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,