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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> |
1c3f1e73 | 35 | |
36 | #include "configuration.h" | |
37 | #include "syscalls.h" | |
38 | #include "log.h" | |
39 | #include "addr.h" | |
40 | #include "timeval.h" | |
41 | #include "rtp.h" | |
81b1bf12 | 42 | #include "ifreq.h" |
1c3f1e73 | 43 | #include "speaker-protocol.h" |
44 | #include "speaker.h" | |
45 | ||
46 | /** @brief Network socket | |
47 | * | |
48 | * This is the file descriptor to write to for @ref BACKEND_NETWORK. | |
49 | */ | |
50 | static int bfd = -1; | |
51 | ||
52 | /** @brief RTP timestamp | |
53 | * | |
54 | * This counts the number of samples played (NB not the number of frames | |
55 | * played). | |
56 | * | |
57 | * The timestamp in the packet header is only 32 bits wide. With 44100Hz | |
58 | * stereo, that only gives about half a day before wrapping, which is not | |
59 | * particularly convenient for certain debugging purposes. Therefore the | |
60 | * timestamp is maintained as a 64-bit integer, giving around six million years | |
61 | * before wrapping, and truncated to 32 bits when transmitting. | |
62 | */ | |
63 | static uint64_t rtp_time; | |
64 | ||
65 | /** @brief RTP base timestamp | |
66 | * | |
67 | * This is the real time correspoding to an @ref rtp_time of 0. It is used | |
68 | * to recalculate the timestamp after idle periods. | |
69 | */ | |
70 | static struct timeval rtp_time_0; | |
71 | ||
72 | /** @brief RTP packet sequence number */ | |
73 | static uint16_t rtp_seq; | |
74 | ||
75 | /** @brief RTP SSRC */ | |
76 | static uint32_t rtp_id; | |
77 | ||
78 | /** @brief Error counter */ | |
79 | static int audio_errors; | |
80 | ||
81 | /** @brief Network backend initialization */ | |
82 | static void network_init(void) { | |
83 | struct addrinfo *res, *sres; | |
84 | static const struct addrinfo pref = { | |
85 | 0, | |
86 | PF_INET, | |
87 | SOCK_DGRAM, | |
88 | IPPROTO_UDP, | |
89 | 0, | |
90 | 0, | |
91 | 0, | |
92 | 0 | |
93 | }; | |
94 | static const struct addrinfo prefbind = { | |
95 | AI_PASSIVE, | |
96 | PF_INET, | |
97 | SOCK_DGRAM, | |
98 | IPPROTO_UDP, | |
99 | 0, | |
100 | 0, | |
101 | 0, | |
102 | 0 | |
103 | }; | |
104 | static const int one = 1; | |
db2c19dc | 105 | int sndbuf, target_sndbuf = 131072; |
1c3f1e73 | 106 | socklen_t len; |
107 | char *sockname, *ssockname; | |
108 | ||
803f6e52 | 109 | /* Override sample format */ |
110 | config->sample_format.rate = 44100; | |
111 | config->sample_format.channels = 2; | |
112 | config->sample_format.bits = 16; | |
113 | config->sample_format.byte_format = AO_FMT_BIG; | |
1c3f1e73 | 114 | res = get_address(&config->broadcast, &pref, &sockname); |
115 | if(!res) exit(-1); | |
116 | if(config->broadcast_from.n) { | |
117 | sres = get_address(&config->broadcast_from, &prefbind, &ssockname); | |
118 | if(!sres) exit(-1); | |
119 | } else | |
120 | sres = 0; | |
121 | if((bfd = socket(res->ai_family, | |
122 | res->ai_socktype, | |
123 | res->ai_protocol)) < 0) | |
124 | fatal(errno, "error creating broadcast socket"); | |
23205f9c RK |
125 | if((res->ai_family == PF_INET |
126 | && IN_MULTICAST( | |
127 | ntohl(((struct sockaddr_in *)res->ai_addr)->sin_addr.s_addr) | |
128 | )) | |
129 | || (res->ai_family == PF_INET6 | |
130 | && IN6_IS_ADDR_MULTICAST( | |
131 | &((struct sockaddr_in6 *)res->ai_addr)->sin6_addr | |
132 | ))) { | |
133 | /* Multicasting */ | |
134 | switch(res->ai_family) { | |
135 | case PF_INET: { | |
136 | const int mttl = config->multicast_ttl; | |
137 | if(setsockopt(bfd, IPPROTO_IP, IP_MULTICAST_TTL, &mttl, sizeof mttl) < 0) | |
138 | fatal(errno, "error setting IP_MULTICAST_TTL on multicast socket"); | |
139 | break; | |
140 | } | |
141 | case PF_INET6: { | |
142 | const int mttl = config->multicast_ttl; | |
143 | if(setsockopt(bfd, IPPROTO_IPV6, IPV6_MULTICAST_HOPS, | |
144 | &mttl, sizeof mttl) < 0) | |
145 | fatal(errno, "error setting IPV6_MULTICAST_HOPS on multicast socket"); | |
146 | break; | |
147 | } | |
148 | default: | |
149 | fatal(0, "unsupported address family %d", res->ai_family); | |
150 | } | |
81b1bf12 | 151 | info("multicasting on %s", sockname); |
23205f9c | 152 | } else { |
db2c19dc | 153 | struct ifaddrs *ifs; |
81b1bf12 | 154 | |
db2c19dc RK |
155 | if(getifaddrs(&ifs) < 0) |
156 | fatal(errno, "error calling getifaddrs"); | |
157 | while(ifs) { | |
158 | if((ifs->ifa_flags & IFF_BROADCAST) | |
159 | && sockaddr_equal(ifs->ifa_broadaddr, res->ai_addr)) | |
81b1bf12 | 160 | break; |
db2c19dc | 161 | ifs = ifs->ifa_next; |
81b1bf12 | 162 | } |
db2c19dc | 163 | if(ifs) { |
81b1bf12 RK |
164 | if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0) |
165 | fatal(errno, "error setting SO_BROADCAST on broadcast socket"); | |
db2c19dc | 166 | info("broadcasting on %s (%s)", sockname, ifs->ifa_name); |
81b1bf12 RK |
167 | } else |
168 | info("unicasting on %s", sockname); | |
23205f9c | 169 | } |
1c3f1e73 | 170 | len = sizeof sndbuf; |
171 | if(getsockopt(bfd, SOL_SOCKET, SO_SNDBUF, | |
172 | &sndbuf, &len) < 0) | |
173 | fatal(errno, "error getting SO_SNDBUF"); | |
174 | if(target_sndbuf > sndbuf) { | |
175 | if(setsockopt(bfd, SOL_SOCKET, SO_SNDBUF, | |
176 | &target_sndbuf, sizeof target_sndbuf) < 0) | |
177 | error(errno, "error setting SO_SNDBUF to %d", target_sndbuf); | |
178 | else | |
179 | info("changed socket send buffer size from %d to %d", | |
180 | sndbuf, target_sndbuf); | |
181 | } else | |
182 | info("default socket send buffer is %d", | |
183 | sndbuf); | |
184 | /* We might well want to set additional broadcast- or multicast-related | |
185 | * options here */ | |
186 | if(sres && bind(bfd, sres->ai_addr, sres->ai_addrlen) < 0) | |
187 | fatal(errno, "error binding broadcast socket to %s", ssockname); | |
188 | if(connect(bfd, res->ai_addr, res->ai_addrlen) < 0) | |
189 | fatal(errno, "error connecting broadcast socket to %s", sockname); | |
190 | /* Select an SSRC */ | |
191 | gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM); | |
192 | info("selected network backend, sending to %s", sockname); | |
1c3f1e73 | 193 | } |
194 | ||
195 | /** @brief Play over the network */ | |
196 | static size_t network_play(size_t frames) { | |
197 | struct rtp_header header; | |
198 | struct iovec vec[2]; | |
199 | size_t bytes = frames * device_bpf, written_frames; | |
200 | int written_bytes; | |
201 | /* We transmit using RTP (RFC3550) and attempt to conform to the internet | |
202 | * AVT profile (RFC3551). */ | |
203 | ||
204 | if(idled) { | |
205 | /* There may have been a gap. Fix up the RTP time accordingly. */ | |
206 | struct timeval now; | |
207 | uint64_t delta; | |
208 | uint64_t target_rtp_time; | |
209 | ||
210 | /* Find the current time */ | |
211 | xgettimeofday(&now, 0); | |
212 | /* Find the number of microseconds elapsed since rtp_time=0 */ | |
213 | delta = tvsub_us(now, rtp_time_0); | |
214 | assert(delta <= UINT64_MAX / 88200); | |
215 | target_rtp_time = (delta * playing->format.rate | |
216 | * playing->format.channels) / 1000000; | |
217 | /* Overflows at ~6 years uptime with 44100Hz stereo */ | |
218 | ||
219 | /* rtp_time is the number of samples we've played. NB that we play | |
220 | * RTP_AHEAD_MS ahead of ourselves, so it may legitimately be ahead of | |
221 | * the value we deduce from time comparison. | |
222 | * | |
223 | * Suppose we have 1s track started at t=0, and another track begins to | |
224 | * play at t=2s. Suppose RTP_AHEAD_MS=1000 and 44100Hz stereo. In that | |
225 | * case we'll send 1s of audio as fast as we can, giving rtp_time=88200. | |
226 | * rtp_time stops at this point. | |
227 | * | |
228 | * At t=2s we'll have calculated target_rtp_time=176400. In this case we | |
229 | * set rtp_time=176400 and the player can correctly conclude that it | |
230 | * should leave 1s between the tracks. | |
231 | * | |
232 | * Suppose instead that the second track arrives at t=0.5s, and that | |
233 | * we've managed to transmit the whole of the first track already. We'll | |
234 | * have target_rtp_time=44100. | |
235 | * | |
236 | * The desired behaviour is to play the second track back to back with | |
237 | * first. In this case therefore we do not modify rtp_time. | |
238 | * | |
239 | * Is it ever right to reduce rtp_time? No; for that would imply | |
240 | * transmitting packets with overlapping timestamp ranges, which does not | |
241 | * make sense. | |
242 | */ | |
243 | target_rtp_time &= ~(uint64_t)1; /* stereo! */ | |
244 | if(target_rtp_time > rtp_time) { | |
245 | /* More time has elapsed than we've transmitted samples. That implies | |
246 | * we've been 'sending' silence. */ | |
247 | info("advancing rtp_time by %"PRIu64" samples", | |
248 | target_rtp_time - rtp_time); | |
249 | rtp_time = target_rtp_time; | |
250 | } else if(target_rtp_time < rtp_time) { | |
251 | const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS | |
252 | * config->sample_format.rate | |
253 | * config->sample_format.channels | |
254 | / 1000); | |
255 | ||
256 | if(target_rtp_time + samples_ahead < rtp_time) { | |
257 | info("reversing rtp_time by %"PRIu64" samples", | |
258 | rtp_time - target_rtp_time); | |
259 | } | |
260 | } | |
261 | } | |
262 | header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */ | |
263 | header.seq = htons(rtp_seq++); | |
264 | header.timestamp = htonl((uint32_t)rtp_time); | |
265 | header.ssrc = rtp_id; | |
266 | header.mpt = (idled ? 0x80 : 0x00) | 10; | |
267 | /* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from | |
268 | * the sample rate (in a library somewhere so that configuration.c can rule | |
269 | * out invalid rates). | |
270 | */ | |
271 | idled = 0; | |
272 | if(bytes > NETWORK_BYTES - sizeof header) { | |
273 | bytes = NETWORK_BYTES - sizeof header; | |
274 | /* Always send a whole number of frames */ | |
275 | bytes -= bytes % device_bpf; | |
276 | } | |
277 | /* "The RTP clock rate used for generating the RTP timestamp is independent | |
278 | * of the number of channels and the encoding; it equals the number of | |
279 | * sampling periods per second. For N-channel encodings, each sampling | |
280 | * period (say, 1/8000 of a second) generates N samples. (This terminology | |
281 | * is standard, but somewhat confusing, as the total number of samples | |
282 | * generated per second is then the sampling rate times the channel | |
283 | * count.)" | |
284 | */ | |
285 | vec[0].iov_base = (void *)&header; | |
286 | vec[0].iov_len = sizeof header; | |
287 | vec[1].iov_base = playing->buffer + playing->start; | |
288 | vec[1].iov_len = bytes; | |
289 | do { | |
290 | written_bytes = writev(bfd, vec, 2); | |
291 | } while(written_bytes < 0 && errno == EINTR); | |
292 | if(written_bytes < 0) { | |
293 | error(errno, "error transmitting audio data"); | |
294 | ++audio_errors; | |
295 | if(audio_errors == 10) | |
296 | fatal(0, "too many audio errors"); | |
297 | return 0; | |
298 | } else | |
299 | audio_errors /= 2; | |
300 | written_bytes -= sizeof (struct rtp_header); | |
301 | written_frames = written_bytes / device_bpf; | |
302 | /* Advance RTP's notion of the time */ | |
303 | rtp_time += written_frames * playing->format.channels; | |
304 | return written_frames; | |
305 | } | |
306 | ||
307 | static int bfd_slot; | |
308 | ||
309 | /** @brief Set up poll array for network play */ | |
310 | static void network_beforepoll(void) { | |
311 | struct timeval now; | |
312 | uint64_t target_us; | |
313 | uint64_t target_rtp_time; | |
314 | const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS | |
315 | * config->sample_format.rate | |
316 | * config->sample_format.channels | |
317 | / 1000); | |
318 | ||
319 | /* If we're starting then initialize the base time */ | |
320 | if(!rtp_time) | |
321 | xgettimeofday(&rtp_time_0, 0); | |
322 | /* We send audio data whenever we get RTP_AHEAD seconds or more | |
323 | * behind */ | |
324 | xgettimeofday(&now, 0); | |
325 | target_us = tvsub_us(now, rtp_time_0); | |
326 | assert(target_us <= UINT64_MAX / 88200); | |
327 | target_rtp_time = (target_us * config->sample_format.rate | |
328 | * config->sample_format.channels) | |
329 | / 1000000; | |
330 | if((int64_t)(rtp_time - target_rtp_time) < samples_ahead) | |
331 | bfd_slot = addfd(bfd, POLLOUT); | |
332 | } | |
333 | ||
334 | /** @brief Process poll() results for network play */ | |
335 | static int network_ready(void) { | |
336 | if(fds[bfd_slot].revents & (POLLOUT | POLLERR)) | |
337 | return 1; | |
338 | else | |
339 | return 0; | |
340 | } | |
341 | ||
342 | const struct speaker_backend network_backend = { | |
343 | BACKEND_NETWORK, | |
344 | FIXED_FORMAT, | |
345 | network_init, | |
346 | 0, /* activate */ | |
347 | network_play, | |
348 | 0, /* deactivate */ | |
349 | network_beforepoll, | |
350 | network_ready | |
351 | }; | |
352 | ||
353 | /* | |
354 | Local Variables: | |
355 | c-basic-offset:2 | |
356 | comment-column:40 | |
357 | fill-column:79 | |
358 | indent-tabs-mode:nil | |
359 | End: | |
360 | */ |