[dlaloum]

OK I know I'm new here - so please be gentle...

And yes I am raising one of the blasphemous themes...

But - just so you understand where I am coming from.... my main listening system is Room Equalised using Audissey.
This means that all my sounds get digitised at some point - including Vinyl. (my Receiver does have a "Pure Audio" mode where I can bypass the processing but I seldom use it, as the room is a major (huge) factor in the sound, and adjusting that comes first....
So - says I - why not digitise in raw form straight from the cartridge, then do all EQ digitally - both Room and RIAA. (Potentially this would also allow for adjustment of individual cartridge EQ as well)

This brings me to my next obvious question - what software is there that a) does Digital RIAA, and b) doesn't mess up the sound / maintains transparency, final item c) is that the software should work in real time (as opposed to recording then EQ'ing before listening)

Has anyone played with this? Had good results? Have other comments?

thanks

David

P.S. My initial digitising experiments were done on an M-Audio 2496, which lost a channel a couple of weeks back, my new ADC will be a e-MU 1616m, which should be delivered any day now
PPS. My Rig: Gallo Nucleus Ref3.1, Biamped with 2xQuad 606 & Harrison Labs 50Hz LP Line Filter, Onkyo TX-SR876 Receiver (acting as Pre-Pro), Phono Stage on TX-SR876 and  Creek phono stage OBH18 with OBH2 PSU; Revox B795 & JVC QL-Y5F turntables; Cartridge - Shure 1000E with N97xE-SAS stylus, Ortofon 320u & TM20; AT12Sa, Signet TK6Ep, Stanton L720EE, Empire/Benz MC1HO, Sony XL-MC104
Other sources - Squeezebox3 feeding SPDIF into TX-SR876, Media PC feeding HDMI Audio(/Video) and SPDIF audio to TX-SR876

[firedog]

I've used Diamond Cut for this purpose. Works well. But you have to apply the RIAA after recording. Don't know of any software that does it during recording.

[analognut]

I also have used Diamond Cut (extensively). DC6 is the only software I know of offhand that does RIAA inside the computer. It does RIAA as well as many other obscure eq's. I work at digitizing vinyl every day but don't use DC6 at all any more because to my ears the program gives everything a grating digital sound, compared to other software that's around.

For me it's a lot handier to do the RIAA in a high quality phono preamp right at the turntable. I use Waves audio plug-ins exclusively for all my processing and restoration. The Waves Q10 Paragraphic Eq comes with a preset that does RIAA eq, but I don't use it.       waves.com   

[bunnyma357]

If you are on a Mac, Pure Vinyl is an excellent choice for this.

http://www.channld.com/purevinyl/

I have been very happy using a mic preamp/interface for inputing the straight signal from a MC cartridge, and doing RIAA in software in real time with Pure Vinyl, I also use IK Multimedia's ARC plug-in for room correction, and at some point plan to use the PV software to provide active crossovers for my speakers.

Here are a few threads with more details on my system and also PV as a Phono Preamp:

http://www.audiocircle.com/index.php?topic=80564.0

http://www.audiocircle.com/index.php?topic=81037.0

http://www.audiocircle.com/index.php?topic=81402.0



Jim C

[dlaloum]

Thanks for those replies.... I am on a Windows system, so PV is unfortunately not an option...
currently collating a list of options, there appear to be a range of plugin VST filters available....

Came across this reference page too:
http://jiiteepee.fortunecity.com/riaafilter/index.html

Which unfortunately doesn't tell me much about how they sound.
For good results the Digital RIAA needs to not just EQ frequencies but also adjust phase... So a minimum phase EQ is require (so I am reading...)

[JDUBS]

Jim C, perhaps this can be in another thread as the OP says that he needs a Windows-based solutution, but I am curious to hear more about your experiences with Pure Vinyl.  I am (still) looking at it in conjunction with a Mac Mini and an RME Fireface 400 as an alternative to the DEQX.

Thanks
Jim

If you are on a Mac, Pure Vinyl is an excellent choice for this.

http://www.channld.com/purevinyl/

I have been very happy using a mic preamp/interface for inputing the straight signal from a MC cartridge, and doing RIAA in software in real time with Pure Vinyl, I also use IK Multimedia's ARC plug-in for room correction, and at some point plan to use the PV software to provide active crossovers for my speakers.

Here are a few threads with more details on my system and also PV as a Phono Preamp:

http://www.audiocircle.com/index.php?topic=80564.0

http://www.audiocircle.com/index.php?topic=81037.0

http://www.audiocircle.com/index.php?topic=81402.0



Jim C

[analognut]

dlaloum-
You've brought up a really good point. Phase smearing is something that lesser software will do and that's a big risk you take when you decide to go farther than simply recording an LP and saving it to a file. How do you know which software does and which doesn't? Maybe your program works in one instance but in another instance with different settings it doesn't work correctly? Chute, I've even had that kind of trouble with de-clickers. Thought it was working great, then somewhere down the road find out that in certain situations, maybe a throaty growl from a saxophone say, or a certain piano note at low volume the thing would distort the music. The only way I know to be sure is use something that the pros use and endorse. The day I discovered Waves all my worries went out the window. I have seen them claim on their website numerous times that their s/ware introduces no phase errors. Most of it is artifact free. If there is an issue with a product that can introduce artifacts at extreme settings it is always outlined in the manuals, which are on the website, free to download.

Doing RIAA in the computer is your choice, just beware that you have a good product. As I noted earlier, the Waves Q10 Paragraphic Eq has a preset for doing RIAA. Waves makes a claim that the Q10 is the world's most powerful eq. I can tell you its has +/- 18db of gain and Q adjustable from 0.5 to 100, all adjustments in 0.1db steps. If you're just buying PV for the RIAA my recommendation would be to check out the Q10. At times Waves offers it as a standalone plug-in, rather than as part of a bundle. They constantly change their website around, they may have it that way now, I don't know. All Waves products work on TDM or Native (Mac or PC) systems. My plug-ins are DirectX, but I think they offer them as VST also, not sure. Last time I looked, the Q10 was $300. Bunnyma357 and JDUBS, I think PV is more than that, isn't it, and you don't REALLY know how good it is, do you? Here is a pic of the Q10:



Assuming you get set up to do RIAA in the computer, if you want to REALLY go all the way, try this:

Play and record a pinknoise track off of a test record into your computer. Do your RIAA. But you still don't have a perfect RIAA curve because your computer program hasn't compensated for the frequency response of the cartridge. If your cart is lacking in the bass, or has a tipped up high end your music will still sound that way.

So you use a software spectrum analyzer that can do a logarithmic display in conjunction with a powerful equalizer and you adjust your eq for the straightest possible line. If your cartridge had perfect response the graph would be a straight line when frequency response is plotted logarithmically against amplitude and the graph would look something like this:




This picture is displaying frequency response from 10Hz to 20,000, where the amplitude falls to zero. This is a pic from the manual of the DC6 software, which I don't use because I don't like the sound of DC6, but you get the idea. If you can get a perfectly straight line you have perfect response. The concept is awesome. I use the concept but I don't do eq in the computer. I use an old Behringer DSP-8000 hardware eq which is a 24bit dual processor (one for Left and one for Right) having 31 bands of "constant Q" eq, adjustable for up to plus or minus 16db in 0.5db increments. I output my phono stage into it and then into the computer, where I used the spectrum analyzer to tell me what adjustments to make on the Behringer. I'm able to do this for L and R channels individually and my line is flat +/- 0.5db from around 19kHz down to about 25 Hz and appears straighter than this one. The difference it makes in the sound of the music is astounding. Keep in mind here that the point is to straighten out the response of the cartridge, not to have a theoretically perfectly straight line. One db is said to be the smallest unit of a volume change that humans can hear anyway. A certain gentleman over at AA who shall remain nameless has some of my CDs and has told me in several different ways they are equal to the best he's heard, so I know we are on the right track! 

[dlaloum]

Indeed  Great minds think alike  (and fools seldom differ... )

I did record a pink noise track (and did some FFT analysis and EQ for flat Freq Resp - it was shocking how far off a linear F/R most of the cartridges are!)- and was playing with cool-edit pro's eq function.... But I struck a couple of limitations.

1) My ADC's (M-Audio 2496) were not good enough
2) EQ was a bit crude - could be made to work, but would not adjust for phase as proper RIAA does
3) My Phono stages were a limiting factor !
a)Working with and researching my cartridges (MM/MI and HO MC), I soon found that I can't get the best out of them without having a higher input impedance. Recomendations vary between 60k and 100k ohms - but differs with each cartridge (oddballs like grado may need 10k to 30k) - but reality is that you cannot get the best from your setup without moving away from the 47k "standard". (which was never standard!!!)
b)To go with the high impedance I need to lower the capacitance hugely - down to total capacitance of circa 100pf.
Both my phono stages (and the one internal to the 1616m) are 47k / 220pf - add to this at least 50pf internal to the tables, and say 70pf for a cable (minimum for a decent but not low C cable) - so my starting point at the moment is 47k/340pf miles away from what it takes for extended Frequency response.
So I am also looking for a new phono stage...

Luckily  for me my M-Audio blew an input channel - hopefully a shiny new e-mu 1616m will appear on my doorstep tomorrow morning... (I sure hope so, if it doesn't make it tomorrow it won't be till after X-mas and I will be peeved  )

So issue 1 is partly resolved, I am moving to a more transparent, higher resolution ADC within the limits of my budget (I really wanted a Lavry AD10 or Mytek Stereo192.... but at 4 times the $ it wasn't reachable!)

I seriously considered a Behringer DEQ2496 - it is capable of onboard EQ (later version of your 8000?)... but - I would have had to have its analogue inputs modded both to improve transparency and to provide a sufficiently high impedance (1M) to allow for impedance plugs. Secondly it was unclear without further research as to whether its EQ could or would correct phase.... cost of the Behringer with mods was starting to get close to the cost of the Lavry/Mytek - so that option got binned.

Interestingly - the 1616m has a HiZ Line input with an impedance of 1M as one option on its MicPre - this is spot on the required spec to use a MicPre for phono purposes....
It also has a reputation for transparency, and having high quality analogue in its MicPre circuitry. - So that made the decision easy.

Having theoretically sourced the amplification and impedance/capacitance matching components required at one end of the chain - I need to find really good reliable software to match at the back end of the chain.

So my next exercise is to find good software to try out - if it all works well (as it should!) - then I won't have to spend the next chunk of my budget ... allocated to the phono stage.

I am already familiar (and happy) with the functionality of CoolEdit Pro - so if I can find some plugins to provide the EQ, and another for bitperfect resampling - then I am set. I already purchased ClickFix which works very well... cleaning up some of the older LP's I have from the late 50's takes days per LP, as best results are achieved manually - by listening, identifying the clicks and pops manually (spectral view) and zapping them. (an initial high level desensitised automatic run over does a good job with the big stuff, but after that it is all manual). My experiments with 4-5 LP's showed me how much work there was per recording.... which started me down this path - I want the initial recording to be as good as I can possibly make it for starters - GarbageIn-GarbageOut - I don't want to have to redo weeks or months of work! (and I still want to stay within a reasonable budget)

I am not trusting my hearing.... primarily because hearing can be tricked... the end result must please my ear - but the instrumentation measurements have to first convince me that all is right with the world.... (otherwise how do I know that what I am enjoying is not just a hump at a certain frequency - which I can replicate any time with an EQ.... and which might sound great on one thing - and a few weeks later when I listen to something altogether different, it would sound disappointing!) - experimenting blindly is a frustrating, lengthy and expensive exercise.... I prefer to measure first then listen.

Looks like we have arrived at a very similar conclusion through differing routes...
Although the Behringer DEQ2496 has a poor reputation in terms of ADC/DAC (without modding that is) - it has an excellent reputation as a DSP... and if I can't get the software doing what I want it to, that may be another alternative, possibly along with something like a Korora phono stage (where I can fit a high impedance resistor) - or a JLTI with external plugs (if I can stretch to its price).

One step at a time though.... my next steps are

1) Receive and install 1616m ADC
2) Receive and install Low C Cables
3) Solder up some impedance plugs (47k, 56k, 62k, 68k, 75k, 100k)
4) Source and Test an EQ PlugIn with CoolEdit
5) Compare the MicPre impedance/capacitance adjusted, EQ'd result to results with the Phono stages.....

This is all with my Revox TT - in the meantime I have another project looking at getting a JVC QL-Y5F working.....

[analognut]

"I did record a pink noise track (and did some FFT analysis and EQ for flat Freq Resp - it was shocking how far off a linear F/R most of the cartridges are!)- and was playing with cool-edit pro's eq function.... But I struck a couple of limitations."
Actually, 1 or 2db is all it takes to screw up the sound. I am running an AT-OC9ML/II on an SME V/SotaStar vacuum table. The cart has about a 2.5db tip up at the top end, not to mention an grossly imperfect bottom as well. But this is the rule rather than the exception when it comes to cartridges. They are a long way from being perfect. I was really captivated by the OC9ML/II's ability to finely resolve low-level high frequency sounds and having it eq'd to flat has made it a true gem in my experience.

"Working with and researching my cartridges (MM/MI and HO MC), I soon found that I can't get the best out of them without having a higher input impedance"
I went through all of that for about 8 months after I got my Star/SME and stuck a cartridge on it. I ran a DL-103R for several months, then bought the OC9ML/II, could immediately hear that it had more potential resolving power and spent the next 5 months getting it right! In the end, the mfr's recommendation sounds the best- a load of 20ohms and VTF of 1.75gm. I use no damping of the SME arm whatsoever, but I learned this can be cartridge dependent-  with the DL-103R I needed the damping to get the best sound out of it. Your cart needs to be fully broken in before you can know how to load it.
 
I am not familiar with CoolEdit Pro, but have heard it's good. You need a program that can do 32bit Floating Point math  when you make changes to your 24bit files. Does CE-Pro do 32bitFP?

I have read many reviews from people who bought the DEQ8024 and thought it was crap. Apparently the relays that switch the power when the unit is turned on are crap. They stick or don't work at all. The DEQ is a totally different animal than the DSP series. Whereas I have the DSP8000, the later version was the DSP 8024. The same machine, but had more memory for stored presets, etc. You can find them on ebay for a couple hundred bucks.

I would definitely say I'm the opposite when it comes to measuring. I listen and then I know if the sound is right or not. I suppose that playing keyboards, guitar and trumpet would be a big factor in knowing when the sound is right or not. I played in bands all the way through high school and college and don't need to be told when the sound of a cymbal crash or an acoustic piano is right or not. A lot of people will not have had that experience.

"cleaning up some of the older LP's I have from the late 50's takes days per LP, as best results are achieved manually...I want the initial recording to be as good as I can possibly make it for starters - GarbageIn-GarbageOut"
Yeah, big 10/4 on the grabage in, garbage out. Having quality source material makes this hobby a lot of fun! That's why I quit buying used LPs- it takes too much time, or it's impossible to begin with, to clean up someone else's dirty groove-damaged records. But de-clicking taking days? Let me tell you a little story:

I have had the Waves Restoration bundle for close to 10 years. Prior to that for 4 or 5 years I tried everything I could find. I found something that seemed to work pretty well and stuck with that. At the time I thought "no way will I ever pay $1200 for the Waves restoration bundle- that's insane". I had a Waves CD that had every product they make on it. All of the Waves products are free to try for 2 weeks. After that they just quit working. One day I couldn't stand it any more and took that CD and installed the  Restoration Bundle of plug-ins into my computer. I guess I immediately went insane because I HAD to have that. The Waves software was the best-sounding, sweetest-sounding stuff I'd ever heard. But the real kick in the ass was the fact that it WORKED. Worked 10 times better than anything else I had tried. I tried out 6 or 8 different systems and paid money for a lot of them. But you want to know the big secret, what the pros forget to mention? It is the EASIEST software to use I've ever seen. Some de-clickers can be a real nightmare trying to get them to work, and then they still don't work right. DC6 comes to mind here, among others. For me, de-clicking a 45 minute LP recorded at 24/88.2 takes about 6 minutes for the software to automatically run over it. I always under-do rather than over-do, so the next step is to listen to the file for any clicks that are left and I manually remove those. So it takes me roughly an hour to completely de-click one of my LPs. It is astounding how powerful the Waves RB is. The ability to go in and remove MAJOR clicks without taking anything out of the music. The RB consists of 4 plug-ins: Xclick, Xcrackle, Xhum, and Xnoise. To get an idea of what those do you can download the manuals on the Waves website.

Keep us posted on how you progress! 

[bunnyma357]

Doing RIAA in the computer is your choice, just beware that you have a good product. As I noted earlier, the Waves Q10 Paragraphic Eq has a preset for doing RIAA. Waves makes a claim that the Q10 is the world's most powerful eq. I can tell you its has +/- 18db of gain and Q adjustable from 0.5 to 100, all adjustments in 0.1db steps. If you're just buying PV for the RIAA my recommendation would be to check out the Q10. At times Waves offers it as a standalone plug-in, rather than as part of a bundle. They constantly change their website around, they may have it that way now, I don't know. All Waves products work on TDM or Native (Mac or PC) systems. My plug-ins are DirectX, but I think they offer them as VST also, not sure. Last time I looked, the Q10 was $300. Bunnyma357 and JDUBS, I think PV is more than that, isn't it, and you don't REALLY know how good it is, do you?

Pure Vinyl is Currently $229, I got mine on sale for $189.  After talking to the designer and reading the info on the site, I feel pretty comfortable that PV is doing a REALLY good job of RIAA. The designer has presented White Papers to AES on digital RIAA, so he is welcoming peer review by audio engineers. He also raises questions about a PEQ or GEQ using FIR vs. IIR - I don't really understand all this, but it seems he has given it considerable thought.

http://www.channld.com/aes123.pdf

http://www.channld.com/pure-vinyl_support_faq.html#anchor34359

Also Channel D's other products are all engineering and testing applications - so they seem to come from a "Professional" mindset.

The question was also initially asked about doing RIAA in real time, which PV does, not sure if Q10 requires post processing, or if it can do things live - Also Q10 is just the EQ software and would require the cost of a Host DAW application to function.

I'm sure your solution does a great job at what you want it to do, as Pure Vinyl does for me -

"a) does Digital RIAA, and b) doesn't mess up the sound / maintains transparency, final item c) is that the software should work in real time (as opposed to recording then EQ'ing before listening)"

When looking at solutions, even though PV is Mac only the value it provides might make the investment in a Mac worth it. For me, it allowed me to upgrade from a MM cartridge to a Low Output MC cartridge, without having to buy a new phono pre-amp or step-up transformer. Having been extremely pleased with it's capabilities -  It may eventually end up allowing me to get rid of my Pre-Amp, MM Phono Pre-amp, DAC, Squeezebox, Behringer DSP-1124P PEQ, while also improving the quality of sound.

(Also I just saw that Waves Plug-Ins are 40% off through 12/31  -  so, if going that route, now would be a good time to buy)

Jim C

[dlaloum]

PV would be perfect - but installing MAC OS would not be!...

Q10 sounds nice but from everything I can tell it is an FIR EQ rather than an IIR - which means it will not correct phase the way an RIAA EQ should... (sigh)

The RIAA curve is defined both by frequency amplitude and phase shift - most of the EQ's do frequency only..... then there is the debate as to whether it is audible - which is similar to the debate on tone controls.... it sounds nice is all good - but doesn't mean it is a reproduction of the original recording.

If the objective is to first reproduce the original with the best possible fidelity then we need to phase EQ as well as Freq EQ..... - and that is where it is hard to find good software!

The Audacity development team discussed it as part of their EQ development and decided that most EQ users (ie other than vinyl) had no interest in the phase aspects - so they left it out. (Discussion seemed to imply that it was under consideration as a seperate RIAA plugin rather than as an EQ).

I think the same will apply to any development team building an EQ rather than an RIAA module.

It is a shame that PV's attitude to Windows users is "install dual boot mac os" - cos their software is the right stuff.

[analognut]

Hi bunnyma357,

"Pure Vinyl is Currently $229, I got mine on sale for $189".
Apparently I'm confused or there is some misinformation going around or the price has dropped. My impression was the price is in Euro dollars and it works out to something like $500+ American. To get a pro product for under $200 is a great bargain.

"He also raises questions about a PEQ or GEQ using FIR vs. IIR - I don't really understand all this, but it seems he has given it considerable thought."
I'm like you, I don't understand all the techno-speak. I can't figure what you mean by PEQ and GEQ, but I think you're talking about 2 different processors using different ways of doing the calculations, which would be either Fourier or Infinite Impulse Response based. The Help section in one of my audio editors talks about Fourier (FFT) and IIR filters. A de-hiss algorithm states "This function does not use FFTs but instead IIR filters meaning that the transients are perfectly preserved." This implies that FFTs have a tendency to change transients. A graphic eq algorithm states "This is an FFT filter and does therefor not introduce any phase distortion". Another eq which they call "IIR Filter" states "IIR means Infinite Impulse Response. This kind of filter is not FFT based but is applied on each sample at a time. It is very quick. A drawback with IIR filters is that it introduces a slight phase error which can usually be ignored".

So what I get from all this is FFT's can change transients but have no phase distortion. IIR's perfectly preserve transients but introduce slight phase errors which "can usually" be ignored", are quick, and applied to each sample. So we have different ways of accomplishing a task. Ways that have different strengths and weaknesses. I suppose it's kind of pointless for us to worry a lot about this kind of thing. We aren't software designers, we are audiophools with EARS that we use to decide what we like! If one goes out and buys a $50 program to do audio processing, one will end up with CD's that sound like 50 cents.

There is at least one notable exception to this: Audacity. According to the raves I've read from people who use it, it is a truly great program, and it's COMPLETELY free. A quick glance at Audacity's website revealed that it does what you want in a high-quality software.  http://audacity.sourceforge.net/about/features   

[bunnyma357]

Apparently I'm confused or there is some misinformation going around or the price has dropped. My impression was the price is in Euro dollars and it works out to something like $500+ American. To get a pro product for under $200 is a great bargain.

I can't figure what you mean by PEQ and GEQ...

Parametric EQ and Graphic EQ.

As far as I know Pure Vinyl hasn't been priced higher than $229 - but it was already at version 2 when I discovered it, right now the current version is a preview release of version 3, and when the final version is released the price is slated to go up to $299. They also haven't charged for any upgrades to new versions yet.


Jim C

[analognut]

Hi dlaloum-
I didn't see your last post which is right above this for some odd reason. I want to give you my take on your last post. I think you should just use a regular phono preamp after your TT. Then you wouldn't need to lose sleep over IIR's, FFT's, and phase distortions. Once you have the phono preamp RIAA'd signal in the computer you can do the pinknoise/cartridge thing I talked about and you can use a hardware eq like I do, or invest in some more expensive software to get the job done. Take my word for this: you can find the Behringer DSP8000 or DSP 8024 on fleabay for a couple hundred bucks and it WILL get the job done. These are very cool boxes. Not only eq, but you can use them to isolate and eliminate feedback in live applications. You can use them as real-time spectrum analyzers in live and home applications to broadcast pinknoise or whitenoise through your speakers and automatically adjust the eq controls for flat response at the position you have placed the microphone pickup. If you don't have a flat response mic, the unit can even draw the response in the eq and compensate for this if you input the frequency response of your mic. Hard to believe but there is MUCH more- a limiter, parametric eq, etc, etc. I tried the room eq trick for awhile. Very cool. It made individual instruments in orchestral recordings just seem to jump right out at you and you could point your finger at their location. For some reason, not all recordings seemed to improve by this adjustment. I eventually gave up on using it. Simpler might be better, I'm not really sure. You seem like a tinkerer. I think if you could pick up a DSP 8XXX on fleabay you'd get your money's worth. The only downside I would say there is to the unit is the user interface is a little frustrating. Some things are buried one or two steps inside the menu. And the old Green LCD display could be more visible. You need to get your viewing angle right. There is an adjustment for the angle of viewing, but it's easy to get off of ideal when you look at it.

FFTs and IIRs aside, I use my ears as the final judge. The sound of this setup blows me away, and has blown away JE over at AA as well. I'll send you some CDs so you can judge for yourself if you like. 

[dlaloum]

Hi analognut,

that was my original plan - phono pre to adc then perhaps to EQ.... when my ADC went belly up a few weeks back, I started looking around.
What I find difficult with the phono pre's is getting something with the requisite flexibility to allow me to experiment with a series of cartridges.... Which means I need to be able to raise impedance to up to 100k max, and drop capacitance to 100pf min.... and then be able to vary it.

Current products are either fixed (sometimes with a MM/MC switch) - or variable within the MC ranges (low impedance, cap irrelevant) - but without the variability in the MM area.

There also seemed to be some concensus amongst very well regarded recording engineers that the cartridges that got closest to their master tape sound profile were MM/MI.

Also with a focus on digitising old vinyl - I researched stylus types, and obtained a small collection of vintage (80's) cartridges (MM/MI persuasions mostly) with styli ranging from conical through various eliptical radii to Shibata, and MicroRidge/SAS Line contact.

So my search was originally for a phono pre with the necessary features and a high level of transparency... (along with some low c cables..... I have a set of Blue Jeans LC-1's on their way to me).

Then my ADC headed for the hills  - so I was looking for a new ADC.... I've chosen and ordered it (delivery due today) - it seems the MicPre on this may provide an opportunity to try out Digital RIAA - with the potential for saving more than a few bucks if it works well! (The cheapest alternative is the Korora at around $400 - including shipping, Phonomena circa $700, JLTI, KAB Aqvox are all over $1500).

In a purely theoretical sense, working at 24bit recording (yes I know actual resolution is circa 20 bit), with 32bit FP processing - the Digital RIAA option SHOULD provide a truly excellent option...... if I can't get that working the way I want it to - I will probably order the Korora from china - with a custom 100k impedance fitted to the optional slot...

I agree with you about the impact of room equalisation - and I use the Audissey system on my Receiver to handle that. I would prefer the Audissey pro system where I can opt to review check and fine tune the end curve, as opposed to living with the manufacturers choice of a "house curve" - but the same receiver/prepro with that option was double the price (more than $2000 extra!!). The house curves tend to be a touch up in the lows and a touch down in the highs... the standard curve that most custom installers have found consumers prefer.... they are not flat.  (at some stage, I will do some measurement and see whether a tiny tweak will bring it into flat)

The behringers remain on my list of interesting products.... but given that everything goes through my MediaCentre PC, if I can do it in software, that is one less piece of stuff to clutter the room... (I am getting flack from the other half about the 3 TT's !!)
The other EQ box that is very interesting is the MetricHalo LIO-8, apparently they will also modify the Mic Inputs for appropriate Phono loading, and they have already made an RIAA onboard EQ (software) module available. But this is megabuck territory gear. (according to people who heave heard it, very very good) I also looked at the similar MOTU boxes - but there are very variable reports about the quality of their analogue circuits... the digital has a good rep.

Simpler may indeed be better (but then you end up keeping the electronics simple, and messing about hugely with the listening room to get things right.... is it really simpler?) - but with a shared HT / Stereo environment, running through a Digital based Prepro for room EQ.... the purist approach is already far out the window.

Apparently the first AES paper on Digital RIAA was delivered around 1991 - at the time the paper (I havn't got the references) talked about a hybrid system with partial EQ being done analogue and the rest Digital... don't remember the details - but the scientist involved believed 16bit was adequate.
I honestly believe that Digital RIAA is the way forward - it has to be - the real issue is that since the late 80's there has been no major investment in studying this.
So progress is going forward very slowly as tinkerers / hobbyists mess about and gradually develop improved methods. But a lot of the progress comes from technology developed for other areas where investment is booming - studio technology, home studio tools, DJ equipment, home theatre, etc... these are the leggo blocks available to us... are they perfect - not really - but can we use them to build something better than what we currently have at any particular pricepoint.... quite likely!

There are now several cheap USB Turntable interfaces.... the high end is for the most part far too snobby about digital to take it seriously... but when a top end brand like Burmester puts out a Phono stage with digital output (http://www.burmester.de/en/produkte/index.php?product=1,3,52) - the writing is clearly on the wall! (no they are not doing digital RIAA, just a very nice phono with built in ADC)

bye for now

David

[analognut]

dlaloum-

"Which means I need to be able to raise impedance to up to 100k max, and drop capacitance to 100pf min.... and then be able to vary it."
Yeah, see? What did I tell ya? You are a tinkerer. Although I've been doing digital vinyl for many years and seem to have a life-long obsession with recording said vinyl to every format know to man, I only entered the high-end vinyl arena a couple years ago. Prior to then my Pro-ject 2.9 Wood and Ortofon X5-MC were all I used. Then I heard some of JE's CDs and was totally blown away. He took an interest in me and coached me with my purchase and setup of the new table. All this is to say that I don't know much about capacitance or anything that's not related to the lo-output mc's that I now run. So I have no suggestions for you. I use the phono stage of my Yamaha CX-1 preamp, and load it with resistor plugs wired in parallel with the phono input.

"There also seemed to be some concensus amongst very well regarded recording engineers that the cartridges that got closest to their master tape sound profile were MM/MI."
I'm going to blindly guess that's because of the irregular frequency response of many MC carts. I don't have the experience with different carts that you have, but I don't worry about it. JE told me he uses the lo-out MC's because they sound better. And apparently the laws of physics support this idea. I just took his word for it. As for a master tape sound profile I wonder what else the engineers could have meant. I have my Studer A810 calibrated flat. With modern day tapes it records flat up to 16kHz at +4db and from there falls down to approx -1.0db at20kHz.

"In a purely theoretical sense, working at 24bit recording (yes I know actual resolution is circa 20 bit), with 32bit FP processing - the Digital RIAA option SHOULD provide a truly excellent option...... if I can't get that working the way I want it to - I will probably order the Korora from china - with a custom 100k impedance fitted to the optional slot..."
I agree with you wholeheartedly there. Good luck with that endeavor! I did digital RIAA for a couple of years with an inexpensive flat preamp that I got from Tracertek along with their DC6 software. Being able to adjust for the response of my cartridge was the reason I did it. Then I got my new table and arm and everything I had worked on for years sounded so dead compared to the new equipment. So for the last year and at the present I'm working at re-doing my close to 500 LPs, using my CX-1 preamp and having a helluva lotta fun doing it. I'm so pleased with the sound I imagine I'll stick with what I've got for awhile and keep buying more vinyl!

"Apparently the first AES paper on Digital RIAA was delivered around 1991 - at the time the paper (I havn't got the references) talked about a hybrid system with partial EQ being done analogue and the rest Digital... don't remember the details - but the scientist involved believed 16bit was adequate."
Your whole post was an interesting read. My comment here is that I can't believe 16bits is adequate without losing something. Have you ever tried re-sampling a 16bit file? Big difference in quality. Every calculation you make truncates the last few bits, changing the value. Example: Lets represent a 16bit number by using only two integers: divide 6.0 into 7.0. The answer is 1.66666667. Our 2 digit 16bit make-believe calculation is going to round this up to 1.7. This value is way off. I forget how many decimal places correspond to a single bit, but the idea is that 24bits is way more decimal places than 16bits. Enough more that if you downsample the file to 16/44.1 at the end of your processing, no data is lost. Can you process a 24bit file and re-save it as 24bit with no data loss? I could, because I have never heard the difference between a 16bit recording and the same file recorded at 24bits to begin with. Perhaps the scientist you speak of actually meant to refer to a sampling rate of 44.1 when he commented about 16bits. I am guessing you could actually record at 24/44.1 and then downgrade the file to 16/44.1 at the end of processing with no data loss. But hey, when you can get a 1Tb HDD for $100+ bucks, who needs to? 

[dlaloum]

"Your whole post was an interesting read."
Glad you enjoy it ! (it's blather straight from the brain..... so always risky, but makes the 2 grey cells I've got dance)

My comment here is that I can't believe 16bits is adequate without losing something. Have you ever tried re-sampling a 16bit file? Big difference in quality. Every calculation you make truncates the last few bits, changing the value. Example: Lets represent a 16bit number by using only two integers: divide 6.0 into 7.0. The answer is 1.66666667. Our 2 digit 16bit make-believe calculation is going to round this up to 1.2. This value is way off. I forget how many decimal places correspond to a single bit, but the idea is that 24bits is way more decimal places than 16bits. Enough more that if you downsample the file to 16/44.1 at the end of your processing, no data is lost. Can you process a 24bit file and re-save it as 24bit with no data loss? I could, because I have never heard the difference between a 16bit recording and the same file recorded at 24bits to begin with. "

Actually I don't know what he was refering to or the details - I came across the summary of his paper a few weeks back - but getting the actual paper required $ to be spent so I passed!

I agree for processing, we definitely need more bits to give room for it (hence I record at 24bits and process at 32...).

But I still have my Revox B225 - it was in its day considered the best CD player of its generation (1st generation!). - And it does sound great even today - even though it is a 14bit player, non oversampling....

There's a lot of fuss made over bits and bytes - but the evidence of listening to the B225 tells me very clearly that execution is EVERYTHING.... sure these were hand selected matched DAC's, and nothing but the best in the analogue section too... I used to sell these around 1985 - and at the time they were $2500 - Aus$ of course. (which reminds me - I need to find a new home for mine.... no longer play CD's everything is on the server!)

The similarly chipped MarantZ CD44/Philips 104 (clone badge engineering) - used almost identical circuits - with cheap components and construction.... they were a modders delight!

Also some of the early CD's were very well Mastered.... and even though they were only using 14bits in most of the first generation players, the results were (are) quite stunning. - I think the Mastering was all done at 16bits, and studio gear quickly moved to 18 and 20 bit resolution for processing purposes.... By the early 90's most studio gear was up around 20 or 24 bits.

This is all very very digital for a "Vinyl Circle" - but the reason Vinyl is so interesting has a lot to do with the rubbish quality of most of the post 1995 mastering of recordings.

We can get recordings from the 50's and 60's which may have tape hiss present, but the overall dynamics are far superior to the compressed "fast food" served today.

I'm blathering again.... gives me a break from setting up my new ADC - all the recabling, installation, trip to electronics store for part, come back discover I'm out of solder, go back to store... get solder, soldering of impedance plugs etc has taken me most of the day.... and I still havn't tested it. I'm now up to driver installation (all the hardware is in place)...

bye for now

David

[dlaloum]

Update:

ADC (1616m) up and running, TT connected to HiZ input, with Impedance plugs soldered and working well...

I dowloaded a series of EQ VST's (jiiteepee's recommended list) - and have them hosted in SAVIHost - so I have the 1616m patchmix application do a send/return to and from SAVIHost and the selected VST.

So far it seems that (outside of PV & a MAC environment) only jiiteepee's Max/MSP module covers all the bases....

Having it integrating into the sound card drivers means that I can loop it straight out for listening - ie the PC becomes the Phono stage.... with the potential for further EQ when I get there...

Still not happy with the setup - but I have seen (heard) it do what I was aiming for. - So this is not just theoretical, but practically possible - whether it is easy to use and user friendly is a completely different question. For Starters e-Mu's Patchmix DSP application is NOT user friendly.... but it is functionally rich - and seems to work as advertised.... once you work out how to make it do what you want!

If anyone out there has already trodden this path - some pointers would be appreciated.... otherwise I will publish some signposts along the path I am meandering down for others.... (2 days of messing about, very little listening done!)