Windows 10 (64 bit)
Rega PLANAR 1 Plus
Creative Sound Blaster Z Sound Card
I have the above equipment that I plan to use to digitize some records and I have some questions relating to recording levels and actually listening to the audio I’m recording.
-I made a topic earlier last year on Hydrogen Audio asking how the entire process of recording vinyl records works. The topic in question is here…
One particular suggestion I was given was to record at 24 bit / 96.0 Khz, as this would allow me to record at a much lower volume whilst still being able to use Audacity’s Amplify tool to bring the volume of the highest peaks to 0 db without distortion. More specifically, quoting from the topic…
But if your usb output from the preamp is limited to 16 bits I would personally also see that as a deal breaker, and look to capture in 24 bit.
This is linked to the issue of levels, and I’ll try to explain again. While the goal is get the final version of the file as close to 0 dB as possible to maximize loudness, there are different requirements for the initial capture which basically involve getting the best definition without suffering digital clipping.
There is some legitimate disagreement about this, but for argument sake let’s say @ 16 bits you need to peak at -6 dB to fully capture the resolution (dynamic range) of your vinyl, with the wind behind on a sunny day, etc.
But as mentioned @ 24 bit resolution you can peak at significantly lower levels and still capture all the detail needed to digitally amplify the signal without any loss of quality when post processing “inside the box” say in Audacity.
This distinction is important for real world applications, because if you are aiming to peak at -6 dB @ 16bit resolution you have a relatively small margin of error to avoid clipping at signal levels that exceed 0 dB.
For example you may not have leveled your input perfectly and find that you’re actually peaking closer to -4 to -3 dB when playing main sections of tracks rather than just short sections at the beginning when spot checking levels before capture (otherwise you would have to play the record twice to precisely measure your peak to calibrate off). That could then lead to clipping if there is suddenly a much louder section or say if there is a pop or a scratch, and you would then have compromised audio that should be re-ripped at lower levels.
With input levels set much lower @ 24 bits you have a bullet proof margin of error, and these types of risks are eliminated and you will get reliable results and crucially almost certainly save a lot of time ultimately from avoiding mistakes.
I was wondering if anyone here could vouch for whether or not this is true. Additionally, what level should I be peaking at if I’m capturing at 24 bit and is it better to record at 24 bit in Audacity rather than 32 bit since my sound card is not capable of native 32 bit recording?
Additionally, I want to be able to hear what I am recording whilst recording. I know of the “Software Passthrough” or whatever it is called. However, I seem to recall hearing of a “hardware accelerated” mode in Audacity that doesn’t have Software Passthrough’s lag and are wondering how that works and how I determine if my sound card supports it.
Finally, I was reading the manual and saw this…
Audacity cannot record at greater 16-bit with MME or WASAPI as the host - you need to use Windows Direct Sound if your sound device supports a bit depth of more than 16-bits.
Does this mean if I want my recordings to be native 24 bit, I need to use DirectSound? Reading the manual further, it seems DirectSound is flat out superior to MME and that there is no point in sticking with MME. Am I correct in this assumption?