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JACKTRIP(1) User Commands JACKTRIP(1)

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 © 2008-2021 Juan-Pablo Caceres, Chris Chafe. SoundWIRE group at CCRMA, Stanford University This build of JackTrip is subject to LGPL license. JackTrip source code is released under MIT and GPL licenses. See LICENSE.md file for more information. VERSION: 1.4.1

OPTIONS

REQUIRED ARGUMENTS: One of:

Run in P2P Server Mode
Run in P2P Client Mode
Run in Hub Server Mode
Run in Hub Client Mode

OPTIONAL ARGUMENTS:

Number of Input and Output Channels (# greater than 0, default: 2)
Number of receive Channels from the network (# greater than 0)
Number of send Channels to the network (# greater than 0)
Queue Buffer Length, in Packet Size (default: 4)
Packet Redundancy to avoid glitches with packet losses (default: 1)
Receiving bind port and peer port offset from default 4464
Set only the bind port number (default: 4464)
Set only the peer port number (default: 4464)
Set only the server udp base port number (default: 61002)
Audio Bit Rate Resolutions (default: 16, 32 uses floating-point)
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)
Set buffer to zeros when underrun occurs (default: wavetable)
Quit after 10 seconds of no network activity
Run in Loop-Back Mode
Run in JamLink Mode (Connect to a JamLink Box)
Change default client name (default: JackTrip)
Change default remote client name when connecting to a hub server (the default is derived from this computer's external facing IP address)
Append thread ID to client names
Change default local host IP address (default: 127.0.0.1)
Don't connect default audio ports in jack
Use alternative jitter buffer
Duplicate receive ports with the specified broadcast_queue length. Broadcast outputs have higher latency but less packet loss.
Use RT thread priority for network I/O

OPTIONAL SIGNAL PROCESSING:

Turn on incoming and/or outgoing compressor and/or reverb in Client - see `-f help' for details
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.
Assumed number of Clients (sources) mixing at Hub Server (otherwise 2 assumed by -O)

ARGUMENTS TO DISPLAY IO STATISTICS:

Turn on IO stat reporting with specified interval (in seconds)
Save stat log into a file (default: print in stdout)
Print round-trip audio delay statistics. See `-x help' for details.

ARGUMENTS TO SIMULATE NETWORK ISSUES:

Simulate packet loss
Simulate jitter, d is max delay in packets

ARGUMENTS FOR HUB CLIENT/SERVER AUTHENTICATION:

Use authentication on the client side, or require it on the server side
The certificate file to use on the hub server
The private key file to use on the hub server
The file containing the stored usernames and passwords
The username to use when connecting as a hub client (if not supplied here, this is read from standard input)
The password to use when connecting as a hub client (if not supplied here, this is read from standard input)

HELP ARGUMENTS:

Prints Version Number
Verbose mode, prints debug messages
Prints this Help

COPYRIGHT

Copyright © 2008-2021 Juan-Pablo Caceres, Chris Chafe. SoundWIRE group at CCRMA, Stanford University This build of JackTrip is subject to LGPL license. JackTrip source code is released under MIT and GPL licenses. See LICENSE.md file for more information.

November 2021 JackTrip VERSION: 1.4.1