Commit | Line | Data |
---|---|---|
1c3f1e73 | 1 | /* |
2 | * This file is part of DisOrder | |
3 | * Copyright (C) 2005, 2006, 2007 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 2 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, but | |
11 | * WITHOUT ANY WARRANTY; without even the implied warranty of | |
12 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
13 | * 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, write to the Free Software | |
17 | * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 | |
18 | * USA | |
19 | */ | |
20 | /** @file server/speaker-network.c | |
21 | * @brief Support for @ref BACKEND_NETWORK */ | |
22 | ||
23 | #include <config.h> | |
24 | #include "types.h" | |
25 | ||
26 | #include <unistd.h> | |
27 | #include <poll.h> | |
28 | #include <netdb.h> | |
29 | #include <gcrypt.h> | |
30 | #include <sys/socket.h> | |
31 | #include <sys/uio.h> | |
32 | #include <assert.h> | |
81b1bf12 | 33 | #include <net/if.h> |
db2c19dc | 34 | #include <ifaddrs.h> |
6d2d327c | 35 | #include <errno.h> |
1c3f1e73 | 36 | |
37 | #include "configuration.h" | |
38 | #include "syscalls.h" | |
39 | #include "log.h" | |
40 | #include "addr.h" | |
41 | #include "timeval.h" | |
42 | #include "rtp.h" | |
81b1bf12 | 43 | #include "ifreq.h" |
1c3f1e73 | 44 | #include "speaker-protocol.h" |
45 | #include "speaker.h" | |
46 | ||
47 | /** @brief Network socket | |
48 | * | |
49 | * This is the file descriptor to write to for @ref BACKEND_NETWORK. | |
50 | */ | |
51 | static int bfd = -1; | |
52 | ||
53 | /** @brief RTP timestamp | |
54 | * | |
55 | * This counts the number of samples played (NB not the number of frames | |
56 | * played). | |
57 | * | |
58 | * The timestamp in the packet header is only 32 bits wide. With 44100Hz | |
59 | * stereo, that only gives about half a day before wrapping, which is not | |
60 | * particularly convenient for certain debugging purposes. Therefore the | |
61 | * timestamp is maintained as a 64-bit integer, giving around six million years | |
62 | * before wrapping, and truncated to 32 bits when transmitting. | |
63 | */ | |
64 | static uint64_t rtp_time; | |
65 | ||
66 | /** @brief RTP base timestamp | |
67 | * | |
68 | * This is the real time correspoding to an @ref rtp_time of 0. It is used | |
69 | * to recalculate the timestamp after idle periods. | |
70 | */ | |
71 | static struct timeval rtp_time_0; | |
72 | ||
73 | /** @brief RTP packet sequence number */ | |
74 | static uint16_t rtp_seq; | |
75 | ||
76 | /** @brief RTP SSRC */ | |
77 | static uint32_t rtp_id; | |
78 | ||
79 | /** @brief Error counter */ | |
80 | static int audio_errors; | |
81 | ||
82 | /** @brief Network backend initialization */ | |
83 | static void network_init(void) { | |
84 | struct addrinfo *res, *sres; | |
85 | static const struct addrinfo pref = { | |
86 | 0, | |
87 | PF_INET, | |
88 | SOCK_DGRAM, | |
89 | IPPROTO_UDP, | |
90 | 0, | |
91 | 0, | |
92 | 0, | |
93 | 0 | |
94 | }; | |
95 | static const struct addrinfo prefbind = { | |
96 | AI_PASSIVE, | |
97 | PF_INET, | |
98 | SOCK_DGRAM, | |
99 | IPPROTO_UDP, | |
100 | 0, | |
101 | 0, | |
102 | 0, | |
103 | 0 | |
104 | }; | |
105 | static const int one = 1; | |
db2c19dc | 106 | int sndbuf, target_sndbuf = 131072; |
1c3f1e73 | 107 | socklen_t len; |
108 | char *sockname, *ssockname; | |
109 | ||
110 | res = get_address(&config->broadcast, &pref, &sockname); | |
111 | if(!res) exit(-1); | |
112 | if(config->broadcast_from.n) { | |
113 | sres = get_address(&config->broadcast_from, &prefbind, &ssockname); | |
114 | if(!sres) exit(-1); | |
115 | } else | |
116 | sres = 0; | |
117 | if((bfd = socket(res->ai_family, | |
118 | res->ai_socktype, | |
119 | res->ai_protocol)) < 0) | |
120 | fatal(errno, "error creating broadcast socket"); | |
6fba990c | 121 | if(multicast(res->ai_addr)) { |
23205f9c RK |
122 | /* Multicasting */ |
123 | switch(res->ai_family) { | |
124 | case PF_INET: { | |
125 | const int mttl = config->multicast_ttl; | |
126 | if(setsockopt(bfd, IPPROTO_IP, IP_MULTICAST_TTL, &mttl, sizeof mttl) < 0) | |
127 | fatal(errno, "error setting IP_MULTICAST_TTL on multicast socket"); | |
128 | break; | |
129 | } | |
130 | case PF_INET6: { | |
131 | const int mttl = config->multicast_ttl; | |
132 | if(setsockopt(bfd, IPPROTO_IPV6, IPV6_MULTICAST_HOPS, | |
133 | &mttl, sizeof mttl) < 0) | |
134 | fatal(errno, "error setting IPV6_MULTICAST_HOPS on multicast socket"); | |
135 | break; | |
136 | } | |
137 | default: | |
138 | fatal(0, "unsupported address family %d", res->ai_family); | |
139 | } | |
81b1bf12 | 140 | info("multicasting on %s", sockname); |
23205f9c | 141 | } else { |
db2c19dc | 142 | struct ifaddrs *ifs; |
81b1bf12 | 143 | |
db2c19dc RK |
144 | if(getifaddrs(&ifs) < 0) |
145 | fatal(errno, "error calling getifaddrs"); | |
146 | while(ifs) { | |
3aa6f359 RK |
147 | /* (At least on Darwin) IFF_BROADCAST might be set but ifa_broadaddr |
148 | * still a null pointer. It turns out that there's a subsequent entry | |
149 | * for he same interface which _does_ have ifa_broadaddr though... */ | |
db2c19dc | 150 | if((ifs->ifa_flags & IFF_BROADCAST) |
3aa6f359 | 151 | && ifs->ifa_broadaddr |
db2c19dc | 152 | && sockaddr_equal(ifs->ifa_broadaddr, res->ai_addr)) |
81b1bf12 | 153 | break; |
db2c19dc | 154 | ifs = ifs->ifa_next; |
81b1bf12 | 155 | } |
db2c19dc | 156 | if(ifs) { |
81b1bf12 RK |
157 | if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0) |
158 | fatal(errno, "error setting SO_BROADCAST on broadcast socket"); | |
db2c19dc | 159 | info("broadcasting on %s (%s)", sockname, ifs->ifa_name); |
81b1bf12 RK |
160 | } else |
161 | info("unicasting on %s", sockname); | |
23205f9c | 162 | } |
1c3f1e73 | 163 | len = sizeof sndbuf; |
164 | if(getsockopt(bfd, SOL_SOCKET, SO_SNDBUF, | |
165 | &sndbuf, &len) < 0) | |
166 | fatal(errno, "error getting SO_SNDBUF"); | |
167 | if(target_sndbuf > sndbuf) { | |
168 | if(setsockopt(bfd, SOL_SOCKET, SO_SNDBUF, | |
169 | &target_sndbuf, sizeof target_sndbuf) < 0) | |
170 | error(errno, "error setting SO_SNDBUF to %d", target_sndbuf); | |
171 | else | |
172 | info("changed socket send buffer size from %d to %d", | |
173 | sndbuf, target_sndbuf); | |
174 | } else | |
175 | info("default socket send buffer is %d", | |
176 | sndbuf); | |
177 | /* We might well want to set additional broadcast- or multicast-related | |
178 | * options here */ | |
179 | if(sres && bind(bfd, sres->ai_addr, sres->ai_addrlen) < 0) | |
180 | fatal(errno, "error binding broadcast socket to %s", ssockname); | |
181 | if(connect(bfd, res->ai_addr, res->ai_addrlen) < 0) | |
182 | fatal(errno, "error connecting broadcast socket to %s", sockname); | |
183 | /* Select an SSRC */ | |
184 | gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM); | |
1c3f1e73 | 185 | } |
186 | ||
187 | /** @brief Play over the network */ | |
188 | static size_t network_play(size_t frames) { | |
189 | struct rtp_header header; | |
190 | struct iovec vec[2]; | |
6d2d327c | 191 | size_t bytes = frames * bpf, written_frames; |
1c3f1e73 | 192 | int written_bytes; |
193 | /* We transmit using RTP (RFC3550) and attempt to conform to the internet | |
194 | * AVT profile (RFC3551). */ | |
195 | ||
196 | if(idled) { | |
197 | /* There may have been a gap. Fix up the RTP time accordingly. */ | |
198 | struct timeval now; | |
199 | uint64_t delta; | |
200 | uint64_t target_rtp_time; | |
201 | ||
202 | /* Find the current time */ | |
203 | xgettimeofday(&now, 0); | |
204 | /* Find the number of microseconds elapsed since rtp_time=0 */ | |
205 | delta = tvsub_us(now, rtp_time_0); | |
206 | assert(delta <= UINT64_MAX / 88200); | |
6d2d327c RK |
207 | target_rtp_time = (delta * config->sample_format.rate |
208 | * config->sample_format.channels) / 1000000; | |
1c3f1e73 | 209 | /* Overflows at ~6 years uptime with 44100Hz stereo */ |
210 | ||
211 | /* rtp_time is the number of samples we've played. NB that we play | |
212 | * RTP_AHEAD_MS ahead of ourselves, so it may legitimately be ahead of | |
213 | * the value we deduce from time comparison. | |
214 | * | |
215 | * Suppose we have 1s track started at t=0, and another track begins to | |
216 | * play at t=2s. Suppose RTP_AHEAD_MS=1000 and 44100Hz stereo. In that | |
217 | * case we'll send 1s of audio as fast as we can, giving rtp_time=88200. | |
218 | * rtp_time stops at this point. | |
219 | * | |
220 | * At t=2s we'll have calculated target_rtp_time=176400. In this case we | |
221 | * set rtp_time=176400 and the player can correctly conclude that it | |
222 | * should leave 1s between the tracks. | |
223 | * | |
224 | * Suppose instead that the second track arrives at t=0.5s, and that | |
225 | * we've managed to transmit the whole of the first track already. We'll | |
226 | * have target_rtp_time=44100. | |
227 | * | |
228 | * The desired behaviour is to play the second track back to back with | |
229 | * first. In this case therefore we do not modify rtp_time. | |
230 | * | |
231 | * Is it ever right to reduce rtp_time? No; for that would imply | |
232 | * transmitting packets with overlapping timestamp ranges, which does not | |
233 | * make sense. | |
234 | */ | |
235 | target_rtp_time &= ~(uint64_t)1; /* stereo! */ | |
236 | if(target_rtp_time > rtp_time) { | |
237 | /* More time has elapsed than we've transmitted samples. That implies | |
238 | * we've been 'sending' silence. */ | |
239 | info("advancing rtp_time by %"PRIu64" samples", | |
240 | target_rtp_time - rtp_time); | |
241 | rtp_time = target_rtp_time; | |
242 | } else if(target_rtp_time < rtp_time) { | |
243 | const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS | |
244 | * config->sample_format.rate | |
245 | * config->sample_format.channels | |
246 | / 1000); | |
247 | ||
248 | if(target_rtp_time + samples_ahead < rtp_time) { | |
249 | info("reversing rtp_time by %"PRIu64" samples", | |
250 | rtp_time - target_rtp_time); | |
251 | } | |
252 | } | |
253 | } | |
254 | header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */ | |
255 | header.seq = htons(rtp_seq++); | |
256 | header.timestamp = htonl((uint32_t)rtp_time); | |
257 | header.ssrc = rtp_id; | |
258 | header.mpt = (idled ? 0x80 : 0x00) | 10; | |
259 | /* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from | |
260 | * the sample rate (in a library somewhere so that configuration.c can rule | |
261 | * out invalid rates). | |
262 | */ | |
263 | idled = 0; | |
264 | if(bytes > NETWORK_BYTES - sizeof header) { | |
265 | bytes = NETWORK_BYTES - sizeof header; | |
266 | /* Always send a whole number of frames */ | |
6d2d327c | 267 | bytes -= bytes % bpf; |
1c3f1e73 | 268 | } |
269 | /* "The RTP clock rate used for generating the RTP timestamp is independent | |
270 | * of the number of channels and the encoding; it equals the number of | |
271 | * sampling periods per second. For N-channel encodings, each sampling | |
272 | * period (say, 1/8000 of a second) generates N samples. (This terminology | |
273 | * is standard, but somewhat confusing, as the total number of samples | |
274 | * generated per second is then the sampling rate times the channel | |
275 | * count.)" | |
276 | */ | |
277 | vec[0].iov_base = (void *)&header; | |
278 | vec[0].iov_len = sizeof header; | |
279 | vec[1].iov_base = playing->buffer + playing->start; | |
280 | vec[1].iov_len = bytes; | |
281 | do { | |
282 | written_bytes = writev(bfd, vec, 2); | |
283 | } while(written_bytes < 0 && errno == EINTR); | |
284 | if(written_bytes < 0) { | |
285 | error(errno, "error transmitting audio data"); | |
286 | ++audio_errors; | |
287 | if(audio_errors == 10) | |
288 | fatal(0, "too many audio errors"); | |
289 | return 0; | |
290 | } else | |
291 | audio_errors /= 2; | |
292 | written_bytes -= sizeof (struct rtp_header); | |
6d2d327c | 293 | written_frames = written_bytes / bpf; |
1c3f1e73 | 294 | /* Advance RTP's notion of the time */ |
6d2d327c | 295 | rtp_time += written_frames * config->sample_format.channels; |
1c3f1e73 | 296 | return written_frames; |
297 | } | |
298 | ||
299 | static int bfd_slot; | |
300 | ||
301 | /** @brief Set up poll array for network play */ | |
e84fb5f0 | 302 | static void network_beforepoll(int *timeoutp) { |
1c3f1e73 | 303 | struct timeval now; |
304 | uint64_t target_us; | |
305 | uint64_t target_rtp_time; | |
e84fb5f0 RK |
306 | const int64_t samples_per_second = config->sample_format.rate |
307 | * config->sample_format.channels; | |
1c3f1e73 | 308 | const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS |
e84fb5f0 | 309 | * samples_per_second |
1c3f1e73 | 310 | / 1000); |
e84fb5f0 | 311 | int64_t lead, ahead_ms; |
1c3f1e73 | 312 | |
313 | /* If we're starting then initialize the base time */ | |
314 | if(!rtp_time) | |
315 | xgettimeofday(&rtp_time_0, 0); | |
316 | /* We send audio data whenever we get RTP_AHEAD seconds or more | |
317 | * behind */ | |
318 | xgettimeofday(&now, 0); | |
319 | target_us = tvsub_us(now, rtp_time_0); | |
320 | assert(target_us <= UINT64_MAX / 88200); | |
321 | target_rtp_time = (target_us * config->sample_format.rate | |
322 | * config->sample_format.channels) | |
323 | / 1000000; | |
e84fb5f0 RK |
324 | lead = rtp_time - target_rtp_time; |
325 | if(lead < samples_ahead) | |
326 | /* We've not reached the desired lead, write as fast as we can */ | |
1c3f1e73 | 327 | bfd_slot = addfd(bfd, POLLOUT); |
e84fb5f0 RK |
328 | else { |
329 | /* We've reached the desired lead, we can afford to wait a bit even if the | |
330 | * IP stack thinks it can accept more. */ | |
331 | ahead_ms = 1000 * (lead - samples_ahead) / samples_per_second; | |
332 | if(ahead_ms < *timeoutp) | |
333 | *timeoutp = ahead_ms; | |
334 | } | |
1c3f1e73 | 335 | } |
336 | ||
337 | /** @brief Process poll() results for network play */ | |
338 | static int network_ready(void) { | |
339 | if(fds[bfd_slot].revents & (POLLOUT | POLLERR)) | |
340 | return 1; | |
341 | else | |
342 | return 0; | |
343 | } | |
344 | ||
345 | const struct speaker_backend network_backend = { | |
346 | BACKEND_NETWORK, | |
6d2d327c | 347 | 0, |
1c3f1e73 | 348 | network_init, |
349 | 0, /* activate */ | |
350 | network_play, | |
351 | 0, /* deactivate */ | |
352 | network_beforepoll, | |
353 | network_ready | |
354 | }; | |
355 | ||
356 | /* | |
357 | Local Variables: | |
358 | c-basic-offset:2 | |
359 | comment-column:40 | |
360 | fill-column:79 | |
361 | indent-tabs-mode:nil | |
362 | End: | |
363 | */ |