lock is held.
294 static void drop_first_packet(void) {
295 if(pheap_count(&packets
)) {
296 struct packet
*const p
= pheap_remove(&packets
);
297 nsamples
-= p
->nsamples
;
298 playrtp_free_packet(p
);
299 pthread_cond_broadcast(&cond
);
303 /** @brief Background thread adding packets to heap
305 * This just transfers packets from @ref received_packets to @ref packets. It
306 * is important that it holds @ref receive_lock for as little time as possible,
307 * in order to minimize the interval between calls to read() in
310 static void *queue_thread(void attribute((unused
)) *arg
) {
314 /* Get the next packet */
315 pthread_mutex_lock(&receive_lock
);
316 while(!received_packets
) {
317 pthread_cond_wait(&receive_cond
, &receive_lock
);
319 p
= received_packets
;
320 received_packets
= p
->next
;
321 if(!received_packets
)
322 received_tail
= &received_packets
;
324 pthread_mutex_unlock(&receive_lock
);
325 /* Add it to the heap */
326 pthread_mutex_lock(&lock
);
327 pheap_insert(&packets
, p
);
328 nsamples
+= p
->nsamples
;
329 pthread_cond_broadcast(&cond
);
330 pthread_mutex_unlock(&lock
);
334 /** @brief Background thread collecting samples
336 * This function collects samples, perhaps converts them to the target format,
337 * and adds them to the packet list.
339 * It is crucial that the gap between successive calls to read() is as small as
340 * possible: otherwise packets will be dropped.
342 * We use a binary heap to ensure that the unavoidable effort is at worst
343 * logarithmic in the total number of packets - in fact if packets are mostly
344 * received in order then we will largely do constant work per packet since the
345 * newest packet will always be last.
347 * Of more concern is that we must acquire the lock on the heap to add a packet
348 * to it. If this proves a problem in practice then the answer would be
349 * (probably doubly) linked list with new packets added the end and a second
350 * thread which reads packets off the list and adds them to the heap.
352 * We keep memory allocation (mostly) very fast by keeping pre-allocated
353 * packets around; see @ref playrtp_new_packet().
355 static void *listen_thread(void attribute((unused
)) *arg
) {
356 struct packet
*p
= 0;
358 struct rtp_header header
;
365 p
= playrtp_new_packet();
366 iov
[0].iov_base
= &header
;
367 iov
[0].iov_len
= sizeof header
;
368 iov
[1].iov_base
= p
->samples_raw
;
369 iov
[1].iov_len
= sizeof p
->samples_raw
/ sizeof *p
->samples_raw
;
370 n
= readv(rtpfd
, iov
, 2);
376 fatal(errno
, "error reading from socket");
379 /* Ignore too-short packets */
380 if((size_t)n
<= sizeof (struct rtp_header
)) {
381 info("ignored a short packet");
384 timestamp
= htonl(header
.timestamp
);
385 seq
= htons(header
.seq
);
386 /* Ignore packets in the past */
387 if(active
&& lt(timestamp
, next_timestamp
)) {
388 info("dropping old packet, timestamp=%"PRIx32
" < %"PRIx32
,
389 timestamp
, next_timestamp
);
392 /* Ignore packets with the extension bit set. */
393 if(header
.vpxcc
& 0x10)
397 p
->timestamp
= timestamp
;
398 /* Convert to target format */
399 if(header
.mpt
& 0x80)
401 switch(header
.mpt
& 0x7F) {
403 p
->nsamples
= (n
- sizeof header
) / sizeof(uint16_t);
405 /* TODO support other RFC3551 media types (when the speaker does) */
407 fatal(0, "unsupported RTP payload type %d",
411 fprintf(logfp
, "sequence %u timestamp %"PRIx32
" length %"PRIx32
" end %"PRIx32
"\n",
412 seq
, timestamp
, p
->nsamples
, timestamp
+ p
->nsamples
);
413 /* Stop reading if we've reached the maximum.
415 * This is rather unsatisfactory: it means that if packets get heavily
416 * out of order then we guarantee dropouts. But for now... */
417 if(nsamples
>= maxbuffer
) {
418 pthread_mutex_lock(&lock
);
419 while(nsamples
>= maxbuffer
) {
420 pthread_cond_wait(&cond
, &lock
);
422 pthread_mutex_unlock(&lock
);
424 /* Add the packet to the receive queue */
425 pthread_mutex_lock(&receive_lock
);
427 received_tail
= &p
->next
;
429 pthread_cond_signal(&receive_cond
);
430 pthread_mutex_unlock(&receive_lock
);
431 /* We'll need a new packet */
436 /** @brief Wait until the buffer is adequately full
438 * Must be called with @ref lock held.
440 void playrtp_fill_buffer(void) {
443 info("Buffering...");
444 while(nsamples
< readahead
) {
445 pthread_cond_wait(&cond
, &lock
);
447 next_timestamp
= pheap_first(&packets
)->timestamp
;
451 /** @brief Find next packet
452 * @return Packet to play or NULL if none found
454 * The return packet is merely guaranteed not to be in the past: it might be
455 * the first packet in the future rather than one that is actually suitable to
458 * Must be called with @ref lock held.
460 struct packet
*playrtp_next_packet(void) {
461 while(pheap_count(&packets
)) {
462 struct packet
*const p
= pheap_first(&packets
);
463 if(le(p
->timestamp
+ p
->nsamples
, next_timestamp
)) {
464 /* This packet is in the past. Drop it and try another one. */
467 /* This packet is NOT in the past. (It might be in the future
474 /* display usage message and terminate */
475 static void help(void) {
477 " disorder-playrtp [OPTIONS] [[ADDRESS] PORT]\n"
479 " --device, -D DEVICE Output device\n"
480 " --min, -m FRAMES Buffer low water mark\n"
481 " --buffer, -b FRAMES Buffer high water mark\n"
482 " --max, -x FRAMES Buffer maximum size\n"
483 " --rcvbuf, -R BYTES Socket receive buffer size\n"
484 " --config, -C PATH Set configuration file\n"
485 #if HAVE_ALSA_ASOUNDLIB_H
486 " --alsa, -a Use ALSA to play audio\n"
488 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
489 " --oss, -o Use OSS to play audio\n"
491 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
492 " --core-audio, -c Use Core Audio to play audio\n"
494 " --help, -h Display usage message\n"
495 " --version, -V Display version number\n"
501 static size_t playrtp_callback(void *buffer
,
503 void attribute((unused
)) *userdata
) {
506 pthread_mutex_lock(&lock
);
507 /* Get the next packet, junking any that are now in the past */
508 const struct packet
*p
= playrtp_next_packet();
509 if(p
&& contains(p
, next_timestamp
)) {
510 /* This packet is ready to play; the desired next timestamp points
511 * somewhere into it. */
513 /* Timestamp of end of packet */
514 const uint32_t packet_end
= p
->timestamp
+ p
->nsamples
;
516 /* Offset of desired next timestamp into current packet */
517 const uint32_t offset
= next_timestamp
- p
->timestamp
;
519 /* Pointer to audio data */
520 const uint16_t *ptr
= (void *)(p
->samples_raw
+ offset
);
522 /* Compute number of samples left in packet, limited to output buffer
524 samples
= packet_end
- next_timestamp
;
525 if(samples
> max_samples
)
526 samples
= max_samples
;
528 /* Copy into buffer, converting to native endianness */
530 int16_t *bufptr
= buffer
;
532 *bufptr
++ = (int16_t)ntohs(*ptr
++);
535 /* We don't junk the packet here; a subsequent call to
536 * playrtp_next_packet() will dispose of it (if it's actually done with). */
538 /* There is no suitable packet. We introduce 0s up to the next packet, or
539 * to fill the buffer if there's no next packet or that's too many. The
540 * comparison with max_samples deals with the otherwise troubling overflow
542 samples
= p ? p
->timestamp
- next_timestamp
: max_samples
;
543 if(samples
> max_samples
)
544 samples
= max_samples
;
545 //info("infill by %zu", samples);
546 memset(buffer
, 0, samples
* uaudio_sample_size
);
550 for(size_t i
= 0; i
< samples
; ++i
) {
551 dump_buffer
[dump_index
++] = ((int16_t *)buffer
)[i
];
552 dump_index
%= dump_size
;
555 /* Advance timestamp */
556 next_timestamp
+= samples
;
557 pthread_mutex_unlock(&lock
);
561 int main(int argc
, char **argv
) {
563 struct addrinfo
*res
;
564 struct stringlist sl
;
566 int rcvbuf
, target_rcvbuf
= 131072;
569 struct ipv6_mreq mreq6
;
571 char *address
, *port
;
575 struct sockaddr_in in
;
576 struct sockaddr_in6 in6
;
578 union any_sockaddr mgroup
;
579 const char *dumpfile
= 0;
580 const char *device
= 0;
583 static const struct addrinfo prefs
= {
584 .ai_flags
= AI_PASSIVE
,
585 .ai_family
= PF_INET
,
586 .ai_socktype
= SOCK_DGRAM
,
587 .ai_protocol
= IPPROTO_UDP
591 if(!setlocale(LC_CTYPE
, "")) fatal(errno
, "error calling setlocale");
592 backend
= uaudio_apis
[0];
593 while((n
= getopt_long(argc
, argv
, "hVdD:m:b:x:L:R:M:aocC:r", options
, 0)) >= 0) {
596 case 'V': version("disorder-playrtp");
597 case 'd': debugging
= 1; break;
598 case 'D': device
= optarg
; break;
599 case 'm': minbuffer
= 2 * atol(optarg
); break;
600 case 'b': readahead
= 2 * atol(optarg
); break;
601 case 'x': maxbuffer
= 2 * atol(optarg
); break;
602 case 'L': logfp
= fopen(optarg
, "w"); break;
603 case 'R': target_rcvbuf
= atoi(optarg
); break;
604 #if HAVE_ALSA_ASOUNDLIB_H
605 case 'a': backend
= &uaudio_alsa
; break;
607 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
608 case 'o': backend
= &uaudio_oss
; break;
610 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
611 case 'c': backend
= &uaudio_coreaudio
; break;
613 case 'C': configfile
= optarg
; break;
614 case 's': control_socket
= optarg
; break;
615 case 'r': dumpfile
= optarg
; break;
616 default: fatal(0, "invalid option");
619 if(config_read(0)) fatal(0, "cannot read configuration");
621 maxbuffer
= 4 * readahead
;
626 /* Get configuration from server */
627 if(!(c
= disorder_new(1))) exit(EXIT_FAILURE
);
628 if(disorder_connect(c
)) exit(EXIT_FAILURE
);
629 if(disorder_rtp_address(c
, &address
, &port
)) exit(EXIT_FAILURE
);
631 sl
.s
= xcalloc(2, sizeof *sl
.s
);
637 /* Use command-line ADDRESS+PORT or just PORT */
642 fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]");
644 /* Look up address and port */
645 if(!(res
= get_address(&sl
, &prefs
, &sockname
)))
647 /* Create the socket */
648 if((rtpfd
= socket(res
->ai_family
,
650 res
->ai_protocol
)) < 0)
651 fatal(errno
, "error creating socket");
652 /* Stash the multicast group address */
653 if((is_multicast
= multicast(res
->ai_addr
))) {
654 memcpy(&mgroup
, res
->ai_addr
, res
->ai_addrlen
);
655 switch(res
->ai_addr
->sa_family
) {
657 mgroup
.in
.sin_port
= 0;
660 mgroup
.in6
.sin6_port
= 0;
665 switch(res
->ai_addr
->sa_family
) {
667 memset(&((struct sockaddr_in
*)res
->ai_addr
)->sin_addr
, 0,
668 sizeof (struct in_addr
));
671 memset(&((struct sockaddr_in6
*)res
->ai_addr
)->sin6_addr
, 0,
672 sizeof (struct in6_addr
));
675 fatal(0, "unsupported family %d", (int)res
->ai_addr
->sa_family
);
677 if(bind(rtpfd
, res
->ai_addr
, res
->ai_addrlen
) < 0)
678 fatal(errno
, "error binding socket to %s", sockname
);
680 switch(mgroup
.sa
.sa_family
) {
682 mreq
.imr_multiaddr
= mgroup
.in
.sin_addr
;
683 mreq
.imr_interface
.s_addr
= 0; /* use primary interface */
684 if(setsockopt(rtpfd
, IPPROTO_IP
, IP_ADD_MEMBERSHIP
,
685 &mreq
, sizeof mreq
) < 0)
686 fatal(errno
, "error calling setsockopt IP_ADD_MEMBERSHIP");
689 mreq6
.ipv6mr_multiaddr
= mgroup
.in6
.sin6_addr
;
690 memset(&mreq6
.ipv6mr_interface
, 0, sizeof mreq6
.ipv6mr_interface
);
691 if(setsockopt(rtpfd
, IPPROTO_IPV6
, IPV6_JOIN_GROUP
,
692 &mreq6
, sizeof mreq6
) < 0)
693 fatal(errno
, "error calling setsockopt IPV6_JOIN_GROUP");
696 fatal(0, "unsupported address family %d", res
->ai_family
);
698 info("listening on %s multicast group %s",
699 format_sockaddr(res
->ai_addr
), format_sockaddr(&mgroup
.sa
));
701 info("listening on %s", format_sockaddr(res
->ai_addr
));
703 if(getsockopt(rtpfd
, SOL_SOCKET
, SO_RCVBUF
, &rcvbuf
, &len
) < 0)
704 fatal(errno
, "error calling getsockopt SO_RCVBUF");
705 if(target_rcvbuf
> rcvbuf
) {
706 if(setsockopt(rtpfd
, SOL_SOCKET
, SO_RCVBUF
,
707 &target_rcvbuf
, sizeof target_rcvbuf
) < 0)
708 error(errno
, "error calling setsockopt SO_RCVBUF %d",
710 /* We try to carry on anyway */
712 info("changed socket receive buffer from %d to %d",
713 rcvbuf
, target_rcvbuf
);
715 info("default socket receive buffer %d", rcvbuf
);
717 info("WARNING: -L option can impact performance");
721 if((err
= pthread_create(&tid
, 0, control_thread
, 0)))
722 fatal(err
, "pthread_create control_thread");
726 unsigned char buffer
[65536];
729 if((fd
= open(dumpfile
, O_RDWR
|O_TRUNC
|O_CREAT
, 0666)) < 0)
730 fatal(errno
, "opening %s", dumpfile
);
731 /* Fill with 0s to a suitable size */
732 memset(buffer
, 0, sizeof buffer
);
733 for(written
= 0; written
< dump_size
* sizeof(int16_t);
734 written
+= sizeof buffer
) {
735 if(write(fd
, buffer
, sizeof buffer
) < 0)
736 fatal(errno
, "clearing %s", dumpfile
);
738 /* Map the buffer into memory for convenience */
739 dump_buffer
= mmap(0, dump_size
* sizeof(int16_t), PROT_READ
|PROT_WRITE
,
741 if(dump_buffer
== (void *)-1)
742 fatal(errno
, "mapping %s", dumpfile
);
743 info("dumping to %s", dumpfile
);
745 /* Choose output device */
747 uaudio_set("device", device
);
748 /* Set up output. Currently we only support L16 so there's no harm setting
749 * the format before we know what it is! */
750 uaudio_set_format(44100/*Hz*/, 2/*channels*/,
751 16/*bits/channel*/, 1/*signed*/);
752 backend
->start(playrtp_callback
, NULL
);
753 /* We receive and convert audio data in a background thread */
754 if((err
= pthread_create(<id
, 0, listen_thread
, 0)))
755 fatal(err
, "pthread_create listen_thread");
756 /* We have a second thread to add received packets to the queue */
757 if((err
= pthread_create(<id
, 0, queue_thread
, 0)))
758 fatal(err
, "pthread_create queue_thread");
759 pthread_mutex_lock(&lock
);
761 /* Wait for the buffer to fill up a bit */
762 playrtp_fill_buffer();
763 /* Start playing now */
765 next_timestamp
= pheap_first(&packets
)->timestamp
;
768 /* Wait until the buffer empties out */
769 while(nsamples
>= minbuffer
771 && contains(pheap_first(&packets
), next_timestamp
))) {
772 pthread_cond_wait(&cond
, &lock
);
774 /* Stop playing for a bit until the buffer re-fills */
775 backend
->deactivate();