MBL Noble Line N31 CD player-DAC Measurements

Sidebar 3: Measurements

I measured the MBL N31 with my Audio Precision SYS2722 system (see the January 2008 As We See It"), using both the Audio Precision's optical and electrical digital outputs and USB data sourced from my MacBook Pro running on battery power with Pure Music 3.0 playing WAV and AIFF test-tone files. Apple's USB Prober utility identified the N31 as "MBL USB Audio Class 2" from "MBL Akustikgerate, Berlin," and its serial number as "Streamlength(tm)." The MBL's USB port operated in the optimal isochronous asynchronous mode. Apple's AudioMIDI utility revealed that, via USB, the N31 accepted 24-bit integer data. The optical input accepted datastreams with sample rates up to 96kHz, and the USB 2.0, AES/EBU, and S/PDIF inputs accepted streams of up to 192kHz.

The CD drive's error correction was good rather than great, glitches appearing in the player's output when the gaps in the data spiral on the Pierre Verany Digital Test CD reached 1.25mm in length at standard track pitch or 1mm with minimum track pitch. This was still much better than the CD standard, the so-called "Red Book," which requires only that a player cope with gaps of up to 0.2mm. The maximum output level at 1kHz was 4.15V from the balanced outputs, 2.04V from the unbalanced, and both outputs preserved absolute polarity. (The XLR jacks are wired with pin 2 hot.) The output impedance conformed to the specification at 198 ohms balanced and 93 ohms unbalanced.

The Fast filter's impulse response (fig.1) indicated that it is a conventional linear-phase finite impulse response (FIR) type, with symmetrical ringing either side of the single sample at 0dBFS. The Slow filter (fig.2) is also an FIR type but is much shorter, while the Min filter (fig.3) is also short but is a minimum-phase type, with all the ringing following the single high sample. The Fast filter rolled off rapidly above half the sampling frequency when the N31 decoded white noise sampled at 44.1kHz (fig.4, footnote 1), and reached the full stop-band suppression at 25kHz. By contrast, the Slow filter did indeed roll off slowly above the audioband (fig.5, magenta and red traces). As a result, there was very little suppression of the aliased image of a full-scale 19.1kHz tone (cyan and blue traces). The Min filter (fig.6) rolled off a little faster than the Slow, but the ultrasonic noise floor has an odd-looking, scalloped appearance.

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Fig.1 MBL N31, Fast filter, impulse response (one sample at 0dBFS, 44.1kHz sampling, 4ms time window).

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Fig.2 MBL N31, Slow filter, impulse response (one sample at 0dBFS, 44.1kHz sampling, 4ms time window).

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Fig.3 MBL N31, Min filter, impulse response (one sample at 0dBFS, 44.1kHz sampling, 4ms time window).

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Fig.4 MBL N31, Fast filter, wideband spectrum of white noise at –4dBFS (left channel red, right magenta) and 19.1kHz tone at 0dBFS (left blue, right cyan), with data sampled at 44.1kHz (20dB/vertical div.).

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Fig.5 MBL N31, Slow filter, wideband spectrum of white noise at –4dBFS (left channel red, right magenta) and 19.1kHz tone at 0dBFS (left blue, right cyan), with data sampled at 44.1kHz (20dB/vertical div.).

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Fig.6 MBL N31, Min filter, wideband spectrum of white noise at –4dBFS (left channel red, right magenta) and 19.1kHz tone at 0dBFS (left blue, right cyan), with data sampled at 44.1kHz (20dB/vertical div.).

With the Min filter and 44.1kHz data, the response was flat up to 18kHz but was then –3dB at 20kHz (fig.7, gray and green traces). With the Slow filter (fig.8), the rolloff began lower in the audioband, above 12kHz. At higher sample rates with all three filters, the rolloff continued smoothly until half of each sample rate, at which point the response dropped rapidly. Channel separation was >125dB in both directions below 1kHz, decreasing slightly to a still-superb 113dB at 20kHz. The N31's analog noise floor was both free from power-supply–related spuriae and extraordinarily low in level. When I increased the bit depth from 16 to 24 with a dithered 1kHz tone at –90dBFS (fig.9), the noise floor dropped by almost 30dB, meaning that the MBL offers 21 bits' worth of resolution—the current state of the DAC art. With undithered data representing a tone at exactly –90.31dBFS (fig.10), the three DC voltage levels described by the data were well resolved, as was the Min filter's minimum-phase ringing at the level transitions. With undithered 24-bit data, the result was a clean sinewave (fig.11).

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Fig.7 MBL N31, Min filter, frequency response at –12dBFS into 100k ohms with data sampled at: 44.1kHz (left channel green, right gray), 96kHz (left cyan, right magenta), 192kHz (left blue, right red), (1dB/vertical div.).

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Fig.8 MBL N31, Slow filter, frequency response at –12dBFS into 100k ohms with data sampled at: 44.1kHz (left channel green, right gray), 96kHz (left cyan, right magenta), 192kHz (left blue, right red), (1dB/vertical div.).

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Fig.9 MBL N31, spectrum with noise and spuriae of dithered 1kHz tone at –90dBFS with: 16-bit data (left channel cyan, right magenta), 24-bit data (left blue, right red) (20dB/vertical div.).

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Fig.10 MBL N31, waveform of undithered 1kHz sinewave at –90.31dBFS, 16-bit TosLink data (left channel blue, right red).

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Fig.11 MBL N31, waveform of undithered 1kHz sinewave at –90.31dBFS, 24-bit TosLink data (left channel blue, right red).

Not only did the N31 offer very low levels of analog noise, its distortion was also superbly low. With a full-scale 50Hz tone, the distortion harmonics all lay below –120B (0.0001%), even into the punishing 600 ohm load (fig.12). Intermodulation distortion was also vanishingly low in level (fig.13), though with the Slow and Min filters, the aliased images of the 19 and 20kHz tones with which I test IMD were not well suppressed (fig.14). The MBL's rejection of word-clock jitter, with both CD (fig.15) and 24-bit AES/EBU data (fig.16), was superb. Digital audio engineering doesn't get any better.

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Fig.12 MBL N31, spectrum of 50Hz sinewave, DC–1kHz, at 0dBFS into 600 ohms (left channel blue, right red; linear frequency scale).

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Fig.13 MBL N31, Min filter, HF intermodulation spectrum, DC–30kHz, 19+20kHz at 0dBFS into 600 ohms, 44.1kHz data (left channel blue, right red; linear frequency scale).

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Fig.14 MBL N31, Slow filter, HF intermodulation spectrum, DC–30kHz, 19+20kHz at 0dBFS into 600 ohms, 44.1kHz data (left channel blue, right red; linear frequency scale).

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Fig.15 MBL N31, high-resolution jitter spectrum of analog output signal, 11.025kHz at –6dBFS, sampled at 44.1kHz with LSB toggled at 229Hz: 16-bit CD data (left channel blue, right red). Center frequency of trace, 11.025kHz; frequency range, ±3.5kHz.

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Fig.16 MBL N31, high-resolution jitter spectrum of analog output signal, 11.025kHz at –6dBFS, sampled at 44.1kHz with LSB toggled at 229Hz: 24-bit AES/EBU data (left channel blue, right red). Center frequency of trace, 11.025kHz; frequency range, ±3.5kHz.

Finally, as Jürgen Reis emphasizes the N31's ability to cope gracefully with intersample overs, I looked at the CD track of Steely Dan's "Gaslighting Abbie" using the no-longer-available BIAS Peak program, which doesn't interpolate between samples for its waveform display. There were indeed multiple instances of consecutive samples at 0dBFS that would result in intersample overs. The red arrows in fig.17, for example, show two such instances in the non-interpolated waveform at 3:27, the one in the left channel (top) having three consecutive samples at 0dBFS, and that in the right (bottom) two consecutive samples.—John Atkinson

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Fig.17 Steely Dan's "Gaslighting Abbie," sample-level waveform display. Red arrows indicate consecutive samples at 0dBFS that would result in intersample overs.



Footnote 1: My thanks for Jürgen Reis for suggesting this test, which I have been using for five years.
COMPANY INFO
MBL Akustikgeräte GmbH & Co.
US distributor: MBL North America, Inc.
217 N. Seacrest Boulevard #276
Boynton Beach, FL 33425
(561) 735-9300
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COMMENTS
Axiom05's picture

Please correct me if I'm wrong, but isn't this whole issue a consequence of the mastering being done poorly? If the people doing the mastering didn't "push" the recorded levels so high, right up against 0 dB, then there wouldn't be any intersample overs? Isn't this another issue caused by the poor quality of the recordings that we are being offered? A higher quality of engineering would eliminate so many of the problems that we are facing in recorded music. These are not "digital" defects, these are symbolic of poor quality work. IMHO, of course...

CG's picture

Yes!

But, this is being done deliberately by somebody in the recording chain. I can only speculate that this is what they feel they need to do in order for the recording to play well over the devices of the time. So, it's not "poor work", although we might think so.

It's funny... People wonder why young folks today don't show as much interest in "hifi" as kids did back in the 70's or thereabouts. Maybe part of the reason is that the contemporary music they like doesn't reproduce well over what we might call "hifi" systems.

supamark's picture

and mastering engineers (and consumers). Labels want it to sound "good" on earbuds and crappy bluetooth speakers because that's how most pop music is consumed so it's heavily compressed/limited, and mastering engineers do it because that's how you keep the lights on and the rent paid.

The upside, if you like vinyl, is that records are mastered with far less compression of the dynamic range (you simply can't cut a record as hot as modern CD's and expect it to be playable). Those young hipsters buying vinyl are the next crop of audiophiles. Unfortunately, a good vinyl playback chain is significantly more expensive (and fiddly) than a comparable DAC/transport.

supamark's picture

that white/gold finish looks TACKY in pictures. The black/silver is a bit cold, but at least it doesn't look like it has a comb-over and tiny fingers [rimshot].

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