2 * This file is part of DisOrder.
3 * Copyright (C) 2009, 2013 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 lib/uaudio-rtp.c
19 * @brief Support for RTP network play backend */
23 #include <sys/socket.h>
26 #include <arpa/inet.h>
27 #include <netinet/in.h>
42 #include "configuration.h"
44 /** @brief Bytes to send per network packet
46 * This is the maximum number of bytes we pass to write(2); to determine actual
47 * packet sizes, add a UDP header and an IP header (and a link layer header if
48 * it's the link layer size you care about).
50 * Don't make this too big or arithmetic will start to overflow.
52 #define NETWORK_BYTES (1500-8/*UDP*/-40/*IP*/-8/*conservatism*/)
54 /** @brief RTP payload type */
55 static int rtp_payload
;
57 /** @brief RTP broadcast/multicast output socket */
58 static int rtp_fd
= -1;
60 /** @brief RTP unicast output socket (IPv4) */
61 static int rtp_fd4
= -1;
63 /** @brief RTP unicast output socket (IPv6) */
64 static int rtp_fd6
= -1;
66 /** @brief RTP SSRC */
67 static uint32_t rtp_id
;
69 /** @brief Base for timestamp */
70 static uint32_t rtp_base
;
72 /** @brief RTP sequence number */
73 static uint16_t rtp_sequence
;
75 /** @brief Network error count
77 * If too many errors occur in too short a time, we give up.
79 static int rtp_errors
;
81 /** @brief RTP mode */
84 #define RTP_BROADCAST 1
85 #define RTP_MULTICAST 2
90 /** @brief A unicast client */
91 struct rtp_recipient
{
92 struct rtp_recipient
*next
;
93 struct sockaddr_storage sa
;
96 /** @brief List of unicast clients */
97 static struct rtp_recipient
*rtp_recipient_list
;
99 /** @brief Mutex protecting data structures */
100 static pthread_mutex_t rtp_lock
= PTHREAD_MUTEX_INITIALIZER
;
102 static const char *const rtp_options
[] = {
104 "rtp-destination-port",
113 static void rtp_get_netconfig(const char *af
,
116 struct netaddress
*na
) {
119 vec
[0] = uaudio_get(af
, NULL
);
120 vec
[1] = uaudio_get(addr
, NULL
);
121 vec
[2] = uaudio_get(port
, NULL
);
125 if(netaddress_parse(na
, 3, vec
))
126 disorder_fatal(0, "invalid RTP address");
129 static void rtp_set_netconfig(const char *af
,
132 const struct netaddress
*na
) {
133 uaudio_set(af
, NULL
);
134 uaudio_set(addr
, NULL
);
135 uaudio_set(port
, NULL
);
140 netaddress_format(na
, &nvec
, &vec
);
142 uaudio_set(af
, vec
[0]);
146 uaudio_set(addr
, vec
[1]);
150 uaudio_set(port
, vec
[2]);
157 static size_t rtp_play(void *buffer
, size_t nsamples
, unsigned flags
) {
158 struct rtp_header header
;
162 if(flags
& (UAUDIO_PAUSE
|UAUDIO_RESUME
))
163 fprintf(stderr
, "rtp_play %zu samples%s%s%s%s\n", nsamples
,
164 flags
& UAUDIO_PAUSE ?
" UAUDIO_PAUSE" : "",
165 flags
& UAUDIO_RESUME ?
" UAUDIO_RESUME" : "",
166 flags
& UAUDIO_PLAYING ?
" UAUDIO_PLAYING" : "",
167 flags
& UAUDIO_PAUSED ?
" UAUDIO_PAUSED" : "");
170 /* We do as much work as possible before checking what time it is */
171 /* Fill out header */
172 header
.vpxcc
= 2 << 6; /* V=2, P=0, X=0, CC=0 */
173 header
.seq
= htons(rtp_sequence
++);
174 header
.ssrc
= rtp_id
;
175 header
.mpt
= rtp_payload
;
176 /* If we've come out of a pause, set the marker bit */
177 if(flags
& UAUDIO_RESUME
)
180 /* Convert samples to network byte order */
181 uint16_t *u
= buffer
, *const limit
= u
+ nsamples
;
187 vec
[0].iov_base
= (void *)&header
;
188 vec
[0].iov_len
= sizeof header
;
189 vec
[1].iov_base
= buffer
;
190 vec
[1].iov_len
= nsamples
* uaudio_sample_size
;
191 const uint32_t timestamp
= uaudio_schedule_sync();
192 header
.timestamp
= htonl(rtp_base
+ (uint32_t)timestamp
);
194 /* We send ~120 packets a second with current arrangements. So if we log
195 * once every 8192 packets we log about once a minute. */
197 if(!(ntohs(header
.seq
) & 8191)
198 && config
->rtp_verbose
)
199 disorder_info("RTP: seq %04"PRIx16
" %08"PRIx32
"+%08"PRIx32
"=%08"PRIx32
" ns %zu%s",
205 flags
& UAUDIO_PAUSED ?
" [paused]" : "");
207 /* If we're paused don't actually end a packet, we just pretend */
208 if(flags
& UAUDIO_PAUSED
) {
209 uaudio_schedule_sent(nsamples
);
212 /* Send stuff to explicitly registerd unicast addresses unconditionally */
213 struct rtp_recipient
*r
;
215 memset(&m
, 0, sizeof m
);
218 pthread_mutex_lock(&rtp_lock
);
219 for(r
= rtp_recipient_list
; r
; r
= r
->next
) {
221 m
.msg_namelen
= r
->sa
.ss_family
== AF_INET ?
222 sizeof(struct sockaddr_in
) : sizeof (struct sockaddr_in6
);
223 sendmsg(r
->sa
.ss_family
== AF_INET ? rtp_fd4
: rtp_fd6
,
224 &m
, MSG_DONTWAIT
|MSG_NOSIGNAL
);
225 // TODO similar error handling to other case?
227 pthread_mutex_unlock(&rtp_lock
);
228 if(rtp_mode
!= RTP_REQUEST
) {
231 written_bytes
= writev(rtp_fd
, vec
, 2);
232 } while(written_bytes
< 0 && errno
== EINTR
);
233 if(written_bytes
< 0) {
234 disorder_error(errno
, "error transmitting audio data");
237 disorder_fatal(0, "too many audio transmission errors");
240 rtp_errors
/= 2; /* gradual decay */
242 /* TODO what can we sensibly do about short writes here? Really that's just
243 * an error and we ought to be using smaller packets. */
244 uaudio_schedule_sent(nsamples
);
248 static void hack_send_buffer_size(int fd
, const char *what
) {
249 int sndbuf
, target_sndbuf
= 131072;
250 socklen_t len
= sizeof sndbuf
;
252 if(getsockopt(fd
, SOL_SOCKET
, SO_SNDBUF
,
254 disorder_fatal(errno
, "error getting SO_SNDBUF on %s socket", what
);
255 if(target_sndbuf
> sndbuf
) {
256 if(setsockopt(fd
, SOL_SOCKET
, SO_SNDBUF
,
257 &target_sndbuf
, sizeof target_sndbuf
) < 0)
258 disorder_error(errno
, "error setting SO_SNDBUF on %s socket to %d",
259 what
, target_sndbuf
);
261 disorder_info("changed socket send buffer size on %socket from %d to %d",
262 what
, sndbuf
, target_sndbuf
);
264 disorder_info("default socket send buffer on %s socket is %d",
268 static void rtp_open(void) {
269 struct addrinfo
*dres
, *sres
;
270 static const int one
= 1;
271 struct netaddress dst
[1], src
[1];
275 mode
= uaudio_get("rtp-mode", "auto");
276 if(!strcmp(mode
, "broadcast")) rtp_mode
= RTP_BROADCAST
;
277 else if(!strcmp(mode
, "multicast")) rtp_mode
= RTP_MULTICAST
;
278 else if(!strcmp(mode
, "unicast")) rtp_mode
= RTP_UNICAST
;
279 else if(!strcmp(mode
, "request")) rtp_mode
= RTP_REQUEST
;
280 else rtp_mode
= RTP_AUTO
;
281 /* Get the source and destination addresses (which might be missing) */
282 rtp_get_netconfig("rtp-destination-af",
284 "rtp-destination-port",
286 rtp_get_netconfig("rtp-source-af",
291 dres
= netaddress_resolve(dst
, 0, IPPROTO_UDP
);
297 sres
= netaddress_resolve(src
, 1, IPPROTO_UDP
);
302 /* _AUTO inspects the destination address and acts accordingly */
303 if(rtp_mode
== RTP_AUTO
) {
305 rtp_mode
= RTP_REQUEST
;
306 else if(multicast(dres
->ai_addr
))
307 rtp_mode
= RTP_MULTICAST
;
311 if(getifaddrs(&ifs
) < 0)
312 disorder_fatal(errno
, "error calling getifaddrs");
314 /* (At least on Darwin) IFF_BROADCAST might be set but ifa_broadaddr
315 * still a null pointer. It turns out that there's a subsequent entry
316 * for he same interface which _does_ have ifa_broadaddr though... */
317 if((ifs
->ifa_flags
& IFF_BROADCAST
)
318 && ifs
->ifa_broadaddr
319 && sockaddr_equal(ifs
->ifa_broadaddr
, dres
->ai_addr
))
324 rtp_mode
= RTP_BROADCAST
;
326 rtp_mode
= RTP_UNICAST
;
329 /* Create the sockets */
330 if(rtp_mode
!= RTP_REQUEST
) {
331 if((rtp_fd
= socket(dres
->ai_family
,
333 dres
->ai_protocol
)) < 0)
334 disorder_fatal(errno
, "error creating RTP transmission socket");
336 if((rtp_fd4
= socket(AF_INET
, SOCK_DGRAM
, IPPROTO_UDP
)) < 0)
337 disorder_fatal(errno
, "error creating v4 RTP transmission socket");
338 if((rtp_fd6
= socket(AF_INET6
, SOCK_DGRAM
, IPPROTO_UDP
)) < 0)
339 disorder_fatal(errno
, "error creating v6 RTP transmission socket");
340 /* Configure the socket according to the desired mode */
342 case RTP_MULTICAST
: {
343 /* Enable multicast options */
344 const int ttl
= atoi(uaudio_get("multicast-ttl", "1"));
345 const int loop
= !strcmp(uaudio_get("multicast-loop", "yes"), "yes");
346 switch(dres
->ai_family
) {
348 if(setsockopt(rtp_fd
, IPPROTO_IP
, IP_MULTICAST_TTL
,
349 &ttl
, sizeof ttl
) < 0)
350 disorder_fatal(errno
, "error setting IP_MULTICAST_TTL on multicast socket");
351 if(setsockopt(rtp_fd
, IPPROTO_IP
, IP_MULTICAST_LOOP
,
352 &loop
, sizeof loop
) < 0)
353 disorder_fatal(errno
, "error setting IP_MULTICAST_LOOP on multicast socket");
357 if(setsockopt(rtp_fd
, IPPROTO_IPV6
, IPV6_MULTICAST_HOPS
,
358 &ttl
, sizeof ttl
) < 0)
359 disorder_fatal(errno
, "error setting IPV6_MULTICAST_HOPS on multicast socket");
360 if(setsockopt(rtp_fd
, IPPROTO_IP
, IPV6_MULTICAST_LOOP
,
361 &loop
, sizeof loop
) < 0)
362 disorder_fatal(errno
, "error setting IPV6_MULTICAST_LOOP on multicast socket");
366 disorder_fatal(0, "unsupported address family %d", dres
->ai_family
);
368 disorder_info("multicasting on %s TTL=%d loop=%s",
369 format_sockaddr(dres
->ai_addr
), ttl
, loop ?
"yes" : "no");
373 disorder_info("unicasting on %s", format_sockaddr(dres
->ai_addr
));
376 case RTP_BROADCAST
: {
377 if(setsockopt(rtp_fd
, SOL_SOCKET
, SO_BROADCAST
, &one
, sizeof one
) < 0)
378 disorder_fatal(errno
, "error setting SO_BROADCAST on broadcast socket");
379 disorder_info("broadcasting on %s",
380 format_sockaddr(dres
->ai_addr
));
384 disorder_info("will transmit on request");
388 /* Enlarge the socket buffers */
389 if (rtp_fd
!= -1) hack_send_buffer_size(rtp_fd
, "master socket");
390 hack_send_buffer_size(rtp_fd4
, "IPv4 on-demand socket");
391 hack_send_buffer_size(rtp_fd6
, "IPv6 on-demand socket");
392 /* We might well want to set additional broadcast- or multicast-related
394 if(rtp_mode
!= RTP_REQUEST
) {
395 if(sres
&& bind(rtp_fd
, sres
->ai_addr
, sres
->ai_addrlen
) < 0)
396 disorder_fatal(errno
, "error binding broadcast socket to %s",
397 format_sockaddr(sres
->ai_addr
));
398 if(connect(rtp_fd
, dres
->ai_addr
, dres
->ai_addrlen
) < 0)
399 disorder_fatal(errno
, "error connecting broadcast socket to %s",
400 format_sockaddr(dres
->ai_addr
));
402 if(config
->rtp_verbose
)
403 disorder_info("RTP: prepared socket");
406 static void rtp_start(uaudio_callback
*callback
,
408 /* We only support L16 (but we do stereo and mono and will convert sign) */
409 if(uaudio_channels
== 2
411 && uaudio_rate
== 44100)
413 else if(uaudio_channels
== 1
415 && uaudio_rate
== 44100)
418 disorder_fatal(0, "asked for %d/%d/%d 16/44100/1 and 16/44100/2",
419 uaudio_bits
, uaudio_rate
, uaudio_channels
);
420 if(config
->rtp_verbose
)
421 disorder_info("RTP: %d channels %d bits %d Hz payload type %d",
422 uaudio_channels
, uaudio_bits
, uaudio_rate
, rtp_payload
);
423 /* Various fields are required to have random initial values by RFC3550. The
424 * packet contents are highly public so there's no point asking for very
425 * strong randomness. */
426 gcry_create_nonce(&rtp_id
, sizeof rtp_id
);
427 gcry_create_nonce(&rtp_base
, sizeof rtp_base
);
428 gcry_create_nonce(&rtp_sequence
, sizeof rtp_sequence
);
429 if(config
->rtp_verbose
)
430 disorder_info("RTP: id %08"PRIx32
" base %08"PRIx32
" initial seq %08"PRIx16
,
431 rtp_id
, rtp_base
, rtp_sequence
);
433 uaudio_schedule_init();
434 if(config
->rtp_verbose
)
435 disorder_info("RTP: initialized schedule");
436 uaudio_thread_start(callback
,
439 256 / uaudio_sample_size
,
440 (NETWORK_BYTES
- sizeof(struct rtp_header
))
441 / uaudio_sample_size
,
443 if(config
->rtp_verbose
)
444 disorder_info("RTP: created thread");
447 static void rtp_stop(void) {
448 uaudio_thread_stop();
449 if(rtp_fd
>= 0) { close(rtp_fd
); rtp_fd
= -1; }
450 if(rtp_fd4
>= 0) { close(rtp_fd4
); rtp_fd4
= -1; }
451 if(rtp_fd6
>= 0) { close(rtp_fd6
); rtp_fd6
= -1; }
454 static void rtp_configure(void) {
457 uaudio_set("rtp-mode", config
->rtp_mode
);
458 rtp_set_netconfig("rtp-destination-af",
460 "rtp-destination-port", &config
->broadcast
);
461 rtp_set_netconfig("rtp-source-af",
463 "rtp-source-port", &config
->broadcast_from
);
464 snprintf(buffer
, sizeof buffer
, "%ld", config
->multicast_ttl
);
465 uaudio_set("multicast-ttl", buffer
);
466 uaudio_set("multicast-loop", config
->multicast_loop ?
"yes" : "no");
467 if(config
->rtp_verbose
)
468 disorder_info("RTP: configured");
471 /** @brief Add an RTP recipient address
472 * @param sa Pointer to recipient address
473 * @return 0 on success, -1 on error
475 int rtp_add_recipient(const struct sockaddr_storage
*sa
) {
476 struct rtp_recipient
*r
;
478 pthread_mutex_lock(&rtp_lock
);
479 for(r
= rtp_recipient_list
;
480 r
&& sockaddrcmp((struct sockaddr
*)sa
,
481 (struct sockaddr
*)&r
->sa
);
487 r
= xmalloc(sizeof *r
);
488 memcpy(&r
->sa
, sa
, sizeof *sa
);
489 r
->next
= rtp_recipient_list
;
490 rtp_recipient_list
= r
;
493 pthread_mutex_unlock(&rtp_lock
);
497 /** @brief Remove an RTP recipient address
498 * @param sa Pointer to recipient address
499 * @return 0 on success, -1 on error
501 int rtp_remove_recipient(const struct sockaddr_storage
*sa
) {
502 struct rtp_recipient
*r
, **rr
;
504 pthread_mutex_lock(&rtp_lock
);
505 for(rr
= &rtp_recipient_list
;
506 (r
= *rr
) && sockaddrcmp((struct sockaddr
*)sa
,
507 (struct sockaddr
*)&r
->sa
);
515 disorder_error(0, "bogus rtp_remove_recipient");
518 pthread_mutex_unlock(&rtp_lock
);
522 const struct uaudio uaudio_rtp
= {
524 .options
= rtp_options
,
527 .activate
= uaudio_thread_activate
,
528 .deactivate
= uaudio_thread_deactivate
,
529 .configure
= rtp_configure
,
530 .flags
= UAUDIO_API_SERVER
,