Ideas for Version Seven

[Audio Yoda]

I may have missed this in the manual, but, is there a way to highlight multiple areas of a waveform so that a filter can be applied to different sections in one pass?

[Craig Maier]

No. Each area of a file that needs selective filtering must be highlighted and then processed individually - - - one at a time.

[Walter]

Version 7

Is there a way to make the 20 or thirty band equalizers
self adjusting? Back a few years there was a graphic analog equalizer that did this using a calabrated mike and pink noise.The auto function was to even out the interactions of all the band settings,The demo sales pitch was that it was computer controlled to adjust the sliders.

[Craig Maier]

I think that I understand the auto graphic idea. You set up a microphone in the room and play random noise and the system adjusts itself to produce a flat response; I am just not sure what the band interaction elimination concept is, unless you are referring to the elimination of that effect when running in auto mode. I think that if a high resolution fft based measurement system (maybe 32,000 band) is used to adjust the graphic(s), then band interaction would be automatically dialed out of the system.

[Walter]

Version 7

As I understand band interaction ,when a band is adjusted it also affects adjacent bands, so you have to re adjust ubtil the response is flattened out across all bands.

[Craig Maier]

Of course, if we provide this feature, a computer would become a permanent part of your audio system. Unless - - - you take all of the final readings off of the Diamond Cut graphic eq and translate them manually into a standard hardware based eq.

[Walter]

Version 7

From DC LIVE/Forensics Manual p309
Q.I noticed that a Live Mode is now included in SIX/LIVE
What's it for?
A. This allows you to take a "live" audio stream and process it without first recording to hard disk.With this function you can make your computer system an intergral part of your home stereo system. Imagine adding the sound of a $15,000 vacuum tube amplifier to all your audio .

[Craig Maier]

Just for edification purposes, you can do that now - - - real time feedthrough is one of the features of DC6. Not only can you run EQ's real time, but if you use the Multifilter, you can EQ, De-Noise (Impulse Filter, Continious Noise Filter, etc) and, of course, insert the VVA to obtain any type of tube sound that you prefer. All of these operations can be done in real time with a computer connected to your stereo system. We know of some DC6 users who have dedicated a computer just to their main audio system and use it in the abovementioned manner.

[Walter]

Version 7

I already have this setup, that is why I was asking if it was
possible to automate the eq. for this room?

[Craig Maier]

Right now, you would have to do it manually. In the future, we can look into automating the process - - - maybe for version 7.

In the meantime, consider going to the following post which indicates one method for tuning up a room with Diamond Cut.

http://www.diamondcut.com/vforum/sho...light=Sounding

Consider making a CD having wideband random noise as the stimulus rather than weighted noise. This will allow you to use one instance of Diamond Cut in Spectrum Analyzer mode as the measurement instrument while using a second instance of the software as the 20 or 30 band EQ.

The previous description of this, using a set of Diamond Cut CD's and an SPL meter is located at:

[jdedrick]

Remember - it's all about the filtering!

Craig:

I'm a recent purchaser, and my use of the product is in digitizing and restoring a large collection of vinyl albums. I think DC6 is a good product and an excellent value, but so far my experience has been mostly one of frustration at my inability to find filter settings that do an acceptable job at getting rid of pops/crackle without destroying the music, so take what follows in that context.

I'd like to stand back and give my perspective on your product roadmap overall. In this thread there have been many suggestions for usability improvements, changes to the UI, etc. All of these are valid and valuable inputs. However I think you should always try to keep focus on the fact that faced with a choice between the worlds slickest user interface, but a product that requires a lot of manual intervention to get the job done, and a command-line-driven DOS program that can do as good a job as I can at figuring out where the imperfections are and getting rid of them, I'd buy the latter, and I suspect most others would too.

A case in point is that I bought your product not because of anything having to do with ease of use, but because I thought of the tools out there it had the widest variety of filtering approaches available, and therefore was the most likely to be able to get the job done. What I want to emphasize is this:

Improving your filter technology is far more important than the sum of any other improvements or feature additions you might conceive of.

You've got a good set of possibilities now, but you're very far from the end of the road of possibilities. To name a few:

There are wavelet based approaches to doing a better job of localizing defects: http://dafx04.na.infn.it/WebProc/Proc/P_038.pdf

and fuzzy-logic approaches to improving median filters by combining time and rank ordering information:
http://www.hindawi.com/GetPDF.aspx?p...10865701000312

and a huge body of research on removing impulse noise from digitized photographs, most of which is could be directly applied to audio.

And finally, there remain opportunities to do something totally new, like (as i suggested previously) implementing a filter that takes advantage of the unique information available to you, like the fact that impulse defects on a record will probably span more than one revolution of the turntable.

To harp on this last one a bit: It seems you could do a nearly perfect job of identifying scratches that span more than one groove. Having done so you could be very aggressive about filtering them - far more aggressive than you could be generally since the chances that you'll have sharp musical transients that happen to be *exactly* one revolution (1.8 seconds) apart has to be vanishingly low.

This kind of innovation is the sort of thing that could make DC7 emerge from the pack to become not only the best buy among audio restoration tools (as it is now) but simply the best.

Regards,
J

[Craig Maier]

J,

Our priority in new product development has always maintained improvement of the algorithms as the primary driver. Everything else is secondary to that. As you can tell from the Diamond Cut GUI, we are not artists at all, but engineers. With each release, we have made improvements to each of the more complex non-linear routines such as the impulse filter, CNF, etc. We continue along that same pathway of prioritization with the work being done on DC7.

The concepts that you mentioned have been known to us for a long time. At general issue is that there is no such thing as a "one size fits all" routine. Wouldn't it be wonderful if there was! Each audio file presents a different challenge, thus, we have not as yet found a "magic bullet" but are still looking for it. Right now, the best tradeoff has been to provide a wide array of filter types and controls so that a user can address all of the various situations that exist out there.

We do have some new ideas that we are working on right now which will hopefully provide more uniform results over a wider range of source material. But, only history will reveal if we will have hit the mark without compromising some other aspect of the performance of the system on certain materials.

Lastly, we do not use "canned" routines in our software. The routines are developed strictly by ourselves. That approach has allowed us the flexibility to do the most with the smallest number of controls possible.

Craig

ps - One last note. I think that it was Einstein who said ~

"Make everything as simple as possible, but never simpler."

We agree with that assessment.

[cromag]

I'm still far from exhausting the capabilities of DC-6, but there is one thing I'd REALLY like to do that I can't do right now. (And if it turns out that the program is capable of doing this, please accept my apologies and tell me how! )

I'm cleaning up a lot of LPs from the 60s through the 80s. When I'm in the final stages of fine tuning I listen to the "source" file and "destination" file alternately to make sure that I have actually cleaned up some noise without introducing artifacts -- since I remember these songs from vinyl, on cheap stereos at that, I can tolerate some hiss and crackle but any digital artifacts like metallic ringing, echoes or deadening seem much more out of place.

What I'd really like to do, after running a filter in the final stages of cleaning, is start a "synchronized" playback that would start the playback cursor advancing through both files "in sync." I could then alternate between source and destination without stopping, re-positioning the cursor and starting up.

The one song would play all the way through, but I could hop back and forth between the two files while it was playing.

So far, that's the only thing I wish I could do that I haven't found a way to do.

[Dan McDonald]

cromag -
that sounds like it would be very useful. Sort of like the 'bypass' box on filters. I think the only way to do that now is to start up two instances of the software and start them at almost the same time.

It might take quite a bit of processing power though - especially if you had 24-bit files.

Dan

[Walter]

Switch source

Try file split and re combine, this will only be mono files
but it works well if you convert it to stereo,then you can chose R+L ,L or R.
I have used this method to compare original with processed.