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Old 10th November 2012, 04:07 PM   #761
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precisely. Time is all that is saved. The temptation in active filtration, for me, is the possibility to use class A amps for the majority of audible bandwidth leaving a class B for helper woofer. That is the route to signal purity in my book. Switching amps just arent viable in the same way. My experience of such amps is that their only advantage is efficiency, and sonically they stink. For the price of a 'good' class D/H i may aswell go analogue active with class A and live with the electricity bill.
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Old 10th November 2012, 04:26 PM   #762
dewardh is offline dewardh  United States
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Originally Posted by AndrewT View Post
How much signal can a 6dB/octave remove from the signal being handled by the bass/mid amplifier?
Why the limit to 6dB/octave? It is easy to build higher order passive filters at line level . . . the inductors required are conveniently small and easy to hand wind.
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Old 10th November 2012, 04:44 PM   #763
dewardh is offline dewardh  United States
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Originally Posted by mondogenerator View Post
The temptation in active filtration, for me, is the possibility to use class A amps
The ability to use (and match) appropriate amps to individual drivers is indeed one of the many advantages of "active" crossovers.
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Old 12th November 2012, 02:04 PM   #764
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Originally Posted by Barleywater View Post
high loudness studio monitors = deaf engineer = poor mix.
The forumers deserve to read better arguments than that.

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Originally Posted by Barleywater View Post
These are post 1980 drivers. 4%-10% 2nd harmonic is not natural or reasonable for high fidelity, and whatever a "pure amplifier" is, it's not going to fix this.
What is mechanism for "pure" amplifiers elimination of driver distortion?

Regards,

Andrew
Andrew, have you taken notice of the SPL levels quoted?
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Old 12th November 2012, 02:11 PM   #765
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Originally Posted by john k... View Post
Being that I am a user of both and a proponent of neither let me pose a hypothetical. The task is to design a midrange band pass filter either active of passive. One person takes the passive approach, another the active. Both achieve the same, identical bandpass response. So, which is superior and why?.
This is not a good comparison because what we want to optimize is the system, which includes at least two drivers and one (or more) amplifiers.
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Old 12th November 2012, 02:13 PM   #766
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Originally Posted by mondogenerator View Post
The temptation in active filtration, for me, is the possibility to use class A amps for the majority of audible bandwidth leaving a class B for helper woofer. That is the route to signal purity in my book..
Agree.
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Old 12th November 2012, 03:28 PM   #767
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Originally Posted by flavio81 View Post
The forumers deserve to read better arguments than that.



Andrew, have you taken notice of the SPL levels quoted?
Yes, I have. Compression driver horn sound, distortion and efficiency is engineering compromise accepted for easy access to hearing impairment inducing SPL.

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Originally Posted by krivium View Post
Far too simplified situation. I don't want to sound harsh but work in studio is divided into phase and each phase as differents needs relative to monitoring and you don't seem to understand this requirements.

In recording phase you need a playback system able to reproduce 'real' dynamic of the instrument recorded, and in case of drum in big control room highly dead (RT60 <0,3seconds with sweet spot 3/4 meters distance from loudspeakers ) high efficiency is the only way imho. Recording through small monitors (NS10 or Genelec or Adam or...you name it) give numerous artifacts and big problems during mixing with as a result = poor mix, because sound of album is made during takes, not really at mixing stage (because crap in=crap out).

For mixing situation is different as the action is relative to levels and equilibrium between instruments or group of instruments, fx, etc,etc... and small nearfield monitors well known by enginneers can give great results at moderate reproduction levels. But in this case big mains monitors are used to check work regularly ( not for high SPL but mainly for bandwith and soundstage) same rooms used as for recording with same acoustical characteristics.

For mastering you need systems as transparent as possible and without any compression artifacts on any peaks, with ability to discern very subtle corrections (eq or compressions) in great acoustics (not as dead as recording studios, more live and closer to 'livingroom' acoustics) . No nearfield allowed, mainly midfield reproduction and only high quality systems used (Dunlavy, B&W, Pmc, Atc, Klinger-Favre, Tannoy, Quested, Kinoshita/Rey Audio,...) with moderate levels of approximately 85db spl (with 20db peaks allowed: at max 105db spl peak at sweet spot, search for K-system on google).....
In tracking phase, recording engineer's job is to capture clean signal, because "crap in = crap out". Monitoring at excessive SPL is not necessary for assessing microphone placement issues. It is producer/artists determination if captured recording has usable sound; to that end if they need deafening levels to make their determination, so be it.

For many types of music, mixing phase is major point at which the desired sound is created often requiring long hours, where high SPL is fatiguing, and destructive to hearing.

Sound stage performance doesn't require high SPL either.

Mastering engineer has a nice job.

Andrew
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Old 12th November 2012, 05:25 PM   #768
krivium is offline krivium  France
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Originally Posted by Barleywater View Post
Monitoring at excessive SPL is not necessary for assessing microphone placement issues. It is producer/artists determination if captured recording has usable sound; to that end if they need deafening levels to make their determination, so be it.
Andrew,
essentially i agree with what you said, but for relatively short period of time* true level monitoring is nescessary for assessing microphone placement and for aesthetical reason too :even if technical, choices of microphone couple or microphone themself define aesthetic results. And to be sure of choices made and behavior regarding peak distortion (dynamic), imaging, 'color' you need to listem to original instrument and compare with reproduced sound at equivalent level.

* as you work with your ears you can't be exposed too long at high pressure as sessions usually last 10 to 12 hours a day and repeated nearly each day a week... Most of professionnals are now aware and take seriously this concerns about deafning levels... but it wasn't nessecerely true 20/30 years ago, and as Fletcher and Munson stand the louder the better (the more linear our ears we should say)...so let's say 10minutes every couple of hours is usual and relatively 'safe'. Most of the time this day, when clients want to be pushed against diffusors of rear wall by spl enginneers leaves the control room...

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For many types of music, mixing phase is major point at which the desired sound is created often requiring long hours, where high SPL is fatiguing, and destructive to hearing.
I wasn't precise enough: for acoustic music tracking is where the sound is made. Including rock, pop, variety,... As you said mixing is an important and major phase aesthetically, but being lucky enough to had access of original tape multitracks of major artists/bands (from the 80's mainly) i can swear you only need to put faders to unity gain to have the 'sound' of the albums... without the refinements added by mixers and mastering engineers however.
We agree about SPl, and this is one of reasons nearfield monitoring took place in studios.

Quote:
Mastering engineer has a nice job.
About listening conditions i fully agree!
By far the most impressive listening conditions i've heard was in mastering rooms (mainly the couple system/room's acoustic responsible for that). For the rest it's a job... with enjoyments and pain in the *** !

Last edited by krivium; 12th November 2012 at 05:29 PM. Reason: precision added.
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Old 12th November 2012, 11:08 PM   #769
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Originally Posted by john k... View Post
Being that I am a user of both and a proponent of neither let me pose a hypothetical. The task is to design a midrange band pass filter either active of passive. One person takes the passive approach, another the active. Both achieve the same, identical bandpass response. So, which is superior and why?
What is the significance of the steep crossover slopes that are possible with active? Are many of the 'black arts' of passive crossover design geared towards circumventing the problems of shallow crossover slopes? (i.e. drivers working outside their range, mis-aligned phase of drivers throughout the crossover region, acoustic lobing effects of drivers combining over a broad crossover region?)

With steep crossovers many of these problems are reduced, and the only issue seems to be the ringing of the filters - which at least one report on the web suggests is not a problem until you have incredibly sharp filters, which is also how it seems to me. As Barleywater suggests, crossover design then becomes a scientific, methodical procedure involving measurements and automatic computation of the perfect filter - in fact it's more than mere crossover design, and becomes a fine correction for the deficiencies of each driver. Are we really just talking about the difference between passive and active crossovers, or also the fine tuning of the drivers individually?

My own experiments (gradually heading towards this ideal) suggest to me that the automatically corrected and crossed-over speaker, far from sounding flat, neutral (in a bad way) or 'grey' as I might have imagined, sounds magnificently colourful (in a good way) and fills the room with sound.

I'm comparing my two way activated Mission speakers (digital filters modified in response to microphone measurements) with some, on paper superior, conventional three way Tannoy speakers. If I've been playing the actives for a while and swap to the passives, the passives seem more immediate and 'raw' with music like this:
1979 Semi-Finalist by The Bad Plus on Spotify

At moderate volumes the passives don't sound compressed at all with music like this, but just the opposite: notes seem shorter and the recording seems dry, reverb-wise. Swapping to the actives, the sound is smoother and the effect of the recording room's ambience seems greater. If anything, I prefer the passives, initially at least. I'm guessing that the brighter spectrum of the attack of the notes lines up with the peaks in the response of these passives, while the spectrum of the note decays and reverb are suppressed. The Tannoys are very good at emphasising the rasp of horse hair on catgut, that sort of thing, but I think it's just an artificial effect, like EQ.

If I change to larger-scale orchestral music like this Shostakovich: Symphony No.5 in D minor, Op.47 - 4. Allegro non troppo by Shostakovich, Dmitri, Royal Philharmonic Orchestra, Vladimir Ashkenazy and Vladimir Ashkenazy on Spotify, the Tannoys fall apart, sounding like two boxes under stress, and quite tiring to listen to. Swapping to the actives (with measured, corrected frequency response and 12th order crossovers) as I have just done while writing this, the transformation is extraordinary, and the room is filled with the sound of a real orchestra in a concert hall. It's quite amazing, and you immedately know that it's 'right'. The bass percussion towards the end of the piece is stunning.

I can't say that it's passive vs. active, as such, that accounts for this difference, but getting the actives to this state of rightness didn't involve any black arts or tweaking in response to listening, nor a long apprenticeship; merely measurements and automatic software (and a ready-made box and driver selection, of course - but they were rubbish with the original passive crossovers). I think I'm almost at the stage of being perfectly happy with my system, and it cost almost nothing!

Last edited by CopperTop; 12th November 2012 at 11:11 PM.
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Old 14th November 2012, 04:53 AM   #770
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^^ That about sums it up for me too.
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