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In Reply to: RE: LP recordings: Normalization and limiter posted by tketcham on November 10, 2021 at 06:24:25
I really wouldn't be worrying about losing "resolution" with the LP when setting the level. I would be more concerned about clipped samples and the effect of overload on the input buffer to the ADC.
The maximum dynamic range for human hearing is 20 bits. A typical consumer 24 bit ADC only has about 21 bits dynamic range and this still vastly exceeds the capability of a vinyl LP/phono stage when looking at the signal level relative to the noise floor.
I don't bother adjusting the input sensitivity per record - I set it once according to the cartridge output relative to the worst case maximum output from a test disc so that the overload margin is never hit even on clicks and pops. Typically I would look for a maximum output of -3 or -4dBFS.
I use RX so the functions have slightly different names - "Normalize" in RX is a little different to what you might expect in that it calculates the gain to hit a desired maximum level. However, I essentially do what you do (without the limiting) after editing/processing my file; I set a maximum level of -4dBFS for all my recordings so that I don't risk intersample overs when downconverting my files. I wouldn't go above -3dBFS myself, but you can choose what you want if you aren't using a DAC that does funky curve fitting.
If you really wanted to maximize fidelity, then you would transfer with flat gain and apply EQ digitally as we have talked about in the past.
"Beauty is Truth, Truth Beauty.." Keats
The ability to edit waveforms to set a desired amplitude is a modern day wonder in my experience, making recording levels less critical than the old days. The only time I pay attention to input levels is if I've changed cartridges and/or phono stage settings. Using a 0.0db input sensitivity on the Tascam I typically end up with peaks at -3 to -4dB, with a range from -2 to -8dB depending on the LP.
For normalizing, I'm using the Loudness Normalization perceived loudness LUFS feature in Audacity to set average amplitude for the entire side. I do take an extra step to reduce or increase amplitude of a song or two if they are dramatically different than the rest of the side prior to normalizing.
I've been using 0.01dB as my peak limiter threshold in post-processing, just to avoid 0db glitches but perhaps I should be using a lower value. I've read that 0.3dB might be a safer margin. I haven't run into any problems with playback using two different DACs but I should check.
I record at 24-192 PCM WAV but then resample to 24-96 FLAC for NAS storage and playback, mainly because the input signal to the Tascam is 24-192 SPDIF from the SugarCube and I didn't want to use the onboard resampler of the Tascam. The sound quality at 24-96 for playback is quite good and takes up much less hard drive space than 24-192 files. I figure it won't hurt to have the finer resolution during recording in case I decide to reprocess certain files using different processes or settings or I find that some recordings warrant the 24-192 resolution for playback.
Thanks again for the comments and suggestions.
I sort of do what you are doing for Normalization except I do it manually to ensure that the original dynamic range is preserved as far as possible.
I search for transient pops and clicks that would upset the actual signal peak determination and remove them, then analyse the waveform for loudness in the same way as you - the results give me all the information I need on peak sample amplitude, true signal peak, average rms level etc. With singles or LPs that have sides or tracks recorded at obviously different levels, I compare the average RMS level and then use the Gain function to rescale the tracks/sides to a similar level as the reference. Once combined, I then apply the Normalize function in RX to set my preferred maximum level.
As far as my preferred sample rate goes, I originally went for 24176 as my reference which allows a straightfoward downsample to Redbook and easy conversion to single or double speed DSD. However, I shifted to just going straight to 2496 and not fuss with DSD - I just create a DVD Audio ISO file and playback that back with a PC through my Hifi or burn a DVD if I want to give it to a friend who still uses an Oppo universal player.
I couldn't be bothered chopping up the DSD file into individual tracks!
How old is your Tascam? Sadly my DA-3000 is failing - the headphone amplifier stopped working some years ago. Now, periodically, it will stall and stop writing files to the memory card (which is fine). I bought it around 2014 and got about 5 years out of it before it started playing up. I hope yours lasts better!
I use an RME ADI-2 Pro FS now as well as a Benchmark ADC1 USB (which was bought in 2006 and still going strong!)
"Beauty is Truth, Truth Beauty.." Keats
"I search for transient pops and clicks that would upset the actual signal peak determination and remove them..."
I do something similar, i.e., reducing amplitude of the atypical peaks before loudness normalization. But with some LPs I still end up with a number of peaks that get pushed into clipping and use the limiter function to "fix" them. I'm conscientious about overuse of the limiter but I'm still learning. The example I provided in my original post was at the upper end of what I consider reasonable.
My DA-3000 is only a couple years old and I only use it for recording. So far, so good.
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