ASR Audio Science Review forum YouTube Channel

[pkane2001]

3 hours ago, PeterSt said:

 

Paul, you should start making DACs. This is really the strangest BS 😉 of all times. Really. Undoubtedly unintentional, but it is (again) getting more and more crazy in this thread.

 

Ask Miska why he implemented a filter A, B, C etc. Try to drag out of him that he did that to create a certain sound. I'll bet you he won't say that. He merely would say something along the lines of:

- Distortion shifts to an other location, less harmful to you if you are sensitive to it.

- Filter B is an improvement of filter A. A remains for those who like it.

- Filter C is a general improvement for XYZ.

And no filter is perfect because that can't exist in reconstruction.

 

Happy to lose the bet. I don't care. I have my own filters. 🤗

 

Peter, I don't know what you're arguing against. Filters are designed. Certain filters are designed poorly, less accurately, causing images, and ringing, oscillations in frequency and phase in the pass band, etc. Some of these are designed because someone thought they sounded better. Some are designed this way because they're easier to implement. If you're arguing that these don't exist, you've not measured many DACs.

[Miska]

Regarding HQPlayer, too many posts about the topic, so instead of replying to many I explain how I look at and do things.

 

Let's say DAC chip datasheet states that it's digital filter has stop-band attenuation of -100 dB (already very good for a DAC chip) at f=0.55 and pass-band ripple of 0.01 dB. And this produces 352.8 kHz output rate from 44.1k input. Above that, the sample is just copied multiple times (S/H aka zero-order hold) to produce MHz rate.

 

And then I create filter that has -240 dB attenuation at f=0.5 and pass-band ripple less than 0.000000001 dB. And this is used to produce 11.2 MHz sample rate. No S/H, linear interpolation or such. Is it more accurate? (note though that you can design multiple different kind of filters with these same properties)

 

I also measure performance of various DACs, both with their built-in DSP and with my external DSP to see how they perform.

 

My goal is to squeeze as much performance as possible out of a DAC and overall playback system. In a way that also subjectively sounds good, to different people, listening different kinds of music.

 

[pkane2001]

3 hours ago, PeterSt said:

 

There is nothing like "poorer". There's just an enormous difference in approach between e.g. genuine NOS (no filter at all) and Rob Watts with 1M filter taps. Let's say that DSD is somewhere in that spectrum too, working with different properties (like upsampling way more, right into the DSD domain).

 

What you guys seem to have fortotten (in the midst of what should be 10K posts about it, if not way more) is that NOS is accurate towards the time domain and 1M taps is accurate towards the frequency domain. The more "accurate" in either of these ends of the spectrum, the more inaccurate the other end will be.

So it is the skill to be something in the middle (anywhere but not at one of the ends) and make the one the best for your (designer's liking) and the other the least harmful technically.

 

Miska will tell you the same. It is only that my liking is towards the time domain and his liking is towards the frequency domain.

Nice eh ? two blokes ever back starting the same thing, but with different technical aims. Both worked out to its best. Both overruling what happens in-DAC. that preferably being a NOS/filterless DAC because it does nothing in the first place.

 

Are you telling me that 1M tap linear-phase FIR filter will be inaccurate in the time domain? Really? Can you explain how that happen when phase is kept linear? I'm curious.

[PeterSt]

1 hour ago, pkane2001 said:

 

Peter, I don't know what you're arguing against. Filters are designed. Certain filters are designed poorly, less accurately, causing images, and ringing, oscillations in frequency and phase in the pass band, etc.

 

That is what *I* all said or clearly implied (not you).

 

1 hour ago, pkane2001 said:

Some of these are designed because someone thought they sounded better.

 

That is what *I* said too (with Miska as subject) (not you).

 

You said this:

 

Quote

Some are "designer" filters meant to add something to the sound

 

And that is what I called "BS" between quotes and with wink.

I don't see anybody do that explicitly. Not even myself. Haha.

 

Btw, I am never into endlessly repeating what someone said (differently). So this is enough for me for today. :-)

[PeterSt]

1 hour ago, pkane2001 said:

Are you telling me that 1M tap linear-phase FIR filter will be inaccurate in the time domain? Really?

 

I see. You just invented the perfect filter. It is 100% good in the frequency domain as that it is 100% good in the time domain (assumed that a 1M tap filter is any sort of 100%). A world's first.

Or would you attest that the 1M tap filter sucks in the frequency domain ? Only then you could theoretically be right.

 

How are the measurements doing ? (decoy attempt)

[John Dyson]

2 minutes ago, PeterSt said:

 

I see. You just invented the perfect filter. It is 100% good in the frequency domain as that it is 100% good in the time domain (assumed that a 1M tap filter is any sort of 100%). A world's first.

Or would you attest that the 1M tap filter sucks in the frequency domain ? Only then you could theoretically be right.

 

How are the measurements doing ? (decoy attempt)

1M tap filter wouldn't likely theoretically meet the specs of a 1Mtap filter, but would likely be pretty good.

I use double precision math above about 1500 taps for a wideband Hilbert transform, but the coefficents are a little different, and my need for precsision is extreme (errors create distortion in my application.)

 

 

[pkane2001]

4 minutes ago, PeterSt said:

 

I see. You just invented the perfect filter. It is 100% good in the frequency domain as that it is 100% good in the time domain (assumed that a 1M tap filter is any sort of 100%). A world's first.

Or would you attest that the 1M tap filter sucks in the frequency domain ? Only then you could theoretically be right.

 

How are the measurements doing ? (decoy attempt)

 

I didn't say it's perfect, but it can be as perfect as you want and doesn't require any new invention or guessing. A 1M tap linear-phase FIR filter doesn't suck. Theoretically speaking :)

 

Measurements are going well. I have a few samples of Lush^2 ready for blind testing, just going through some final adjustments and testing :)

 

[pkane2001]

18 minutes ago, PeterSt said:

And that is what I called "BS" between quotes and with wink.

I don't see anybody do that explicitly. Not even myself. Haha.

 

What do you think about these two filters? Actual implementations, measured at the output:

 

 

 

[Rexp]

2 hours ago, Miska said:

 

Except that in practice nobody will ever have the exact PCM playback chain the Artist had in mastering studio. Already for the reason that there are way too many studios that people could possess the same gear and playback chain. Even less so the same acoustics.

 

But say the consumer did have the same PCM chain and listened on the same headphones, the goal of recreating what was heard in the mastering studio would be achieved. 

[Miska]

45 minutes ago, PeterSt said:

That is what *I* said too (with Miska as subject) (not you).

 

Yes, it doesn't mean giving up on objective side. As I said earlier, you can have two filters, both objectively equally good and sound different.

 

36 minutes ago, John Dyson said:

1M tap filter wouldn't likely theoretically meet the specs of a 1Mtap filter, but would likely be pretty good.

 

Sometimes it exists because people want to have a specific number (and because one company happens to have such filter in their hardware product). Not that it would be my longest one though.

 

But pretty decent in my opinion

for it's purpose.

 

[Miska]

1 minute ago, Rexp said:

But say the consumer did have the same PCM chain and listened on the same headphones, the goal of recreating what was heard in the mastering studio would be achieved. 

 

That is pretty theoretical, or you need to stick to listening very few recordings. Or end up buying a lot of hardware. You'd likely also need to play through the same DAW as well with the same plugins.

 

If that's your goal in first place. My goal is to exceed the quality and performance that was available in the mastering studio and hear also the details that were not even audible in the mastering studio. To really hear what is in the recording.

 

 

P.S. Many times the mastering guy didn't involve the artist, especially for all the export formats such as RedBook and MP3. Could be that the artist was already dead when the latest mastering was done.

 

[The Computer Audiophile]

10 minutes ago, Rexp said:

But say the consumer did have the same PCM chain and listened on the same headphones, the goal of recreating what was heard in the mastering studio would be achieved. 

 

What are the odds anyone has the exact same system, same power, same cables, same DAW, and same room, same hearing profile in your ears/brain, etc... ?

 

[Rexp]

35 minutes ago, Miska said:

 

 

 

My goal is to exceed the quality and performance that was available in the mastering studio and hear also the details that were not even audible in the mastering studio. To really hear what is in the recording.

 

 

 

 

You don't think its possible to hear whats on the recording at a mastering studio? 

[Miska]

8 minutes ago, Rexp said:

You don't think its possible to hear whats on the recording at a mastering studio? 

 

Maybe, maybe not. But if you have better system and better ears, you can hear more than they did.

 

[jparvio]

1 hour ago, Miska said:

 

...My goal is to exceed the quality and performance that was available in the mastering studio and hear also the details that were not even audible in the mastering studio. To really hear what is in the recording.

To emphasize this; about three decades ago I had a chance to hear one of my then fav records played through a hefty high end system. The cd had a famous classical guitar pieces played by a capable chap. In fact the cd was from a rather famous studio. What I had not heard before through my tiny ProAc´s and Cyruses was now as clear as day; apparently during the recording sessions there had been a garbage truck emptying the trash. First I thought it was outside during my listening, but pausing the cd made the truck pause... So, I seems to happen to the best of us - or them to be more precise :)

 

[danadam]

22 minutes ago, jparvio said:

What I had not heard before through my tiny ProAc´s and Cyruses was now as clear as day; apparently during the recording sessions there had been a garbage truck emptying the trash.

How about after that? Did you still not hear it, now that you knew it's there?

[jparvio]

18 minutes ago, danadam said:

How about after that? Did you still not hear it, now that you knew it's there?

 

No, those small speakers at that time and in those acoustics were not able to reach low (or loud) enough no matter what I did. I even tried Adcom´s 500-series amps. It was not to be. But luckily enough I was not forced to settle for those circumstances for too long. Knowing it was there made me hasten the steps for better (equipment and acoustics).

 

EDIT: Don't get me wrong; ProAc´s are excellent speakers and during the Years I've owned several models. I still have fond memories of SC´s and Tablettes and would welcome any of them again. Different horses for different courses... 

[Bill Brown]

Reminds me of the subway under Kingsway Hall on the old EMI classical recordings.  I had read about it, but even then when I heard them the first time it took me a while to recall what they were/realize it was them.

 

Bill

[charlesphoto]

1 hour ago, Rexp said:

You don't think its possible to hear whats on the recording at a mastering studio? 

No, not really, but it probably depends on the music as well. Back in the day, the likes of Phil Spector would have an AM radio wired into the studio. If it didn't sound good through that, no matter how good it sounded through the studio speakers, then it wasn't going out the door because that's how most people listened to music and how hits were made.

 

I still think some of that thinking goes on. For example, I found Beyonce's big record from a few years back to sound dull and veiled on my system, but I'm sure it probably sounded much better on a phone or a Spotify mp3 via a Sonos, Bose, AV receiver, etc. where it would be most consumed. 

 

So there's no magical straight wire from the recording studio to your home hifi. Lots of different ears and processes in-between. The best you can do is get it to sound good, and even that baseline will vary from listener to listener, no matter the budget or gear. 

[Samuel T Cogley]

14 hours ago, The Computer Audiophile said:

Are you suggesting better measurements and shifting noise way up into the inaudible band isn’t more accurate?

 

But that's a really good point, Chris.  Which HQP filter is the "most accurate"?  Based on my limited understanding of digital filters, I think it's the Closed Form filters, which not many like.  How about you?

[pkane2001]

6 minutes ago, Samuel T Cogley said:

But that's a really good point, Chris.  Which HQP filter is the "most accurate"?  Based on my limited understanding of digital filters, I think it's the Closed Form filters, which not many like.  How about you?

 

"Like" and "accurate" are not necessarily the same. My PKHarmonic VST plugin lets one add arbitrary levels of harmonic distortion to audio. There are many who prefer their music with a dash of distortion, even among the objectivist crowd :)

 

A filter has very well defined mathematical properties. The one that introduces less distortion in the audible band wins the accuracy competition. This can be evaluated mathematically, or by measuring. What makes the closed-form filter better than others?

[57gold]

9 minutes ago, pkane2001 said:

 

"Like" and "accurate" are not necessarily the same. My PKHarmonic VST plugin lets one add arbitrary levels of harmonic distortion to audio. There are many who prefer their music with a dash of distortion, even among the objectivist crowd :)

 

 

Interesting, understand that Mytek's "improved" analog volume on Brooklyn+ adds some distortion to "warm up" the sonics.  

 

ASR disliked that feature, as opposed to just recommending users to stick to "cleaner" digital volume control.

[The Computer Audiophile]

1 minute ago, 57gold said:

 

Interesting, understand that Mytek's "improved" analog volume on Brooklyn+ adds some distortion to "warm up" the sonics.  

 

ASR disliked that feature, as opposed to just recommending users to stick to "cleaner" digital volume control.

 

The digital volume control on the Brooklyn Bridge is far noisier than the analog VC. Connecting my IEMs to the headphone output and switching between the volume controls without anything playing, the noise floor with the digital very audibly increases. 

[Samuel T Cogley]

21 minutes ago, pkane2001 said:

What makes the closed-form filter better than others?

 

I never said "better".  Closed-form is non-apodizing.  I think the whole "accuracy" claim in this context is more hubris than science.

[pkane2001]

7 minutes ago, Samuel T Cogley said:

I never said "better".  Closed-form is non-apodizing.  I think the whole "accuracy" claim in this context is more hubris than science.

 

Not hubris, mathematics.