[disorder] / server / speaker-network.c

/*
 * This file is part of DisOrder
 * Copyright (C) 2005-2008 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 3 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, see <http://www.gnu.org/licenses/>.
 */
/** @file server/speaker-network.c
 * @brief Support for @ref BACKEND_NETWORK */

#include "common.h"

#include <unistd.h>
#include <poll.h>
#include <netdb.h>
#include <gcrypt.h>
#include <sys/socket.h>
#include <sys/uio.h>
#include <net/if.h>
#include <ifaddrs.h>
#include <errno.h>
#include <netinet/in.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 = {
    .ai_flags = 0,
    .ai_family = PF_INET,
    .ai_socktype = SOCK_DGRAM,
    .ai_protocol = IPPROTO_UDP,
  };
  static const struct addrinfo prefbind = {
    .ai_flags = AI_PASSIVE,
    .ai_family = PF_INET,
    .ai_socktype = SOCK_DGRAM,
    .ai_protocol = IPPROTO_UDP,
  };
  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 we're starting then initialize the base time */
  if(!rtp_time)
    xgettimeofday(&rtp_time_0, 0);
  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);
    if(delta > UINT64_MAX / 88200)
      fatal(0, "rtp_time=%llu now=%ld.%06ld rtp_time_0=%ld.%06ld delta=%llu (%lld)",
            rtp_time,
            (long)now.tv_sec, (long)now.tv_usec,
            (long)rtp_time_0.tv_sec, (long)rtp_time_0.tv_usec,
            delta, delta);
    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 44100Hz stereo.  We send 1s of audio over the
     * next (about) one second, 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.
     *
     * It's never right to reduce rtp_time, for that would imply 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) {
      info("would reverse 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;
  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 would otherwise get behind */
  xgettimeofday(&now, 0);
  target_us = tvsub_us(now, rtp_time_0);
  if(target_us > UINT64_MAX / 88200)
    fatal(0, "rtp_time=%llu rtp_time_0=%ld.%06ld now=%ld.%06ld target_us=%llu (%lld)\n",
          rtp_time,
          (long)rtp_time_0.tv_sec, (long)rtp_time_0.tv_usec,
          (long)now.tv_sec, (long)now.tv_usec,
          target_us, target_us);
  target_rtp_time = (target_us * config->sample_format.rate
                               * config->sample_format.channels)
                     / 1000000;
  /* Lead is how far ahead we are */
  lead = rtp_time - target_rtp_time;
  if(lead <= 0)
    /* We're behind or even, so we'll need to write as soon as we can */
    bfd_slot = addfd(bfd, POLLOUT);
  else {
    /* We've ahead, we can afford to wait a bit even if the IP stack thinks it
     * can accept more. */
    ahead_ms = 1000 * lead / 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	/*
1c3f1e73	2	* This file is part of DisOrder
5aff007d	3	* Copyright (C) 2005-2008 Richard Kettlewell
1c3f1e73	4	*
e7eb3a27	5	* This program is free software: you can redistribute it and/or modify
1c3f1e73	6	* it under the terms of the GNU General Public License as published by
e7eb3a27	7	* the Free Software Foundation, either version 3 of the License, or
1c3f1e73	8	* (at your option) any later version.
1c3f1e73	9	*
e7eb3a27 RK	10	* This program is distributed in the hope that it will be useful,
	11	* but WITHOUT ANY WARRANTY; without even the implied warranty of
	12	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
	13	* GNU General Public License for more details.
	14	*
1c3f1e73	15	* You should have received a copy of the GNU General Public License
e7eb3a27	16	* along with this program. If not, see <http://www.gnu.org/licenses/>.
1c3f1e73	17	*/
	18	/** @file server/speaker-network.c
	19	* @brief Support for @ref BACKEND_NETWORK */
	20
05b75f8d	21	#include "common.h"
1c3f1e73	22
	23	#include <unistd.h>
	24	#include <poll.h>
	25	#include <netdb.h>
	26	#include <gcrypt.h>
	27	#include <sys/socket.h>
	28	#include <sys/uio.h>
81b1bf12	29	#include <net/if.h>
db2c19dc	30	#include <ifaddrs.h>
6d2d327c	31	#include <errno.h>
edbd470f	32	#include <netinet/in.h>
1c3f1e73	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"
81b1bf12	40	#include "ifreq.h"
1c3f1e73	41	#include "speaker-protocol.h"
	42	#include "speaker.h"
	43
	44	/** @brief Network socket
	45	*
	46	* This is the file descriptor to write to for @ref BACKEND_NETWORK.
	47	*/
	48	static int bfd = -1;
	49
	50	/** @brief RTP timestamp
	51	*
	52	* This counts the number of samples played (NB not the number of frames
	53	* played).
	54	*
	55	* The timestamp in the packet header is only 32 bits wide. With 44100Hz
	56	* stereo, that only gives about half a day before wrapping, which is not
	57	* particularly convenient for certain debugging purposes. Therefore the
	58	* timestamp is maintained as a 64-bit integer, giving around six million years
	59	* before wrapping, and truncated to 32 bits when transmitting.
	60	*/
	61	static uint64_t rtp_time;
	62
	63	/** @brief RTP base timestamp
	64	*
	65	* This is the real time correspoding to an @ref rtp_time of 0. It is used
	66	* to recalculate the timestamp after idle periods.
	67	*/
	68	static struct timeval rtp_time_0;
	69
	70	/** @brief RTP packet sequence number */
	71	static uint16_t rtp_seq;
	72
	73	/** @brief RTP SSRC */
	74	static uint32_t rtp_id;
	75
	76	/** @brief Error counter */
	77	static int audio_errors;
	78
	79	/** @brief Network backend initialization */
	80	static void network_init(void) {
	81	struct addrinfo res, sres;
	82	static const struct addrinfo pref = {
66613034 RK	83	.ai_flags = 0,
	84	.ai_family = PF_INET,
	85	.ai_socktype = SOCK_DGRAM,
	86	.ai_protocol = IPPROTO_UDP,
1c3f1e73	87	};
1c3f1e73	88	static const struct addrinfo prefbind = {
66613034 RK	89	.ai_flags = AI_PASSIVE,
	90	.ai_family = PF_INET,
	91	.ai_socktype = SOCK_DGRAM,
	92	.ai_protocol = IPPROTO_UDP,
1c3f1e73	93	};
1c3f1e73	94	static const int one = 1;
db2c19dc	95	int sndbuf, target_sndbuf = 131072;
1c3f1e73	96	socklen_t len;
	97	char sockname, ssockname;
	98
	99	res = get_address(&config->broadcast, &pref, &sockname);
	100	if(!res) exit(-1);
	101	if(config->broadcast_from.n) {
	102	sres = get_address(&config->broadcast_from, &prefbind, &ssockname);
	103	if(!sres) exit(-1);
	104	} else
	105	sres = 0;
	106	if((bfd = socket(res->ai_family,
	107	res->ai_socktype,
	108	res->ai_protocol)) < 0)
	109	fatal(errno, "error creating broadcast socket");
6fba990c	110	if(multicast(res->ai_addr)) {
23205f9c RK	111	/* Multicasting */
	112	switch(res->ai_family) {
	113	case PF_INET: {
	114	const int mttl = config->multicast_ttl;
	115	if(setsockopt(bfd, IPPROTO_IP, IP_MULTICAST_TTL, &mttl, sizeof mttl) < 0)
	116	fatal(errno, "error setting IP_MULTICAST_TTL on multicast socket");
61941295 RK	117	if(setsockopt(bfd, IPPROTO_IP, IP_MULTICAST_LOOP,
	118	&config->multicast_loop, sizeof one) < 0)
	119	fatal(errno, "error setting IP_MULTICAST_LOOP on multicast socket");
23205f9c RK	120	break;
	121	}
	122	case PF_INET6: {
	123	const int mttl = config->multicast_ttl;
	124	if(setsockopt(bfd, IPPROTO_IPV6, IPV6_MULTICAST_HOPS,
	125	&mttl, sizeof mttl) < 0)
	126	fatal(errno, "error setting IPV6_MULTICAST_HOPS on multicast socket");
61941295 RK	127	if(setsockopt(bfd, IPPROTO_IP, IPV6_MULTICAST_LOOP,
	128	&config->multicast_loop, sizeof (int)) < 0)
	129	fatal(errno, "error setting IPV6_MULTICAST_LOOP on multicast socket");
23205f9c RK	130	break;
	131	}
	132	default:
	133	fatal(0, "unsupported address family %d", res->ai_family);
	134	}
81b1bf12	135	info("multicasting on %s", sockname);
23205f9c	136	} else {
db2c19dc	137	struct ifaddrs *ifs;
81b1bf12	138
db2c19dc RK	139	if(getifaddrs(&ifs) < 0)
	140	fatal(errno, "error calling getifaddrs");
	141	while(ifs) {
3aa6f359 RK	142	/* (At least on Darwin) IFF_BROADCAST might be set but ifa_broadaddr
	143	* still a null pointer. It turns out that there's a subsequent entry
	144	* for he same interface which _does_ have ifa_broadaddr though... */
db2c19dc	145	if((ifs->ifa_flags & IFF_BROADCAST)
3aa6f359	146	&& ifs->ifa_broadaddr
db2c19dc	147	&& sockaddr_equal(ifs->ifa_broadaddr, res->ai_addr))
81b1bf12	148	break;
db2c19dc	149	ifs = ifs->ifa_next;
81b1bf12	150	}
db2c19dc	151	if(ifs) {
81b1bf12 RK	152	if(setsockopt(bfd, SOL_SOCKET, SO_BROADCAST, &one, sizeof one) < 0)
81b1bf12 RK	153	fatal(errno, "error setting SO_BROADCAST on broadcast socket");
db2c19dc	154	info("broadcasting on %s (%s)", sockname, ifs->ifa_name);
81b1bf12 RK	155	} else
81b1bf12 RK	156	info("unicasting on %s", sockname);
23205f9c	157	}
1c3f1e73	158	len = sizeof sndbuf;
	159	if(getsockopt(bfd, SOL_SOCKET, SO_SNDBUF,
	160	&sndbuf, &len) < 0)
	161	fatal(errno, "error getting SO_SNDBUF");
	162	if(target_sndbuf > sndbuf) {
	163	if(setsockopt(bfd, SOL_SOCKET, SO_SNDBUF,
	164	&target_sndbuf, sizeof target_sndbuf) < 0)
	165	error(errno, "error setting SO_SNDBUF to %d", target_sndbuf);
	166	else
	167	info("changed socket send buffer size from %d to %d",
	168	sndbuf, target_sndbuf);
	169	} else
	170	info("default socket send buffer is %d",
	171	sndbuf);
	172	/* We might well want to set additional broadcast- or multicast-related
	173	* options here */
	174	if(sres && bind(bfd, sres->ai_addr, sres->ai_addrlen) < 0)
	175	fatal(errno, "error binding broadcast socket to %s", ssockname);
	176	if(connect(bfd, res->ai_addr, res->ai_addrlen) < 0)
	177	fatal(errno, "error connecting broadcast socket to %s", sockname);
	178	/* Select an SSRC */
	179	gcry_randomize(&rtp_id, sizeof rtp_id, GCRY_STRONG_RANDOM);
1c3f1e73	180	}
	181
	182	/** @brief Play over the network */
	183	static size_t network_play(size_t frames) {
	184	struct rtp_header header;
	185	struct iovec vec[2];
6d2d327c	186	size_t bytes = frames * bpf, written_frames;
1c3f1e73	187	int written_bytes;
	188	/* We transmit using RTP (RFC3550) and attempt to conform to the internet
	189	* AVT profile (RFC3551). */
	190
23741390 RK	191	/* If we're starting then initialize the base time */
	192	if(!rtp_time)
	193	xgettimeofday(&rtp_time_0, 0);
1c3f1e73	194	if(idled) {
	195	/* There may have been a gap. Fix up the RTP time accordingly. */
	196	struct timeval now;
	197	uint64_t delta;
	198	uint64_t target_rtp_time;
	199
	200	/* Find the current time */
	201	xgettimeofday(&now, 0);
	202	/* Find the number of microseconds elapsed since rtp_time=0 */
	203	delta = tvsub_us(now, rtp_time_0);
23741390 RK	204	if(delta > UINT64_MAX / 88200)
	205	fatal(0, "rtp_time=%llu now=%ld.%06ld rtp_time_0=%ld.%06ld delta=%llu (%lld)",
	206	rtp_time,
	207	(long)now.tv_sec, (long)now.tv_usec,
	208	(long)rtp_time_0.tv_sec, (long)rtp_time_0.tv_usec,
	209	delta, delta);
6d2d327c RK	210	target_rtp_time = (delta * config->sample_format.rate
6d2d327c RK	211	* config->sample_format.channels) / 1000000;
1c3f1e73	212	/* Overflows at ~6 years uptime with 44100Hz stereo */
	213
	214	/* rtp_time is the number of samples we've played. NB that we play
	215	* RTP_AHEAD_MS ahead of ourselves, so it may legitimately be ahead of
	216	* the value we deduce from time comparison.
	217	*
	218	* Suppose we have 1s track started at t=0, and another track begins to
24d6fd25 RK	219	* play at t=2s. Suppose 44100Hz stereo. We send 1s of audio over the
	220	* next (about) one second, giving rtp_time=88200. rtp_time stops at this
	221	* point.
1c3f1e73	222	*
	223	* At t=2s we'll have calculated target_rtp_time=176400. In this case we
	224	* set rtp_time=176400 and the player can correctly conclude that it
	225	* should leave 1s between the tracks.
	226	*
24d6fd25 RK	227	* It's never right to reduce rtp_time, for that would imply packets with
24d6fd25 RK	228	* overlapping timestamp ranges, which does not make sense.
1c3f1e73	229	*/
	230	target_rtp_time &= ~(uint64_t)1; /* stereo! */
	231	if(target_rtp_time > rtp_time) {
	232	/* More time has elapsed than we've transmitted samples. That implies
	233	* we've been 'sending' silence. */
	234	info("advancing rtp_time by %"PRIu64" samples",
	235	target_rtp_time - rtp_time);
	236	rtp_time = target_rtp_time;
	237	} else if(target_rtp_time < rtp_time) {
24d6fd25 RK	238	info("would reverse rtp_time by %"PRIu64" samples",
24d6fd25 RK	239	rtp_time - target_rtp_time);
1c3f1e73	240	}
	241	}
	242	header.vpxcc = 2 << 6; /* V=2, P=0, X=0, CC=0 */
	243	header.seq = htons(rtp_seq++);
	244	header.timestamp = htonl((uint32_t)rtp_time);
	245	header.ssrc = rtp_id;
	246	header.mpt = (idled ? 0x80 : 0x00) \| 10;
	247	/* 10 = L16 = 16-bit x 2 x 44100KHz. We ought to deduce this value from
	248	* the sample rate (in a library somewhere so that configuration.c can rule
	249	* out invalid rates).
	250	*/
	251	idled = 0;
	252	if(bytes > NETWORK_BYTES - sizeof header) {
	253	bytes = NETWORK_BYTES - sizeof header;
	254	/* Always send a whole number of frames */
6d2d327c	255	bytes -= bytes % bpf;
1c3f1e73	256	}
	257	/* "The RTP clock rate used for generating the RTP timestamp is independent
	258	* of the number of channels and the encoding; it equals the number of
	259	* sampling periods per second. For N-channel encodings, each sampling
	260	* period (say, 1/8000 of a second) generates N samples. (This terminology
	261	* is standard, but somewhat confusing, as the total number of samples
	262	* generated per second is then the sampling rate times the channel
	263	* count.)"
	264	*/
	265	vec[0].iov_base = (void *)&header;
	266	vec[0].iov_len = sizeof header;
	267	vec[1].iov_base = playing->buffer + playing->start;
	268	vec[1].iov_len = bytes;
	269	do {
	270	written_bytes = writev(bfd, vec, 2);
	271	} while(written_bytes < 0 && errno == EINTR);
	272	if(written_bytes < 0) {
	273	error(errno, "error transmitting audio data");
	274	++audio_errors;
	275	if(audio_errors == 10)
	276	fatal(0, "too many audio errors");
	277	return 0;
	278	} else
	279	audio_errors /= 2;
	280	written_bytes -= sizeof (struct rtp_header);
6d2d327c	281	written_frames = written_bytes / bpf;
1c3f1e73	282	/* Advance RTP's notion of the time */
6d2d327c	283	rtp_time += written_frames * config->sample_format.channels;
1c3f1e73	284	return written_frames;
	285	}
	286
	287	static int bfd_slot;
	288
	289	/** @brief Set up poll array for network play */
e84fb5f0	290	static void network_beforepoll(int *timeoutp) {
1c3f1e73	291	struct timeval now;
	292	uint64_t target_us;
	293	uint64_t target_rtp_time;
e84fb5f0 RK	294	const int64_t samples_per_second = config->sample_format.rate
e84fb5f0 RK	295	* config->sample_format.channels;
e84fb5f0	296	int64_t lead, ahead_ms;
1c3f1e73	297
	298	/* If we're starting then initialize the base time */
	299	if(!rtp_time)
	300	xgettimeofday(&rtp_time_0, 0);
24d6fd25	301	/* We send audio data whenever we would otherwise get behind */
1c3f1e73	302	xgettimeofday(&now, 0);
1c3f1e73	303	target_us = tvsub_us(now, rtp_time_0);
23741390 RK	304	if(target_us > UINT64_MAX / 88200)
	305	fatal(0, "rtp_time=%llu rtp_time_0=%ld.%06ld now=%ld.%06ld target_us=%llu (%lld)\n",
	306	rtp_time,
	307	(long)rtp_time_0.tv_sec, (long)rtp_time_0.tv_usec,
	308	(long)now.tv_sec, (long)now.tv_usec,
	309	target_us, target_us);
1c3f1e73	310	target_rtp_time = (target_us * config->sample_format.rate
	311	* config->sample_format.channels)
	312	/ 1000000;
24d6fd25	313	/* Lead is how far ahead we are */
e84fb5f0	314	lead = rtp_time - target_rtp_time;
24d6fd25 RK	315	if(lead <= 0)
24d6fd25 RK	316	/* We're behind or even, so we'll need to write as soon as we can */
1c3f1e73	317	bfd_slot = addfd(bfd, POLLOUT);
e84fb5f0	318	else {
24d6fd25 RK	319	/* We've ahead, we can afford to wait a bit even if the IP stack thinks it
	320	* can accept more. */
	321	ahead_ms = 1000 * lead / samples_per_second;
e84fb5f0 RK	322	if(ahead_ms < *timeoutp)
	323	*timeoutp = ahead_ms;
	324	}
1c3f1e73	325	}
	326
	327	/** @brief Process poll() results for network play */
	328	static int network_ready(void) {
	329	if(fds[bfd_slot].revents & (POLLOUT \| POLLERR))
	330	return 1;
	331	else
	332	return 0;
	333	}
	334
	335	const struct speaker_backend network_backend = {
	336	BACKEND_NETWORK,
6d2d327c	337	0,
1c3f1e73	338	network_init,
	339	0, /* activate */
	340	network_play,
	341	0, /* deactivate */
	342	network_beforepoll,
	343	network_ready
	344	};
	345
	346	/*
	347	Local Variables:
	348	c-basic-offset:2
	349	comment-column:40
	350	fill-column:79
	351	indent-tabs-mode:nil
	352	End:
	353	*/