Yes see my post above:
For years now, if I record anything from vinyl (via my Technics SL1200MK2s) once itâs on the computer Iâll use audio editing s/w to filter out the (under 30Hz) rumble then remove crackles if theyâre audible. Maybe tweak the EQ as required, then a touch of modern limiting.
Maybe youâre thinking of the option to change the SPDIF between 16 and 24bit. If the gear receiving it can do both, thereâs no reason to use 16bit.
The only time the older CDJâs prior to the 3000 do SRC is on some of the models that allow you to do a hot cue from a track thatâs not currently loaded but is accessible, and only then when that hot cue is a different frequency sample rate than the track playing.
The CDJ-2000NXS2 added 96khz capability, with the same auto-adaptation of the onboard sample rate and SPDIF. So, if you were sending a 24/96 track on the CDJ-2000NXS2 into a PPD9000 using SPDIF, it was going to malfunction. All the fully digital Pioneers from the DJM-800 on can take 96khz, though. Even the A&H Xone DB4 has 96khz SPDIF input capability.
I think noticing the IMD+noise and treble roll-off has a lot to do with how much is going on in the music youâre listening to, how much inherent distortion + noise is in the recording (or baked into the production), and even the loudspeakersâ environment. If youâre in a barn blasting bleep techno to the point itâs causing your ears / brain to distort and compress, itâs unlikely anyone will notice.
The primary advantage with higher sample rates is for analog to digital conversion, anti-aliasing filters in processing and DACs, and digital signal processing, and that last one has a lot of anti-aliasing filters. All sample rate converting has side effects of some sort or another, even if you use non-real-time methods with massive computational overhead that minimize them. Whatever you record & produce at, itâs usually best to stay with that rate, and even bit depth. It is true, however, that most preamps, power-amplifiers, and tweeters donât respond as well from a linearity and distortion standpoint if they have to transmit / reproduce frequencies that are near the limits of their frequency bandwidth. You donât have to use the entire dynamic range and bandwidth of a digital recording medium to get benefit out of it, though, but a lot of what people are hearing with HiRes stuff that truly is lossless is either simply the fact thereâs been no further conversions, and / or, sadly enough, they sometime are just hearing their downstream gear performing worse and they interpret the difference as being better. Personally, I think true studio versions that people prefer usually sound better because of the former, but itâs undeniable that some of the latter is happening sometimes.
Oh man, that classic HP Dynamic Signal Analyzer in the photo is bringing me such a warm fuzzy feeling. That is a stone cold CLASSIC lab instrument. I used those day in day out for years while working on early Blu-ray recording drives. Itâs a lovely, lovely instrument from when HP were still HP before being diluted by selloffs. Basking in the nostalgia of an HP 3561A and my Yokagawa scope. Usually I used the newer Agilent version, but the old ones were still capable.
That definitely sounds interesting! Can I ask you what your background is in that?
I used to work on optical storage drives for a large corporation. Like one of the drive types in the OG Xbox for example. Control Theory, Lapace Transforms, Bode plots, S-domain, z-domain is my bag.
Ok, that sounds awesome for sure!
Null test ?
Nothing else matters to support any argument.