jacktrip - Man Page

-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

-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

-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)

-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)

-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.

--simloss <rate>

Simulate packet loss

--simjitter <rate>,<d>

Simulate jitter, d is max delay in packets

-v,  --version

Prints Version Number

-V,  --verbose

Verbose mode, prints debug messages

-h,  --help

Prints this Help