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bbff2d1efb565ab5f578a4c7e715555d7bfcfcd3
shairport-sync/rtp.c
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Mike Brady bbff2d1efb Get IPv6 working properly
2016-05-02 13:58:17 +01:00

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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 <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 char self_ip_string[INET6_ADDRSTRLEN]; // the ip string being used by this program -- it could be one of many, so we need to know it
static short connection_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(int ip_family,const char *self_ip_address, 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);
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!");
}
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);
}
*sock = local_socket;
return sport;
}
void rtp_setup(SOCKADDR *local, SOCKADDR *remote, int cport, int tport, uint32_t active_remote, int *lsport,
int *lcport, int *ltport) {
// 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 (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 *client_addr,*self_addr;
int client_port,self_port;
char client_port_str[64];
char self_addr_str[64];
connection_ip_family = remote->SAFAMILY; // keep information about the kind of ip of the client
#ifdef AF_INET6
if (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);
}
#endif
if (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(connection_ip_family, client_addr, client_ip_string,
sizeof(client_ip_string));
inet_ntop(connection_ip_family, self_addr, self_ip_string,
sizeof(self_ip_string));
debug(1, "Set up play connection from %s to self at %s.", client_ip_string,self_ip_string);
// 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 = connection_ip_family;
hints.ai_socktype = SOCK_DGRAM;
char portstr[20];
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 = connection_ip_family;
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(connection_ip_family,self_ip_string,&audio_socket);
*lcport = bind_port(connection_ip_family,self_ip_string,&control_socket);
*ltport = bind_port(connection_ip_family,self_ip_string,&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.");
}
}
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