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 res = get_address(&config->broadcast, &pref, &sockname);
108 if(config->broadcast_from.n) {
109 sres = get_address(&config->broadcast_from, &prefbind, &ssockname);
113 if((bfd = socket(res->ai_family,
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");
120 if(getsockopt(bfd, SOL_SOCKET, SO_SNDBUF,
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);
128 info("changed socket send buffer size from %d to %d",
129 sndbuf, target_sndbuf);
131 info("default socket send buffer is %d",
133 /* We might well want to set additional broadcast- or multicast-related
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);
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;
148 /** @brief Play over the network */
149 static size_t network_play(size_t frames) {
150 struct rtp_header header;
152 size_t bytes = frames * device_bpf, written_frames;
154 /* We transmit using RTP (RFC3550) and attempt to conform to the internet
155 * AVT profile (RFC3551). */
158 /* There may have been a gap. Fix up the RTP time accordingly. */
161 uint64_t target_rtp_time;
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 */
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.
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.
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.
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.
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.
192 * Is it ever right to reduce rtp_time? No; for that would imply
193 * transmitting packets with overlapping timestamp ranges, which does not
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
209 if(target_rtp_time + samples_ahead < rtp_time) {
210 info("reversing rtp_time by %"PRIu64" samples",
211 rtp_time - target_rtp_time);
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).
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;
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
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;
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");
248 if(audio_errors == 10)
249 fatal(0, "too many audio errors");
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;
262 /** @brief Set up poll array for network play */
263 static void network_beforepoll(void) {
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
272 /* If we're starting then initialize the base time */
274 xgettimeofday(&rtp_time_0, 0);
275 /* We send audio data whenever we get RTP_AHEAD seconds or more
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)
283 if((int64_t)(rtp_time - target_rtp_time) < samples_ahead)
284 bfd_slot = addfd(bfd, POLLOUT);
287 /** @brief Process poll() results for network play */
288 static int network_ready(void) {
289 if(fds[bfd_slot].revents & (POLLOUT | POLLERR))
295 const struct speaker_backend network_backend = {