44.1kHz to 48kHz laserdisc PCM track conversion

[spoRv]

Capturing bit-perfect preserves the audio at the source so to speak. That way as resampling software continues to improve one can always go back to the original to make a new conversion vs the 'baked' in conversion done by analogue capture. Obviously if you're not burning to disc then 48kHz may not even be necessary especially if using a server/media player/htpc.
If you are determined to go the analogue route there may be some merit in using a newer DAC from a high end CD player or A/V receiver, something that outperforms even the best LD player, this could be fed by a 44.1kHz bit-perfect capture or the optical out of a LD player, essentially using it as a transport.

First observation is very right - a bit like "capturing" laserdisc video (and audio, and everything) using the raw converter - you get the whole raw signal and if/when software improves, you can get better conversion.

Second one, I did not think about - and I must admit it's a great idea! So if someone has only the laserdisc player and a so-and-so capture card could capture PCM laserdisc track bit-perfect, and another one without a laserdisc player but a state-of-the-art CD transport and capture card could capture that same PCM track ported on a compact disc, at 48kHz 24bit... so, the best case scenario could be:

1) bit-perfect capture "as is" - not even in sync or with "glued" sides, but as side1 and side2 (and side3 and so on)
2) same as before, but in sync with a given release - possibly with detailed info how to put it in sync
3) analog capture of 1) or 2) depending on which method used (LD-to-CD tracks or DAC reading from the file)

(why do I always look for the most intricate and difficult way to solve easy problems, I can't understand it... )

[spoRv]

One other thing to be aware of is even if your MKV contains 44.1kHz audio, it doesn't mean its not being tampered with before it reaches your ears. I have all my HDMI devices routed into my LG OLED, then use an optical cable to my amp. After recording the output of the TV I discovered that everything is getting resampled to 48kHz regardless. I have a feeling its a worse conversion than an offline one..

Another thing to take in account... so, releasing at only at 44.1kHz bit-perfect leaves anyone to (eventually) convert it to 48kHz, but releasing both can be better, as if someone does not like the conversion could always do it by himself, avoiding the task of doing it for the others - even if frankly is not a difficult task, finding things already prepared is always easier than cook them by yourself, risking to burning something in the process!

[little-endian]

Thoughts?

Yes, plenty actually even if joining the discussion a little late and good counter-arguments already have been made by bronan.

It is common ground that the simplest way to convert 44.1kHz laserdisc PCM track to 48kHz is to capture it via digital cable by a bit-perfect audio card, then convert it to 48kHz

Not only the simplest, but technically the best. Although in practise, one cascade will hardly make any practical difference given nowadays ADC/DAC performances, especially if you're after theoretical academics, no realtime electronic device partly using the analog reconstruction domain will be better than a purely number-based mathematical conversion process done on a PC where memory and time is no big factor.

this introduces a very tiny error in the signal

Yeah, followed by magnitudes of higher errors in the DAC, amplifier, speakers, room acoustics and to more extend than people want to literally hear it - the ears.

But keep also in mind that you will never hear numbers and any sample rate has to provide the "manual" to eventually reconstruct the analog waveform and here the question arises if DAC x 's performance is really best at the source sample rate or if it cannot be higher when using a different one. For instance, the manufacturer Benchmark with their DAC1s back then explained that they voluntarily decided to resample everything to about 110 kHz or so as the "sweet spot" of the used DAC chipset by Analog Devices. Hence, CD sources were upsampled and stuff like 192 kHz (also considered to be entirely silly as a distribution format by the way) downsampled - both also used for decoupling the DAC clock from the input signal to give jitter* immunity.

* also the audiophiles' major buzzword to explain virtually everything. While it seems to be not complete voodoo (even companies like Intel had white papers about it back in the days), most probably, also that issue is vastly exaggerated.

if we slowdown PAL to NTSC (or speedup 23.976fps to 24fps) we introduce further quality degradation

True, but similar category as above.

if the PCM on NTSC laserdisc is not 44100Hz but 44056Hz then any bit-perfect capture is, well, not perfect!

The condition is not fulfilled as non-existing, but even if it would be by other sources, a successful capture isn't entirely unlikely as the actual clock is determined by the source and not the receiver via S/PDIF and hence it even remains "bit perfect" as long as no other mangling occurs if one sets e.g. 192 kHz but playing back 44.1 kHz. Only side effect then is that the audio appears to run "slower" in the recording editor as for one second, less samples are "collected", but when flagging the whole thing correctly at the end, it won't make a difference. Hence, if the hardware is able to synchronise with your beloved 44056 Hz, nothing speaks against doing so in principle. Never had such a source myself though so I couldn't test it yet.

why don't we capture at 48kHz 24bit both analog and PCM tracks via analog cables?

Because it would be inferior to capturing it at the native rates and then post-process it. Also, in general, as an end product, 24 bit don't make sense, despite all the marketing voodoo around it. Even the SNR and effective dynamic range of 16 bit by far exceeds virtually all applications and hearing capabilities and 24 bit cannot even electronically be fully archived due to the electronic noise floor and hence is pure marketing bullshit.

• no quality loss converting 44.1kHz to 48kHz
  
• less quality loss converting from 25fps to 23.976fps - or 23.976fps to 24fps
  
• if the laserdisc NTSC PCM frequency is really 44056Hz then capture it digitally at 44.1kHz, bit-perfect or not, is futile - unless it's possible to capture at 44056hz bit perfect
  

Non of them will manifest themselves here for the reasons given.

sure, it depends all from the player's DAC quality but AFAIK mid models had good DACs, up to excellent ones on top and high end models.

Also that point is vastly overrated and most will fail passing a blind test between cheap an expensive lets say CD players.

Don't think I am totally immune against such irrationality myself - it took me some time to finally encode the audio to AAC multichannel from all those bloated TrueHD/DTS-HD MA / whatnot codecs to save significant space and I still refrain from changing the audio tracks between 24/1.001 fps vs. 24 fps and rather reflag the video although I'm aware I will never hear the difference, but neither am I cheating myself and am at least aware of the cognitive dissonances.

My verdict: to record the AC3/DTS/PCM from LaserDiscs digitally, any cheap-ass C-media soundcard in the few dollars range suffices as long as the data remains intact and for analog recordings, something like a AUREON XFIRE8.0 HD is already way above what is needed for the FM tracks of LaserDiscs to record decently.