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:
29 * listen_thread() is responsible for reading RTP packets off the wire and
30 * adding them to the linked list @ref received_packets, assuming they are
33 * queue_thread() takes packets off this linked list and adds them to @ref
34 * packets (an operation which might be much slower due to contention for @ref
37 * control_thread() accepts commands from Disobedience (or anything else).
39 * The main thread activates and deactivates audio playing via the @ref
40 * lib/uaudio.h API (which probably implies at least one further thread).
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 /** @brief Obsolete synonym */
85 #ifndef IPV6_JOIN_GROUP
86 # define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP
89 /** @brief RTP socket */
92 /** @brief Log output */
95 /** @brief Output device */
97 /** @brief Buffer low watermark in samples */
98 unsigned minbuffer = 4 * (2 * 44100) / 10; /* 0.4 seconds */
100 /** @brief Maximum buffer size in samples
102 * We'll stop reading from the network if we have this many samples.
104 static unsigned maxbuffer;
106 /** @brief Received packets
107 * Protected by @ref receive_lock
109 * Received packets are added to this list, and queue_thread() picks them off
110 * it and adds them to @ref packets. Whenever a packet is added to it, @ref
111 * receive_cond is signalled.
113 struct packet *received_packets;
115 /** @brief Tail of @ref received_packets
116 * Protected by @ref receive_lock
118 struct packet **received_tail = &received_packets;
120 /** @brief Lock protecting @ref received_packets
122 * Only listen_thread() and queue_thread() ever hold this lock. It is vital
123 * that queue_thread() not hold it any longer than it strictly has to. */
124 pthread_mutex_t receive_lock = PTHREAD_MUTEX_INITIALIZER;
126 /** @brief Condition variable signalled when @ref received_packets is updated
128 * Used by listen_thread() to notify queue_thread() that it has added another
129 * packet to @ref received_packets. */
130 pthread_cond_t receive_cond = PTHREAD_COND_INITIALIZER;
132 /** @brief Length of @ref received_packets */
135 /** @brief Binary heap of received packets */
136 struct pheap packets;
138 /** @brief Total number of samples available
140 * We make this volatile because we inspect it without a protecting lock,
141 * so the usual pthread_* guarantees aren't available.
143 volatile uint32_t nsamples;
145 /** @brief Timestamp of next packet to play.
147 * This is set to the timestamp of the last packet, plus the number of
148 * samples it contained. Only valid if @ref active is nonzero.
150 uint32_t next_timestamp;
152 /** @brief True if actively playing
154 * This is true when playing and false when just buffering. */
157 /** @brief Lock protecting @ref packets */
158 pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER;
160 /** @brief Condition variable signalled whenever @ref packets is changed */
161 pthread_cond_t cond = PTHREAD_COND_INITIALIZER;
163 /** @brief Backend to play with */
164 static const struct uaudio *backend;
166 HEAP_DEFINE(pheap, struct packet *, lt_packet);
168 /** @brief Control socket or NULL */
169 const char *control_socket;
171 /** @brief Buffer for debugging dump
173 * The debug dump is enabled by the @c --dump option. It records the last 20s
174 * of audio to the specified file (which will be about 3.5Mbytes). The file is
175 * written as as ring buffer, so the start point will progress through it.
177 * Use clients/dump2wav to convert this to a WAV file, which can then be loaded
178 * into (e.g.) Audacity for further inspection.
180 * All three backends (ALSA, OSS, Core Audio) now support this option.
182 * The idea is to allow the user a few seconds to react to an audible artefact.
184 int16_t *dump_buffer;
186 /** @brief Current index within debugging dump */
189 /** @brief Size of debugging dump in samples */
190 size_t dump_size = 44100/*Hz*/ * 2/*channels*/ * 20/*seconds*/;
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 { "rcvbuf", required_argument, 0, 'R' },
200 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
201 { "oss", no_argument, 0, 'o' },
203 #if HAVE_ALSA_ASOUNDLIB_H
204 { "alsa", no_argument, 0, 'a' },
206 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
207 { "core-audio", no_argument, 0, 'c' },
209 { "dump", required_argument, 0, 'r' },
210 { "command", required_argument, 0, 'e' },
211 { "pause-mode", required_argument, 0, 'P' },
212 { "socket", required_argument, 0, 's' },
213 { "config", required_argument, 0, 'C' },
217 /** @brief Control thread
219 * This thread is responsible for accepting control commands from Disobedience
220 * (or other controllers) over an AF_UNIX stream socket with a path specified
221 * by the @c --socket option. The protocol uses simple string commands and
224 * - @c stop will shut the player down
225 * - @c query will send back the reply @c running
226 * - anything else is ignored
228 * Commands and response strings terminated by shutting down the connection or
229 * by a newline. No attempt is made to multiplex multiple clients so it is
230 * important that the command be sent as soon as the connection is made - it is
231 * assumed that both parties to the protocol are entirely cooperating with one
234 static void *control_thread(void attribute((unused)) *arg) {
235 struct sockaddr_un sa;
241 assert(control_socket);
242 unlink(control_socket);
243 memset(&sa, 0, sizeof sa);
244 sa.sun_family = AF_UNIX;
245 strcpy(sa.sun_path, control_socket);
246 sfd = xsocket(PF_UNIX, SOCK_STREAM, 0);
247 if(bind(sfd, (const struct sockaddr *)&sa, sizeof sa) < 0)
248 fatal(errno, "error binding to %s", control_socket);
249 if(listen(sfd, 128) < 0)
250 fatal(errno, "error calling listen on %s", control_socket);
251 info("listening on %s", control_socket);
254 cfd = accept(sfd, (struct sockaddr *)&sa, &salen);
261 fatal(errno, "error calling accept on %s", control_socket);
264 if(!(fp = fdopen(cfd, "r+"))) {
265 error(errno, "error calling fdopen for %s connection", control_socket);
269 if(!inputline(control_socket, fp, &line, '\n')) {
270 if(!strcmp(line, "stop")) {
271 info("stopped via %s", control_socket);
272 exit(0); /* terminate immediately */
274 if(!strcmp(line, "query"))
275 fprintf(fp, "running");
279 error(errno, "error closing %s connection", control_socket);
283 /** @brief Drop the first packet
285 * Assumes that @ref lock is held.
287 static void drop_first_packet(void) {
288 if(pheap_count(&packets)) {
289 struct packet *const p = pheap_remove(&packets);
290 nsamples -= p->nsamples;
291 playrtp_free_packet(p);
292 pthread_cond_broadcast(&cond);
296 /** @brief Background thread adding packets to heap
298 * This just transfers packets from @ref received_packets to @ref packets. It
299 * is important that it holds @ref receive_lock for as little time as possible,
300 * in order to minimize the interval between calls to read() in
303 static void *queue_thread(void attribute((unused)) *arg) {
307 /* Get the next packet */
308 pthread_mutex_lock(&receive_lock);
309 while(!received_packets) {
310 pthread_cond_wait(&receive_cond, &receive_lock);
312 p = received_packets;
313 received_packets = p->next;
314 if(!received_packets)
315 received_tail = &received_packets;
317 pthread_mutex_unlock(&receive_lock);
318 /* Add it to the heap */
319 pthread_mutex_lock(&lock);
320 pheap_insert(&packets, p);
321 nsamples += p->nsamples;
322 pthread_cond_broadcast(&cond);
323 pthread_mutex_unlock(&lock);
327 /** @brief Background thread collecting samples
329 * This function collects samples, perhaps converts them to the target format,
330 * and adds them to the packet list.
332 * It is crucial that the gap between successive calls to read() is as small as
333 * possible: otherwise packets will be dropped.
335 * We use a binary heap to ensure that the unavoidable effort is at worst
336 * logarithmic in the total number of packets - in fact if packets are mostly
337 * received in order then we will largely do constant work per packet since the
338 * newest packet will always be last.
340 * Of more concern is that we must acquire the lock on the heap to add a packet
341 * to it. If this proves a problem in practice then the answer would be
342 * (probably doubly) linked list with new packets added the end and a second
343 * thread which reads packets off the list and adds them to the heap.
345 * We keep memory allocation (mostly) very fast by keeping pre-allocated
346 * packets around; see @ref playrtp_new_packet().
348 static void *listen_thread(void attribute((unused)) *arg) {
349 struct packet *p = 0;
351 struct rtp_header header;
358 p = playrtp_new_packet();
359 iov[0].iov_base = &header;
360 iov[0].iov_len = sizeof header;
361 iov[1].iov_base = p->samples_raw;
362 iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw;
363 n = readv(rtpfd, iov, 2);
369 fatal(errno, "error reading from socket");
372 /* Ignore too-short packets */
373 if((size_t)n <= sizeof (struct rtp_header)) {
374 info("ignored a short packet");
377 timestamp = htonl(header.timestamp);
378 seq = htons(header.seq);
379 /* Ignore packets in the past */
380 if(active && lt(timestamp, next_timestamp)) {
381 info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
382 timestamp, next_timestamp);
385 /* Ignore packets with the extension bit set. */
386 if(header.vpxcc & 0x10)
390 p->timestamp = timestamp;
391 /* Convert to target format */
392 if(header.mpt & 0x80)
394 switch(header.mpt & 0x7F) {
396 p->nsamples = (n - sizeof header) / sizeof(uint16_t);
398 /* TODO support other RFC3551 media types (when the speaker does) */
400 fatal(0, "unsupported RTP payload type %d",
404 fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
405 seq, timestamp, p->nsamples, timestamp + p->nsamples);
406 /* Stop reading if we've reached the maximum.
408 * This is rather unsatisfactory: it means that if packets get heavily
409 * out of order then we guarantee dropouts. But for now... */
410 if(nsamples >= maxbuffer) {
411 pthread_mutex_lock(&lock);
412 while(nsamples >= maxbuffer) {
413 pthread_cond_wait(&cond, &lock);
415 pthread_mutex_unlock(&lock);
417 /* Add the packet to the receive queue */
418 pthread_mutex_lock(&receive_lock);
420 received_tail = &p->next;
422 pthread_cond_signal(&receive_cond);
423 pthread_mutex_unlock(&receive_lock);
424 /* We'll need a new packet */
429 /** @brief Wait until the buffer is adequately full
431 * Must be called with @ref lock held.
433 void playrtp_fill_buffer(void) {
434 /* Discard current buffer contents */
436 //fprintf(stderr, "%8u/%u (%u) DROPPING\n", nsamples, maxbuffer, minbuffer);
439 info("Buffering...");
440 /* Wait until there's at least minbuffer samples available */
441 while(nsamples < minbuffer) {
442 //fprintf(stderr, "%8u/%u (%u) FILLING\n", nsamples, maxbuffer, minbuffer);
443 pthread_cond_wait(&cond, &lock);
445 /* Start from whatever is earliest */
446 next_timestamp = pheap_first(&packets)->timestamp;
450 /** @brief Find next packet
451 * @return Packet to play or NULL if none found
453 * The return packet is merely guaranteed not to be in the past: it might be
454 * the first packet in the future rather than one that is actually suitable to
457 * Must be called with @ref lock held.
459 struct packet *playrtp_next_packet(void) {
460 while(pheap_count(&packets)) {
461 struct packet *const p = pheap_first(&packets);
462 if(le(p->timestamp + p->nsamples, next_timestamp)) {
463 /* This packet is in the past. Drop it and try another one. */
466 /* This packet is NOT in the past. (It might be in the future
473 /* display usage message and terminate */
474 static void help(void) {
476 " disorder-playrtp [OPTIONS] [[ADDRESS] PORT]\n"
478 " --device, -D DEVICE Output device\n"
479 " --min, -m FRAMES Buffer low water mark\n"
480 " --max, -x FRAMES Buffer maximum size\n"
481 " --rcvbuf, -R BYTES Socket receive buffer size\n"
482 " --config, -C PATH Set configuration file\n"
483 #if HAVE_ALSA_ASOUNDLIB_H
484 " --alsa, -a Use ALSA to play audio\n"
486 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
487 " --oss, -o Use OSS to play audio\n"
489 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
490 " --core-audio, -c Use Core Audio to play audio\n"
492 " --command, -e COMMAND Pipe audio to command.\n"
493 " --pause-mode, -P silence For -e: pauses send silence (default)\n"
494 " --pause-mode, -P suspend For -e: pauses suspend writes\n"
495 " --help, -h Display usage message\n"
496 " --version, -V Display version number\n"
502 static size_t playrtp_callback(void *buffer,
504 void attribute((unused)) *userdata) {
507 pthread_mutex_lock(&lock);
508 /* Get the next packet, junking any that are now in the past */
509 const struct packet *p = playrtp_next_packet();
510 if(p && contains(p, next_timestamp)) {
511 /* This packet is ready to play; the desired next timestamp points
512 * somewhere into it. */
514 /* Timestamp of end of packet */
515 const uint32_t packet_end = p->timestamp + p->nsamples;
517 /* Offset of desired next timestamp into current packet */
518 const uint32_t offset = next_timestamp - p->timestamp;
520 /* Pointer to audio data */
521 const uint16_t *ptr = (void *)(p->samples_raw + offset);
523 /* Compute number of samples left in packet, limited to output buffer
525 samples = packet_end - next_timestamp;
526 if(samples > max_samples)
527 samples = max_samples;
529 /* Copy into buffer, converting to native endianness */
531 int16_t *bufptr = buffer;
533 *bufptr++ = (int16_t)ntohs(*ptr++);
537 /* There is no suitable packet. We introduce 0s up to the next packet, or
538 * to fill the buffer if there's no next packet or that's too many. The
539 * comparison with max_samples deals with the otherwise troubling overflow
541 samples = p ? p->timestamp - next_timestamp : max_samples;
542 if(samples > max_samples)
543 samples = max_samples;
544 //info("infill by %zu", samples);
545 memset(buffer, 0, samples * uaudio_sample_size);
549 for(size_t i = 0; i < samples; ++i) {
550 dump_buffer[dump_index++] = ((int16_t *)buffer)[i];
551 dump_index %= dump_size;
554 /* Advance timestamp */
555 next_timestamp += samples;
556 /* Junk obsolete packets */
557 playrtp_next_packet();
558 pthread_mutex_unlock(&lock);
562 int main(int argc, char **argv) {
564 struct addrinfo *res;
565 struct stringlist sl;
567 int rcvbuf, target_rcvbuf = 0;
570 struct ipv6_mreq mreq6;
572 char *address, *port;
576 struct sockaddr_in in;
577 struct sockaddr_in6 in6;
579 union any_sockaddr mgroup;
580 const char *dumpfile = 0;
582 static const int one = 1;
584 static const struct addrinfo prefs = {
585 .ai_flags = AI_PASSIVE,
586 .ai_family = PF_INET,
587 .ai_socktype = SOCK_DGRAM,
588 .ai_protocol = IPPROTO_UDP
591 /* Timing information is often important to debugging playrtp, so we include
592 * timestamps in the logs */
595 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
596 backend = uaudio_apis[0];
597 while((n = getopt_long(argc, argv, "hVdD:m:x:L:R:M:aocC:re:P:", options, 0)) >= 0) {
600 case 'V': version("disorder-playrtp");
601 case 'd': debugging = 1; break;
602 case 'D': uaudio_set("device", optarg); break;
603 case 'm': minbuffer = 2 * atol(optarg); break;
604 case 'x': maxbuffer = 2 * atol(optarg); break;
605 case 'L': logfp = fopen(optarg, "w"); break;
606 case 'R': target_rcvbuf = atoi(optarg); break;
607 #if HAVE_ALSA_ASOUNDLIB_H
608 case 'a': backend = &uaudio_alsa; break;
610 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
611 case 'o': backend = &uaudio_oss; break;
613 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
614 case 'c': backend = &uaudio_coreaudio; break;
616 case 'C': configfile = optarg; break;
617 case 's': control_socket = optarg; break;
618 case 'r': dumpfile = optarg; break;
619 case 'e': backend = &uaudio_command; uaudio_set("command", optarg); break;
620 case 'P': uaudio_set("pause-mode", optarg); break;
621 default: fatal(0, "invalid option");
624 if(config_read(0)) fatal(0, "cannot read configuration");
626 maxbuffer = 2 * minbuffer;
631 /* Get configuration from server */
632 if(!(c = disorder_new(1))) exit(EXIT_FAILURE);
633 if(disorder_connect(c)) exit(EXIT_FAILURE);
634 if(disorder_rtp_address(c, &address, &port)) exit(EXIT_FAILURE);
636 sl.s = xcalloc(2, sizeof *sl.s);
642 /* Use command-line ADDRESS+PORT or just PORT */
647 fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]");
649 /* Look up address and port */
650 if(!(res = get_address(&sl, &prefs, &sockname)))
652 /* Create the socket */
653 if((rtpfd = socket(res->ai_family,
655 res->ai_protocol)) < 0)
656 fatal(errno, "error creating socket");
657 /* Allow multiple listeners */
658 xsetsockopt(rtpfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one);
659 is_multicast = multicast(res->ai_addr);
660 /* The multicast and unicast/broadcast cases are different enough that they
661 * are totally split. Trying to find commonality between them causes more
662 * trouble that it's worth. */
664 /* Stash the multicast group address */
665 memcpy(&mgroup, res->ai_addr, res->ai_addrlen);
666 switch(res->ai_addr->sa_family) {
668 mgroup.in.sin_port = 0;
671 mgroup.in6.sin6_port = 0;
674 fatal(0, "unsupported family %d", (int)res->ai_addr->sa_family);
676 /* Bind to to the multicast group address */
677 if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
678 fatal(errno, "error binding socket to %s", format_sockaddr(res->ai_addr));
679 /* Add multicast group membership */
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 /* Report what we did */
699 info("listening on %s multicast group %s",
700 format_sockaddr(res->ai_addr), format_sockaddr(&mgroup.sa));
703 switch(res->ai_addr->sa_family) {
705 struct sockaddr_in *in = (struct sockaddr_in *)res->ai_addr;
707 memset(&in->sin_addr, 0, sizeof (struct in_addr));
708 if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
709 fatal(errno, "error binding socket to 0.0.0.0 port %d",
710 ntohs(in->sin_port));
714 struct sockaddr_in6 *in6 = (struct sockaddr_in6 *)res->ai_addr;
716 memset(&in6->sin6_addr, 0, sizeof (struct in6_addr));
720 fatal(0, "unsupported family %d", (int)res->ai_addr->sa_family);
722 if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
723 fatal(errno, "error binding socket to %s", format_sockaddr(res->ai_addr));
724 /* Report what we did */
725 info("listening on %s", format_sockaddr(res->ai_addr));
728 if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0)
729 fatal(errno, "error calling getsockopt SO_RCVBUF");
730 if(target_rcvbuf > rcvbuf) {
731 if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF,
732 &target_rcvbuf, sizeof target_rcvbuf) < 0)
733 error(errno, "error calling setsockopt SO_RCVBUF %d",
735 /* We try to carry on anyway */
737 info("changed socket receive buffer from %d to %d",
738 rcvbuf, target_rcvbuf);
740 info("default socket receive buffer %d", rcvbuf);
741 //info("minbuffer %u maxbuffer %u", minbuffer, maxbuffer);
743 info("WARNING: -L option can impact performance");
747 if((err = pthread_create(&tid, 0, control_thread, 0)))
748 fatal(err, "pthread_create control_thread");
752 unsigned char buffer[65536];
755 if((fd = open(dumpfile, O_RDWR|O_TRUNC|O_CREAT, 0666)) < 0)
756 fatal(errno, "opening %s", dumpfile);
757 /* Fill with 0s to a suitable size */
758 memset(buffer, 0, sizeof buffer);
759 for(written = 0; written < dump_size * sizeof(int16_t);
760 written += sizeof buffer) {
761 if(write(fd, buffer, sizeof buffer) < 0)
762 fatal(errno, "clearing %s", dumpfile);
764 /* Map the buffer into memory for convenience */
765 dump_buffer = mmap(0, dump_size * sizeof(int16_t), PROT_READ|PROT_WRITE,
767 if(dump_buffer == (void *)-1)
768 fatal(errno, "mapping %s", dumpfile);
769 info("dumping to %s", dumpfile);
771 /* Set up output. Currently we only support L16 so there's no harm setting
772 * the format before we know what it is! */
773 uaudio_set_format(44100/*Hz*/, 2/*channels*/,
774 16/*bits/channel*/, 1/*signed*/);
775 backend->start(playrtp_callback, NULL);
776 /* We receive and convert audio data in a background thread */
777 if((err = pthread_create(<id, 0, listen_thread, 0)))
778 fatal(err, "pthread_create listen_thread");
779 /* We have a second thread to add received packets to the queue */
780 if((err = pthread_create(<id, 0, queue_thread, 0)))
781 fatal(err, "pthread_create queue_thread");
782 pthread_mutex_lock(&lock);
784 /* Wait for the buffer to fill up a bit */
785 playrtp_fill_buffer();
786 /* Start playing now */
788 next_timestamp = pheap_first(&packets)->timestamp;
790 pthread_mutex_unlock(&lock);
792 pthread_mutex_lock(&lock);
793 /* Wait until the buffer empties out
795 * If there's a packet that we can play right now then we definitely
798 * Also if there's at least minbuffer samples we carry on regardless and
799 * insert silence. The assumption is there's been a pause but more data
802 while(nsamples >= minbuffer
804 && contains(pheap_first(&packets), next_timestamp))) {
805 //fprintf(stderr, "%8u/%u (%u) PLAYING\n", nsamples, maxbuffer, minbuffer);
806 pthread_cond_wait(&cond, &lock);
810 struct packet *p = pheap_first(&packets);
811 fprintf(stderr, "nsamples=%u (%u) next_timestamp=%"PRIx32", first packet is [%"PRIx32",%"PRIx32")\n",
812 nsamples, minbuffer, next_timestamp,p->timestamp,p->timestamp+p->nsamples);
815 /* Stop playing for a bit until the buffer re-fills */
816 pthread_mutex_unlock(&lock);
817 backend->deactivate();
818 pthread_mutex_lock(&lock);