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High Resolution Audio: Does It Matter?

j_j

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Oh bless - what is the purpose of our audio systems? I assume we are all music lovers who want to enjoy and appreciate the talents of our preferred musicians.

Or maybe it really is just about the science . Pretty sure the performers on the recordings didn’t have that in mind :(

In general, we want to hear what we prefer. Enter science.
 

CapMan

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It’s science that made it possible to enjoy all that fabulous music.
A fair point, well made. That said, ‘science’ has enabled many fine performances to be captured that would not be considered hi-fidelity by today’s benchmarks. Listen to Louis Armstrong‘s Hot Five recordings - sublime musicianship - basic fidelity.

I hate the nonsense that the HiFi industry spouts which this forum serves to expose , my sincere thanks for that. It is refreshing.

I also think that the quality of musical performance trumps everything - even science, and that just felt a little lacking in this thread! Peace.
 

Raindog123

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“I just like to go in that car (airplane, boat) wherever I want! Fast, reliably, and safely! Who needs those scientists and engineers that made it happen!?”
 
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voodooless

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I also think that the quality of musical performance trumps everything - even science, and that just felt a little lacking in this thread!
Well, yes :) Because this thread is not about the performance of the artist. There are other threads that are about that though.
Most def!
 

CapMan

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“I just want to go in that car (airplane, boat) wherever I want! Fast, reliably, and safely! Who needs those scientists and engineers that made it happen!?”
Oh Lord. What did I start. I have a PhD In Chemical Engineering, I love science and engineering. No offence was intended.
 

voodooless

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Typical Poe’s Law at work here :rolleyes: Shit happens…
 

BDWoody

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Oh bless - what is the purpose of our audio systems? I assume we are all music lovers who want to enjoy and appreciate the talents of our preferred musicians.

Or maybe it really is just about the science . Pretty sure the performers on the recordings didn’t have that in mind :(

I suppose talking about some of the science related to the reproduction of music, with one of the handful of leading scientists in the field, on a site called Audio Science Review, is too far a chasm for you to cross?

Sorry you weren't entertained.


Indeed.
 

CapMan

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I suppose talking about some of the science related to the reproduction of music, with one of the handful of leading scientists in the field, on a site called Audio Science Review, is too far a chasm for you to cross?

Sorry you weren't entertained.



Indeed.
Yep - I am a numpty . Sorry I couldn’t cross that space.

Yours truly
A patronised forum member xx
 

Raindog123

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I think I said I would listen to some music. Hey ho. Sad to get such reaction to a post. I’m done . Peace .

This is what you said, literally: “I think I’ll just go away and enjoy listening to some music.” We still have our memory intact, well most of us. :)

Seriously, you can go or stay. We welcome everyone - especially PhDs in Chem Engineering, ie people with a potential for critical thinking. What we welcome less is primadonna‘s with the “how dare you‘re not agreeing with me, I am leaving!“ tude.

But I am not your dad… I hope. :)
 

CapMan

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Why so aggressive Raindog - I have a point of view, maybe not the same as yours. I’m not trying to be offensive or contrary,
I just love music.

It’s appreciated that you have set out the rules for membership.
 

voodooless

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Actually no. I did wonder about the thread from the perspective of musical performance
Can you elaborate how that perspective works? What in your mind does the musical performance have to do with high resolution audio?
 

CapMan

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Hi - On reflection, absolutely nothing. A great performance is a great performance , which I will happily listen to on my iPhone , in my car, on my Sonos or on a Bluetooth speaker.

Thanks for clearing my mind. Have a great weekend all.
 

theREALdotnet

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It's that 'first detection' that matters, and the "leakage" in the filters (small for the FIR, not so much for the most HF cochlear filters) that also can matter.

Most interesting!

From a practical perspective, would it be generally advisable to avoid those digital filters inside DACs that produce substantial pre-ringing? Or are there other trade-offs that muddy this pond?

As an aside and not directly related to high-res audio, I found this presentation by Jim Hudspeth about the physiology of hearing, down to the molecular level, very fascinating and entertaining:
 

RichB

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From a Secrets interview with Benchmark Media @John_Siau:

Q: ESS offers several oversampling filters on their DAC chips. Per the Secrets review, you expose one filter which is a “textbook Fast Roff-Off Linear Phase type with excellent attenuation of the first reconstruction tone”.

Was this choice driven by subjective and/or objective evaluation?

Digital Filter test result, Benchmark DAC3 B from Secrets review
Digital Filter test result, Benchmark DAC3 B from Secrets review

Both.

Some engineers attempt to “improve” the time-domain response by relaxing the brick-wall filter. This appears to work when looking at the square wave response with an oscilloscope, but it changes the sound of the recording by causing phase and amplitude modulation of high-frequency signals. If you relax the response of the filter, transients move toward the nearest clock transition, damaging the time-domain response. Relaxed filters will sound different because they are changing the sound. You can cascade our A/D and D/A converters through many A/D – D/A cycles without causing a change in sound. Try the same thing with a relaxed filter response and it will not work!

As a side note, we are not using the ESS filters, nor are we using the ESS DAC in a conventional manner.


Q: You mention in an article that you operate the DAC2 and DAC3 at a fixed 211 kHz sample rate. This is interesting in that fractional sample rate converters are often avoided. What was the reasoning behind this choice?

We frequency-shift the ESS lowpass filter so that it is always above the Nyquist frequency of the incoming digital audio. We replace this lowpass filter with a sin(x)/x reconstruction filter that is implemented at a gigahertz rate within the ASRC processing. We upsample to gigahertz frequencies, apply the sin(x)/x and then downsample to 211 kHz while applying a proper low-pass filter. The key to this process is the extreme oversampling ratio that we use for reconstruction.

I recently picked up a used DAC3 and love it connected to the LA4.

- Rich
 

j_j

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Most interesting!

From a practical perspective, would it be generally advisable to avoid those digital filters inside DACs that produce substantial pre-ringing? Or are there other trade-offs that muddy this pond?

As an aside and not directly related to high-res audio, I found this presentation by Jim Hudspeth about the physiology of hearing, down to the molecular level, very fascinating and entertaining:


Unfortunately, using a minimum-phase filter can change the envelope of the high frequency components (as it must). So lose some one way, lose some another. Consider a three-tone complex at 18khz, change one of the sidebands, you change the envelope. Can one hear this? Maybe yes, certainly if one contrives the right signal to make it audible. Same power spectrum, different phase, sounds different because the signal envelope in a particular ERB is changed. That is, audible if one can still hear that frequency range. It all comes down to the issue that if one has a particular transition bandwidth, there is also a central lobe of the filter with corresponding inverse width (i.e. for a steeper filter, the impulse response is longer). Mathematics is very clear about that.

Some folks have proposed a combination of filter, some minimum phase, some constant delay, that might be better. That's the claim, at least.

To explain further, a constant delay filter has a zero pair inside the unit circle matched exactly by a zero pair OUTSIDE the unit circle at 1/roots of the one inside. When one flips one set of roots inside, that changes the phase response of the filter toward minimum phase, one zero pair at a time, while holding absolutely (mathematically to limits of calculation in fact) the same frequency response in terms of magnitude (but not phase). Given that we are talking of filters with many, many pairs of zero pairs, it is possible to have many "optimum" responses that have variation between minimum phase, constant delay, and maximum phase. (where all of the zeros are outside the unit circle).

I will point out that the whole discussion here is based on POSSIBLE mechanisms, there is no evidence (beyond some old rate convertors and DAC's with terrible filters that are not in the present day) yet that there is anything to this, beyond perhaps a slight difference at 44 vs. 48, and doing that in a fair test is an outright (*(&*((&*. So remember that this is to some extent speculation, based on an unpublished work by Tom Stockham, who is quite sadly no longer with us to help out. In the modern day, nobody is concerned, some defend the status quo, and some ask for ridiculous sampling rates. There is a major lack of dialog. Even a negative result in a proper test would advance the science.

On the other hand, I'm a bit skeptical about the "active amplification", there are other hypotheses, for instance Zwislocki's, that explain the same behavior, and that provide the compression via other mechanisms, and also explains the frequency resolution. The problem with active amplification comes from signal to noise ratio issues. This is a long discussion and I won't say for either side is completely right (I suspect some joint solution, frankly, with both detuning via outer hair cells as well as perhaps less amplification via the "active" method. But, in either case, there surely is active behavior, the question "how" remains.) Likewise, it's clear that there are several sources for tinnitus, not just "oscillation". But this is to some extent picking nits, both models result in both the right sensitivity and frequency resolution, as well as as modelling things like auditory masking.

I like that he points out that 0dB SPL is effectively a pressure variation of about 2 * 10^-10 atmospheres, and that 100dB SPL is about 2*10^-5 Atmospheres. Many people do not understand how close to the very unavoidable molecular noise that we can hear.
 
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Mnyb

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It’s my understanding that the pre ringing seen in the typical curves presented in for example DAC tests appears when using the step impulse signal ? The signal is designed to test filters and the curve is a telltale for the expert to interpret which kind of filter we have and how well excecuted it is in a given product ?

But it does not happen at al with properly bandwidth limited signals such as the music you actually play ?

But it’s used in hifi marketing to give us noobs the impression that there is some kind of pre echo of transients going on all the time when using “normal filters” and sell us everything from MQA to filterless NOS DAC’s and every other unorthodox filter approach?
This pre ringing impulse and ringing square waves is often used deceptively in marketing against audiophiles with limited understanding of the theory (such as myself ).

Bandwidth limiting and filters are part of digital audio in its practical use, but not necessarily the other way around.
You can demonstrate many of these effects in an analog system to ?
 
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