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);
194 /** @brief Play over the network */
195 static size_t network_play(size_t frames) {
196 struct rtp_header header;
198 size_t bytes = frames * bpf, written_frames;
200 /* We transmit using RTP (RFC3550) and attempt to conform to the internet
201 * AVT profile (RFC3551). */
204 /* There may have been a gap. Fix up the RTP time accordingly. */
207 uint64_t target_rtp_time;
209 /* Find the current time */
210 xgettimeofday(&now, 0);
211 /* Find the number of microseconds elapsed since rtp_time=0 */
212 delta = tvsub_us(now, rtp_time_0);
213 assert(delta <= UINT64_MAX / 88200);
214 target_rtp_time = (delta * config->sample_format.rate
215 * config->sample_format.channels) / 1000000;
216 /* Overflows at ~6 years uptime with 44100Hz stereo */
218 /* rtp_time is the number of samples we've played. NB that we play
219 * RTP_AHEAD_MS ahead of ourselves, so it may legitimately be ahead of
220 * the value we deduce from time comparison.
222 * Suppose we have 1s track started at t=0, and another track begins to
223 * play at t=2s. Suppose RTP_AHEAD_MS=1000 and 44100Hz stereo. In that
224 * case we'll send 1s of audio as fast as we can, giving rtp_time=88200.
225 * rtp_time stops at this point.
227 * At t=2s we'll have calculated target_rtp_time=176400. In this case we
228 * set rtp_time=176400 and the player can correctly conclude that it
229 * should leave 1s between the tracks.
231 * Suppose instead that the second track arrives at t=0.5s, and that
232 * we've managed to transmit the whole of the first track already. We'll
233 * have target_rtp_time=44100.
235 * The desired behaviour is to play the second track back to back with
236 * first. In this case therefore we do not modify rtp_time.
238 * Is it ever right to reduce rtp_time? No; for that would imply
239 * transmitting packets with overlapping timestamp ranges, which does not
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) {
250 const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS
251 * config->sample_format.rate
252 * config->sample_format.channels
255 if(target_rtp_time + samples_ahead < rtp_time) {
256 info("reversing rtp_time by %"PRIu64" samples",
257 rtp_time - target_rtp_time);
261 header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */
262 header.seq = htons(rtp_seq++);
263 header.timestamp = htonl((uint32_t)rtp_time);
264 header.ssrc = rtp_id;
265 header.mpt = (idled ? 0x80 : 0x00) | 10;
266 /* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from
267 * the sample rate (in a library somewhere so that configuration.c can rule
268 * out invalid rates).
271 if(bytes > NETWORK_BYTES - sizeof header) {
272 bytes = NETWORK_BYTES - sizeof header;
273 /* Always send a whole number of frames */
274 bytes -= bytes % bpf;
276 /* "The RTP clock rate used for generating the RTP timestamp is independent
277 * of the number of channels and the encoding; it equals the number of
278 * sampling periods per second. For N-channel encodings, each sampling
279 * period (say, 1/8000 of a second) generates N samples. (This terminology
280 * is standard, but somewhat confusing, as the total number of samples
281 * generated per second is then the sampling rate times the channel
284 vec[0].iov_base = (void *)&header;
285 vec[0].iov_len = sizeof header;
286 vec[1].iov_base = playing->buffer + playing->start;
287 vec[1].iov_len = bytes;
289 written_bytes = writev(bfd, vec, 2);
290 } while(written_bytes < 0 && errno == EINTR);
291 if(written_bytes < 0) {
292 error(errno, "error transmitting audio data");
294 if(audio_errors == 10)
295 fatal(0, "too many audio errors");
299 written_bytes -= sizeof (struct rtp_header);
300 written_frames = written_bytes / bpf;
301 /* Advance RTP's notion of the time */
302 rtp_time += written_frames * config->sample_format.channels;
303 return written_frames;
308 /** @brief Set up poll array for network play */
309 static void network_beforepoll(int *timeoutp) {
312 uint64_t target_rtp_time;
313 const int64_t samples_per_second = config->sample_format.rate
314 * config->sample_format.channels;
315 const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS
318 int64_t lead, ahead_ms;
320 /* If we're starting then initialize the base time */
322 xgettimeofday(&rtp_time_0, 0);
323 /* We send audio data whenever we get RTP_AHEAD seconds or more
325 xgettimeofday(&now, 0);
326 target_us = tvsub_us(now, rtp_time_0);
327 assert(target_us <= UINT64_MAX / 88200);
328 target_rtp_time = (target_us * config->sample_format.rate
329 * config->sample_format.channels)
331 lead = rtp_time - target_rtp_time;
332 if(lead < samples_ahead)
333 /* We've not reached the desired lead, write as fast as we can */
334 bfd_slot = addfd(bfd, POLLOUT);
336 /* We've reached the desired lead, we can afford to wait a bit even if the
337 * IP stack thinks it can accept more. */
338 ahead_ms = 1000 * (lead - samples_ahead) / samples_per_second;
339 if(ahead_ms < *timeoutp)
340 *timeoutp = ahead_ms;
344 /** @brief Process poll() results for network play */
345 static int network_ready(void) {
346 if(fds[bfd_slot].revents & (POLLOUT | POLLERR))
352 const struct speaker_backend network_backend = {