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shairport-sync/rtp.c
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Mike Brady 54cb9de183 Handle active_remote_id as a string rather than an unsigned 32-bit number -- seems ROON uses a longer character sequence. 
Tidy up some error messages about which backend is selected.
    Tighten up memory allocatons in DACP.c.
    Add a default name for the pipe backend: /tmp/shairport-sync-audio
    Ensure the metadata pipe is created, if necessary, when the --with-metadata option is chosen without dbus or mpris
    Make the first output backend in the list of backends the default and make its name the default output_name. Clang-format everything
    Add the word 'jack' to the version string if Jack Audio support is compiled in.
    Fix a flaw in resyncing -- a flush is set up but not triggered.
    Open metadata and audio pipes with 666 permissions.
    Ensure metadata and cover art are enabled if metadata support is included at compilation.
    Reword some misleading error messages.
    Set convolution defaults even if no configuration file is found.
2020-09-30 15:02:35 +01:00

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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>
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;
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->reference_timestamp = 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) {
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 % 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(seqno, actual_timestamp, pktp, plen, conn);
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 {
debug(1, "Error receiving an audio packet.");
}
}
/*
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(3, "Control Receiver Cleanup Done.");
}
void *rtp_control_receiver(void *arg) {
pthread_cleanup_push(rtp_control_handler_cleanup_handler, arg);
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;
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(3, "Latency derived just 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(3, "A fixed latency offset of %d frames has been added, giving a latency of
// "
// "%" PRId64
// " frames with flags: %d and AirPlay version %d (triggers if 353 or
// less).",
// config.fixedLatencyOffset, la, flags, conn->AirPlayVersion);
}
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 {
if (la != conn->latency) {
conn->latency = la;
debug(3,
"New latency detected: %" PRIu32 ", sync latency: %" PRIu32
", minimum latency: %" PRIu32 ", maximum "
"latency: %" PRIu32 ", fixed offset: %" PRIu32 ".",
la, sync_rtp_timestamp - rtp_timestamp_less_latency, conn->minimum_latency,
conn->maximum_latency, config.fixedLatencyOffset);
}
}
}
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->remote_reference_timestamp_time -
conn->initial_reference_time; // here, this should never be zero
if (remote_frame_time_interval) {
conn->remote_frame_rate =
(1.0E9 * (conn->reference_timestamp - 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->remote_reference_timestamp_time;
uint32_t old_reference_timestamp = conn->reference_timestamp;
// int64_t old_latency_delayed_timestamp = conn->latency_delayed_timestamp;
conn->remote_reference_timestamp_time = remote_time_of_sync;
// conn->reference_timestamp_time =
// remote_time_of_sync - local_to_remote_time_difference_now(conn);
conn->reference_timestamp = 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(seqno, actual_timestamp, pktp, plen, conn);
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 {
debug(1, "Control Receiver -- error receiving a packet.");
}
}
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) {
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) {
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);
// 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(3, "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);
}
static uint16_t 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 ret = 0;
int local_socket = socket(ip_family, SOCK_DGRAM, IPPROTO_UDP);
if (local_socket == -1)
die("Could not allocate a socket.");
/*
int val = 1;
ret = setsockopt(local_socket, SOL_SOCKET, SO_REUSEADDR, &val, sizeof(val));
if (ret < 0) {
char errorstring[1024];
strerror_r(errno, (char *)errorstring, sizeof(errorstring));
debug(1, "Error %d: \"%s\". Couldn't set SO_REUSEADDR");
}
*/
SOCKADDR myaddr;
int tryCount = 0;
uint16_t desired_port;
do {
tryCount++;
desired_port = nextFreeUDPPort();
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) &&
(tryCount < config.udp_port_range));
// debug(1,"UDP port chosen: %d.",desired_port);
if (ret < 0) {
close(local_socket);
char errorstring[1024];
strerror_r(errno, (char *)errorstring, sizeof(errorstring));
die("error %d: \"%s\". Could not bind a UDP port! Check the udp_port_range is large enough -- "
"it must be "
"at least 3, and 10 or more is suggested -- or "
"check for restrictive firewall settings or a bad router! UDP base is %u, range is %u and "
"current suggestion is %u.",
errno, errorstring, config.udp_port_base, config.udp_port_range, desired_port);
}
uint16_t 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, 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,
sizeof(conn->self_ip_string));
debug(2, "Connection %d: SETUP -- Connection from %s to self at %s.", conn->connection_number,
conn->client_ip_string, conn->self_ip_string);
// 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
conn->remote_control_port = cport;
conn->remote_timing_port = tport;
conn->local_control_port = bind_port(conn->connection_ip_family, conn->self_ip_string,
conn->self_scope_id, &conn->control_socket);
conn->local_timing_port = bind_port(conn->connection_ip_family, conn->self_ip_string,
conn->self_scope_id, &conn->timing_socket);
conn->local_audio_port = bind_port(conn->connection_ip_family, conn->self_ip_string,
conn->self_scope_id, &conn->audio_socket);
debug(3, "listening for audio, control and timing on ports %d, %d, %d.", conn->local_audio_port,
conn->local_control_port, conn->local_timing_port);
conn->reference_timestamp = 0;
conn->request_sent = 0;
conn->rtp_running = 1;
#ifdef CONFIG_METADATA
send_ssnc_metadata('clip', conn->client_ip_string, strlen(conn->client_ip_string), 1);
send_ssnc_metadata('svip', conn->self_ip_string, strlen(conn->self_ip_string), 1);
#endif
}
}
void get_reference_timestamp_stuff(uint32_t *timestamp, uint64_t *timestamp_time,
uint64_t *remote_timestamp_time, rtsp_conn_info *conn) {
// types okay
debug_mutex_lock(&conn->reference_time_mutex, 1000, 0);
*timestamp = conn->reference_timestamp;
*remote_timestamp_time = conn->remote_reference_timestamp_time;
*timestamp_time =
conn->remote_reference_timestamp_time - local_to_remote_time_difference_now(conn);
debug_mutex_unlock(&conn->reference_time_mutex, 0);
}
void clear_reference_timestamp(rtsp_conn_info *conn) {
debug_mutex_lock(&conn->reference_time_mutex, 1000, 1);
conn->reference_timestamp = 0;
conn->remote_reference_timestamp_time = 0;
debug_mutex_unlock(&conn->reference_time_mutex, 3);
}
int have_timestamp_timing_information(rtsp_conn_info *conn) {
if (conn->reference_timestamp == 0)
return 0;
else
return 1;
}
// set this to zero to use the rates supplied by the sources, which might not always be completely
// right...
const int use_nominal_rate = 0; // specify whether to use the nominal input rate, usually 44100 fps
int sanitised_source_rate_information(uint32_t *frames, uint64_t *time, rtsp_conn_info *conn) {
int result = 1;
uint32_t fs = conn->input_rate;
*frames = fs; // default value to return
*time = 1000000000; // default value to return
if ((conn->initial_reference_time) && (conn->initial_reference_timestamp)) {
// uint32_t local_frames = conn->reference_timestamp - conn->initial_reference_timestamp;
uint32_t local_frames =
modulo_32_offset(conn->initial_reference_timestamp, conn->reference_timestamp);
uint64_t local_time = conn->remote_reference_timestamp_time - conn->initial_reference_time;
if ((local_frames == 0) || (local_time == 0) || (use_nominal_rate)) {
result = 1;
} else {
double calculated_frame_rate = conn->input_rate;
if (local_time)
calculated_frame_rate = (1.0E9 * local_frames) / local_time;
else
debug(1, "sanitised_source_rate_information: local_time is zero");
if ((local_time == 0) || ((calculated_frame_rate / conn->input_rate) > 1.002) ||
((calculated_frame_rate / conn->input_rate) < 0.998)) {
debug(3, "input frame rate out of bounds at %.2f fps.", calculated_frame_rate);
result = 1;
} else {
*frames = local_frames;
*time = local_time;
result = 0;
}
}
}
return result;
}
// the timestamp is a timestamp calculated at the input rate
// the reference timestamps are denominated in terms of the input rate
int frame_to_local_time(uint32_t timestamp, uint64_t *time, rtsp_conn_info *conn) {
debug_mutex_lock(&conn->reference_time_mutex, 1000, 0);
int result = 0;
uint64_t time_difference;
uint32_t frame_difference;
result = sanitised_source_rate_information(&frame_difference, &time_difference, conn);
uint64_t timestamp_interval_time;
uint64_t remote_time_of_timestamp;
uint32_t timestamp_interval = modulo_32_offset(conn->reference_timestamp, timestamp);
if (timestamp_interval <=
conn->input_rate * 3600) { // i.e. timestamp was really after the reference timestamp
timestamp_interval_time = (timestamp_interval * time_difference) /
frame_difference; // this is the nominal time, based on the
// fps specified between current and
// previous sync frame.
remote_time_of_timestamp = conn->remote_reference_timestamp_time +
timestamp_interval_time; // based on the reference timestamp time
// plus the time interval calculated based
// on the specified fps.
} else { // i.e. timestamp was actually before the reference timestamp
timestamp_interval =
modulo_32_offset(timestamp, conn->reference_timestamp); // fix the calculation
timestamp_interval_time = (timestamp_interval * time_difference) /
frame_difference; // this is the nominal time, based on the
// fps specified between current and
// previous sync frame.
remote_time_of_timestamp = conn->remote_reference_timestamp_time -
timestamp_interval_time; // based on the reference timestamp time
// plus the time interval calculated based
// on the specified fps.
}
*time = remote_time_of_timestamp - local_to_remote_time_difference_now(conn);
debug_mutex_unlock(&conn->reference_time_mutex, 0);
return result;
}
int local_time_to_frame(uint64_t time, uint32_t *frame, rtsp_conn_info *conn) {
debug_mutex_lock(&conn->reference_time_mutex, 1000, 0);
int result = 0;
uint64_t time_difference;
uint32_t frame_difference;
result = sanitised_source_rate_information(&frame_difference, &time_difference, conn);
// first, get from [local] time to remote time.
uint64_t remote_time = time + local_to_remote_time_difference_now(conn);
// next, get the remote time interval from the remote_time to the reference time
uint64_t time_interval;
// here, we calculate the time interval, in terms of remote time
uint64_t offset = modulo_64_offset(conn->remote_reference_timestamp_time, remote_time);
int reference_time_was_earlier = (offset <= (uint64_t)3600000000000);
if (reference_time_was_earlier) // if we haven't had a reference within the last hour, it'll be
// taken as afterwards
time_interval = remote_time - conn->remote_reference_timestamp_time;
else
time_interval = conn->remote_reference_timestamp_time - remote_time;
// now, convert the remote time interval into frames using the frame rate we have observed or
// which has been nominated
uint32_t frame_interval = 0;
if (time_difference)
frame_interval = (time_interval * frame_difference) / time_difference;
else
debug(1, "local_time_to_frame: time_difference is zero");
if (reference_time_was_earlier) {
// debug(1,"Frame interval is %" PRId64 " frames.",frame_interval);
*frame = (conn->reference_timestamp + frame_interval);
} else {
// debug(1,"Frame interval is %" PRId64 " frames.",-frame_interval);
*frame = (conn->reference_timestamp - frame_interval);
}
debug_mutex_unlock(&conn->reference_time_mutex, 0);
return result;
}
void rtp_request_resend(seq_t first, uint32_t count, rtsp_conn_info *conn) {
if (conn->rtp_running) {
// if (!request_sent) {
// debug(2, "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 sequence number
*(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
uint64_t time_of_sending_ns = get_absolute_time_in_ns();
uint64_t resend_error_backoff_time = 300000000; // 0.3 seconds
if ((conn->rtp_time_of_last_resend_request_error_ns == 0) ||
((time_of_sending_ns - conn->rtp_time_of_last_resend_request_error_ns) >
resend_error_backoff_time)) {
if ((config.diagnostic_drop_packet_fraction == 0.0) ||
(drand48() > config.diagnostic_drop_packet_fraction)) {
// put a time limit on the sendto
struct timeval timeout;
timeout.tv_sec = 0;
timeout.tv_usec = 100000;
if (setsockopt(conn->control_socket, SOL_SOCKET, SO_SNDTIMEO, (char *)&timeout,
sizeof(timeout)) < 0)
debug(1, "Can't set timeout on resend request socket.");
if (sendto(conn->control_socket, req, sizeof(req), 0,
(struct sockaddr *)&conn->rtp_client_control_socket, msgsize) == -1) {
char em[1024];
strerror_r(errno, em, sizeof(em));
debug(2, "Error %d using sendto to request a resend: \"%s\".", errno, em);
conn->rtp_time_of_last_resend_request_error_ns = time_of_sending_ns;
} else {
conn->rtp_time_of_last_resend_request_error_ns = 0;
}
} else {
debug(3, "Dropping resend request packet to simulate a bad network. Backing off for 0.3 "
"second.");
conn->rtp_time_of_last_resend_request_error_ns = time_of_sending_ns;
}
} else {
debug(1,
"Suppressing a resend request due to a resend sendto error in the last 0.3 seconds.");
}
} else {
// if (!request_sent) {
debug(2, "rtp_request_resend called without active stream!");
// request_sent = 1;
//}
}
}
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