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 Minimum low watermark
99 * We'll stop playing if there's only this many samples in the buffer. */
100 unsigned minbuffer
= 2 * 44100 / 10; /* 0.2 seconds */
102 /** @brief Maximum buffer size
104 * We'll stop reading from the network if we have this many samples. */
105 static unsigned maxbuffer
;
107 /** @brief Received packets
108 * Protected by @ref receive_lock
110 * Received packets are added to this list, and queue_thread() picks them off
111 * it and adds them to @ref packets. Whenever a packet is added to it, @ref
112 * receive_cond is signalled.
114 struct packet
*received_packets
;
116 /** @brief Tail of @ref received_packets
117 * Protected by @ref receive_lock
119 struct packet
**received_tail
= &received_packets
;
121 /** @brief Lock protecting @ref received_packets
123 * Only listen_thread() and queue_thread() ever hold this lock. It is vital
124 * that queue_thread() not hold it any longer than it strictly has to. */
125 pthread_mutex_t receive_lock
= PTHREAD_MUTEX_INITIALIZER
;
127 /** @brief Condition variable signalled when @ref received_packets is updated
129 * Used by listen_thread() to notify queue_thread() that it has added another
130 * packet to @ref received_packets. */
131 pthread_cond_t receive_cond
= PTHREAD_COND_INITIALIZER
;
133 /** @brief Length of @ref received_packets */
136 /** @brief Binary heap of received packets */
137 struct pheap packets
;
139 /** @brief Total number of samples available
141 * We make this volatile because we inspect it without a protecting lock,
142 * so the usual pthread_* guarantees aren't available.
144 volatile uint32_t nsamples
;
146 /** @brief Timestamp of next packet to play.
148 * This is set to the timestamp of the last packet, plus the number of
149 * samples it contained. Only valid if @ref active is nonzero.
151 uint32_t next_timestamp
;
153 /** @brief True if actively playing
155 * This is true when playing and false when just buffering. */
158 /** @brief Lock protecting @ref packets */
159 pthread_mutex_t lock
= PTHREAD_MUTEX_INITIALIZER
;
161 /** @brief Condition variable signalled whenever @ref packets is changed */
162 pthread_cond_t cond
= PTHREAD_COND_INITIALIZER
;
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' },
218 /** @brief Control thread
220 * This thread is responsible for accepting control commands from Disobedience
221 * (or other controllers) over an AF_UNIX stream socket with a path specified
222 * by the @c --socket option. The protocol uses simple string commands and
225 * - @c stop will shut the player down
226 * - @c query will send back the reply @c running
227 * - anything else is ignored
229 * Commands and response strings terminated by shutting down the connection or
230 * by a newline. No attempt is made to multiplex multiple clients so it is
231 * important that the command be sent as soon as the connection is made - it is
232 * assumed that both parties to the protocol are entirely cooperating with one
235 static void *control_thread(void attribute((unused
)) *arg
) {
236 struct sockaddr_un sa
;
242 assert(control_socket
);
243 unlink(control_socket
);
244 memset(&sa
, 0, sizeof sa
);
245 sa
.sun_family
= AF_UNIX
;
246 strcpy(sa
.sun_path
, control_socket
);
247 sfd
= xsocket(PF_UNIX
, SOCK_STREAM
, 0);
248 if(bind(sfd
, (const struct sockaddr
*)&sa
, sizeof sa
) < 0)
249 fatal(errno
, "error binding to %s", control_socket
);
250 if(listen(sfd
, 128) < 0)
251 fatal(errno
, "error calling listen on %s", control_socket
);
252 info("listening on %s", control_socket
);
255 cfd
= accept(sfd
, (struct sockaddr
*)&sa
, &salen
);
262 fatal(errno
, "error calling accept on %s", control_socket
);
265 if(!(fp
= fdopen(cfd
, "r+"))) {
266 error(errno
, "error calling fdopen for %s connection", control_socket
);
270 if(!inputline(control_socket
, fp
, &line
, '\n')) {
271 if(!strcmp(line
, "stop")) {
272 info("stopped via %s", control_socket
);
273 exit(0); /* terminate immediately */
275 if(!strcmp(line
, "query"))
276 fprintf(fp
, "running");
280 error(errno
, "error closing %s connection", control_socket
);
284 /** @brief Drop the first packet
286 * Assumes that @ref lock is held.
288 static void drop_first_packet(void) {
289 if(pheap_count(&packets
)) {
290 struct packet
*const p
= pheap_remove(&packets
);
291 nsamples
-= p
->nsamples
;
292 playrtp_free_packet(p
);
293 pthread_cond_broadcast(&cond
);
297 /** @brief Background thread adding packets to heap
299 * This just transfers packets from @ref received_packets to @ref packets. It
300 * is important that it holds @ref receive_lock for as little time as possible,
301 * in order to minimize the interval between calls to read() in
304 static void *queue_thread(void attribute((unused
)) *arg
) {
308 /* Get the next packet */
309 pthread_mutex_lock(&receive_lock
);
310 while(!received_packets
) {
311 pthread_cond_wait(&receive_cond
, &receive_lock
);
313 p
= received_packets
;
314 received_packets
= p
->next
;
315 if(!received_packets
)
316 received_tail
= &received_packets
;
318 pthread_mutex_unlock(&receive_lock
);
319 /* Add it to the heap */
320 pthread_mutex_lock(&lock
);
321 pheap_insert(&packets
, p
);
322 nsamples
+= p
->nsamples
;
323 pthread_cond_broadcast(&cond
);
324 pthread_mutex_unlock(&lock
);
328 /** @brief Background thread collecting samples
330 * This function collects samples, perhaps converts them to the target format,
331 * and adds them to the packet list.
333 * It is crucial that the gap between successive calls to read() is as small as
334 * possible: otherwise packets will be dropped.
336 * We use a binary heap to ensure that the unavoidable effort is at worst
337 * logarithmic in the total number of packets - in fact if packets are mostly
338 * received in order then we will largely do constant work per packet since the
339 * newest packet will always be last.
341 * Of more concern is that we must acquire the lock on the heap to add a packet
342 * to it. If this proves a problem in practice then the answer would be
343 * (probably doubly) linked list with new packets added the end and a second
344 * thread which reads packets off the list and adds them to the heap.
346 * We keep memory allocation (mostly) very fast by keeping pre-allocated
347 * packets around; see @ref playrtp_new_packet().
349 static void *listen_thread(void attribute((unused
)) *arg
) {
350 struct packet
*p
= 0;
352 struct rtp_header header
;
359 p
= playrtp_new_packet();
360 iov
[0].iov_base
= &header
;
361 iov
[0].iov_len
= sizeof header
;
362 iov
[1].iov_base
= p
->samples_raw
;
363 iov
[1].iov_len
= sizeof p
->samples_raw
/ sizeof *p
->samples_raw
;
364 n
= readv(rtpfd
, iov
, 2);
370 fatal(errno
, "error reading from socket");
373 /* Ignore too-short packets */
374 if((size_t)n
<= sizeof (struct rtp_header
)) {
375 info("ignored a short packet");
378 timestamp
= htonl(header
.timestamp
);
379 seq
= htons(header
.seq
);
380 /* Ignore packets in the past */
381 if(active
&& lt(timestamp
, next_timestamp
)) {
382 info("dropping old packet, timestamp=%"PRIx32
" < %"PRIx32
,
383 timestamp
, next_timestamp
);
386 /* Ignore packets with the extension bit set. */
387 if(header
.vpxcc
& 0x10)
391 p
->timestamp
= timestamp
;
392 /* Convert to target format */
393 if(header
.mpt
& 0x80)
395 switch(header
.mpt
& 0x7F) {
397 p
->nsamples
= (n
- sizeof header
) / sizeof(uint16_t);
399 /* TODO support other RFC3551 media types (when the speaker does) */
401 fatal(0, "unsupported RTP payload type %d",
405 fprintf(logfp
, "sequence %u timestamp %"PRIx32
" length %"PRIx32
" end %"PRIx32
"\n",
406 seq
, timestamp
, p
->nsamples
, timestamp
+ p
->nsamples
);
407 /* Stop reading if we've reached the maximum.
409 * This is rather unsatisfactory: it means that if packets get heavily
410 * out of order then we guarantee dropouts. But for now... */
411 if(nsamples
>= maxbuffer
) {
412 pthread_mutex_lock(&lock
);
413 while(nsamples
>= maxbuffer
) {
414 pthread_cond_wait(&cond
, &lock
);
416 pthread_mutex_unlock(&lock
);
418 /* Add the packet to the receive queue */
419 pthread_mutex_lock(&receive_lock
);
421 received_tail
= &p
->next
;
423 pthread_cond_signal(&receive_cond
);
424 pthread_mutex_unlock(&receive_lock
);
425 /* We'll need a new packet */
430 /** @brief Wait until the buffer is adequately full
432 * Must be called with @ref lock held.
434 void playrtp_fill_buffer(void) {
435 /* Discard current buffer contents */
438 info("Buffering...");
439 /* Wait until there's at least minbuffer samples available */
440 while(nsamples
< minbuffer
) {
441 pthread_cond_wait(&cond
, &lock
);
443 /* Start from whatever is earliest */
444 next_timestamp
= pheap_first(&packets
)->timestamp
;
448 /** @brief Find next packet
449 * @return Packet to play or NULL if none found
451 * The return packet is merely guaranteed not to be in the past: it might be
452 * the first packet in the future rather than one that is actually suitable to
455 * Must be called with @ref lock held.
457 struct packet
*playrtp_next_packet(void) {
458 while(pheap_count(&packets
)) {
459 struct packet
*const p
= pheap_first(&packets
);
460 if(le(p
->timestamp
+ p
->nsamples
, next_timestamp
)) {
461 /* This packet is in the past. Drop it and try another one. */
464 /* This packet is NOT in the past. (It might be in the future
471 /* display usage message and terminate */
472 static void help(void) {
474 " disorder-playrtp [OPTIONS] [[ADDRESS] PORT]\n"
476 " --device, -D DEVICE Output device\n"
477 " --min, -m FRAMES Buffer low water mark\n"
478 " --max, -x FRAMES Buffer maximum size\n"
479 " --rcvbuf, -R BYTES Socket receive buffer size\n"
480 " --config, -C PATH Set configuration file\n"
481 #if HAVE_ALSA_ASOUNDLIB_H
482 " --alsa, -a Use ALSA to play audio\n"
484 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
485 " --oss, -o Use OSS to play audio\n"
487 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
488 " --core-audio, -c Use Core Audio to play audio\n"
490 " --command, -e COMMAND Pipe audio to command.\n"
491 " --pause-mode, -P silence For -e: pauses send silence (default)\n"
492 " --pause-mode, -P suspend For -e: pauses suspend writes\n"
493 " --help, -h Display usage message\n"
494 " --version, -V Display version number\n"
500 static size_t playrtp_callback(void *buffer
,
502 void attribute((unused
)) *userdata
) {
505 pthread_mutex_lock(&lock
);
506 /* Get the next packet, junking any that are now in the past */
507 const struct packet
*p
= playrtp_next_packet();
508 if(p
&& contains(p
, next_timestamp
)) {
509 /* This packet is ready to play; the desired next timestamp points
510 * somewhere into it. */
512 /* Timestamp of end of packet */
513 const uint32_t packet_end
= p
->timestamp
+ p
->nsamples
;
515 /* Offset of desired next timestamp into current packet */
516 const uint32_t offset
= next_timestamp
- p
->timestamp
;
518 /* Pointer to audio data */
519 const uint16_t *ptr
= (void *)(p
->samples_raw
+ offset
);
521 /* Compute number of samples left in packet, limited to output buffer
523 samples
= packet_end
- next_timestamp
;
524 if(samples
> max_samples
)
525 samples
= max_samples
;
527 /* Copy into buffer, converting to native endianness */
529 int16_t *bufptr
= buffer
;
531 *bufptr
++ = (int16_t)ntohs(*ptr
++);
534 /* We don't junk the packet here; a subsequent call to
535 * playrtp_next_packet() will dispose of it (if it's actually done with). */
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 pthread_mutex_unlock(&lock
);
560 int main(int argc
, char **argv
) {
562 struct addrinfo
*res
;
563 struct stringlist sl
;
565 int rcvbuf
, target_rcvbuf
= 0;
568 struct ipv6_mreq mreq6
;
570 char *address
, *port
;
574 struct sockaddr_in in
;
575 struct sockaddr_in6 in6
;
577 union any_sockaddr mgroup
;
578 const char *dumpfile
= 0;
580 static const int one
= 1;
582 static const struct addrinfo prefs
= {
583 .ai_flags
= AI_PASSIVE
,
584 .ai_family
= PF_INET
,
585 .ai_socktype
= SOCK_DGRAM
,
586 .ai_protocol
= IPPROTO_UDP
589 /* Timing information is often important to debugging playrtp, so we include
590 * timestamps in the logs */
593 if(!setlocale(LC_CTYPE
, "")) fatal(errno
, "error calling setlocale");
594 backend
= uaudio_apis
[0];
595 while((n
= getopt_long(argc
, argv
, "hVdD:m:x:L:R:M:aocC:re:P:", options
, 0)) >= 0) {
598 case 'V': version("disorder-playrtp");
599 case 'd': debugging
= 1; break;
600 case 'D': uaudio_set("device", optarg
); break;
601 case 'm': minbuffer
= 2 * atol(optarg
); break;
602 case 'x': maxbuffer
= 2 * atol(optarg
); break;
603 case 'L': logfp
= fopen(optarg
, "w"); break;
604 case 'R': target_rcvbuf
= atoi(optarg
); break;
605 #if HAVE_ALSA_ASOUNDLIB_H
606 case 'a': backend
= &uaudio_alsa
; break;
608 #if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST
609 case 'o': backend
= &uaudio_oss
; break;
611 #if HAVE_COREAUDIO_AUDIOHARDWARE_H
612 case 'c': backend
= &uaudio_coreaudio
; break;
614 case 'C': configfile
= optarg
; break;
615 case 's': control_socket
= optarg
; break;
616 case 'r': dumpfile
= optarg
; break;
617 case 'e': backend
= &uaudio_command
; uaudio_set("command", optarg
); break;
618 case 'P': uaudio_set("pause-mode", optarg
); break;
619 default: fatal(0, "invalid option");
622 if(config_read(0)) fatal(0, "cannot read configuration");
624 maxbuffer
= 2 * minbuffer
;
629 /* Get configuration from server */
630 if(!(c
= disorder_new(1))) exit(EXIT_FAILURE
);
631 if(disorder_connect(c
)) exit(EXIT_FAILURE
);
632 if(disorder_rtp_address(c
, &address
, &port
)) exit(EXIT_FAILURE
);
634 sl
.s
= xcalloc(2, sizeof *sl
.s
);
640 /* Use command-line ADDRESS+PORT or just PORT */
645 fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]");
647 /* Look up address and port */
648 if(!(res
= get_address(&sl
, &prefs
, &sockname
)))
650 /* Create the socket */
651 if((rtpfd
= socket(res
->ai_family
,
653 res
->ai_protocol
)) < 0)
654 fatal(errno
, "error creating socket");
655 /* Allow multiple listeners */
656 xsetsockopt(rtpfd
, SOL_SOCKET
, SO_REUSEADDR
, &one
, sizeof one
);
657 is_multicast
= multicast(res
->ai_addr
);
658 /* The multicast and unicast/broadcast cases are different enough that they
659 * are totally split. Trying to find commonality between them causes more
660 * trouble that it's worth. */
662 /* Stash the multicast group address */
663 memcpy(&mgroup
, res
->ai_addr
, res
->ai_addrlen
);
664 switch(res
->ai_addr
->sa_family
) {
666 mgroup
.in
.sin_port
= 0;
669 mgroup
.in6
.sin6_port
= 0;
672 fatal(0, "unsupported family %d", (int)res
->ai_addr
->sa_family
);
674 /* Bind to to the multicast group address */
675 if(bind(rtpfd
, res
->ai_addr
, res
->ai_addrlen
) < 0)
676 fatal(errno
, "error binding socket to %s", format_sockaddr(res
->ai_addr
));
677 /* Add multicast group membership */
678 switch(mgroup
.sa
.sa_family
) {
680 mreq
.imr_multiaddr
= mgroup
.in
.sin_addr
;
681 mreq
.imr_interface
.s_addr
= 0; /* use primary interface */
682 if(setsockopt(rtpfd
, IPPROTO_IP
, IP_ADD_MEMBERSHIP
,
683 &mreq
, sizeof mreq
) < 0)
684 fatal(errno
, "error calling setsockopt IP_ADD_MEMBERSHIP");
687 mreq6
.ipv6mr_multiaddr
= mgroup
.in6
.sin6_addr
;
688 memset(&mreq6
.ipv6mr_interface
, 0, sizeof mreq6
.ipv6mr_interface
);
689 if(setsockopt(rtpfd
, IPPROTO_IPV6
, IPV6_JOIN_GROUP
,
690 &mreq6
, sizeof mreq6
) < 0)
691 fatal(errno
, "error calling setsockopt IPV6_JOIN_GROUP");
694 fatal(0, "unsupported address family %d", res
->ai_family
);
696 /* Report what we did */
697 info("listening on %s multicast group %s",
698 format_sockaddr(res
->ai_addr
), format_sockaddr(&mgroup
.sa
));
701 switch(res
->ai_addr
->sa_family
) {
703 struct sockaddr_in
*in
= (struct sockaddr_in
*)res
->ai_addr
;
705 memset(&in
->sin_addr
, 0, sizeof (struct in_addr
));
706 if(bind(rtpfd
, res
->ai_addr
, res
->ai_addrlen
) < 0)
707 fatal(errno
, "error binding socket to 0.0.0.0 port %d",
708 ntohs(in
->sin_port
));
712 struct sockaddr_in6
*in6
= (struct sockaddr_in6
*)res
->ai_addr
;
714 memset(&in6
->sin6_addr
, 0, sizeof (struct in6_addr
));
718 fatal(0, "unsupported family %d", (int)res
->ai_addr
->sa_family
);
720 if(bind(rtpfd
, res
->ai_addr
, res
->ai_addrlen
) < 0)
721 fatal(errno
, "error binding socket to %s", format_sockaddr(res
->ai_addr
));
722 /* Report what we did */
723 info("listening on %s", format_sockaddr(res
->ai_addr
));
726 if(getsockopt(rtpfd
, SOL_SOCKET
, SO_RCVBUF
, &rcvbuf
, &len
) < 0)
727 fatal(errno
, "error calling getsockopt SO_RCVBUF");
728 if(target_rcvbuf
> rcvbuf
) {
729 if(setsockopt(rtpfd
, SOL_SOCKET
, SO_RCVBUF
,
730 &target_rcvbuf
, sizeof target_rcvbuf
) < 0)
731 error(errno
, "error calling setsockopt SO_RCVBUF %d",
733 /* We try to carry on anyway */
735 info("changed socket receive buffer from %d to %d",
736 rcvbuf
, target_rcvbuf
);
738 info("default socket receive buffer %d", rcvbuf
);
740 info("WARNING: -L option can impact performance");
744 if((err
= pthread_create(&tid
, 0, control_thread
, 0)))
745 fatal(err
, "pthread_create control_thread");
749 unsigned char buffer
[65536];
752 if((fd
= open(dumpfile
, O_RDWR
|O_TRUNC
|O_CREAT
, 0666)) < 0)
753 fatal(errno
, "opening %s", dumpfile
);
754 /* Fill with 0s to a suitable size */
755 memset(buffer
, 0, sizeof buffer
);
756 for(written
= 0; written
< dump_size
* sizeof(int16_t);
757 written
+= sizeof buffer
) {
758 if(write(fd
, buffer
, sizeof buffer
) < 0)
759 fatal(errno
, "clearing %s", dumpfile
);
761 /* Map the buffer into memory for convenience */
762 dump_buffer
= mmap(0, dump_size
* sizeof(int16_t), PROT_READ
|PROT_WRITE
,
764 if(dump_buffer
== (void *)-1)
765 fatal(errno
, "mapping %s", dumpfile
);
766 info("dumping to %s", dumpfile
);
768 /* Set up output. Currently we only support L16 so there's no harm setting
769 * the format before we know what it is! */
770 uaudio_set_format(44100/*Hz*/, 2/*channels*/,
771 16/*bits/channel*/, 1/*signed*/);
772 backend
->start(playrtp_callback
, NULL
);
773 /* We receive and convert audio data in a background thread */
774 if((err
= pthread_create(<id
, 0, listen_thread
, 0)))
775 fatal(err
, "pthread_create listen_thread");
776 /* We have a second thread to add received packets to the queue */
777 if((err
= pthread_create(<id
, 0, queue_thread
, 0)))
778 fatal(err
, "pthread_create queue_thread");
779 pthread_mutex_lock(&lock
);
781 /* Wait for the buffer to fill up a bit */
782 playrtp_fill_buffer();
783 /* Start playing now */
785 next_timestamp
= pheap_first(&packets
)->timestamp
;
787 pthread_mutex_unlock(&lock
);
789 pthread_mutex_lock(&lock
);
790 /* Wait until the buffer empties out
792 * If there's a packet that we can play right now then we definitely
795 * Also if there's at least minbuffer samples we carry on regardless and
796 * insert silence. The assumption is there's been a pause but more data
799 while(nsamples
>= minbuffer
801 && contains(pheap_first(&packets
), next_timestamp
))) {
802 pthread_cond_wait(&cond
, &lock
);
804 /* Stop playing for a bit until the buffer re-fills */
805 pthread_mutex_unlock(&lock
);
806 backend
->deactivate();
807 pthread_mutex_lock(&lock
);