@Rob Watts We met at CanJam SoCal 2022, and it was great talking to you. I'm eagerly anticipating the Quartet MScaler. Based on my experience listening to Chord DACs and observing the progression from Mojo to TT, TT2, DAVE, and Blu MK2/MScaler, it became quite evident how the longer WTA...

I agree with this If you really narrow the scope of what Signal processing means to your previous statement then yes. Because applying a digital filter for the purpose of removing out of band images is still signal processing. I agree with this. However, showing the relationship between...

Thank you for your suggestion, I am already a member of AES and thanks for your time.

But, if there is not an established method to measure transparency that is published and researched, where does one draw a line on audibility? Nothing is as clear cut, there is always a fuzzy boundary and the width of the boundary may depend on what is being tested and also the system as a whole...

No it is just one parameters, which is why I said everything else being equal. You can have a nimble transducer, but if you have a poorly designed magnetic circuit and introduce local modes, then distortion is assured. If you have two of the same design with the same distortion issues, the...

That just brings it back a full circle and my original concern that there is not enough focus on time domain response and more work needs to be done in measurements and understanding. We know what true transparency in the theoretical sense is but a true test for the degree of transparency does...

Transparent means you cannot see it is there and in the case of audio, you should not hear it and all you hear is the source (i.e., the sound that was recorded) If a system's response is a Dirac delta function it means it simply reproduces the input at the output, that is both frequency and...

I was commenting on only using frequency response to form an opinion. Yes the frequency domain response includes the phase response but a phase response is rarely shown during measurements and it is even harder to interpret or use to form any kind of onion. I view measuring the digital filter as...

Yes impulse response holds the key to some of the questions, which is why I have been talking about the time domain response. I have looked at impulse responses from different headphones and overlaid them too, once you EQ the headphones and look at the impulse response you know they are not the...

Is there a way to tell this from measurements? or measure the degree of openness? If an electrostatic headphone, a planar and a dynamic headphone are all eqed to have the same bass will they all sound equally 'dynamic' is there a way to measure this? Are there any measurements that help one...

On the contrary, that is exactly my point, but most of the analysis and measurements are done to highlight the frequency response. In theory yes you can go back and forth an infinite number of times, but for measurements, a lot of information is lost as FFTs are done with fixed window sizes and...

I was just helping keep the 'eye on the ball' so to speak. It is easy to loose focus when there are varying opinions and many of them can be correct too. Theory helps understand what is happening to the signals, how we perceive it is another story. For example if a process improves a...

In theory, does it correct artifacts? yes, it can correct some artifacts. Have I heard it correct artifacts? I can't hear past 16kHz. What question was this whose answer is oversampling? PS: Both my questions were meant to be rhetorical.

A case can be made for an apodizing filter where the filter cuts off at 20kHz and opening up the possibility of reducing any aliased frequencies that folded back due to the ADC's filters not being good enough. So, objectively it could be argued that an apodizing filter can 'correct' for some of...

An impulse or a square wave or a step function are some examples of 'illegal' signals because their analog equivalent will have an infinite bandwidth (it is a sudden instantaneous change with infinite slope). If they had ever truly passed an ADC (even a mediocre one at that), you would still see...

A non-apodizing filter makes most sense when it is sinc based, following Whittaker-Shannon interpolation, which would necessitate that it is non-apodizing. Any deviation from this will no longer guarantee perfect reconstruction as per Whittaker-Shannon. The idea is quite simple, you need to use...

Yes, latency is an issue with linear filters, but for normal music listening, it is not an issue (like one would do with Roon). I do not know of a way to use Roon for video.

I understand the frustration, we have always outsourced the Reveal plugin development as we are not a software shop. We provide the presets (just like we do to Roon). Initially we used contractors to create the Reveal plugin, later we collaborated with Embody and they integrated our presets...

Tonally they should be really close but from my experience, the difference is mostly in clarity and imaging. With a resolving setup, the difference in clarity between PEQ and linear filters becomes more obvious.

I recommend using Roon's Audeze presets instead of plugging in the PEQ values. Especially, using the linear presets will result in more accurate reproduction of Bob's EQ. They were derived directly from DMG Equilibrium that I too like to use, based on Bob's recommendations. A lot of work went...

One would need 16M taps at least to go directly from 1FS to 256FS and still retain the same performance. I see cascaded filters as a compromise similar to adding more elements to a lens, it may make the design easier but degrades transparency unless they are perfect, which will be very hard to...

But you assume the issue is one of speed and not space. FFTs will help speed for sure, but will not require any less space, if anything it would require more space. Using FFTs also come with it's own set of challenges, fixed point FFT algorithms need to be implemented and OA or OS algorithm...

Noise shaping was invented for a reason, it is also part of the A/D process. When a signal is processed (like in an upsampler or the DAW), you add more noise due to requantization. Say you take the 44.1kHz 16bit signal to equalize it and do the math in 32 bit or 64 bit float and then convert it...

The previous poster implied that the timing accuracy was affected by bit-depth and not the sample rate, while that may look like it is true at first glance, all I am doing is pointing out that higher sample rates can help timing accuracy because noise shaping will allow a higher effective bit...

I don't think one can dismiss the importance of higher sample rate that easily. Depending on the type of processing, the effective bit-depth can be influenced by the sample rate. DSD is single bit, and can only represent two levels 0 or 1, and yet thanks to noise shaping (done at 64x CD rate)...