NEWS: Hugo M Scaler announced

Discussion in 'Hi-Fi Stereo Systems & Separates' started by hodg100, Jul 22, 2018.


    1. hodg100

      hodg100
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    2. muljao

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      Interesting. I wonder would there be a market for this. Personally it reminds me of upscaling an image in photoshop, the detail isn't really there, but it tricks you into thinking it is. I reckon it's a bit of a rip off
       
    3. Welwynnick

      Welwynnick
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      I think the idea is that you push the sampling frequency high that the DAC reconstruction filter has an easy job, and you're not left with any in-band amplitude or phase non-linearities.

      It's not snake oil.

      Nick
       
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      Last edited: Jul 22, 2018
    4. xxGBHxx

      xxGBHxx
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      Care to expand or point me at an article that explains it? Genuinely curious.

      Thanks

      G
       
    5. B17NNS

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      Snake oil.
       
    6. waspy

      waspy
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      Can someone explain this? How can you upscale a 16bit 44.1khz signal? to add in more detail? - The SPDIF signal going into it will be what's on the disc, what is it "filling" in.. I am very suspicious about it and I am a chord electronics fan!
       
    7. Evinger

      Evinger
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      Yep I'm suspicious as well. You can't add back in what isn't there. Or maybe it contains an infinite number of Digital Musical Monkeys that can then recreate the original ;)
       
    8. spinaltap

      spinaltap
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      For £1,500, the TEAC NT-505 can also offer upsampling - plus MQA decoding. Thus, unlike the Chord, it can properly decode/upsample a USB fed signal up to full-fat MQA. That’s, typically 24/352. However, the TEAC NT-505 can upsample to 32/768.

      TEAC | NT-505
       
    9. waspy

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    10. dannnielll

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      It's good to be suspicious.. and you cannot add back something which is not there.. good points.
      However if you have two samples taken in adjacent time intervals, and they have different values, which is highly likely, you can interpolate, and guess what the sample inbetween should be, and insert that.
      Is it cheating? Yes and no. It is reasonable to assume that had the intermediate sample been taken , it would have been close to the inserted one.
      We are very happy for our Full HD TVs to do that with DVD data or 4K TVs with BluRay.

      Now xxGBHxx, the maths of digital filtering is rather heavy, so rather than, get outside my comfort zone, could the following explanation work for you.?
      .. the same 16 bit CD quality data is fed into a high speed DAC and converted into the analogue voltage perhaps 32 times with the same data. Now every time a conversation takes place there is a little momentary glitch. By doing the conversion 32 times faster, and having a low pass filter on the output it is much easier to filter out these gliches.
       
    11. Evinger

      Evinger
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      Yep totally agree with the analogy with Blu Ray upscaling, & see that a quality unit upscaling will allow the DAC to give a better approximation of the Analogue.
      In the end, like many things, as you get closer & closer to a good equivalent of the actual original, the cost increases exponentially for small improvements, & I guess it is up to the individual as to how much they are willing to spend for these ever more & more minute improvements
       
    12. waspy

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      Well this is it.. I think I'll need to a home loan of this to see how much better it is than a standard transport - DAC combo.
       
    13. Welwynnick

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      The caption should read CanJam rather than CamJam 2018.

      It was 33 minutes long, but it was worth every minute for anyone who's really interested in digital audio. It's not really about the 768k upscaling so much as the million tap filtering, which has almost historical significance.

      Paraphrasing very heavily, the story seems to be that Chord's digital designer Rob Watts realised 35 years ago that the theoretical performance of CD audio could be realised if you had a good enough reconstruction filter. The filter should be a Sync filter, but it would need a million taps to realise the full 44.1/16 performance. Hardware limitations have prevented that until now. Chord use a large Xilinx Artix-7 FPGA with 740 DSP cores to do the digital filtering. The output of that filtering is fed to a suitable (very good) DAC for conversion. Or the filtering can be performed in an integrated transport, the Blu MkII, launched last year.

      Rob was quite dismissive about what had been achieved by digital audio to date (including his own designs) saying that it was some way short of what should theoretically be achieved. In particular, current reconstruction filters were never able to get transient timing right, and this wasn't good enough. Now however, the M scaler can filter sufficiently well to realise the theoretical 44.1/16 envelope, and this is first time that this has been done. He says the improvements are considerable, and improve everything.

      This does sound like the real thing.

      Nick
       
    14. waspy

      waspy
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      Thanks Nick, you succinctly put together what was said. I guess proof will be in the pudding come autumn!
       
    15. dannnielll

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      If as it likely appears, this is the real deal, it is very good news in the long run for everyone. In the short term Watts and his team stand to make a good profit, as wealthy patrons buy the newest and best. In the medium term the technology will be licenced out and in the longer term,it will be either public domain or pirated.
      3000 quid is a substantial sum of money but not exorbitant, .. I have seen plasma TVs at 40k quid dropping to 20k ,then 10k and then almost free fall as LCD TVs came on stream. .. and they were being purchased at the 10k and 5k levels in quantity. Many musical instruments are in that ballpark.. pianos, branded guitars,as indeed are turntables and cartridges
       
      Last edited: Jul 23, 2018
    16. B17NNS

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      :rotfl: Russ Andrews couldn't have said it better.
       
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    17. larkone

      larkone
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      What is guaranteed is those that spend £3000 on this will tell you it is wonderful, whilst the rest of us argue about its efficacy.
       
    18. dannnielll

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      .. yes but There will be one difference between that and the afore mentioned RA products. .. it will be possible to measure on a scope whether the transients are in or out of phase... What
      measurement will not tell us however, is whether it matters
       
    19. tony @ suffolk

      tony @ suffolk
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      This scaler is the same one that's inside the Blu2 CD player, which I've heard being used with a Chord Dave DAC. Adding the scaler in gave a surprising uplift in sound quality. I currently use a Dave, so I'm very keen to try this device in my own system.
       
    20. waspy

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      I got a response from the man him self to the same question I posed here on head-fi :)

      "Interesting post - I am actually not adding in any more information - the information is all contained in the original bandwidth limited sampled signal. but what we have is sampled data, and what we need is a continuous un-sampled signal - and we need to do a huge amount of processing to do this without error. I will give you an example. Imagine a sine wave. You can state it's a sine wave; give its frequency and its amplitude. So the information content is fixed; but if you want a waveform of infinite length, and precision, then you would need an infinite amount of processing to create the infinite number of points. And with the sampled data, we can convert it to a continuous signal - with exactly the same information content - and recover the original bandwidth limited continuous signal - if and only if you do an infinite amount of processing and use an ideal sinc function interpolation filter. So I am not trying to create new information - actually we are converting from a sampled bandwidth limited signal to a continuous with exactly the same information. The problem with conventional filters is they are the ones that are adding extra information, as the interpolated signal is different from the original. What I am trying to do is merely reduce these errors, which are audible as it degrades the timing of transients - something which is essential from human psychoacoustics."

      Chord Electronics Hugo M-Scaler
       
    21. pressure

      pressure
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      Ok, there’s something here that’s potentially very good news. I’d have to hear it to say for sure, but the theory looks sound.
       
    22. waspy

      waspy
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      Same, the difference in the Mojo to the Qutest was huge so hoping the MScaler opens things up massively. Time to stock up on CDs :p
       
    23. AndrewH13

      AndrewH13
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      Picture I took from Canjam unveiling along with Hugo TT2

      322FF0FF-3D49-4C67-83E6-0AD72B304759.jpeg
       
    24. fyonn

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      While I’ve no idea if it makes any difference or not, I’d think this ought to be part of the input stage to one of their own dac’s rather than a stand-alone bit of kit..?
       
      Last edited: Jul 28, 2018
    25. Welwynnick

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      …… or part of the output stage of one of their transports.

      I thought that, too. Apparently the reason for not integrating it with the DAVE DAC is that the power consumption is too high.

      Nick
       
    26. fyonn

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      Power consumption too high? Unlike their 1000w reference power amp for example...?
       
    27. Trollslayer

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      I can see the principle, it remains to be seen how effective (and expensive) it is.
      Basically if the FPGA can detect what the original waveform before digitisation might be then it can interpolate sub samples to make reproduction more faithful to the original.
       
    28. MoonHamster

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      Yes I think the same and why have they made it look so cheap it looks like it cost
      £30/$40
       
    29. BigAl217

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      Wow, I’ve always felt that Chord equipment looks and feels amongst the highest quality and finish equipment in the hi-fi world. Guess I must have cheap taste but don’t tell the wife.
       
    30. Khazul

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      The thing you need to get right with a DAC output is the filter (whether it be analog or DSP). If you get that right, then oversampling makes no difference at all.

      The reason for oversampling is to make it easier to implement better a better band limiting DSP filter with respect to it effects in the audio range.

      I find it rather odd that someone feels there is merit in creating a box that just does the over-sampling bit when an ideal filter will precisely throw away all of its efforts.

      OTOH, if you happen to have an audiophile pet bat or cat, then by all means knock yourself out.
       

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