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>
39 #include "configuration.h"
41 /** @brief Bytes to send per network packet
43 * This is the maximum number of bytes we pass to write(2); to determine actual
44 * packet sizes, add a UDP header and an IP header (and a link layer header if
45 * it's the link layer size you care about).
47 * Don't make this too big or arithmetic will start to overflow.
49 #define NETWORK_BYTES (1500-8/*UDP*/-40/*IP*/-8/*conservatism*/)
51 /** @brief RTP payload type */
52 static int rtp_payload;
54 /** @brief RTP output socket */
57 /** @brief RTP SSRC */
58 static uint32_t rtp_id;
60 /** @brief RTP sequence number */
61 static uint16_t rtp_sequence;
63 /** @brief Network error count
65 * If too many errors occur in too short a time, we give up.
67 static int rtp_errors;
69 /** @brief Delay threshold in microseconds
71 * rtp_play() never attempts to introduce a delay shorter than this.
73 static int64_t rtp_delay_threshold;
75 static const char *const rtp_options[] = {
77 "rtp-destination-port",
86 static size_t rtp_play(void *buffer, size_t nsamples) {
87 struct rtp_header header;
90 /* We do as much work as possible before checking what time it is */
92 header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */
93 header.seq = htons(rtp_sequence++);
95 header.mpt = (uaudio_schedule_reactivated ? 0x80 : 0x00) | rtp_payload;
97 /* Convert samples to network byte order */
98 uint16_t *u = buffer, *const limit = u + nsamples;
104 vec[0].iov_base = (void *)&header;
105 vec[0].iov_len = sizeof header;
106 vec[1].iov_base = buffer;
107 vec[1].iov_len = nsamples * uaudio_sample_size;
108 uaudio_schedule_synchronize();
109 header.timestamp = htonl((uint32_t)uaudio_schedule_timestamp);
112 written_bytes = writev(rtp_fd, vec, 2);
113 } while(written_bytes < 0 && errno == EINTR);
114 if(written_bytes < 0) {
115 error(errno, "error transmitting audio data");
118 fatal(0, "too many audio tranmission errors");
121 rtp_errors /= 2; /* gradual decay */
122 written_bytes -= sizeof (struct rtp_header);
123 const size_t written_samples = written_bytes / uaudio_sample_size;
124 uaudio_schedule_update(written_samples);
125 return written_samples;
128 static void rtp_open(void) {
129 struct addrinfo *res, *sres;
130 static const struct addrinfo pref = {
132 .ai_family = PF_INET,
133 .ai_socktype = SOCK_DGRAM,
134 .ai_protocol = IPPROTO_UDP,
136 static const struct addrinfo prefbind = {
137 .ai_flags = AI_PASSIVE,
138 .ai_family = PF_INET,
139 .ai_socktype = SOCK_DGRAM,
140 .ai_protocol = IPPROTO_UDP,
142 static const int one = 1;
143 int sndbuf, target_sndbuf = 131072;
145 char *sockname, *ssockname;
146 struct stringlist dst, src;
148 /* Get configuration */
150 dst.s = xcalloc(2, sizeof *dst.s);
151 dst.s[0] = uaudio_get("rtp-destination", NULL);
152 dst.s[1] = uaudio_get("rtp-destination-port", NULL);
154 src.s = xcalloc(2, sizeof *dst.s);
155 src.s[0] = uaudio_get("rtp-source", NULL);
156 src.s[1] = uaudio_get("rtp-source-port", NULL);
158 fatal(0, "'rtp-destination' not set");
160 fatal(0, "'rtp-destination-port' not set");
163 fatal(0, "'rtp-source-port' not set");
167 rtp_delay_threshold = atoi(uaudio_get("rtp-delay-threshold", "1000"));
168 /* ...microseconds */
170 /* Resolve addresses */
171 res = get_address(&dst, &pref, &sockname);
174 sres = get_address(&src, &prefbind, &ssockname);
178 /* Create the socket */
179 if((rtp_fd = socket(res->ai_family,
181 res->ai_protocol)) < 0)
182 fatal(errno, "error creating broadcast socket");
183 if(multicast(res->ai_addr)) {
184 /* Enable multicast options */
185 const int ttl = atoi(uaudio_get("multicast-ttl", "1"));
186 const int loop = !strcmp(uaudio_get("multicast-loop", "yes"), "yes");
187 switch(res->ai_family) {
189 if(setsockopt(rtp_fd, IPPROTO_IP, IP_MULTICAST_TTL,
190 &ttl, sizeof ttl) < 0)
191 fatal(errno, "error setting IP_MULTICAST_TTL on multicast socket");
192 if(setsockopt(rtp_fd, IPPROTO_IP, IP_MULTICAST_LOOP,
193 &loop, sizeof loop) < 0)
194 fatal(errno, "error setting IP_MULTICAST_LOOP on multicast socket");
198 if(setsockopt(rtp_fd, IPPROTO_IPV6, IPV6_MULTICAST_HOPS,
199 &ttl, sizeof ttl) < 0)
200 fatal(errno, "error setting IPV6_MULTICAST_HOPS on multicast socket");
201 if(setsockopt(rtp_fd, IPPROTO_IP, IPV6_MULTICAST_LOOP,
202 &loop, sizeof loop) < 0)
203 fatal(errno, "error setting IPV6_MULTICAST_LOOP on multicast socket");
207 fatal(0, "unsupported address family %d", res->ai_family);
209 info("multicasting on %s TTL=%d loop=%s",
210 sockname, ttl, loop ? "yes" : "no");
214 if(getifaddrs(&ifs) < 0)
215 fatal(errno, "error calling getifaddrs");
217 /* (At least on Darwin) IFF_BROADCAST might be set but ifa_broadaddr
218 * still a null pointer. It turns out that there's a subsequent entry
219 * for he same interface which _does_ have ifa_broadaddr though... */
220 if((ifs->ifa_flags & IFF_BROADCAST)
221 && ifs->ifa_broadaddr
222 && sockaddr_equal(ifs->ifa_broadaddr, res->ai_addr))
227 if(setsockopt(rtp_fd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0)
228 fatal(errno, "error setting SO_BROADCAST on broadcast socket");
229 info("broadcasting on %s (%s)", sockname, ifs->ifa_name);
231 info("unicasting on %s", sockname);
233 /* Enlarge the socket buffer */
235 if(getsockopt(rtp_fd, SOL_SOCKET, SO_SNDBUF,
237 fatal(errno, "error getting SO_SNDBUF");
238 if(target_sndbuf > sndbuf) {
239 if(setsockopt(rtp_fd, SOL_SOCKET, SO_SNDBUF,
240 &target_sndbuf, sizeof target_sndbuf) < 0)
241 error(errno, "error setting SO_SNDBUF to %d", target_sndbuf);
243 info("changed socket send buffer size from %d to %d",
244 sndbuf, target_sndbuf);
246 info("default socket send buffer is %d",
248 /* We might well want to set additional broadcast- or multicast-related
250 if(sres && bind(rtp_fd, sres->ai_addr, sres->ai_addrlen) < 0)
251 fatal(errno, "error binding broadcast socket to %s", ssockname);
252 if(connect(rtp_fd, res->ai_addr, res->ai_addrlen) < 0)
253 fatal(errno, "error connecting broadcast socket to %s", sockname);
256 static void rtp_start(uaudio_callback *callback,
258 /* We only support L16 (but we do stereo and mono and will convert sign) */
259 if(uaudio_channels == 2
261 && uaudio_rate == 44100)
263 else if(uaudio_channels == 1
265 && uaudio_rate == 44100)
268 fatal(0, "asked for %d/%d/%d 16/44100/1 and 16/44100/2",
269 uaudio_bits, uaudio_rate, uaudio_channels);
270 /* Various fields are required to have random initial values by RFC3550. The
271 * packet contents are highly public so there's no point asking for very
272 * strong randomness. */
273 gcry_create_nonce(&rtp_id, sizeof rtp_id);
274 gcry_create_nonce(&rtp_sequence, sizeof rtp_sequence);
276 uaudio_schedule_init();
277 uaudio_thread_start(callback,
280 256 / uaudio_sample_size,
281 (NETWORK_BYTES - sizeof(struct rtp_header))
282 / uaudio_sample_size);
285 static void rtp_stop(void) {
286 uaudio_thread_stop();
291 static void rtp_activate(void) {
292 uaudio_schedule_reactivated = 1;
293 uaudio_thread_activate();
296 static void rtp_deactivate(void) {
297 uaudio_thread_deactivate();
300 static void rtp_configure(void) {
303 uaudio_set("rtp-destination", config->broadcast.s[0]);
304 uaudio_set("rtp-destination-port", config->broadcast.s[1]);
305 if(config->broadcast_from.n) {
306 uaudio_set("rtp-source", config->broadcast_from.s[0]);
307 uaudio_set("rtp-source-port", config->broadcast_from.s[0]);
309 uaudio_set("rtp-source", NULL);
310 uaudio_set("rtp-source-port", NULL);
312 snprintf(buffer, sizeof buffer, "%ld", config->multicast_ttl);
313 uaudio_set("multicast-ttl", buffer);
314 uaudio_set("multicast-loop", config->multicast_loop ? "yes" : "no");
315 snprintf(buffer, sizeof buffer, "%ld", config->rtp_delay_threshold);
316 uaudio_set("delay-threshold", buffer);
319 const struct uaudio uaudio_rtp = {
321 .options = rtp_options,
324 .activate = rtp_activate,
325 .deactivate = rtp_deactivate,
326 .configure = rtp_configure,