Editing - Still "bit-perfect"?

Hey everyone,

I've been capturing bit-perfect PCM using Bronan's guide (Reaper set to 44/24 and then exporting at 44/16). I've got some questions about editing.

Although I capture in Reaper, I've been editing out side breaks/silence between discs using Audacity (a program I know a lot better). I'm wondering if that is messing up my files at all.

I've got two options for editing:

Option 1: Capture in Reaper at 44/24. Edit out side breaks/disc swaps in Reapar. Export edited WAV file in 44/16.
Option 2: Capture in Reaper at 44/24. Export file in 44/16. Put it into Audacity (which seems to upsample file from 44/16 to 44/32). Edit in Audacity. Export from Audacity at 44/16.

Will either option impact the bit-perfect nature of the audio itself? I know upsampling the khz is bad but is there any problem with upsampling the bit depth? 

Any help would be great. Thank you!

[spoRv]

If you capture laserdisc at 44/24, it's already not bit-perfect...

If you capture laserdisc at 44/24, it's already not bit-perfect...

Ahhh. I was following Bronan's guide: https://forum.fanres.com/thread-2742.html

I've used the 44/24 capture method to capture both DTS CDs and LDs (which then are able to be converted from PCM to DTS) which would seem to suggest it doesn't harm the audio in any way. I could be totally wrong about that though.

[spoRv]

If you capture laserdisc at 44/24, it's already not bit-perfect...

Ahhh. I was following Bronan's guide: https://forum.fanres.com/thread-2742.html

I've used the 44/24 capture method to capture both DTS CDs and LDs (which then are able to be converted from PCM to DTS) which would seem to suggest it doesn't harm the audio in any way. I could be totally wrong about that though.

It "could" not harm PCM (read: even if not bit perfect, you could get the signal more or less the same), but it would definitely harm it for DTS (and AC-3, too)

If you capture laserdisc at 44/24, it's already not bit-perfect...

Ahhh. I was following Bronan's guide: https://forum.fanres.com/thread-2742.html

I've used the 44/24 capture method to capture both DTS CDs and LDs (which then are able to be converted from PCM to DTS) which would seem to suggest it doesn't harm the audio in any way. I could be totally wrong about that though.

It "could" not harm PCM (read: even if not bit perfect, you could get the signal more or less the same), but it would definitely harm it for DTS (and AC-3, too)

I see. I've been able to play my edited PCM capture of a DTS soundtrack on Foobar no problem. I did just put that same file into BeSplit and got a weird DTS file. It was all the content but at like 2.2x speed. No idea what is going on there.

Do we have a guide about highest quality editing? Silence before and after, side breaks, and potentially small edits need to be done in order to sync. Is the goal "bit-perfect for as long as posible... until you can't be bit-perfect anymore"?

[BusterD]

In my setup at least, it doesn't matter if I set it to a higher bit-depth or sampling rate when capturing PCM, as it's essentially capturing the same data stream regardless.

I once accidentally had Sound Forge set to 24/96 for one LD, but all I had to do was go to "Resample" and then "Set the sampling rate only (do not resample)" to change it back to its proper 44.1kHz, and then used eac3to ("eac3to input.wav output.wav") and it automatically changed it back to a 16-bit file, since Sound Forge simply padded the 16-bit stream with zeroes to make it 24-bit.