Best way of recording vinyl to pc

[sbo6]

not trying to be a contrarian, but these days with so much streaming content out there for the taking, my expectation would be that about 95% of any vinyl you could rip would be out there accessible from a streaming platform; and likely be tape based, so at least having equivalent performance to your particular vinyl set-up.

Mike, when you say 95% of any vinyl is out there and accessible from a streaming platform, can you elaborate?

IME, some of the best quality vinyl is / has been made available via Mobile Fidelity, Impex, Analog Productions, etc., most of which isn't remastered and available to purchase digitally, disc and/ or streaming. To me, that's the real benefit of vinyl these days is to get the best version of rock and jazz in particular, being remastered to yield the best sonics via aforementioned companies and others. That best version, I think others would be interested in ripping to digital for convenience's sake.

 

[audiobomber]

IME, remasters are too often inferior to the original, whether the original was on vinyl or CD.

 

[mtemur]

I hear significant improvements as the sample rate rises, by which I mean the digital rip sounds ever closer to the native vinyl in my system the higher the sampling rate goes. I've done tests where I rip and 768/24 and then downsample (so there is only one "read" of the record) to the various sample rates and tested. And I've done multiple rips of the same record at various rates all the way up to 768/24 (accepting there may be variation between plays) and I hear the same thing both times - the higher the sample rate the closer it sounds to native vinyl.

You're right. I made a couple of rips using VinylStudio Pro and compared it to Izotope RX record function. VinylStudio might be just a smidge clearer. More importantly there is a noticeable difference between 768/32 and 96/24. There is a slight air and freshness with 768KHz that lower sampling rates cannot capture. Unfortunately it loses all that magic when you resample to lower, 96KHz and/or dithered to lower bit rate. 4X DSD recording and playback isn't as good as 768KHz either.

This also proves any digital process is audible and smears the SQ. Thanks for the tip.

 

[rDin]

This also proves any digital process is audible and smears the SQ. Thanks for the tip.

You're welcome!

For me, when doing these comparisons, the most obvious differences are in the air and the space around the instruments. As the resolution climbs, the air, the acoustic space, becomes more obvious - you start to see the space between instruments clearer, and finally into the depths and behind them - it all becomes 3D. And this is critical information too. As the resolution drops the acoustic space suffers first, you lose the "visible" air and space, and when you jump back to 44.1 it's obvious - the acoustic space is smeared out and *everything* becomes "out of focus" and "wider" - for instance vocals - at high resolutions you can almost see the lips of the singer focussed in space in front of you, at 44.1 the voice becomes wide and out of focus - a blurred presentation of a singer.

As to why? Well, my conclusion is that has to be reconstruction. If digital reconstruction was perfect you would be able to take a higher resolution and downsample it and still reconstruct it back to the original sound. This works in sampling theory, but not yet in physical DACs. Information gets lost at lower sampling rates - and we can clearly hear this. This is why HQPlayer and Chord MScaler throw millions of taps at the problem to try and get better reconstruction, which subjectively they do, but still not perfect reconstruction. So while we wait for implementation to catch up with theory we can assist by using higher resolutions which helps make reconstruction less error prone...

And if we are brute forcing sampling rates to solve reconstruction, then I feel that 768 is also not quite there. Perhaps 1536 would be 100% transparent?!

 

[mtemur]

And if we are brute forcing sampling rates to solve reconstruction, then I feel that 768 is also not quite there. Perhaps 1536 would be 100% transparent?!

IMHO/IME 768K isn't transparent enough and there is still a huge difference in comparison to original analog. The difference is higher than the difference between sample rates. I don't think 1536K will be %100 transparent.

This is why HQPlayer and Chord MScaler throw millions of taps at the problem to try and get better reconstruction, which subjectively they do, but still not perfect reconstruction. So while we wait for implementation to catch up with theory we can assist by using higher resolutions which helps make reconstruction less error prone...

I use Audirvana and it makes a huge difference at upscaling/upsampling using a minimum phase filter. There is still a slight, but noticeable difference between 96K upsampled to 768K and native 768K. I tried HQPlayer before but I liked Audirvana more. It sounds livelier to my ears. BTW it's Audirvana 3.50 not studio.

 

[Rexp]

IMHO/IME 768K isn't transparent enough and there is still a huge difference in comparison to original analog. The difference is higher than the difference between sample rates. I don't think 1536K will be %100 transparent.


I use Audirvana and it makes a huge difference at upscaling/upsampling using a minimum phase filter. There is still a slight, but noticeable difference between 96K upsampled 768K and native 768K. I tried HQPlayer before but I liked Audirvana more. It sounds livelier to my ears. BTW it's Audirvana 3.50 not studio.

Thanks for putting it into perspective, I thought for a moment the holy grail of the perfect analog capture device had been found...have you tried DSD or any early ADC's that didn't use delta-sigma chips?

 

[mtemur]

Thanks for putting it into perspective, I thought for a moment the holy grail of the perfect analog capture device had been found...have you tried DSD or any early ADC's that didn't use delta-sigma chips?

I tried DSD. IME it is not as transparent as 768K but I haven’t tried early ADCs before delta sigma chips. YMMV.

 

[audiobomber]

I tried HQPlayer before but I liked Audirvana more. It sounds livelier to my ears.

There are about half a million filter options in HQP.

 

[Don_Camillo]

not trying to be a contrarian, but these days with so much streaming content out there for the taking, my expectation would be that about 95% of any vinyl you could rip would be out there accessible from a streaming platform

and >99% of them can be found on CD at sensationally low prices.

Years ago, I bought an M-Audio Audiophile 24/96 PCI for ripping, which no longer works on modern motherboards. That's why I'm looking for a modern audio interface and am torn between internal and external, but I won´t go over 24/192 kHz. For me, the digital world has far too little value and I still prefer listening to a record much more.

Nevertheless, instead of pondering the best sampling rate, I think it's much more important to establish the key points, which, in my opinion, would be:

- Separate D/A converter chips for recording and playback. Gaming sound cards, for example, can convert D to A very well, but not A to D, so you should definitely pay attention to this feature.
- The signal-to-noise ratio should be >110 dB, as this benefits the dynamic range and resolution.
- If the audio interface provides a phono input, it must be adjustable to the cartridge to ensure a linear frequency response, or at least the same frequency response as the normally used phono stage; otherwise, the recording will sound significantly different than normal.
If the normally used phono stage is connected to the audio interface, the input impedance of the sound card and the cable used should match that of the normally used preamp and cable, as otherwise you may get a different sound than you are used to.

In terms of software, I have always used Audiograbber Pro with the professional filters from Algorithmix. The filters are better than any others I have tried. Audiograbber automatically detects the pauses between tracks, for example.

 

[Bruce B]

Here's my 2cents...

I've been involved with mastering from vinyl on quite a few projects and have found the more emphasis you put on set up, the less you'll have to work in Post .... De-noise, ... De-click... etc. I've had to, on a couple of occasions, even play the vinyl wet!
My workflow is: Coralstone... Series V... SME 30... Doshi... Merging Horus -> Pyramix

As Mike would remember, we even did a CD for FIM using his room/equipment comparing cartridges.

Jun Fukamachi - At Steinway

First we had to figure out which A/D to use. Pacific Microsonics against a Merging/DAD converter. All preferred the Merging (except the owner of the PM).

If you listen to the above video, she does have the Mytek converter, but the label says "Hapi".

I feel I ... and most other engineers, can make a digital copy of vinyl and during A/B listening, you can't tell the source....

 

[Mike Lavigne]

As Mike would remember, we even did a CD for FIM using his room/equipment comparing cartridges.

Jun Fukamachi - At Steinway

First we had to figure out which A/D to use. Pacific Microsonics against a Merging/DAD converter. All preferred the Merging (except the owner of the PM).

that was a fun session. i still miss Winston.

I feel I ... and most other engineers, can make a digital copy of vinyl and during A/B listening, you can't tell the source....

with your mixing board and pro tools all things are possible. not real world for hobbyist vinyl rippers regardless of talents. and Merging is pretty spendy too. tape transfers and streaming are better choices for most. especially if you are starting out and have not already invested in the gear. money better placed else ware.

 

[audiobomber]

I've been involved with mastering from vinyl on quite a few projects and have found the more emphasis you put on set up, the less you'll have to work in Post .... De-noise, ... De-click... etc.

Which reminds me, warm up your phono system prior to recording if at all possible. I found it took around fifteen minutes of play to get everything up to snuff.

 

[rDin]

I feel I ... and most other engineers, can make a digital copy of vinyl and during A/B listening, you can't tell the source....

What minimum sampling rate would you say is required for transparent capture?

 

[Bruce B]

What minimum sampling rate would you say is required for transparent capture?

DSD128

 

[rDin]

DSD128

So that's roughly equivalent to PCM 368? Very interesting, thanks.

 

[Mike Lavigne]

So that's roughly equivalent to PCM 368? Very interesting, thanks.

to my ears dsd128 is more like PCM 24/192. i have lots of files of each. i have lots of dxd PCM (24/352) and dsd256 which to my ears are equivalent. mostly i prefer the native transfer format for sound quality.....which is more significant than the particular format. more math applied to the file is not your friend to my ears.

but at these format levels your DAC, sever, and ADC are going to be pretty significant in terms of what sounds best. and what amount of data storage makes sense. higher is not necessarily better.......just has a higher theoretical ceiling for less noise if your digital chain is low enough noise to take advantage of it.

 

[DasguteOhr]

DSD is only worthwhile if your AD converters use the delta-sigma principle, as then bit word lengths don't need to be converted, resulting in fewer losses in the summing and noise shaping filters. You then practically have a 1-bit DSD stream.

 

[Rexp]

DSD is only worthwhile if your AD converters use the delta-sigma principle, as then bit word lengths don't need to be converted, resulting in fewer losses in the summing and noise shaping filters. You then practically have a 1-bit DSD stream.View attachment 150186

Interesting, have you ripped any records using such an ADC and if so, which ADC?

 

[DasguteOhr]

Interesting, have you ripped any records using such an ADC and if so, which ADC?

No ..only heard it sound really good in my ears.
Korg mr series does it exsample.

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[Don_Camillo]

DSD is only worthwhile if your AD converters use the delta-sigma principle, as then bit word lengths don't need to be converted, resulting in fewer losses in the summing and noise shaping filters. You then practically have a 1-bit DSD stream.View attachment 150186

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