hi, im transferring a few songs to pc from vinyl.
using technics 1210mk2 a mixer and connected to saffire pro 24 to pc,recording to wavelab 6
the level on mixer not hitting red,
does it matter what level i have mixer at( obviously not hittng red) i can just normalize the file after its in wavelab yeah?

 

Software and Recording Format

Regarding the software we want to use to record, it's fair to say that there's now quite a considerable selection available. Audacity for PC and Mac is free to try. SOUND FORGE Audio Studio from MAGIX (PC only) or Steinberg's WaveLab Elements (PC & Mac) are practical options because they allow recording at 24-bit and 96 kHz. SOUND FORGE PRO and WaveLab Pro are very suitable, although they are admittedly expensive. However, both applications offer countless plug-ins for efficient restoration and are excellent for mastering.

Lossless Uncompressed Recording

Lossless uncompressed recording of PCM data, such as AIFF, WAV, or even BWF, is suitable for recording. Free tools usually only allow recording at 16-bit and 44.1 kHz, which is basically sufficient if you don't want to edit the files afterwards. Anyone who wants to subsequently restore the files will have to use 24-bit and 88.2 or 96 kHz sample rates, and will have to spend between €60 and €100 for this.

The state-of-the-art are definitely the "software battleships." They burn a medium-sized hole in your pocket, but allow recordings at up to 32-bit (floating point) and sample rates of up to 192 kHz or more, which is a huge advantage for mastering the files. They also come with the necessary tools for restoration and mastering.

Digital Recording and Editing

After starting the program, for digital recording, you need to select your connected audio interface as the recording source in the preferences and then create a new "empty" file. You always need to specify the number of channels (2 for stereo), the resolution in bits (at least 24 bits), and the sample rate (at least 88.2 kHz). Now you're ready to go.

Typically, you first need to put the program into record mode. If the interface and application support ASIO, you can now view the level of the incoming signal on the software meter, and the signal will be looped from the input to the output of the audio interface, allowing you to listen to the music on the connected speakers or headphones. I would set the line signal level with a bit of headroom (4-6 dB). I would definitely record the pause before and after the track first; we can always edit the whole thing later.
Once the track is over or the LP is "finished," you can stop recording. Now we should cut out everything except for 5 seconds before/after. Select, delete – done.

Next

... we save the file as a separate audio file. Better safe than sorry. I would save the file with the same key data as preselected during the recording, in this case: WAV, stereo, 24-bit, and 176.4 kHz. From experience, I recommend being specific when naming the file, such as the artist, title, and editing status, but that wasn't done in this case. The latter is missing: a suffix like "_Raw Material" or "_Recording" or something similar would be quite appropriate.

Digitizing Records: DC Offset

In the next step, we remove the DC component, the so-called "DC offset." The electrical signal is nothing other than a resulting AC voltage in which various individual frequencies overlap. Often, pickups, preamplifiers (both phono and line), and even AD converters cause a DC component, which is mathematically simply added to the signal. If you imagine the voltage over time in an X/Y coordinate system (time on the X axis), the resulting curve does not have its origin at the zero point.

This point is then either above the X axis and thus positive, or below it and thus negative. We simply set the curve to the X axis for DC correction. Nothing else happens during this editing step. This function is usually found in the Effects menu or under "Edit." In SOUND FORGE PRO, we find this step under "Process."

To determine the DC component, the application must scan the file once. Basically, it's sufficient to use only the first few seconds for the calculation, because the same DC offset is usually found throughout the entire file. Here, too, the rule is: scan first, then remove by pressing the virtual OK button. And: save!

Once the DC component has been "calculated out," we move on to normalization. Only then is it really worthwhile. During this process, we adjust the recorded file to the values of the specified quantization. Huh!? This means that we use the full 24 bits for the digital representation, or more simply: we digitally adjust the levels. The recording has already been completed, but we can still utilize the entire headroom by subsequently digitally "setting" it to full scale. Yes, it works!
The program bases its work on the highest measured peak, which means that the software has to scan the file completely (this time, really).

Normalization

In MAGIX, we find normalization under "Process." First, select the peak value method, then scan. Now you have to select the desired value. I always use -1 dB, some even choose -3 dB. Broadcasters are particularly cautious and even use -6 dB, but we shouldn't go that far. We don't need to send the file in real time to a satellite or something similar. So: Set to -1 dB and then OK.

Digitizing Records – Target Format

Ultimately, it's important to create a separate file using the "Save As" function, which we can listen to in everyday life. This is entirely individual, and there is no single solution. Some people have a Hi-Res audio server and can listen to the original WAV at 176k and 24-bit over the internal network. Many users, especially my age, would like to burn this to an audio CD, while frequent travelers need a format they can transfer to their iPhone.

In the latter case, I would recommend an AC3 file at 128 kbps; for a large iPod, perhaps an MP3 at 320 kbps. The audio CD requires a WAV or AIFF file at 44.1 kHz and 16-bit. If you'd like to share lossless, high-quality files with your friends on your home server, I recommend FLAC or Ogg Vorbis. So it is rather complicated and depends on the accompanying circumstances – just like in “real” life.

Source / German: https://www.bonedo.de/artikel/vinyl...hbearbeiten-und-mastern-schritt-fuer-schritt/

 

I've done a lot of vinyl recording myself in the past and I came to the conclusion that you want the input signal to be as hot as possible. So that means that the absolute loudest peaks should reach somewhere between 0 and +2 db (Or maybe even +3 sometimes, depending on the type of music?). It then mostly sounds better after removing crackles, hiss and noise etc. But I would say just try it for yourself and record a few samples with various levels and see what works best for you?

 

I've always used SoundForge, it works swell.. also available at sister site at a very reasonable price..
has good tools for filtering/removing noise, clicks,scratches etc.

clean record before ripping vinyl
I use discwasher system... modern equivalents available on ebay/amazon/etc

 

I use Wavelab12 with Technics SL1200MKII into a Tascam US1800 from little mixer. I have anywhere from -6 to -2 with the occasional flash in the red depending on the music and loudness of the vinyl. Adjust to taste. I was recording a couple of Miss Nelson & Bruce Haack LPs last night which are quite quiet so took a couple of goes to get a good balance. As it's just talking with bits of music had to check where it was loud. If its quiet and you keep turning it up you'll start hearing the surface noise and any other artifacts present. Once it sounds ok keep that setting for next time as a starting point.

 

In vinyl digitization, the key, what characterizes the sound is the analog path of the signal: a clean and well preserved vinyl, a needle in good shape, a quality cartridge and good riaa preamps, also avoid placing your monitors anywhere near the turntable. If you can do your best in that, the hard part is done. Your Saffire has good converters and more than 100db on input so you will have perfect recordings with it, stay away from the zero Db, you can normalize afterwards. Generally speaking, I am not a fan of audio restoration, I prefer to have the raw recording without any intervention. In the export of the file it is essential to use a lossless format