New User for using VS to add to my LMS system

[Ecoli-557]

Hello All, very New User here.
I have VS Premium V14.0.4 and plan on using it to add to my audio files on lyrion Music Server (formally known as Logitech Music Server).
I have several question and pardon if this is the wrong place to ask.
First of all it seems so very cool and I have played with it for a week now but am wanting to get serious with it.
I am using a, ADL preamp/digitizer which works well - after trying serval cartridges for the best 'sound'.
Work Flow:
1. Digitize both sides and then
2. Split tracks (wish this was more accurate its works pretty good)
3. Cleanup Audio - this is where I have questions - of course.
I understand the pop and crack part, but the equalizer I would like to use.  I do this AFTER the recording but does this get saved when saving the tracks?  It seems weird to Slit Tracks, then Cleanup Audio, then go BACK to Split Tracks in order to save them.

Last question for now (I suppose), is there a rule of thumb on how much to add on the low end from an album?  The cuts can not get the full base (RIAA help) and even though everyone's taste for the low end is different, is there an perceived 'average' people use?
My system is all Bob Carver except for the graphic equalizer bot tube and solid state with 4 Carver AL3+ speakers so the sound stage is excellent according to other audiophiles who have listened in.

I am asking quite basic questions on a totally subjective topic but I could use a starting place.

Great program and I plan on adding hundreds of albums and only play them just to see the Michell go around......

Spin an album with an Adult Beverage........

[Ecoli-557]

Also, anyone here managed to recreate the 'warm tone' of an album while converting to digital?
I am using FLAC, 24 bits, and 96KHz sampling.  My equalizer is:
31Hz - +9
63hZ - +5
125hZ - +5
250hZ - +8
500hZ +9
1khZ - +9
2kHZ - +6
4kHZ - +3
8kHZ - +1
16khZ - +5
Normalize at -1dB

This gives me in my and a few friends estimation about 75% of the sound from vinyl.
There is a difference in my cartridges; table has a MC and the recording table has a MM.
I am wondering if I need to kick up my sampling rate?
Anyone who has re-created the vinyl 'sound' please weigh in.
Regards to All.
 

[Indy33]

I'm surprised that Paul has not responded by now, maybe he is out of pocket right now.

After reading thru both your posts and the perceived quality of your system, I'm surprised you are not recording to a DSD file. But maybe it's because you can't edit a DSD file and then save it as such.

If the vinyl surface quality is good enough, try recording to DSD.
.

[Ecoli-557]

Thanks Indy33.  I have asked this question on a Facebook group which is international in scope.  Interesting responses to say the least as well as informational.
One guy who is a professional sound engineer for movies, etc. has given good information.
He is saying that 24 bit will not get me close to my goal - which is what I am 'hearing'.
He uses a RME ADI-2/4 SE unit which uses 32 bit floating.  He further states that a MC is not needed.
I have found a 32-bit floating ADC; Zoom UAC-232 and have asked him if he has had any experience with it - awaiting his response.
He says that the bandwidth of a 32-bit floating unit will get me 'just a hair below live listening'.....
He further states that 96KHz and 24 bit tops out in terms of frequency response and low signal resolution.
My wonder is if VS can work with either the RME unit or a lower cost but still 32-bit floater such as the Zoom UAC-232?
Hopefully someone will know.

[Indy33]

My wonder is if VS can work with either the RME unit or a lower cost but still 32-bit floater such as the Zoom UAC-232?
Hopefully someone will know.

Looks like it should work according to this website.

https://www.alpinesoft.co.uk/VinylStudio/screenshots.aspx

Record directly to WAV, FLAC, Ogg, MP3, or AIFF format.

On the Mac, CAF, AAC and Apple Lossless files are also supported.
Recording stops automatically at the end of the record

Supports up to 192 kHz, 16, 24 or 32 bit

Monitor your recording through your PC's speakers

Record 78's on turntables with no 78 RPM speed setting

Apply RIAA and other recording equalisation curves

Lookup and edit the track listing while recording is in progress

This also might be of interest.

https://www.alpinesoft.co.uk/VinylStudio/dsd.aspx

As of version 8.6, VinylStudio can record DSD from a suitable ADC, such as the PS Audio NuWave Phono Converter or (with a firmware upgrade) the Ayre QA-9. These are DSD over PCM (DoP) devices. Under Windows, 'DSD native' recording is also supported; devices tested include the Korg DS-DAC-10R (DSD supported on Windows only), the RME ADI-2 Pro, Mytek's Brooklyn ADC, the Ayre QA9, Pro-Ject's Debut Carbon RecordMaster HiRes Turntable and Playback Designs' Pinot ADC.
.

[LtMandella]

[[Also, anyone here managed to recreate the 'warm tone' of an album while converting to digital?]]

Try digitizing to DSD.  That worked for me to avoid digital "harshness"...

[Ecoli-557]

LtMandella, how does DSD compare to listening to the same cut of music on vinyl?

[LtMandella]

I find DSD to be consistently less harsh and "scratchy" sounding in high volume mids and highs vs. 24x96 flac or wave.

Take a listen sometime to Springsteen's "She's the One".  The background rythym guitar on all the
 PCM digital tracks I have heard sounds more like hiss than guitar chords. 

But when I ripped to DSD I hear the tone of a chord, not just scratchiness.  Sony's commercial DSD discs sound overall much "smoother" than CDs.  I have the Dylan SACD box set and the smoothness is definitely there.  Some complain that SACDs are too laid back, but I love it.

[Ecoli-557]

LtMandella, I just downloaded the beta update and will try it.  I have a SACD player and are familiar with its sound.  There HAS to be a way to do this says a retired electronics design engineer!

I have looked for a DSD file option and do not see it, where IS it located?
Never mind - its DSF/DSD.  Got it.

[Indy33]

There are two types of DSD files. DSF and DFF file extensions, but only the DSF file has metadata. I don't know why anyone would use the DFF type.

Normally you can't edit a DSD files, except to break it into tracks. But if there are only a few clicks/pops then I will "cut out" the few milliseconds containing the click so I can retain the DSD format. But at some point this is just too time consuming.
.

[LtMandella]

I did not say DSD was better for editing.  Far from it.  However the digital grit I often hear with PCM is aggravating enough to me at this point I would _much_ rather hear a few clicks and pops (I don't get many of those after I started using an Ultrasonic cleaner) than hear grit as almost the norm.

[Indy33]

I did not say DSD was better for editing.  Far from it.  H

I don't know why you thought I was talking to you, your post was not even directly above mine. I was just adding information to the thread.

Sorry if I offended you.

[Ecoli-557]

I appreciate everyone helping in this - a wee bit more complicated than I had given thought.
As a newbie, DSD has more 'sound stage presence' possibly but, at the risk of not being able to edit the pops and clicks or any equalizing?  Do I have this correct?
LtMandella, I too have an ultrasonic cleaner and while most are fairly clean, there are a few that get in...... will try it however without editing.
Indy33, thanks for the notion to 'cut out' microsecond clicks/pops - may try that as well.

As you both understand this much better than me, why are we not able to edit DSD?  I still just do not have the big picture even after looking online.

[Ecoli-557]

Rats.
My ADL GT40 apparently does NOT do DSD.
Back to the drawing board.......

[LtMandella]

why are we not able to edit DSD?

My limited understanding is that each encoded value in DSD is computed from the previous moment's encoded value.  It is not simply a "snapshot" if you will of whatever waveforms were detected at any moment.

So cutting and pasting of encoded moments ends up being gibberish.