rtp.c

/*
 * Apple RTP protocol handler. This file is part of Shairport.
 * Copyright (c) James Laird 2013
 * All rights reserved.
 *
 * Permission is hereby granted, free of charge, to any person
 * obtaining a copy of this software and associated documentation
 * files (the "Software"), to deal in the Software without
 * restriction, including without limitation the rights to use,
 * copy, modify, merge, publish, distribute, sublicense, and/or
 * sell copies of the Software, and to permit persons to whom the
 * Software is furnished to do so, subject to the following conditions:
 *
 * The above copyright notice and this permission notice shall be
 * included in all copies or substantial portions of the Software.
 *
 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
 * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES
 * OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND
 * NONINFRINGEMENT. IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT
 * HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY,
 * WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
 * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR
 * OTHER DEALINGS IN THE SOFTWARE.
 */

#include <time.h>
#include <pthread.h>
#include <signal.h>
#include <unistd.h>
#include <memory.h>
#include <math.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#include <netdb.h>
#include <stdio.h>
#include <errno.h>

#include "common.h"
#include "player.h"
#include "rtp.h"

/*
// this does not compile properly with OpenWrt Barrier Breaker...
#if defined(__linux__)
#include <linux/in6.h>
#endif
*/
typedef struct time_ping_record {
  uint64_t local_to_remote_difference;
  uint64_t dispersion;
  uint64_t local_time;
  uint64_t remote_time;
} time_ping_record;

// only one RTP session can be active at a time.
static int running = 0;

static char client_ip_string[INET6_ADDRSTRLEN]; // the ip string pointing to the client
static short client_ip_family;                  // AF_INET / AF_INET6
static uint32_t client_active_remote;           // used when you want to control the client...

static SOCKADDR rtp_client_control_socket; // a socket pointing to the control port of the client
static SOCKADDR rtp_client_timing_socket;  // a socket pointing to the timing port of the client
static int audio_socket;                   // our local [server] audio socket
static int control_socket;                 // our local [server] control socket
static int timing_socket;                  // local timing socket
//static pthread_t rtp_audio_thread, rtp_control_thread, rtp_timing_thread;

static uint32_t reference_timestamp;
static uint64_t reference_timestamp_time;
static uint64_t remote_reference_timestamp_time;

// debug variables
static int request_sent;

#define time_ping_history 8
#define time_ping_fudge_factor 100000

static uint8_t time_ping_count;
struct time_ping_record time_pings[time_ping_history];

// static struct timespec dtt; // dangerous -- this assumes that there will never be two timing
// request in flight at the same time
static uint64_t departure_time; // dangerous -- this assumes that there will never be two timing
                                // request in flight at the same time

static pthread_mutex_t reference_time_mutex = PTHREAD_MUTEX_INITIALIZER;

uint64_t static local_to_remote_time_difference; // used to switch between local and remote clocks

void *rtp_audio_receiver(void *arg) {
  debug(2, "Audio receiver -- Server RTP thread starting.");

  // we inherit the signal mask (SIGUSR1)
  struct inter_threads_record *itr = arg;

  int32_t last_seqno = -1;
  uint8_t packet[2048], *pktp;

  uint64_t time_of_previous_packet_fp = 0;
  float longest_packet_time_interval_us = 0.0;
  
  // mean and variance calculations from "online_variance" algorithm at https://en.wikipedia.org/wiki/Algorithms_for_calculating_variance#Online_algorithm
  
  int32_t stat_n = 0;
  float stat_mean = 0.0;
  float stat_M2 = 0.0;
  
  ssize_t nread;
  while (itr->please_stop==0) {
    nread = recv(audio_socket, packet, sizeof(packet), 0);
    
    uint64_t local_time_now_fp = get_absolute_time_in_fp();
    if (time_of_previous_packet_fp) {
      float time_interval_us = (((local_time_now_fp - time_of_previous_packet_fp)*1000000)>>32)*1.0;
      time_of_previous_packet_fp = local_time_now_fp;
      if (time_interval_us>longest_packet_time_interval_us)
        longest_packet_time_interval_us=time_interval_us;
      stat_n+=1;
      float stat_delta = time_interval_us - stat_mean;
      stat_mean += stat_delta/stat_n;
      stat_M2 += stat_delta*(time_interval_us - stat_mean);
      if (stat_n % 2500 == 0) {
        debug(2,"Packet reception interval stats: mean, standard deviation and max for the last 2,500 packets in microseconds: %10.1f, %10.1f, %10.1f.",stat_mean, sqrtf(stat_M2 / (stat_n-1)),longest_packet_time_interval_us);
        stat_n = 0;
        stat_mean = 0.0;
        stat_M2 = 0.0;
        time_of_previous_packet_fp = 0;
        longest_packet_time_interval_us = 0.0;
      }
    } else {
      time_of_previous_packet_fp = local_time_now_fp;
    }

    if (nread < 0)
      break;

    ssize_t plen = nread;
    uint8_t type = packet[1] & ~0x80;
    if (type == 0x60 || type == 0x56) { // audio data / resend
      pktp = packet;
      if (type == 0x56) {
        pktp += 4;
        plen -= 4;
      }
      seq_t seqno = ntohs(*(unsigned short *)(pktp + 2));
      // increment last_seqno and see if it's the same as the incoming seqno

      if (last_seqno == -1)
        last_seqno = seqno;
      else {
        last_seqno = (last_seqno + 1) & 0xffff;
        //if (seqno != last_seqno)
        //  debug(3, "RTP: Packets out of sequence: expected: %d, got %d.", last_seqno, seqno);
        last_seqno = seqno; // reset warning...
      }
      uint32_t timestamp = ntohl(*(unsigned long *)(pktp + 4));

      // if (packet[1]&0x10)
      //	debug(1,"Audio packet Extension bit set.");

      pktp += 12;
      plen -= 12;

      // check if packet contains enough content to be reasonable
      if (plen >= 16) {
        player_put_packet(seqno, timestamp, pktp, plen);
        continue;
      }
      if (type == 0x56 && seqno == 0) {
        debug(2, "resend-related request packet received, ignoring.");
        continue;
      }
      debug(1, "Audio receiver -- Unknown RTP packet of type 0x%02X length %d seqno %d", type,
            nread, seqno);
    }
    warn("Audio receiver -- Unknown RTP packet of type 0x%02X length %d.", type, nread);
  }

  debug(1, "Audio receiver -- Server RTP thread interrupted. terminating.");
  close(audio_socket);

  return NULL;
}

void *rtp_control_receiver(void *arg) {
  // we inherit the signal mask (SIGUSR1)

  debug(2, "Control receiver -- Server RTP thread starting.");
  struct inter_threads_record *itr = arg;

  reference_timestamp = 0; // nothing valid received yet
  uint8_t packet[2048], *pktp;
  struct timespec tn;
  uint64_t remote_time_of_sync, local_time_now, remote_time_now;
  uint32_t sync_rtp_timestamp, rtp_timestamp_less_latency;
  ssize_t nread;
  while (itr->please_stop==0) {
    nread = recv(control_socket, packet, sizeof(packet), 0);
    local_time_now = get_absolute_time_in_fp();
    //        clock_gettime(CLOCK_MONOTONIC,&tn);
    //        local_time_now=((uint64_t)tn.tv_sec<<32)+((uint64_t)tn.tv_nsec<<32)/1000000000;

    if (nread < 0)
      break;

    ssize_t plen = nread;
    if (packet[1] == 0xd4) { // sync data
      /*
      char obf[4096];
      char *obfp = obf;
      int obfc;
      for (obfc=0;obfc<plen;obfc++) {
        sprintf(obfp,"%02X",packet[obfc]);
        obfp+=2;
      };
      *obfp=0;
      debug(1,"Sync Packet Received: \"%s\"",obf);
      */
      if (local_to_remote_time_difference) { // need a time packet to be interchanged first...

        remote_time_of_sync = (uint64_t)ntohl(*((uint32_t *)&packet[8])) << 32;
        remote_time_of_sync += ntohl(*((uint32_t *)&packet[12]));

        // debug(1,"Remote Sync Time: %0llx.",remote_time_of_sync);

        rtp_timestamp_less_latency = ntohl(*((uint32_t *)&packet[4]));
        sync_rtp_timestamp = ntohl(*((uint32_t *)&packet[16]));
        
        if (config.use_negotiated_latencies) {
          uint32_t la = sync_rtp_timestamp-rtp_timestamp_less_latency+11025;
          if (la!=config.latency) {
            config.latency = la;
            // debug(1,"Using negotiated latency of %u frames.",config.latency);
          }
        }
        
        if (packet[0] & 0x10) {
          // if it's a packet right after a flush or resume
          sync_rtp_timestamp += 352; // add frame_size -- can't see a reference to this anywhere,
                                     // but it seems to get everything into sync.
          // it's as if the first sync after a flush or resume is the timing of the next packet
          // after the one whose RTP is given. Weird.
        }
        pthread_mutex_lock(&reference_time_mutex);
        remote_reference_timestamp_time = remote_time_of_sync;
        reference_timestamp_time = remote_time_of_sync - local_to_remote_time_difference;
        reference_timestamp = sync_rtp_timestamp;
        pthread_mutex_unlock(&reference_time_mutex);
        // debug(1,"New Reference timestamp and timestamp time...");
        // get estimated remote time now
        // remote_time_now = local_time_now + local_to_remote_time_difference;

        // debug(1,"Sync Time is %lld us late (remote
        // times).",((remote_time_now-remote_time_of_sync)*1000000)>>32);
        // debug(1,"Sync Time is %lld us late (local
        // times).",((local_time_now-reference_timestamp_time)*1000000)>>32);
      } else {
        debug(1, "Sync packet received before we got a timing packet back.");
      }
    } else if (packet[1] == 0xd6) { // resent audio data in the control path -- whaale only?
      // debug(1, "Control Port -- Retransmitted Audio Data Packet received.");
      pktp = packet+4;
      plen -= 4;
      seq_t seqno = ntohs(*(unsigned short *)(pktp + 2));

      uint32_t timestamp = ntohl(*(unsigned long *)(pktp + 4));

      pktp += 12;
      plen -= 12;

      // check if packet contains enough content to be reasonable
      if (plen >= 16) {
        player_put_packet(seqno, timestamp, pktp, plen);
        continue;
      } else {
        debug(1, "Too-short retransmitted audio packet received in control port, ignored.");
      }
    } else
      debug(1, "Control Port -- Unknown RTP packet of type 0x%02X length %d.", packet[1], nread);
  }

  debug(1, "Control RTP thread interrupted. terminating.");
  close(control_socket);

  return NULL;
}

void *rtp_timing_sender(void *arg) {
  debug(2, "Timing sender thread starting.");
	int *stop = arg; // the parameter points to this request to stop thing
  struct timing_request {
    char leader;
    char type;
    uint16_t seqno;
    uint32_t filler;
    uint64_t origin, receive, transmit;
  };

  uint64_t request_number = 0;

  struct timing_request req; // *not* a standard RTCP NACK

  req.leader = 0x80;
  req.type = 0xd2; // Timing request
  req.filler = 0;
  req.seqno = htons(7);

  time_ping_count = 0;

  // we inherit the signal mask (SIGUSR1)
  while (*stop==0) {
  	// debug(1,"Send a timing request");

    if (!running)
      die("rtp_timing_sender called without active stream!");

    // debug(1, "Requesting ntp timestamp exchange.");

    req.filler = 0;
    req.origin = req.receive = req.transmit = 0;

    //    clock_gettime(CLOCK_MONOTONIC,&dtt);
    departure_time = get_absolute_time_in_fp();
    socklen_t msgsize = sizeof(struct sockaddr_in);
#ifdef AF_INET6
    if (rtp_client_timing_socket.SAFAMILY == AF_INET6) {
      msgsize = sizeof(struct sockaddr_in6);
    }
#endif
    if (sendto(timing_socket, &req, sizeof(req), 0, (struct sockaddr *)&rtp_client_timing_socket,
               msgsize) == -1) {
      perror("Error sendto-ing to timing socket");
    }
    request_number++;
    if (request_number <= 4)
      usleep(500000);
    else
      sleep(3);
  }
  debug(1, "rtp_timing_sender thread interrupted. terminating.");
  return NULL;
}

void *rtp_timing_receiver(void *arg) {
  debug(2, "Timing receiver -- Server RTP thread starting.");
  // we inherit the signal mask (SIGUSR1)

	struct inter_threads_record *itr = arg;

  uint8_t packet[2048], *pktp;
  ssize_t nread;
  int request_stop = 0;
  pthread_t timer_requester;
  pthread_create(&timer_requester, NULL, &rtp_timing_sender, (void *)&request_stop);
  //    struct timespec att;
  uint64_t distant_receive_time, distant_transmit_time, arrival_time, return_time, transit_time,
      processing_time;
  local_to_remote_time_jitters = 0;
  local_to_remote_time_jitters_count = 0;
  uint64_t first_remote_time = 0;
  uint64_t first_local_time = 0;
  
  uint64_t first_local_to_remote_time_difference = 0;
  uint64_t first_local_to_remote_time_difference_time;
  uint64_t l2rtd = 0;
  while (itr->please_stop==0) {
    nread = recv(timing_socket, packet, sizeof(packet), 0);
    arrival_time = get_absolute_time_in_fp();
    //      clock_gettime(CLOCK_MONOTONIC,&att);

    if (nread < 0)
      break;

    ssize_t plen = nread;
    // debug(1,"Packet Received on Timing Port.");
    if (packet[1] == 0xd3) { // timing reply
      /*
      char obf[4096];
      char *obfp = obf;
      int obfc;
      for (obfc=0;obfc<plen;obfc++) {
        sprintf(obfp,"%02X",packet[obfc]);
        obfp+=2;
      };
      *obfp=0;
      //debug(1,"Timing Packet Received: \"%s\"",obf);
      */

      // arrival_time = ((uint64_t)att.tv_sec<<32)+((uint64_t)att.tv_nsec<<32)/1000000000;
      // departure_time = ((uint64_t)dtt.tv_sec<<32)+((uint64_t)dtt.tv_nsec<<32)/1000000000;

      return_time = arrival_time - departure_time;

      // uint64_t rtus = (return_time*1000000)>>32; debug(1,"Time ping turnaround time: %lld
      // us.",rtus);

      // distant_receive_time =
      // ((uint64_t)ntohl(*((uint32_t*)&packet[16])))<<32+ntohl(*((uint32_t*)&packet[20]));

      distant_receive_time = (uint64_t)ntohl(*((uint32_t *)&packet[16])) << 32;
      distant_receive_time += ntohl(*((uint32_t *)&packet[20]));

      // distant_transmit_time =
      // ((uint64_t)ntohl(*((uint32_t*)&packet[24])))<<32+ntohl(*((uint32_t*)&packet[28]));

      distant_transmit_time = (uint64_t)ntohl(*((uint32_t *)&packet[24])) << 32;
      distant_transmit_time += ntohl(*((uint32_t *)&packet[28]));

      processing_time = distant_transmit_time - distant_receive_time;

      // debug(1,"Return trip time: %lluuS, remote processing time:
      // %lluuS.",(return_time*1000000)>>32,(processing_time*1000000)>>32);

      uint64_t local_time_by_remote_clock = distant_transmit_time + return_time / 2;

      unsigned int cc;
      for (cc = time_ping_history - 1; cc > 0; cc--) {
        time_pings[cc] = time_pings[cc - 1];
        time_pings[cc].dispersion = (time_pings[cc].dispersion * 133) /
                                    100; // make the dispersions 'age' by this rational factor
      }
      // these are for diagnostics only -- not used
      time_pings[0].local_time = arrival_time;
      time_pings[0].remote_time = distant_transmit_time;
      
      time_pings[0].local_to_remote_difference = local_time_by_remote_clock - arrival_time;
      time_pings[0].dispersion = return_time;
      if (time_ping_count < time_ping_history)
        time_ping_count++;
      
      uint64_t local_time_chosen = arrival_time;;
      uint64_t remote_time_chosen = distant_transmit_time;
      // now pick the timestamp with the lowest dispersion
      uint64_t l2rtd = time_pings[0].local_to_remote_difference;
      uint64_t tld = time_pings[0].dispersion;
      for (cc = 1; cc < time_ping_count; cc++)
        if (time_pings[cc].dispersion < tld) {
          l2rtd = time_pings[cc].local_to_remote_difference;
          tld = time_pings[cc].dispersion;
          local_time_chosen = time_pings[cc].local_time;
          remote_time_chosen = time_pings[cc].remote_time;
        }
      int64_t ji;

      if (time_ping_count > 1) {
        if (l2rtd > local_to_remote_time_difference) {
          local_to_remote_time_jitters =
              local_to_remote_time_jitters + l2rtd - local_to_remote_time_difference;
          ji = l2rtd - local_to_remote_time_difference;
        } else {
          local_to_remote_time_jitters =
              local_to_remote_time_jitters + local_to_remote_time_difference - l2rtd;
          ji = -(local_to_remote_time_difference - l2rtd);
        }
        local_to_remote_time_jitters_count += 1;
      }
      // uncomment below to print jitter between client's clock and oour clock
      // int64_t rtus = (tld*1000000)>>32; ji = (ji*1000000)>>32; debug(1,"Choosing time difference
      // with dispersion of %lld us with delta of %lld us",rtus,ji);

      local_to_remote_time_difference = l2rtd;
      if (first_local_to_remote_time_difference==0) {
        first_local_to_remote_time_difference = local_to_remote_time_difference;
        first_local_to_remote_time_difference_time = get_absolute_time_in_fp();
      }
      
      int64_t clock_drift, clock_drift_in_usec;
      if (first_local_time==0) {
        first_local_time = local_time_chosen;
        first_remote_time = remote_time_chosen;
        uint64_t clock_drift = 0;
      } else {
        uint64_t local_time_change = local_time_chosen - first_local_time;
        uint64_t remote_time_change = remote_time_chosen - first_remote_time;
        

        if (remote_time_change >= local_time_change)
          clock_drift = remote_time_change - local_time_change;
        else
          clock_drift = -(local_time_change - remote_time_change);
      }
      if (clock_drift>=0)
        clock_drift_in_usec = (clock_drift * 1000000)>>32;
      else
        clock_drift_in_usec = -(((-clock_drift) * 1000000)>>32);
      
      
     
     int64_t source_drift_usec;
     if (play_segment_reference_frame!=0) {
      uint32_t reference_timestamp;
      uint64_t reference_timestamp_time,remote_reference_timestamp_time;
      get_reference_timestamp_stuff(&reference_timestamp, &reference_timestamp_time, &remote_reference_timestamp_time);
      uint64_t frame_difference = 0;
      if (reference_timestamp>=play_segment_reference_frame)
        frame_difference = (uint64_t)reference_timestamp-(uint64_t)play_segment_reference_frame;
      else // rollover
        frame_difference = (uint64_t)reference_timestamp+0x100000000-(uint64_t)play_segment_reference_frame;
      uint64_t frame_time_difference_calculated = (((uint64_t)frame_difference<<32)/44100);
      uint64_t frame_time_difference_actual = remote_reference_timestamp_time-play_segment_reference_frame_remote_time; // this is all done by reference to the sources' system clock
      // debug(1,"%llu frames since play started, %llu usec calculated, %llu usec actual",frame_difference, (frame_time_difference_calculated*1000000)>>32, (frame_time_difference_actual*1000000)>>32);
      if (frame_time_difference_calculated>=frame_time_difference_actual) // i.e. if the time it should have taken to send the packets is greater than the actual time difference measured on the source clock
        // then the source DAC's clock is running fast relative to the source system clock
        source_drift_usec = frame_time_difference_calculated-frame_time_difference_actual;
      else
        // otherwise the source DAC's clock is running slow relative to the source system clock
        source_drift_usec = -(frame_time_difference_actual-frame_time_difference_calculated);
     } else
       source_drift_usec = 0;
     source_drift_usec = (source_drift_usec*1000000)>>32; // turn it to microseconds
      
     //long current_delay = 0;
     //if (config.output->delay) {
     //       config.output->delay(&current_delay);
     //}
     //  Useful for troubleshooting:
     //    clock_drift between source and local clock -- +ve means source is faster
     //    session_corrections -- the amount of correction done, in microseconds. +ve means frames added
     //    current_delay = delay in DAC buffer in frames
     //    source_drift_usec = how much faster (+ve) or slower the source DAC is running relative to the source clock
     //    buffer_occupancy = the number of buffers occupied. Crude, but should show no long term trend if source and device are in sync.
     //    return_time = the time from soliciting a timing packet to getting it back. It should be short ( < 5 ms) and pretty consistent.
     // debug(1, "%lld\t%lld\t%ld\t%lld\t%u\t%llu", clock_drift_in_usec,(session_corrections*1000000)/44100,current_delay,source_drift_usec,buffer_occupancy,(return_time*1000000)>>32);
      
    } else {
      debug(1, "Timing port -- Unknown RTP packet of type 0x%02X length %d.", packet[1], nread);
    }
  }

  debug(1, "Timing thread interrupted. terminating.");
  request_stop = 1;
  void *retval;
  pthread_kill(timer_requester, SIGUSR1);
  pthread_join(timer_requester, &retval);
  debug(1, "Closed and terminated timer requester thread.");
  debug(1, "Timing RTP thread terminated.");
  close(timing_socket);

  return NULL;
}

static int bind_port(SOCKADDR *remote, int *sock) {
  struct addrinfo hints, *info;

  memset(&hints, 0, sizeof(hints));
  hints.ai_family = remote->SAFAMILY;
  hints.ai_socktype = SOCK_DGRAM;
  hints.ai_flags = AI_PASSIVE;

  char buffer[10];
  
  // look for a port in the range, if any was specified.
  int desired_port = config.udp_port_base;
  int ret;
  do {
    snprintf(buffer, 10, "%d", desired_port);

    ret = getaddrinfo(NULL, buffer, &hints, &info);

    if (ret < 0)
      die("failed to get usable addrinfo?! %s.", gai_strerror(ret));

    *sock = socket(remote->SAFAMILY, SOCK_DGRAM, IPPROTO_UDP);
    
/*
  // this doesn't compile properly with OpenWrt Barrier Breaker.
    #if defined(__linux__)
    #ifdef AF_INET6
    // now, if we are on IPv6, prefer a public ipv6 address
    if (remote->SAFAMILY==AF_INET6) {
      int value = IPV6_PREFER_SRC_PUBLIC;
      ret = setsockopt(*sock, IPPROTO_IPV6, IPV6_ADDR_PREFERENCES, &value, sizeof(value));
      if (ret<0)
        die("error: could not select a preference for public IPv6 address");
    }
    #endif
    #endif
*/
    ret = bind(*sock, info->ai_addr, info->ai_addrlen);

    freeaddrinfo(info);

  } while ((ret<0) && (errno==EADDRINUSE) && (desired_port!=0) && (desired_port++ < config.udp_port_base+config.udp_port_range));
  
  // debug(1,"UDP port chosen: %d.",desired_port);
  
  if (ret < 0) {
     die("error: could not bind a UDP port!");
  }
 
  
  int sport;
  SOCKADDR local;
  socklen_t local_len = sizeof(local);
  getsockname(*sock, (struct sockaddr *)&local, &local_len);
#ifdef AF_INET6
  if (local.SAFAMILY == AF_INET6) {
    struct sockaddr_in6 *sa6 = (struct sockaddr_in6 *)&local;
    sport = ntohs(sa6->sin6_port);
  } else
#endif
  {
    struct sockaddr_in *sa = (struct sockaddr_in *)&local;
    sport = ntohs(sa->sin_port);
  }

  return sport;
}

void rtp_setup(SOCKADDR *remote, int cport, int tport, uint32_t active_remote, int *lsport,
               int *lcport, int *ltport) {
  if (running)
    die("rtp_setup called with active stream!");

  debug(2, "rtp_setup: cport=%d tport=%d.", cport, tport);

  client_active_remote = active_remote;

  // print out what we know about the client
  void *addr;
  char *ipver;
  int port;
  char portstr[20];
  client_ip_family = remote->SAFAMILY; // keep information about the kind of ip of the client
#ifdef AF_INET6
  if (remote->SAFAMILY == AF_INET6) {
    struct sockaddr_in6 *sa6 = (struct sockaddr_in6 *)remote;
    addr = &(sa6->sin6_addr);
    port = ntohs(sa6->sin6_port);
    ipver = "IPv6";
  }
#endif
  if (remote->SAFAMILY == AF_INET) {
    struct sockaddr_in *sa4 = (struct sockaddr_in *)remote;
    addr = &(sa4->sin_addr);
    port = ntohs(sa4->sin_port);
    ipver = "IPv4";
  }
  inet_ntop(remote->SAFAMILY, addr, client_ip_string,
            sizeof(client_ip_string)); // keep the client's ip number
  debug(1, "Connection from %s: %s:%d", ipver, client_ip_string, port);

  // set up a the record of the remote's control socket
  struct addrinfo hints;
  struct addrinfo *servinfo;

  memset(&rtp_client_control_socket, 0, sizeof(rtp_client_control_socket));
  memset(&hints, 0, sizeof hints);
  hints.ai_family = remote->SAFAMILY;
  hints.ai_socktype = SOCK_DGRAM;
  snprintf(portstr, 20, "%d", cport);
  if (getaddrinfo(client_ip_string, portstr, &hints, &servinfo) != 0)
    die("Can't get address of client's control port");

#ifdef AF_INET6
  if (servinfo->ai_family == AF_INET6)
    memcpy(&rtp_client_control_socket, servinfo->ai_addr, sizeof(struct sockaddr_in6));
  else
#endif
    memcpy(&rtp_client_control_socket, servinfo->ai_addr, sizeof(struct sockaddr_in));
  freeaddrinfo(servinfo);

  // set up a the record of the remote's timing socket
  memset(&rtp_client_timing_socket, 0, sizeof(rtp_client_timing_socket));
  memset(&hints, 0, sizeof hints);
  hints.ai_family = remote->SAFAMILY;
  hints.ai_socktype = SOCK_DGRAM;
  snprintf(portstr, 20, "%d", tport);
  if (getaddrinfo(client_ip_string, portstr, &hints, &servinfo) != 0)
    die("Can't get address of client's timing port");
#ifdef AF_INET6
  if (servinfo->ai_family == AF_INET6)
    memcpy(&rtp_client_timing_socket, servinfo->ai_addr, sizeof(struct sockaddr_in6));
  else
#endif
    memcpy(&rtp_client_timing_socket, servinfo->ai_addr, sizeof(struct sockaddr_in));
  freeaddrinfo(servinfo);

  // now, we open three sockets -- one for the audio stream, one for the timing and one for the
  // control

  *lsport = bind_port(remote, &audio_socket);
  *lcport = bind_port(remote, &control_socket);
  *ltport = bind_port(remote, &timing_socket);

  debug(2, "listening for audio, control and timing on ports %d, %d, %d.", *lsport, *lcport,
        *ltport);

  reference_timestamp = 0;
  //pthread_create(&rtp_audio_thread, NULL, &rtp_audio_receiver, NULL);
  //pthread_create(&rtp_control_thread, NULL, &rtp_control_receiver, NULL);
  //pthread_create(&rtp_timing_thread, NULL, &rtp_timing_receiver, NULL);

  running = 1;
  request_sent = 0;
}

void get_reference_timestamp_stuff(uint32_t *timestamp, uint64_t *timestamp_time, uint64_t *remote_timestamp_time) {
  // types okay
  pthread_mutex_lock(&reference_time_mutex);
  *timestamp = reference_timestamp;
  *timestamp_time = reference_timestamp_time;
  *remote_timestamp_time = remote_reference_timestamp_time;
  pthread_mutex_unlock(&reference_time_mutex);
}

void clear_reference_timestamp(void) {
  pthread_mutex_lock(&reference_time_mutex);
  reference_timestamp = 0;
  reference_timestamp_time = 0;
  pthread_mutex_unlock(&reference_time_mutex);
}

void rtp_shutdown(void) {
  if (!running)
    debug(1,"rtp_shutdown called without active stream!");

  debug(2, "shutting down RTP thread");
  clear_reference_timestamp();
//  debug(1,"Shut down audio, control and timing threads");
//  usleep(3000000); // hack
//  pthread_kill(rtp_audio_thread, SIGUSR1);
//  pthread_kill(rtp_control_thread, SIGUSR1);
//  pthread_kill(rtp_timing_thread, SIGUSR1);
//  pthread_join(rtp_audio_thread, &retval);
//  pthread_join(rtp_control_thread, &retval);
//  pthread_join(rtp_timing_thread, &retval);
  running = 0;
}

void rtp_request_resend(seq_t first, uint32_t count) {
  if (running) {
    //if (!request_sent) {
      debug(3, "requesting resend of %d packets starting at %u.", count, first);
    //  request_sent = 1;
    //}

    char req[8]; // *not* a standard RTCP NACK
    req[0] = 0x80;
    req[1] = 0x55 | 0x80;                        // Apple 'resend'
    *(unsigned short *)(req + 2) = htons(1);     // our seqnum
    *(unsigned short *)(req + 4) = htons(first); // missed seqnum
    *(unsigned short *)(req + 6) = htons(count); // count
    socklen_t msgsize = sizeof(struct sockaddr_in);
#ifdef AF_INET6
    if (rtp_client_control_socket.SAFAMILY == AF_INET6) {
      msgsize = sizeof(struct sockaddr_in6);
    }
#endif
    if (sendto(audio_socket, req, sizeof(req), 0, (struct sockaddr *)&rtp_client_control_socket,
               msgsize) == -1) {
      perror("Error sendto-ing to audio socket");
    }
  } else {
    //if (!request_sent) {
      debug(2, "rtp_request_resend called without active stream!");
    //  request_sent = 1;
    //}
  }
}

void rtp_request_client_pause() {
  if (running) {
    if (client_active_remote == 0) {
      debug(1, "Can't request a client pause: no valid active remote.");
    } else {
      // debug(1,"Send a client pause request to %s:3689 with active remote
      // %u.",client_ip_string,client_active_remote);

      struct addrinfo hints, *res;
      int sockfd;

      char message[1000], server_reply[2000];

      // first, load up address structs with getaddrinfo():

      memset(&hints, 0, sizeof hints);
      hints.ai_family = AF_UNSPEC;
      hints.ai_socktype = SOCK_STREAM;

      getaddrinfo(client_ip_string, "3689", &hints, &res);

      // make a socket:

      sockfd = socket(res->ai_family, res->ai_socktype, res->ai_protocol);

      if (sockfd == -1) {
        die("Could not create socket");
      }
      // debug(1,"Socket created");

      // connect!

      if (connect(sockfd, res->ai_addr, res->ai_addrlen) < 0) {
        die("connect failed. Error");
      }
      // debug(1,"Connect successful");

      sprintf(message,
              "GET /ctrl-int/1/pause HTTP/1.1\r\nHost: %s:3689\r\nActive-Remote: %u\r\n\r\n",
              client_ip_string, client_active_remote);
      // debug(1,"Sending this message: \"%s\".",message);

      // Send some data
      if (send(sockfd, message, strlen(message), 0) < 0) {
        debug(1, "Send failed");
      }

      // Receive a reply from the server
      if (recv(sockfd, server_reply, 2000, 0) < 0) {
        debug(1, "recv failed");
      }

      // debug(1,"Server replied: \"%s\".",server_reply);

      if (strstr(server_reply, "HTTP/1.1 204 No Content") != server_reply)
        debug(1, "Client pause request failed.");
      // debug(1,"Client pause request failed: \"%s\".",server_reply);
      close(sockfd);
    }
  } else {
    debug(1, "Request to pause non-existent play stream -- ignored.");
  }
}