Help explain intersample overs, please?

[192kbps]

Firstly yes it will truncated some bits. What matters is the noise at the output or the SNR of the actual DAC. Then some dacs like ESS based can preserve most of the SNR when reducing volume internally. For AK based I think(correct me if I'm wrong) will have a trade off between SNR and else. Most likely reducing a little bit of SNR can give better SINAD because distortion may also decrease.
So in the end you may avoid the intersample clipping and gaining SINAD at the same time. So this may be worth the trade off. More over, one can always use two dacs with each for parallel to gain back the 3db of SNR.

Two dacs?
Do you mean two D90SE? Or is it the eight channels (>2) used by D90SE?

 

[little-endian]

Also there is a big difference between AKM and ESS due to differences in their internal architecture. AKM chips typically provide about +2 dB headroom at their analog output. And when it gets too much all you see in a wideband FFT is out-of-band harmonics - no big deal. ESS on the contrary also pollute the audio band with spikes - a totally different behaviour.

Out of technical interest, it would be great to know how the implementations of the ADI-2s differ depending on the used DAC (AKM vs. ESS).

If I understand it correctly, for the ESS versions, you provide an additional headroom of 2.5dB. Is that an additional headroom on top of what is used for the AKM versions or don't those have any in particular?

I'm aware that this is probably a rather academic and theoretical scenario, but given that with especially prepared test inputs, even way higher intersample peaks can be constructed, aren't AKM chips still in slight advantage in terms of their general behavior when intersample clipping occurs for those cases where even the additional headroom of 2.5dB is exceeded on the ESS versions?

 

[AnalogSteph]

Here's a little illustration of what @MC_RME meant by "pollute the audio band with spikes", which is typical ASRC behavior:

Teac VRDS-20 Review (CD Player)

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You're right, it’s mentioned in the datasheet

"Clipping
Under certain rare input conditions, it is possible for the AD1893 to produce a clipped output sample. This situation is best comprehended by employing the interpolation/decimation model. If two consecutive samples happened to have full-scale amplitudes (representing the peak of a full-scale sine wave, for example), the interpolated sample (or samples) between these two samples might have an amplitude greater than full scale."

And it is that ugly:

 

[MC_RME]

Out of technical interest, it would be great to know how the implementations of the ADI-2s differ depending on the used DAC (AKM vs. ESS).

They don't. Both have that digital headroom of 2.5 dB

If I understand it correctly, for the ESS versions, you provide an additional headroom of 2.5dB. Is that an additional headroom on top of what is used for the AKM versions or don't those have any in particular?

I'm aware that this is probably a rather academic and theoretical scenario, but given that with especially prepared test inputs, even way higher intersample peaks can be constructed, aren't AKM chips still in slight advantage in terms of their general behavior when intersample clipping occurs for those cases where even the additional headroom of 2.5dB is exceeded on the ESS versions?

In theory - yes. In real-world it's barely measurable, so not much difference. And as both chips don't pollute the audio band, but generate proper out-of-band harmonics, there is definitely no audible difference.

 

[j_j]

They don't. Both have that digital headroom of 2.5 dB



In theory - yes. In real-world it's barely measurable, so not much difference. And as both chips don't pollute the audio band, but generate proper out-of-band harmonics, there is definitely no audible difference.

If I understand correctly, you must be rendering the unclipped signal all the way into the continuous domain before any clipping can happen, yes?

The error signal of a symmetrically clipped sine wave does not roll off particularly fast, as I recall, something like down 10dB for 3rd, 24 for 5th, next two around -30 or so, then more slowly.

For insignificant aliasing due to clipping in sampled domain, this would require any oversampled processing to be way, way way oversampled, yes? I'm not sure how much that would be, but it would seem to be a lot.

 

[little-endian]

They don't. Both have that digital headroom of 2.5 dB

Thanks for the confirmation. Is that the point in the signal chain where Benchmark came to the conclusion to use 3.5 dB? If so, what's the reason for RME's choice? Not questioning anything, only interested to learn.

Furthermore, I assume the headroom is on top of the 0dB setting of the DAC, right? As long as no SRC is active with its only headroom implementation if not mistaken, one is probably left with actually a lot more headroom as the digital signal is lowered before the DAC input, correct?

In theory - yes. In real-world it's barely measurable, so not much difference. And as both chips don't pollute the audio band, but generate proper out-of-band harmonics, there is definitely no audible difference.

What is the essential, theoretical advantage of the AK4493 then if it is said to handle intersample overs more "graciously" than the ESS 9039Q2M?

 

[JRT]

An informative on-topic video (see below) was posted on Youtube yesterday, a conversation between Gene DellaSala of Audioholics and John Siau of Benchmark Media, including demonstration simulations, measurements, and explanatory discussion.

Maybe obvious to some, but... One takeaway from the video was that Roon provides facility for reducing digital signal level -3dB prior to other downstream digital signal processing and DA conversion, and that doing that may be a good workaround to ameliorate much of the worst of the problem with little downside tradeoff in the compromise.

Wednesday, 06 November 2024
Intersample Clipping - A Problem with Most Digital Playback Systems?
(77 min, 09 sec)

"Intersample clipping is the biggest problem with PCM systems." - John Siau

Oversampled sigma-delta D/A converters and fixed-point sample rate converters can add significant amounts of distortion when playing CD-format recordings. This distortion is caused by intersample musical peaks that overload our digital playback systems. These overloads often occur many times per second, even in a well-recorded track. This distortion can reach high amplitudes, and it can easily be measured, but it is completely invisible to traditional audio measurements. This distortion can easily be eliminated from our playback hardware, but most of the audio industry has failed to recognize this issue. Intersample clipping may be the single most audible defect in fixed-point PCM playback systems. It is the elephant in the room.

We need to change the way we build and test our digital playback systems.

...