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shairport-sync/rtp.c
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/*
* Apple RTP protocol handler. This file is part of Shairport.
* Copyright (c) James Laird 2013
* Copyright (c) Mike Brady 2014 -- 2017
* 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 <arpa/inet.h>
#include <errno.h>
#include <fcntl.h>
#include <math.h>
#include <memory.h>
#include <netdb.h>
#include <netinet/in.h>
#include <pthread.h>
#include <stdio.h>
#include <sys/socket.h>
#include <sys/types.h>
#include <time.h>
#include <unistd.h>
#include "common.h"
#include "player.h"
#include "rtp.h"
void memory_barrier();
void rtp_initialise(rtsp_conn_info *conn) {
conn->rtp_running = 0;
// initialise the timer mutex
int rc = pthread_mutex_init(&conn->reference_time_mutex, NULL);
if (rc)
debug(1, "Error initialising reference_time_mutex.");
}
void rtp_terminate(rtsp_conn_info *conn) {
// destroy the timer mutex
int rc = pthread_mutex_destroy(&conn->reference_time_mutex);
if (rc)
debug(1, "Error destroying reference_time_mutex variable.");
}
void *rtp_audio_receiver(void *arg) {
debug(2, "Audio receiver -- Server RTP thread starting.");
// we inherit the signal mask (SIGUSR1)
rtsp_conn_info *conn = (rtsp_conn_info *)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 (conn->please_stop == 0) {
fd_set readfds;
FD_ZERO(&readfds);
FD_SET(conn->audio_socket, &readfds);
do {
memory_barrier();
} while (conn->please_stop == 0 &&
pselect(conn->audio_socket + 1, &readfds, NULL, NULL, NULL, &pselect_sigset) <= 0);
if (conn->please_stop != 0) {
break;
}
nread = recv(conn->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...
}
int64_t timestamp = monotonic_timestamp(ntohl(*(unsigned long *)(pktp + 4)), conn);
// 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, conn);
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(3, "Audio receiver -- Server RTP thread interrupted. terminating.");
close(conn->audio_socket);
return NULL;
}
void *rtp_control_receiver(void *arg) {
// we inherit the signal mask (SIGUSR1)
debug(2, "Control receiver -- Server RTP thread starting.");
rtsp_conn_info *conn = (rtsp_conn_info *)arg;
conn->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;
int64_t sync_rtp_timestamp, rtp_timestamp_less_latency;
ssize_t nread;
while (conn->please_stop == 0) {
fd_set readfds;
FD_ZERO(&readfds);
FD_SET(conn->control_socket, &readfds);
do {
memory_barrier();
} while (conn->please_stop == 0 &&
pselect(conn->control_socket + 1, &readfds, NULL, NULL, NULL, &pselect_sigset) <= 0);
if (conn->please_stop != 0) {
break;
}
nread = recv(conn->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 (conn->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 = monotonic_timestamp(ntohl(*((uint32_t *)&packet[4])), conn);
sync_rtp_timestamp = monotonic_timestamp(ntohl(*((uint32_t *)&packet[16])), conn);
if (config.use_negotiated_latencies) {
int64_t la = sync_rtp_timestamp - rtp_timestamp_less_latency + conn->staticLatencyCorrection;
if (la != config.latency) {
config.latency = la;
debug(1,"Using negotiated latency of %lld frames and a static latency correction of %lld",sync_rtp_timestamp - rtp_timestamp_less_latency,conn->staticLatencyCorrection);
}
}
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(&conn->reference_time_mutex);
conn->remote_reference_timestamp_time = remote_time_of_sync;
conn->reference_timestamp_time =
remote_time_of_sync - conn->local_to_remote_time_difference;
conn->reference_timestamp = sync_rtp_timestamp;
pthread_mutex_unlock(&conn->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));
int64_t timestamp = monotonic_timestamp(ntohl(*(unsigned long *)(pktp + 4)), conn);
pktp += 12;
plen -= 12;
// check if packet contains enough content to be reasonable
if (plen >= 16) {
player_put_packet(seqno, timestamp, pktp, plen, conn);
continue;
} else {
debug(3, "Too-short retransmitted audio packet received in control port, ignored.");
}
} else
debug(1, "Control Port -- Unknown RTP packet of type 0x%02X length %d, ignored.", packet[1],
nread);
}
debug(3, "Control RTP thread interrupted. terminating.");
close(conn->control_socket);
return NULL;
}
void *rtp_timing_sender(void *arg) {
debug(2, "Timing sender thread starting.");
rtsp_conn_info *conn = (rtsp_conn_info *)arg;
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);
conn->time_ping_count = 0;
// we inherit the signal mask (SIGUSR1)
while (conn->timing_sender_stop == 0) {
// debug(1,"Send a timing request");
if (!conn->rtp_running)
debug(1, "rtp_timing_sender called without active stream in RTSP conversation thread %d!",
conn->connection_number);
// debug(1, "Requesting ntp timestamp exchange.");
req.filler = 0;
req.origin = req.receive = req.transmit = 0;
// clock_gettime(CLOCK_MONOTONIC,&dtt);
conn->departure_time = get_absolute_time_in_fp();
socklen_t msgsize = sizeof(struct sockaddr_in);
#ifdef AF_INET6
if (conn->rtp_client_timing_socket.SAFAMILY == AF_INET6) {
msgsize = sizeof(struct sockaddr_in6);
}
#endif
fd_set writefds;
FD_ZERO(&writefds);
FD_SET(conn->timing_socket, &writefds);
do {
memory_barrier();
} while (conn->timing_sender_stop == 0 &&
pselect(conn->timing_socket + 1, NULL, &writefds, NULL, NULL, &pselect_sigset) <= 0);
if (conn->timing_sender_stop != 0) {
break;
}
if (sendto(conn->timing_socket, &req, sizeof(req), 0,
(struct sockaddr *)&conn->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(3, "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)
rtsp_conn_info *conn = (rtsp_conn_info *)arg;
uint8_t packet[2048], *pktp;
ssize_t nread;
conn->timing_sender_stop = 0;
pthread_t timer_requester;
pthread_create(&timer_requester, NULL, &rtp_timing_sender, arg);
// 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 (conn->please_stop == 0) {
fd_set readfds;
FD_ZERO(&readfds);
FD_SET(conn->timing_socket, &readfds);
do {
memory_barrier();
} while (conn->please_stop == 0 &&
pselect(conn->timing_socket + 1, &readfds, NULL, NULL, NULL, &pselect_sigset) <= 0);
if (conn->please_stop != 0) {
break;
}
nread = recv(conn->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 - conn->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, chosen;
for (cc = time_ping_history - 1; cc > 0; cc--) {
conn->time_pings[cc] = conn->time_pings[cc - 1];
conn->time_pings[cc].dispersion = (conn->time_pings[cc].dispersion * 110) /
100; // make the dispersions 'age' by this rational factor
}
// these are for diagnostics only -- not used
conn->time_pings[0].local_time = arrival_time;
conn->time_pings[0].remote_time = distant_transmit_time;
conn->time_pings[0].local_to_remote_difference = local_time_by_remote_clock - arrival_time;
conn->time_pings[0].dispersion = return_time;
if (conn->time_ping_count < time_ping_history)
conn->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 = conn->time_pings[0].local_to_remote_difference;
uint64_t tld = conn->time_pings[0].dispersion;
chosen = 0;
for (cc = 1; cc < conn->time_ping_count; cc++)
if (conn->time_pings[cc].dispersion < tld) {
l2rtd = conn->time_pings[cc].local_to_remote_difference;
chosen = cc;
tld = conn->time_pings[cc].dispersion;
local_time_chosen = conn->time_pings[cc].local_time;
remote_time_chosen = conn->time_pings[cc].remote_time;
}
int64_t ji;
if (conn->time_ping_count > 1) {
if (l2rtd > conn->local_to_remote_time_difference) {
local_to_remote_time_jitters =
local_to_remote_time_jitters + l2rtd - conn->local_to_remote_time_difference;
ji = l2rtd - conn->local_to_remote_time_difference;
} else {
local_to_remote_time_jitters =
local_to_remote_time_jitters + conn->local_to_remote_time_difference - l2rtd;
ji = -(conn->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);
conn->local_to_remote_time_difference = l2rtd;
if (first_local_to_remote_time_difference == 0) {
first_local_to_remote_time_difference = conn->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;
double clock_drift_ppm = 0.0;
if (first_local_time == 0) {
first_local_time = local_time_chosen;
first_remote_time = remote_time_chosen;
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);
clock_drift_ppm = (1.0 * clock_drift_in_usec) / (local_time_change >> 32);
}
int64_t source_drift_usec;
if (conn->play_segment_reference_frame != 0) {
int64_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, conn);
uint64_t frame_difference = 0;
if (reference_timestamp >= conn->play_segment_reference_frame)
frame_difference =
(uint64_t)reference_timestamp - (uint64_t)conn->play_segment_reference_frame;
else // rollover
frame_difference = (uint64_t)reference_timestamp + 0x100000000 -
(uint64_t)conn->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 -
conn->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:
// debug(1, "clock_drift_ppm %f\tchosen %5d\tsource_drift_usec %10.1lld\treturn_time_in_usec
// %10.1llu",
// clock_drift_ppm,
// chosen,
//(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(3, "Timing thread interrupted. terminating.");
conn->timing_sender_stop = 1;
void *retval;
pthread_kill(timer_requester, SIGUSR1);
debug(3, "Wait for timer requester to exit.");
pthread_join(timer_requester, &retval);
debug(3, "Closed and terminated timer requester thread.");
debug(3, "Timing RTP thread terminated.");
close(conn->timing_socket);
return NULL;
}
static int bind_port(int ip_family, const char *self_ip_address, uint32_t scope_id, int *sock) {
// look for a port in the range, if any was specified.
int desired_port = config.udp_port_base;
int ret;
int local_socket = socket(ip_family, SOCK_DGRAM, IPPROTO_UDP);
if (local_socket == -1)
die("Could not allocate a socket.");
SOCKADDR myaddr;
do {
memset(&myaddr, 0, sizeof(myaddr));
if (ip_family == AF_INET) {
struct sockaddr_in *sa = (struct sockaddr_in *)&myaddr;
sa->sin_family = AF_INET;
sa->sin_port = ntohs(desired_port);
inet_pton(AF_INET, self_ip_address, &(sa->sin_addr));
ret = bind(local_socket, (struct sockaddr *)sa, sizeof(struct sockaddr_in));
}
#ifdef AF_INET6
if (ip_family == AF_INET6) {
struct sockaddr_in6 *sa6 = (struct sockaddr_in6 *)&myaddr;
sa6->sin6_family = AF_INET6;
sa6->sin6_port = ntohs(desired_port);
inet_pton(AF_INET6, self_ip_address, &(sa6->sin6_addr));
sa6->sin6_scope_id = scope_id;
ret = bind(local_socket, (struct sockaddr *)sa6, sizeof(struct sockaddr_in6));
}
#endif
} 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) {
close(local_socket);
die("error: could not bind a UDP port! Check the udp_port_range is large enough (>= 10) or "
"check for restrictive firewall settings or a bad router!");
}
int sport;
SOCKADDR local;
socklen_t local_len = sizeof(local);
getsockname(local_socket, (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);
}
fcntl(local_socket, F_SETFL, O_NONBLOCK);
*sock = local_socket;
return sport;
}
void rtp_setup(SOCKADDR *local, SOCKADDR *remote, int cport, int tport, int *lsport, int *lcport,
int *ltport, rtsp_conn_info *conn) {
// this gets the local and remote ip numbers (and ports used for the TCD stuff)
// we use the local stuff to specify the address we are coming from and
// we use the remote stuff to specify where we're goint to
if (conn->rtp_running)
die("rtp_setup called with active stream!");
debug(2, "rtp_setup: cport=%d tport=%d.", cport, tport);
// print out what we know about the client
void *client_addr, *self_addr;
int client_port, self_port;
char client_port_str[64];
char self_addr_str[64];
conn->connection_ip_family =
remote->SAFAMILY; // keep information about the kind of ip of the client
#ifdef AF_INET6
if (conn->connection_ip_family == AF_INET6) {
struct sockaddr_in6 *sa6 = (struct sockaddr_in6 *)remote;
client_addr = &(sa6->sin6_addr);
client_port = ntohs(sa6->sin6_port);
sa6 = (struct sockaddr_in6 *)local;
self_addr = &(sa6->sin6_addr);
self_port = ntohs(sa6->sin6_port);
conn->self_scope_id = sa6->sin6_scope_id;
}
#endif
if (conn->connection_ip_family == AF_INET) {
struct sockaddr_in *sa4 = (struct sockaddr_in *)remote;
client_addr = &(sa4->sin_addr);
client_port = ntohs(sa4->sin_port);
sa4 = (struct sockaddr_in *)local;
self_addr = &(sa4->sin_addr);
self_port = ntohs(sa4->sin_port);
}
inet_ntop(conn->connection_ip_family, client_addr, conn->client_ip_string,
sizeof(conn->client_ip_string));
inet_ntop(conn->connection_ip_family, self_addr, conn->self_ip_string,
sizeof(conn->self_ip_string));
debug(1, "Set up play connection from %s to self at %s on RTSP conversation thread %d.",
conn->client_ip_string, conn->self_ip_string, conn->connection_number);
// set up a the record of the remote's control socket
struct addrinfo hints;
struct addrinfo *servinfo;
memset(&conn->rtp_client_control_socket, 0, sizeof(conn->rtp_client_control_socket));
memset(&hints, 0, sizeof hints);
hints.ai_family = conn->connection_ip_family;
hints.ai_socktype = SOCK_DGRAM;
char portstr[20];
snprintf(portstr, 20, "%d", cport);
if (getaddrinfo(conn->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(&conn->rtp_client_control_socket, servinfo->ai_addr, sizeof(struct sockaddr_in6));
// ensure the scope id matches that of remote. this is needed for link-local addresses.
struct sockaddr_in6 *sa6 = (struct sockaddr_in6 *)&conn->rtp_client_control_socket;
sa6->sin6_scope_id = conn->self_scope_id;
} else
#endif
memcpy(&conn->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(&conn->rtp_client_timing_socket, 0, sizeof(conn->rtp_client_timing_socket));
memset(&hints, 0, sizeof hints);
hints.ai_family = conn->connection_ip_family;
hints.ai_socktype = SOCK_DGRAM;
snprintf(portstr, 20, "%d", tport);
if (getaddrinfo(conn->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(&conn->rtp_client_timing_socket, servinfo->ai_addr, sizeof(struct sockaddr_in6));
// ensure the scope id matches that of remote. this is needed for link-local addresses.
struct sockaddr_in6 *sa6 = (struct sockaddr_in6 *)&conn->rtp_client_timing_socket;
sa6->sin6_scope_id = conn->self_scope_id;
} else
#endif
memcpy(&conn->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(conn->connection_ip_family, conn->self_ip_string, conn->self_scope_id,
&conn->audio_socket);
*lcport = bind_port(conn->connection_ip_family, conn->self_ip_string, conn->self_scope_id,
&conn->control_socket);
*ltport = bind_port(conn->connection_ip_family, conn->self_ip_string, conn->self_scope_id,
&conn->timing_socket);
debug(2, "listening for audio, control and timing on ports %d, %d, %d.", *lsport, *lcport,
*ltport);
conn->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);
conn->request_sent = 0;
conn->rtp_running = 1;
}
void get_reference_timestamp_stuff(int64_t *timestamp, uint64_t *timestamp_time,
uint64_t *remote_timestamp_time, rtsp_conn_info *conn) {
// types okay
pthread_mutex_lock(&conn->reference_time_mutex);
*timestamp = conn->reference_timestamp;
*timestamp_time = conn->reference_timestamp_time;
// if ((*timestamp == 0) && (*timestamp_time == 0)) {
// debug(1,"Reference timestamp is invalid.");
//}
*remote_timestamp_time = conn->remote_reference_timestamp_time;
pthread_mutex_unlock(&conn->reference_time_mutex);
}
void clear_reference_timestamp(rtsp_conn_info *conn) {
pthread_mutex_lock(&conn->reference_time_mutex);
conn->reference_timestamp = 0;
conn->reference_timestamp_time = 0;
pthread_mutex_unlock(&conn->reference_time_mutex);
}
void rtp_request_resend(seq_t first, uint32_t count, rtsp_conn_info *conn) {
if (conn->rtp_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] = (char)0x55 | (char)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 (conn->rtp_client_control_socket.SAFAMILY == AF_INET6) {
msgsize = sizeof(struct sockaddr_in6);
}
#endif
if (sendto(conn->audio_socket, req, sizeof(req), 0,
(struct sockaddr *)&conn->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;
//}
}
}
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