Provided by: jacktrip_1.3.0+ds0-1_amd64 
NAME
JackTrip - high-quality system for audio network performances
SYNOPSIS
jacktrip [-s|-c|-S|-C hostIPAddressOrURL] [options]
DESCRIPTION
JackTrip: A System for High-Quality Audio Network Performance over the Internet Copyright
© 2008-2020 Juan-Pablo Caceres, Chris Chafe. SoundWIRE group at CCRMA, Stanford
University VERSION: 1.3.0
OPTIONS
REQUIRED ARGUMENTS: One of:
-s, --server
Run in P2P Server Mode
-c, --client <peer_hostname_or_IP_num>
Run in P2P Client Mode
-S, --jacktripserver
Run in Hub Server Mode
-C, --pingtoserver <peer_name_or_IP>
Run in Hub Client Mode
OPTIONAL ARGUMENTS:
-n, --numchannels #
Number of Input and Output Channels (default: 2)
-q, --queue
# (2 or more) Queue Buffer Length, in Packet Size (default: 4)
-r, --redundancy
# (1 or more) Packet Redundancy to avoid glitches with packet losses
(default: 1)
-o, --portoffset
# Receiving bind port and peer port offset from default 4464
-B, --bindport
# Set only the bind port number (default: 4464)
-P, --peerport
# Set only the peer port number (default: 4464)
-U, --udpbaseport
Set only the server udp base port number (default: 61002)
-b, --bitres
# (8, 16, 24, 32) Audio Bit Rate Resolutions (default: 16, 32 uses
floating-point)
-p, --hubpatch
# (0, 1, 2, 3, 4, 5) Hub auto audio patch, only has effect if running HUB SERVER
mode, 0=server-to-clients, 1=client loopback, 2=client fan out/in but not loopback,
3=reserved for TUB, 4=full mix, 5=no auto patching (default: 0)
-z, --zerounderrun
Set buffer to zeros when underrun occurs (default: wavetable)
-t, --timeout
Quit after 10 seconds of no network activity
-l, --loopback
Run in Loop-Back Mode
-j, --jamlink
Run in JamLink Mode (Connect to a JamLink Box)
-J, --clientname
Change default client name (default: JackTrip)
-K, --remotename
Change default remote client name when connecting to a hub server (the default is
derived from this computer's external facing IP address)
-L, --localaddress
Change default local host IP address (default: 127.0.0.1)
-D, --nojackportsconnect
Don't connect default audio ports in jack
--bufstrategy
# (0, 1, 2) Use alternative jitter buffer
--broadcast <broadcast_queue>
Turn on broadcast output ports with extra queue (requires new jitter buffer)
--udprt
Use RT thread priority for network I/O
OPTIONAL SIGNAL PROCESSING:
-f, --effects # | paramString | help
Turn on incoming and/or outgoing compressor and/or reverb in Client - see `-f help'
for details
-O, --overflowlimiting i|o[w]|io[w]|n|help
Use audio limiter(s) in Client, i=incoming from network, o=outgoing to network,
io=both, n=no limiters, w=warn if limiting (default=n). Say -O help for more.
-a, --assumednumclients help|# (1,2,...) Assumed number of Clients (sources) mixing at Hub
Server (otherwise 2 assumed by -O)
ARGUMENTS TO USE JACKTRIP WITHOUT JACK:
-R, --rtaudio
Use system's default sound system instead of Jack
-T, --srate
# Set the sampling rate, works on --rtaudio mode only (default:
48000)
-F, --bufsize
# Set the buffer size, works on --rtaudio mode only (default:
128)
-d, --deviceid
# The rtaudio device id --rtaudio mode only (default: 0)
ARGUMENTS TO DISPLAY IO STATISTICS:
-I, --iostat <time_in_secs>
Turn on IO stat reporting with specified interval (in seconds)
-G, --iostatlog <log_file>
Save stat log into a file (default: print in stdout)
-x, --examine-audio-delay <print_interval_in_secs> | help
Print round-trip audio delay statistics. See `-x help' for details.
ARGUMENTS TO SIMULATE NETWORK ISSUES:
--simloss <rate>
Simulate packet loss
--simjitter <rate>,<d>
Simulate jitter, d is max delay in packets
HELP ARGUMENTS:
-v, --version
Prints Version Number
-V, --verbose
Verbose mode, prints debug messages
-h, --help
Prints this Help
COPYRIGHT
Copyright © 2008-2020 Juan-Pablo Caceres, Chris Chafe. SoundWIRE group at CCRMA, Stanford
University