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 #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
= 44100; /* 0.5 seconds */
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 { "command", required_argument
, 0, 'e' },
220 { "pause-mode", required_argument
, 0, 'P' },
221 { "socket", required_argument
, 0, 's' },
222 { "config", required_argument
, 0, 'C' },
226 /** @brief Control thread
228 * This thread is responsible for accepting control commands from Disobedience
229 * (or other controllers) over an AF_UNIX stream socket with a path specified
230 * by the @c --socket option. The protocol uses simple string commands and
233 * - @c stop will shut the player down
234 * - @c query will send back the reply @c running
235 * - anything else is ignored
237 * Commands and response strings terminated by shutting down the connection or
238 * by a newline. No attempt is made to multiplex multiple clients so it is
239 * important that the command be sent as soon as the connection is made - it is
240 * assumed that both parties to the protocol are entirely cooperating with one
243 static void *control_thread(void attribute((unused
)) *arg
) {
244 struct sockaddr_un sa
;
250 assert(control_socket
);
251 unlink(control_socket
);
252 memset(&sa
, 0, sizeof sa
);
253 sa
.sun_family
= AF_UNIX
;
254 strcpy(sa
.sun_path
, control_socket
);
255 sfd
= xsocket(PF_UNIX
, SOCK_STREAM
, 0);
256 if(bind(sfd
, (const struct sockaddr
*)&sa
, sizeof sa
) < 0)
257 fatal(errno
, "error binding to %s", control_socket
);
258 if(listen(sfd
, 128) < 0)
259 fatal(errno
, "error calling listen on %s", control_socket
);
260 info("listening on %s", control_socket
);
263 cfd
= accept(sfd
, (struct sockaddr
*)&sa
, &salen
);
270 fatal(errno
, "error calling accept on %s", control_socket
);
273 if(!(fp
= fdopen(cfd
, "r+"))) {
274 error(errno
, "error calling fdopen for %s connection", control_socket
);
278 if(!inputline(control_socket
, fp
, &line
, '\n')) {
279 if(!strcmp(line
, "stop")) {
280 info("stopped via %s", control_socket
);
281 exit(0); /* terminate immediately */
283 if(!strcmp(line
, "query"))
284 fprintf(fp
, "running");
288 error(errno
, "error closing %s connection", control_socket
);
292 /** @brief Drop the first packet
294 * Assumes that @ref lock is held.
296 static void drop_first_packet(void) {
297 if(pheap_count(&packets
)) {
298 struct packet
*const p
= pheap_remove(&packets
);
299 nsamples
-= p
->nsamples
;
300 playrtp_free_packet(p
);
301 pthread_cond_broadcast(&cond
);
305 /** @brief Background thread adding packets to heap
307 * This just transfers packets from @ref received_packets to @ref packets. It
308 * is important that it holds @ref receive_lock for as little time as possible,
309 * in order to minimize the interval between calls to read() in
312 static void *queue_thread(void attribute((unused
)) *arg
) {
316 /* Get the next packet */
317 pthread_mutex_lock(&receive_lock
);
318 while(!received_packets
) {
319 pthread_cond_wait(&receive_cond
, &receive_lock
);
321 p
= received_packets
;
322 received_packets
= p
->next
;
323 if(!received_packets
)
324 received_tail
= &received_packets
;
326 pthread_mutex_unlock(&receive_lock
);
327 /* Add it to the heap */
328 pthread_mutex_lock(&lock
);
329 pheap_insert(&packets
, p
);
330 nsamples
+= p
->nsamples
;
331 pthread_cond_broadcast(&cond
);
332 pthread_mutex_unlock(&lock
);
336 /** @brief Background thread collecting samples
338 * This function collects samples, perhaps converts them to the target format,
339 * and adds them to the packet list.
341 * It is crucial that the gap between successive calls to read() is as small as
342 * possible: otherwise packets will be dropped.
344 * We use a binary heap to ensure that the unavoidable effort is at worst
345 * logarithmic in the total number of packets - in fact if packets are mostly
346 * received in order then we will largely do constant work per packet since the
347 * newest packet will always be last.
349 * Of more concern is that we must acquire the lock on the heap to add a packet
350 * to it. If this proves a problem in practice then the answer would be
351 * (probably doubly) linked list with new packets added the end and a second
352 * thread which reads packets off the list and adds them to the heap.
354 * We keep memory allocation (mostly) very fast by keeping pre-allocated
355 * packets around; see @ref playrtp_new_packet().
357 static void *listen_thread(void attribute((unused
)) *arg
) {
358 struct packet
*p
= 0;
360 struct rtp_header header
;
367 p
= playrtp_new_packet();
368 iov
[0].iov_base
= &header
;
369 iov
[0].iov_len
= sizeof header
;
370 iov
[1].iov_base
= p
->samples_raw
;
371 iov
[1].iov_len
= sizeof p
->samples_raw
/ sizeof *p
->samples_raw
;
372 n
= readv(rtpfd
, iov
, 2);
378 fatal(errno
, "error reading from socket");
381 /* Ignore too-short packets */
382 if((size_t)n
<= sizeof (struct rtp_header
)) {
383 info("ignored a short packet");
386 timestamp
= htonl(header
.timestamp
);
387 seq
= htons(header
.seq
);
388 /* Ignore packets in the past */
389 if(active
&& lt(timestamp
, next_timestamp
)) {
390 info("dropping old packet, timestamp=%"PRIx32
" < %"PRIx32
,
391 timestamp
, next_timestamp
);
394 /* Ignore packets with the extension bit set. */
395 if(header
.vpxcc
& 0x10)
399 p
->timestamp
= timestamp
;
400 /* Convert to target format */
401 if(header
.mpt
& 0x80)
403 switch(header
.mpt
& 0x7F) {
405 p
->nsamples
= (n
- sizeof header
) / sizeof(uint16_t);
407 /* TODO support other RFC3551 media types (when the speaker does) */
409 fatal(0, "unsupported RTP payload type %d",
413 fprintf(logfp
, "sequence %u timestamp %"PRIx32
" length %"PRIx32
" end %"PRIx32
"\n",
414 seq
, timestamp
, p
->nsamples
, timestamp
+ p
->nsamples
);
415 /* Stop reading if we've reached the maximum.
417 * This is rather unsatisfactory: it means that if packets get heavily
418 * out of order then we guarantee dropouts. But for now... */
419 if(nsamples
>= maxbuffer
) {
420 pthread_mutex_lock(&lock
);
421 while(nsamples
>= maxbuffer
) {
422 pthread_cond_wait(&cond
, &lock
);
424 pthread_mutex_unlock(&lock
);
426 /* Add the packet to the receive queue */
427 pthread_mutex_lock(&receive_lock
);
429 received_tail
= &p
->next
;
431 pthread_cond_signal(&receive_cond
);
432 pthread_mutex_unlock(&receive_lock
);
433 /* We'll need a new packet */
438 /** @brief Wait until the buffer is adequately full
440 * Must be called with @ref lock held.
442 void playrtp_fill_buffer(void) {
445 info("Buffering...");
446 while(nsamples
< readahead
) {
447 pthread_cond_wait(&cond
, &lock
);
449 next_timestamp
= pheap_first(&packets
)->timestamp
;
453 /** @brief Find next packet
454 * @return Packet to play or NULL if none found
456 * The return packet is merely guaranteed not to be in the past: it might be
457 * the first packet in the future rather than one that is actually suitable to
460 * Must be called with @ref lock held.
462 struct packet
*playrtp_next_packet(void) {
463 while(pheap_count(&packets
)) {
464 struct packet
*const p
= pheap_first(&packets
);
465 if(le(p
->timestamp
+ p
->nsamples
, next_timestamp
)) {
466 /* This packet is in the past. Drop it and try another one. */
469 /* This packet is NOT in the past. (It might be in the future
476 /* display usage message and terminate */
477 static void help(void) {
479 " disorder-playrtp [OPTIONS] [[ADDRESS] PORT]\n"
481 " --device, -D DEVICE Output device\n"
482 " --min, -m FRAMES Buffer low water mark\n"
483 " --buffer, -b FRAMES Buffer high water mark\n"
484 " --max, -x FRAMES Buffer maximum size\n"
485 " --rcvbuf, -R BYTES Socket receive buffer size\n"
486 " --config, -C PATH Set configuration file\n"
487 #if HAVE_ALSA_ASOUNDLIB_H
488 " --alsa, -a Use ALSA to play audio\n"
490 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
491 " --oss, -o Use OSS to play audio\n"
493 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
494 " --core-audio, -c Use Core Audio to play audio\n"
496 " --command, -e COMMAND Pipe audio to command.\n"
497 " --pause-mode, -P silence For -e: pauses send silence (default)\n"
498 " --pause-mode, -P suspend For -e: pauses suspend writes\n"
499 " --help, -h Display usage message\n"
500 " --version, -V Display version number\n"
506 static size_t playrtp_callback(void *buffer
,
508 void attribute((unused
)) *userdata
) {
511 pthread_mutex_lock(&lock
);
512 /* Get the next packet, junking any that are now in the past */
513 const struct packet
*p
= playrtp_next_packet();
514 if(p
&& contains(p
, next_timestamp
)) {
515 /* This packet is ready to play; the desired next timestamp points
516 * somewhere into it. */
518 /* Timestamp of end of packet */
519 const uint32_t packet_end
= p
->timestamp
+ p
->nsamples
;
521 /* Offset of desired next timestamp into current packet */
522 const uint32_t offset
= next_timestamp
- p
->timestamp
;
524 /* Pointer to audio data */
525 const uint16_t *ptr
= (void *)(p
->samples_raw
+ offset
);
527 /* Compute number of samples left in packet, limited to output buffer
529 samples
= packet_end
- next_timestamp
;
530 if(samples
> max_samples
)
531 samples
= max_samples
;
533 /* Copy into buffer, converting to native endianness */
535 int16_t *bufptr
= buffer
;
537 *bufptr
++ = (int16_t)ntohs(*ptr
++);
540 /* We don't junk the packet here; a subsequent call to
541 * playrtp_next_packet() will dispose of it (if it's actually done with). */
543 /* There is no suitable packet. We introduce 0s up to the next packet, or
544 * to fill the buffer if there's no next packet or that's too many. The
545 * comparison with max_samples deals with the otherwise troubling overflow
547 samples
= p ? p
->timestamp
- next_timestamp
: max_samples
;
548 if(samples
> max_samples
)
549 samples
= max_samples
;
550 //info("infill by %zu", samples);
551 memset(buffer
, 0, samples
* uaudio_sample_size
);
555 for(size_t i
= 0; i
< samples
; ++i
) {
556 dump_buffer
[dump_index
++] = ((int16_t *)buffer
)[i
];
557 dump_index
%= dump_size
;
560 /* Advance timestamp */
561 next_timestamp
+= samples
;
562 pthread_mutex_unlock(&lock
);
566 int main(int argc
, char **argv
) {
568 struct addrinfo
*res
;
569 struct stringlist sl
;
571 int rcvbuf
, target_rcvbuf
= 131072;
574 struct ipv6_mreq mreq6
;
576 char *address
, *port
;
580 struct sockaddr_in in
;
581 struct sockaddr_in6 in6
;
583 union any_sockaddr mgroup
;
584 const char *dumpfile
= 0;
586 static const int one
= 1;
588 static const struct addrinfo prefs
= {
589 .ai_flags
= AI_PASSIVE
,
590 .ai_family
= PF_INET
,
591 .ai_socktype
= SOCK_DGRAM
,
592 .ai_protocol
= IPPROTO_UDP
595 /* Timing information is often important to debugging playrtp, so we include
596 * timestamps in the logs */
599 if(!setlocale(LC_CTYPE
, "")) fatal(errno
, "error calling setlocale");
600 backend
= uaudio_apis
[0];
601 while((n
= getopt_long(argc
, argv
, "hVdD:m:b:x:L:R:M:aocC:re:P:", options
, 0)) >= 0) {
604 case 'V': version("disorder-playrtp");
605 case 'd': debugging
= 1; break;
606 case 'D': uaudio_set("device", optarg
); break;
607 case 'm': minbuffer
= 2 * atol(optarg
); break;
608 case 'b': readahead
= 2 * atol(optarg
); break;
609 case 'x': maxbuffer
= 2 * atol(optarg
); break;
610 case 'L': logfp
= fopen(optarg
, "w"); break;
611 case 'R': target_rcvbuf
= atoi(optarg
); break;
612 #if HAVE_ALSA_ASOUNDLIB_H
613 case 'a': backend
= &uaudio_alsa
; break;
615 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
616 case 'o': backend
= &uaudio_oss
; break;
618 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
619 case 'c': backend
= &uaudio_coreaudio
; break;
621 case 'C': configfile
= optarg
; break;
622 case 's': control_socket
= optarg
; break;
623 case 'r': dumpfile
= optarg
; break;
624 case 'e': backend
= &uaudio_command
; uaudio_set("command", optarg
); break;
625 case 'P': uaudio_set("pause-mode", optarg
); break;
626 default: fatal(0, "invalid option");
629 if(config_read(0)) fatal(0, "cannot read configuration");
631 maxbuffer
= 4 * readahead
;
636 /* Get configuration from server */
637 if(!(c
= disorder_new(1))) exit(EXIT_FAILURE
);
638 if(disorder_connect(c
)) exit(EXIT_FAILURE
);
639 if(disorder_rtp_address(c
, &address
, &port
)) exit(EXIT_FAILURE
);
641 sl
.s
= xcalloc(2, sizeof *sl
.s
);
647 /* Use command-line ADDRESS+PORT or just PORT */
652 fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]");
654 /* Look up address and port */
655 if(!(res
= get_address(&sl
, &prefs
, &sockname
)))
657 /* Create the socket */
658 if((rtpfd
= socket(res
->ai_family
,
660 res
->ai_protocol
)) < 0)
661 fatal(errno
, "error creating socket");
662 /* Allow multiple listeners */
663 xsetsockopt(rtpfd
, SOL_SOCKET
, SO_REUSEADDR
, &one
, sizeof one
);
664 is_multicast
= multicast(res
->ai_addr
);
665 /* The multicast and unicast/broadcast cases are different enough that they
666 * are totally split. Trying to find commonality between them causes more
667 * trouble that it's worth. */
669 /* Stash the multicast group address */
670 memcpy(&mgroup
, res
->ai_addr
, res
->ai_addrlen
);
671 switch(res
->ai_addr
->sa_family
) {
673 mgroup
.in
.sin_port
= 0;
676 mgroup
.in6
.sin6_port
= 0;
679 fatal(0, "unsupported family %d", (int)res
->ai_addr
->sa_family
);
681 /* Bind to to the multicast group address */
682 if(bind(rtpfd
, res
->ai_addr
, res
->ai_addrlen
) < 0)
683 fatal(errno
, "error binding socket to %s", format_sockaddr(res
->ai_addr
));
684 /* Add multicast group membership */
685 switch(mgroup
.sa
.sa_family
) {
687 mreq
.imr_multiaddr
= mgroup
.in
.sin_addr
;
688 mreq
.imr_interface
.s_addr
= 0; /* use primary interface */
689 if(setsockopt(rtpfd
, IPPROTO_IP
, IP_ADD_MEMBERSHIP
,
690 &mreq
, sizeof mreq
) < 0)
691 fatal(errno
, "error calling setsockopt IP_ADD_MEMBERSHIP");
694 mreq6
.ipv6mr_multiaddr
= mgroup
.in6
.sin6_addr
;
695 memset(&mreq6
.ipv6mr_interface
, 0, sizeof mreq6
.ipv6mr_interface
);
696 if(setsockopt(rtpfd
, IPPROTO_IPV6
, IPV6_JOIN_GROUP
,
697 &mreq6
, sizeof mreq6
) < 0)
698 fatal(errno
, "error calling setsockopt IPV6_JOIN_GROUP");
701 fatal(0, "unsupported address family %d", res
->ai_family
);
703 /* Report what we did */
704 info("listening on %s multicast group %s",
705 format_sockaddr(res
->ai_addr
), format_sockaddr(&mgroup
.sa
));
708 switch(res
->ai_addr
->sa_family
) {
710 struct sockaddr_in
*in
= (struct sockaddr_in
*)res
->ai_addr
;
712 memset(&in
->sin_addr
, 0, sizeof (struct in_addr
));
713 if(bind(rtpfd
, res
->ai_addr
, res
->ai_addrlen
) < 0)
714 fatal(errno
, "error binding socket to 0.0.0.0 port %d",
715 ntohs(in
->sin_port
));
719 struct sockaddr_in6
*in6
= (struct sockaddr_in6
*)res
->ai_addr
;
721 memset(&in6
->sin6_addr
, 0, sizeof (struct in6_addr
));
725 fatal(0, "unsupported family %d", (int)res
->ai_addr
->sa_family
);
727 if(bind(rtpfd
, res
->ai_addr
, res
->ai_addrlen
) < 0)
728 fatal(errno
, "error binding socket to %s", format_sockaddr(res
->ai_addr
));
729 /* Report what we did */
730 info("listening on %s", format_sockaddr(res
->ai_addr
));
733 if(getsockopt(rtpfd
, SOL_SOCKET
, SO_RCVBUF
, &rcvbuf
, &len
) < 0)
734 fatal(errno
, "error calling getsockopt SO_RCVBUF");
735 if(target_rcvbuf
> rcvbuf
) {
736 if(setsockopt(rtpfd
, SOL_SOCKET
, SO_RCVBUF
,
737 &target_rcvbuf
, sizeof target_rcvbuf
) < 0)
738 error(errno
, "error calling setsockopt SO_RCVBUF %d",
740 /* We try to carry on anyway */
742 info("changed socket receive buffer from %d to %d",
743 rcvbuf
, target_rcvbuf
);
745 info("default socket receive buffer %d", rcvbuf
);
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
);
788 /* Wait for the buffer to fill up a bit */
789 playrtp_fill_buffer();
790 /* Start playing now */
792 next_timestamp
= pheap_first(&packets
)->timestamp
;
794 pthread_mutex_unlock(&lock
);
796 pthread_mutex_lock(&lock
);
797 /* Wait until the buffer empties out */
798 while(nsamples
>= minbuffer
800 && contains(pheap_first(&packets
), next_timestamp
))) {
801 pthread_cond_wait(&cond
, &lock
);
803 /* Stop playing for a bit until the buffer re-fills */
804 pthread_mutex_unlock(&lock
);
805 backend
->deactivate();
806 pthread_mutex_lock(&lock
);