Editing - Still "bit-perfect"?

[zoidberg]

Fair enough, it's good that we have extra options for bit-perfect capture. I knew audacity could edit bit-perfect but not that it could capture bit-perfect via padded 24bit. I just use Reaper (I'm still evaluating it lol)

[Falcon]

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.

I still think that in this scenario, you have lost samples, even if the ones you have are perfect.
If you capture a 44.1 stream at 96 khz, you will duplicate the samples 2 or 3 times a priori. Then if you remove samples to get to 44.1 (no resampling or interpolation, just remove the excess) these will of course be 16 bit intact.
But 96 not being a multiple of 44.1, not sure that you don't have some cadence problem (slight acceleration and deceleration) or lost samples.
If you had captured in 88.2 this wouldn't have been a problem in my opinion 

[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.

I still think that in this scenario, you have lost samples, even if the ones you have are perfect.
If you capture a 44.1 stream at 96 khz, you will duplicate the samples 2 or 3 times a priori. Then if you remove samples to get to 44.1 (no resampling or interpolation, just remove the excess) these will of course be 16 bit intact.
But 96 not being a multiple of 44.1, not sure that you don't have some cadence problem (slight acceleration and deceleration) or lost samples.
If you had captured in 88.2 this wouldn't have been a problem in my opinion 

Sound Forge doesn't duplicate the samples, it just speeds them up by compressing them into a shorter time length. So when I used "Set the sampling rate only (do not resample)" on the sped up file, it slowed it back down to the proper rate without changing the samples.

[Falcon]

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.

I still think that in this scenario, you have lost samples, even if the ones you have are perfect.
If you capture a 44.1 stream at 96 khz, you will duplicate the samples 2 or 3 times a priori. Then if you remove samples to get to 44.1 (no resampling or interpolation, just remove the excess) these will of course be 16 bit intact.
But 96 not being a multiple of 44.1, not sure that you don't have some cadence problem (slight acceleration and deceleration) or lost samples.
If you had captured in 88.2 this wouldn't have been a problem in my opinion 

Sound Forge doesn't duplicate the samples, it just speeds them up by compressing them into a shorter time length. So when I used "Set the sampling rate only (do not resample)" on the sped up file, it slowed it back down to the proper rate without changing the samples.

Good to know that