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Okay, fair point – but are there any technical parameters that typically correlate with this kind of listening impression? For example: slight bass lift, gentle treble roll-off, higher second-order harmonic distortion? Would be interesting to see if there’s a measurable basis for what...

We already know about things like: Higher price Fancier design Wider dispersion More suited for far-field listening But what exactly does "more forgiving playback" mean in technical terms? Is it just a matter of equalizing? Or are harmonic distortions involved?

So, I can ignore any Dithering options in EQs or digital Volume Control and resampling?

Hi everyone, I have a question regarding bit depth reduc-tion, quantization errors, and dithering in a typical audio signal path. As I understand it, dithering should ideally be applied only once-typically during the final export from a DAW, when converting from 64/32-bit floating point to...

In MIDI Management, the clock source is always set to default. Can Core Audio clock directly to the hardware like ASIO, or is there always an asynchronous layer in between?

If I download a song in Qobuz and play it offline and output it to the Dante Virtual Soundcard, will the Dante Clock be adopted from the start or are there any other transfer protocols from Qobuz in the background?

Question for Genelec the Ones users. Is every incoming signal generally resampled to 96 kHz, so also 96 kHz? Are there any advantages to only feeding them 96 kHz?

How? They have to decide on a form of roll off and a filter length and accordingly on a time frequency behavior

How is it possible to judge songs objectively if the native band-limiting characteristics of a song are ultimately overwritten by the DAC filter?

Thank you very much for your time, the matter was unnecessary for me anyway, since a DSP probably performs an ASRC anyway and always resamples everything :facepalm::D

a dac that is clocked to the dsp rate and goes directly into the zero order hold interpolation and dispenses with a regular previous upsampling step so that I can control how it is rolled off after the nyqvist limit

I would use for resampling 44.1 -> 96, 48 -> 96, 96 -> 96 Do you think it is superior to most DAC filters? Do I have to worry about interference? The DAC chip on my Genelec The Ones is AK4621EF.However, I don't know which filter is used internally. Possibly a low...

I had hoped that the DSP would do the basic work and the DAC chip would only be used for pure conversion

Strange. For other AKM codecs or DAC chips, the diagrams always show "X INTERPOLATO" as the processing step. I always assumed that interpolation does not occur in DSP monitors in favor of signal purity. Thank you for the clarification

AK4621EF I'm not a professional, but what can you deduce from this diagram?

But if DSP works at 96 kHz, like in the Genelec The Ones, that means the integrated DAC doesn't oversample, right?

would it possibly make sense to use ascr in an audio player to only feed the monitor with 96 khz? or a halfband filter. I can imagine that the monitor expects a 96 khz signal from external sources and its own internal resampling is only a nice to have

While your explanation is technically correct in that the resampling filter at ~48 kHz does not affect the frequency content below Nyquist (22.05 kHz for a 44.1 kHz signal), it overlooks a critical aspect: the potential time-domain effects of the filter. Specifically, the pre-ringing and...

Hello everyone, I have a question regarding DSP studio monitors that operate at fixed sample rates like 96 kHz. When listening to music in 44.1 kHz, the signal is typically resampled. During this process, the native reconstruction filter of the original 44.1 kHz signal is replaced by the...