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7a2c7068 RK |
1 | /* |
2 | * This file is part of DisOrder. | |
3 | * Copyright (C) 2009 Richard Kettlewell | |
4 | * | |
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. | |
9 | * | |
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. | |
14 | * | |
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/>. | |
17 | */ | |
e8c185c3 | 18 | /** @file lib/uaudio-rtp.c |
7a2c7068 RK |
19 | * @brief Support for RTP network play backend */ |
20 | #include "common.h" | |
21 | ||
dfa51bb7 RK |
22 | #include <errno.h> |
23 | #include <ifaddrs.h> | |
24 | #include <net/if.h> | |
25 | #include <gcrypt.h> | |
26 | #include <unistd.h> | |
27 | #include <time.h> | |
7a2c7068 RK |
28 | |
29 | #include "uaudio.h" | |
30 | #include "mem.h" | |
31 | #include "log.h" | |
32 | #include "syscalls.h" | |
dfa51bb7 RK |
33 | #include "rtp.h" |
34 | #include "addr.h" | |
35 | #include "ifreq.h" | |
36 | #include "timeval.h" | |
37 | ||
38 | /** @brief Bytes to send per network packet | |
39 | * | |
40 | * This is the maximum number of bytes we pass to write(2); to determine actual | |
41 | * packet sizes, add a UDP header and an IP header (and a link layer header if | |
42 | * it's the link layer size you care about). | |
43 | * | |
44 | * Don't make this too big or arithmetic will start to overflow. | |
45 | */ | |
46 | #define NETWORK_BYTES (1500-8/*UDP*/-40/*IP*/-8/*conservatism*/) | |
47 | ||
48 | /** @brief RTP payload type */ | |
49 | static int rtp_payload; | |
50 | ||
51 | /** @brief RTP output socket */ | |
52 | static int rtp_fd; | |
53 | ||
54 | /** @brief RTP SSRC */ | |
55 | static uint32_t rtp_id; | |
56 | ||
57 | /** @brief RTP sequence number */ | |
58 | static uint16_t rtp_sequence; | |
59 | ||
60 | /** @brief RTP timestamp | |
61 | * | |
62 | * This is the timestamp that will be used on the next outbound packet. | |
63 | * | |
64 | * The timestamp in the packet header is only 32 bits wide. With 44100Hz | |
65 | * stereo, that only gives about half a day before wrapping, which is not | |
66 | * particularly convenient for certain debugging purposes. Therefore the | |
67 | * timestamp is maintained as a 64-bit integer, giving around six million years | |
68 | * before wrapping, and truncated to 32 bits when transmitting. | |
69 | */ | |
70 | static uint64_t rtp_timestamp; | |
71 | ||
72 | /** @brief Actual time corresponding to @ref rtp_timestamp | |
73 | * | |
74 | * This is the time, on this machine, at which the sample at @ref rtp_timestamp | |
75 | * ought to be sent, interpreted as the time the last packet was sent plus the | |
76 | * time length of the packet. */ | |
77 | static struct timeval rtp_timeval; | |
78 | ||
79 | /** @brief Set when we (re-)activate, to provoke timestamp resync */ | |
80 | static int rtp_reactivated; | |
81 | ||
82 | /** @brief Network error count | |
83 | * | |
84 | * If too many errors occur in too short a time, we give up. | |
85 | */ | |
86 | static int rtp_errors; | |
87 | ||
88 | /** @brief Delay threshold in microseconds | |
89 | * | |
90 | * rtp_play() never attempts to introduce a delay shorter than this. | |
91 | */ | |
92 | static int64_t rtp_delay_threshold; | |
7a2c7068 RK |
93 | |
94 | static const char *const rtp_options[] = { | |
dfa51bb7 RK |
95 | "rtp-destination", |
96 | "rtp-destination-port", | |
97 | "rtp-source", | |
98 | "rtp-source-port", | |
99 | "multicast-ttl", | |
100 | "multicast-loop", | |
101 | "rtp-delay-threshold", | |
7a2c7068 RK |
102 | NULL |
103 | }; | |
104 | ||
dfa51bb7 RK |
105 | static size_t rtp_play(void *buffer, size_t nsamples) { |
106 | struct rtp_header header; | |
107 | struct iovec vec[2]; | |
108 | struct timeval now; | |
109 | ||
110 | /* We do as much work as possible before checking what time it is */ | |
111 | /* Fill out header */ | |
112 | header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */ | |
113 | header.seq = htons(rtp_sequence++); | |
114 | header.ssrc = rtp_id; | |
115 | header.mpt = (rtp_reactivated ? 0x80 : 0x00) | rtp_payload; | |
116 | #if !WORDS_BIGENDIAN | |
117 | /* Convert samples to network byte order */ | |
118 | uint16_t *u = buffer, *const limit = u + nsamples; | |
119 | while(u < limit) { | |
120 | *u = htons(*u); | |
121 | ++u; | |
122 | } | |
123 | #endif | |
124 | vec[0].iov_base = (void *)&header; | |
125 | vec[0].iov_len = sizeof header; | |
126 | vec[1].iov_base = buffer; | |
127 | vec[1].iov_len = nsamples * uaudio_sample_size; | |
128 | retry: | |
129 | xgettimeofday(&now, NULL); | |
130 | if(rtp_reactivated) { | |
131 | /* We've been deactivated for some unknown interval. We need to advance | |
132 | * rtp_timestamp to account for the dead air. */ | |
133 | /* On the first run through we'll set the start time. */ | |
134 | if(!rtp_timeval.tv_sec) | |
135 | rtp_timeval = now; | |
136 | /* See how much time we missed. | |
137 | * | |
138 | * This will be 0 on the first run through, in which case we'll not modify | |
139 | * anything. | |
140 | * | |
141 | * It'll be negative in the (rare) situation where the deactivation | |
142 | * interval is shorter than the last packet we sent. In this case we wait | |
143 | * for that much time and then return having sent no samples, which will | |
144 | * cause uaudio_play_thread_fn() to retry. | |
145 | * | |
146 | * In the normal case it will be positive. | |
147 | */ | |
148 | const int64_t delay = tvsub_us(now, rtp_timeval); /* microseconds */ | |
149 | if(delay < 0) { | |
150 | usleep(-delay); | |
151 | goto retry; | |
152 | } | |
153 | /* Advance the RTP timestamp to the present. With 44.1KHz stereo this will | |
154 | * overflow the intermediate value with a delay of a bit over 6 years. | |
155 | * This seems acceptable. */ | |
156 | uint64_t update = (delay * uaudio_rate * uaudio_channels) / 1000000; | |
157 | /* Don't throw off channel synchronization */ | |
158 | update -= update % uaudio_channels; | |
159 | /* We log nontrivial changes */ | |
160 | if(update) | |
161 | info("advancing rtp_time by %"PRIu64" samples", update); | |
162 | rtp_timestamp += update; | |
163 | rtp_timeval = now; | |
164 | rtp_reactivated = 0; | |
165 | } else { | |
166 | /* Chances are we've been called right on the heels of the previous packet. | |
167 | * If we just sent packets as fast as we got audio data we'd get way ahead | |
168 | * of the player and some buffer somewhere would fill (or at least become | |
169 | * unreasonably large). | |
170 | * | |
171 | * First find out how far ahead of the target time we are. | |
172 | */ | |
173 | const int64_t ahead = tvsub_us(now, rtp_timeval); /* microseconds */ | |
174 | /* Only delay at all if we are nontrivially ahead. */ | |
175 | if(ahead > rtp_delay_threshold) { | |
176 | /* Don't delay by the full amount */ | |
177 | usleep(ahead - rtp_delay_threshold / 2); | |
178 | /* Refetch time (so we don't get out of step with reality) */ | |
179 | xgettimeofday(&now, NULL); | |
180 | } | |
181 | } | |
182 | header.timestamp = htonl((uint32_t)rtp_timestamp); | |
183 | int written_bytes; | |
184 | do { | |
185 | written_bytes = writev(rtp_fd, vec, 2); | |
186 | } while(written_bytes < 0 && errno == EINTR); | |
187 | if(written_bytes < 0) { | |
188 | error(errno, "error transmitting audio data"); | |
189 | ++rtp_errors; | |
190 | if(rtp_errors == 10) | |
191 | fatal(0, "too many audio tranmission errors"); | |
192 | return 0; | |
193 | } else | |
194 | rtp_errors /= 2; /* gradual decay */ | |
195 | written_bytes -= sizeof (struct rtp_header); | |
196 | size_t written_samples = written_bytes / uaudio_sample_size; | |
197 | /* rtp_timestamp and rtp_timestamp are supposed to refer to the first sample | |
198 | * of the next packet */ | |
199 | rtp_timestamp += written_samples; | |
200 | const unsigned usec = (rtp_timeval.tv_usec | |
201 | + 1000000 * written_samples / (uaudio_rate | |
202 | * uaudio_channels)); | |
203 | /* ...will only overflow 32 bits if one packet is more than about half an | |
204 | * hour long, which is not plausible. */ | |
205 | rtp_timeval.tv_sec += usec / 1000000; | |
206 | rtp_timeval.tv_usec = usec % 1000000; | |
207 | return written_samples; | |
208 | } | |
209 | ||
210 | static void rtp_open(void) { | |
211 | struct addrinfo *res, *sres; | |
212 | static const struct addrinfo pref = { | |
213 | .ai_flags = 0, | |
214 | .ai_family = PF_INET, | |
215 | .ai_socktype = SOCK_DGRAM, | |
216 | .ai_protocol = IPPROTO_UDP, | |
217 | }; | |
218 | static const struct addrinfo prefbind = { | |
219 | .ai_flags = AI_PASSIVE, | |
220 | .ai_family = PF_INET, | |
221 | .ai_socktype = SOCK_DGRAM, | |
222 | .ai_protocol = IPPROTO_UDP, | |
223 | }; | |
224 | static const int one = 1; | |
225 | int sndbuf, target_sndbuf = 131072; | |
226 | socklen_t len; | |
227 | char *sockname, *ssockname; | |
228 | struct stringlist dst, src; | |
229 | const char *delay; | |
230 | ||
231 | /* Get configuration */ | |
232 | dst.n = 2; | |
233 | dst.s = xcalloc(2, sizeof *dst.s); | |
234 | dst.s[0] = uaudio_get("rtp-destination"); | |
235 | dst.s[1] = uaudio_get("rtp-destination-port"); | |
236 | src.n = 2; | |
237 | src.s = xcalloc(2, sizeof *dst.s); | |
238 | src.s[0] = uaudio_get("rtp-source"); | |
239 | src.s[1] = uaudio_get("rtp-source-port"); | |
240 | if(!dst.s[0]) | |
241 | fatal(0, "'rtp-destination' not set"); | |
242 | if(!dst.s[1]) | |
243 | fatal(0, "'rtp-destination-port' not set"); | |
244 | if(src.s[0]) { | |
245 | if(!src.s[1]) | |
246 | fatal(0, "'rtp-source-port' not set"); | |
247 | src.n = 2; | |
248 | } else | |
249 | src.n = 0; | |
250 | if((delay = uaudio_get("rtp-delay-threshold"))) | |
251 | rtp_delay_threshold = atoi(delay); | |
252 | else | |
253 | rtp_delay_threshold = 1000; /* microseconds */ | |
254 | ||
255 | /* Resolve addresses */ | |
256 | res = get_address(&dst, &pref, &sockname); | |
257 | if(!res) exit(-1); | |
258 | if(src.n) { | |
259 | sres = get_address(&src, &prefbind, &ssockname); | |
260 | if(!sres) exit(-1); | |
261 | } else | |
262 | sres = 0; | |
263 | /* Create the socket */ | |
264 | if((rtp_fd = socket(res->ai_family, | |
265 | res->ai_socktype, | |
266 | res->ai_protocol)) < 0) | |
267 | fatal(errno, "error creating broadcast socket"); | |
268 | if(multicast(res->ai_addr)) { | |
269 | /* Enable multicast options */ | |
270 | const char *ttls = uaudio_get("multicast-ttl"); | |
271 | const int ttl = ttls ? atoi(ttls) : 1; | |
272 | const char *loops = uaudio_get("multicast-loop"); | |
273 | const int loop = loops ? !strcmp(loops, "yes") : 1; | |
274 | switch(res->ai_family) { | |
275 | case PF_INET: { | |
276 | if(setsockopt(rtp_fd, IPPROTO_IP, IP_MULTICAST_TTL, | |
277 | &ttl, sizeof ttl) < 0) | |
278 | fatal(errno, "error setting IP_MULTICAST_TTL on multicast socket"); | |
279 | if(setsockopt(rtp_fd, IPPROTO_IP, IP_MULTICAST_LOOP, | |
280 | &loop, sizeof loop) < 0) | |
281 | fatal(errno, "error setting IP_MULTICAST_LOOP on multicast socket"); | |
282 | break; | |
283 | } | |
284 | case PF_INET6: { | |
285 | if(setsockopt(rtp_fd, IPPROTO_IPV6, IPV6_MULTICAST_HOPS, | |
286 | &ttl, sizeof ttl) < 0) | |
287 | fatal(errno, "error setting IPV6_MULTICAST_HOPS on multicast socket"); | |
288 | if(setsockopt(rtp_fd, IPPROTO_IP, IPV6_MULTICAST_LOOP, | |
289 | &loop, sizeof loop) < 0) | |
290 | fatal(errno, "error setting IPV6_MULTICAST_LOOP on multicast socket"); | |
291 | break; | |
292 | } | |
293 | default: | |
294 | fatal(0, "unsupported address family %d", res->ai_family); | |
295 | } | |
296 | info("multicasting on %s TTL=%d loop=%s", | |
297 | sockname, ttl, loop ? "yes" : "no"); | |
298 | } else { | |
299 | struct ifaddrs *ifs; | |
300 | ||
301 | if(getifaddrs(&ifs) < 0) | |
302 | fatal(errno, "error calling getifaddrs"); | |
303 | while(ifs) { | |
304 | /* (At least on Darwin) IFF_BROADCAST might be set but ifa_broadaddr | |
305 | * still a null pointer. It turns out that there's a subsequent entry | |
306 | * for he same interface which _does_ have ifa_broadaddr though... */ | |
307 | if((ifs->ifa_flags & IFF_BROADCAST) | |
308 | && ifs->ifa_broadaddr | |
309 | && sockaddr_equal(ifs->ifa_broadaddr, res->ai_addr)) | |
310 | break; | |
311 | ifs = ifs->ifa_next; | |
312 | } | |
313 | if(ifs) { | |
314 | if(setsockopt(rtp_fd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0) | |
315 | fatal(errno, "error setting SO_BROADCAST on broadcast socket"); | |
316 | info("broadcasting on %s (%s)", sockname, ifs->ifa_name); | |
317 | } else | |
318 | info("unicasting on %s", sockname); | |
319 | } | |
320 | /* Enlarge the socket buffer */ | |
321 | len = sizeof sndbuf; | |
322 | if(getsockopt(rtp_fd, SOL_SOCKET, SO_SNDBUF, | |
323 | &sndbuf, &len) < 0) | |
324 | fatal(errno, "error getting SO_SNDBUF"); | |
325 | if(target_sndbuf > sndbuf) { | |
326 | if(setsockopt(rtp_fd, SOL_SOCKET, SO_SNDBUF, | |
327 | &target_sndbuf, sizeof target_sndbuf) < 0) | |
328 | error(errno, "error setting SO_SNDBUF to %d", target_sndbuf); | |
329 | else | |
330 | info("changed socket send buffer size from %d to %d", | |
331 | sndbuf, target_sndbuf); | |
332 | } else | |
333 | info("default socket send buffer is %d", | |
334 | sndbuf); | |
335 | /* We might well want to set additional broadcast- or multicast-related | |
336 | * options here */ | |
337 | if(sres && bind(rtp_fd, sres->ai_addr, sres->ai_addrlen) < 0) | |
338 | fatal(errno, "error binding broadcast socket to %s", ssockname); | |
339 | if(connect(rtp_fd, res->ai_addr, res->ai_addrlen) < 0) | |
340 | fatal(errno, "error connecting broadcast socket to %s", sockname); | |
341 | /* Various fields are required to have random initial values by RFC3550. The | |
342 | * packet contents are highly public so there's no point asking for very | |
343 | * strong randomness. */ | |
344 | gcry_create_nonce(&rtp_id, sizeof rtp_id); | |
345 | gcry_create_nonce(&rtp_sequence, sizeof rtp_sequence); | |
346 | gcry_create_nonce(&rtp_timestamp, sizeof rtp_timestamp); | |
347 | /* rtp_play() will spot this and choose an initial value */ | |
348 | rtp_timeval.tv_sec = 0; | |
349 | } | |
350 | ||
7a2c7068 RK |
351 | static void rtp_start(uaudio_callback *callback, |
352 | void *userdata) { | |
dfa51bb7 RK |
353 | /* We only support L16 (but we do stereo and mono and will convert sign) */ |
354 | if(uaudio_channels == 2 | |
355 | && uaudio_bits == 16 | |
356 | && uaudio_rate == 44100) | |
357 | rtp_payload = 10; | |
358 | else if(uaudio_channels == 1 | |
359 | && uaudio_bits == 16 | |
360 | && uaudio_rate == 44100) | |
361 | rtp_payload = 11; | |
362 | else | |
363 | fatal(0, "asked for %d/%d/%d 16/44100/1 and 16/44100/2", | |
364 | uaudio_bits, uaudio_rate, uaudio_channels); | |
365 | rtp_open(); | |
366 | uaudio_thread_start(callback, | |
367 | userdata, | |
368 | rtp_play, | |
369 | 256 / uaudio_sample_size, | |
370 | (NETWORK_BYTES - sizeof(struct rtp_header)) | |
371 | / uaudio_sample_size); | |
7a2c7068 RK |
372 | } |
373 | ||
374 | static void rtp_stop(void) { | |
dfa51bb7 RK |
375 | uaudio_thread_stop(); |
376 | close(rtp_fd); | |
377 | rtp_fd = -1; | |
7a2c7068 RK |
378 | } |
379 | ||
380 | static void rtp_activate(void) { | |
dfa51bb7 RK |
381 | rtp_reactivated = 1; |
382 | uaudio_thread_activate(); | |
7a2c7068 RK |
383 | } |
384 | ||
385 | static void rtp_deactivate(void) { | |
dfa51bb7 | 386 | uaudio_thread_deactivate(); |
7a2c7068 RK |
387 | } |
388 | ||
389 | const struct uaudio uaudio_rtp = { | |
390 | .name = "rtp", | |
391 | .options = rtp_options, | |
392 | .start = rtp_start, | |
393 | .stop = rtp_stop, | |
394 | .activate = rtp_activate, | |
395 | .deactivate = rtp_deactivate | |
396 | }; | |
397 | ||
398 | /* | |
399 | Local Variables: | |
400 | c-basic-offset:2 | |
401 | comment-column:40 | |
402 | fill-column:79 | |
403 | indent-tabs-mode:nil | |
404 | End: | |
405 | */ |