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(multicast(res
->ai_addr
)) {
123 switch(res
->ai_family
) {
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");
131 const int mttl
= config
->multicast_ttl
;
132 if(setsockopt(bfd
, IPPROTO_IPV6
, IPV6_MULTICAST_HOPS
,
133 &mttl
, sizeof mttl
) < 0)
134 fatal(errno
, "error setting IPV6_MULTICAST_HOPS on multicast socket");
138 fatal(0, "unsupported address family %d", res
->ai_family
);
140 info("multicasting on %s", sockname
);
144 if(getifaddrs(&ifs
) < 0)
145 fatal(errno
, "error calling getifaddrs");
147 /* (At least on Darwin) IFF_BROADCAST might be set but ifa_broadaddr
148 * still a null pointer. It turns out that there's a subsequent entry
149 * for he same interface which _does_ have ifa_broadaddr though... */
150 if((ifs
->ifa_flags
& IFF_BROADCAST
)
151 && ifs
->ifa_broadaddr
152 && sockaddr_equal(ifs
->ifa_broadaddr
, res
->ai_addr
))
157 if(setsockopt(bfd
, SOL_SOCKET
, SO_BROADCAST
, &one
, sizeof one
) < 0)
158 fatal(errno
, "error setting SO_BROADCAST on broadcast socket");
159 info("broadcasting on %s (%s)", sockname
, ifs
->ifa_name
);
161 info("unicasting on %s", sockname
);
164 if(getsockopt(bfd
, SOL_SOCKET
, SO_SNDBUF
,
166 fatal(errno
, "error getting SO_SNDBUF");
167 if(target_sndbuf
> sndbuf
) {
168 if(setsockopt(bfd
, SOL_SOCKET
, SO_SNDBUF
,
169 &target_sndbuf
, sizeof target_sndbuf
) < 0)
170 error(errno
, "error setting SO_SNDBUF to %d", target_sndbuf
);
172 info("changed socket send buffer size from %d to %d",
173 sndbuf
, target_sndbuf
);
175 info("default socket send buffer is %d",
177 /* We might well want to set additional broadcast- or multicast-related
179 if(sres
&& bind(bfd
, sres
->ai_addr
, sres
->ai_addrlen
) < 0)
180 fatal(errno
, "error binding broadcast socket to %s", ssockname
);
181 if(connect(bfd
, res
->ai_addr
, res
->ai_addrlen
) < 0)
182 fatal(errno
, "error connecting broadcast socket to %s", sockname
);
184 gcry_randomize(&rtp_id
, sizeof rtp_id
, GCRY_STRONG_RANDOM
);
187 /** @brief Play over the network */
188 static size_t network_play(size_t frames
) {
189 struct rtp_header header
;
191 size_t bytes
= frames
* bpf
, written_frames
;
193 /* We transmit using RTP (RFC3550) and attempt to conform to the internet
194 * AVT profile (RFC3551). */
197 /* There may have been a gap. Fix up the RTP time accordingly. */
200 uint64_t target_rtp_time
;
202 /* Find the current time */
203 xgettimeofday(&now
, 0);
204 /* Find the number of microseconds elapsed since rtp_time=0 */
205 delta
= tvsub_us(now
, rtp_time_0
);
206 assert(delta
<= UINT64_MAX
/ 88200);
207 target_rtp_time
= (delta
* config
->sample_format
.rate
208 * config
->sample_format
.channels
) / 1000000;
209 /* Overflows at ~6 years uptime with 44100Hz stereo */
211 /* rtp_time is the number of samples we've played. NB that we play
212 * RTP_AHEAD_MS ahead of ourselves, so it may legitimately be ahead of
213 * the value we deduce from time comparison.
215 * Suppose we have 1s track started at t=0, and another track begins to
216 * play at t=2s. Suppose RTP_AHEAD_MS=1000 and 44100Hz stereo. In that
217 * case we'll send 1s of audio as fast as we can, giving rtp_time=88200.
218 * rtp_time stops at this point.
220 * At t=2s we'll have calculated target_rtp_time=176400. In this case we
221 * set rtp_time=176400 and the player can correctly conclude that it
222 * should leave 1s between the tracks.
224 * Suppose instead that the second track arrives at t=0.5s, and that
225 * we've managed to transmit the whole of the first track already. We'll
226 * have target_rtp_time=44100.
228 * The desired behaviour is to play the second track back to back with
229 * first. In this case therefore we do not modify rtp_time.
231 * Is it ever right to reduce rtp_time? No; for that would imply
232 * transmitting packets with overlapping timestamp ranges, which does not
235 target_rtp_time
&= ~(uint64_t)1; /* stereo! */
236 if(target_rtp_time
> rtp_time
) {
237 /* More time has elapsed than we've transmitted samples. That implies
238 * we've been 'sending' silence. */
239 info("advancing rtp_time by %"PRIu64
" samples",
240 target_rtp_time
- rtp_time
);
241 rtp_time
= target_rtp_time
;
242 } else if(target_rtp_time
< rtp_time
) {
243 const int64_t samples_ahead
= ((uint64_t)RTP_AHEAD_MS
244 * config
->sample_format
.rate
245 * config
->sample_format
.channels
248 if(target_rtp_time
+ samples_ahead
< rtp_time
) {
249 info("reversing rtp_time by %"PRIu64
" samples",
250 rtp_time
- target_rtp_time
);
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).
264 if(bytes
> NETWORK_BYTES
- sizeof header
) {
265 bytes
= NETWORK_BYTES
- sizeof header
;
266 /* Always send a whole number of frames */
267 bytes
-= bytes
% bpf
;
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
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
;
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");
287 if(audio_errors
== 10)
288 fatal(0, "too many audio errors");
292 written_bytes
-= sizeof (struct rtp_header
);
293 written_frames
= written_bytes
/ bpf
;
294 /* Advance RTP's notion of the time */
295 rtp_time
+= written_frames
* config
->sample_format
.channels
;
296 return written_frames
;
301 /** @brief Set up poll array for network play */
302 static void network_beforepoll(int *timeoutp
) {
305 uint64_t target_rtp_time
;
306 const int64_t samples_per_second
= config
->sample_format
.rate
307 * config
->sample_format
.channels
;
308 const int64_t samples_ahead
= ((uint64_t)RTP_AHEAD_MS
311 int64_t lead
, ahead_ms
;
313 /* If we're starting then initialize the base time */
315 xgettimeofday(&rtp_time_0
, 0);
316 /* We send audio data whenever we get RTP_AHEAD seconds or more
318 xgettimeofday(&now
, 0);
319 target_us
= tvsub_us(now
, rtp_time_0
);
320 assert(target_us
<= UINT64_MAX
/ 88200);
321 target_rtp_time
= (target_us
* config
->sample_format
.rate
322 * config
->sample_format
.channels
)
324 lead
= rtp_time
- target_rtp_time
;
325 if(lead
< samples_ahead
)
326 /* We've not reached the desired lead, write as fast as we can */
327 bfd_slot
= addfd(bfd
, POLLOUT
);
329 /* We've reached the desired lead, we can afford to wait a bit even if the
330 * IP stack thinks it can accept more. */
331 ahead_ms
= 1000 * (lead
- samples_ahead
) / samples_per_second
;
332 if(ahead_ms
< *timeoutp
)
333 *timeoutp
= ahead_ms
;
337 /** @brief Process poll() results for network play */
338 static int network_ready(void) {
339 if(fds
[bfd_slot
].revents
& (POLLOUT
| POLLERR
))
345 const struct speaker_backend network_backend
= {