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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 | */ |
47 | static 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 | */ |
60 | static 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 | */ |
67 | static struct timeval rtp_time_0; |
68 | |
69 | /** @brief RTP packet sequence number */ |
70 | static uint16_t rtp_seq; |
71 | |
72 | /** @brief RTP SSRC */ |
73 | static uint32_t rtp_id; |
74 | |
75 | /** @brief Error counter */ |
76 | static int audio_errors; |
77 | |
78 | /** @brief Network backend initialization */ |
79 | static 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 | |
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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; |
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111 | res = get_address(&config->broadcast, &pref, &sockname); |
112 | if(!res) exit(-1); |
113 | if(config->broadcast_from.n) { |
114 | sres = get_address(&config->broadcast_from, &prefbind, &ssockname); |
115 | if(!sres) exit(-1); |
116 | } else |
117 | sres = 0; |
118 | if((bfd = socket(res->ai_family, |
119 | res->ai_socktype, |
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"); |
124 | len = sizeof sndbuf; |
125 | if(getsockopt(bfd, SOL_SOCKET, SO_SNDBUF, |
126 | &sndbuf, &len) < 0) |
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); |
132 | else |
133 | info("changed socket send buffer size from %d to %d", |
134 | sndbuf, target_sndbuf); |
135 | } else |
136 | info("default socket send buffer is %d", |
137 | sndbuf); |
138 | /* We might well want to set additional broadcast- or multicast-related |
139 | * options here */ |
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); |
144 | /* Select an SSRC */ |
145 | gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM); |
146 | info("selected network backend, sending to %s", sockname); |
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147 | } |
148 | |
149 | /** @brief Play over the network */ |
150 | static size_t network_play(size_t frames) { |
151 | struct rtp_header header; |
152 | struct iovec vec[2]; |
153 | size_t bytes = frames * device_bpf, written_frames; |
154 | int written_bytes; |
155 | /* We transmit using RTP (RFC3550) and attempt to conform to the internet |
156 | * AVT profile (RFC3551). */ |
157 | |
158 | if(idled) { |
159 | /* There may have been a gap. Fix up the RTP time accordingly. */ |
160 | struct timeval now; |
161 | uint64_t delta; |
162 | uint64_t target_rtp_time; |
163 | |
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 */ |
172 | |
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. |
176 | * |
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. |
181 | * |
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. |
185 | * |
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. |
189 | * |
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. |
192 | * |
193 | * Is it ever right to reduce rtp_time? No; for that would imply |
194 | * transmitting packets with overlapping timestamp ranges, which does not |
195 | * make sense. |
196 | */ |
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 |
208 | / 1000); |
209 | |
210 | if(target_rtp_time + samples_ahead < rtp_time) { |
211 | info("reversing rtp_time by %"PRIu64" samples", |
212 | rtp_time - target_rtp_time); |
213 | } |
214 | } |
215 | } |
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). |
224 | */ |
225 | idled = 0; |
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; |
230 | } |
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 |
237 | * count.)" |
238 | */ |
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; |
243 | do { |
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"); |
248 | ++audio_errors; |
249 | if(audio_errors == 10) |
250 | fatal(0, "too many audio errors"); |
251 | return 0; |
252 | } else |
253 | audio_errors /= 2; |
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; |
259 | } |
260 | |
261 | static int bfd_slot; |
262 | |
263 | /** @brief Set up poll array for network play */ |
264 | static void network_beforepoll(void) { |
265 | struct timeval now; |
266 | uint64_t target_us; |
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 |
271 | / 1000); |
272 | |
273 | /* If we're starting then initialize the base time */ |
274 | if(!rtp_time) |
275 | xgettimeofday(&rtp_time_0, 0); |
276 | /* We send audio data whenever we get RTP_AHEAD seconds or more |
277 | * behind */ |
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) |
283 | / 1000000; |
284 | if((int64_t)(rtp_time - target_rtp_time) < samples_ahead) |
285 | bfd_slot = addfd(bfd, POLLOUT); |
286 | } |
287 | |
288 | /** @brief Process poll() results for network play */ |
289 | static int network_ready(void) { |
290 | if(fds[bfd_slot].revents & (POLLOUT | POLLERR)) |
291 | return 1; |
292 | else |
293 | return 0; |
294 | } |
295 | |
296 | const struct speaker_backend network_backend = { |
297 | BACKEND_NETWORK, |
298 | FIXED_FORMAT, |
299 | network_init, |
300 | 0, /* activate */ |
301 | network_play, |
302 | 0, /* deactivate */ |
303 | network_beforepoll, |
304 | network_ready |
305 | }; |
306 | |
307 | /* |
308 | Local Variables: |
309 | c-basic-offset:2 |
310 | comment-column:40 |
311 | fill-column:79 |
312 | indent-tabs-mode:nil |
313 | End: |
314 | */ |