chiark / gitweb /
split backends out into their own speaker-*.c
[disorder] / server / speaker-network.c
CommitLineData
1c3f1e73 1/*
2 * This file is part of DisOrder
3 * Copyright (C) 2005, 2006, 2007 Richard Kettlewell
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>
33
34#include "configuration.h"
35#include "syscalls.h"
36#include "log.h"
37#include "addr.h"
38#include "timeval.h"
39#include "rtp.h"
40#include "speaker-protocol.h"
41#include "speaker.h"
42
43/** @brief Network socket
44 *
45 * This is the file descriptor to write to for @ref BACKEND_NETWORK.
46 */
47static int bfd = -1;
48
49/** @brief RTP timestamp
50 *
51 * This counts the number of samples played (NB not the number of frames
52 * played).
53 *
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.
59 */
60static uint64_t rtp_time;
61
62/** @brief RTP base timestamp
63 *
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.
66 */
67static struct timeval rtp_time_0;
68
69/** @brief RTP packet sequence number */
70static uint16_t rtp_seq;
71
72/** @brief RTP SSRC */
73static uint32_t rtp_id;
74
75/** @brief Error counter */
76static int audio_errors;
77
78/** @brief Network backend initialization */
79static void network_init(void) {
80 struct addrinfo *res, *sres;
81 static const struct addrinfo pref = {
82 0,
83 PF_INET,
84 SOCK_DGRAM,
85 IPPROTO_UDP,
86 0,
87 0,
88 0,
89 0
90 };
91 static const struct addrinfo prefbind = {
92 AI_PASSIVE,
93 PF_INET,
94 SOCK_DGRAM,
95 IPPROTO_UDP,
96 0,
97 0,
98 0,
99 0
100 };
101 static const int one = 1;
102 int sndbuf, target_sndbuf = 131072;
103 socklen_t len;
104 char *sockname, *ssockname;
105
106 res = get_address(&config->broadcast, &pref, &sockname);
107 if(!res) exit(-1);
108 if(config->broadcast_from.n) {
109 sres = get_address(&config->broadcast_from, &prefbind, &ssockname);
110 if(!sres) exit(-1);
111 } else
112 sres = 0;
113 if((bfd = socket(res->ai_family,
114 res->ai_socktype,
115 res->ai_protocol)) < 0)
116 fatal(errno, "error creating broadcast socket");
117 if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0)
118 fatal(errno, "error setting SO_BROADCAST on broadcast socket");
119 len = sizeof sndbuf;
120 if(getsockopt(bfd, SOL_SOCKET, SO_SNDBUF,
121 &sndbuf, &len) < 0)
122 fatal(errno, "error getting SO_SNDBUF");
123 if(target_sndbuf > sndbuf) {
124 if(setsockopt(bfd, SOL_SOCKET, SO_SNDBUF,
125 &target_sndbuf, sizeof target_sndbuf) < 0)
126 error(errno, "error setting SO_SNDBUF to %d", target_sndbuf);
127 else
128 info("changed socket send buffer size from %d to %d",
129 sndbuf, target_sndbuf);
130 } else
131 info("default socket send buffer is %d",
132 sndbuf);
133 /* We might well want to set additional broadcast- or multicast-related
134 * options here */
135 if(sres && bind(bfd, sres->ai_addr, sres->ai_addrlen) < 0)
136 fatal(errno, "error binding broadcast socket to %s", ssockname);
137 if(connect(bfd, res->ai_addr, res->ai_addrlen) < 0)
138 fatal(errno, "error connecting broadcast socket to %s", sockname);
139 /* Select an SSRC */
140 gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM);
141 info("selected network backend, sending to %s", sockname);
142 if(config->sample_format.byte_format != AO_FMT_BIG) {
143 info("forcing big-endian sample format");
144 config->sample_format.byte_format = AO_FMT_BIG;
145 }
146}
147
148/** @brief Play over the network */
149static size_t network_play(size_t frames) {
150 struct rtp_header header;
151 struct iovec vec[2];
152 size_t bytes = frames * device_bpf, written_frames;
153 int written_bytes;
154 /* We transmit using RTP (RFC3550) and attempt to conform to the internet
155 * AVT profile (RFC3551). */
156
157 if(idled) {
158 /* There may have been a gap. Fix up the RTP time accordingly. */
159 struct timeval now;
160 uint64_t delta;
161 uint64_t target_rtp_time;
162
163 /* Find the current time */
164 xgettimeofday(&now, 0);
165 /* Find the number of microseconds elapsed since rtp_time=0 */
166 delta = tvsub_us(now, rtp_time_0);
167 assert(delta <= UINT64_MAX / 88200);
168 target_rtp_time = (delta * playing->format.rate
169 * playing->format.channels) / 1000000;
170 /* Overflows at ~6 years uptime with 44100Hz stereo */
171
172 /* rtp_time is the number of samples we've played. NB that we play
173 * RTP_AHEAD_MS ahead of ourselves, so it may legitimately be ahead of
174 * the value we deduce from time comparison.
175 *
176 * Suppose we have 1s track started at t=0, and another track begins to
177 * play at t=2s. Suppose RTP_AHEAD_MS=1000 and 44100Hz stereo. In that
178 * case we'll send 1s of audio as fast as we can, giving rtp_time=88200.
179 * rtp_time stops at this point.
180 *
181 * At t=2s we'll have calculated target_rtp_time=176400. In this case we
182 * set rtp_time=176400 and the player can correctly conclude that it
183 * should leave 1s between the tracks.
184 *
185 * Suppose instead that the second track arrives at t=0.5s, and that
186 * we've managed to transmit the whole of the first track already. We'll
187 * have target_rtp_time=44100.
188 *
189 * The desired behaviour is to play the second track back to back with
190 * first. In this case therefore we do not modify rtp_time.
191 *
192 * Is it ever right to reduce rtp_time? No; for that would imply
193 * transmitting packets with overlapping timestamp ranges, which does not
194 * make sense.
195 */
196 target_rtp_time &= ~(uint64_t)1; /* stereo! */
197 if(target_rtp_time > rtp_time) {
198 /* More time has elapsed than we've transmitted samples. That implies
199 * we've been 'sending' silence. */
200 info("advancing rtp_time by %"PRIu64" samples",
201 target_rtp_time - rtp_time);
202 rtp_time = target_rtp_time;
203 } else if(target_rtp_time < rtp_time) {
204 const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS
205 * config->sample_format.rate
206 * config->sample_format.channels
207 / 1000);
208
209 if(target_rtp_time + samples_ahead < rtp_time) {
210 info("reversing rtp_time by %"PRIu64" samples",
211 rtp_time - target_rtp_time);
212 }
213 }
214 }
215 header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */
216 header.seq = htons(rtp_seq++);
217 header.timestamp = htonl((uint32_t)rtp_time);
218 header.ssrc = rtp_id;
219 header.mpt = (idled ? 0x80 : 0x00) | 10;
220 /* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from
221 * the sample rate (in a library somewhere so that configuration.c can rule
222 * out invalid rates).
223 */
224 idled = 0;
225 if(bytes > NETWORK_BYTES - sizeof header) {
226 bytes = NETWORK_BYTES - sizeof header;
227 /* Always send a whole number of frames */
228 bytes -= bytes % device_bpf;
229 }
230 /* "The RTP clock rate used for generating the RTP timestamp is independent
231 * of the number of channels and the encoding; it equals the number of
232 * sampling periods per second. For N-channel encodings, each sampling
233 * period (say, 1/8000 of a second) generates N samples. (This terminology
234 * is standard, but somewhat confusing, as the total number of samples
235 * generated per second is then the sampling rate times the channel
236 * count.)"
237 */
238 vec[0].iov_base = (void *)&header;
239 vec[0].iov_len = sizeof header;
240 vec[1].iov_base = playing->buffer + playing->start;
241 vec[1].iov_len = bytes;
242 do {
243 written_bytes = writev(bfd, vec, 2);
244 } while(written_bytes < 0 && errno == EINTR);
245 if(written_bytes < 0) {
246 error(errno, "error transmitting audio data");
247 ++audio_errors;
248 if(audio_errors == 10)
249 fatal(0, "too many audio errors");
250 return 0;
251 } else
252 audio_errors /= 2;
253 written_bytes -= sizeof (struct rtp_header);
254 written_frames = written_bytes / device_bpf;
255 /* Advance RTP's notion of the time */
256 rtp_time += written_frames * playing->format.channels;
257 return written_frames;
258}
259
260static int bfd_slot;
261
262/** @brief Set up poll array for network play */
263static void network_beforepoll(void) {
264 struct timeval now;
265 uint64_t target_us;
266 uint64_t target_rtp_time;
267 const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS
268 * config->sample_format.rate
269 * config->sample_format.channels
270 / 1000);
271
272 /* If we're starting then initialize the base time */
273 if(!rtp_time)
274 xgettimeofday(&rtp_time_0, 0);
275 /* We send audio data whenever we get RTP_AHEAD seconds or more
276 * behind */
277 xgettimeofday(&now, 0);
278 target_us = tvsub_us(now, rtp_time_0);
279 assert(target_us <= UINT64_MAX / 88200);
280 target_rtp_time = (target_us * config->sample_format.rate
281 * config->sample_format.channels)
282 / 1000000;
283 if((int64_t)(rtp_time - target_rtp_time) < samples_ahead)
284 bfd_slot = addfd(bfd, POLLOUT);
285}
286
287/** @brief Process poll() results for network play */
288static int network_ready(void) {
289 if(fds[bfd_slot].revents & (POLLOUT | POLLERR))
290 return 1;
291 else
292 return 0;
293}
294
295const struct speaker_backend network_backend = {
296 BACKEND_NETWORK,
297 FIXED_FORMAT,
298 network_init,
299 0, /* activate */
300 network_play,
301 0, /* deactivate */
302 network_beforepoll,
303 network_ready
304};
305
306/*
307Local Variables:
308c-basic-offset:2
309comment-column:40
310fill-column:79
311indent-tabs-mode:nil
312End:
313*/