Little Labs IBP

[djgout]

can anyone explain how these awesome boxes work? Is it essentially a pan pot that is between the positive and negative leads? would it be difficult to build a box that does this?

 

[soundguy]

Its an all pass filter. Not necessarily hard to build, but its WAY more complicated than just a pot, its a completely active filter circuit. I just used on of those to correct a snare track, its great on bass, but I can really hear the slewing with a snare drum. Every studio should have one of those though.

dave

 

[PRR]

From the Operators Manual :

?? how does the ibp work ??

The IBP phase shift circuitry works by running through two passive symmetrical all pass filters. It works entirely in the analog domain. All pass filters are like high pass or lo pass filters as in your EQ or speaker crossover network, but only the phase is affected not amplitude. The IBP has relatively flat amplitude response well past 96kHz. In most but not all cases at least two all pass filters are needed to recreate what is necessary to correct a phase problem or emulate phase shift in nature. One filter is used for the bottom of the frequency spectrum. The other is used in series for the top, creating a somewhat linear phase vs. frequency response. This is because you need to create a phase curve that is approximately twice the phase shift when you double the frequency. So for 45 degrees of phase shift at 40 Hz you need approximately 90 degrees of Phase shift at 80 Hz for it to sound natural. I say approximately because every situation is different with sound bouncing around, EQ added (with corresponding phase shift), phase response of speakers, microphones etc. Phase curves in nature have many variables that determine the amount of phase shift vs. frequency.

The switch Phase adjust 90 degrees /180 degrees selects between one or two series all pass filters in circuit. Phase center lo/hi selects between a higher bandwidth phase adjust utilizing one all pass filter for lo and the second for hi frequencies or a lower bandwidth phase adjust which uses both all pass filters for the lo frequencies. The IBPs electronics are for buffering and gain make up only. The IBP is entirely balanced and symmetrical from input to output. The output transformer in the ibp allows you to plug freely into balanced or unbalanced circuitry without any interface problems and also allows you to lift the ground.

The DI section puts a very high impedance input circuit specifically designed for instrument pickups ahead of the phase adjust circuitry. This hi impedance presents no load on your pickups allowing you to hear the full tonality of your instrument. The gain of the DI section can be varied from 12 to 26 dB of gain. The buffered out put is tapped after the instrument input circuit, before the phase adjust circuit, and can be used independently, even when the IBP is used in line level mode.

The re-amp output will bring the +4dB db output to guitar level (about -14dB)and is tapped after the phase adjust circuitry and after the transformer. It is the same output as the line output but at a lower level and higher impedance. The re-amp output and line out share the same transformer, they are not intended to be used at the same time. The earth lift switch on the front panel lifts the ground of the line out xlr pin 1 and the sleeve of the re-amp output jack. If you are installing the ibp in a rack and will be extending the reamp output jack to a remote location use a stereo jack and plug with 3 conductor wiring. This will allow the earth lift switch to function correctly whether using the re-amp or line output.

> a pan pot that is between the positive and negative leads?

No, that just gives in-phase, weak, dead, weak, and out-of-phase.

> would it be difficult to build a box that does this?

The essential idea is clear. If I recall from days when we did FM and SSB modulation the hard way, it isn't trivial to assemble a full-range constant phase shift. In this case I'm not sure it needs to be constant across the audio band (some of that switching suggests an inexact approach), but it does have to be variable, which is difficult to do, especially with a passive all-pass filter, and especially at that price (allowing for all the other useful frills you get).

The price seems very fair for what it is: a complete box-solution for the musician.

If you wanted just the basic idea integrated into say a desk channel, a lot of those frills are not needed. But re-inventing filter parameters that work musically may be very hard. I suspect that designer went to a lot of recording sessions with a hay-wire prototype and a soldering iron, trimming and tweaking to get best effect.