Time resolution of digital sampling

[Miska]

9 hours ago, pkane2001 said:

... and Redbook format can record delays of much less than 5 microseconds.

 

Down to about 346 ps...

 

[Miska]

On 10/8/2020 at 6:05 AM, JoshM said:

How much do you think we should worry about intersample overs? Are they audible? Should we apply digital volume attenuation in Audirvana, HQ Player, etc. to avoid overs? Or does the extra processing involved in using digital volume attenuation outweigh the benefits of avoiding overs?

 

Yes you should. And you have no way of avoiding "extra processing involved in using digital volume" with HQPlayer.

 

On 10/8/2020 at 5:35 PM, Jud said:

I don't know whether HQPlayer has any sort of "graceful failure" mode for intersample overs that may have affected my experience.

 

It has, every time "Limited" counter is incremented you have triggered the soft-knee limiter that has been designed to be as inaudible as possible.

 

On 10/9/2020 at 12:59 AM, Don Hills said:

I don't worry too much about intersample overs. I feel that music with aspirations to high fidelity won't have been pushed to the limit in mastering, and conversely music that has been pushed hard will likely already have more audible damage.

 

To what I've seen, over 90% of RedBook material has overs. And for over 90% of it, -3 dBFS is enough to avoid it. And there are clear mathematical reasons why this is the case.

 

There are some albums though where -3 dBFS is not enough.

 

[Miska]

Intersample overs happen at least because someone decides to run "normalize" function for music at 44.1k sampling rate. Since actual sampling points rarely coincide with the actual waveform peaks, this results in values higher than 0 dBFS when the actual waveform is reconstructed. The reconstructed values are more likely to reach the highest point, higher the digital filter oversampling factor is.

 

Another common reason is RedBook content driven to clipping, which also seems to be about 90+% of modern content.

 

[Miska]

3 hours ago, John Dyson said:

I haven't thought about this -- but what if you immediately convert to floating point in your math?   I always use floating point, even floating point .wav files.  Therefore, worrying about clipping only happens when dropping to the +-1 type formats.   Then, can worry about the -0.8dB or -3dB or whatever the fashion is today.

 

DACs take only integer format data, and also delivery containers such as RedBook or FLAC. You have strict and clear value range boundaries.