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Re: "passive" vs "active"

Posted by Presto (A) on November 21, 2006 at 17:15:03

In Reply to: "passive" vs "active" posted by Christine Tham on November 21, 2006 at 13:50:38:

CT "However, filtering prior to the amp is not necessarily an improvement, since it's not the complexity of the input signal that challenges an amp, it's the complexity of the output load."

Presto > Correct. But when doing active filtering before amplification, you are reducing the effective bandwidth that each amp will see because you are reducing the bandwidth of the voltages for which it can will provide gain. When multi-amping with filters post-amp, each amp still sees the NET impedance of both the driver in the pass-band, the crossover in the stop-band and a combination of both in the crossover region. When filtering "pre-amp", there are only voltages present for the pass-band, which reduce proportionately through the crossover region(s) to zero voltage in the stop band. So, a big difference is that for active (pre-amp) filtering systems, the amps do NOT see crossover impedances in the stop band region(s), since there is simply no voltage AT those frequencies being amplified.

CT "The reason I don't like digital filtering (and DSPs in general) is that in my experience they seem to add a veil or a "haze" to the sound."

Presto > I think digital/active filters are much much more transparent than their op-amp based analog/active predecessors. Passive networks are not necessarily 100% transparent either. In fact, some of "deadest" sounding speakers are ones with elaborate passive networks that have every impedance compensation and EQ network under the sun and a component count like a laptop computer, and poor transparency and dynamics as a result. They measure ruler flat, but sound totally lifeless. The trick is to find DSP algorithms that do what "they are supposed to do" but also afford the listener the greatest possible transparency. Certain types of "digital mathmatics" seem to suck more life out of music than others, and even similar processes (like filtering or convolution) are not all created equal. Truth be told, I was about to "hang up my experimentation hat" with DSP filters and just stick to convolution/room correction stuff until I heard what the Thuneau Allocator can do. Al Jordon here at the computer audio asylum is getting similar results: the Allocator is the single most transparent DSP based crossover program we have encountered, and to top it all off, it's IIR based and performes "phase arbitration" for any filter within it's usable range.

There are also many cases where crossover frequencies for the low end are just not practical to do with passive components - this is why we see so so many crossover points in three way designs that are 300-500 hertz - right smack in the lower registers of the vocal range. Going active can often preclude the need for physically large and expensive passive components for lower crossover frequencies - meaning that the designer is less inclined to make a "design trade-off" due to prohibitive cost, size, or insertion losses associated with certain size inductors.

CT: "I even listen to Dolby Digital 2.0 soundtracks without applying Pro Logic II/IIx (for concerts and anything that's not surround encoded, they sound better that way)."

Presto > I thought I was the only one! ;)

CT: "And I've given up on equalization, convolution, bass management, upsampling."

Presto > I'm far too interested in that stuff to give up on it as a hobby. There are days I wish I had only a turntable and an integrated amp so I can put all of the technology away and just listen to music. But I am convinced that one of these days, all of the tweaking and experimenting will slowly turn into longer and longer listening sessions as DSP filters and room correction (AND my ability to implement them) get better and better. I'm really not a big fan of upsampling. I simply play all digital files back in their native resolution and bitdepth. I find that upsampling (on the PC anyways) is just a waste of computer resources that could be used for convolution or filtering! :D One thing people miss is that if you tax your system too much, you ARE getting degradation of SQ long before you are actually hearing dropouts or clicks and pops. So, for sure as you "Add DSP" into your signal chain you could be hearing lower SQ. But this does not mean there are additive "losses" (like how noise is additive in an analog signal chain). It just means one needs sufficient processing power to all of the DSP one wants to do.

CT "I know, it's heresy on this forum, but hey, at the end of the day it's the music that matters."

Presto > True - but in my mind, taking a digital master from a soundstage that was digital down to the mic mixers and pressing it into a vinyl disc that slowly wears out is just insanity. No less insane than digitizing a pristine analog recording anyways! For those of us who believe that digital can eventually come out of its infancy and sound really good, I think we're almost there - and some are there already. For those who believe it will never ever work, well, for them it will never ever work!

Cheers,
Presto

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