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Author Topic: Metrum Acoustics Quad Measurements (NOS Mini DAC)  (Read 12357 times)

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xnor

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Re oversampling (resampling) in software: then what's the point of a non-oversampling DAC to begin with.
Sigma-delta DACs have clocks operating at several MHz and oversample by a large ratio. In software you can do what, 4x oversampling with 44.1 kHz, probably even 8x to 352.8 kHz?

I don't know where the THD is coming from in your measurements, but if it's the DAC then you'll always get IMD at according levels plus whatever the zero-order hold reconstruction / DAC adds additionally. IMD is something to minimize in my book, since it's a lot more offensive than harmonic distortion.
« Last Edit: December 08, 2013, 09:04:13 PM by xnor »
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For one, software upsampling is more interesting to me because it gives me greater control and flexibility in what methods I use. It's interesting to experiment with. In theory, would it not be possible to devise software-based methods that do a better job than hardware-based methods?
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I have no (auditive) experience with NOS DAC's.

software upsampling (both in frequency and bit depth) will yield smaller 'steps' and the frequency of those steps will be higher as well moving the HF energy far above the audible range and the headphone amp / headphone will act as a reconstruction filter.
The spikes you see in the THD plots (which in music would translate in a raised noise floor WHEN PLAYING will be lower in amplitude.

However, it will ALL depend on who has made the algorithm.
Most upsampling filters will have post and ore-ringing and all advantages of the NOS concept will have gone.
The roll-off in the highs will be gone, the exemplary squarewave and needle pulse behaviour will be gone, pre-post ringing will be back etc.
Of course as Xnor explained these pulses do NOT exist in audio and 'attack' of instruments is NEVER a jump from LSB to MSB.
An NOS DAC without a proper reconstruction/brickwall filter 'invents' things that aren't in a music signal and have never been in the original music signal as well.

What you will have gained is flexibility in that you can try different algorithms/ideas from different people which may lead to measurable and perhaps even audible differences (I will leave that to golden eared people, I honestly can't tell which seems a blessing and saves me loads of money)

There will be no difference between hardware and software 'upsampling' when the same algorithm is used in the same way.
Of course it is easier to change/revise software than it is to change hardware.
Yes, you could flash hardware with other firmware but in software one could do tricks that may not be possible with existing hardware chips.

Haven't experimenting with recording files for Arta but don't think it would be impossible and really interesting to see the differences with other upsampling algorithms.

Afterall, Peter uses it on his own NOS DAC as well and that may yield measurable differences that may not be measurable with OS DAC's at all.

As I have questioned before the audibility of (pre) and post ringing at inaudible frequencies (well above 20kHz even for redbook) that incidentally are NOT related to the audio signal, baffles me.
I could go along with HF energy that has a relation to fundamentals/harmonics in certain music providing transducers can actually reproduce that but I have never seen research that 'links' improved or degraded sound with unrelated HF energy above the audible limit.
I still hope someone can show me pre and post ringing artefacts in say a 5kHz sinewave or even lower frequencies.
I once asked Archimago and he did do such a test and could find no ringing at these frequencies.

But as you say... very fun to experiment with and suggest to now and then test 'blind' when in doubt about findings.
To me that is just as enlightening as suitable analysis equipment.

Just remember most test are 'limited' in what they can show and in most cases only show one or two 'measurable' aspects.
You need a plethora of tests to draw conclusions.


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There will be no difference between hardware and software 'upsampling' when the same algorithm is used in the same way.
Of course it is easier to change/revise software than it is to change hardware.
Yes, you could flash hardware with other firmware but in software one could do tricks that may not be possible with existing hardware chips.

Haven't experimenting with recording files for Arta but don't think it would be impossible and really interesting to see the differences with other upsampling algorithms.

I still hope someone can show me pre and post ringing artefacts in say a 5kHz sinewave or even lower frequencies.
I once asked Archimago and he did do such a test and could find no ringing at these frequencies.

Trimmed some of that. That was more what I was getting at. You should have more flexibility and ease of development with software-based methods, or, at least, this is what I assumed from my background with programming (B.S. in Computer Science - but I think programming is too frustrating! haha).

I'll look into recording external samples with ARTA. I'm fairly sure it's possible and might have even noticed options for it when originally playing around.

I'm willing to test nearly anything someone asks so long as I have the means and time to do it. I might not always know how to do it immediately or what it necessarily means, but I'm willing! The more information we have, the better we can objectively assess something, even if just single elements.

Would it be at all possible to devise a measurement test that takes an actual music file for recording and measurement purposes? I mean, I know I only listen to square waves and impulses (hence the NOS DAC purchase...kidding!), but I think that would tell more about a DAC's musical performance than anything else. I could see that being a very complex process, though.
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Would it be at all possible to devise a measurement test that takes an actual music file for recording and measurement purposes? I mean, I know I only listen to square waves and impulses (hence the NOS DAC purchase...kidding!), but I think that would tell more about a DAC's musical performance than anything else. I could see that being a very complex process, though.

This would mean a nulling test with either the original digital file (but that only represents sample points in timespacing) or the original analog waveform. The original analog waveform has to pass through a brickwall at the ADC stage and thus alters slightly anyway during the encoding.

Archimago has made a 'correlation' test with a few 'representative' audio samples.
The problem with such null testing is the clocks of the recording and playback step that will be compared at a digital level (and thus only partly relevant by itself) differ as the sample points differ in time of that analog waveform (see Xnors plot about sampling frequency and input frequency not being synchronised).
This causes some errors which will always show up and ensure perfect nulling is NOT possible in practice nor theory.
Nulling 2 pure analog waveforms can reach better results IF you can actually null the 1kHz extremely exactly.

Once you know the limits of this test you can make an educated guess of how different the sound would/could be and you can actually have a LISTEN to the differences and see which volume level of the original music is actually needed before you can actually hear the differences.
Be ready for some surprises.

Everyone should decide for themselves which 'method' they prefer based on their own experience.
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I see. Even more difficulties than I initially assumed!
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This is not directly related to the Metrum, BUT I did just find out the HiFiMan HM-601 and 602 both use the TDA1543 NOS DAC. Found RightMark measurements for the 602:

http://sonove.angry.jp/Comparison_HM602_HM801_iPhone4.htm

The 602 does not measure as well as the Metrum Quad (not sure if this is due to the 16ohm load or not?), but the thing that stuck out most to me was how similar the IMD measurements looked. Interesting, and that might validate what my measurements show as well.

In measurement terms, it seems like the Metrum sits somewhere in between NOS DACs that measure fairly poorly, like the TDA1543, and the more expensive options that show better numbers, generally those running PCM1704 chips. Again, it's unfortunate the chips used in the Metrum DACs have their labels scratched off.
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xnor

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Another curiosity is the stronger odd-order harmonics, which are said to be more offensive.

Rightmark seems to use a sweep with two tones 1 kHz apart for the swept IMD test. I wonder how high the IMD is with 5 tones, 10 tones, 100 tones. It could be much worse with a more music-like test signal with more tones.
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Additional and Revised Metrum Quad Measurements - Auzen Coax Out
« Reply #38 on: December 28, 2013, 05:11:36 PM »

Given the interesting results I got with my Auzentech Bravura sound card's coax out with the NOS1704 DAC (see that thread, and it'll have more details on my desktop rig and measurement setup), I wanted to try it with the Metrum Quad. I had only tried the JKSPDIF prior. For these measurements:

Desktop rig -> Auzentech Bravura sound card's coax out -> Metrum Quad DAC -> SB1240 ADC -> tweaked laptop on battery
24/96

I'm also posting revised ARTA spectrum measurements, so it's easier to see details. This is also on top of better narrowing down on DAC and ADC volume settings in Windows to get the best, most reliable measurements. They'll look nearly identical but with slightly better numbers, generally.

The first two files are JTest jitter measurements played in JRiver Media Center using a 24/48 file. The Metrum has a noticeably lower floor than the NOS1704. The first picture is just standard playback through JRMC. The second is what happened when I enabled upsampling to 24/96.

Spectrum files are all 24/96 and in order are 1KHz sine, 100Hz sine, 60Hz + 7KHz at 4:1 ratio, and 25Hz + 66Hz at 1:1 ratio.
« Last Edit: December 28, 2013, 05:45:16 PM by hans030390 »
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Additional and Revised Metrum Quad Measurements - RMAA - Auzen Out
« Reply #39 on: December 28, 2013, 05:16:06 PM »

RMAA tests at 24/96 using the same Auzen coax out setup (keep in mind the results look better than earlier ones because I better adjusted volume settings):

Frequency response (from 40 Hz to 15 kHz), dB | +0.03, -0.84 | Good
Noise level, dB (A) | -102.6 | Excellent
Dynamic range, dB (A) | 102.9 | Excellent
THD, % | 0.037 | Good
THD + Noise, dB (A) | -66.7 | Average
IMD + Noise, % | 0.044 | Good
Stereo crosstalk, dB | -97.9 | Excellent
IMD at 10 kHz, % | 0.170 | Average
General performance | Good
« Last Edit: December 28, 2013, 05:46:07 PM by hans030390 »
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