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>
71 #include "configuration.h"
81 #include "inputline.h"
85 /** @brief Obsolete synonym */
86 #ifndef IPV6_JOIN_GROUP
87 # define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP
90 /** @brief RTP socket */
93 /** @brief Log output */
96 /** @brief Output device */
98 /** @brief Buffer low watermark in samples */
99 unsigned minbuffer = 4 * (2 * 44100) / 10; /* 0.4 seconds */
101 /** @brief Maximum buffer size in samples
103 * 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 /** @brief Backend to play with */
165 static const struct uaudio *backend;
167 HEAP_DEFINE(pheap, struct packet *, lt_packet);
169 /** @brief Control socket or NULL */
170 const char *control_socket;
172 /** @brief Buffer for debugging dump
174 * The debug dump is enabled by the @c --dump option. It records the last 20s
175 * of audio to the specified file (which will be about 3.5Mbytes). The file is
176 * written as as ring buffer, so the start point will progress through it.
178 * Use clients/dump2wav to convert this to a WAV file, which can then be loaded
179 * into (e.g.) Audacity for further inspection.
181 * All three backends (ALSA, OSS, Core Audio) now support this option.
183 * The idea is to allow the user a few seconds to react to an audible artefact.
185 int16_t *dump_buffer;
187 /** @brief Current index within debugging dump */
190 /** @brief Size of debugging dump in samples */
191 size_t dump_size = 44100/*Hz*/ * 2/*channels*/ * 20/*seconds*/;
193 static const struct option options[] = {
194 { "help", no_argument, 0, 'h' },
195 { "version", no_argument, 0, 'V' },
196 { "debug", no_argument, 0, 'd' },
197 { "device", required_argument, 0, 'D' },
198 { "min", required_argument, 0, 'm' },
199 { "max", required_argument, 0, 'x' },
200 { "rcvbuf", required_argument, 0, 'R' },
201 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
202 { "oss", no_argument, 0, 'o' },
204 #if HAVE_ALSA_ASOUNDLIB_H
205 { "alsa", no_argument, 0, 'a' },
207 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
208 { "core-audio", no_argument, 0, 'c' },
210 { "dump", required_argument, 0, 'r' },
211 { "command", required_argument, 0, 'e' },
212 { "pause-mode", required_argument, 0, 'P' },
213 { "socket", required_argument, 0, 's' },
214 { "config", required_argument, 0, 'C' },
215 { "monitor", no_argument, 0, 'M' },
219 /** @brief Control thread
221 * This thread is responsible for accepting control commands from Disobedience
222 * (or other controllers) over an AF_UNIX stream socket with a path specified
223 * by the @c --socket option. The protocol uses simple string commands and
226 * - @c stop will shut the player down
227 * - @c query will send back the reply @c running
228 * - anything else is ignored
230 * Commands and response strings terminated by shutting down the connection or
231 * by a newline. No attempt is made to multiplex multiple clients so it is
232 * important that the command be sent as soon as the connection is made - it is
233 * assumed that both parties to the protocol are entirely cooperating with one
236 static void *control_thread(void attribute((unused)) *arg) {
237 struct sockaddr_un sa;
243 assert(control_socket);
244 unlink(control_socket);
245 memset(&sa, 0, sizeof sa);
246 sa.sun_family = AF_UNIX;
247 strcpy(sa.sun_path, control_socket);
248 sfd = xsocket(PF_UNIX, SOCK_STREAM, 0);
249 if(bind(sfd, (const struct sockaddr *)&sa, sizeof sa) < 0)
250 fatal(errno, "error binding to %s", control_socket);
251 if(listen(sfd, 128) < 0)
252 fatal(errno, "error calling listen on %s", control_socket);
253 info("listening on %s", control_socket);
256 cfd = accept(sfd, (struct sockaddr *)&sa, &salen);
263 fatal(errno, "error calling accept on %s", control_socket);
266 if(!(fp = fdopen(cfd, "r+"))) {
267 error(errno, "error calling fdopen for %s connection", control_socket);
271 if(!inputline(control_socket, fp, &line, '\n')) {
272 if(!strcmp(line, "stop")) {
273 info("stopped via %s", control_socket);
274 exit(0); /* terminate immediately */
276 if(!strcmp(line, "query"))
277 fprintf(fp, "running");
281 error(errno, "error closing %s connection", control_socket);
285 /** @brief Drop the first packet
287 * Assumes that @ref lock is held.
289 static void drop_first_packet(void) {
290 if(pheap_count(&packets)) {
291 struct packet *const p = pheap_remove(&packets);
292 nsamples -= p->nsamples;
293 playrtp_free_packet(p);
294 pthread_cond_broadcast(&cond);
298 /** @brief Background thread adding packets to heap
300 * This just transfers packets from @ref received_packets to @ref packets. It
301 * is important that it holds @ref receive_lock for as little time as possible,
302 * in order to minimize the interval between calls to read() in
305 static void *queue_thread(void attribute((unused)) *arg) {
309 /* Get the next packet */
310 pthread_mutex_lock(&receive_lock);
311 while(!received_packets) {
312 pthread_cond_wait(&receive_cond, &receive_lock);
314 p = received_packets;
315 received_packets = p->next;
316 if(!received_packets)
317 received_tail = &received_packets;
319 pthread_mutex_unlock(&receive_lock);
320 /* Add it to the heap */
321 pthread_mutex_lock(&lock);
322 pheap_insert(&packets, p);
323 nsamples += p->nsamples;
324 pthread_cond_broadcast(&cond);
325 pthread_mutex_unlock(&lock);
329 /** @brief Background thread collecting samples
331 * This function collects samples, perhaps converts them to the target format,
332 * and adds them to the packet list.
334 * It is crucial that the gap between successive calls to read() is as small as
335 * possible: otherwise packets will be dropped.
337 * We use a binary heap to ensure that the unavoidable effort is at worst
338 * logarithmic in the total number of packets - in fact if packets are mostly
339 * received in order then we will largely do constant work per packet since the
340 * newest packet will always be last.
342 * Of more concern is that we must acquire the lock on the heap to add a packet
343 * to it. If this proves a problem in practice then the answer would be
344 * (probably doubly) linked list with new packets added the end and a second
345 * thread which reads packets off the list and adds them to the heap.
347 * We keep memory allocation (mostly) very fast by keeping pre-allocated
348 * packets around; see @ref playrtp_new_packet().
350 static void *listen_thread(void attribute((unused)) *arg) {
351 struct packet *p = 0;
353 struct rtp_header header;
360 p = playrtp_new_packet();
361 iov[0].iov_base = &header;
362 iov[0].iov_len = sizeof header;
363 iov[1].iov_base = p->samples_raw;
364 iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw;
365 n = readv(rtpfd, iov, 2);
371 fatal(errno, "error reading from socket");
374 /* Ignore too-short packets */
375 if((size_t)n <= sizeof (struct rtp_header)) {
376 info("ignored a short packet");
379 timestamp = htonl(header.timestamp);
380 seq = htons(header.seq);
381 /* Ignore packets in the past */
382 if(active && lt(timestamp, next_timestamp)) {
383 info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32,
384 timestamp, next_timestamp);
387 /* Ignore packets with the extension bit set. */
388 if(header.vpxcc & 0x10)
392 p->timestamp = timestamp;
393 /* Convert to target format */
394 if(header.mpt & 0x80)
396 switch(header.mpt & 0x7F) {
398 p->nsamples = (n - sizeof header) / sizeof(uint16_t);
400 /* TODO support other RFC3551 media types (when the speaker does) */
402 fatal(0, "unsupported RTP payload type %d",
406 fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n",
407 seq, timestamp, p->nsamples, timestamp + p->nsamples);
408 /* Stop reading if we've reached the maximum.
410 * This is rather unsatisfactory: it means that if packets get heavily
411 * out of order then we guarantee dropouts. But for now... */
412 if(nsamples >= maxbuffer) {
413 pthread_mutex_lock(&lock);
414 while(nsamples >= maxbuffer) {
415 pthread_cond_wait(&cond, &lock);
417 pthread_mutex_unlock(&lock);
419 /* Add the packet to the receive queue */
420 pthread_mutex_lock(&receive_lock);
422 received_tail = &p->next;
424 pthread_cond_signal(&receive_cond);
425 pthread_mutex_unlock(&receive_lock);
426 /* We'll need a new packet */
431 /** @brief Wait until the buffer is adequately full
433 * Must be called with @ref lock held.
435 void playrtp_fill_buffer(void) {
436 /* Discard current buffer contents */
438 //fprintf(stderr, "%8u/%u (%u) DROPPING\n", nsamples, maxbuffer, minbuffer);
441 info("Buffering...");
442 /* Wait until there's at least minbuffer samples available */
443 while(nsamples < minbuffer) {
444 //fprintf(stderr, "%8u/%u (%u) FILLING\n", nsamples, maxbuffer, minbuffer);
445 pthread_cond_wait(&cond, &lock);
447 /* Start from whatever is earliest */
448 next_timestamp = pheap_first(&packets)->timestamp;
452 /** @brief Find next packet
453 * @return Packet to play or NULL if none found
455 * The return packet is merely guaranteed not to be in the past: it might be
456 * the first packet in the future rather than one that is actually suitable to
459 * Must be called with @ref lock held.
461 struct packet *playrtp_next_packet(void) {
462 while(pheap_count(&packets)) {
463 struct packet *const p = pheap_first(&packets);
464 if(le(p->timestamp + p->nsamples, next_timestamp)) {
465 /* This packet is in the past. Drop it and try another one. */
468 /* This packet is NOT in the past. (It might be in the future
475 /* display usage message and terminate */
476 static void help(void) {
478 " disorder-playrtp [OPTIONS] [[ADDRESS] PORT]\n"
480 " --device, -D DEVICE Output device\n"
481 " --min, -m FRAMES Buffer low 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 " --command, -e COMMAND Pipe audio to command.\n"
495 " --pause-mode, -P silence For -e: pauses send silence (default)\n"
496 " --pause-mode, -P suspend For -e: pauses suspend writes\n"
497 " --help, -h Display usage message\n"
498 " --version, -V Display version number\n"
504 static size_t playrtp_callback(void *buffer,
506 void attribute((unused)) *userdata) {
509 pthread_mutex_lock(&lock);
510 /* Get the next packet, junking any that are now in the past */
511 const struct packet *p = playrtp_next_packet();
512 if(p && contains(p, next_timestamp)) {
513 /* This packet is ready to play; the desired next timestamp points
514 * somewhere into it. */
516 /* Timestamp of end of packet */
517 const uint32_t packet_end = p->timestamp + p->nsamples;
519 /* Offset of desired next timestamp into current packet */
520 const uint32_t offset = next_timestamp - p->timestamp;
522 /* Pointer to audio data */
523 const uint16_t *ptr = (void *)(p->samples_raw + offset);
525 /* Compute number of samples left in packet, limited to output buffer
527 samples = packet_end - next_timestamp;
528 if(samples > max_samples)
529 samples = max_samples;
531 /* Copy into buffer, converting to native endianness */
533 int16_t *bufptr = buffer;
535 *bufptr++ = (int16_t)ntohs(*ptr++);
539 /* There is no suitable packet. We introduce 0s up to the next packet, or
540 * to fill the buffer if there's no next packet or that's too many. The
541 * comparison with max_samples deals with the otherwise troubling overflow
543 samples = p ? p->timestamp - next_timestamp : max_samples;
544 if(samples > max_samples)
545 samples = max_samples;
546 //info("infill by %zu", samples);
547 memset(buffer, 0, samples * uaudio_sample_size);
551 for(size_t i = 0; i < samples; ++i) {
552 dump_buffer[dump_index++] = ((int16_t *)buffer)[i];
553 dump_index %= dump_size;
556 /* Advance timestamp */
557 next_timestamp += samples;
558 /* Junk obsolete packets */
559 playrtp_next_packet();
560 pthread_mutex_unlock(&lock);
564 int main(int argc, char **argv) {
566 struct addrinfo *res;
567 struct stringlist sl;
569 int rcvbuf, target_rcvbuf = 0;
572 struct ipv6_mreq mreq6;
574 char *address, *port;
578 struct sockaddr_in in;
579 struct sockaddr_in6 in6;
581 union any_sockaddr mgroup;
582 const char *dumpfile = 0;
585 static const int one = 1;
587 static const struct addrinfo prefs = {
588 .ai_flags = AI_PASSIVE,
589 .ai_family = PF_INET,
590 .ai_socktype = SOCK_DGRAM,
591 .ai_protocol = IPPROTO_UDP
594 /* Timing information is often important to debugging playrtp, so we include
595 * timestamps in the logs */
598 if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale");
599 backend = uaudio_apis[0];
600 while((n = getopt_long(argc, argv, "hVdD:m:x:L:R:aocC:re:P:M", options, 0)) >= 0) {
603 case 'V': version("disorder-playrtp");
604 case 'd': debugging = 1; break;
605 case 'D': uaudio_set("device", optarg); break;
606 case 'm': minbuffer = 2 * atol(optarg); break;
607 case 'x': maxbuffer = 2 * atol(optarg); break;
608 case 'L': logfp = fopen(optarg, "w"); break;
609 case 'R': target_rcvbuf = atoi(optarg); break;
610 #if HAVE_ALSA_ASOUNDLIB_H
611 case 'a': backend = &uaudio_alsa; break;
613 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
614 case 'o': backend = &uaudio_oss; break;
616 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
617 case 'c': backend = &uaudio_coreaudio; break;
619 case 'C': configfile = optarg; break;
620 case 's': control_socket = optarg; break;
621 case 'r': dumpfile = optarg; break;
622 case 'e': backend = &uaudio_command; uaudio_set("command", optarg); break;
623 case 'P': uaudio_set("pause-mode", optarg); break;
624 case 'M': monitor = 1; break;
625 default: fatal(0, "invalid option");
628 if(config_read(0, NULL)) fatal(0, "cannot read configuration");
630 maxbuffer = 2 * minbuffer;
635 /* Get configuration from server */
636 if(!(c = disorder_new(1))) exit(EXIT_FAILURE);
637 if(disorder_connect(c)) exit(EXIT_FAILURE);
638 if(disorder_rtp_address(c, &address, &port)) exit(EXIT_FAILURE);
640 sl.s = xcalloc(2, sizeof *sl.s);
646 /* Use command-line ADDRESS+PORT or just PORT */
651 fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]");
653 /* Look up address and port */
654 if(!(res = get_address(&sl, &prefs, &sockname)))
656 /* Create the socket */
657 if((rtpfd = socket(res->ai_family,
659 res->ai_protocol)) < 0)
660 fatal(errno, "error creating socket");
661 /* Allow multiple listeners */
662 xsetsockopt(rtpfd, SOL_SOCKET, SO_REUSEADDR, &one, sizeof one);
663 is_multicast = multicast(res->ai_addr);
664 /* The multicast and unicast/broadcast cases are different enough that they
665 * are totally split. Trying to find commonality between them causes more
666 * trouble that it's worth. */
668 /* Stash the multicast group address */
669 memcpy(&mgroup, res->ai_addr, res->ai_addrlen);
670 switch(res->ai_addr->sa_family) {
672 mgroup.in.sin_port = 0;
675 mgroup.in6.sin6_port = 0;
678 fatal(0, "unsupported family %d", (int)res->ai_addr->sa_family);
680 /* Bind to to the multicast group address */
681 if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
682 fatal(errno, "error binding socket to %s", format_sockaddr(res->ai_addr));
683 /* Add multicast group membership */
684 switch(mgroup.sa.sa_family) {
686 mreq.imr_multiaddr = mgroup.in.sin_addr;
687 mreq.imr_interface.s_addr = 0; /* use primary interface */
688 if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP,
689 &mreq, sizeof mreq) < 0)
690 fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP");
693 mreq6.ipv6mr_multiaddr = mgroup.in6.sin6_addr;
694 memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface);
695 if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP,
696 &mreq6, sizeof mreq6) < 0)
697 fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP");
700 fatal(0, "unsupported address family %d", res->ai_family);
702 /* Report what we did */
703 info("listening on %s multicast group %s",
704 format_sockaddr(res->ai_addr), format_sockaddr(&mgroup.sa));
707 switch(res->ai_addr->sa_family) {
709 struct sockaddr_in *in = (struct sockaddr_in *)res->ai_addr;
711 memset(&in->sin_addr, 0, sizeof (struct in_addr));
712 if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
713 fatal(errno, "error binding socket to 0.0.0.0 port %d",
714 ntohs(in->sin_port));
718 struct sockaddr_in6 *in6 = (struct sockaddr_in6 *)res->ai_addr;
720 memset(&in6->sin6_addr, 0, sizeof (struct in6_addr));
724 fatal(0, "unsupported family %d", (int)res->ai_addr->sa_family);
726 if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0)
727 fatal(errno, "error binding socket to %s", format_sockaddr(res->ai_addr));
728 /* Report what we did */
729 info("listening on %s", format_sockaddr(res->ai_addr));
732 if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0)
733 fatal(errno, "error calling getsockopt SO_RCVBUF");
734 if(target_rcvbuf > rcvbuf) {
735 if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF,
736 &target_rcvbuf, sizeof target_rcvbuf) < 0)
737 error(errno, "error calling setsockopt SO_RCVBUF %d",
739 /* We try to carry on anyway */
741 info("changed socket receive buffer from %d to %d",
742 rcvbuf, target_rcvbuf);
744 info("default socket receive buffer %d", rcvbuf);
745 //info("minbuffer %u maxbuffer %u", minbuffer, maxbuffer);
747 info("WARNING: -L option can impact performance");
751 if((err = pthread_create(&tid, 0, control_thread, 0)))
752 fatal(err, "pthread_create control_thread");
756 unsigned char buffer[65536];
759 if((fd = open(dumpfile, O_RDWR|O_TRUNC|O_CREAT, 0666)) < 0)
760 fatal(errno, "opening %s", dumpfile);
761 /* Fill with 0s to a suitable size */
762 memset(buffer, 0, sizeof buffer);
763 for(written = 0; written < dump_size * sizeof(int16_t);
764 written += sizeof buffer) {
765 if(write(fd, buffer, sizeof buffer) < 0)
766 fatal(errno, "clearing %s", dumpfile);
768 /* Map the buffer into memory for convenience */
769 dump_buffer = mmap(0, dump_size * sizeof(int16_t), PROT_READ|PROT_WRITE,
771 if(dump_buffer == (void *)-1)
772 fatal(errno, "mapping %s", dumpfile);
773 info("dumping to %s", dumpfile);
775 /* Set up output. Currently we only support L16 so there's no harm setting
776 * the format before we know what it is! */
777 uaudio_set_format(44100/*Hz*/, 2/*channels*/,
778 16/*bits/channel*/, 1/*signed*/);
779 backend->start(playrtp_callback, NULL);
780 /* We receive and convert audio data in a background thread */
781 if((err = pthread_create(<id, 0, listen_thread, 0)))
782 fatal(err, "pthread_create listen_thread");
783 /* We have a second thread to add received packets to the queue */
784 if((err = pthread_create(<id, 0, queue_thread, 0)))
785 fatal(err, "pthread_create queue_thread");
786 pthread_mutex_lock(&lock);
789 /* Wait for the buffer to fill up a bit */
790 playrtp_fill_buffer();
791 /* Start playing now */
793 next_timestamp = pheap_first(&packets)->timestamp;
795 pthread_mutex_unlock(&lock);
797 pthread_mutex_lock(&lock);
798 /* Wait until the buffer empties out
800 * If there's a packet that we can play right now then we definitely
803 * Also if there's at least minbuffer samples we carry on regardless and
804 * insert silence. The assumption is there's been a pause but more data
807 while(nsamples >= minbuffer
809 && contains(pheap_first(&packets), next_timestamp))) {
811 time_t now = time(0);
813 if(now >= lastlog + 60) {
814 int offset = nsamples - minbuffer;
815 double offtime = (double)offset / (uaudio_rate * uaudio_channels);
816 info("%+d samples off (%d.%02ds, %d bytes)",
818 (int)fabs(offtime) * (offtime < 0 ? -1 : 1),
819 (int)(fabs(offtime) * 100) % 100,
820 offset * uaudio_bits / CHAR_BIT);
824 //fprintf(stderr, "%8u/%u (%u) PLAYING\n", nsamples, maxbuffer, minbuffer);
825 pthread_cond_wait(&cond, &lock);
829 struct packet *p = pheap_first(&packets);
830 fprintf(stderr, "nsamples=%u (%u) next_timestamp=%"PRIx32", first packet is [%"PRIx32",%"PRIx32")\n",
831 nsamples, minbuffer, next_timestamp,p->timestamp,p->timestamp+p->nsamples);
834 /* Stop playing for a bit until the buffer re-fills */
835 pthread_mutex_unlock(&lock);
836 backend->deactivate();
837 pthread_mutex_lock(&lock);