TuneInSoul
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It depends
BTW, interesting your list of DAC with minimum phase and slow (my choice too). The logical thing is to assume that any modern DAC has the minimum and linear implemented in its two variants but...
https://yabb.jriver.com/interact/in...HPSESSID=d92m3jk4nn3sf3s1krtru594f5#msg785295
For that topic with JRiver I searched Stereophile for known brands of DACs that have the minimum phase, slow roll off so that I could argue the relevance of it. But actually, from my previous post, you can see that I came to these settings that have lower post ringing as the typical minimum phase slow roll off.
This is not true on several levels. The traditional SNR definition of an A/D does not tell the right story here and you need to consider capturing the noise spectral density (NSD) over the audio range. You then need to factor in dynamic headroom and equalization. The folks at Channel D (http://www.channld.com/puremusic/ ) have an AES presentation on the issues and I wrote a short article with some data and actual measurements. If you want to do digital RIAA 24/48 would be minimal IMO. If you want to do real time using IIR filters 24/96 is better since it reduces the frequency warping effects. There is a table with IIR coefficients that I computed for all the classical equalization curves at https://linearaudio.net/articles , you never need more than two basic bi-quad filters to get (I admit) absurdly close to perfect.I am sure you are aware that 12 bits at 44.1 kHz is enough for vinyl....
This is not true on several levels. The traditional SNR definition of an A/D does not tell the right story here and you need to consider capturing the noise spectral density (NSD) over the audio range. You then need to factor in dynamic headroom and equalization. The folks at Channel D (http://www.channld.com/puremusic/ ) have an AES presentation on the issues and I wrote a short article with some data and actual measurements. If you want to do digital RIAA 24/48 would be minimal IMO. If you want to do real time using IIR filters 24/96 is better since it reduces the frequency warping effects.
If you want to do digital RIAA 24/48 would be minimal IMO. If you want to do real time using IIR filters 24/96 is better since it reduces the frequency warping effects. There is a table with IIR coefficients that I computed for all the classical equalization curves at https://linearaudio.net/articles , you never need more than two basic bi-quad filters to get (I admit) absurdly close to perfect...
Hmmmm. Sorry for being late to the party but this seems to me to be an odd summary. Out of three music samples, in two there was a mild non-statistically significant preference for MP and in one it was for LP. The nearest to a significant result was a preference in one track for LP amongst headphone listeners (I'm not sure that multi factor adjustment was made to the significance threshold). Again there was an overall (across the three tracks) non significant preference among headphone listners for LP and among loudspeaker listeners for MP.Archimago had the following conclusion in part III:
As we know last week, statistically, indeed there was a preference for minimum phase filtering, but not strongly significant based on statistical data.
Sure, for recording and processing, you do want to use 24 bits and higher sample rates - but once you are done, you can definitely normalize to 16/44.1 without losing any audible information.
The choice of the 16 bits and 44 kHz was due to a compromise of what was known at the time and what the technology allowed. If they could change what was decided then they would have risen to 48 kHz. At that time 24-bit was not possible.
Many recent DAC's allow user control over these parameters as this feature may be enabled in AKM and ESS chips. I do recall Bryston makes a DAC with an AKM chip which is permanently set to minimum phase, slow roll off.
Yes, many DACs have these filters for users to choose. Wyred 4 Sound and Matrix Audio too, for example.
But my point is that the DAC filters are too "binary", all or nothing, combinations of fast/slow roll off, linear/minimum phase. The beauty of Resampler-V is choosing intermediate values and seeing how they behave on the graphs.
To finish the vinyl story, it is better to stay in the 24-bit to avoid subsequent dithering, which implies a new soft processing of the audio.
Yeah, the old story is the Head of Sony wanted to be able to fit Beethoven's 9th on a single CD so he could listen to it in his car so we wound up with the odd number of 44.1. DVD's and most other cinema are done at 48k, although they use compressed formats.
A caution about bit depth, I don't know of any A/D with and ENOB near 24bits so 32 bits can mean two things 24 bits in a 32 bit integer container with the lower 8 bits 0 or meaningless, or it could be 24bit integers converted to 32bit floating point. Since the mantissa here is 23bits when normalized 32bit floats don't have much more real resolution than 24bit integers.
That is a non-issue with vinyl - vinyl already has a noise floor that is so high that it is pretty much self dithered at a less-than-16-bit level.
What is heard in the attached file is the following:
1- Without "Dither", that is, the quantization errors to hair.
2- iZotope Ozone 5, MBIT + algorithm, noise shaping "ULTRA", dither amount "NORMAL".
3- PSP Xenon, noiseshape "B".
4- Voxengo Elephant, noise "GAUSS", shaping "EQUAL".
5- FabFilter Pro-L, noise shaping "Optimized".
6- Waves IDR.
7- Waves L2, type 1, shape "NORMAL".
8- Steinberg / Apogee UV22 HR.
9- Sonnox Limiter, type 3.
10- Kjaerhus audio MPL-1 Pro.
My DAC has a switch to select linear vs. minimum phase reconstruction filter. It selects either of 2 filters that are built into the WM8741 chip. With most music I can't hear any difference, much like the previously referenced Keith Howard article about the guys at Stereophile. It's possible that most music just doesn't highlight the difference and we're wasting our time trying to hear differences unless we're listening to tracks having the right kind of sounds.... I invite everyone to choose a 44/16 lossless song of your choice with a loud and clear sounding drum kit, with a consistent snare rhythm. ...
Listen to the 3 new files, paying attention to the snare sound, and see if you can hear the difference. Have fun!
Within "academic only" relevance, I found that current 24-bit D/S-DACs like the AKM4490 actually DO output all 24bits with great linearity when extreme time-domain synchronuous averaging is used (look this thread for pics) to reduce all uncorrelated analog noise. Therefore I don't think it is impossible to get even more from a D/S-modulator even though this is utterly useless in normal practice, of course.A caution about bit depth, ...