Hello to everyone, looking forward to seeing the really cool projects that people have committed time to and looking to learn something new about preservation.

TL;DR: You can stretch/resample audio originally intended to run along with 24/1.001 fps (instead of "Captain Jack") to do so with 24.000 fps or also only reflag the sample rate to run slightly faster, theoretically preserving the quality.

Hi there,

the following is for sure rather academic and "nerdy" in terms of being most probably inaudible anyway. Technically, resampling audio is nothing bad per se, contrary to common audiophile misbelief, as most DACs for example will already to that for the sake of oversampling, jitter suppression (e.g. via ASRC) or matching some sweet spot of a crystal's/DAC's native sample rate (as Benchmark did prominently with their "DAC1"), etc.

However, from an archival's and psychological standpoint, it makes sense to mangle the data as little as possible.

That was the preamble so to say. Now recently, I again edited and synchronized a few Cinema DTS tracks to certain movie releases. Those usually run at 24/1.001 (~ 23.976) fps (thanks, NTSC legacy!) and are encoded at 48000 Hz. Cinema DTS however, being sampled at 44100 Hz, is intended to run along 24.000 fps as standard in the cinema and thus will be too fast by a factor of 1.001.

Most people here would thus stretch the Cinema DTS 44100 Hz audio to be a bit slower and match the 24/1.001 fps timing and maybe also resample the whole thing to become 48000 Hz to somewhat better match the usual DVD/BD/UHD BD specifications. Either step will be lossy by definition, but as mentioned in the introduction, not a real issue.

If one wants to avoid that and only keep the (usually best anyway) Cinema DTS track, one can quite easily go ahead and just change the corresponding nominal frame rate of the video track to be 24.000 fps instead of 24/1.001 fps by e.g. eac3to (the parameter "-changeto24.000" will do). Remux everything back and voila, one ends up with 24.000 fps video (which accordingly to eac3to is "unusual", but actually used as such officially for often European releases) and the proper untouched audio.

But, if one also wants to keep the usually provided audio tracks from the 24/1.001 fps source, one runs into the problem of having to alter the video playback frame rate depending on the used audio track (24.000 fps for the Cinema DTS track and 24/1.001 fps for "the others" (movie pun intended, Kidman is great)). Unfortunately, no usual player I'm aware of, supports that on the fly.

So I thought of another solution which turns out to work surprising well: instead of resampling or changing the speed of the 24/1.001 fps audio tracks, one can instead reflag the 48000 Hz of those to become 48048 Hz (e.g. with Audacity), covering for exactly the timing difference, as 48000 Hz for 24/1.001 fps equals 48048 Hz for 24 fps. It is a nice mathematical coincidence that the speedup works out to perfectly fit integer-wise as with 44100 Hz, it would already be fractorial.

48048 Hz are definitely non-standard and it could be very well that most players internally mangle the 48048 Hz to 48000 Hz anyway (then providing not even the theoretical advantage of having left the audio material untouched) but the data itself one will still have kept unaltered which is always nice for archiving. Sadly, my Denon's status information isn't exact enough as it only displays "48 kHz", so without further measurements, I don't know if it actually gets 48048 Hz from the Ugoos or resampled 48000 Hz. Maybe, I'll check that via S/PDIF some other time.

I haven't found a way to reflag AC3 or DTS files to run at 48048 Hz though and it could be very well possible that it's simply not allowed by any standard, although technically possible of course as it's only a matter of eventual processing and DAC timing and not the actual encoded data. So one would have to decode that stuff to PCM beforehand.

Feel free to let me know what you think of that, just some idea I came up with.

Hello everyone. Glad to be here and very interested in exploring what other members and accomplished with the wonderful edits and restorations.

The German Blu-ray has very different opening then I remember watching on TV as a kid. Then I saw a version on Youtube and it has the opening I remember. Can anyone provide more information what the source is and if it's available?

Thanks

New Opening(I call it that way)
https://www.youtube.com/watch?v=niXOGQWZ_TE

Original Opening
https://www.youtube.com/watch?v=7SvAV9U4bwc

Does anyone have the mono track from The Time Machine (1960) LD? I don't like the 5.1 from the bluray and would like to make my own with the best audio possible. I'm aware it was synced by emunateD, but he hasn't responded to my PM.

Hello everyone

Hi, i am mostly into original cinema sound mixes and color timings, but there seems to be alot of fun going on in this forum like fancuts and other things to enjoy.

Long live 35mm.

1993 Special Edition Laserdisc [1988-85] PCM 2.0 decoded into 5.0 using a Shure HTS5300.  Synced to the UHD and saved as FLAC.  Watched it last night, it sounded awesome.  PM me for link.

I also have the 1996 LD AC3 [0896885] synced without re-encoding.  

Some info on the PCM recording process:
Panasonic LX600 digital out goes to an ESI U24XL.  The U24XL's stereo analogue outputs go into the HTS5300.  The HTS5300's L-C-R-SL-SR outputs go into an ESI Maya44, which is recorded in Reaper at 44.1khz.  Each disc must be recorded twice since the Maya44 only has 4 inputs.  The first pass records L-C-R.  The second pass records C-SL-SR.  Both passes are then synced together in Audacity by matching the two Center recordings.  The resulting 5 channels are then synced to UHD audio.  Finally, an empty LFE channel is added for playback compatibility (I prefer to let the AVR handle the bass management rather than record the HTS5300's LFE output).  The mix is peak-normalized as a whole to -1db thus preserving the channel balances while increasing the overall gain (the volume output on the Shure is kept at a conservative level to avoid the possibility of clipping), then exported as 48khz 5.0 FLAC.  The two ESI units are operating on separate computers to avoid ground loop hum and interference.  There might be an easier way to do all this but this what I had on hand to work with.  

Normally, I would prefer to record a mono surround in a simple 4.0 but so far as I know, besides lowering the volume pot on one Surround channel, there's no way to stop the Shure from doing it's surround magic, so I was afraid that recording only one surround channel would be detrimental to the overall cohesiveness of the sound.  Maybe downmixing the surrounds into mono after the fact would be more "accurate" but I don't think it makes much difference; its not like the Shure is suddenly making the surrounds completely discrete.  Besides, the track sounded really great last night.

The AC3 was recorded using the LX600's AC3 Mod output going into a Sony EP9ES acting as a demodulator.  The Sony output is recorded in Reaper at 48khz using the U24XL, then decoded into AC3 using BeSplit, then synced using Eac3to.

Hey everyone. Just introducing myself here.

I'm an old sod from before the Blockbuster era who's getting sick and tired of revised colour grades that destroy the original feeling. It irks me. So I'm trying my hand at "premastering" Aliens to make it look like clean footage straight out of the bath.

It's a playful way to learn DaVinci Resolve and correct a trend that occurred around the time everything went "Blu".

If I'm allowed I'll post a few screenshots of my progress looking for feedback. I've not been through the site rules as yet though so we'll see.

Cheers

Hey, guys.

Am I the only one who have this bug on the homepage on my PC? It's been like that for about a week now. It's fine on my phone.