DAC that handles intersample overs Options

[pbelkner]

I did some sample editing in CEP 2.1 and found that I could create peaks that were more than twice (> +6 dB) the values of any of the samples.

Creating this wave needs some, err coincidences to occur if it were to happen in the real world. For a positive peak >> FS:

To give you a real world example consider the second track of VH's latest release:

• The track's maximum sample peak is -0.1 dBFS.
• The track's maximum inter-sample peak is 3.3 dBFS.
• The maximum sample peak of a corresponding MP3 (via "lame -V2") is 2.0 dBFS.
• The maximum inter-sample peak of a corresponding MP3 (via "lame -V2") is even 3.5 dBFS.

CODE

pb@PETER-PC ~/test
$ r128gain --sample-peak 02_she_s_the_woman.wav
SoX sucessfully loaded.
FFmpeg sucessfully loaded.
analyzing ...
  [1/1] "02_she_s_the_woman.wav": -5.6 LUFS (-17.4 LU)
      peak: -0.1 SPFS, range: 2.1 LU
  [ALBUM]: -5.6 LUFS (-17.4 LU)
      peak: -0.1 SPFS, range: 2.1 LU
done.

pb@PETER-PC ~/test
$ r128gain --true-peak 02_she_s_the_woman.wav
SoX sucessfully loaded.
FFmpeg sucessfully loaded.
analyzing ...
  [1/1] "02_she_s_the_woman.wav": -5.6 LUFS (-17.4 LU)
      peak: 3.3 TPFS, range: 2.1 LU
  [ALBUM]: -5.6 LUFS (-17.4 LU)
      peak: 3.3 TPFS, range: 2.1 LU
done.

pb@PETER-PC ~/test
$ r128gain --sample-peak 02_she_s_the_woman.mp3
SoX sucessfully loaded.
FFmpeg sucessfully loaded.
analyzing ...
  [1/1] "02_she_s_the_woman.mp3": -5.6 LUFS (-17.4 LU)
      peak: 2.0 SPFS, range: 2.1 LU
  [ALBUM]: -5.6 LUFS (-17.4 LU)
      peak: 2.0 SPFS, range: 2.1 LU
done.

pb@PETER-PC ~/test
$ r128gain --true-peak 02_she_s_the_woman.mp3
SoX sucessfully loaded.
FFmpeg sucessfully loaded.
analyzing ...
  [1/1] "02_she_s_the_woman.mp3": -5.6 LUFS (-17.4 LU)
      peak: 3.5 TPFS, range: 2.1 LU
  [ALBUM]: -5.6 LUFS (-17.4 LU)
      peak: 3.5 TPFS, range: 2.1 LU
done.

pb@PETER-PC ~/test
$

[Wombat]

I wonder if these inter-sample peaks are a problem on modern playback systems at all.
With my network player i stream 16bit flacs and use its build in 24bit volume control. Normal listening happens between -8db to -18dB.
I don´t know exactly how it works with PC playback but i guess here also the data gets calculated down in level before playback.
If so the most problematic playback may happen with classic CD-Players or players without digital volume control that go full scale on these peaks.

Besides that i often did play with clipped pieces of music but did not find they sound cleaner when simply lowered in volume, samples welcome.

[pbelkner]

I wonder if these inter-sample peaks are a problem on modern playback systems at all.

As it was discussed several times in this forum and as it can be found elsewhere contemporary DACs up-sample in the course of reconstruction, i.e. in the course of "smoothing the digital staircase". I can hardly imagine that clipping during the up-sampling stage is improving reconstruction.

I don´t know exactly how it works with PC playback but i guess here also the data gets calculated down in level before playback.

I simply don't know. Moreover, as far as I can see that's exactly where this thread is about, quoting the OP:

Benchmark has released a consumer DAC that (among other features) allows 3.5dB of headroom to account for correct reconstruction of intersample overs (peaks between samples that exceed 0dbFS, which are said to be "common in commercial releases")

Reading through the posts so far nobody else seems to know.

[lvqcl]

Several posts in this thread are relevant to this discussion

[bandpass]

Sounds like this DAC is just what I need for my amp that goes to 11.

[Dynamic]

Thomas Lund has measured intersample over distortion in many high end DACs he tested, according to this portion of his technical presentation in a YouTube video where he demonstrates intersample over distortion in a NAD C 520 playing a 11.025 kHz sinusoid constructed to produce continuous intersample overs, viewed on an oscilloscope.

However, he's addressing a lot of issues from the precautionary principle and the idea of preventive measures in audio production engineering, hence the reason for measuring distortion in iTunes AAC and playing the S "side" signal of M/S stereo decomposition, with the message (eventually) coming that use of lossy sources in production should be avoided, but is perfectly fine for distribution (it felt like a massive TOS8 violation for most of the video until that became clear!).

In other words this is all about measurability, not proving audibility, and having tried and failed to ABX digital clipping of a sample-or-two's duration once or twice in the past, I'm tempted to think that audibility is unlikely. It certainly pales against the clear audible damage done by the Loudness War.

Nonetheless, for good engineering design, since I saw this video while bearing EBU-R128 intersample peak measures in mind, I've also thought it's eminently sensible to design any oversampling DAC to incorporate a fixed attenuation during the upsampling phase, potentially compensated for by the analogue circuitry. The upsampling and reconstruction filtering is performed with many multiplications in any case in floating point or high-depth fixed point in DSP, so a fixed multiplication or scaling of the filter coefficients to provide maybe 2-6 dB of headroom above digital full scale to the rail voltage of the op-amp is potentially relatively easy.