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[disorder] / server / speaker-network.c
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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>
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 */
51static 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 */
64static 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 */
71static struct timeval rtp_time_0;
72
73/** @brief RTP packet sequence number */
74static uint16_t rtp_seq;
75
76/** @brief RTP SSRC */
77static uint32_t rtp_id;
78
79/** @brief Error counter */
80static int audio_errors;
81
82/** @brief Network backend initialization */
83static 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)) {
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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");
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128 if(setsockopt(bfd, IPPROTO_IP, IP_MULTICAST_LOOP,
129 &config->multicast_loop, sizeof one) < 0)
130 fatal(errno, "error setting IP_MULTICAST_LOOP on multicast socket");
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131 break;
132 }
133 case PF_INET6: {
134 const int mttl = config->multicast_ttl;
135 if(setsockopt(bfd, IPPROTO_IPV6, IPV6_MULTICAST_HOPS,
136 &mttl, sizeof mttl) < 0)
137 fatal(errno, "error setting IPV6_MULTICAST_HOPS on multicast socket");
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138 if(setsockopt(bfd, IPPROTO_IP, IPV6_MULTICAST_LOOP,
139 &config->multicast_loop, sizeof (int)) < 0)
140 fatal(errno, "error setting IPV6_MULTICAST_LOOP on multicast socket");
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141 break;
142 }
143 default:
144 fatal(0, "unsupported address family %d", res->ai_family);
145 }
81b1bf12 146 info("multicasting on %s", sockname);
23205f9c 147 } else {
db2c19dc 148 struct ifaddrs *ifs;
81b1bf12 149
db2c19dc
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150 if(getifaddrs(&ifs) < 0)
151 fatal(errno, "error calling getifaddrs");
152 while(ifs) {
3aa6f359
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153 /* (At least on Darwin) IFF_BROADCAST might be set but ifa_broadaddr
154 * still a null pointer. It turns out that there's a subsequent entry
155 * for he same interface which _does_ have ifa_broadaddr though... */
db2c19dc 156 if((ifs->ifa_flags & IFF_BROADCAST)
3aa6f359 157 && ifs->ifa_broadaddr
db2c19dc 158 && sockaddr_equal(ifs->ifa_broadaddr, res->ai_addr))
81b1bf12 159 break;
db2c19dc 160 ifs = ifs->ifa_next;
81b1bf12 161 }
db2c19dc 162 if(ifs) {
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163 if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0)
164 fatal(errno, "error setting SO_BROADCAST on broadcast socket");
db2c19dc 165 info("broadcasting on %s (%s)", sockname, ifs->ifa_name);
81b1bf12
RK
166 } else
167 info("unicasting on %s", sockname);
23205f9c 168 }
1c3f1e73 169 len = sizeof sndbuf;
170 if(getsockopt(bfd, SOL_SOCKET, SO_SNDBUF,
171 &sndbuf, &len) < 0)
172 fatal(errno, "error getting SO_SNDBUF");
173 if(target_sndbuf > sndbuf) {
174 if(setsockopt(bfd, SOL_SOCKET, SO_SNDBUF,
175 &target_sndbuf, sizeof target_sndbuf) < 0)
176 error(errno, "error setting SO_SNDBUF to %d", target_sndbuf);
177 else
178 info("changed socket send buffer size from %d to %d",
179 sndbuf, target_sndbuf);
180 } else
181 info("default socket send buffer is %d",
182 sndbuf);
183 /* We might well want to set additional broadcast- or multicast-related
184 * options here */
185 if(sres && bind(bfd, sres->ai_addr, sres->ai_addrlen) < 0)
186 fatal(errno, "error binding broadcast socket to %s", ssockname);
187 if(connect(bfd, res->ai_addr, res->ai_addrlen) < 0)
188 fatal(errno, "error connecting broadcast socket to %s", sockname);
189 /* Select an SSRC */
190 gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM);
1c3f1e73 191}
192
193/** @brief Play over the network */
194static size_t network_play(size_t frames) {
195 struct rtp_header header;
196 struct iovec vec[2];
6d2d327c 197 size_t bytes = frames * bpf, written_frames;
1c3f1e73 198 int written_bytes;
199 /* We transmit using RTP (RFC3550) and attempt to conform to the internet
200 * AVT profile (RFC3551). */
201
202 if(idled) {
203 /* There may have been a gap. Fix up the RTP time accordingly. */
204 struct timeval now;
205 uint64_t delta;
206 uint64_t target_rtp_time;
207
208 /* Find the current time */
209 xgettimeofday(&now, 0);
210 /* Find the number of microseconds elapsed since rtp_time=0 */
211 delta = tvsub_us(now, rtp_time_0);
212 assert(delta <= UINT64_MAX / 88200);
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213 target_rtp_time = (delta * config->sample_format.rate
214 * config->sample_format.channels) / 1000000;
1c3f1e73 215 /* Overflows at ~6 years uptime with 44100Hz stereo */
216
217 /* rtp_time is the number of samples we've played. NB that we play
218 * RTP_AHEAD_MS ahead of ourselves, so it may legitimately be ahead of
219 * the value we deduce from time comparison.
220 *
221 * Suppose we have 1s track started at t=0, and another track begins to
222 * play at t=2s. Suppose RTP_AHEAD_MS=1000 and 44100Hz stereo. In that
223 * case we'll send 1s of audio as fast as we can, giving rtp_time=88200.
224 * rtp_time stops at this point.
225 *
226 * At t=2s we'll have calculated target_rtp_time=176400. In this case we
227 * set rtp_time=176400 and the player can correctly conclude that it
228 * should leave 1s between the tracks.
229 *
230 * Suppose instead that the second track arrives at t=0.5s, and that
231 * we've managed to transmit the whole of the first track already. We'll
232 * have target_rtp_time=44100.
233 *
234 * The desired behaviour is to play the second track back to back with
235 * first. In this case therefore we do not modify rtp_time.
236 *
237 * Is it ever right to reduce rtp_time? No; for that would imply
238 * transmitting packets with overlapping timestamp ranges, which does not
239 * make sense.
240 */
241 target_rtp_time &= ~(uint64_t)1; /* stereo! */
242 if(target_rtp_time > rtp_time) {
243 /* More time has elapsed than we've transmitted samples. That implies
244 * we've been 'sending' silence. */
245 info("advancing rtp_time by %"PRIu64" samples",
246 target_rtp_time - rtp_time);
247 rtp_time = target_rtp_time;
248 } else if(target_rtp_time < rtp_time) {
249 const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS
250 * config->sample_format.rate
251 * config->sample_format.channels
252 / 1000);
253
254 if(target_rtp_time + samples_ahead < rtp_time) {
255 info("reversing rtp_time by %"PRIu64" samples",
256 rtp_time - target_rtp_time);
257 }
258 }
259 }
260 header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */
261 header.seq = htons(rtp_seq++);
262 header.timestamp = htonl((uint32_t)rtp_time);
263 header.ssrc = rtp_id;
264 header.mpt = (idled ? 0x80 : 0x00) | 10;
265 /* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from
266 * the sample rate (in a library somewhere so that configuration.c can rule
267 * out invalid rates).
268 */
269 idled = 0;
270 if(bytes > NETWORK_BYTES - sizeof header) {
271 bytes = NETWORK_BYTES - sizeof header;
272 /* Always send a whole number of frames */
6d2d327c 273 bytes -= bytes % bpf;
1c3f1e73 274 }
275 /* "The RTP clock rate used for generating the RTP timestamp is independent
276 * of the number of channels and the encoding; it equals the number of
277 * sampling periods per second. For N-channel encodings, each sampling
278 * period (say, 1/8000 of a second) generates N samples. (This terminology
279 * is standard, but somewhat confusing, as the total number of samples
280 * generated per second is then the sampling rate times the channel
281 * count.)"
282 */
283 vec[0].iov_base = (void *)&header;
284 vec[0].iov_len = sizeof header;
285 vec[1].iov_base = playing->buffer + playing->start;
286 vec[1].iov_len = bytes;
287 do {
288 written_bytes = writev(bfd, vec, 2);
289 } while(written_bytes < 0 && errno == EINTR);
290 if(written_bytes < 0) {
291 error(errno, "error transmitting audio data");
292 ++audio_errors;
293 if(audio_errors == 10)
294 fatal(0, "too many audio errors");
295 return 0;
296 } else
297 audio_errors /= 2;
298 written_bytes -= sizeof (struct rtp_header);
6d2d327c 299 written_frames = written_bytes / bpf;
1c3f1e73 300 /* Advance RTP's notion of the time */
6d2d327c 301 rtp_time += written_frames * config->sample_format.channels;
1c3f1e73 302 return written_frames;
303}
304
305static int bfd_slot;
306
307/** @brief Set up poll array for network play */
e84fb5f0 308static void network_beforepoll(int *timeoutp) {
1c3f1e73 309 struct timeval now;
310 uint64_t target_us;
311 uint64_t target_rtp_time;
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RK
312 const int64_t samples_per_second = config->sample_format.rate
313 * config->sample_format.channels;
1c3f1e73 314 const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS
e84fb5f0 315 * samples_per_second
1c3f1e73 316 / 1000);
e84fb5f0 317 int64_t lead, ahead_ms;
1c3f1e73 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;
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330 lead = rtp_time - target_rtp_time;
331 if(lead < samples_ahead)
332 /* We've not reached the desired lead, write as fast as we can */
1c3f1e73 333 bfd_slot = addfd(bfd, POLLOUT);
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RK
334 else {
335 /* We've reached the desired lead, we can afford to wait a bit even if the
336 * IP stack thinks it can accept more. */
337 ahead_ms = 1000 * (lead - samples_ahead) / samples_per_second;
338 if(ahead_ms < *timeoutp)
339 *timeoutp = ahead_ms;
340 }
1c3f1e73 341}
342
343/** @brief Process poll() results for network play */
344static int network_ready(void) {
345 if(fds[bfd_slot].revents & (POLLOUT | POLLERR))
346 return 1;
347 else
348 return 0;
349}
350
351const struct speaker_backend network_backend = {
352 BACKEND_NETWORK,
6d2d327c 353 0,
1c3f1e73 354 network_init,
355 0, /* activate */
356 network_play,
357 0, /* deactivate */
358 network_beforepoll,
359 network_ready
360};
361
362/*
363Local Variables:
364c-basic-offset:2
365comment-column:40
366fill-column:79
367indent-tabs-mode:nil
368End:
369*/