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Is Someone Playing Fast & Loose With Measurements?


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One friend from Google is sending me 3 more good DACs, including Herus, micro streamer and D3.

 

I did measurements on my Herus too, we can then compare... :)

 

Btw, I would always suggest to also make standard IMD measurement and spectrum analysis plot of at least 3.1 MHz band to detect leaky digital filters.

 

(I also did some measurements on hiFace DAC which uses same DAC chip as the Explorer)

 

 

P.S. Generating 32-bit J-test signal is easy, but you can of course as well use padded 16- or 24-bit ones.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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The only DAC design I know of which advertises this fact is Benchmark with their DAC2 series that have 3.5dB headroom in the digital domain, which should be enough to avoid inter-sample clipping with most lossless files.

 

Top Wolfson DAC chips have option to leave 2 dB headroom to avoid inter-sample clipping.

 

And I recommend my users to use max -3 dB volume setting when upsampling to avoid limiter kicking in at inter-sample overs. But in any case this is made visible through indicator display.

 

JRiver measures inter-sample clipping (listed as Peak Level (R128)) in its audio analysis

 

Hmmh, I don't think EBU R128 has anything to do with inter-sample overs... Inter-sample clipping also depends on the oversampling filter and factor used, so it is not easily determined.

 

However, about 40% of my library has inter-sample peaks above 0dB - so it could be argued that most DACs are going to be clipping the signal even with lossless files if you are leaving the volume at 0dB

 

Yes, inter-sample overs are guaranteed whenever there is digital clipping, and that's these days happening on lot of content!

 

You have to be very careful with DSD playback as well - many devices or players will automatically add 6dB of gain, which can clip the signal if the track is not mastered correctly.

 

That is not problem when DAC is properly designed true DSD DAC. It is mostly problem only with PCM conversion and then there should be option to not apply 6 dB gain.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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SPDIF and AES signal is based on NRZ protocol and they embedded the clock signal within the audio data. On receiver side, it needs to make sure the input signal as jitter-free as possible, so when we use PLL to recover the master clock from the data, it will have the best performance for the next stage, which is digital-to-analog stage.

 

But if you want to make any jitter measurements on it, you cannot measure it based on the recovered clock, because the result would always look perfect. Instead you need to measure it against external unrelated reference clock. So practically you would run spectrum analysis against frequency difference of the two clocks. Recovered master clock and reference - in optimal case reference being the original transmitter master clock.

 

Making analysis on digital output of S/PDIF receiver as-is, is completely useless. Unless something is really badly wrong, it always looks perfect.

 

I have been asked many times for the similar question in USB interface too. People ask since they use USB Hard drive or USB printer that never got a bit of error. Why bother with Asynchronous USB interface or buffer... this kind of stuff. The answer is the same: the real time transmission nature of digital music, and the SPDIF or USB Audio protocol design itself could not guarantee the bit perfect, 100% error free transmission. That is why people are investing their time and money here. :-)

 

I have moved over to use ethernet/WLAN as transfer media, guaranteed to be 100% error free.

 

Error rate is not really an issue, I don't remember having transfer errors even with S/PDIF... It is very likely that any transmission error would result in clearly audible error.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Local hifi magazine "Hifimaailma" had group test of USB DACs with headphone amp and Dragonfly v1.2 was included.

 

Some of the Dragonfly v1.2 measurement results:

SNR: 95 dB

Noise: -120 dBV

THD 0 dBFS, 500 mV, 1 kHz 33/330 ohm: 0.18/0.17 %

Linearity: -92 dBFS

IMD: 19/20 kHz, 2nd, 33/330 ohm: 0.017/0.016%

 

Headphone output of MacBook Pro was included for comparison:

SNR: 97 dB

Noise: -120 dBV

THD: 0.0043/0.0013 %

Linearity: -93 dBFS

IMD: 0.0009/0.0008%

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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First time I see lower IMD than THD unless the whole thing was oscillating.

 

I've seen it quite a lot. For example my Resonessence Labs HERUS when playing DSD128 has 2nd order IMD at around -100 dB (3rd @-87.5 dB) while 3rd THD component (highest) is at -85 dB. With PCM 2nd IMD is around -90 dB (3rd -85 dB) while 3rd harmonic is at -82 dB. Curious thing is also that with PCM THD spectrum is dominated by odd harmonics while with DSD it becomes more even harmonics.

 

2nd order IMD was lower than THD also in number of other devices. They listed 2nd and 3rd order IMD separately, so for total IMD you need to sum the two. For Dragonfly, 3rd order IMD was 0.017/0.014%.

 

Some devices had significantly higher 3rd order IMD than 2nd order: ADL X1, CA DacMagic XS, Meridian Explorer (0.30/0.23%!), Meridian Prime and Pro-Ject Headbox DS. MacBook Pro has challenge to lower 33 ohm impedance since 3rd IMD increases to 0.01% compared to 0.0008% to 330 ohm.

 

Do you have more details how the measurements where conducted?

 

Apart from what I already quoted, measurement is done using Miller Audio Research analyzer (IOW, same as used by HiFi-News magazine).

 

SNR is A-weighted while the background noise dBV figure must be unweighted.

 

Background noise is probably measured while playing silence. Hopefully dithered one, because otherwise most DACs would just mute output. (from past experience I know MacBook Pro mutes headphone output one second after playback has stopped)

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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You would have to look at Jitter test signal (J-Test - cough, cough) to see if your device actually allowed low jitter.

 

That is only useful for digital input if you discard the recovered clock and use some other reference clock. It would of course still look perfect if you check it against the recovered clock.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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THD or THD+N?

 

The first, IMD can never be greater than THD (mathematical rules), THD & N implies not just distortion but any noise, so THD & N can be greater than any IMD intercept, but only in the presence of noise greater than the actual distortion.

 

Story doesn't tell...

 

Jumping to some measurements I did...

Here's how THD looks like on Resonessence Labs HERUS @DSD128, 0 dBFS:

Herus-1k-0dB_DSD128.png

 

Dropping volume to -10 dBFS lowers the distortion quite drastically, as is case for many DACs:

Herus-1k-10dB_DSD128.png

 

And here's how IMD looks like @DSD128, 0 dBFS:

Herus-IMD-0dB_DSD128.png

 

 

How can unweighed noise be lower than the A-weighted SNR? This truly seems an enigma to rival the Sphinx.

 

At least SNR contains all the distortion components while my guess is that background noise is the idle noise level.

 

You mean you are not sure how these results where arrived at, yet you still choose to share them "because they use Miller Beer Analyser"?

 

You cannot be cerious.

 

Damn, I am trying to answer your questions, I didn't make the measurements, I don't have any affiliation to the magazine in question. I'm just a subscriber and quoted figures from the table of results and referenced the source.

 

I don't take sides on validity of measurements of anybody else's except my own. Just one more set of figures to already existing bunch.

 

I don't have Dragonfly, so I cannot measure it myself and I'm not interested to buy it because it doesn't fulfill my feature requirements (DXD & DSD128).

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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* Output load is not 100k Ohm or 200K, which is usually not the real headphone load situation. I set 300Ohm in Audio Precision which is more real. By this alone, THD+N will be different.

 

This makes sense for a device with headphone output, it is quite close to typical hifi headphone impedance.

 

The numbers I quoted were with 33 and 330 ohm load, but the difference is that Dragonfly was version 1.2 that supposedly had the clipping issue fixed and other improvements too.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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