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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 -- 2019
* 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 "rtp.h"
#include "common.h"
#include "player.h"
#include "rtsp.h"
#include <arpa/inet.h>
#include <errno.h>
#include <fcntl.h>
#include <inttypes.h>
#include <math.h>
#include <memory.h>
#include <netdb.h>
#include <netinet/in.h>
#include <pthread.h>
#include <stdio.h>
#include <stdlib.h>
#include <sys/socket.h>
#include <sys/types.h>
#include <time.h>
#include <unistd.h>
#ifdef CONFIG_AIRPLAY_2
#include "ptp-utilities.h"
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
#include <libavutil/opt.h>
#include <libswresample/swresample.h>
#include <sodium.h>
#endif
struct Nvll {
char *name;
double value;
struct Nvll *next;
};
typedef struct Nvll nvll;
uint64_t local_to_remote_time_jitter;
uint64_t local_to_remote_time_jitter_count;
typedef struct {
int closed;
int error_code;
int sock_fd;
char *buffer;
char *toq;
char *eoq;
size_t buffer_max_size;
size_t buffer_occupancy;
pthread_mutex_t mutex;
pthread_cond_t not_empty_cv;
pthread_cond_t not_full_cv;
} buffered_tcp_desc;
void check64conversion(const char *prompt, const uint8_t *source, uint64_t value) {
char converted_value[128];
sprintf(converted_value, "%" PRIx64 "", value);
char obf[32];
char *obfp = obf;
int obfc;
int suppress_zeroes = 1;
for (obfc = 0; obfc < 8; obfc++) {
if ((suppress_zeroes == 0) || (source[obfc] != 0)) {
if (suppress_zeroes != 0) {
if (source[obfc] < 0x10) {
snprintf(obfp, 3, "%1x", source[obfc]);
obfp += 1;
} else {
snprintf(obfp, 3, "%02x", source[obfc]);
obfp += 2;
}
} else {
snprintf(obfp, 3, "%02x", source[obfc]);
obfp += 2;
}
suppress_zeroes = 0;
}
};
*obfp = 0;
if (strcmp(converted_value, obf) != 0) {
debug(1, "%s check64conversion error converting \"%s\" to %" PRIx64 ".", prompt, obf, value);
}
}
void check32conversion(const char *prompt, const uint8_t *source, uint32_t value) {
char converted_value[128];
sprintf(converted_value, "%" PRIx32 "", value);
char obf[32];
char *obfp = obf;
int obfc;
int suppress_zeroes = 1;
for (obfc = 0; obfc < 4; obfc++) {
if ((suppress_zeroes == 0) || (source[obfc] != 0)) {
if (suppress_zeroes != 0) {
if (source[obfc] < 0x10) {
snprintf(obfp, 3, "%1x", source[obfc]);
obfp += 1;
} else {
snprintf(obfp, 3, "%02x", source[obfc]);
obfp += 2;
}
} else {
snprintf(obfp, 3, "%02x", source[obfc]);
obfp += 2;
}
suppress_zeroes = 0;
}
};
*obfp = 0;
if (strcmp(converted_value, obf) != 0) {
debug(1, "%s check32conversion error converting \"%s\" to %" PRIx32 ".", prompt, obf, value);
}
}
void rtp_initialise(rtsp_conn_info *conn) {
conn->rtp_time_of_last_resend_request_error_ns = 0;
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) {
conn->anchor_rtptime = 0;
// destroy the timer mutex
int rc = pthread_mutex_destroy(&conn->reference_time_mutex);
if (rc)
debug(1, "Error destroying reference_time_mutex variable.");
}
uint64_t local_to_remote_time_difference_now(rtsp_conn_info *conn) {
// this is an attempt to compensate for clock drift since the last time ping that was used
// so, if we have a non-zero clock drift, we will calculate the drift there would
// be from the time of the last time ping
uint64_t time_since_last_local_to_remote_time_difference_measurement =
get_absolute_time_in_ns() - conn->local_to_remote_time_difference_measurement_time;
uint64_t result = conn->local_to_remote_time_difference;
if (conn->local_to_remote_time_gradient >= 1.0) {
result = conn->local_to_remote_time_difference +
(uint64_t)((conn->local_to_remote_time_gradient - 1.0) *
time_since_last_local_to_remote_time_difference_measurement);
} else {
result = conn->local_to_remote_time_difference -
(uint64_t)((1.0 - conn->local_to_remote_time_gradient) *
time_since_last_local_to_remote_time_difference_measurement);
}
return result;
}
void rtp_audio_receiver_cleanup_handler(__attribute__((unused)) void *arg) {
debug(3, "Audio Receiver Cleanup Done.");
}
void *rtp_audio_receiver(void *arg) {
debug(3, "rtp_audio_receiver start");
pthread_cleanup_push(rtp_audio_receiver_cleanup_handler, arg);
rtsp_conn_info *conn = (rtsp_conn_info *)arg;
int32_t last_seqno = -1;
uint8_t packet[2048], *pktp;
uint64_t time_of_previous_packet_ns = 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;
int frame_count = 0;
ssize_t nread;
while (1) {
nread = recv(conn->audio_socket, packet, sizeof(packet), 0);
frame_count++;
uint64_t local_time_now_ns = get_absolute_time_in_ns();
if (time_of_previous_packet_ns) {
float time_interval_us = (local_time_now_ns - time_of_previous_packet_ns) * 0.001;
time_of_previous_packet_ns = local_time_now_ns;
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 != 1) && (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_ns = 0;
longest_packet_time_interval_us = 0.0;
}
} else {
time_of_previous_packet_ns = local_time_now_ns;
}
if (nread >= 0) {
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(*(uint16_t *)(pktp + 2));
// increment last_seqno and see if it's the same as the incoming seqno
if (type == 0x60) { // regular audio data
/*
char obf[4096];
char *obfp = obf;
int obfc;
for (obfc=0;obfc<plen;obfc++) {
snprintf(obfp, 3, "%02X", pktp[obfc]);
obfp+=2;
};
*obfp=0;
debug(1,"Audio Packet Received: \"%s\"",obf);
*/
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...
}
} else {
debug(3, "Audio Receiver -- Retransmitted Audio Data Packet %u received.", seqno);
}
uint32_t actual_timestamp = ntohl(*(uint32_t *)(pktp + 4));
// uint32_t ssid = ntohl(*(uint32_t *)(pktp + 8));
// debug(1, "Audio packet SSID: %08X,%u", ssid,ssid);
// 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) {
if ((config.diagnostic_drop_packet_fraction == 0.0) ||
(drand48() > config.diagnostic_drop_packet_fraction))
player_put_packet(1, seqno, actual_timestamp, pktp, plen,
conn); // the '1' means is original format
else
debug(3, "Dropping audio packet %u to simulate a bad connection.", seqno);
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);
} else {
char em[1024];
strerror_r(errno, em, sizeof(em));
debug(1, "Error %d receiving an audio packet: \"%s\".", errno, em);
}
}
/*
debug(3, "Audio receiver -- Server RTP thread interrupted. terminating.");
close(conn->audio_socket);
*/
debug(1, "Audio receiver thread \"normal\" exit -- this can't happen. Hah!");
pthread_cleanup_pop(0); // don't execute anything here.
debug(2, "Audio receiver thread exit.");
pthread_exit(NULL);
}
void rtp_control_handler_cleanup_handler(__attribute__((unused)) void *arg) {
debug(2, "Control Receiver Cleanup Done.");
}
void *rtp_control_receiver(void *arg) {
debug(2, "rtp_control_receiver start");
pthread_cleanup_push(rtp_control_handler_cleanup_handler, arg);
rtsp_conn_info *conn = (rtsp_conn_info *)arg;
conn->anchor_rtptime = 0; // nothing valid received yet
uint8_t packet[2048], *pktp;
// struct timespec tn;
uint64_t remote_time_of_sync;
uint32_t sync_rtp_timestamp;
ssize_t nread;
while (1) {
nread = recv(conn->control_socket, packet, sizeof(packet), 0);
if (nread >= 0) {
if ((config.diagnostic_drop_packet_fraction == 0.0) ||
(drand48() > config.diagnostic_drop_packet_fraction)) {
ssize_t plen = nread;
if (packet[1] == 0xd4) { // sync data
/*
// the following stanza is for debugging only -- normally commented out.
{
char obf[4096];
char *obfp = obf;
int obfc;
for (obfc = 0; obfc < plen; obfc++) {
snprintf(obfp, 3, "%02X", packet[obfc]);
obfp += 2;
};
*obfp = 0;
// get raw timestamp information
// I think that a good way to understand these timestamps is that
// (1) the rtlt below is the timestamp of the frame that should be playing at the
// client-time specified in the packet if there was no delay
// and (2) that the rt below is the timestamp of the frame that should be playing
// at the client-time specified in the packet on this device taking account of
// the delay
// Thus, (3) the latency can be calculated by subtracting the second from the
// first.
// There must be more to it -- there something missing.
// In addition, it seems that if the value of the short represented by the second
// pair of bytes in the packet is 7
// then an extra time lag is expected to be added, presumably by
// the AirPort Express.
// Best guess is that this delay is 11,025 frames.
uint32_t rtlt = nctohl(&packet[4]); // raw timestamp less latency
uint32_t rt = nctohl(&packet[16]); // raw timestamp
uint32_t fl = nctohs(&packet[2]); //
debug(1,"Sync Packet of %d bytes received: \"%s\", flags: %d, timestamps %u and %u,
giving a latency of %d frames.",plen,obf,fl,rt,rtlt,rt-rtlt);
//debug(1,"Monotonic timestamps are: %" PRId64 " and %" PRId64 "
respectively.",monotonic_timestamp(rt, conn),monotonic_timestamp(rtlt, conn));
}
*/
if (conn->local_to_remote_time_difference) { // need a time packet to be interchanged
// first...
uint64_t ps, pn;
ps = nctohl(&packet[8]);
ps = ps * 1000000000; // this many nanoseconds from the whole seconds
pn = nctohl(&packet[12]);
pn = pn * 1000000000;
pn = pn >> 32; // this many nanoseconds from the fractional part
remote_time_of_sync = ps + pn;
// debug(1,"Remote Sync Time: " PRIu64 "",remote_time_of_sync);
sync_rtp_timestamp = nctohl(&packet[16]);
uint32_t rtp_timestamp_less_latency = nctohl(&packet[4]);
// debug(1,"Sync timestamp is %u.",ntohl(*((uint32_t *)&packet[16])));
if (config.userSuppliedLatency) {
if (config.userSuppliedLatency != conn->latency) {
debug(1, "Using the user-supplied latency: %" PRIu32 ".",
config.userSuppliedLatency);
}
conn->latency = config.userSuppliedLatency;
} else {
// It seems that the second pair of bytes in the packet indicate whether a fixed
// delay of 11,025 frames should be added -- iTunes set this field to 7 and
// AirPlay sets it to 4.
// However, on older versions of AirPlay, the 11,025 frames seem to be necessary too
// The value of 11,025 (0.25 seconds) is a guess based on the "Audio-Latency"
// parameter
// returned by an AE.
// Sigh, it would be nice to have a published protocol...
uint16_t flags = nctohs(&packet[2]);
uint32_t la = sync_rtp_timestamp - rtp_timestamp_less_latency; // note, this might
// loop around in
// modulo. Not sure if
// you'll get an error!
// debug(1, "Latency from the sync packet is %" PRIu32 " frames.", la);
if ((flags == 7) || ((conn->AirPlayVersion > 0) && (conn->AirPlayVersion <= 353)) ||
((conn->AirPlayVersion > 0) && (conn->AirPlayVersion >= 371))) {
la += config.fixedLatencyOffset;
// debug(1, "Latency offset by %" PRIu32" frames due to the source flags and version
// giving a latency of %" PRIu32 " frames.", config.fixedLatencyOffset, la);
}
if ((conn->maximum_latency) && (conn->maximum_latency < la))
la = conn->maximum_latency;
if ((conn->minimum_latency) && (conn->minimum_latency > la))
la = conn->minimum_latency;
const uint32_t max_frames = ((3 * BUFFER_FRAMES * 352) / 4) - 11025;
if (la > max_frames) {
warn("An out-of-range latency request of %" PRIu32
" frames was ignored. Must be %" PRIu32
" frames or less (44,100 frames per second). "
"Latency remains at %" PRIu32 " frames.",
la, max_frames, conn->latency);
} else {
// here we have the latency but it does not yet account for the
// audio_backend_latency_offset
int32_t latency_offset =
(int32_t)(config.audio_backend_latency_offset * conn->input_rate);
// debug(1,"latency offset is %" PRId32 ", input rate is %u", latency_offset,
// conn->input_rate);
int32_t adjusted_latency = latency_offset + (int32_t)la;
if ((adjusted_latency < 0) ||
(adjusted_latency >
(int32_t)(conn->max_frames_per_packet *
(BUFFER_FRAMES - config.minimum_free_buffer_headroom))))
warn("audio_backend_latency_offset out of range -- ignored.");
else
la = adjusted_latency;
if (la != conn->latency) {
conn->latency = la;
debug(2,
"New latency: %" PRIu32 ", sync latency: %" PRIu32
", minimum latency: %" PRIu32 ", maximum "
"latency: %" PRIu32 ", fixed offset: %" PRIu32
", audio_backend_latency_offset: %f.",
conn->latency, sync_rtp_timestamp - rtp_timestamp_less_latency,
conn->minimum_latency, conn->maximum_latency, config.fixedLatencyOffset,
config.audio_backend_latency_offset);
}
}
}
// here, we apply the latency to the sync_rtp_timestamp
sync_rtp_timestamp = sync_rtp_timestamp - conn->latency;
debug_mutex_lock(&conn->reference_time_mutex, 1000, 0);
if (conn->initial_reference_time == 0) {
if (conn->packet_count_since_flush > 0) {
conn->initial_reference_time = remote_time_of_sync;
conn->initial_reference_timestamp = sync_rtp_timestamp;
}
} else {
uint64_t remote_frame_time_interval =
conn->anchor_time -
conn->initial_reference_time; // here, this should never be zero
if (remote_frame_time_interval) {
conn->remote_frame_rate =
(1.0E9 * (conn->anchor_rtptime - conn->initial_reference_timestamp)) /
remote_frame_time_interval;
} else {
conn->remote_frame_rate = 0.0; // use as a flag.
}
}
// this is for debugging
uint64_t old_remote_reference_time = conn->anchor_time;
uint32_t old_reference_timestamp = conn->anchor_rtptime;
// int64_t old_latency_delayed_timestamp = conn->latency_delayed_timestamp;
conn->anchor_time = remote_time_of_sync;
// conn->reference_timestamp_time =
// remote_time_of_sync - local_to_remote_time_difference_now(conn);
conn->anchor_rtptime = sync_rtp_timestamp;
conn->latency_delayed_timestamp = rtp_timestamp_less_latency;
debug_mutex_unlock(&conn->reference_time_mutex, 0);
conn->reference_to_previous_time_difference =
remote_time_of_sync - old_remote_reference_time;
if (old_reference_timestamp == 0)
conn->reference_to_previous_frame_difference = 0;
else
conn->reference_to_previous_frame_difference =
sync_rtp_timestamp - old_reference_timestamp;
} else {
debug(2, "Sync packet received before we got a timing packet back.");
}
} else if (packet[1] == 0xd6) { // resent audio data in the control path -- whaale only?
pktp = packet + 4;
plen -= 4;
seq_t seqno = ntohs(*(uint16_t *)(pktp + 2));
debug(3, "Control Receiver -- Retransmitted Audio Data Packet %u received.", seqno);
uint32_t actual_timestamp = ntohl(*(uint32_t *)(pktp + 4));
pktp += 12;
plen -= 12;
// check if packet contains enough content to be reasonable
if (plen >= 16) {
player_put_packet(1, seqno, actual_timestamp, pktp, plen,
conn); // the '1' means is original format
continue;
} else {
debug(3, "Too-short retransmitted audio packet received in control port, ignored.");
}
} else
debug(1, "Control Receiver -- Unknown RTP packet of type 0x%02X length %d, ignored.",
packet[1], nread);
} else {
debug(3, "Control Receiver -- dropping a packet to simulate a bad network.");
}
} else {
char em[1024];
strerror_r(errno, em, sizeof(em));
debug(1, "Control Receiver -- error %d receiving a packet: \"%s\".", errno, em);
}
}
debug(1, "Control RTP thread \"normal\" exit -- this can't happen. Hah!");
pthread_cleanup_pop(0); // don't execute anything here.
debug(2, "Control RTP thread exit.");
pthread_exit(NULL);
}
void rtp_timing_sender_cleanup_handler(void *arg) {
rtsp_conn_info *conn = (rtsp_conn_info *)arg;
debug(3, "Connection %d: Timing Sender Cleanup.", conn->connection_number);
}
void *rtp_timing_sender(void *arg) {
debug(2, "rtp_timing_sender start");
pthread_cleanup_push(rtp_timing_sender_cleanup_handler, arg);
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;
while (1) {
// 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;
conn->departure_time = get_absolute_time_in_ns();
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
if ((config.diagnostic_drop_packet_fraction == 0.0) ||
(drand48() > config.diagnostic_drop_packet_fraction)) {
if (sendto(conn->timing_socket, &req, sizeof(req), 0,
(struct sockaddr *)&conn->rtp_client_timing_socket, msgsize) == -1) {
char em[1024];
strerror_r(errno, em, sizeof(em));
debug(1, "Error %d using send-to to the timing socket: \"%s\".", errno, em);
}
} else {
debug(3, "Timing Sender Thread -- dropping outgoing packet to simulate bad network.");
}
request_number++;
if (request_number <= 6)
usleep(300000); // these are thread cancellation points
else
usleep(3000000);
}
debug(3, "rtp_timing_sender thread interrupted. This should never happen.");
pthread_cleanup_pop(0); // don't execute anything here.
pthread_exit(NULL);
}
void rtp_timing_receiver_cleanup_handler(void *arg) {
rtsp_conn_info *conn = (rtsp_conn_info *)arg;
debug(3, "Timing Receiver Cleanup.");
// walk down the list of DACP / gradient pairs, if any
nvll *gradients = config.gradients;
if (conn->dacp_id)
while ((gradients) && (strcasecmp((const char *)&conn->client_ip_string, gradients->name) != 0))
gradients = gradients->next;
// if gradients comes out of this non-null, it is pointing to the DACP and it's last-known
// gradient
if (gradients) {
gradients->value = conn->local_to_remote_time_gradient;
// debug(1,"Updating a drift of %.2f ppm for \"%s\".", (conn->local_to_remote_time_gradient
// - 1.0)*1000000, gradients->name);
} else {
nvll *new_entry = (nvll *)malloc(sizeof(nvll));
if (new_entry) {
new_entry->name = strdup((const char *)&conn->client_ip_string);
new_entry->value = conn->local_to_remote_time_gradient;
new_entry->next = config.gradients;
config.gradients = new_entry;
// debug(1,"Setting a new drift of %.2f ppm for \"%s\".", (conn->local_to_remote_time_gradient
// - 1.0)*1000000, new_entry->name);
}
}
debug(3, "Cancel Timing Requester.");
pthread_cancel(conn->timer_requester);
int oldState;
pthread_setcancelstate(PTHREAD_CANCEL_DISABLE, &oldState);
debug(3, "Join Timing Requester.");
pthread_join(conn->timer_requester, NULL);
debug(3, "Timing Receiver Cleanup Successful.");
pthread_setcancelstate(oldState, NULL);
}
void *rtp_timing_receiver(void *arg) {
debug(3, "rtp_timing_receiver start");
pthread_cleanup_push(rtp_timing_receiver_cleanup_handler, arg);
rtsp_conn_info *conn = (rtsp_conn_info *)arg;
uint8_t packet[2048];
ssize_t nread;
pthread_create(&conn->timer_requester, NULL, &rtp_timing_sender, arg);
// struct timespec att;
uint64_t distant_receive_time, distant_transmit_time, arrival_time, return_time;
local_to_remote_time_jitter = 0;
local_to_remote_time_jitter_count = 0;
// uint64_t first_remote_time = 0;
// uint64_t first_local_time = 0;
uint64_t first_local_to_remote_time_difference = 0;
conn->local_to_remote_time_gradient = 1.0; // initial value.
// walk down the list of DACP / gradient pairs, if any
nvll *gradients = config.gradients;
while ((gradients) && (strcasecmp((const char *)&conn->client_ip_string, gradients->name) != 0))
gradients = gradients->next;
// if gradients comes out of this non-null, it is pointing to the IP and it's last-known gradient
if (gradients) {
conn->local_to_remote_time_gradient = gradients->value;
// debug(1,"Using a stored drift of %.2f ppm for \"%s\".", (conn->local_to_remote_time_gradient
// - 1.0)*1000000, gradients->name);
}
// calculate diffusion factor
// at the end of the array of time pings, the diffusion factor
// must be diffusion_expansion_factor
// this, at each step, the diffusion multiplication constant must
// be the nth root of diffusion_expansion_factor
// where n is the number of elements in the array
const double diffusion_expansion_factor = 10;
double log_of_multiplier = log10(diffusion_expansion_factor) / time_ping_history;
double multiplier = pow(10, log_of_multiplier);
uint64_t dispersion_factor = (uint64_t)(multiplier * 100);
if (dispersion_factor == 0)
die("dispersion factor is zero!");
// debug(1,"dispersion factor is %" PRIu64 ".", dispersion_factor);
// uint64_t first_local_to_remote_time_difference_time;
// uint64_t l2rtd = 0;
int sequence_number = 0;
// for getting mean and sd of return times
int32_t stat_n = 0;
double stat_mean = 0.0;
// double stat_M2 = 0.0;
while (1) {
nread = recv(conn->timing_socket, packet, sizeof(packet), 0);
if (nread >= 0) {
if ((config.diagnostic_drop_packet_fraction == 0.0) ||
(drand48() > config.diagnostic_drop_packet_fraction)) {
arrival_time = get_absolute_time_in_ns();
// ssize_t plen = nread;
// debug(1,"Packet Received on Timing Port.");
if (packet[1] == 0xd3) { // timing reply
return_time = arrival_time - conn->departure_time;
debug(2, "clock synchronisation request: return time is %8.3f milliseconds.",
0.000001 * return_time);
if (return_time < 200000000) { // must be less than 0.2 seconds
// distant_receive_time =
// ((uint64_t)ntohl(*((uint32_t*)&packet[16])))<<32+ntohl(*((uint32_t*)&packet[20]));
uint64_t ps, pn;
ps = nctohl(&packet[16]);
ps = ps * 1000000000; // this many nanoseconds from the whole seconds
pn = nctohl(&packet[20]);
pn = pn * 1000000000;
pn = pn >> 32; // this many nanoseconds from the fractional part
distant_receive_time = ps + pn;
// distant_transmit_time =
// ((uint64_t)ntohl(*((uint32_t*)&packet[24])))<<32+ntohl(*((uint32_t*)&packet[28]));
ps = nctohl(&packet[24]);
ps = ps * 1000000000; // this many nanoseconds from the whole seconds
pn = nctohl(&packet[28]);
pn = pn * 1000000000;
pn = pn >> 32; // this many nanoseconds from the fractional part
distant_transmit_time = ps + pn;
uint64_t remote_processing_time = 0;
if (distant_transmit_time >= distant_receive_time)
remote_processing_time = distant_transmit_time - distant_receive_time;
else {
debug(1, "Yikes: distant_transmit_time is before distant_receive_time; remote "
"processing time set to zero.");
}
// debug(1,"Return trip time: %" PRIu64 " nS, remote processing time: %" PRIu64 "
// nS.",return_time, remote_processing_time);
if (remote_processing_time < return_time)
return_time -= remote_processing_time;
else
debug(1, "Remote processing time greater than return time -- ignored.");
int cc;
// debug(1, "time ping history is %d entries.", time_ping_history);
for (cc = time_ping_history - 1; cc > 0; cc--) {
conn->time_pings[cc] = conn->time_pings[cc - 1];
// if ((conn->time_ping_count) && (conn->time_ping_count < 10))
// conn->time_pings[cc].dispersion =
// conn->time_pings[cc].dispersion * pow(2.14,
// 1.0/conn->time_ping_count);
if (conn->time_pings[cc].dispersion > UINT64_MAX / dispersion_factor)
debug(1, "dispersion factor is too large at %" PRIu64 ".");
else
conn->time_pings[cc].dispersion =
(conn->time_pings[cc].dispersion * dispersion_factor) /
100; // make the dispersions 'age' by this rational factor
}
// these are used for doing a least squares calculation to get the drift
conn->time_pings[0].local_time = arrival_time;
conn->time_pings[0].remote_time = distant_transmit_time + return_time / 2;
conn->time_pings[0].sequence_number = sequence_number++;
conn->time_pings[0].chosen = 0;
conn->time_pings[0].dispersion = return_time;
if (conn->time_ping_count < time_ping_history)
conn->time_ping_count++;
// here, calculate the mean and standard deviation of the return times
// mean and variance calculations from "online_variance" algorithm at
// https://en.wikipedia.org/wiki/Algorithms_for_calculating_variance#Online_algorithm
stat_n += 1;
double stat_delta = return_time - stat_mean;
stat_mean += stat_delta / stat_n;
// stat_M2 += stat_delta * (return_time - stat_mean);
// debug(1, "Timing packet return time stats: current, mean and standard deviation over
// %d packets: %.1f, %.1f, %.1f (nanoseconds).",
// stat_n,return_time,stat_mean, sqrtf(stat_M2 / (stat_n - 1)));
// here, pick the record with the least dispersion, and record that it's been chosen
// 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 rt = conn->time_pings[0].remote_time;
uint64_t lt = conn->time_pings[0].local_time;
uint64_t tld = conn->time_pings[0].dispersion;
int chosen = 0;
for (cc = 1; cc < conn->time_ping_count; cc++)
if (conn->time_pings[cc].dispersion < tld) {
chosen = cc;
rt = conn->time_pings[cc].remote_time;
lt = conn->time_pings[cc].local_time;
tld = conn->time_pings[cc].dispersion;
// local_time_chosen = conn->time_pings[cc].local_time;
// remote_time_chosen = conn->time_pings[cc].remote_time;
}
// debug(1,"Record %d has the lowest dispersion with %0.2f us
// dispersion.",chosen,1.0*((tld * 1000000) >> 32));
conn->time_pings[chosen].chosen = 1; // record the fact that it has been used for timing
conn->local_to_remote_time_difference =
rt - lt; // make this the new local-to-remote-time-difference
conn->local_to_remote_time_difference_measurement_time = lt; // done at this time.
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();
}
// here, let's try to use the timing pings that were selected because of their short
// return times to
// estimate a figure for drift between the local clock (x) and the remote clock (y)
// if we plug in a local interval, we will get back what that is in remote time
// calculate the line of best fit for relating the local time and the remote time
// we will calculate the slope, which is the drift
// see https://www.varsitytutors.com/hotmath/hotmath_help/topics/line-of-best-fit
uint64_t y_bar = 0; // remote timestamp average
uint64_t x_bar = 0; // local timestamp average
int sample_count = 0;
// approximate time in seconds to let the system settle down
const int settling_time = 60;
// number of points to have for calculating a valid drift
const int sample_point_minimum = 8;
for (cc = 0; cc < conn->time_ping_count; cc++)
if ((conn->time_pings[cc].chosen) &&
(conn->time_pings[cc].sequence_number >
(settling_time / 3))) { // wait for a approximate settling time
// have to scale them down so that the sum, possibly over
// every term in the array, doesn't overflow
y_bar += (conn->time_pings[cc].remote_time >> time_ping_history_power_of_two);
x_bar += (conn->time_pings[cc].local_time >> time_ping_history_power_of_two);
sample_count++;
}
conn->local_to_remote_time_gradient_sample_count = sample_count;
if (sample_count > sample_point_minimum) {
y_bar = y_bar / sample_count;
x_bar = x_bar / sample_count;
int64_t xid, yid;
double mtl, mbl;
mtl = 0;
mbl = 0;
for (cc = 0; cc < conn->time_ping_count; cc++)
if ((conn->time_pings[cc].chosen) &&
(conn->time_pings[cc].sequence_number > (settling_time / 3))) {
uint64_t slt = conn->time_pings[cc].local_time >> time_ping_history_power_of_two;
if (slt > x_bar)
xid = slt - x_bar;
else
xid = -(x_bar - slt);
uint64_t srt = conn->time_pings[cc].remote_time >> time_ping_history_power_of_two;
if (srt > y_bar)
yid = srt - y_bar;
else
yid = -(y_bar - srt);
mtl = mtl + (1.0 * xid) * yid;
mbl = mbl + (1.0 * xid) * xid;
}
if (mbl)
conn->local_to_remote_time_gradient = mtl / mbl;
else {
// conn->local_to_remote_time_gradient = 1.0;
debug(1, "mbl is zero. Drift remains at %.2f ppm.",
(conn->local_to_remote_time_gradient - 1.0) * 1000000);
}
// scale the numbers back up
uint64_t ybf = y_bar << time_ping_history_power_of_two;
uint64_t xbf = x_bar << time_ping_history_power_of_two;
conn->local_to_remote_time_difference =
ybf - xbf; // make this the new local-to-remote-time-difference
conn->local_to_remote_time_difference_measurement_time = xbf;
} else {
debug(3, "not enough samples to estimate drift -- remaining at %.2f ppm.",
(conn->local_to_remote_time_gradient - 1.0) * 1000000);
// conn->local_to_remote_time_gradient = 1.0;
}
// debug(1,"local to remote time gradient is %12.2f ppm, based on %d
// samples.",conn->local_to_remote_time_gradient*1000000,sample_count);
} else {
debug(1,
"Time ping turnaround time: %" PRIu64
" ns -- it looks like a timing ping was lost.",
return_time);
}
} else {
debug(1, "Timing port -- Unknown RTP packet of type 0x%02X length %d.", packet[1], nread);
}
} else {
debug(3, "Timing Receiver Thread -- dropping incoming packet to simulate a bad network.");
}
} else {
debug(1, "Timing receiver -- error receiving a packet.");
}
}
debug(1, "Timing Receiver RTP thread \"normal\" exit -- this can't happen. Hah!");
pthread_cleanup_pop(0); // don't execute anything here.
debug(2, "Timing Receiver RTP thread exit.");
pthread_exit(NULL);
}
void rtp_setup(SOCKADDR *local, SOCKADDR *remote, uint16_t cport, uint16_t tport,
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)
warn("rtp_setup has been called with al already-active stream -- ignored. Possible duplicate "
"SETUP call?");
else {
debug(3, "rtp_setup: cport=%d tport=%d.", cport, tport);
// print out what we know about the client
void *client_addr = NULL, *self_addr = NULL;
// 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,