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disorder-choose rejects tracks in queue/recent list
[disorder] / server / speaker-network.c
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1c3f1e73 1/*
2 * This file is part of DisOrder
5aff007d 3 * Copyright (C) 2005-2008 Richard Kettlewell
1c3f1e73 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>
edbd470f 36#include <netinet/in.h>
1c3f1e73 37
38#include "configuration.h"
39#include "syscalls.h"
40#include "log.h"
41#include "addr.h"
42#include "timeval.h"
43#include "rtp.h"
81b1bf12 44#include "ifreq.h"
1c3f1e73 45#include "speaker-protocol.h"
46#include "speaker.h"
47
48/** @brief Network socket
49 *
50 * This is the file descriptor to write to for @ref BACKEND_NETWORK.
51 */
52static int bfd = -1;
53
54/** @brief RTP timestamp
55 *
56 * This counts the number of samples played (NB not the number of frames
57 * played).
58 *
59 * The timestamp in the packet header is only 32 bits wide. With 44100Hz
60 * stereo, that only gives about half a day before wrapping, which is not
61 * particularly convenient for certain debugging purposes. Therefore the
62 * timestamp is maintained as a 64-bit integer, giving around six million years
63 * before wrapping, and truncated to 32 bits when transmitting.
64 */
65static uint64_t rtp_time;
66
67/** @brief RTP base timestamp
68 *
69 * This is the real time correspoding to an @ref rtp_time of 0. It is used
70 * to recalculate the timestamp after idle periods.
71 */
72static struct timeval rtp_time_0;
73
74/** @brief RTP packet sequence number */
75static uint16_t rtp_seq;
76
77/** @brief RTP SSRC */
78static uint32_t rtp_id;
79
80/** @brief Error counter */
81static int audio_errors;
82
83/** @brief Network backend initialization */
84static void network_init(void) {
85 struct addrinfo *res, *sres;
86 static const struct addrinfo pref = {
87 0,
88 PF_INET,
89 SOCK_DGRAM,
90 IPPROTO_UDP,
91 0,
92 0,
93 0,
94 0
95 };
96 static const struct addrinfo prefbind = {
97 AI_PASSIVE,
98 PF_INET,
99 SOCK_DGRAM,
100 IPPROTO_UDP,
101 0,
102 0,
103 0,
104 0
105 };
106 static const int one = 1;
db2c19dc 107 int sndbuf, target_sndbuf = 131072;
1c3f1e73 108 socklen_t len;
109 char *sockname, *ssockname;
110
111 res = get_address(&config->broadcast, &pref, &sockname);
112 if(!res) exit(-1);
113 if(config->broadcast_from.n) {
114 sres = get_address(&config->broadcast_from, &prefbind, &ssockname);
115 if(!sres) exit(-1);
116 } else
117 sres = 0;
118 if((bfd = socket(res->ai_family,
119 res->ai_socktype,
120 res->ai_protocol)) < 0)
121 fatal(errno, "error creating broadcast socket");
6fba990c 122 if(multicast(res->ai_addr)) {
23205f9c
RK
123 /* Multicasting */
124 switch(res->ai_family) {
125 case PF_INET: {
126 const int mttl = config->multicast_ttl;
127 if(setsockopt(bfd, IPPROTO_IP, IP_MULTICAST_TTL, &mttl, sizeof mttl) < 0)
128 fatal(errno, "error setting IP_MULTICAST_TTL on multicast socket");
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RK
129 if(setsockopt(bfd, IPPROTO_IP, IP_MULTICAST_LOOP,
130 &config->multicast_loop, sizeof one) < 0)
131 fatal(errno, "error setting IP_MULTICAST_LOOP on multicast socket");
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132 break;
133 }
134 case PF_INET6: {
135 const int mttl = config->multicast_ttl;
136 if(setsockopt(bfd, IPPROTO_IPV6, IPV6_MULTICAST_HOPS,
137 &mttl, sizeof mttl) < 0)
138 fatal(errno, "error setting IPV6_MULTICAST_HOPS on multicast socket");
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139 if(setsockopt(bfd, IPPROTO_IP, IPV6_MULTICAST_LOOP,
140 &config->multicast_loop, sizeof (int)) < 0)
141 fatal(errno, "error setting IPV6_MULTICAST_LOOP on multicast socket");
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RK
142 break;
143 }
144 default:
145 fatal(0, "unsupported address family %d", res->ai_family);
146 }
81b1bf12 147 info("multicasting on %s", sockname);
23205f9c 148 } else {
db2c19dc 149 struct ifaddrs *ifs;
81b1bf12 150
db2c19dc
RK
151 if(getifaddrs(&ifs) < 0)
152 fatal(errno, "error calling getifaddrs");
153 while(ifs) {
3aa6f359
RK
154 /* (At least on Darwin) IFF_BROADCAST might be set but ifa_broadaddr
155 * still a null pointer. It turns out that there's a subsequent entry
156 * for he same interface which _does_ have ifa_broadaddr though... */
db2c19dc 157 if((ifs->ifa_flags & IFF_BROADCAST)
3aa6f359 158 && ifs->ifa_broadaddr
db2c19dc 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);
1c3f1e73 192}
193
194/** @brief Play over the network */
195static size_t network_play(size_t frames) {
196 struct rtp_header header;
197 struct iovec vec[2];
6d2d327c 198 size_t bytes = frames * bpf, written_frames;
1c3f1e73 199 int written_bytes;
200 /* We transmit using RTP (RFC3550) and attempt to conform to the internet
201 * AVT profile (RFC3551). */
202
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203 /* If we're starting then initialize the base time */
204 if(!rtp_time)
205 xgettimeofday(&rtp_time_0, 0);
1c3f1e73 206 if(idled) {
207 /* There may have been a gap. Fix up the RTP time accordingly. */
208 struct timeval now;
209 uint64_t delta;
210 uint64_t target_rtp_time;
211
212 /* Find the current time */
213 xgettimeofday(&now, 0);
214 /* Find the number of microseconds elapsed since rtp_time=0 */
215 delta = tvsub_us(now, rtp_time_0);
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216 if(delta > UINT64_MAX / 88200)
217 fatal(0, "rtp_time=%llu now=%ld.%06ld rtp_time_0=%ld.%06ld delta=%llu (%lld)",
218 rtp_time,
219 (long)now.tv_sec, (long)now.tv_usec,
220 (long)rtp_time_0.tv_sec, (long)rtp_time_0.tv_usec,
221 delta, delta);
6d2d327c
RK
222 target_rtp_time = (delta * config->sample_format.rate
223 * config->sample_format.channels) / 1000000;
1c3f1e73 224 /* Overflows at ~6 years uptime with 44100Hz stereo */
225
226 /* rtp_time is the number of samples we've played. NB that we play
227 * RTP_AHEAD_MS ahead of ourselves, so it may legitimately be ahead of
228 * the value we deduce from time comparison.
229 *
230 * Suppose we have 1s track started at t=0, and another track begins to
24d6fd25
RK
231 * play at t=2s. Suppose 44100Hz stereo. We send 1s of audio over the
232 * next (about) one second, giving rtp_time=88200. rtp_time stops at this
233 * point.
1c3f1e73 234 *
235 * At t=2s we'll have calculated target_rtp_time=176400. In this case we
236 * set rtp_time=176400 and the player can correctly conclude that it
237 * should leave 1s between the tracks.
238 *
24d6fd25
RK
239 * It's never right to reduce rtp_time, for that would imply packets with
240 * overlapping timestamp ranges, which does not make sense.
1c3f1e73 241 */
242 target_rtp_time &= ~(uint64_t)1; /* stereo! */
243 if(target_rtp_time > rtp_time) {
244 /* More time has elapsed than we've transmitted samples. That implies
245 * we've been 'sending' silence. */
246 info("advancing rtp_time by %"PRIu64" samples",
247 target_rtp_time - rtp_time);
248 rtp_time = target_rtp_time;
249 } else if(target_rtp_time < rtp_time) {
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RK
250 info("would reverse rtp_time by %"PRIu64" samples",
251 rtp_time - target_rtp_time);
1c3f1e73 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
299static int bfd_slot;
300
301/** @brief Set up poll array for network play */
e84fb5f0 302static 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;
e84fb5f0 308 int64_t lead, ahead_ms;
1c3f1e73 309
310 /* If we're starting then initialize the base time */
311 if(!rtp_time)
312 xgettimeofday(&rtp_time_0, 0);
24d6fd25 313 /* We send audio data whenever we would otherwise get behind */
1c3f1e73 314 xgettimeofday(&now, 0);
315 target_us = tvsub_us(now, rtp_time_0);
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RK
316 if(target_us > UINT64_MAX / 88200)
317 fatal(0, "rtp_time=%llu rtp_time_0=%ld.%06ld now=%ld.%06ld target_us=%llu (%lld)\n",
318 rtp_time,
319 (long)rtp_time_0.tv_sec, (long)rtp_time_0.tv_usec,
320 (long)now.tv_sec, (long)now.tv_usec,
321 target_us, target_us);
1c3f1e73 322 target_rtp_time = (target_us * config->sample_format.rate
323 * config->sample_format.channels)
324 / 1000000;
24d6fd25 325 /* Lead is how far ahead we are */
e84fb5f0 326 lead = rtp_time - target_rtp_time;
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RK
327 if(lead <= 0)
328 /* We're behind or even, so we'll need to write as soon as we can */
1c3f1e73 329 bfd_slot = addfd(bfd, POLLOUT);
e84fb5f0 330 else {
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331 /* We've ahead, we can afford to wait a bit even if the IP stack thinks it
332 * can accept more. */
333 ahead_ms = 1000 * lead / samples_per_second;
e84fb5f0
RK
334 if(ahead_ms < *timeoutp)
335 *timeoutp = ahead_ms;
336 }
1c3f1e73 337}
338
339/** @brief Process poll() results for network play */
340static int network_ready(void) {
341 if(fds[bfd_slot].revents & (POLLOUT | POLLERR))
342 return 1;
343 else
344 return 0;
345}
346
347const struct speaker_backend network_backend = {
348 BACKEND_NETWORK,
6d2d327c 349 0,
1c3f1e73 350 network_init,
351 0, /* activate */
352 network_play,
353 0, /* deactivate */
354 network_beforepoll,
355 network_ready
356};
357
358/*
359Local Variables:
360c-basic-offset:2
361comment-column:40
362fill-column:79
363indent-tabs-mode:nil
364End:
365*/