2 * This file is part of DisOrder.
3 * Copyright (C) 2009 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>
41 #include "configuration.h"
43 /** @brief Bytes to send per network packet
45 * This is the maximum number of bytes we pass to write(2); to determine actual
46 * packet sizes, add a UDP header and an IP header (and a link layer header if
47 * it's the link layer size you care about).
49 * Don't make this too big or arithmetic will start to overflow.
51 #define NETWORK_BYTES (1500-8/*UDP*/-40/*IP*/-8/*conservatism*/)
53 /** @brief RTP payload type */
54 static int rtp_payload
;
56 /** @brief RTP output socket */
59 /** @brief RTP SSRC */
60 static uint32_t rtp_id
;
62 /** @brief RTP sequence number */
63 static uint16_t rtp_sequence
;
65 /** @brief Network error count
67 * If too many errors occur in too short a time, we give up.
69 static int rtp_errors
;
71 /** @brief Delay threshold in microseconds
73 * rtp_play() never attempts to introduce a delay shorter than this.
75 static int64_t rtp_delay_threshold
;
77 static const char *const rtp_options
[] = {
79 "rtp-destination-port",
88 static void rtp_get_netconfig(const char *af
,
91 struct netaddress
*na
) {
94 vec
[0] = uaudio_get(af
, NULL
);
95 vec
[1] = uaudio_get(addr
, NULL
);
96 vec
[2] = uaudio_get(port
, NULL
);
100 if(netaddress_parse(na
, 3, vec
))
101 fatal(0, "invalid RTP address");
104 static void rtp_set_netconfig(const char *af
,
107 const struct netaddress
*na
) {
108 uaudio_set(af
, NULL
);
109 uaudio_set(addr
, NULL
);
110 uaudio_set(port
, NULL
);
115 netaddress_format(na
, &nvec
, &vec
);
117 uaudio_set(af
, vec
[0]);
121 uaudio_set(addr
, vec
[1]);
125 uaudio_set(port
, vec
[2]);
132 static size_t rtp_play(void *buffer
, size_t nsamples
) {
133 struct rtp_header header
;
136 /* We do as much work as possible before checking what time it is */
137 /* Fill out header */
138 header
.vpxcc
= 2 << 6; /* V=2, P=0, X=0, CC=0 */
139 header
.seq
= htons(rtp_sequence
++);
140 header
.ssrc
= rtp_id
;
141 header
.mpt
= (uaudio_schedule_reactivated ?
0x80 : 0x00) | rtp_payload
;
143 /* Convert samples to network byte order */
144 uint16_t *u
= buffer
, *const limit
= u
+ nsamples
;
150 vec
[0].iov_base
= (void *)&header
;
151 vec
[0].iov_len
= sizeof header
;
152 vec
[1].iov_base
= buffer
;
153 vec
[1].iov_len
= nsamples
* uaudio_sample_size
;
154 uaudio_schedule_synchronize();
155 header
.timestamp
= htonl((uint32_t)uaudio_schedule_timestamp
);
158 written_bytes
= writev(rtp_fd
, vec
, 2);
159 } while(written_bytes
< 0 && errno
== EINTR
);
160 if(written_bytes
< 0) {
161 error(errno
, "error transmitting audio data");
164 fatal(0, "too many audio tranmission errors");
167 rtp_errors
/= 2; /* gradual decay */
168 written_bytes
-= sizeof (struct rtp_header
);
169 const size_t written_samples
= written_bytes
/ uaudio_sample_size
;
170 uaudio_schedule_update(written_samples
);
171 return written_samples
;
174 static void rtp_open(void) {
175 struct addrinfo
*res
, *sres
;
176 static const int one
= 1;
177 int sndbuf
, target_sndbuf
= 131072;
179 struct netaddress dst
[1], src
[1];
181 /* Get configuration */
182 rtp_get_netconfig("rtp-destination-af",
184 "rtp-destination-port",
186 rtp_get_netconfig("rtp-source-af",
190 rtp_delay_threshold
= atoi(uaudio_get("rtp-delay-threshold", "1000"));
191 /* ...microseconds */
193 /* Resolve addresses */
194 res
= netaddress_resolve(dst
, 0, IPPROTO_UDP
);
198 sres
= netaddress_resolve(src
, 1, IPPROTO_UDP
);
203 /* Create the socket */
204 if((rtp_fd
= socket(res
->ai_family
,
206 res
->ai_protocol
)) < 0)
207 fatal(errno
, "error creating broadcast socket");
208 if(multicast(res
->ai_addr
)) {
209 /* Enable multicast options */
210 const int ttl
= atoi(uaudio_get("multicast-ttl", "1"));
211 const int loop
= !strcmp(uaudio_get("multicast-loop", "yes"), "yes");
212 switch(res
->ai_family
) {
214 if(setsockopt(rtp_fd
, IPPROTO_IP
, IP_MULTICAST_TTL
,
215 &ttl
, sizeof ttl
) < 0)
216 fatal(errno
, "error setting IP_MULTICAST_TTL on multicast socket");
217 if(setsockopt(rtp_fd
, IPPROTO_IP
, IP_MULTICAST_LOOP
,
218 &loop
, sizeof loop
) < 0)
219 fatal(errno
, "error setting IP_MULTICAST_LOOP on multicast socket");
223 if(setsockopt(rtp_fd
, IPPROTO_IPV6
, IPV6_MULTICAST_HOPS
,
224 &ttl
, sizeof ttl
) < 0)
225 fatal(errno
, "error setting IPV6_MULTICAST_HOPS on multicast socket");
226 if(setsockopt(rtp_fd
, IPPROTO_IP
, IPV6_MULTICAST_LOOP
,
227 &loop
, sizeof loop
) < 0)
228 fatal(errno
, "error setting IPV6_MULTICAST_LOOP on multicast socket");
232 fatal(0, "unsupported address family %d", res
->ai_family
);
234 info("multicasting on %s TTL=%d loop=%s",
235 format_sockaddr(res
->ai_addr
), ttl
, loop ?
"yes" : "no");
239 if(getifaddrs(&ifs
) < 0)
240 fatal(errno
, "error calling getifaddrs");
242 /* (At least on Darwin) IFF_BROADCAST might be set but ifa_broadaddr
243 * still a null pointer. It turns out that there's a subsequent entry
244 * for he same interface which _does_ have ifa_broadaddr though... */
245 if((ifs
->ifa_flags
& IFF_BROADCAST
)
246 && ifs
->ifa_broadaddr
247 && sockaddr_equal(ifs
->ifa_broadaddr
, res
->ai_addr
))
252 if(setsockopt(rtp_fd
, SOL_SOCKET
, SO_BROADCAST
, &one
, sizeof one
) < 0)
253 fatal(errno
, "error setting SO_BROADCAST on broadcast socket");
254 info("broadcasting on %s (%s)",
255 format_sockaddr(res
->ai_addr
), ifs
->ifa_name
);
257 info("unicasting on %s", format_sockaddr(res
->ai_addr
));
259 /* Enlarge the socket buffer */
261 if(getsockopt(rtp_fd
, SOL_SOCKET
, SO_SNDBUF
,
263 fatal(errno
, "error getting SO_SNDBUF");
264 if(target_sndbuf
> sndbuf
) {
265 if(setsockopt(rtp_fd
, SOL_SOCKET
, SO_SNDBUF
,
266 &target_sndbuf
, sizeof target_sndbuf
) < 0)
267 error(errno
, "error setting SO_SNDBUF to %d", target_sndbuf
);
269 info("changed socket send buffer size from %d to %d",
270 sndbuf
, target_sndbuf
);
272 info("default socket send buffer is %d",
274 /* We might well want to set additional broadcast- or multicast-related
276 if(sres
&& bind(rtp_fd
, sres
->ai_addr
, sres
->ai_addrlen
) < 0)
277 fatal(errno
, "error binding broadcast socket to %s",
278 format_sockaddr(sres
->ai_addr
));
279 if(connect(rtp_fd
, res
->ai_addr
, res
->ai_addrlen
) < 0)
280 fatal(errno
, "error connecting broadcast socket to %s",
281 format_sockaddr(res
->ai_addr
));
284 static void rtp_start(uaudio_callback
*callback
,
286 /* We only support L16 (but we do stereo and mono and will convert sign) */
287 if(uaudio_channels
== 2
289 && uaudio_rate
== 44100)
291 else if(uaudio_channels
== 1
293 && uaudio_rate
== 44100)
296 fatal(0, "asked for %d/%d/%d 16/44100/1 and 16/44100/2",
297 uaudio_bits
, uaudio_rate
, uaudio_channels
);
298 /* Various fields are required to have random initial values by RFC3550. The
299 * packet contents are highly public so there's no point asking for very
300 * strong randomness. */
301 gcry_create_nonce(&rtp_id
, sizeof rtp_id
);
302 gcry_create_nonce(&rtp_sequence
, sizeof rtp_sequence
);
304 uaudio_schedule_init();
305 uaudio_thread_start(callback
,
308 256 / uaudio_sample_size
,
309 (NETWORK_BYTES
- sizeof(struct rtp_header
))
310 / uaudio_sample_size
,
314 static void rtp_stop(void) {
315 uaudio_thread_stop();
320 static void rtp_activate(void) {
321 uaudio_schedule_reactivated
= 1;
322 uaudio_thread_activate();
325 static void rtp_deactivate(void) {
326 uaudio_thread_deactivate();
329 static void rtp_configure(void) {
332 rtp_set_netconfig("rtp-destination-af",
334 "rtp-destination-port", &config
->broadcast
);
335 rtp_set_netconfig("rtp-source-af",
337 "rtp-source-port", &config
->broadcast_from
);
338 snprintf(buffer
, sizeof buffer
, "%ld", config
->multicast_ttl
);
339 uaudio_set("multicast-ttl", buffer
);
340 uaudio_set("multicast-loop", config
->multicast_loop ?
"yes" : "no");
341 snprintf(buffer
, sizeof buffer
, "%ld", config
->rtp_delay_threshold
);
342 uaudio_set("delay-threshold", buffer
);
345 const struct uaudio uaudio_rtp
= {
347 .options
= rtp_options
,
350 .activate
= rtp_activate
,
351 .deactivate
= rtp_deactivate
,
352 .configure
= rtp_configure
,