[disorder] / server / speaker-network.c

/*
 * 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 <net/if.h>
#include <ifaddrs.h>
#include <errno.h>

#include "configuration.h"
#include "syscalls.h"
#include "log.h"
#include "addr.h"
#include "timeval.h"
#include "rtp.h"
#include "ifreq.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(multicast(res->ai_addr)) {
    /* Multicasting */
    switch(res->ai_family) {
    case PF_INET: {
      const int mttl = config->multicast_ttl;
      if(setsockopt(bfd, IPPROTO_IP, IP_MULTICAST_TTL, &mttl, sizeof mttl) < 0)
        fatal(errno, "error setting IP_MULTICAST_TTL on multicast socket");
      if(setsockopt(bfd, IPPROTO_IP, IP_MULTICAST_LOOP,
                    &config->multicast_loop, sizeof one) < 0)
        fatal(errno, "error setting IP_MULTICAST_LOOP on multicast socket");
      break;
    }
    case PF_INET6: {
      const int mttl = config->multicast_ttl;
      if(setsockopt(bfd, IPPROTO_IPV6, IPV6_MULTICAST_HOPS,
                    &mttl, sizeof mttl) < 0)
        fatal(errno, "error setting IPV6_MULTICAST_HOPS on multicast socket");
      if(setsockopt(bfd, IPPROTO_IP, IPV6_MULTICAST_LOOP,
                    &config->multicast_loop, sizeof (int)) < 0)
        fatal(errno, "error setting IPV6_MULTICAST_LOOP on multicast socket");
      break;
    }
    default:
      fatal(0, "unsupported address family %d", res->ai_family);
    }
    info("multicasting on %s", sockname);
  } else {
    struct ifaddrs *ifs;

    if(getifaddrs(&ifs) < 0)
      fatal(errno, "error calling getifaddrs");
    while(ifs) {
      /* (At least on Darwin) IFF_BROADCAST might be set but ifa_broadaddr
       * still a null pointer.  It turns out that there's a subsequent entry
       * for he same interface which _does_ have ifa_broadaddr though... */
      if((ifs->ifa_flags & IFF_BROADCAST)
         && ifs->ifa_broadaddr
         && sockaddr_equal(ifs->ifa_broadaddr, res->ai_addr))
        break;
      ifs = ifs->ifa_next;
    }
    if(ifs) {
      if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0)
        fatal(errno, "error setting SO_BROADCAST on broadcast socket");
      info("broadcasting on %s (%s)", sockname, ifs->ifa_name);
    } else
      info("unicasting on %s", sockname);
  }
  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);
}

/** @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 * 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 * config->sample_format.rate
                       * config->sample_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 % 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 / bpf;
  /* Advance RTP's notion of the time */
  rtp_time += written_frames * config->sample_format.channels;
  return written_frames;
}

static int bfd_slot;

/** @brief Set up poll array for network play */
static void network_beforepoll(int *timeoutp) {
  struct timeval now;
  uint64_t target_us;
  uint64_t target_rtp_time;
  const int64_t samples_per_second = config->sample_format.rate
                                   * config->sample_format.channels;
  const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS
                                 * samples_per_second
                                 / 1000);
  int64_t lead, ahead_ms;
  
  /* 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;
  lead = rtp_time - target_rtp_time;
  if(lead < samples_ahead)
    /* We've not reached the desired lead, write as fast as we can */
    bfd_slot = addfd(bfd, POLLOUT);
  else {
    /* We've reached the desired lead, we can afford to wait a bit even if the
     * IP stack thinks it can accept more. */
    ahead_ms = 1000 * (lead - samples_ahead) / samples_per_second;
    if(ahead_ms < *timeoutp)
      *timeoutp = ahead_ms;
  }
}

/** @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,
  0,
  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:
*/
Commit	Line	Data
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>
81b1bf12	33	#include <net/if.h>
db2c19dc	34	#include <ifaddrs.h>
6d2d327c	35	#include <errno.h>
1c3f1e73	36
	37	#include "configuration.h"
	38	#include "syscalls.h"
	39	#include "log.h"
	40	#include "addr.h"
	41	#include "timeval.h"
	42	#include "rtp.h"
81b1bf12	43	#include "ifreq.h"
1c3f1e73	44	#include "speaker-protocol.h"
	45	#include "speaker.h"
	46
	47	/** @brief Network socket
	48	*
	49	* This is the file descriptor to write to for @ref BACKEND_NETWORK.
	50	*/
	51	static int bfd = -1;
	52
	53	/** @brief RTP timestamp
	54	*
	55	* This counts the number of samples played (NB not the number of frames
	56	* played).
	57	*
	58	* The timestamp in the packet header is only 32 bits wide. With 44100Hz
	59	* stereo, that only gives about half a day before wrapping, which is not
	60	* particularly convenient for certain debugging purposes. Therefore the
	61	* timestamp is maintained as a 64-bit integer, giving around six million years
	62	* before wrapping, and truncated to 32 bits when transmitting.
	63	*/
	64	static uint64_t rtp_time;
	65
	66	/** @brief RTP base timestamp
	67	*
	68	* This is the real time correspoding to an @ref rtp_time of 0. It is used
	69	* to recalculate the timestamp after idle periods.
	70	*/
	71	static struct timeval rtp_time_0;
	72
	73	/** @brief RTP packet sequence number */
	74	static uint16_t rtp_seq;
	75
	76	/** @brief RTP SSRC */
	77	static uint32_t rtp_id;
	78
	79	/** @brief Error counter */
	80	static int audio_errors;
	81
	82	/** @brief Network backend initialization */
	83	static void network_init(void) {
	84	struct addrinfo res, sres;
	85	static const struct addrinfo pref = {
	86	0,
	87	PF_INET,
	88	SOCK_DGRAM,
	89	IPPROTO_UDP,
	90	0,
	91	0,
	92	0,
	93	0
	94	};
	95	static const struct addrinfo prefbind = {
	96	AI_PASSIVE,
	97	PF_INET,
	98	SOCK_DGRAM,
	99	IPPROTO_UDP,
	100	0,
	101	0,
	102	0,
	103	0
	104	};
	105	static const int one = 1;
db2c19dc	106	int sndbuf, target_sndbuf = 131072;
1c3f1e73	107	socklen_t len;
	108	char sockname, ssockname;
	109
	110	res = get_address(&config->broadcast, &pref, &sockname);
	111	if(!res) exit(-1);
	112	if(config->broadcast_from.n) {
	113	sres = get_address(&config->broadcast_from, &prefbind, &ssockname);
	114	if(!sres) exit(-1);
	115	} else
	116	sres = 0;
	117	if((bfd = socket(res->ai_family,
	118	res->ai_socktype,
	119	res->ai_protocol)) < 0)
	120	fatal(errno, "error creating broadcast socket");
6fba990c	121	if(multicast(res->ai_addr)) {
23205f9c RK	122	/* Multicasting */
	123	switch(res->ai_family) {
	124	case PF_INET: {
	125	const int mttl = config->multicast_ttl;
	126	if(setsockopt(bfd, IPPROTO_IP, IP_MULTICAST_TTL, &mttl, sizeof mttl) < 0)
	127	fatal(errno, "error setting IP_MULTICAST_TTL on multicast socket");
61941295 RK	128	if(setsockopt(bfd, IPPROTO_IP, IP_MULTICAST_LOOP,
	129	&config->multicast_loop, sizeof one) < 0)
	130	fatal(errno, "error setting IP_MULTICAST_LOOP on multicast socket");
23205f9c RK	131	break;
	132	}
	133	case PF_INET6: {
	134	const int mttl = config->multicast_ttl;
	135	if(setsockopt(bfd, IPPROTO_IPV6, IPV6_MULTICAST_HOPS,
	136	&mttl, sizeof mttl) < 0)
	137	fatal(errno, "error setting IPV6_MULTICAST_HOPS on multicast socket");
61941295 RK	138	if(setsockopt(bfd, IPPROTO_IP, IPV6_MULTICAST_LOOP,
	139	&config->multicast_loop, sizeof (int)) < 0)
	140	fatal(errno, "error setting IPV6_MULTICAST_LOOP on multicast socket");
23205f9c RK	141	break;
	142	}
	143	default:
	144	fatal(0, "unsupported address family %d", res->ai_family);
	145	}
81b1bf12	146	info("multicasting on %s", sockname);
23205f9c	147	} else {
db2c19dc	148	struct ifaddrs *ifs;
81b1bf12	149
db2c19dc RK	150	if(getifaddrs(&ifs) < 0)
	151	fatal(errno, "error calling getifaddrs");
	152	while(ifs) {
3aa6f359 RK	153	/* (At least on Darwin) IFF_BROADCAST might be set but ifa_broadaddr
	154	* still a null pointer. It turns out that there's a subsequent entry
	155	* for he same interface which _does_ have ifa_broadaddr though... */
db2c19dc	156	if((ifs->ifa_flags & IFF_BROADCAST)
3aa6f359	157	&& ifs->ifa_broadaddr
db2c19dc	158	&& sockaddr_equal(ifs->ifa_broadaddr, res->ai_addr))
81b1bf12	159	break;
db2c19dc	160	ifs = ifs->ifa_next;
81b1bf12	161	}
db2c19dc	162	if(ifs) {
81b1bf12 RK	163	if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0)
81b1bf12 RK	164	fatal(errno, "error setting SO_BROADCAST on broadcast socket");
db2c19dc	165	info("broadcasting on %s (%s)", sockname, ifs->ifa_name);
81b1bf12 RK	166	} else
81b1bf12 RK	167	info("unicasting on %s", sockname);
23205f9c	168	}
1c3f1e73	169	len = sizeof sndbuf;
	170	if(getsockopt(bfd, SOL_SOCKET, SO_SNDBUF,
	171	&sndbuf, &len) < 0)
	172	fatal(errno, "error getting SO_SNDBUF");
	173	if(target_sndbuf > sndbuf) {
	174	if(setsockopt(bfd, SOL_SOCKET, SO_SNDBUF,
	175	&target_sndbuf, sizeof target_sndbuf) < 0)
	176	error(errno, "error setting SO_SNDBUF to %d", target_sndbuf);
	177	else
	178	info("changed socket send buffer size from %d to %d",
	179	sndbuf, target_sndbuf);
	180	} else
	181	info("default socket send buffer is %d",
	182	sndbuf);
	183	/* We might well want to set additional broadcast- or multicast-related
	184	* options here */
	185	if(sres && bind(bfd, sres->ai_addr, sres->ai_addrlen) < 0)
	186	fatal(errno, "error binding broadcast socket to %s", ssockname);
	187	if(connect(bfd, res->ai_addr, res->ai_addrlen) < 0)
	188	fatal(errno, "error connecting broadcast socket to %s", sockname);
	189	/* Select an SSRC */
	190	gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM);
1c3f1e73	191	}
	192
	193	/** @brief Play over the network */
	194	static size_t network_play(size_t frames) {
	195	struct rtp_header header;
	196	struct iovec vec[2];
6d2d327c	197	size_t bytes = frames * bpf, written_frames;
1c3f1e73	198	int written_bytes;
	199	/* We transmit using RTP (RFC3550) and attempt to conform to the internet
	200	* AVT profile (RFC3551). */
	201
	202	if(idled) {
	203	/* There may have been a gap. Fix up the RTP time accordingly. */
	204	struct timeval now;
	205	uint64_t delta;
	206	uint64_t target_rtp_time;
	207
	208	/* Find the current time */
	209	xgettimeofday(&now, 0);
	210	/* Find the number of microseconds elapsed since rtp_time=0 */
	211	delta = tvsub_us(now, rtp_time_0);
	212	assert(delta <= UINT64_MAX / 88200);
6d2d327c RK	213	target_rtp_time = (delta * config->sample_format.rate
6d2d327c RK	214	* config->sample_format.channels) / 1000000;
1c3f1e73	215	/* Overflows at ~6 years uptime with 44100Hz stereo */
	216
	217	/* rtp_time is the number of samples we've played. NB that we play
	218	* RTP_AHEAD_MS ahead of ourselves, so it may legitimately be ahead of
	219	* the value we deduce from time comparison.
	220	*
	221	* Suppose we have 1s track started at t=0, and another track begins to
	222	* play at t=2s. Suppose RTP_AHEAD_MS=1000 and 44100Hz stereo. In that
	223	* case we'll send 1s of audio as fast as we can, giving rtp_time=88200.
	224	* rtp_time stops at this point.
	225	*
	226	* At t=2s we'll have calculated target_rtp_time=176400. In this case we
	227	* set rtp_time=176400 and the player can correctly conclude that it
	228	* should leave 1s between the tracks.
	229	*
	230	* Suppose instead that the second track arrives at t=0.5s, and that
	231	* we've managed to transmit the whole of the first track already. We'll
	232	* have target_rtp_time=44100.
	233	*
	234	* The desired behaviour is to play the second track back to back with
	235	* first. In this case therefore we do not modify rtp_time.
	236	*
	237	* Is it ever right to reduce rtp_time? No; for that would imply
	238	* transmitting packets with overlapping timestamp ranges, which does not
	239	* make sense.
	240	*/
	241	target_rtp_time &= ~(uint64_t)1; /* stereo! */
	242	if(target_rtp_time > rtp_time) {
	243	/* More time has elapsed than we've transmitted samples. That implies
	244	* we've been 'sending' silence. */
	245	info("advancing rtp_time by %"PRIu64" samples",
	246	target_rtp_time - rtp_time);
	247	rtp_time = target_rtp_time;
	248	} else if(target_rtp_time < rtp_time) {
	249	const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS
	250	* config->sample_format.rate
	251	* config->sample_format.channels
	252	/ 1000);
	253
	254	if(target_rtp_time + samples_ahead < rtp_time) {
	255	info("reversing rtp_time by %"PRIu64" samples",
	256	rtp_time - target_rtp_time);
	257	}
	258	}
	259	}
	260	header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */
	261	header.seq = htons(rtp_seq++);
	262	header.timestamp = htonl((uint32_t)rtp_time);
	263	header.ssrc = rtp_id;
	264	header.mpt = (idled ? 0x80 : 0x00) \| 10;
	265	/* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from
	266	* the sample rate (in a library somewhere so that configuration.c can rule
	267	* out invalid rates).
	268	*/
	269	idled = 0;
	270	if(bytes > NETWORK_BYTES - sizeof header) {
	271	bytes = NETWORK_BYTES - sizeof header;
	272	/* Always send a whole number of frames */
6d2d327c	273	bytes -= bytes % bpf;
1c3f1e73	274	}
	275	/* "The RTP clock rate used for generating the RTP timestamp is independent
	276	* of the number of channels and the encoding; it equals the number of
	277	* sampling periods per second. For N-channel encodings, each sampling
	278	* period (say, 1/8000 of a second) generates N samples. (This terminology
	279	* is standard, but somewhat confusing, as the total number of samples
	280	* generated per second is then the sampling rate times the channel
	281	* count.)"
	282	*/
	283	vec[0].iov_base = (void *)&header;
	284	vec[0].iov_len = sizeof header;
	285	vec[1].iov_base = playing->buffer + playing->start;
	286	vec[1].iov_len = bytes;
	287	do {
	288	written_bytes = writev(bfd, vec, 2);
	289	} while(written_bytes < 0 && errno == EINTR);
	290	if(written_bytes < 0) {
	291	error(errno, "error transmitting audio data");
	292	++audio_errors;
	293	if(audio_errors == 10)
	294	fatal(0, "too many audio errors");
	295	return 0;
	296	} else
	297	audio_errors /= 2;
	298	written_bytes -= sizeof (struct rtp_header);
6d2d327c	299	written_frames = written_bytes / bpf;
1c3f1e73	300	/* Advance RTP's notion of the time */
6d2d327c	301	rtp_time += written_frames * config->sample_format.channels;
1c3f1e73	302	return written_frames;
	303	}
	304
	305	static int bfd_slot;
	306
	307	/** @brief Set up poll array for network play */
e84fb5f0	308	static void network_beforepoll(int *timeoutp) {
1c3f1e73	309	struct timeval now;
	310	uint64_t target_us;
	311	uint64_t target_rtp_time;
e84fb5f0 RK	312	const int64_t samples_per_second = config->sample_format.rate
e84fb5f0 RK	313	* config->sample_format.channels;
1c3f1e73	314	const int64_t samples_ahead = ((uint64_t)RTP_AHEAD_MS
e84fb5f0	315	* samples_per_second
1c3f1e73	316	/ 1000);
e84fb5f0	317	int64_t lead, ahead_ms;
1c3f1e73	318
	319	/* If we're starting then initialize the base time */
	320	if(!rtp_time)
	321	xgettimeofday(&rtp_time_0, 0);
	322	/* We send audio data whenever we get RTP_AHEAD seconds or more
	323	* behind */
	324	xgettimeofday(&now, 0);
	325	target_us = tvsub_us(now, rtp_time_0);
	326	assert(target_us <= UINT64_MAX / 88200);
	327	target_rtp_time = (target_us * config->sample_format.rate
	328	* config->sample_format.channels)
	329	/ 1000000;
e84fb5f0 RK	330	lead = rtp_time - target_rtp_time;
	331	if(lead < samples_ahead)
	332	/* We've not reached the desired lead, write as fast as we can */
1c3f1e73	333	bfd_slot = addfd(bfd, POLLOUT);
e84fb5f0 RK	334	else {
	335	/* We've reached the desired lead, we can afford to wait a bit even if the
	336	* IP stack thinks it can accept more. */
	337	ahead_ms = 1000 * (lead - samples_ahead) / samples_per_second;
	338	if(ahead_ms < *timeoutp)
	339	*timeoutp = ahead_ms;
	340	}
1c3f1e73	341	}
	342
	343	/** @brief Process poll() results for network play */
	344	static int network_ready(void) {
	345	if(fds[bfd_slot].revents & (POLLOUT \| POLLERR))
	346	return 1;
	347	else
	348	return 0;
	349	}
	350
	351	const struct speaker_backend network_backend = {
	352	BACKEND_NETWORK,
6d2d327c	353	0,
1c3f1e73	354	network_init,
	355	0, /* activate */
	356	network_play,
	357	0, /* deactivate */
	358	network_beforepoll,
	359	network_ready
	360	};
	361
	362	/*
	363	Local Variables:
	364	c-basic-offset:2
	365	comment-column:40
	366	fill-column:79
	367	indent-tabs-mode:nil
	368	End:
	369	*/