less harsh dropping of near-empty buffers

[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 "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
};

/*
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*/