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In Reply to: RE: The idea that the posted by Presto on May 25, 2012 at 10:13:36
The reason why people prefer slightly rolled off response is that the studio monitors used have similar response. Most CDs (including classical and probably even some so-called "purist audiophile" recordings) are EQ'd at some point in the "mastering stage". (The correct term for this stage is "pre-mastering" as the actual mastering takes place at the manufacturing plant.) This EQ is needed because the earlier stages of processing produced imbalance results, either due to accidents or because errors were deliberately made to offset errors in the inferior monitoring and room acoustics at the recording venue or mixing stage.
There are some very subtle and complex relationships between equalization and sound stage. After making lots of adjustments on the electronic crossovers of my Focal satellites and sub woofer I still was left with a few peaks due to room modes, at 40 Hz and 127 Hz and a small peak at 781 Hz. After living with these for the past two months I decided yesterday to see what would happen if I took these out with a parametric equalizer. As expected a few recordings that were bass heavy because of peaks at similar frequencies were tonally improved. This was to be expected, but what was much more surprising is that the sound stage improved greatly in depth, as my mind was no longer cuing in on the room resonances of my small listening room and hearing past them to more of the ambiance on the recordings.
To do this EQ I had to use the iZotope parametric equalizer that comes with Soundforge, a pain in the ass process that takes a minute to process each recording to be played. Unfortunately, this software is tied to Soundforge and is not usable as a VST plugin that I can use with my player software such as cPlay.
Do you know of any good parametric equalizers that run as VST plug ins? Are you still recommending the one from aixcoustic.com?
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
Follow Ups:
The ReaEq plug in from the Reaper folk. Works at 64 bit precision, and is minimum phase. There is also ReaFIR, a linear-phase para-graphic (meaning you can draw your own curve if you aren't trying to be precise at the moment).
The plugins are free. Reaper has a package for either 64 bit or 32 bit hosts. They are not pretty, but are extremely effective. I have worked with some of the expensive plugins, and always come back to these two.
Give me Ambiguity or give me something else!
Tony:
Wow, that was a while ago "AIX Acoustic". Good memory!
I am running DSP crossovers almost exclusively now, except when I pop in my passive-crossover based reference monitors to see how far I've wandered off the beaten path. ;) As such, I can do pretty much any equalization imaginable right in the crossover.
I might have to listen to the AIX acoustic crossover again - it's been years!
Cheers,
Presto
Not memory on my part. AA search. :-)
I got it to work and set it up for a similar response to what I was getting out of the iZotope parametric EQ in Soundforge. There were a few differences, however, i.e. "bandwidth" is specified instead of Q. Also I am suspicious of the AIX plugin because it's "flat" position had about 0.5 dB of gain. Not something any serious DSP expert would ever do.
I need to do some more tests to see if this gain problem is the only problem. At least it interfaces with Soundforge and cPlay.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
Thanks for bringing that to my attention Tony. Some basic testing (in versus out) is called for here. Simply recording the output of the plugin in the digital domain should do it... I'll see what version of this EQ I have.
Cheers,
Presto
I am no longer seeing this difference in gain, so perhaps the problem only appears in certain conditions. Or perhaps it could have been "cockpit error" on my part when I first started playing around with the program.
I've got the EQ running as a VST plugin under cPlay and this is definitely improving my sound, cleaning up some muddy bass on some recordings as well as improving imaging. I've done some In-out tests and the amplitude response appears to be more or less as displayed on the graphs. I've looked at impulse response and there is no pre-ringing, looks like minimum phase. I didn't observe distortion products on some sine wave tests, but this was not done at high precision, so there could be non-linear results at -130 dB that I might have missed.
My next step will be to try some time correction as well, but this is much more complex and I'm not sure what programs are available for testing that aren't expensive.
Tony Lauck
"Diversity is the law of nature; no two entities in this universe are uniform." - P.R. Sarkar
I think you’re on the right track, remember when CD’s were introduced, they “had to” sound better than the LP’s they were targeting to replace and an easy way was making a brighter recording that anyone would notice as “more extended”. Get a copy of Michel Hedges Breakfast in the field and hear an early a very clear but very bright recording.
Part B I think Is that harmonic distortion is normally to the high side of the fundamental producing it so in the case of loudspeakers, one finds the harder they are driven, the brighter they sound and eventually sound bad. As the size of loudspeakers has fallen and power handling climbed and lacking a standard measure of linearity or usable loudness, what we have now are speakers that are often less signal faithful than the old days.
An Altec horn for example had essentially no power compression because if you drove it that hard, the wire came off the voice coil at ~ 125-150C, now days some VC adhesive will tolerate > > 350C (Rdc doubles at about 230C).
Also. I do think we “hear through” obstructions without being conscious of it and so if you remove a que that the speakers or room add, what you hear is a more faithful image while you were previously unaware of that same que.
I would offer that for the most part, if you can measure and correct your speakers based on outdoor / anechoic measurement, then you can be pretty sure that what your doing will fix both mag and phase simultaneously.
To the degree what you’re trying to eq is caused by a delayed signal (reflection) combining with the direct, you can’t really fix these as they are not a minimum phase problem. In the old days, they said only cut peaks and bumps, NEVER try to fill a sharp deep notch as that is the signature of a comb filter (caused by a delayed signal).
While the miracle of DSP and incomplete explanation or limited measurement resolution will make it appear you can “fix everything” the absolute best one can do is fix it in one specific spot where the measurement was taken and often making it worse everywhere else.
The “fixed” location is limited to about ¼ wavelength in size for the highest frequency being corrected so it is truly futile (acoustically) fixing it with dsp when the listening area is much larger than the wl in size. At 20Khz, the wave length is about 5/8 inch..
Fixing the source is the best way I think. Can’t help on a plug in but would offer that LSPcad can emulate a number of speaker controllers which have parametric’s , the down side is most dsp units are somewhat different when you call for a given filter set or alignment.
You can also listen to music through that alignment if you want. For work, I take the actual unit and measure / adjust the eq until it overlays on the transfer function or response I need.
Also, you could save the impulse response for the correction and convolve it with the music, the ”Gratisvolver” at Cat acoustics is free anyway. That also appears to be a way of transferring “what a speaker sounds like” as well as the speakers impulse response can be convolved with music too, sort of a software way of doing loudspeaker generation loss recordings we do at work.
Best,
Tom
Get a copy of Michel Hedges Breakfast in the field and hear an early a very clear but very bright recording.
The original analog version from 1981 or the later CD copy? I have both. In this case, I think the "villain" is the minimal processing done on the recording. Your post piqued my interest because I find that overall, the original analog copy is an incredibly natural sounding recording - albeit a touch bright (an easy thing to cure). On The Happy Couple , you can so easily visualize his hands moving in his inimitable way. I've had to good fortune to see Hedges live three times before his untimely accident.
In another post, you mention the KEF Blade. I was in the Bay area earlier this year and noticed a hi-fi shop in downtown SFO nearby the Ruth's Chris where my wife and I had dinner. Since it had literally been years since I've set foot in an audio dealer, we walked over. They graciously played a couple of tracks on the Blades. As a coherency freak, I found them good in that respect. But, the apparent image size was tiny and very directional. Not my cup of tea. :)
Hi
I have an early CD of it (and one of his others).
I like the recordings very much but are what I was talking about as to how they made the early CD voiced differently.
In that case, on speakers that measure flat is a very bright recording and bright I think “so it sounded better”.
I have not heard the Blade speakers myself but they did talk about hearing the radiation shape but they didn’t provide any directivity measurements that I recall.
While I noticed the effect developing the Synergy Horns speakers for work, they are constant directivity and radiate as if they had one driver..
...they are constant directivity and radiate as if they had one driver..
Yes. Back to the Blades for a moment, they seemed quite coherent even if it was a coaxial mid/tweeter and two side firing woofers. It just suffered from shrunken image size.
I'm thinking that would not be the case with an SH-60. Maybe two would be better.
I agree, the recording is not bright. Same is true of all or most Windham Hill Recordings.
Edits: 05/27/12
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