Baffled and Frustrated: Streaming/DAC Sound Issues

@cleeds 

Wikepedia:    A Fourier transform (FT) is a mathematical transform that decomposes functions into frequency components, which are represented by the output of the transform as a function of frequency. Most commonly functions of time or space are transformed, which will output a function depending on temporal frequency or spatial frequency respectively. That process is also called analysis. An example application would be decomposing the waveform of a musical chord into terms of the intensity of its constituent pitches. The term Fourier transform refers to both the frequency domain representation and the mathematical operation that associates the frequency domain representation to a function of space or time.

You can read above that the Fourier Transform decomposes the signal.  The example given is the decomposition of the waveform of a musical chord.  That is exactly what I said.

In analogue audio, the waveform is not decomposed.

Clock error in a DA converter is the timing error in recomposing the analogue waveform from the decomposed waveform for expression by an analogue music system.  I am given to believe that it is impossible to remove clock error entirely.  Hence my comment that the pieces cannot be put together again correctly.

All the cheap oscilloscopes in the world cannot change this.  Nor is their definition fine enough to reproduce the results of clock error.

clearthinker

Wikepedia:    A Fourier transform (FT) is a mathematical transform ...

Oh, I understand what a Fourier Transform is, and that it's a theorem, not a theory. I understand both how it works with digital (conceptually) and I understand why the same principle applies to the squiggles in a vinyl LP. You appear to understand neither.

I am given to believe that it is impossible to remove clock error entirely.  Hence my comment that the pieces cannot be put together again correctly.

I understand that you hold certain beliefs. Digital audio is not intuitive - at least not for most of us - so it can take some understanding to reveal that belief and fact can be two different things.

Did you watch the video? it kinda blows apart your theory, doesn't it?

The reason this stuff is important is because digital audio is not perfect and if we want to improve it, we won't be successful by trying to remedy imaginary faults.

I’ve just been reading about "intersample overs" and how they can cause some dacs to "spaz out" on certain tracks. I vaguely understand that if there is a sample that corresponds to minus zero dB in the digital signal, or close to it, the resulting analog waveform can actually exceed minus zero dB, which can cause the output stage to go into clipping or compression. Some dacs have headroom to handle it, some don’t. This matches my experiences, where I’ve had to digitally attenuate some tracks by as much as 6dB to avoid clipping lights coming on.

The dacs that spaz on these overly compressed or too loudly recorded tracks can be called "revealing." One argument is that this is all the fault of the recording because anybody who actually knows what they are doing would never include samples approaching so closely to minus zero dB. We have lots of headroom in digital so there’s no need for it.

@audphile1  mentioned making sure sample rate in Roon settings match what your DAC is capable of, and this was a problem I had.  Better matching of those levels fixed it for me.  Worth a try.

@brewerslaw so…any updates?