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Author Topic: real sound lab Coneq ?  (Read 38662 times)

damon gold

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Re: real sound lab Coneq ?
« Reply #20 on: October 14, 2007, 01:41:19 PM »

NOT ONLY HAVE I CHECKD IT OUT, I HAVE USED 3 IN REAL LIFE!!!. YOU WOULD HAVE TO DRAG ME KICKING AND SCREAMING TO MIX AT A GIG THAT DID NOT USE ONE. i have three of the apeq2 units and use them on house or monitors (whichever the situation calls for) for monitors the results are astounding at venues that provide unmatched monitors it takes only a few minutes to get jbl caberets to sound so similar to l'acoustic 115XT's that only a ture Seasoned pro could distingush between the two. if you are touring and using house racke and stacks it saves hours of eq work. for anyone still using wedges dont leave home without at least 6 of these in your quiver. As for house what i have experienced is a more even (consistant) and intellageable mix, especially where i have multiple zones and center fill, i have not had a chance to try it on a line array in an outdoor venue yet but i have a VDOSC and an Adamson Y10/T21 system i will be working with for the next couple of months so i will post more as my experience with CONEQ grows. as for now i am a believer.
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Geri O'Neil

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Re: real sound lab Coneq ?
« Reply #21 on: October 14, 2007, 01:46:54 PM »



Oh, Lawd. You registered for this forum just to tell us this??

damon gold wrote on Sun, 14 October 2007 12:41

 YOU WOULD HAVE TO DRAG ME KICKING AND SCREAMING TO MIX AT A GIG THAT DID NOT USE ONE.


Note that several of us are actually capable of doing just that... Twisted Evil

Geri "adding nothing productive as usual" O


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Bob McCarthy

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Re: real sound lab Coneq ?
« Reply #22 on: October 14, 2007, 11:19:05 PM »

Too tall,

Admiitedly I did not go into depth reading the whole tactlab info  - busy packing for a trip - but I just don't get it - and didn't get it 25 years ago when I first heard about it - compensation for the equal loudness contours. Why do I want speakers to change their eq every time the level goes up or down? If someone unamplpified talks to me loudly and quietly, I expect the eq to change (in my head - not at the source), naturally, since it is in my natural perception. Why would I want a sound system to do what does not occur to an unamplified source?  

Obviously I am dense on this so I will need some help.

6o6
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Raimonds Skuruls

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Re: real sound lab Coneq ?
« Reply #23 on: October 15, 2007, 12:17:26 AM »

Message from Real Sound Lab:

Thanks to those of you who attended our Demo at AES in NY, and thanks for the interest towards CONEQ technology. I’m sorry, if some of you considered Demo session too short and insufficiently informative. I also hope that you’re enjoying not only the USB drive received but also CONEQ measurement software and CONEQ FIR filter DirectX plug-in demos that it contains – please try them out by using Quick Start Guide enclosed in the drive. For those of you who haven’t had chance to come to the Demo – you can download this demo software from our website. I would like to emphasise that we’ll highly appreciate all your constructive opinions and reviews, no matter how critical they are. They always help us to explain the concept better and in more detail, sometimes it even helps us to develop new features of our technology, which currently is only in its early days and there is much more to come.

Here I would like to make the following promise: everyone who will contribute to detailed reviewing of CONEQ and helping us to develop it further will receive free copy of the most functional professional version of CONEQ measurement software CONEQ TOURING. Please send your opinions to info@realsoundlab.com. We also hope to hear about all your experience when using CONEQ in practice.

Below are my short comments/ explanations triggered by forum discussion so far:
1)   First and foremost I would repeatedly emphasise that CONEQ is NOT a room correction tool. Influence of the room has been excluded/ minimised to negligible. But of course by correcting the loudspeaker we take into account imaginary loudspeaker images of the nearest reflective surfaces (first typically is floor). The essence of CONEQ correction is to create undistorted copy picture of original source of signal (sound) in any room to match this original source in the same room as close as possible. We assume that this task could be accomplished by making the loudspeaker to emit exactly identical acoustic (sound) power as to its frequency spectrum, as the original sound source would emit in the same environment. Therefore we work with correction of loudspeakers’ acoustic (sound) power frequency curve. We assume that the human ear (or more precisely hearing analyzer) analyses timbre characteristics of the sound by sound power and not sound pressure. Many of you perhaps know already that by a simple experiment it could be easily demonstrated that in the location where sound pressure mic shows a picture of heavy interference, human ear does not hear any problem. Regarding the room ‘issue’ – every room, good or bad, treats the sound equally regardless of from where comes – from a loudspeaker or e.g. from a violin.
2)   CONEQ measurement technology is NOT time blind. Transfer function of loudspeaker system is obtainable both trough dual FFT or by calculating FFT from system’s impulse response. The last approach is used in CONEQ commercial software.
3)   It has been misunderstood by some of you that there are thousands of measurements – it’s our fault – we should be better at explaining. The shortest period of a test signal is 0.341 sec, allowing for 176 measurements per minute. Practice shows that good repeatability is achieved through about 80 measurements already. We suggest using 150, whereas 250 would be realistic maximum. Measurement points should not be dispersed throughout the room, they should be taken equidistantly from the speaker (+/- 0.5m).
4)   Time error of measurements: it should be carefully observed that distance of different measurement points to the speaker does not differ for more than 1m (3ms). Normally it is easy to meet this criterion. But in next versions of CONEQ we will go further by removing even this restriction. Currently the success of observation of the relative equidistance could be judged by looking on the ‘Impulse Response’ graph, where the first peak of IR should be depicted at about 5ms. If it is shown remarkably earlier (1…2ms), we suggest to repeat the measurement.
5)   Speaking of 4096 correction points – they should not be viewed as parameters of a classical parametric EQ – it is number of FIR filter coefficients in the first version of CONEQ used for our gear APEQ, allowing using FIR filters of 6Hz resolution. CONEQ software for measurement and filter synthesis makes possible synthesising of FIR filters of any length and they could be further in DirectX plug-in. Use of longer filters increases filter resolution in the low frequencies.
6)   Loss of dynamic range: even if the correction is deep (up to + 10dB) but is not wide (around an octave), there should be no concern about loss of dynamic range. If the loudspeaker is suboptimally emitting some wide part of the range, good sound engineer will fill this deficiency with other means or tools and the range there will be somewhat suboptimal anyway will have loss of dynamic range and this will be entirely determined by this not so good speaker. As some say – first one should ‘tighten screws’ and tune crossovers, and only then do proper equalisation, not vice versa.
7)   The curve shown in our brochure (and reproduced in the beginning of this discussion) is Yamaha NS10, it corrects greatly. With regard to our demo in NY I would like to compliment Genelec… Their uncorrected curves are almost flat and it really needed some focussed listening to hear the CONEQ introduced correction. The message here is – regardless of how good the studio monitors are, there have never been any adverse effects introduced by this entirely automated correction. It has required no human interference at all.
Cool   Phase response: let me begin with a little joking – could anyone of you send us his most troublesome phase response curve – and we’ll prepare correction of it? But back to being serious – FIR filter allows for phase correction as well. At this moment we understand the need and are able to implement correction of group delay of subwoofers and phase response related to it. In order to develop a commercially usable phase correction tool, we would like to know the opinion of users (i.e. you) about the usefulness and need of such correction. We perceive another use of phase correction – achieving of phase identicality of both channels of a stereo system. And yet again – question to you – how much it is needed?
9)   Effect of Comb Filter by measuring of acoustic (sound) power is NOT present. This could be explained either by analysing of the question from ‘energy’ perspective, as it represents the summary (omnidirectional) view, or by exploring interference picture of each separate measurement, as in each measurement this Comb Filter is shifted on frequency axis against a Comb Filter of another measurement so that it ‘fills in’ the gaps of frequency curve caused by Comb Filter of this another measurement. Presumably exactly the fact that our measurement excludes interference, allows to rely fully on measurements and to automatically synthesise direct EQ from that measurement, without any interactive controlling of the measurement peak and dip points, as opposed to RTA SPL measurements.
10)   Acoustic (sound) power (AP): it seems to me that the concept of AP is somewhat new to many. And thank you for some explanations already provided within this discussion. I would like to add only that acoustic power has been and still is measured in acoustic laboratories, which have reverberation chamber along with an anechoic one. We offer doing pretty much the same in a field environment. When measuring of AP frequency response, one could use an ordinary room to serve purpose of reverberation chamber (if only is not heavily damped). In such case measurement points should be taken on the far-end diagonals of the room (as from loudspeaker’s position). The obtained result should be corrected by reverberation time frequency response of the given room. Currently we do not have a tool for measuring of this reverberation time curve but we may prepare it, if the market would be interested. For those of you, who would like to know more on AP concept, I would like to suggest the following source: S.J. Yang and A.J. Ellison, Machinery noise measurement, Oxford, 1985. It is not that much about machinery noise, but well explains all concepts of acoustic (sound) power and its measurements.

And at the very end I would like to apologise for not having an extensive User’s Manual for CONEQ – it will arrive soon.

Raimonds Skuruls, CTO of Real Sound Lab, Inventor of the technology

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BruceOlson

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EASERA SysTune
« Reply #24 on: October 15, 2007, 04:00:03 PM »

theo mack wrote on Sat, 13 October 2007 21:13

I look forward to the day when Smaart makes the process of averaging many test mic positions easier.


Would it be alright for a new software called EASERA SysTune that was shown at the same AES to do this? See http://www.EASERASysTune.com

Quote:

Or if you had the budget and time, a real time display using a dozen or so mics.
Is 8 mics enough?

Quote:

Not realistic with a paying audience but for setup in a room like mine, it would be really interesting.
I've done that with SysTune using both speech and music as the source.

Quote:

I know it can be done now with big expensive test rigs, But with cpu's getting so fast I hope a compact affordable fast way to do this comes soon to our price point.
I agree, is $750 reasonable enough?

Quote:

AES 2011
Can you really wait that long? Why not this year?

Too Tall (Curtis H. List)

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Re: real sound lab Coneq ?
« Reply #25 on: October 15, 2007, 05:24:42 PM »

Bob McCarthy wrote on Sun, 14 October 2007 23:19

Too tall,

Admiitedly I did not go into depth reading the whole tactlab info  - busy packing for a trip - but I just don't get it - and didn't get it 25 years ago when I first heard about it - compensation for the equal loudness contours. Why do I want speakers to change their eq every time the level goes up or down? If someone unamplpified talks to me loudly and quietly, I expect the eq to change (in my head - not at the source), naturally, since it is in my natural perception. Why would I want a sound system to do what does not occur to an unamplified source?  

Obviously I am dense on this so I will need some help.

6o6



Dynamic automatic room correction???
This is new to me also. Last time I looked I did not know they were doing this.
I may have missed it.

I was interested on what he figured was the right way to do room correction. From what I have read Radomir was one of the top people in the Western world on DSP. Creating a turn key FFT analyzer when only people like B&K had such a thing impressed me also, but I never saw a sign one way or the other if he knew anything about acoustics.

I suppose the other companies forced him to come up with something "different" to keep his place top of the heap for Home Audio room correction.
I know this from having bought a couple HT recievers in the last year.
The cheapest HT reciever out there now comes with a microphone and DSP that will do automatic room correction.
Just press a button and sit quiet.

I was too scared to let the little monster out of the box.
You can also put in EQ manual. I may try that if we find a nasty room mode.

Speaking of room correction the blurb on the magic the Easera Systune is capable of is a little...overboard???

Oops, I don't mean to hijack the thread.
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Too Tall
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Ivan Beaver

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Re: real sound lab Coneq ?
« Reply #26 on: October 15, 2007, 06:30:11 PM »

So life pretty much sucked for you untill recently? Laughing

Anybody who spends hours "eqing" a monitor has other problems.

There is a lot more to a particular loudspdeakers "sound" than a flat response.
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Bob McCarthy

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Re: real sound lab Coneq ?
« Reply #27 on: October 15, 2007, 07:37:18 PM »

Raimonds Skuruls wrote:
1)  We assume that the human ear (or more precisely hearing analyzer) analyses timbre characteristics of the sound by sound power and not sound pressure.

--- That is a very BIG assumption. How exactly does a human spread their ear all over the listening area?  

Raimonds Skuruls wrote:
Many of you perhaps know already that by a simple experiment it could be easily demonstrated that in the location where sound pressure mic shows a picture of heavy interference, human ear does not hear any problem.

--- Amazing claim. Not my ear. And not the ears of my clients.

Raimonds Skuruls wrote:
Regarding the room ‘issue’ – every room, good or bad, treats the sound equally regardless of from where comes – from a loudspeaker or e.g. from a violin.

--- another unsubstantiated claim. As if speaker placement and directionality have no effect on speaker/room interaction

Raimonds Skuruls wrote:
4) Time error of measurements: it should be carefully observed that distance of different measurement points to the speaker does not differ for more than 1m (3ms). Normally it is easy to meet this criterion.

--- This seems to indicate that an area no larger than 2 rows in depth could be factored into the equalization. Most PA applications (hopefully) have substantially more depth than two rows.


Raimonds Skuruls wrote:
Phase response: let me begin with a little joking – could anyone of you send us his most troublesome phase response curve – and we’ll prepare correction of it?

----I could send you boatloads of troublesome phase responses and the point is that you CANNOT prepare a correction on your DSP. The relevant info of these phase traces is that it indicates solutions that cannot be treated with 4096 filters, or all-pass phase shifters or whatever they think of next. Solutions in speaker position, delay setting, splay angle and relative level are often strongly influenced by the phase response.

Raimonds Skuruls wrote:
9) Effect of Comb Filter by measuring of acoustic (sound) power is NOT present. This could be explained either by analysing of the question from ‘energy’ perspective, as it represents the summary (omnidirectional) view, or by exploring interference picture of each separate measurement, as in each measurement this Comb Filter is shifted on frequency axis against a Comb Filter of another measurement so that it ‘fills in’ the gaps of frequency curve caused by Comb Filter of this another measurement. Presumably exactly the fact that our measurement excludes interference, allows to rely fully on measurements and to automatically synthesise direct EQ from that measurement, without any interactive controlling of the measurement peak and dip points, as opposed to RTA SPL measurements.

---- No form of mathematical averaging, no matter how many points you take or how you massage the data REDUCES the comb filtering. There is no best EQ for a comb filtered region - because no matter what, it is only best for a single point, with all other areas having substantial differences. The treatment lies in taking steps to reduce the combing - but since you said earlier that we can't hear it, I am not sure why point was carried forward.

6o6
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Scott Fahy

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Re: EASERA SysTune
« Reply #28 on: October 16, 2007, 01:06:38 AM »

Hey Bruce welcome to the LAB-  how's Mr. Yost doing these days

Scott
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Bennett Prescott

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Re: EASERA SysTune
« Reply #29 on: October 16, 2007, 11:37:58 AM »

Bruce! Great to see you here. Here's a digital beer: |=3 (I dunno... how do you make foam?)
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Re: EASERA SysTune
« Reply #29 on: October 16, 2007, 11:37:58 AM »


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