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How to decode 6-track APTX-100 (cinema DTS) with the correct channel levels
from The Projectionist’s Guide to the DFP-3000:
Quote:all channels are full range 20kHz (even the subwoofer)

from the Sony DFP-D3000 SDDS decoder manual:
Quote:Setting the digital subwoofer low pass filter frequency to 100 to 200 Hz should be acceptable

from the DTS 6AD digital processor manual:
Quote:DTS derives the digital subwoofer by filtering out the surround signals from 80Hz and below.

from the Dolby CP65 Digital Cinema Processor manual:
Quote:Digital Subwoofer Channel
Pink noise is now present on the subwoofer channel
only (100 Hz bandwidth)
so my educated guess is that subwoofer frequency bandwidth is 100Hz as well
Sadly my projects are lost due to an HDD crash... Sad
Fundamental Collection | Vimeo channel | My blog
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Idk if this is misunderstanding the discussion but I always thought it would be interesting to do the Cinema DTS decoding into a higher bit depth so that there's more headroom. My reasoning is ... maybe the original track was NOT clipped, but the loss of precision through the band splitting, encoding etc. might have resulted in clipping when everything is put back together.

So basically, the individual bands and whatever would be decoded and then added back together and maybe that's the point where the clipping is introduced perhaps. Now if you decoded into, say, a floating point buffer, for the summing of the bands, then maybe this could be avoided.

I actually asked the developer of the plugin if he could do that but he wasn't willing to do it and it's not open source either so I can't do it myself. Has bummed me for a while ... maybe if someone else were to bum him about it in a way that does not seem coordinated, he might reconsider?
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I don't know enough about the APTX100 codec to say for sure, but if is a 16bit process by design then decoding to a higher bit-depth may not be possible
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My thoughts: as the Cinema DTS codec APT-X100 uses basically a refined ADPCM conversion, even if it's good - actually very good, considering they did not use psychoacoustic principles at all, a great feature in my book - it could clip in rare case; this "baked in" the clip into the actual encoding, so it's not possible to "unclip" it, unless you repair the track manually.

And even when it happens, it's usually limited to few single-sample peaks here and there over a 1.5/2hrs 6-track movie - not that difficult to repair.

Fact is: a proper hardware decoder made by the producer should always sound better than a software version made by a third party - even if I must admit I am very grateful for that!!!

Side note: maybe the clipping is not due to ADPCM conversion, or the software, but to the fact that DTS "pumped" their tracks at the beginning, to give them their "signature" (in particular exploding bass!)

From sdurani (a reputable member of AVSforum):
Quote:Their CAE-4 encoder was tested by Warner Home Video on 5 titles and found to add a .6 dB level boost, just enough to not be perceived as a level difference but instead sound like a difference in sound quality (old audio sales trick). Weird part is that the level boost didn't show up with the internal test tones that were used for level matching to other encoders, only in the program material. Clever. Warners was able to catch this for the three 'Lethal Weapon' DVDs, so the included Dolby and DTS tracks ended up being encoded at the same level. But it was too late for 'Interview With a Vampire' and 'Twister' DVDs, both of which had already shipped with the DTS level boost intact (can still be measured against the Dolby track).
Sadly my projects are lost due to an HDD crash... Sad
Fundamental Collection | Vimeo channel | My blog
Thanks given by: borisanddoris
Well my theory (or maybe I should call it a hunch to not overstate it) is that the data inside each band is still not clipped. Basically, the codec, afaik, separates the audio into different bands by frequency before applying further compression. Each band gets its own specific compression ratio and whatnot.

Now, the process would be: separate bands, then encode each band, and write all into CDTS.

So the reverse process would be: decode each band, sum them, and output 16 bit.

Once you have decoded each band and you're summing them up again, there would be nothing stopping you from just allowing for some headroom. Specifically, let's say the normal output is 16 bit. Well, that's a so-called short in programming. You can just use a normal 32-bit integer to write the data into and anything that would have clipped at 16 bit no longer does.

Whether that's actually how it works.... idk. Just a hunch.

Edit: On a sidenote, I was trying to find videos on youtube explaining ADPCM and I gave up after trying 10 videos. Every. Single. One of them is by some Indian mumbling in broken English into a really bad microphone with the volume cranked up to 200000. I'm not doing that to myself! Big Grin
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I don't know if different decoding would make some difference; I took a look at ADPCM and I found several new encoding methods, but if they rely on legacy compatibility, they do not involve also a new decoding method, so *maybe* you could improve the encoding, but not the decoding that *should* be easier than the whole encoding process, that implies a lot of predictions - hence why sometimes it could clip... I guess?!? Wink
Sadly my projects are lost due to an HDD crash... Sad
Fundamental Collection | Vimeo channel | My blog
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