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>
34 #include "configuration.h"
40 #include "speaker-protocol.h"
43 /** @brief Network socket
45 * This is the file descriptor to write to for @ref BACKEND_NETWORK.
49 /** @brief RTP timestamp
51 * This counts the number of samples played (NB not the number of frames
54 * The timestamp in the packet header is only 32 bits wide. With 44100Hz
55 * stereo, that only gives about half a day before wrapping, which is not
56 * particularly convenient for certain debugging purposes. Therefore the
57 * timestamp is maintained as a 64-bit integer, giving around six million years
58 * before wrapping, and truncated to 32 bits when transmitting.
60 static uint64_t rtp_time
;
62 /** @brief RTP base timestamp
64 * This is the real time correspoding to an @ref rtp_time of 0. It is used
65 * to recalculate the timestamp after idle periods.
67 static struct timeval rtp_time_0
;
69 /** @brief RTP packet sequence number */
70 static uint16_t rtp_seq
;
72 /** @brief RTP SSRC */
73 static uint32_t rtp_id
;
75 /** @brief Error counter */
76 static int audio_errors
;
78 /** @brief Network backend initialization */
79 static void network_init(void) {
80 struct addrinfo
*res
, *sres
;
81 static const struct addrinfo pref
= {
91 static const struct addrinfo prefbind
= {
101 static const int one
= 1;
102 int sndbuf
, target_sndbuf
= 131072;
104 char *sockname
, *ssockname
;
106 /* Override sample format */
107 config
->sample_format
.rate
= 44100;
108 config
->sample_format
.channels
= 2;
109 config
->sample_format
.bits
= 16;
110 config
->sample_format
.byte_format
= AO_FMT_BIG
;
111 res
= get_address(&config
->broadcast
, &pref
, &sockname
);
113 if(config
->broadcast_from
.n
) {
114 sres
= get_address(&config
->broadcast_from
, &prefbind
, &ssockname
);
118 if((bfd
= socket(res
->ai_family
,
120 res
->ai_protocol
)) < 0)
121 fatal(errno
, "error creating broadcast socket");
122 if(setsockopt(bfd
, SOL_SOCKET
, SO_BROADCAST
, &one
, sizeof one
) < 0)
123 fatal(errno
, "error setting SO_BROADCAST on broadcast socket");
125 if(getsockopt(bfd
, SOL_SOCKET
, SO_SNDBUF
,
127 fatal(errno
, "error getting SO_SNDBUF");
128 if(target_sndbuf
> sndbuf
) {
129 if(setsockopt(bfd
, SOL_SOCKET
, SO_SNDBUF
,
130 &target_sndbuf
, sizeof target_sndbuf
) < 0)
131 error(errno
, "error setting SO_SNDBUF to %d", target_sndbuf
);
133 info("changed socket send buffer size from %d to %d",
134 sndbuf
, target_sndbuf
);
136 info("default socket send buffer is %d",
138 /* We might well want to set additional broadcast- or multicast-related
140 if(sres
&& bind(bfd
, sres
->ai_addr
, sres
->ai_addrlen
) < 0)
141 fatal(errno
, "error binding broadcast socket to %s", ssockname
);
142 if(connect(bfd
, res
->ai_addr
, res
->ai_addrlen
) < 0)
143 fatal(errno
, "error connecting broadcast socket to %s", sockname
);
145 gcry_randomize(&rtp_id
, sizeof rtp_id
, GCRY_STRONG_RANDOM
);
146 info("selected network backend, sending to %s", sockname
);
149 /** @brief Play over the network */
150 static size_t network_play(size_t frames
) {
151 struct rtp_header header
;
153 size_t bytes
= frames
* device_bpf
, written_frames
;
155 /* We transmit using RTP (RFC3550) and attempt to conform to the internet
156 * AVT profile (RFC3551). */
159 /* There may have been a gap. Fix up the RTP time accordingly. */
162 uint64_t target_rtp_time
;
164 /* Find the current time */
165 xgettimeofday(&now
, 0);
166 /* Find the number of microseconds elapsed since rtp_time=0 */
167 delta
= tvsub_us(now
, rtp_time_0
);
168 assert(delta
<= UINT64_MAX
/ 88200);
169 target_rtp_time
= (delta
* playing
->format
.rate
170 * playing
->format
.channels
) / 1000000;
171 /* Overflows at ~6 years uptime with 44100Hz stereo */
173 /* rtp_time is the number of samples we've played. NB that we play
174 * RTP_AHEAD_MS ahead of ourselves, so it may legitimately be ahead of
175 * the value we deduce from time comparison.
177 * Suppose we have 1s track started at t=0, and another track begins to
178 * play at t=2s. Suppose RTP_AHEAD_MS=1000 and 44100Hz stereo. In that
179 * case we'll send 1s of audio as fast as we can, giving rtp_time=88200.
180 * rtp_time stops at this point.
182 * At t=2s we'll have calculated target_rtp_time=176400. In this case we
183 * set rtp_time=176400 and the player can correctly conclude that it
184 * should leave 1s between the tracks.
186 * Suppose instead that the second track arrives at t=0.5s, and that
187 * we've managed to transmit the whole of the first track already. We'll
188 * have target_rtp_time=44100.
190 * The desired behaviour is to play the second track back to back with
191 * first. In this case therefore we do not modify rtp_time.
193 * Is it ever right to reduce rtp_time? No; for that would imply
194 * transmitting packets with overlapping timestamp ranges, which does not
197 target_rtp_time
&= ~(uint64_t)1; /* stereo! */
198 if(target_rtp_time
> rtp_time
) {
199 /* More time has elapsed than we've transmitted samples. That implies
200 * we've been 'sending' silence. */
201 info("advancing rtp_time by %"PRIu64
" samples",
202 target_rtp_time
- rtp_time
);
203 rtp_time
= target_rtp_time
;
204 } else if(target_rtp_time
< rtp_time
) {
205 const int64_t samples_ahead
= ((uint64_t)RTP_AHEAD_MS
206 * config
->sample_format
.rate
207 * config
->sample_format
.channels
210 if(target_rtp_time
+ samples_ahead
< rtp_time
) {
211 info("reversing rtp_time by %"PRIu64
" samples",
212 rtp_time
- target_rtp_time
);
216 header
.vpxcc
= 2 << 6; /* V=2, P=0, X=0, CC=0 */
217 header
.seq
= htons(rtp_seq
++);
218 header
.timestamp
= htonl((uint32_t)rtp_time
);
219 header
.ssrc
= rtp_id
;
220 header
.mpt
= (idled ?
0x80 : 0x00) | 10;
221 /* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from
222 * the sample rate (in a library somewhere so that configuration.c can rule
223 * out invalid rates).
226 if(bytes
> NETWORK_BYTES
- sizeof header
) {
227 bytes
= NETWORK_BYTES
- sizeof header
;
228 /* Always send a whole number of frames */
229 bytes
-= bytes
% device_bpf
;
231 /* "The RTP clock rate used for generating the RTP timestamp is independent
232 * of the number of channels and the encoding; it equals the number of
233 * sampling periods per second. For N-channel encodings, each sampling
234 * period (say, 1/8000 of a second) generates N samples. (This terminology
235 * is standard, but somewhat confusing, as the total number of samples
236 * generated per second is then the sampling rate times the channel
239 vec
[0].iov_base
= (void *)&header
;
240 vec
[0].iov_len
= sizeof header
;
241 vec
[1].iov_base
= playing
->buffer
+ playing
->start
;
242 vec
[1].iov_len
= bytes
;
244 written_bytes
= writev(bfd
, vec
, 2);
245 } while(written_bytes
< 0 && errno
== EINTR
);
246 if(written_bytes
< 0) {
247 error(errno
, "error transmitting audio data");
249 if(audio_errors
== 10)
250 fatal(0, "too many audio errors");
254 written_bytes
-= sizeof (struct rtp_header
);
255 written_frames
= written_bytes
/ device_bpf
;
256 /* Advance RTP's notion of the time */
257 rtp_time
+= written_frames
* playing
->format
.channels
;
258 return written_frames
;
263 /** @brief Set up poll array for network play */
264 static void network_beforepoll(void) {
267 uint64_t target_rtp_time
;
268 const int64_t samples_ahead
= ((uint64_t)RTP_AHEAD_MS
269 * config
->sample_format
.rate
270 * config
->sample_format
.channels
273 /* If we're starting then initialize the base time */
275 xgettimeofday(&rtp_time_0
, 0);
276 /* We send audio data whenever we get RTP_AHEAD seconds or more
278 xgettimeofday(&now
, 0);
279 target_us
= tvsub_us(now
, rtp_time_0
);
280 assert(target_us
<= UINT64_MAX
/ 88200);
281 target_rtp_time
= (target_us
* config
->sample_format
.rate
282 * config
->sample_format
.channels
)
284 if((int64_t)(rtp_time
- target_rtp_time
) < samples_ahead
)
285 bfd_slot
= addfd(bfd
, POLLOUT
);
288 /** @brief Process poll() results for network play */
289 static int network_ready(void) {
290 if(fds
[bfd_slot
].revents
& (POLLOUT
| POLLERR
))
296 const struct speaker_backend network_backend
= {