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Re: Thanks for taking the effort to find it and read it

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***Actually, it is also likely that the clipped signal level is at 0dB FS and signals at a higher level are samples of the reconstructed analog waveform at points where it exceeds 0dB FS. remember we are taking a recording of a reconstructed analog waveform, it will be highly unlikely we are sampling at exactly the same points as the original digital recording.***

This can be the case. But the wave you showed has **many** samples showing clipping.
The troubles the papers refer to with reconstructed signals between sample points can only occur across a few samples.

***Your statement is contradicted by those of ted smith and graemme previously on the asylum. given that these people have been involved in designing and building audio equipment, or have direct experience recording and mastering in DSD, i choose to believe them over you.
***

You need to approach this logically. Whatever your guru's have posted about clipping in **recordings** isn't of consequence in this discussion.

With 1 bit at 64*fs you can't turn on (or off) more than that number of bits. So it's ***impossible*** in a pure 1 bit environment to get past 0dB. You simply run out of bits. This is true for recording and processing.

With pcm recording there isn't a need to keep the levels below -6dB.
Maxing out at near 0dB with 24bits isn't creating any problems as long as there are more bits available in the processing environment.
You simply don't need to waist dynamic range during recording.

To prevent the problem described in the paper it's only necessairy to keep the maximum output for the production master below -6dB.
In the production stage you can still have the full resolution with pcm.

Also this 6dB processing headroom is a big 'waist' of storage space.
With 1 bit is 'costing' you half the number of possible bit patterns.

With 24bit pcm 6dB headroom is only costing 1/24th of the available bitspace.

Frank


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