I was never happy using pll clocks including dsp-cpu and subsystem clks. Jitter always there.
Several of my friends are 1%ers (measured by net worth) thanks to the PLL clocks they designed. Occasionally they have a get-together at "Conference Room C" -- the bar at the Mountain View Chili's -- and inevitably one of them will raise a toast to Floyd M. Gardner. The man who literally wrote The Book on PLLs.
No I was more asking if there would be a measureable difference between SRC from 44.1/16 to 96/24 in the music server and letting the SRC in miniDSP do it.
Probably easiest to let the miniDSP do it since it will probably do so by default anyway. You might get better performance doing it in a PC, but unless you are very picky about small differences in sound quality miniDSP should probably be satisfactory.
Then again, someone like me might want a different solution. 🙂
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Hi Richard, I am glad to have been shown that digital sources and equipment is so variable. I have been fighting the problems of limited bandwidth, steep filters, bit level, etc for the last 40 years, but I never knew that a really good (reference) digital source was so variable due to other reasons as well. It makes it almost impossible to casually exchange typical files, except for general listening. Now I know that many of you are happy with that, either because your inherent ear-brain capability is more relaxed, or you are just 'used to it' and the problems are ignored. But Richard and I come from listening and producing extremely high quality ANALOG sources, and we are not so easily satisfied. Thanks everybody for showing me this.
just came across this document that delves into the USB audio driver classes. Dated 2015, but I believe is pretty much the current state.
Hopefully it can offer some small bit of clarity.
https://www.diyaudio.com/forums/att...m4222-ak5572-cs5381-etc-usb-audio-class-2-pdf
thanks to Altor for point this out
Cheers
Alan
Hopefully it can offer some small bit of clarity.
https://www.diyaudio.com/forums/att...m4222-ak5572-cs5381-etc-usb-audio-class-2-pdf
thanks to Altor for point this out
Cheers
Alan
AFAIK DSD was originally developed by Sony as a way to store content in their archives, independent of change in future technology.
Then came the idea to bring this "almost analogue" Direct Stream to the market, but there are two reasons why this never happened.
1) To perform the necessary Edit and mix DSD while recording, 1 bit recordings have to be converted into PCM and reconverted back into DSD (this results of course inevitably in some conversion losses).
2) Progress in technology made higher multibit sampling rates no longer an issue, outperforming the 1 bit 2.8Mhz DSD concept.
This all caused the DSD to become a mere marketing issue and real Direct Stream is nowhere in existente at all.
Apart from an added surround sound feature, there is no advantage in DSD or SACD, but you get a lot HF noise to cope with.
This article goes into somewhat more detail.
https://www.grimmaudio.com/site/assets/files/1088/dsd_myth.pdf
Hans
Yes, Eelco's old article is good. He and Grimm no longer make their DSD64 ADC. The DSD recording labels, Pentatone and Channel, now have Horus systems and record in higher bit DSD to shove the HF noise higher. Pentatone now converts to DXD for editing while Jared at Channel continues to do as Eelco describes. Sound Mirror in the US also does some DSD recording for Pentatone and Reference Recordings.
There never has been a lot of DSD recording studios, ever.
BTW, surround sound is a specification for sacd and is not exclusive to DSD recording. Most new sacd's are pcm recorded and have a mch layer on the disc.
...Now I know that many of you are happy with that, either because your inherent ear-brain capability is more relaxed, or you are just 'used to it' and the problems are ignored. But Richard and I come from listening and producing extremely high quality ANALOG sources, and we are not so easily satisfied. Thanks everybody for showing me this.
Come on, man. Is this "golden ears" shade really necessary?
John, with all respect for your analogue contributions in the past, I find your current posting patronizing and completely ignorant on what’s going on in the digital world.Now I know that many of you are happy with that, either because your inherent ear-brain capability is more relaxed, or you are just 'used to it' and the problems are ignored. But Richard and I come from listening and producing extremely high quality ANALOG sources, and we are not so easily satisfied. Thanks everybody for showing me this.
Because, other than you, most of us know what a PC is doing.
That’s why we use software that avoids any of the problems that Windows or Apple can cause.
Therefore all mentioning from your side that 192/24 is so much better has to be taken with a big amount of suspicion.
Hans
Anyone who doesn't use brickwall filters on John's and Richard's posts needs their bits examining
When comparing "extremely high quality ANALOG sources" (as played back on extremely high quality hardware, I assume) one should compare them to extremely high quality digital sources also played back on extremely high quality digital hardware and not on some consumer gear, such as PCs out of the box.
John, with all respect for your analogue contributions in the past, I find your current posting patronizing and completely ignorant on what’s going on in the digital world.
I think it's hilarious when you look closely at his avatar.
...one should compare them to extremely high quality digital sources also played back on extremely high quality digital hardware and not on some consumer gear, such as PCs out of the box.
Chord Hugo TT 2 with M Scaler? Who can afford that?
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Probably easiest to let the miniDSP do it since it will probably do so by default anyway. You might get better performance doing it in a PC, but unless you are very picky about small differences in sound quality miniDSP should probably be satisfactory.
Maybe, but I want to measure it. As at all stages it's digital and I'm not trying to measure jitter it should be relatively easy. The challenge is working out the test file and confounders.
Chord Hugo TT 2 with Upscaler? Who can afford that?
Cheap compared to a 100k full dCS stack. And they seem to sell. At least to hifi reviewers...
Maybe, but I want to measure it. As at all stages it's digital and I'm not trying to measure jitter it should be relatively easy. The challenge is working out the test file and confounders.
Typically, the HD in DSP chip ASRCs is spec'ed a little below -120dB. Of course, incoming jitter and or ASRC PLL power rail noise can have some adverse effects that are audible.
If miniDSP has a USB input and the configuration software lets you turn off the ASRC, then easy to do an experiment.
Could be you will see some difference in HD on the analog audio outputs depending the particular resampler you use.
Like I said, just interested in if there is any detectable difference.Typically, the HD in DSP chip ASRCs is spec'ed a little below -120dB. Of course, incoming jitter and or ASRC PLL power rail noise can have some adverse effects that are audible.
No and noIf miniDSP has a USB input and the configuration software lets you turn off the ASRC, then easy to do an experiment.
As per previous posts I don't have a setup capable of measuring that and only really want to compare upsampling methods so I know which one to use.Could be you will see some difference in HD on the analog audio outputs depending the particular resampler you use.
...At least to hifi reviewers...
They won't tell you the truth about which sounds best, they can't afford to lose the advertising revenue. Not making that up either.
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No. Digital in. Eventually digital out.
Digital how? SPDIF? USB?
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