jack_netsource(1) [debian man page]

JACK_NETSOURCE(1)					      General Commands Manual						 JACK_NETSOURCE(1)

NAME

       jack_netsource - Netjack Master client for one slave

SYNOPSIS

       jack_netsource [ -H hostname ] [ options ]

DESCRIPTION

       jack_netsource The Master side of a netjack connection. Represents the slave jackd -dnet in the master jack graph.  Most connection parame-
       ters are configured via the netsource, and the slave will set itself up according to the commandline option given to jack_netsource.
       Netjack allows low latency audio connections over general IP networks. When using celt for compression, it is even  possible  to  establish
       transatlantic links, with latencies not much over the actual ping time.
       But the main usecase is of course a LAN, where it can achieve one jack period of latency.

OPTIONS

       -h this help text

       -H slave host
	      Host name of the slave JACK

       -o num channels
	      Number of audio playback channels

       -i num channels
	      Number of audio capture channels

       -O num channels
	      Number of midi playback channels

       -I num channels
	      Number of midi capture channels

       -n periods
	      Network latency in JACK periods

       -p port
	      UDP port that the slave is listening on

       -r reply port
	      UDP port that we are listening on

       -B bind port
	      reply port, for use in NAT environments

       -b bitdepth
	      Set transport to use 16bit or 8bit

       -c bytes
	      Use CELT encoding with <bytes> per period and channel

       -m mtu
	      Assume this mtu for the link

       -R N
	      Redundancy: send out packets N times.

       -e
	      skip host-to-network endianness conversion

       -N jack name
	      Reports a different client name to jack

       -s, --server servername
	      Connect to the jack server named servername

       -h, --help
	      Display help/usage message

       -v, --version
	      Output version information and exit

EXAMPLES

       run a 4 audio channel bidirectional link with one period of latency and no midi channels. Audio data is flowing uncompressed over the wire:
       On hostA:

	      jackd -d alsa
	      jack_netsource -H hostB -n1 -i4 -o4 -I0 -O0

       On hostB:

	      jackd -d net

0-122-0 							  September 2012						 JACK_NETSOURCE(1)

ALSA_IO(1)						      General Commands Manual							ALSA_IO(1)

NAME

       alsa_in, alsa_out - Jack clients that perform I/O with an alternate audio interface

SYNOPSIS

       alsa_in [options]
       alsa_out [options]

DESCRIPTION

       A  JACK	client	that opens a specified audio interface (different to the one used by the JACK server, if any) and moves audio data between
       its JACK ports and the interface. alsa_in will provide data from the interface (potentially for capture); alsa_out will deliver data to	it
       (for playback).

       The  audio  interface  used  by	alsa_in/alsa_out  does	not need to be synchronized with JACK backend (or the hardware it might be using).
       alsa_in/alsa_out tries to resample the output stream in an attempt to compensate for drift between the two clocks.

       As of jack-0.116.3 this works almost perfectly. It takes some time, to reach absolute resample-rate stability. So give it some minutes (its
       intended to be running permanently anyways)

OPTIONS

       -j  jack_client_name
	      Set Client Name.

       -d  alsa_device
	      Use this Soundcard.

       -v
	      Verbose, prints out resample coefficient and other parameters useful for debugging, every 500ms.	also reports soft xruns.

       -i
	      Instrumentation. This logs the 4 important parameters of the samplerate control algorithm every 1ms.  You can pipe this into a file,
	      and plot it. Should only be necessary, if it does not work as expected, and we need to adjust some of  the  obscure  parameters,	to
	      make it work.  Find me on irc.freenode.org #jack in order to set this up correctly.

       -c  channels
	      Set Number of channels.

       -r  sample_rate
	      Set  sample_rate.  The program resamples as necessary.  So you can connect a 44k1 jackd to a soundcard only supporting 48k. (default
	      is jack sample_rate)

       -p  period_size
	      Set the period size. It is not related to the jackd period_size.	Sometimes it affects the quality of the delay measurements.   Set-
	      ting this lower than the jackd period_size will only work, if you use a higher number of periods.

       -n  num_period
	      Set number of periods. See note for period_size.

       -q  quality
	      Set the quality of the resampler from 0 to 4. This can significanly reduce CPU usage. Higher values give better quality and more CPU
	      usage.

       -m  max_diff
	      The value when a soft xrun occurs. Basically the window, in which the dma pointer may jitter. I don't think its  necessary  to  play
	      with this anymore.

       -t  target_delay
	      The delay alsa_io should try to approach. Same as for max_diff. It will be setup based on -p and -n which is generally sufficient.

       -s  smooth_array_size
	      This parameter controls the size of the array used for smoothing the delay measurement. Its default is 256.  If you use a pretty low
	      period size, you can lower the CPU usage a bit by decreasing this parameter.  However most CPU time is spent in  the  resampling	so
	      this will not be much.

       -C  P Control Clamp
	      If  you  have a PCI card, then the default value (15) of this parameter is too high for -p64 -n2... Setting it to 5 should fix that.
	      Be aware that setting this parameter too low, lets the hf noise on the delay measurement come through onto the resamplerate, so this
	      might  degrade the quality of the output. (but its a threshold value, and it has been chosen, to mask the noise of a USB card, which
	      has an amplitude which is 50 times higher than that of a PCI card, so 5 wont loose you any quality on a PCI card)

AUTHOR

       Torben Hohn

0-122-0 							  September 2012							ALSA_IO(1)