--- /dev/null
+/*
+ * This file is part of DisOrder
+ * Copyright (C) 2005, 2006, 2007 Richard Kettlewell
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307
+ * USA
+ */
+/** @file server/speaker-network.c
+ * @brief Support for @ref BACKEND_NETWORK */
+
+#include <config.h>
+#include "types.h"
+
+#include <unistd.h>
+#include <poll.h>
+#include <netdb.h>
+#include <gcrypt.h>
+#include <sys/socket.h>
+#include <sys/uio.h>
+#include <assert.h>
+
+#include "configuration.h"
+#include "syscalls.h"
+#include "log.h"
+#include "addr.h"
+#include "timeval.h"
+#include "rtp.h"
+#include "speaker-protocol.h"
+#include "speaker.h"
+
+/** @brief Network socket
+ *
+ * This is the file descriptor to write to for @ref BACKEND_NETWORK.
+ */
+static int bfd = -1;
+
+/** @brief RTP timestamp
+ *
+ * This counts the number of samples played (NB not the number of frames
+ * played).
+ *
+ * The timestamp in the packet header is only 32 bits wide. With 44100Hz
+ * stereo, that only gives about half a day before wrapping, which is not
+ * particularly convenient for certain debugging purposes. Therefore the
+ * timestamp is maintained as a 64-bit integer, giving around six million years
+ * before wrapping, and truncated to 32 bits when transmitting.
+ */
+static uint64_t rtp_time;
+
+/** @brief RTP base timestamp
+ *
+ * This is the real time correspoding to an @ref rtp_time of 0. It is used
+ * to recalculate the timestamp after idle periods.
+ */
+static struct timeval rtp_time_0;
+
+/** @brief RTP packet sequence number */
+static uint16_t rtp_seq;
+
+/** @brief RTP SSRC */
+static uint32_t rtp_id;
+
+/** @brief Error counter */
+static int audio_errors;
+
+/** @brief Network backend initialization */
+static void network_init(void) {
+ struct addrinfo *res, *sres;
+ static const struct addrinfo pref = {
+ 0,
+ PF_INET,
+ SOCK_DGRAM,
+ IPPROTO_UDP,
+ 0,
+ 0,
+ 0,
+ 0
+ };
+ static const struct addrinfo prefbind = {
+ AI_PASSIVE,
+ PF_INET,
+ SOCK_DGRAM,
+ IPPROTO_UDP,
+ 0,
+ 0,
+ 0,
+ 0
+ };
+ static const int one = 1;
+ int sndbuf, target_sndbuf = 131072;
+ socklen_t len;
+ char *sockname, *ssockname;
+
+ res = get_address(&config->broadcast, &pref, &sockname);
+ if(!res) exit(-1);
+ if(config->broadcast_from.n) {
+ sres = get_address(&config->broadcast_from, &prefbind, &ssockname);
+ if(!sres) exit(-1);
+ } else
+ sres = 0;
+ if((bfd = socket(res->ai_family,
+ res->ai_socktype,
+ res->ai_protocol)) < 0)
+ fatal(errno, "error creating broadcast socket");
+ if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0)
+ fatal(errno, "error setting SO_BROADCAST on broadcast socket");
+ len = sizeof sndbuf;
+ if(getsockopt(bfd, SOL_SOCKET, SO_SNDBUF,
+ &sndbuf, &len) < 0)
+ fatal(errno, "error getting SO_SNDBUF");
+ if(target_sndbuf > sndbuf) {
+ if(setsockopt(bfd, SOL_SOCKET, SO_SNDBUF,
+ &target_sndbuf, sizeof target_sndbuf) < 0)
+ error(errno, "error setting SO_SNDBUF to %d", target_sndbuf);
+ else
+ info("changed socket send buffer size from %d to %d",
+ sndbuf, target_sndbuf);
+ } else
+ info("default socket send buffer is %d",
+ sndbuf);
+ /* We might well want to set additional broadcast- or multicast-related
+ * options here */
+ if(sres && bind(bfd, sres->ai_addr, sres->ai_addrlen) < 0)
+ fatal(errno, "error binding broadcast socket to %s", ssockname);
+ if(connect(bfd, res->ai_addr, res->ai_addrlen) < 0)
+ fatal(errno, "error connecting broadcast socket to %s", sockname);
+ /* Select an SSRC */
+ gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM);
+ info("selected network backend, sending to %s", sockname);
+ if(config->sample_format.byte_format != AO_FMT_BIG) {
+ info("forcing big-endian sample format");
+ config->sample_format.byte_format = AO_FMT_BIG;
+ }
+}
+
+/** @brief Play over the network */
+static size_t network_play(size_t frames) {
+ struct rtp_header header;
+ struct iovec vec[2];
+ size_t bytes = frames * device_bpf, written_frames;
+ int written_bytes;
+ /* We transmit using RTP (RFC3550) and attempt to conform to the internet
+ * AVT profile (RFC3551). */
+
+ if(idled) {
+ /* There may have been a gap. Fix up the RTP time accordingly. */
+ struct timeval now;
+ uint64_t delta;
+ uint64_t target_rtp_time;
+
+ /* Find the current time */
+ xgettimeofday(&now, 0);
+ /* Find the number of microseconds elapsed since rtp_time=0 */
+ delta = tvsub_us(now, rtp_time_0);
+ assert(delta <= UINT64_MAX / 88200);
+ target_rtp_time = (delta * playing->format.rate
+ * playing->format.channels) / 1000000;
+ /* Overflows at ~6 years uptime with 44100Hz stereo */
+
+ /* rtp_time is the number of samples we've played. NB that we play
+ * RTP_AHEAD_MS ahead of ourselves, so it may legitimately be ahead of
+ * the value we deduce from time comparison.
+ *
+ * Suppose we have 1s track started at t=0, and another track begins to
+ * play at t=2s. Suppose RTP_AHEAD_MS=1000 and 44100Hz stereo. In that
+ * case we'll send 1s of audio as fast as we can, giving rtp_time=88200.
+ * rtp_time stops at this point.
+ *
+ * At t=2s we'll have calculated target_rtp_time=176400. In this case we
+ * set rtp_time=176400 and the player can correctly conclude that it
+ * should leave 1s between the tracks.
+ *
+ * Suppose instead that the second track arrives at t=0.5s, and that
+ * we've managed to transmit the whole of the first track already. We'll
+ * have target_rtp_time=44100.
+ *
+ * The desired behaviour is to play the second track back to back with
+ * first. In this case therefore we do not modify rtp_time.
+ *
+ * Is it ever right to reduce rtp_time? No; for that would imply
+ * transmitting packets with overlapping timestamp ranges, which does not
+ * make sense.
+ */
+ target_rtp_time &= ~(uint64_t)1; /* stereo! */
+ if(target_rtp_time > rtp_time) {
+ /* More time has elapsed than we've transmitted samples. That implies
+ * we've been 'sending' silence. */
+ info("advancing rtp_time by %"PRIu64" samples",
+ target_rtp_time - rtp_time);
+ rtp_time = target_rtp_time;
+ } else if(target_rtp_time < rtp_time) {
+ const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS
+ * config->sample_format.rate
+ * config->sample_format.channels
+ / 1000);
+
+ if(target_rtp_time + samples_ahead < rtp_time) {
+ info("reversing rtp_time by %"PRIu64" samples",
+ rtp_time - target_rtp_time);
+ }
+ }
+ }
+ header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */
+ header.seq = htons(rtp_seq++);
+ header.timestamp = htonl((uint32_t)rtp_time);
+ header.ssrc = rtp_id;
+ header.mpt = (idled ? 0x80 : 0x00) | 10;
+ /* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from
+ * the sample rate (in a library somewhere so that configuration.c can rule
+ * out invalid rates).
+ */
+ idled = 0;
+ if(bytes > NETWORK_BYTES - sizeof header) {
+ bytes = NETWORK_BYTES - sizeof header;
+ /* Always send a whole number of frames */
+ bytes -= bytes % device_bpf;
+ }
+ /* "The RTP clock rate used for generating the RTP timestamp is independent
+ * of the number of channels and the encoding; it equals the number of
+ * sampling periods per second. For N-channel encodings, each sampling
+ * period (say, 1/8000 of a second) generates N samples. (This terminology
+ * is standard, but somewhat confusing, as the total number of samples
+ * generated per second is then the sampling rate times the channel
+ * count.)"
+ */
+ vec[0].iov_base = (void *)&header;
+ vec[0].iov_len = sizeof header;
+ vec[1].iov_base = playing->buffer + playing->start;
+ vec[1].iov_len = bytes;
+ do {
+ written_bytes = writev(bfd, vec, 2);
+ } while(written_bytes < 0 && errno == EINTR);
+ if(written_bytes < 0) {
+ error(errno, "error transmitting audio data");
+ ++audio_errors;
+ if(audio_errors == 10)
+ fatal(0, "too many audio errors");
+ return 0;
+ } else
+ audio_errors /= 2;
+ written_bytes -= sizeof (struct rtp_header);
+ written_frames = written_bytes / device_bpf;
+ /* Advance RTP's notion of the time */
+ rtp_time += written_frames * playing->format.channels;
+ return written_frames;
+}
+
+static int bfd_slot;
+
+/** @brief Set up poll array for network play */
+static void network_beforepoll(void) {
+ struct timeval now;
+ uint64_t target_us;
+ uint64_t target_rtp_time;
+ const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS
+ * config->sample_format.rate
+ * config->sample_format.channels
+ / 1000);
+
+ /* If we're starting then initialize the base time */
+ if(!rtp_time)
+ xgettimeofday(&rtp_time_0, 0);
+ /* We send audio data whenever we get RTP_AHEAD seconds or more
+ * behind */
+ xgettimeofday(&now, 0);
+ target_us = tvsub_us(now, rtp_time_0);
+ assert(target_us <= UINT64_MAX / 88200);
+ target_rtp_time = (target_us * config->sample_format.rate
+ * config->sample_format.channels)
+ / 1000000;
+ if((int64_t)(rtp_time - target_rtp_time) < samples_ahead)
+ bfd_slot = addfd(bfd, POLLOUT);
+}
+
+/** @brief Process poll() results for network play */
+static int network_ready(void) {
+ if(fds[bfd_slot].revents & (POLLOUT | POLLERR))
+ return 1;
+ else
+ return 0;
+}
+
+const struct speaker_backend network_backend = {
+ BACKEND_NETWORK,
+ FIXED_FORMAT,
+ network_init,
+ 0, /* activate */
+ network_play,
+ 0, /* deactivate */
+ network_beforepoll,
+ network_ready
+};
+
+/*
+Local Variables:
+c-basic-offset:2
+comment-column:40
+fill-column:79
+indent-tabs-mode:nil
+End:
+*/
~mdw / disorder / blobdiff