How to use eac3to to edit AC3 to avoid transcoding

[zoidberg]

That and the 44.1kHz sample rate, although there are DTS audio CDs as well

[NeonBible]

Is editing DTS (not HD MA) supposed to be lossless?

Simply cutting the beginning appears to have filtered out the high end at around 20KHz for the rest of the track. How can this be?

EDIT: I wonder if its just Spek reporting it incorrectly. Because it seems way too quick for re-encoding to occur.

[alleycat]

Is editing DTS (not HD MA) supposed to be lossless?

Simply cutting the beginning appears to have filtered out the high end at around 20KHz for the rest of the track. How can this be?

EDIT: I wonder if its just Spek reporting it incorrectly. Because it seems way too quick for re-encoding to occur.

Just a thought but is this laserdisc DTS you are editing? If so have you used -dontpatchDTS?

[NeonBible]

Just a thought but is this laserdisc DTS you are editing? If so have you used -dontpatchDTS?

It is a laserdisc DTS! But I also used -dontpatchdts.

Whats weird is if I cut only a small value like -60ms it looks fine. But if I cut a large duration like -50000ms it filters the high end. Very strange.

[borisanddoris]

Hmm. That’s concerning. I may have to go back and find my raw recording files on some of my rips and compare.

[NeonBible]

Riiiight so the problem is just Spek's rendering. If I increase the window size so that it re-renders it looks all good.

I feel so stupid haha. Don't need to go back to recheck borisanddoris, nothing to worry about!

[borisanddoris]

Great news. I didn’t want to have to go back and redo all 3 hours of the boat movie.

[ilpubzo5ie]

eac3to is way to complicated for cutting.
LosslessCut usually works well, however it it not time accurate, so you need to check the result. There are usually 1-2 audio frames to many at the cut point.
One AC3 frame usually has 32ms, so it can be off-sync by that value. However, you won't notice that. spoRv said, you could notice 20ms. No, you won't. You'll start noticing the delay at around 80ms.

For AC3 there's an old tool HeadAC3He, which cuts corrently at AC3 frames. You then combine the parts by just using copy /b file1+file2+... dest.ac3

[phadron]

This is something that has bugged me for a while now and it seams many of you don't know how to handle AC-3 audio, without the need to transcode to lossless (FLAC, DTS-HD MA,PCM) after it was synced. In order to sync an AC-3 audio track to a new source, just use eac3to to edit the AC-3 audio. A long time ago zeropc showed me this technique, which he was teached by jj666. Now the method itself wont be always perfect, but the tracks I've heard so far or even edited myself are pretty good. Most of the time you wont even notice a thing. Editing an AC-3 audio track needs to be done in 32ms frame window. This means the audio sometime wont be synced down to the level you might want it to be. It usually drifts between 2ms to 31ms. Nothing the average listener will notice

...

Thanks Bend!
a question:  if I have to do 100 cuts on a 1h 30m film audio will have 32ms x 100 =3200ms = 3.2s delay ?