[HELP] Hybrid introducing audio delay that is not present in source material

[4i4oVi40]

[quote pid="13798" dateline="1633958768"]
-> Since the channel count (usually 2 or 6), the sample rate (usually 48 000Hz) is often the same the delay modifier also often is the same.
[/quote]

OK, I see. So by default (for faac, fdk, etc.) 2048 Kbps are needed for DTS-HR. And if the SR is 48 kHz (almost always), your delay formula is: 2048/48 + 0.5 ≈ 43 ms.

So this is not a bug but a feature :-) So if I have a DTS at 48 kHz sample rate converted to AAC, I'll always end up with 43 ms delay. I read your thread from 2013 https://forum.videohelp.com/threads/3576...AC-Encoder

Obviously, 43 ms is an offset of just 1 frame, so it's barely humanly possible to discern it. Nevertheless, it's useful to know where this minimal delay comes from and why it's always the same, if you use the same encoder and the same SR.

Thanks for the info.