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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
* 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 <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"
typedef struct {
uint32_t seconds;
uint32_t fraction;
} ntp_timestamp;
typedef struct time_ping_record {
uint64_t local_to_remote_difference;
uint64_t dispersion;
} time_ping_record;
// only one RTP session can be active at a time.
static int running = 0;
static int please_shutdown;
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;
// 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
static void *rtp_audio_receiver(void *arg) {
// we inherit the signal mask (SIGUSR1)
int32_t last_seqno = -1;
uint8_t packet[2048], *pktp;
ssize_t nread;
while (1) {
if (please_shutdown)
break;
nread = recv(audio_socket, packet, sizeof(packet), 0);
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(2,"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;
}
static void *rtp_control_receiver(void *arg) {
// we inherit the signal mask (SIGUSR1)
reference_timestamp=0; // nothing valid received yet
uint8_t packet[2048];
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 (1) {
if (please_shutdown)
break;
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 (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);
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
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;
}
static void *rtp_timing_sender(void *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);
time_ping_count = 0;
// we inherit the signal mask (SIGUSR1)
while (1) {
if (please_shutdown)
break;
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;
}
static void *rtp_timing_receiver(void *arg) {
// we inherit the signal mask (SIGUSR1)
uint8_t packet[2048], *pktp;
ssize_t nread;
pthread_t timer_requester;
pthread_create(&timer_requester, NULL, &rtp_timing_sender, NULL);
// 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 l2rtd=0;
while (1) {
if (please_shutdown)
break;
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
}
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++;
// 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;
}
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;
} else {
debug(1, "Timing port -- Unknown RTP packet of type 0x%02X length %d.", packet[1], nread);
}
}
debug(1, "Timing RTP thread interrupted. terminating.");
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, int desired_port ) {
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];
snprintf(buffer, 10, "%d", desired_port);
int 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);
ret = bind(*sock, info->ai_addr, info->ai_addrlen);
freeaddrinfo(info);
if (ret < 0)
die("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, 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);
// print out what we know about the client
void *addr;
char *ipver;
int port;
char ipstr[INET6_ADDRSTRLEN];
char portstr[20];
#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,ipstr,sizeof(ipstr));
debug(1,"Connection from %s: %s:%d",ipver,ipstr,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(ipstr,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(ipstr,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,0);
*lcport = bind_port(remote,&control_socket,0);
*ltport = bind_port(remote,&timing_socket,0);
debug(2, "listening for audio, control and timing on ports %d, %d, %d.", *lsport, *lcport, *ltport);
please_shutdown = 0;
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) {
pthread_mutex_lock(&reference_time_mutex);
*timestamp=reference_timestamp;
*timestamp_time = 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)
die("rtp_shutdown called without active stream!");
debug(2, "shutting down RTP thread");
please_shutdown = 1;
void *retval;
reference_timestamp=0;
pthread_kill(rtp_audio_thread, SIGUSR1);
pthread_join(rtp_audio_thread, &retval);
pthread_kill(rtp_control_thread, SIGUSR1);
pthread_join(rtp_control_thread, &retval);
pthread_kill(rtp_timing_thread, SIGUSR1);
pthread_join(rtp_timing_thread, &retval);
running = 0;
}
void rtp_request_resend(seq_t first, uint32_t count) {
if (running) {
if (!request_sent) {
debug(2, "requesting resend on %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_timing_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;
}
}
}
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