In perusing and experimenting with various methods of doing this, I have occasionally come across the idea (which I have not tried) of recording direct without RIAA equalisation and then applying that equalisation in the digital domain.

I would be interested, as I imagine, would be others, in Grahams thoughts on this approach and its merits or failings.

Hmmmmmmmm, I have been pretty satisfied with the results of feeding the Reflex into an M-audio firewire interface (though not as good as just playing the record) - but being at a loose end for a day or two  I have taken it into my foolish mind to tinker some more.

Source of the question was the Terratec DMX 6 Fire and Phase 26 USB both of which I own (as also the newer - "I want" DMX 6 Fire USB {24/192 I/O, 48V Phantom power for Mic all singing all dancing}), appear, from the rather scant documentation, to utilise this method.

Specs quote the phono inputs as 47Kohm impedance and sensitivity to suit MM or HOMC cartridges, there is a button on the control panel (software) for "RIAA Filter" but the rather cryptic documentation hints that this is for monitoring only and the "howto" guides one to use the provided Wavelab Lite software to apply RIAA correction using a .dll provided as a post-processing filter.

Down the rabbit hole we go - using ASIO drivers is a "good thing", but being Steinberg proprietary, not all software can use them, notably the freeware Audacity (Steinbergs licensing conflicts with the Audacity license) BUT - it is possible to "roll your own" for personal use (I believe).

My usual preferred, minimalist recording software, offers auto level setting, auto start and auto stop - all quite handy in a stripped down interface uncluttered with all the paraphernalia of Audition or Sound Forge and the like.  Sadly, it does not support ASIO nor high bitrates, which I rather prefer for initial "takes", no harm in down sampling to 16/44 later.

Why now ? - some other darn fool wanted to do some archiving and this one opened its big yap and volunteered to help out - I predict much regret of this impetuousness in the coming days.

Trouble with this stuff and software and 'puters in general, it is the stuff they don't tell you that you REALLY DO need to know to get it all to work and the only way to find out that stuff is to try all the wrong ways and any other avenues of frustration that offer themselves in the process.

Probably not much different to designing phono stages and headphone amps and such eh? 

That's about enuff from Groucho as he goes orf to make 'is Marx -  - maybe this should have gone in Krabby Patty.

Lang may yer lum reek.

Oh BTB, had the lid off the Phase 26 (don't ask) and was impressed to see what appeared to be a decent looking digital pulse transformer on the coax out.

Let you all know how it went when I am done driving myself further round the bend. 

Finally used the Phase 26 USB to actually record a few takes of the same album today.

Using the Technics with a DL110 and 1Kohm parallel load into the phono inputs of the unit.

Using Sony Sound Forge Studio as the only software I had that could both use ASIO drivers and VST plugins, other software that looked promising could not use ASIO drivers or was looking like taking too long to work out how to use it (Ableton seemed to have a better interface to both the driver and the plugin but is geared to multitrack recording and mixing).

Whatever, sound level good, without clipping being apparent, so boosted BUT NOT EQUALISED.

This was quite easy to hear on playback, this is an album I already had a fairly good recording of to use as a reference.

Ran the supplied RIAA plugin over the file et voila - does the business.

Much fooling around with different post-processing and different sample rates, 24/88.2 and 16/44.1 both tried.  Results yet to be fully evaluated, but appears to work as described, monitoring during recording via Solo/K701 did not indicate the "tilted" sound that could be heard on playback.

Still seems really odd that they should Eq the direct hardware path from input to output but not the recorded signal sent back via USB.

Edited the above following further more careful evaluation.

As Alice said, "it gets curiouser and curiouser" - stay tuned for further inciting instalments.

Nearly forgot, Graham, that is very interesting what you have done to the Genera output, I am so glad I no longer have the one I built, I would likely think I needed to try that too. 

One unexpected outcome from the above fooling around, I noted the output of one channel to be perhaps -3dB cf the other over several different records.

A careful check of azimuth setting on the headshell in question, followed by a slight adjustment (the headshell offers azimuth adjustment) resulted in nearly perfect channel balance.

With, naturally, a rather better centred image (although that is often suspect in my system due to speaker placement constraints - one reason that I like - where possible - to have a balance control).

The wonders of technology 

Of course I had to then re-record the sides with mismatch 

The hardest part of this whole exercise has been achieving an input level that does not clip.

I would much prefer to use the Reflex for phono pre duties but cannot lower the output sufficiently with means at hand (available preamps are not sufficiently transparent and tend to remove most of the advantage in SQ of using the Reflex) - the latest - USB2 capable - interface from Terratec offers onboard gain controls which might be good.

Perhaps the Aria will be available soon and might be OK for the purpose.

Interesting find on tnt-audio  http://www.tnt-audio.com/sorgenti/rip_it_4_e.html" rel="nofollow - http://www.tnt-audio.com/sorgenti/rip_it_4_e.html a 4 part series covering various experiments over a number of years.

He has tried a few of the same pieces of gear that I have although ending with a rather more expensive solution.

He also identified the same input level control issue and recommends the same click removal software that I prefer.

Of particular interest, to me at least and hopefully to anyone else reading this, the page linked covers various software applications available, with some testing methodology for their accuracy and usage, most especially WRT sample rate conversion performance.

There is also a deal of useful information concerning the use of higher accuracy sampling rates during the digital processing phase, prior to downsampling to Redbook and the desirability of doing all digital manipulation in 32bit floating point.  I found this information both readily accessible and easily digested, there are also links to further reading on the subject.

If there is interest, perhaps this thread could move in the direction of some "how to" type tips.

My first in that direction focuses on the source, GIGO is very apt here and the first step to satisfying results is the cleanest possible record that can be made from what is to be recorded.

This applies regardless of whether the record is new or used.

Wash the record, play it a couple of times and check for improvement in SQ, rewash if it is felt that further improvement may be possible (for new or mint condition records NO extraneous noise should be acceptable), play at least twice more.

During replay after cleaning, there will be noted a continuing improving in SQ for the first 3 - 4 plays, this may be, in part, due to the stylus removing "gunk" loosened by the washing, but I feel there is another mechanism in play here and it is referred to in the excellent articles by Rudolf Bruil on his website http://www.soundfountain.com/amb/rc1.html" rel="nofollow - http://www.soundfountain.com/amb/rc1.html  

Following the cleaning and replay routine, I then apply Last record preservative, which, to my hearing, does "what it says on the tin" - providing a fuller and smoother sound, exactly as one might expect from the stylus pressing against/being moved by, a less yielding surface.

All that and not a computer in sight yet 

Who knows, do all that and you might forget the idea of trying to digitise them altogether.

If not, watch for further installments.

Aside from the various RCMs available at prices from a couple of hundred dollars to many thousands of dollars, there are various manual or semi-manual methods of cleaning records.

The method with which I have been happy for several years combines products from 2 different suppliers.

I use the hardware (trough, drying rack & record label clamp) from the Knosti system, which I tried but was not completely happy with, as a post wash rinse bath.  Use distilled or filtered water in the trough and tip it out after 10 - 15 records or after a cleaning session.

The actual cleaning I do using the Disc Dr cleaning solution and brushes, the brushes have replaceable pads and will last a long time, you may prefer a DIY cleaning solution, there are many recipes to be found for that.  

I clean one side, turn over and clean the other side, then rinse and air dry.  

The record is laid on a soft and fully supportive surface such as a clean towel or similar on a table top during the cleaning and held down to the surface by manual pressure on the label area whilst "scrubbing", the majority of the cleaning solution is removed from the surface using the cleaning brush prior to turning over, at which stage the surface will be moist rather than flowing wet.

Occasionally there will be sticky muck on a record that may require solvent removal, isopropyl alcohol (or even methylated spirit) will often loosen these.  The record is then washed with the normal cleaning solution, rinsed and air dried.

Where records have only a paper sleeve, I will put them in a good quality poly sleeve after cleaning, since paper sleeves will abrade and leave mould feeding debris in the grooves.

I keep on hand a pack of new sleeves for such replacements and for replacement of any no longer serviceable poly sleeves from S/H acquisitions.

There have been articles on the net WRT the use of small portable manual steam cleaners for record cleaning, this is not something I have tried, but could be worth investigation with the obvious proviso that this method may entail some risk if insufficient thought were given to the possible effects of over enthusiastic application.  eg HOT steam + vinyl = potential damage.

I also regularly use a carbon fibre brush to remove static and airborne dust from the record surface immediately prior to play, a light touch on the turning record combined with a rolling action of the brush will lift most of this.

For records holding a static charge, I have found the Milty anti-static pistol effective, this is not commonly a problem for me here and is more common in air-conditioned or centrally heated environments I think, decent anti-static sleeves are also a help.

The bath tub idea might be a lot of fun but could introduce some undesirable factors detrimental to  optimal sound quality, aside from hazards to baby, bathwater and the record label.  

Will get round to some of the hardware stuff next.

Gary,

not to put the dampener on your technical enthusiasm at all, but my reference for improved quality of vinyl digitisation is the original recording.

If I play both on my system I can tell how good my rip (and my DAC) are.

I am not sure how one might use spectral analysis for relative SQ analysis.

If I cannot hear an improvement when I play back and compare to the original, then, to all practical purposes there is none.

If it sounds better (than the original) then there is a problem 

Have fun anyway.

Basically I prefer the attenuator -> buffer arrangement however it might be achieved, this does provide the best impedance matching between devices.

I have long been interested in the passive, buffered linestage in the following article -  http://www.stereophile.com/solidpreamps/54/index.html - http://www.stereophile.com/solidpreamps/54/index.html - unfortunately a number of the components are now no longer available and I have not taken the time to pursue suitable alternatives, but I have  seriously considered that this might be a solution to the preamp issues I have noted above.

In all of this, I am trying to keep the focus at the enthusiast level of equipment and options and not considering more professional grade equipment that might be beyond the budget of many (me included), particularly for the occasional project.

Edited for inclusion of additional content and to correct some spelling.

tg,

 I agree, based on my experience digitizing audio over the past 10 years or so, obtaining an acceptable input signal for recording is the greatest challenge. I have worked with tapes, cassette and reel to reel; and vinyl using various ADCs with a constant issue related to input signal level and noise.

 My experience has convinced me, it is an impedance mismatch issue between our phono preamp outputs and whatever type of ADC we connect with. Your point about the buffer is a good approach to try. It could provide the impedance match or correction allowing signal from your phono to fit, so to speak, nicely into the window provided by your ADC, therefore cleanly digitizing the signal maintaining minimal loss.

 I have been trying to find the matching capability with the output of the phono preamp. What we do, digitizing vinyl, does not generate enough interest in the marketplace for a vendor to spend their resources developing products specifically for our application, especially within your “enthusiast level” focus. Therefore, it is up to us to find the solution and apply it.  Just look at the compromises we make to do this, yes I appreciate the challenge, but I also have years’ worth of vinyl to deal with.  I have modified my phono preamps over the years as my ADCs have changed, but I’ve never felt confident I ‘m getting the best sound.

 I have been very interested in Graham’s discussions concerning a phono preamps design and the theory behind it. His insight has helped answer numerous questions I have had over the past years concerning problems with level matching between my phono output and the ADC.

 I look forward to discussing this down the road.

Tom

Hi Tom, welcome to this forum.

It will be good to have some more contribution on this topic.

enjoy your time here,

Tony G

tg wrote: In perusing and experimenting with various methods of doing this, I have occasionally come across the idea (which I have not tried) of recording direct without RIAA equalisation and then applying that equalisation in the digital domain. I would be interested, as I imagine, would be others, in Grahams thoughts on this approach and its merits or failings.

The output of a magnetic cartridge as established already is a rising one where the highest frequency in the accepted audio range - 20Hz to 20kHz has 1,000 times the amplitude of the lowest frequency. Because of the "tuck and nip" of the RIAA recording curve the actual output playing a record will give a range of 100:1, or in decibels: 40dB.

The analogue input to an ADC (analogue to digital converter) chip is 2 volts peak to peak in the case of a Texas Instruments PCM2902. In r.m.s. that is approximately 700mV. It should be similar for most chips as all now seem to work on a 3.3V supply.

700mV r.m.s. is therefore the maximum the ADC input will take before clipping. The input would have to be adjusted down to cater for the dynamic peaks of the record. If reduced by -14dB which seems to be the "pre loudness wars" input level, the input at -14dB is 140mV.

140mV is therefore the input level to the ADC input at 20kHz and therefore the input at 20Hz is 1/100th of 140mV which is 1.4mV. In decibels that is -54dB.

If the dynamic range of the ADC is 16 bits there is 96dB dynamic range. At -54dB there is 42dB left.

In this 42dB has to fit the low frequency (20Hz) dynamic range less the peaks. Beyond that is the noise floor which is different from an analogue noise floor - in digital nothing can exist at or lower than the noise floor. If 42dB is considered sufficient then noise wise the idea seems OK.

However, please do not forget that the distortion increases in proportion with falling digital signal level and may be in the region of 10% on the quietest lowest bass notes, whilst being much more respectable as frequency increases.

 I have tried an unequalized phono preamp for digitizing vinyl. Adjusting the input levels on my ADC was really a pain. Once I found a good setting, the digitized audio had high frequency distortion. Suspecting an overdrive condition, I reduced the input gain only to find that when the high frequencies were better, but the low frequencies were muddy. After Graham’s explanation it seems maybe the phono preamp was producing a distorted output. The phono preamp manufacturer blamed it on the ADC input circuit being too sensitive. Using a properly equalized phono preamp produces acceptable results. Actually, it was the same device, a switch produced the "flat" circuit vs. the "normal" circuit.

 It seems that just as much care must be given to the design of these unequalized preamps to handle what is basically a highly complex signal as is required of equalized preamps. Something about if it’s not broken don’t fix it, or a solution looking for a problem. I do understand the claims that a software program can be consistent and doesn’t drift, etc., but the original method works just fine with properly designed phono preamps. My software screws around with my sound just about enough as it is.

 Tom

Graham, I was just wondering depending on how the switch activated removal of the equalization circuit was placed in the circuit could cause an output impedance change such that it could cause distortion to the signal. Your explanation with actual electrical theory as apposed the more main stream explanations is refreshing and I now understand what the problems are.

 Could the "100:1 level difference" problem be solved with a phone preamp integrated into the ADC?

Tom

Long time with no input to this thread, I have now moved it to the more recently created digital audio forum under the appropriate heading. 

Continuing on from the last couple of posts, one of the interfaces (2 versions actually) I use are from Terratec and provide a phono specific input and a software plugin to apply the RIAA eq post recording.

At a period earlier this year, I made a number of recordings using this input method with "acceptable" results, it did seem that the output of the Denon DL-110 I was using with it was suited to its input requirements and clipping was not an issue. Higher output cartridges were problematic.
This was prior to my modification of the buffer, which, until now, I have not tried in its intended application.
I have now had opportunity to try the buffered approach on 9 LPs and am rather more happy with the results.
Using the Reflex into the buffer which then feeds an M-Audio Firewire Audiophile 24/96.

On one compilation album showing marked level variation from one track to another, I found I was able to switch attenuation up or down between tracks, while recording, with no switching noise and level the whole album rather effectively.
Admittedly not on the first take.

I have virtually settled on using Sound Forge Audio Studio for recording, it has a number of features I find useful, it can use ASIO drivers, VST plug-ins and set metadata on .wav files to carry useful recording information.
It's handling of metadata is not as complete as dBpoweramp Reference edition, which is also a part of my collection of "must have" audio software.
The recording interface is simple and powerful with many options, the most useful feature being the ability to hold the peak level which will turn red on clipping and stay that way until reset, making it much simpler to run through an album to set level without having to watch it like a hawk.
It also allows very simple compression or expansion of the timeline to view levels of a whole recording in one view or spread a very short interval along the line to aid in delicate cuting or glitch removal.
As part of a video editing package (the HD Platinum Suite), the price is most acceptable for the functionality afforded.

Revisiting the earlier recordings made using the Terratec phono in and comparing them to these made with the buffered arrangement, shows that the Terratec phono applied quite a deal of level reduction in order to achieve its result, in comparison to what was possible using this arrangement.
IOW I was able to record to a higher peak level without clipping and keep the average signal within the recommended range.
Naturally the Reflex provides a much better result with greater dynamics being evident, along with the greater refinement one might expect from a much more accomplished phono stage.
The ADC should be fairly similar as both the Terratec and the M-Audio use the Envy 24 chip and IIRC AKM hard codecs.

A few screenshots illustrating some of the above:

A few more thoughts on this recent exercise when I gather it together.

• Need a FLAT CURVE output and you provided some basic mods to effect same to the GENERA
• Then there is the matter of IMPEDENCE matching/adequate Dynamics
• Next is the issue of Attenuation of  either at the Phono or ADC but at PHONO preferred.
• There is also a debate as to whether the RIAA equalisatio should be added after ADC process.
• There is a debate that the ADC and phono Equalisation/RIAA should be One Unit with a good stepeed attennuator.
• Ther is a Software isse as 'Audacity' is free but with limitations andpoor instructions mong other
• There are other issues but the thread of Forum discussion is split across a number of opics on the Forum.

Regards and enjoy the rain indoors with music.