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However, I've not seen other measurements done on DACs having full scale 20KHz squarewave applied
20 Hz (see his description). 20 kHz would not even look like a square.
However, I've not seen other measurements done on DACs having full scale 20KHz squarewave applied
thank you very muchAtomic Bob has found clipping with the XSP. Very surprised Matrix let this slip through. Comments @MatrixAudio or @amirm? Was this fixed in the MQA version?
https://www.superbestaudiofriends.o...-pro-technical-measurements.9310/#post-300976
Yes.This is simply the intersample clipping.
It's only normal behaviour of a DAC to a clipped recording! It will never happen with unclipped recordings so I wouldn't say that this is normal.It's a very normal behaviour.
Yes, reduce the volume be 3 dB and you're mostly good.For DACs with an adjustable volume, it's a none issue.
At 44.1 kHz samplerate a 20 kHz square would be very nice sinus. Next harmonic component is at 60 kHz and should be (hopefully) suppressed.20 Hz (see his description). 20 kHz would not even look like a square.
Not really:It will never happen with unclipped recordings
It's not a slip (the cause is a clipped input signal so it's no wonder that the DAC clips as well), although some DACs (see here) can handle this situation better.Atomic Bob has found clipping with the XSP. Very surprised Matrix let this slip through. Comments @MatrixAudio or @amirm? Was this fixed in the MQA version?
https://www.superbestaudiofriends.o...-pro-technical-measurements.9310/#post-300976
This is a clipped recording.
Same here. If the white noise was generated by an algorithm using random numbers you have to be very careful not to violate Nyquist and clipping.
If you scroll down, you will find an audio clip I attached. It is not recorded by an ADC, it is a chiptune -- synthesized music.This is a clipped recording.
Did you see the spectrum screenshot I attached? There is a lowpass before Nyquist.Same here. If the white noise was generated by an algorithm using random numbers you have to be very careful not to violate Nyquist and clipping.
My test signal is LOWPASSED. See the screenshot carefully. The one on the right, with purple color:
Then the synthesizer clipped the signal, even with sample values below 0dBFS.If you scroll down, you will find an audio clip I attached. It is not recorded by an ADC, it is a chiptune -- synthesized music.
Yes, but the sample still is clipped.Did you see the spectrum screenshot I attached? There is a lowpass before Nyquist.
I do not dispute this. Still the noise signal may be clipped. The left side does not show the analog signal. It shows the display of the digital values which is not the same as the analog signal it represents.My test signal is LOWPASSED. See the screenshot carefully. The one on the right, with purple color:
Synthesizers don't have to be bandlimited, especially with the oldskool chiptune genre:Then the synthesizer clipped the signal
And therefore the clipping threshold is defined by the DAC, and it is unfair to call the digital file "clipped". If a DAC has a filter that generates higher peaks than others, then that DAC will clip, but not the other DACs.I do not dispute this. Still the noise signal may be clipped. The left side does not show the analog signal. It shows the display of the digital values which is not the same as the analog signal it represents.
This is where our opinions differ.And therefore the clipping threshold is defined by the DAC, and it is unfair to call the digital file "clipped". If a DAC has a filter that generates higher peaks than others, then that DAC will clip, but not the other DACs.
Yes, something like this:This is where our opinions differ.
I look at the complete audio chain, ADC -> processing >- DAC:
- An analog signal captured by an ADC must not contain frequencies above Nyquist, and its level must not be higher than the ADC can handle.
- Such an unprocessed digital signal can be transformed back into the analog world by a DAC without clippling of the DAC or its analog stage as long as the DAC is properly implemented. This is how digital audio should work in the first place (I know it doesn't).
- If the digital signal is processed in any way the processing algorithm must ensure that such a DAC does not clip, otherwise the processing is bad.
- The same is true when generating artificial digital signals, being it test signals like the 20 Hz square wave, or chip tune music.
Let's think in another way. If a DAC does not clip at full volume when playing test signals like square wave, then it just means it has a forced digital attenuation stage before going through the digital filter. It is not really "bypassed" or "disabled".
Jds Labs' view:
On the other hand, if a DAC does not have a forced headroom, then you have full control of the digital attenuation yourself.
See for yourself:How did Benchmark prevent clipping in the DAC3? I don’t think the implication is that thr DAC3 isn’t bit perfect, is it?