Cinema DTS - main thread

[zoidberg]

It's what you think it is, a DTS Stereo CD-ROM contained 2.0 matrixed audio. The release of Jurassic Park was dual inventory with either discrete or matrixed DTS and a Dolby A analogue optical track

[borisanddoris]

Has anyone gotten a hold of any of the DTS stereo CD-ROMS for ripping? I'd be really curious to see how they sound.

[spoRv]

Has anyone gotten a hold of any of the DTS stereo CD-ROMS for ripping?  I'd be really curious to see how they sound.

Second that!

[2001CluelessDianogas]

Personally I don't rely on foobar to split out the LFE. That doesn't even do it all for you anyway. As the surrounds are left the wrong level and I feel the LFE is wrong anyway as well.

The surrounds need to be lowered -3dB in Audacity or something once you get hte output from foobar because home theater designates that you set L,C,R,Ls,Rs to all read the same volume for the same input but in cinemas their Ls and Rs each put out -3dB less volume for the same input than L,C,R so they are calibrated -3dB compared to for home theater. So when listening to CinemaDTS mixes you either play it as is and then set -3dB for each surround or simply leave your home cinema speaker levels alone (safest, since not everyone will know to adjust) and then simply lower the signal -3dB. The cinemas put their Ls and Rs -3dB lower each than L,C,R as an old artifact from when surround was mono and for easy auto compatability if it dropped back to some optical mono surround or whatnot since that gets fed then to Ls and Rs they'd have mono surround volume doubled otherwise. So the Cinema DTS decoder doesn't do any -3dB for surrounds since they don't have to, the cinema surround speakers are already each calibrated -3dB lower unlike for home setups.

I feel like the thing to do is:
First when using foobar turn off it's auto LFE creation and all of it's +dB processing for any channel, let it do nothing at all. I seem to recall that you set it to do LFE and adjust dB of anything that the dB amounbt adjusted seems different numbers than when doing it manually so not sure what it is doing under the hood. And use the most recent version.

Take the Ls,Rs and combine the two tracks and then do a lowpass at 80Hz filter on them and then save that out as the LFE and then do a +3dB gain to it (already have a +3dB from combining the signal stored in both Ls AND Rs and actually I think it might even be wrong to do this +3dB step, I forget, have to go do it again and see, might be -3dB for old discs and 0dB for new rather than 0 and +3). For pre-1999 I think forget the +3dB stage. Smetimes you need to do the 80Hz lowpass filter first before combinging them as the upper frequency power might clip stuff, in those cases, 80Hz filter first, then combine the two channels, then +3dB if post start of 1999 release. Actually it's been a while, I forget, I think maybe in the end I did NO +3dB for the early 1999 title actually and decided it maybe just needed nothing done since they had recorded it in studio with LFE speaker already set to account for+3dB and then they stored it once in each channel and combining gives another +3dB. So maybe it is actually low pass, combine and then -3dB for pre-1999 and as is for 1999+? I forget, will check. I know that whatever I did, in audactity, my LFE looked about the same as what Schorman got for AOTC and all I did was either lowpass each and then combine Ls and Rs and then leave them be or at most then do a +3dB at that point and did no +6dB or +9dB and such for sure.

Then undo those actions on the Ls and Rs and then apply -3dB to each and then apply a high pass filter to them at 80Hz.

I was doing 48dB octave passes in Audacity although that seems to remove more stray high frquency from LFE than what the CinemaDTS box does looking at Schorman ouput so that box maybe does something more like 24dB per octave but it also seems to use a different filter method too so it won't be quite the same whatever way. Mine 48dB/octave in audacity certaily filtred out a lot of higher end stuff from LFE channel that really should not be there and took more bass out of the surrounds. That said a stronger filter perhaps puts in more artifcats of other sort. Not sure what the ideal balance is.

My software method in Audacity left a bit less instances of clipping than the Schroman DTS box output, neither had many clips at all, but the official box output had a few more. Or perhaps it came from when he resampled it from 44kHz to 48kHz? Not sure if resampling can produce clipping, have to think, maybe, in fact I'd bet so. I know applying low and high pass filters can since even though you are removing total energy you can realign some stuff to re-enforce and can phase shift some frequencies relative to others and they can add in new ways and they can also produce peak artifacts near cut frequency, etc. As even his center for one disc had the part that also clipped one mine (apparently clipping in the master!) clip just a bit more, maybe that was from his 44->48Hz resmapling stuff.

To best avoid clipping you may need to play around with the order of steps and balance clipping vs. maintaing max SNR:

For one title (using Audacity and whatever algorithms and methods it uses) reel 1:
I have to first -3dB before high-pass to create surrounds if I don't want any clipping in them.
I can straight add the two and then lowpass and make LFE with no clipping.

for reel 2:
I can just high-pass the surrounds first without clipping and then give them their -3dB.
I can straight add the two and then lowpass and make LFE with no clipping.

for reel 3:
I can just high-pass the surrounds first without clipping and then give them their -3dB. EDIT: nope can only do that for left surround, right surround need to first -3dB and then high-pass to avoid clipping.
I can't just straight add the two surrounds and then lowpass and make the LFE without it clipping. Even doing -3dB first to each surround and adding them still clips. I found I could first low-pass them and then add them as is without clipping though.

etc.

I wonder how the DTS processor does it. For safety does it just first -3dB the surrounds and then high-pass and then raise them back up again 3dB (remember the theater speakers have surrounds set -3dB) and then also add the two -3dB adjusted (EDIT: nope this is not enough to stop clipping for some of them when starting to make the LFE) together to make the LFE and then low pass that and then raise that back up again 3dB? Does it just apply all the stuff and the adding as they are and you end up with some clipping? In which case these manual methods with care may give more distortion free output than the theatrical processor. Maybe it does -3dB to each surround and then high-pass filters them to produce the surrounds. And then it goes back and looks at the original combo surround data and does a low pass filter on them and then adds them together.

One thing to keep in mind is how you are outputting this stuff and whether your PC is doing the +10dB (+6dB depending how you measure stuff) boost for LFE channel to your sub and your sub at home is normally set +0dB and not +10(+6dB) like in a theater, in a theater the sub is the LFE while in most home theaters the sub is the LFE signal (which needs to be +10dB) and the re-reouted deep stuff from all other channels (which needs to be +0dB). Etc.

How does your A/V receiver handle .1 input from analog vs PCM digital inputs vs DTS/DD etc.

Anyway I'd do it all manually and not let foobar do LFE anything. And I'd not resample to 48kHz and I'd leave frame rate at 24 fps exact and just force any blu-ray stuff you sync to to play back at exact 24 (which introduces zero anything and actually gives you originally theatrical frame rate back, OTOH resampling to get the 23.98whatver can put in some more bits of clipping and make small sound artifacts and same for 44->48 resample, so why do that stuff?). And if you match to 35mm scan then of course no need at all.