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  1. #11
    Senior Member Wombat's Avatar
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    Somehow i can┤t believe that just comes froms integer/non- integer. Isn┤t it that you get such spikes as the DAC1 shows when you ignore or don't add enough dither?

    edit: For example the dithered/non-dithered pics of this SRC http://shibatch.sourceforge.net/ssrc/
    Last edited by Wombat; 2013-05-06 at 08:17.
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  2. #12
    Senior Member Archimago's Avatar
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    Quote Originally Posted by Wombat View Post
    Somehow i can┤t believe that just comes froms integer/non- integer. Isn┤t it that you get such spikes as the DAC1 shows when you ignore dither?

    edit: For example the dithered/non-dithered pics of this SRC http://shibatch.sourceforge.net/ssrc/
    IMO the square wave spectrum being shown by the Rockwell DAC1 tests is a reflection of the change in harmonics due to the resampling... I'll have a look at this more soon once I get the Teac UD-501 and try out the ASRC on that device to 192kHz. Obviously I suspect that will look like the DAC1.
    Last edited by Archimago; 2013-05-06 at 08:22.
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  3. #13
    Senior Member Archimago's Avatar
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    Here's we go... Non-integer upsampling of that square wave:

    No upsampling - 1.00227kHz square at 0dBFS:
    Name:  Default.jpg
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    Asynchronous upsampling to 192kHz of same square wave:
    Name:  Upsampled_192kHz.jpg
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    Looks kinda like the DAC1, bunch of extra harmonics courtesy of the ASRC algorithm, not due to volume overload... (I might have to look into this statement a little more actually since it is possible I'm overloading the measuring device... Still getting to know the machine and idiosyncracies!)

    Measurements off TEAC UD-501.
    Last edited by Archimago; 2013-05-08 at 22:59.
    Archimago's Musings: (archimago.blogspot.com) A 'more objective' audiophile blog.

  4. #14
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    Archimago,
    Nice. Can we do a comparison with lowered amplitude though (both integer and non-integer)? I am guessing a square wave will need headroom?
    Darren
    Check it, add to it! http://www.dr.loudness-war.info/

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  5. #15
    Senior Member Wombat's Avatar
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    Quote Originally Posted by Archimago View Post
    Measurements off TEAC UD-501.
    And again i wonder if this is also because of insufficient dither. Can you resample that square wave with SoX and force to disable dither to see what happens?
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  6. #16
    Senior Member Archimago's Avatar
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    Quote Originally Posted by Wombat View Post
    And again i wonder if this is also because of insufficient dither. Can you resample that square wave with SoX and force to disable dither to see what happens?
    The square wave was generated in native 16/44 with Audacity or Audition (don't remember which one now).

    So, basically turn the 16/44 1.00227kHz to say 24/192 in SOX with no dither and see how that looks when played out on the DAC?

    OK for fun, here's what I'll do tonight if I got some time - with the Teac, will put up graphs of:
    1. 16/44 square wave as generated - 0dBFS.
    2. 16/44 square wave generated with -3dB attenuation.
    3. Hardware 24/192 upsampling of said 0dBFS and -3dB attentuated signals.
    4. SOX upsampled without dither & with dither (what significance is this since we're going from 16-bit to 24-bits?).

    Should that settle the matter once and for all? :-)
    Last edited by Archimago; 2013-05-08 at 11:07.
    Archimago's Musings: (archimago.blogspot.com) A 'more objective' audiophile blog.

  7. #17
    Senior Member Wombat's Avatar
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    I don't know how you create these pics. If you do it with DAC output you should make sure no further resampling is done and for comparison it should be nice to see both results with SoX, one with dither and one without. Maybe this hints to something. Sorry if that makes you busy only because i have a wild speculation.
    edit: didn't see your edited post, was typing to long on my pad...
    Last edited by Wombat; 2013-05-08 at 11:14.

  8. #18
    Senior Member Archimago's Avatar
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    Quote Originally Posted by Wombat View Post
    I don't know how you create these pics. If you do it with DAC output you should make sure no further resampling is done and for comparison it should be nice to see both results with SoX, one with dither and one without. Maybe this hints to something. Sorry if that makes you busy only because i have a wild speculation.
    edit: didn't see your edited post, was typing to long on my pad...
    The pics are snapped from real-time FFT done from the direct RCA capture of the DAC output through the E-MU 0404USB at 24/96. No resampling or DSP engaged... Just spectrum analysis of what's coming in.

    Tonight, as I look into this, I realized just how complicated it's getting and how hard it would be to show the pictures unless this was one of my long blog posts! Bottom line:
    It really depends a lot on the resampling algorithm and obviously hardware!

    It looks like *both* integer resampling and dynamic headroom have a part to play in the square wave tests at 0dB.

    IN SOFTWARE: If you do integer resampling on 0dBFS like 44kHz --> 88 --> 176, Adobe Audition, and dBPowerAmp maintains the clean harmonics.
    SOX for some reason is not giving me the same result! Not sure if it's because Audition and dBPowerAmp are anticipating and upsampling with a small amount of volume overhead adjustment.

    IN HARDWARE: Using the TEAC UD-501, integer resampling is not a problem - for example 48kHz --> 192kHz results in a nice spectrum. Likewise the ASUS Essence One "symmetric upsampling" was not a problem. However non-integer 44kHz --> 192kHz results in the additional harmonics I showed last night with the Teac; presumably similar to the DAC1 pictures from Ken Rockwell.

    Now as for volume headroom - YES, this does result in cleaner harmonics including SOX even with non-integer upsampling from what I see using SOX, Audition, and dBPowerAmp. Therefore, Darren / Benchmark is right to allow headroom while upsampling - avoidance from clipping is never a bad thing!

    I couldn't see a difference with dithering since this was upsampling from 16 --> 24-bit which is logical when I tried SOX (both dithered and undithered looked nasty going from 44 --> 192)...

    In any case, as you can see, the situation looks complicated and it'd be tough to put together images of the combinations I tested.

    Bottom line, and I think the *only graph* that would be of benefit for Benchmark DAC1 owners is one where a -3dB 1.00227kHz square wave is passed through the DAC1 and there's demonstration of a clean 1kHz primary with the appropriate clean 3/5/7/9...kHz harmonics to show that the volume reduction helps. Based on my TEAC UD-501, it should look like this:

    -3dB Square Wave 1.00227kHz (16/44):
    Name:  -3dB_Square.jpg
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    -3dB Square Wave upsampled to 24/192 by Teac:
    Name:  -3dB_Square_Upsampled.jpg
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    Nice and clean with minimal even order harmonics... (Notice that the 192kHz upsampling in the Teac actually added a little bit of distortion.)

    Just a reminder... We're talking square waves here, and full volume as well. Great to look for little distortions here and there for the purpose of determining "accuracy" of DAC performance - likely meaningless in terms of actual audible significance :-)
    Last edited by Archimago; 2013-05-08 at 23:03.
    Archimago's Musings: (archimago.blogspot.com) A 'more objective' audiophile blog.

  9. #19
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    Well ferreted Archimago. Am I right to conclude the volume reduction lets the ASRC/upsampling work with minimal impact, then?
    Darren
    Last edited by darrenyeats; 2013-05-09 at 05:46.
    Check it, add to it! http://www.dr.loudness-war.info/

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  10. #20
    Senior Member Archimago's Avatar
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    Quote Originally Posted by darrenyeats View Post
    Well ferreted Archimago. Am I right to conclude the volume reduction lets the ASRC/upsampling work with minimal impact, then?
    Darren
    Yes Darren,
    That's what I'm able to demonstrate with the Teac's 24/192 upsampling. A -3dB attenuation will allow the upsampling to work without adding significant distortion on the square wave at 0dB.

    Peak volume extension on these square waves produced by Adobe Audition are around +2dB so the hardware will need at least that amount to prevent clipping.
    Archimago's Musings: (archimago.blogspot.com) A 'more objective' audiophile blog.

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