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Old 4th March 2011, 12:49 PM   #111
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are you sure they are actually recorded at that speed, or simply upsampled to that speed?
Yes I am sure!

As I wrote - the master recording are not only done in DXD, but DXD surround..
I have listened to the DXD surround masters in the 2L studio and compared to DXD stereo - due to that I would like to implement the ability to play DXD surround my self - then I can play all surround formats up to full DXD..

1.6875MHz are the highest I have been running the Sabre32, but at the time I tested I got some spurious noise issues and went back to the safe and working 1.536MHz.

Last edited by RayCtech; 4th March 2011 at 12:55 PM.
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Old 4th March 2011, 12:52 PM   #112
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cool, i' just never able to find that much info at hdtracks and like i said multiples of 44 make e suspicious. dxd surround is of great interest to me also eventually. but multichannel will only be used for 2 channel digital xo for me for now, thats exciting enough.

i'll have to go check out XL some more now that we upgraded the link here downloading that much is not an issue. i imagine the playback system there is pretty special

Last edited by qusp; 4th March 2011 at 12:57 PM.
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Old 4th March 2011, 03:14 PM   #113
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Hi,

Well I am basing my ramblings on some old datasheets I found regarding the Anagram upsampler. This module is used in some cambridge CD-players capable of 768Khz with AD1955 DACs as well as with some Wolfonson parts.


For example, in 8fs mode, the WM8741, seems capable of 352 or 384 Khz. (page 26)
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Old 4th March 2011, 04:16 PM   #114
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how does that relate to content? actual non upsampled native 354khz+content is whats sparse on the ground and will probably continue to be so, not dacs that can do such resolution through upsampling; theyve been around for years

Last edited by qusp; 4th March 2011 at 04:18 PM.
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Old 4th March 2011, 05:07 PM   #115
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Hi qusp,

sorry, I should have quoted rsdio reply to my question regarding making a dac such as ak4396 or ad1955 to pass 353 or 384 Khz content.

Quote:
Also, the AK4396 operates with a 128x clock with Quad Speed rates (176.4 kHz and 192 kHz). For 384 kHz, that would be 49.152 MHz, but the AK4396 has a maximum of 41.472 MHz for that clock. Again, I don't think it's possible.

Another issue is that, technically, 384 kHz would be Octuple Speed; the AK4396 doesn't have that mode. You'd have to stretch Quad Speed to the top, and the top doesn't reach 384 kHz. It does seem to reach 324 kHz, but that's shy of 352.8 kHz.
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Old 4th March 2011, 06:00 PM   #116
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no problem, apologies for misunderstanding
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Old 4th March 2011, 11:17 PM   #117
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ive seen the files at hdtracks etc and have a few, but there isnt enough information to gather whether or not its recorded or just supplied at 352khz? anything supplied at a multiple of 44.1 rather than 48 as any regular recording gear uses i'm automatically suspicious of and many of these artists simply were not around to be recorded like this and while analogue masters contain arguably more musical information in some areas of sound, the noise floor is high enough that the bandwidth will be effected. so remasters will likely be a bit hit and miss also.
Analog tape masters have a maximum bandwidth of maybe 30 kHz, so sampling at least 60 kHz will be beneficial, but sampling above 120 kHz is somewhat pointless unless you're only concerned with the anti-aliasing filter.

All 24-bit converters have noise, so the noise floor in original masters merely serves as excellent dither.

It's good to be suspicious of high sample rates, whether multiples of 44.1 kHz or 48 kHz. I have found many examples of recordings with no information other than quantization noise in the upper half of the available bandwidth, and why pay for a double-sized file when you can upconvert for yourself? What's worse, I've actually found high sample rate files with aliased frequencies above 44.1 kHz or 48 kHz, showing that they were converted with inferior technology. An aid to this sort of forensic analysis is to use linear frequency and extended dynamic range in the FFT analysis. It's much easier to detect aliasing visually when the frequency is on a linear scale.

That said, HDtracks does have some good recordings. Perhaps our tastes are different, but I went reading through the releases from artists that I already knew, and found a number of titles to purchase that were satisfying. I purchased a combination of new performances and original masters.

The one thing that bugs me is that exaU2I seems to not support 384 kHz, only 352.8 kHz. Why would the hardware go that far and not extend to 8x 48 kHz?
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Old 4th March 2011, 11:22 PM   #118
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Originally Posted by SunRa View Post
I am basing my ramblings on some old datasheets I found regarding the Anagram upsampler. This module is used in some cambridge CD-players capable of 768Khz with AD1955 DACs as well as with some Wolfonson parts.
I would be a little suspicious of an upsampler named 'Anagram.' Does that mean that it aliases frequencies from the original content in order to fill the higher frequencies? Actually, the worst would be a hypothetical product named the 'Palindrome upsampler' - which would surely alias all of the original frequencies.

Sorry, I couldn't resist the pun ... especially since I have seen many so-called high-resolution audio files with aliased frequencies in the ultrasonic range.
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Old 5th March 2011, 03:30 AM   #119
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The one thing that bugs me is that exaU2I seems to not support 384 kHz, only 352.8 kHz. Why would the hardware go that far and not extend to 8x 48 kHz?
Maybe it's limited by the bandwidth of the usb module?

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I would be a little suspicious of an upsampler named 'Anagram.'
Well yes, I see upsampling to 384 a little pointless, however some of things they did seemed to me quite clever. They are out of business, however I've found something based on the same tech, the datasheet is similar at least. Sonic Scrambler and Q5 Upsampler.

Also, on the products page I've noticed they have an USB interface based on XMOS!

Last edited by SunRa; 5th March 2011 at 03:46 AM.
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Old 6th March 2011, 01:59 PM   #120
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Anagram is just an ASRC with polynomial interpolator, unlike the ESS that only uses 2 samples to feed its ASRC, so the waveform in an ESS DA seems to be shaped by the FIR oversampling filter.
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