Provided by: zita-ajbridge_0.6.0-1_amd64 
NAME
zita-a2j, zita-j2a - Use ALSA device as a Jack client, with resampling.
SYNOPSIS
zita-a2j [options]
zita-j2a [options]
DESCRIPTION
These two programs allow one to use an ALSA device as a Jack client, to provide additional capture (zita-
a2j) or playback (zita-j2a) channels. Functionally they are equivalent to alsa_in and alsa_out that come
with Jack, but they will provide much better audio quality.
The resampling ratio will typically be stable within a few parts per million and change only very
smoothly. Delay will be stable as well even under worse case conditions, e.g. the Jack client running
near the end of the cycle. There should be no perceptible loss of quality at all. From version 0.6.0 it
is also possible to disable resampling if the device is word-clock synced.
The ALSA device should be a 'hw:' one, i.e. direct access to a soundcard and not an ALSA 'plug' device. A
well-working Jack system is assumed, running in real-time mode.
The sample rate can be the same as Jack's one, or different. Minimum delay is obtained by running the
alsa device at a lower period size than Jack. This can be done safely as the alsa thread will run at a
higher priority, and apart from copying to an internal buffer no work is done there. There are no
restrictions on the product of period_size and number_of_periods as there are for alsa_in and alsa_out.
Resampling is performed using the zita-resampler library. The -Q option sets the resampling quality. The
parameter value is half the length (i.e. the delay) of the multiphase filter used for resampling,
expressed in samples at the lower of the two sample rates. This mainly affects the shape of the frequency
response near the Nyquist frequency. It is normally not necessary to use this option as an optimum value
depending both the sample rates is chosen automatically. The available range is 16..96.
The -S option disables resampling. This requires that the device is synced via word-clock to the one used
by Jack.
The -L option forces the ALSA interface to 2 channels and 16-bit sample format. This may be required when
a2j or j2a is used with ALSA's 'loop' device, if the other end doesn't accept more than 2 channels or
floating point samples. This option must not be used otherwise.
Both programs will optionally print some information four times per second. The first number is the
average loop error over the last quarter second, in samples. It should be reduced to small randowm values
close to zero after 15 seconds or so. The second is the dynamic correction factor of the nominal
resampling ratio. This should converge to a value close to one and not move much.
Small variations in these numbers will be seen when Jack apps are started or stopped. This is normal.
Anything else isn't - please report.
When starting, and in case of major trouble, the 'Starting synchronisation' message will be printed. A
restart can happen if there is a timeout on the Jack server, e.g. when a client crashed or terminated in
a dirty way.
Jack1 will skip one or more cycles when new apps are started, or when a large number of port connections
is done in a short time. This may interrupt the audio signal, but should otherwise not have any ill
consequences nor require a restart.
Both zita-a2j and zita-j2a will suspend operation while Jack is in 'freewheeling' mode.
OPTIONS
-h Print command line and options summary.
-j <name>
Jack client name [zita-a2j or zita-j2a].
-d <device>
ALSA device [none].
-r <rate>
Sample rate [48000].
-p <period>
Period size [256].
-n <nfrags>
Number of fragments [2].
-c <nchannels>
Number of channels [2].
-Q <quality>
Resampling quality [auto].
-S Word clock sync, disable resampling.
-I <samples> [0]
(zita-a2j only) Latency correction. The value is added to the latency property set on the Jack
ports.
-O <samples> [0]
(zita-j2a only) Latency correction. The value is added to the latency property set on the Jack
ports.
-L Force 2 channels and 16-bit sample format.
-v Print tracing information.
AUTHOR
The zita-a2j and zita-j2a programs were written by Fons Adriaensen <fons@linuxaudio.org>. This manual page was written by Robin Gareus <robin@gareus.org> and Fons Adriaensen. Sept 2013 ZITA-AJBRIDGE(1)