2 * This file is part of DisOrder
3 * Copyright (C) 2005, 2006, 2007 Richard Kettlewell
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.
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.
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
20 /** @file server/speaker-network.c
21 * @brief Support for @ref BACKEND_NETWORK */
30 #include <sys/socket.h>
37 #include "configuration.h"
44 #include "speaker-protocol.h"
47 /** @brief Network socket
49 * This is the file descriptor to write to for @ref BACKEND_NETWORK.
53 /** @brief RTP timestamp
55 * This counts the number of samples played (NB not the number of frames
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.
64 static uint64_t rtp_time;
66 /** @brief RTP base timestamp
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.
71 static struct timeval rtp_time_0;
73 /** @brief RTP packet sequence number */
74 static uint16_t rtp_seq;
76 /** @brief RTP SSRC */
77 static uint32_t rtp_id;
79 /** @brief Error counter */
80 static int audio_errors;
82 /** @brief Network backend initialization */
83 static void network_init(void) {
84 struct addrinfo *res, *sres;
85 static const struct addrinfo pref = {
95 static const struct addrinfo prefbind = {
105 static const int one = 1;
106 int sndbuf, target_sndbuf = 131072;
108 char *sockname, *ssockname;
110 res = get_address(&config->broadcast, &pref, &sockname);
112 if(config->broadcast_from.n) {
113 sres = get_address(&config->broadcast_from, &prefbind, &ssockname);
117 if((bfd = socket(res->ai_family,
119 res->ai_protocol)) < 0)
120 fatal(errno, "error creating broadcast socket");
121 if((res->ai_family == PF_INET
123 ntohl(((struct sockaddr_in *)res->ai_addr)->sin_addr.s_addr)
125 || (res->ai_family == PF_INET6
126 && IN6_IS_ADDR_MULTICAST(
127 &((struct sockaddr_in6 *)res->ai_addr)->sin6_addr
130 switch(res->ai_family) {
132 const int mttl = config->multicast_ttl;
133 if(setsockopt(bfd, IPPROTO_IP, IP_MULTICAST_TTL, &mttl, sizeof mttl) < 0)
134 fatal(errno, "error setting IP_MULTICAST_TTL on multicast socket");
138 const int mttl = config->multicast_ttl;
139 if(setsockopt(bfd, IPPROTO_IPV6, IPV6_MULTICAST_HOPS,
140 &mttl, sizeof mttl) < 0)
141 fatal(errno, "error setting IPV6_MULTICAST_HOPS on multicast socket");
145 fatal(0, "unsupported address family %d", res->ai_family);
147 info("multicasting on %s", sockname);
151 if(getifaddrs(&ifs) < 0)
152 fatal(errno, "error calling getifaddrs");
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... */
157 if((ifs->ifa_flags & IFF_BROADCAST)
158 && ifs->ifa_broadaddr
159 && sockaddr_equal(ifs->ifa_broadaddr, res->ai_addr))
164 if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0)
165 fatal(errno, "error setting SO_BROADCAST on broadcast socket");
166 info("broadcasting on %s (%s)", sockname, ifs->ifa_name);
168 info("unicasting on %s", sockname);
171 if(getsockopt(bfd, SOL_SOCKET, SO_SNDBUF,
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);
179 info("changed socket send buffer size from %d to %d",
180 sndbuf, target_sndbuf);
182 info("default socket send buffer is %d",
184 /* We might well want to set additional broadcast- or multicast-related
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);
191 gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM);
192 info("selected network backend, sending to %s", sockname);
195 /** @brief Play over the network */
196 static size_t network_play(size_t frames) {
197 struct rtp_header header;
199 size_t bytes = frames * bpf, written_frames;
201 /* We transmit using RTP (RFC3550) and attempt to conform to the internet
202 * AVT profile (RFC3551). */
205 /* There may have been a gap. Fix up the RTP time accordingly. */
208 uint64_t target_rtp_time;
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 * config->sample_format.rate
216 * config->sample_format.channels) / 1000000;
217 /* Overflows at ~6 years uptime with 44100Hz stereo */
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.
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.
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.
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.
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.
239 * Is it ever right to reduce rtp_time? No; for that would imply
240 * transmitting packets with overlapping timestamp ranges, which does not
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
256 if(target_rtp_time + samples_ahead < rtp_time) {
257 info("reversing rtp_time by %"PRIu64" samples",
258 rtp_time - target_rtp_time);
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).
272 if(bytes > NETWORK_BYTES - sizeof header) {
273 bytes = NETWORK_BYTES - sizeof header;
274 /* Always send a whole number of frames */
275 bytes -= bytes % bpf;
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
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;
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");
295 if(audio_errors == 10)
296 fatal(0, "too many audio errors");
300 written_bytes -= sizeof (struct rtp_header);
301 written_frames = written_bytes / bpf;
302 /* Advance RTP's notion of the time */
303 rtp_time += written_frames * config->sample_format.channels;
304 return written_frames;
309 /** @brief Set up poll array for network play */
310 static void network_beforepoll(void) {
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
319 /* If we're starting then initialize the base time */
321 xgettimeofday(&rtp_time_0, 0);
322 /* We send audio data whenever we get RTP_AHEAD seconds or more
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)
330 if((int64_t)(rtp_time - target_rtp_time) < samples_ahead)
331 bfd_slot = addfd(bfd, POLLOUT);
334 /** @brief Process poll() results for network play */
335 static int network_ready(void) {
336 if(fds[bfd_slot].revents & (POLLOUT | POLLERR))
342 const struct speaker_backend network_backend = {