Details

Parent Category: Products

Category: Digital-to-Analog Converter Measurements

Created: 15 January 2023

Link: reviewed by Matt Bonaccio on SoundStage! Hi-Fi on January 15, 2023

General Information

All measurements taken using an Audio Precision APx555 B Series analyzer.

The RME ADI-2 DAC FS was conditioned for 30 minutes at 0dBFS (3.6Vrms out) into 200k ohms before any measurements were taken.

The ADI-2 DAC FS offers three digital inputs: one coaxial S/PDIF (RCA), one optical SPDIF (TosLink), and one USB. There are two sets of line-level outputs (balanced XLR and unbalanced RCA) and two headphone outputs (1/4″ TRS labelled “PHONES” and 1/8″ TRS labelled “IEM”). There is a digital volume control for the headphone and line-level outputs. Comparisons were made between unbalanced and balanced line-level outputs, and aside from the 6dB extra voltage over balanced, no differences were seen in terms of THD and noise.

The ADI-2 DAC FS offers a dizzying array of features and settings. The following are the default settings used for the coaxial input, balanced line-level outputs, PHONES and IEM headphone outputs, using a 0dBFS input, unless otherwise specified:

Line-level output:

• Ref level: +7dBu (3.6Vrms out over XLR)
• Auto ref level: off
• Mono: off
• Width: 1.0 (full stereo)
• M/S proc: off
• Polarity: off
• Crossfeed: off
• DA filter: SD Sharp
• De-emphasis: Auto
• Dual EQ: off
• Volume: 0dB
• Lock volume: off
• Balance: center
• Loopback to USB: off

PHONES output:

• Hi-power: on
• Auto ref level: off
• Mono: off
• Width: 1.0 (full stereo)
• M/S proc: off
• Polarity: off
• Crossfeed: off
• DA filter: SD Sharp
• De-emphasis: Auto
• Dual EQ: off
• Volume: set to output 2Vrms
• Lock volume: off
• Balance: center
• Loopback to USB: off

IEM output:

• Hi-power: off
• Auto ref level: off
• Mono: off
• Width: 1.0 (full stereo)
• M/S proc: off
• Polarity: off
• Crossfeed: off
• DA filter: SD Sharp
• De-emphasis: Auto
• Dual EQ: off
• Volume: set to output 0.5Vrms
• Lock volume: off
• Balance: center
• Loopback to USB: off

There are six digital filter settings, labelled: Short Delay (SD) Sharp, SD Slow, Sharp, Slow, Non-oversampling (NOS), and Brickwall. Here are RME’s descriptions for each:

• SD Sharp: the most linear frequency response and lowest latency
• SD Slow: causes a small drop in the higher frequency range
• Sharp and Slow: same as SD Sharp and SD Slow respectively but with higher latency but linear phase over the entire audioband
• NOS: smallest steepness and therefore affects treble more than the other filters, but offers the best impulse response
• Brickwall: sharp filtering and phase-linear

The ADI-2 DAC FS volume control ranges from -93.8dB to +6dB, in steps of 6dB to 0.5dB (most of the range is 0.5dB steps). Channel-to-channel deviation proved exceptional, at around 0.01dB throughout the range.

Volume-control accuracy (measured at line-level outputs): left-right channel tracking

Published specifications vs. our primary measurements

The table below summarizes the measurements published by RME for the ADI-2 DAC FS compared directly against our own. The published specifications are sourced from RME’s website, either directly or from the supplied manual, or a combination thereof. With the exception of frequency response, where the Audio Precision bandwidth was set at its maximum (DC to 1MHz), assume, unless otherwise stated, the coaxial digital input (24/96 1kHz sine wave at 0dBFS), the balanced line-level or unbalanced headphone outputs into 200k ohms (line-level) and 300 ohms (headphone) using a measurement input bandwidth of 10Hz to 90kHz, and the worst-case measured result between the left and right channels.

*due to very low noise of DUT, analyzer self-noise has been removed from measurement to more accurately report value

Our primary measurements revealed the following using the coaxial input and the balanced line-level outputs (unless specified, assume a 1kHz sine wave at 0dBFS, 200k ohms loading, 10Hz to 90kHz bandwidth):

Our primary measurements revealed the following using the coaxial input and the headphone output (unless specified, assume a 1kHz sine wave at 0dBFS, 300-ohm loading, 10Hz to 90kHz bandwidth, and 2Vrms output for the PHONES output, and 0.56Vrms for the IEM output): 

*due to very low noise of DUT, analyzer self-noise has been removed from measurement to more accurately report value

Frequency response (16/44.1, 24/96, 24/192 with SD Sharp filter)

The plot above shows the ADI-2 DAC FS’s frequency response as a function of sample rate. The blue/red traces are for a 16-bit/44.1kHz dithered digital input signal from 5Hz to 22kHz, the purple/green traces are for a 24/96 dithered digital input signal from 5Hz to 48kHz, and, finally, orange/pink represents 24/192 from 5Hz to 96kHz. The behavior at low frequencies is the same for the all input resolutions—perfectly flat down to 5Hz. The behavior at high frequencies for all three digital sample rates is as expected, offering sharp filtering around 22k, 48k, and 96kHz (half the respective sample rate), although the 24/192 data shows softer attenuation around the corner frequency. The -3dB point for each sample rate is roughly 21.2, 46.6, and 92.6kHz, respectively. In the graph above and most of the graphs below, only a single trace may be visible. This is because the left channel (blue, purple or orange trace) is performing identically to the right channel (red, green or pink trace), and so they perfectly overlap, indicating that the two channels are ideally matched.

Frequency response (bass and treble)

Above are frequency-response plots measured at the balanced outputs with the bass and treble controls set to maximum (blue/red plots) and minimum (purple/green plots). We see that roughly +/- 6dB of gain/cut is available for each.

Frequency response (16/44.1 with SD Sharp, SD Slow, and Sharp fiters)

The plots above show frequency responses for a 0dBFS input signal sampled at 44.1kHz for the SD Sharp (blue), SD Slow filter (red), and Sharp (green) filters into a 200k ohm load for the left channel only. The graph is zoomed in from 1kHz to 22kHz to highlight the various responses of the three filters. We can see that the SD Sharp and Sharp filters offer essentially the same frequency response, with a -3dB point at 21.2 kHz. The SD Slow filter offers a much shallower attenuation, with a -3dB point at 17.4kHz.

Frequency response (16/44.1 with Slow, NOS, and Brickwall fiters)

The plots above show frequency responses for a 0dBFS input signal sampled at 44.1kHz for the Slow (blue), NOS (red), and Brickwall (green) filters into a 200k ohm load for the left channel only. The graph is zoomed in from 1kHz to 22kHz to highlight the various responses of the three filters. We can see that the Slow filter is similar to the SD Slow filter above, but with a -3dB point at 20kHz. The NOS filter, predictably, exhibits significant high-frequency roll-off with a -0.8dB response at 10kHz, and -3.4dB at 20kHz. The Brickwall filter exhibits the most, well, brickwall-type behavior, although with a lower -3dB point (19.8kHz) than the filters shown above.

Phase response vs. sample rate (16/44.1, 24/96, 24/192 with SD Sharp filter)

Above are the phase-response plots from 20Hz to 20kHz for the coaxial input, measured at the balanced output, using the SD Sharp filter setting. The blue/red traces are for a dithered 16/44.1 input at 0dBFS, the purple/green for 24/96, and the orange/pink for 24/192. We can see that the ADI-2 DAC FS does not invert polarity, with a worst-case phase shift of 92 degrees at 13kHz for the 16/44.1, 60 degrees at 20kHz for the 24/96 input data, and just 20 degrees of phase shift at 20kHz for the 24/192 input data.

Phase response vs. filter type (16/44.1 with SD Sharp, SD Slow, and Sharp filters)

Above are the phase-response plots from 20Hz to 20kHz for a 0dBFS input signal sampled at 44.1kHz for the SD Sharp (blue), SD Slow (red), and Sharp (green) filters into a 200k ohm load for the left channel only. The SD Slow and Sharp filters yielded significantly less phase shift than the default SD Sharp filter, with -10 and +40 degrees respectively of phase shift at 20kHz.

Phase response vs. filter type (16/44.1 with Slow, NOS, and Brickwall filters)

Above are the phase-response plots from 20Hz to 20kHz for a 0dBFS input signal sampled at 44.1kHz for the Slow (blue), NOS (red), and Brickwall (green) filters into a 200k ohm load for the left channel only. Predictably, the Brickwall filter yields the highest phase shift at over +180 degrees at 20kHz, the Slow filter is at +120 degrees at 20kHz, while the NOS filter is +80 degrees at 20kHz.

Digital linearity (16/44.1 and 24/96 data)

The graph above shows the results of a linearity test for the coaxial digital input (the optical input performed identically) for both 16/44.1 (blue/red) and 24/96 (purple/green) input data, measured at the balanced line-level output of the ADI-2 DAC FS. The digital input is swept with a dithered 1kHz input signal from -120dBFS to 0dBFS, and the output is analyzed by the APx555. The ideal response is a straight flat line at 0dB (i.e., the amplitude of the digital input perfectly matches the amplitude of the measured analog output). The 24/96 input data is essentially pefect down to -120dBFS, while the 16/44.1 input data performed well, only over-responding by 2.5dB at -120dBFS. The 24/96 data yielded such superb results that we extended the sweep down to . . .

Digital linearity (16/44.1 and 24/96 data)

. . . -140dBFS. Above we see that even at -140dBFS, the ADI-2 DAC FS is only undershooting by -1 to -2 dB. This is an exemplary linearity test result.

Impulse response (SD Sharp, SD Slow, Sharp filters)

The graph above shows the impulse responses for a looped 24/44.1 test file that moves from digital silence to full 0dBFS (all “1”s) for one sample period, then back to digital silence, fed to the coaxial digital input, measured at the balanced outputs, for the SD sharp (blue), SD Slow (red), and Sharp (green) filters into a 200k ohm load for the left channel only. We can see that the SD Sharp filter has no pre-ringing, but significant post-ringing, the SD Slow filter has only very minor post-ringing, and the Sharp filter exhibits symmetrical pre- and post-ringing, as seen in a typical sinc function.

Impulse response (Slow, NOS, Brickwall filters)

The graph above shows the impulse responses for a looped 24/44.1 test file that moves from digital silence to full 0dBFS (all “1”s) for one sample period, then back to digital silence, fed to the coaxial digital input, measured at the balanced outputs, for the Slow (blue), NOS (red), and Brickwall (green) filters into a 200k ohm load for the left channel only. We can see that the Slow filter has very minor pre- and post-ringing, the NOS filter is as advertised and shows a single pulse with essentially no pre- or post-ringing, and the Brickwall filter exhibits symmetrical pre/post ringing, as seen in a typical sinc function.

Impulse response (NOS filter)

We decided to investigate the impulse response of the NOS filter in more detail. The graph above shows the impulse responses for a looped 24/44.1 test file that moves from digital silence to full 0dBFS (all “1”s), for one sample period then back to digital silence, fed to the coaxial digital input, measured at the balanced outputs, for the NOS (red) filter only into a 200k ohm load for the left channel only. The graph is zoomed in to show that the NOS filter is as advertised, and yields a single pulse with essentially no pre/post ringing.

J-Test (coaxial input)

The plot above shows the results of the J-Test test for the coaxial digital input measured at the balanced line-level output of the ADI-2 DAC FS. J-Test was developed by Julian Dunn the 1990s. It is a test signal—specifically,  a -3dBFS undithered 12kHz squarewave sampled (in this case) at 48kHz (24 bits). Since even the first odd harmonic (i.e., 36kHz) of the 12kHz squarewave is removed by the bandwidth limitation of the sampling rate, we are left with a 12kHz sine wave (the main peak). In addition, an undithered 250Hz squarewave at -144dBFS is mixed with the signal. This test file causes the 22 least-significant bits to constantly toggle, which produces strong jitter spectral components at the 250Hz rate and its odd harmonics. The test file shows how susceptible the DAC and delivery interface are to jitter, which would manifest as peaks above the noise floor at 500Hz intervals (e.g., 250Hz, 750Hz, 1250Hz, etc). Note that the alternating peaks are in the test file itself, but at levels of -144dBrA (fundamental at 250Hz) down to -170dBrA for the odd harmonics.  The test file can also be used in conjunction with artificially injected sine-wave jitter by the Audio Precision, to also show how well the DAC rejects jitter.

The coaxial input shows an extremely clean J-Test result, with only minor peaks at the -150dBrA level. This is an indication that the ADI-2 DAC FS should not be sensitive to jitter through this input.

J-Test (optical input)

The plot above shows the results of the J-Test test for the coaxial digital input measured at the balanced line-level output of the ADI-2 DAC FS. The optical input shows an extremely clean J-Test result, with only minor peaks at the -150dBrA level. This is an indication that the ADI-2 DAC FS should not be sensitive to jitter through this input.

J-Test (coaxial input, 2kHz sine-wave jitter at 500ns)

The plot above shows the results of the J-Test test for the coaxial digital input (the optical input behaved similarly), measured at the balanced line-level output, with an additional 500ns of 2kHz sine-wave jitter injected by the APx555. The result remains clean with no visible sidebands at 10kHz and 14kHz (12kHz main signal +/-2kHz jitter signal). This is further evidence of the ADI-2 DAC FS’s superb jitter immunity.

J-Test (coaxial input, 2kHz sine-wave jitter at 900ns)

The plot above shows the results of the J-Test test for the coaxial digital input (the optical input behaved similarly), measured at the balanced line-level output, with an additional 900ns of 2kHz sine-wave jitter injected by the APx555. Here sidebands are visible at 10kHz and 14kHz (12kHz main signal +/- 2kHz jitter signal), but remain relatively low at -110dBrA. With jitter above this level, the ADI-2 DAC FS lost sync with the signal.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (SD Sharp filter)

The plot above shows a fast Fourier transform (FFT) of the ADI-2 DAC FS’s balanced line-level output with white noise at -4dBFS (blue/red) and a 19.1kHz sine wave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using the SD Sharp filter setting. The sharp roll-off above 20kHz in the white-noise spectrum shows the implementation of the brick-wall-type reconstruction filter. There are absolutely no imaged aliasing artifacts in the audioband above the -135dBrA noise floor. The primary aliasing signal at 25kHz is at -105dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (SD Slow filter)

The plot above shows a fast Fourier transform (FFT) of the ADI-2 DAC FS’s balanced line-level output with white noise at -4dBFS (blue/red) and a 19.1kHz sine wave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using the SD Slow filter setting. The slow roll-off above 20kHz in the white-noise spectrum shows the implementation of a reconstruction filter with a slow roll-off. Despite this, there are absolutely no imaged aliasing artifacts in the audioband above the -135dBrA noise floor. The primary aliasing signal at 25kHz is near -10dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (Sharp filter)

The plot above shows a fast Fourier transform (FFT) of the ADI-2 DAC FS’s balanced line-level output with white noise at -4dBFS (blue/red) and a 19.1kHz sine wave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using the Sharp filter setting. The sharp roll-off above 20kHz in the white noise spectrum shows the implementation of the brick-wall-type reconstruction filter. There are absolutely no imaged aliasing artifacts in the audioband above the -135dBrA noise floor. The primary aliasing signal at 25kHz is at -105dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (Slow filter)

The plot above shows a fast Fourier transform (FFT) of the ADI-2 DAC FS’s balanced line-level output with white noise at -4dBFS (blue/red) and a 19.1kHz sine wave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using the Slow filter setting. The slow roll-off above 20kHz in the white-noise spectrum shows the implementation of a reconstruction filter with slow roll-off. Despite this, there are absolutely no imaged aliasing artifacts in the audioband above the -135dBrA noise floor. The primary aliasing signal at 25kHz is near -10dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (NOS filter)

The plot above shows a fast Fourier transform (FFT) of the ADI-2 DAC FS’s balanced line-level output with white noise at -4dBFS (blue/red) and a 19.1kHz sine wave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using the NOS filter setting. Due to the lack of a filter, the noise spectrum is mostly constant (i.e., un-attenuated), except at multiples of the 44.1kHz sample rate. Despite this, the imaged aliasing artifacts in the audioband at 13.2kHz is only at -120dBrA. The primary aliasing signal at 25kHz is near -5dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (Brickwall filter)

The plot above shows a fast Fourier transform (FFT) of the ADI-2 DAC FS’s balanced line-level output with white noise at -4dBFS (blue/red) and a 19.1 kHz sine wave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using the Brickwall filter setting. The very sharp roll-off above 20kHz in the white noise spectrum shows the implementation of the brick-wall-type reconstruction filter. There are absolutely no imaged aliasing artifacts in the audioband above the -135dBrA noise floor. The primary aliasing signal at 25kHz is at roughly -100dBrA.

THD ratio (unweighted) vs. frequency vs. load (24/96)

The chart above shows THD ratios at the balanced line-level output into 200k ohms (blue/red) and 600 ohms (purple/green) as a function of frequency for a 24/96 dithered 1kHz signal at the coaxial input. The 200k- and 600-ohm data are nearly identical throughout the audioband, which is in indication that ADI-2 DAC FS’s outputs are robust and can handle loads below 1k ohms with no difficultly. The right channel does outperform the left by about 5dB from 20Hz to 1kHz; however, at these THD levels (0.00005% to 0.0001%), the differences are of absolutely no consequence. It should also be noted that these THD ratios are pushing up against the limits of the AP analyzer, which exhibits just under 0.00002% THD at 3.6Vrms in loopback mode. Above 3kHz, there is a rise in THD, up to 0.0004% at 20kHz.

THD ratio (unweighted) vs. frequency vs. sample rate (16/44.1 and 24/96)

The chart above shows THD ratios at the balanced line-level output into 200k ohms as a function of frequency for a 16/44.1 (blue/red) and a 24/96 (purple/green) dithered 1kHz signal at the coaxial input. The 24/96 data consistently outperformed the 16/44.1 data by about 10dB from 20Hz to about 2kHz due to the inherently higher noise floor at 16 bits (i.e., the analyzer cannot assign a THD value below the noise floor). THD ratios with 16/44.1 data range from 0.0002% at 20Hz, down to 0.0001% at 5kHz, back up to 0.0004% at 20kHz. THD ratios with 24/96 data range from 0.00005% at 20Hz, up to 0.0004% at 20kHz.

THD ratio (unweighted) vs. output (16/44.1 and 24/96)

The chart above shows THD ratios measured at the balanced output as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green). Once again, the 24/96 outperformed the 16/44.1 data, with a THD range from 0.2% to 0.00005%, while the 16/44.1 ranged from 5% down to nearly 0.0002%.

THD+N ratio (unweighted) vs. output (16/44.1 and 24/96)

The chart above shows THD+N ratios measured at the balanced output as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green). The 24/96 outperformed the 16/44.1 data, with a THD+N range from 3% down to  0.0003%, while the 16/44.1 ranged from 40% down to 0.002% at the maximum output voltage of 3.6Vrms.

THD ratio (unweighted) vs. output (24/96) at maximum gain

The chart above shows THD ratios measured at the balanced output as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS at 24/96 (blue/red), this time with the ADI-2 DAC FS gain set to maximum (i.e., Ref Level set +19dBu, volume set to +6dB). The THD ratios ranged from 0.1% at 0.5mVrms, down to 0.00005% at the “knee” at 7Vrms, with the 1% THD mark hit at roughly 10Vrms at the output.

THD+N ratio (unweighted) vs. output (24/96) at maximum gain

Similar to the chart above, this chart shows THD+N ratios (the addition of noise to THD) measured at the balanced output as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS at 24/96 (blue/red), with the ADI-2 DAC FS gain set to maximum (i.e., Ref Level set +19dBu, volume set to +6dB). The THD+N ratios ranged from 2% at 0.5mVrms, down to 0.0003% at the “knee” at 7Vrms, with the 1% THD+N mark hit at roughly 10Vrms at the output.

FFT spectrum – 1kHz (digital input, 16/44.1 data at 0dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 16/44.1. We see no sign of signal harmonics above the -135dBrA noise floor. There are also no power-supply noise peaks to speak of to the left of the main signal peak.

FFT spectrum – 1kHz (digital input, 24/96 data at 0dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 24/96. Due to the increased bit-depth, the noise floor is much lower compared to the 16/44.1 FFT, at a very low -160dBrA. We see very low signal harmonics ranging from -130dBrA, or 0.00003%, at 7kHz, down to below -150dBrA, or 0.000003%. Here also, there are no power-supply noise peaks to speak of to the left of the main signal peak.

FFT spectrum – 1kHz (digital input, 16/44.1 data at -90dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 16/44.1 at -90dBFS. We see the signal peak at the correct amplitude, and no signal harmonics above the noise floor within the audioband.

FFT spectrum – 1kHz (digital input, 24/96 data at -90dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 24/961 at -90dBFS. We see the signal peak at the correct amplitude, and no signal harmonics above the noise floor within the audioband.

Intermodulation distortion vs generator level (SMPTE, 60Hz:4kHz, 4:1, 16/44.1, 24/96)

The chart above shows intermodulation distortion (IMD) ratios measured at balanced output for 16/44.1 (blue/red) and 24/96 input data (purple/green), from -60dBFS to 0dBFS. Here, the SMPTE IMD method is used, where the primary frequency (F1 = 60Hz), and the secondary frequency (F2 = 7kHz), are mixed at a ratio of 4:1. The SMPTE IMD analysis results consider the second (F2 ± F1) through the fifth (F2 ± 4xF1) modulation products. The 16/44.1 data yields IMD ratios from 2% down to 0.002% at 0dBFS. The 24/96 data yields IMD ratios from 0.1% down to 0.0003% at 0dBFS.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, 16/44.1)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sine-wave stimulus tone measured at the balanced output into 200k ohms for the coaxial input at 16/44.1. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 3.6Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz), as well as the third-order modulation products, at 17kHz and 20kHz, are not visible above the -135dBrA noise floor.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, 24/96)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sine-wave stimulus tone measured at the balanced output into 200k ohms for the coaxial input at 24/96. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 3.6Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -135dBrA, or 0.00002%, for the left channel (right channel peak cannot be seen above -150dBrA noise floor), while the third-order modulation products, at 17kHz and 20kHz, are slightly higher, at around -120dBrA to -130dBrA, or 0.0001% to 0.00003%. This is an exceptionally clean IMD result.

Diego Estan
Electronics Measurement Specialist

Details

Parent Category: Products

Category: Digital-to-Analog Converter Measurements

Created: 01 July 2022

Link: reviewed by Gordon Brockhouse on SoundStage! Simplifi on July 1, 2022

General Information

All measurements taken using an Audio Precision APx555 B Series analyzer.

The Neo S was conditioned for 30 minutes at 0dBFS (4.5Vrms out) into 200k ohms before any measurements were taken.

The Neo S offers four digital inputs: one coaxial S/PDIF (RCA), one optical S/PDIF, one AES/EBU (XLR), and one USB. There are two line-level outputs (balanced XLR and unbalanced RCA) and two headphone outputs (1/8″ TRS unbalanced and 3.4mm TRRS balanced). There are also LAN (over ethernet) and Bluetooth inputs, as well as HDMI and coaxial digital outputs. There is also a digital volume control. Comparisons were made between unbalanced and balanced line-level outputs for a 24/96 0dBFS input, and aside from the 6dB extra voltage over balanced, there were no appreciable differences in THD+N. In terms of input types (USB, coaxial, optical), there were no differences in terms of THD+N at 24/96 resolution.

The Neo S offers three different digital filter settings, accessible through the touchscreen user menu. These are: Fast Rolloff, Slow Rolloff Minimum Phase, and Minimum Phase Corrected.

The Neo S volume control can provide adjustments in 0.5, 1, 2, or 3dB steps. The step value size can be selected in the user menu. The range is -60dB to 0dB. At -60dB, the output is effectively muted; at -59.5dB, the output over the balanced connectors measured 4.8mVrms; and at 0dB, the output from the balanced connectors measured 4.5Vmrs.  Using the headphone outputs does not offer any further gain/output voltage. The volume control is implemented in the digital domain, as every step was exactly 0.5dB, and the channel-to-channel deviation was exactly 0.035-0.036dB at every step, throughout the range, as seen in the table below.

Unless otherwise stated, all measurements are with the coaxial digital input, balanced outputs, the Fast Roll-Off filter, and the volume set to 0dB.

Volume-control accuracy (measured at line-level outputs): left-right channel tracking

Published specifications vs. our primary measurements

The table below summarizes the measurements published by Zidoo for the Neo S compared directly against our own. The published specifications are sourced from Zidoo’s website, either directly or from the manual available for download or included in the box, or a combination thereof. With the exception of frequency response, where the Audio Precision bandwidth is set at its maximum (DC to 1MHz), assume, unless otherwise stated, the coaxial digital input (24/96 1kHz sine wave at 0dBFS), the line-level or headphone outputs into 200k ohms (line-level) and 300/32 ohms (headphone high/low gain), using a measurement input bandwidth of 20Hz to 20kHz, and the worst-case measured result between the left and right channels.

Our primary measurements revealed the following using the coaxial input and the balanced line-level outputs (unless specified, assume a 1kHz sine wave at 0dBFS, 200k ohms loading, 10Hz to 90kHz bandwidth):

Our primary measurements revealed the following using the coaxial input and the balanced headphone output (unless specified, assume a 1kHz sine wave at 0dBFS input, 4.5Vrms into 300 ohms, 10Hz to 90kHz bandwidth):

High gain setting

Low gain setting

Frequency response (16/44.1, 24/96, 24/192)

The plot above shows the Neo S frequency response as a function of sample rate. The blue/red traces are for a 16-bit/44.1kHz dithered digital input signal from 5Hz to 22kHz, the purple/green traces are for a 24/96 dithered digital input signal from 5Hz to 48kHz, and finally orange/pink represents 24/192 from 5Hz to 96kHz. The behavior at low frequencies is the same for all sample rates—perfectly flat down to 5Hz. The behavior at high frequencies for all three digital sample rates is as expected, offering sharp filtering around 22, 48, and 96kHz (half the respective sample rate). The -3dB point for each sample rate is roughly 21, 46.2, and 92.2kHz, respectively. It is also obvious from the plots above that all three sample rates offered “brick-wall”- type behavior with the default Fast Rolloff filter. In the graph above and most of the graphs below, only a single trace may be visible. This is because the left channel (blue, purple or orange trace) is performing identically to the right channel (red, green or pink trace), and so they perfectly overlap, indicating that the two channels are ideally matched.

Frequency response (16/44.1, all three filters)

The plots above show frequency-response for a 16/44.1 input for all three filters (Fast Rolloff, Slow Rolloff Minimum Phase, and Minimum Phase Corrected) into a 200k ohm load for the left channel only. Fast Rolloff is in blue, Slow Rolloff Minimum Phase in purple, and Minimum Phase Corrected is in red. The graph is zoomed in from 1kHz to 22kHz, to highlight the various responses of the filters around the “knee” of the response. At 20kHz, the Fast Rolloff filter is at -0.03dB, the Slow Rolloff Minimum Phase filter is at -5dB, and the Minimum Phase Corrected filter is at -12.3dB.

Phase response (16/44.1, 24/96, 24/192 with Fast Rolloff filter)

Above are the phase response plots from 20Hz to 20kHz for the coaxial input, measured at the balanced output, using the Fast Rolloff filter. The blue/red traces are for a dithered 16/44.1 input at -20dBFS, the purple/green for 24/96, and the orange/pink for 24/192. We can see that the NEO S does not invert polarity, with a worst-case phase shift of just under 160 degrees at 20kHz for the 16/44.1 data, and within +/-20 degrees or so for the 24/96 and 24/192 input data.

Phase response (16/44.1, all three filters)

Above are the phase response plots from 20Hz to 20kHz for a 16/44.1 signal at the coaxial input, measured at the balanced output, for all three filters (Fast Rolloff, Slow Rolloff Minimum Phase, and Minimum Phase Corrected) into a 200k ohm load for the left channel only. Fast Rolloff is in blue, Slow Rolloff Minimum Phase in purple and Minimum Phase Corrected in red. We see that both the Slow Rolloff Minimum Phase and Minimum Phase Corrected filters exhibit far less phase shift between 5kHz and 20kHz than the Fast Rolloff filter.

Digital linearity (16/44.1 and 24/96 to -120dB)

The graph above shows the results of a linearity test for the coaxial digital input (the optical input performed identically) for both 16/44.1 (blue/red) and 24/96 (purple/green) input data, measured at the balanced line-level output. The digital input is swept with a dithered 1kHz input signal from -120dBFS to 0dBFS, and the output is analyzed by the APx555. The ideal response is a straight flat line at 0dB (i.e., the amplitude of the digital input perfectly matches the amplitude of the measured analog output). The 24/96 data is perfectly linear down to -120dBFS, while the 16/44.1 data was perfect down to -100dBFS, and only +3dB (left) and +1dB (right) at -120dBFS. Because of this DAC’s exceptional linearity performance, a second measurement was performed extending down to -140dBFS, plotted in the chart below.

Digital linearity (16/44.1 and 24/96 to -140dB)

This shows the 24/96 data remained within 0.5dB or so of flat to -140dB, which is a phenomenal result. The 16/44.1 data results are as expected, showing significant deviations below -120dBFS. But it is worth highlighting that a linearity result that is flat down to -100dBFS for 16-bit input data is also exceptional.

Impulse response (24/44.1, all three filters)

The graph above shows the impulse responses for a looped 24/44.1 test file that moves from digital silence to full 0dBFS (all “1”s) for one sample period then back to digital silence for all three filters (Fast Rolloff, Slow Rolloff Minimum Phase, and Minimum Phase Corrected) into a 200k ohm load for the left channel only. Fast Rolloff is in blue, Slow Rolloff Minimum Phase in purple, and Minimum Phase Corrected in red. The default Fast Rolloff filter exhibits a typical sinc function, with symmetrical pre- and post-ringing behavior. The Slow Rolloff Minimum Phase filter exhibits no pre-ringing and very little post-ringing, where the Minimum Phase Corrected filter is somewhere in between the other two.

J-Test (coaxial input)

The plot above shows the results of the J-Test test for the coaxial digital input measured at the balanced line-level output of the Neo S. J-Test was developed by Julian Dunn the 1990s. It is a test signal—specifically, a -3dBFS undithered 12kHz square wave sampled (in this case) at 48kHz (24 bits). Since even the first odd harmonic (i.e., 36kHz) of the 12kHz square wave is removed by the bandwidth limitation of the sampling rate, we are left with a 12kHz sine wave (the main peak). In addition, an undithered 250Hz square wave at -144dBFS is mixed with the signal. This test file causes the 22 least-significant bits to constantly toggle, which produces strong jitter spectral components at the 250Hz rate and its odd harmonics. The test file shows how susceptible the DAC and delivery interface are to jitter, which would manifest as peaks above the noise floor at 500Hz intervals (e.g., 250Hz, 750Hz, 1250Hz, etc). Note that the alternating peaks are in the test file itself, but at levels of -144dBrA (fundamental at 250Hz) down to -170dBrA for the odd harmonics.  The test file can also be used in conjunction with artificially injected sinewave jitter by the Audio Precision, to also show how well the DAC rejects jitter.

The coaxial S/PDIF input shows some of the alternating 500Hz peaks in the audio band but at low levels; below -130dBrA. This is an indication that the Neo S should not be sensitive to jitter.

J-Test (optical input)

The optical S/PDIF input shows essentially the same result as with the coaxial input above.

J-Test with 10ns of injected jitter (coaxial input)

Both the coaxial and optical inputs were also tested for jitter immunity by injecting artificial sine wave jitter at 2kHz on top of the J-Test test file, which would manifest as sidebands at 10kHz and 14kHz without any jitter rejection. Above is an FFT with jitter injected at the 10ns level, and peaks can be seen at -125dBrA. This demonstrates that the Neo S DAC’s jitter rejection is not as robust as the J-test result alone would have indicated. Only the coaxial input is shown because the optical input showed basically the same result.

J-Test with 100ns of injected jitter (coaxial input)

Both the coaxial and optical inputs were also tested for jitter immunity by injecting artificial sine wave jitter at 2kHz on top of the J-Test test file, which would manifest as sidebands at 10kHz and 14kHz without any jitter rejection. Above is an FFT with jitter injected at the 100ns level, and peaks can be seen at -105dBrA. This demonstrates that the Neo S DAC’s jitter rejection is not as robust as the J-Test result alone would have indicated. Again, the optical input showed pretty much the same result.

Wideband FFT spectrum of white noise and 19.1kHz sine-wave tone (Fast Rolloff filter)

The plot above shows a fast Fourier transform (FFT) of the Neo S balanced-line level output with white noise at -4dBFS (blue/red) and a 19.1kHz sine wave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using the Fast Rolloff filter. The sharp roll-off above 20kHz in the white-noise spectrum shows the implementation of the brick-wall type reconstruction filter. There are absolutely no imaged aliasing artifacts in the audio band above the -135dBrA noise floor. The primary aliasing signal at 25kHz is at -85dBrA, with subsequent harmonics of the 25kHz peak at or below this level.

Wideband FFT spectrum of white noise and 19.1kHz sine-wave tone (Slow Rolloff Minimum Phase filter)

The plot above shows a fast Fourier transform (FFT) of the Neo S balanced-line level output with white noise at -4dBFS (blue/red) and a 19.1kHz sine wave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using the Slow Rolloff Minimum Phase filter. The roll-off above 20kHz in the white-noise spectrum is shallower than what is seen with the filters above and below. There are absolutely no imaged aliasing artifacts in the audio band above the -135dBrA noise floor. The primary aliasing signal at 25kHz is at -35dBrA, with subsequent harmonics of the 25kHz peak at or below this level.

Wideband FFT spectrum of white noise and 19.1kHz sine-wave tone (Minimum Phase Corrected filter)

The plot above shows a fast Fourier transform (FFT) of the Neo S balanced-line level output with white noise at -4dBFS (blue/red) and a 19.1 kHz sine wave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using the Minimum Phase Corrected filter. The FFT is very similar to the one for the Fast Rolloff filter.

THD ratio (unweighted) vs. frequency vs. load (24/96)

The chart above shows THD ratios at the balanced line-level output into 200k ohms (blue/red) and 600 ohms (purple/green) as a function of frequency for a 24/96 dithered 1kHz signal at the coaxial input. The 200k and 600 ohms data are nearly identical; however, the right channel, at 0.0002% and below, did outperform the left channel, which was at 0.0003%. In either case, these are extremely low levels of THD. These data also demonstrate that the Neo S’s line-level outputs are robust and can handle lower impedance loads.

THD ratio (unweighted) vs. frequency vs. sample rate (16/44.1 and 24/96)

The chart above shows THD ratios at the balanced line-level output into 200k ohms as a function of frequency for a 16/44.1 (blue/red) and a 24/96 (purple/green) dithered 1kHz signal at the coaxial input. The 24/96 data (right channel) consistently outperformed the 16/44.1 data by about 5dB. Still, all THD values are very low, between 0.0005% and 0.00015%.

THD ratio (unweighted) vs. output (16/44.1 and 24/96) at 1kHz

The chart above shows THD ratios measured at the balanced output as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS with 16/44.1 (blue/red) and 24/96 (purple/green) dithered input data. Once again, the 24/96 outperformed the 16/44.1 data, with a THD range from 0.2% to nearly 0.0001% (right channel) at 4.5Vrms, while the 16/44.1 ranged from 5% down to 0.0004% at 4.5Vrms. It’s important to highlight that large discrepancies in THD ratios at lower input levels are due to the inherently lower 24-bit noise floor of the 24/96 data (i.e., when no signal harmonics are measurable in an FFT, a THD value can only be assigned as low as the noise floor permits). The 24/96 data also shows a slight rise in THD between around 100mVrms and 1Vrms.

THD+N ratio (unweighted) vs. output (16/44.1 and 24/96) at 1kHz

The chart above shows THD+N ratios measured at the balanced output as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS with 16/44.1 (blue/red) and 24/96 (purple/green) dithered input data. The 24/96 outperformed the 16/44.1 data, with a THD+N range from 3% down to  0.0005%, while the 16/44.1 ranged from 50% down to 0.002% at the maximum output voltage of 4.5Vrms.

Intermodulation distortion vs generator level (SMPTE, 60Hz:4kHz, 4:1, 16/44.1, 24/96)

The chart above shows intermodulation distortion (IMD) ratios measured at the balanced output as a function of generator input level for the coaxial input into 200k ohms from -60dBFS to 0dBFS with 16/44.1 (blue/red) and 24/96 (purple/green) dithered input data. Here, the SMPTE IMD method was used, where the primary frequency (F1 = 60Hz) and the secondary frequency (F2 = 7kHz) are mixed at a ratio of 4:1. The SMPTE IMD analysis results consider the second (F2 ± F1) through the fifth (F2 ± 4xF1) modulation products. The 24/96 outperformed the 16/44.1 data, with an IMD range from 0.06% down to  0.0005% between -10 and -5dBFS, then up to about 0.001% at 0dBFS, while the 16/44.1 ranged from 2% down to 0.002% at the maximum output voltage of 4.5Vrms at 0dBFS. The 24/96 data exhibited a slight rise in IMD between -30dBrA and -15dBrA.

FFT spectrum – 1kHz (digital input, 16/44.1 data at 0dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 16/44.1. We see the third signal harmonic (3kHz) at -110/-120dBrA (left/right), or 0.0003/0.0001%. The second and fifth signal harmonics for the left channel are visible at -125dBrA, or 0.00006%, just above the noise floor. There are also no power-supply noise peaks to speak of to the left of the main signal peak.

FFT spectrum – 1kHz (digital input, 24/96 data at 0dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the balanced output into 200k ohms for the coaxial digital input, sampled at 24/96. Due to the increased bit-depth, the noise floor is much lower compared to the 16/44.1 FFT. We see signal harmonics ranging from -110/-120dBrA (left/right), or 0.0003/0.0001% at 3kHz, down to -140dBrA, or 0.00001%. With the lower noise floor, we can see higher even order harmonics, for example at 4 and 6kHz where the peaks are just below -140dBrA, or 0.00001%. Here we see low level peaks on the left side of the main signal peak, at -130dBrA, or 0.00003%, and below.

FFT spectrum – 1kHz (digital input, 16/44.1 data at -90dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the balanced output into 200k ohms for the coaxial digital input, sampled at 16/44.1 at -90dBFS. We see the main peak at the correct amplitude, and no signal harmonics above the noise floor within the audio band.

FFT spectrum – 1kHz (digital input, 24/96 data at -90dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the balanced output into 200k ohms for the coaxial digital input, sampled at 24/96 at -90dBFS. We see the main peak at the correct amplitude, and power-supply related harmonics at 60Hz, 180Hz, 300Hz, etc., at -130dBrA, or 0.00003%, and below.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, 16/44.1)

Shown above is an FFT of the intermodulation (IMD) products for an 18kHz + 19kHz summed sine-wave stimulus tone measured at the balanced output into 200k ohms for the coaxial input at 16/44.1. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 4.5Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) cannot be seen above the -130dBRA, or 0.00003%, noise floor, while the third-order modulation products, at 17kHz and 20kHz, are just above (left) and below (right) -120dBrA, or 0.0001%.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, 24/96)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sine-wave stimulus tone measured at the balanced output into 200k ohms for the coaxial input at 24/96. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 4.5Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at a vanishingly low -140dBRA, or 0.00001%, while the third-order modulation products, at 17kHz and 20kHz, are just above (left) and below (right) -120dBrA, or 0.0001%.

Diego Estan
Electronics Measurement Specialist

Details

Parent Category: Products

Category: Digital-to-Analog Converter Measurements

Created: 01 June 2022

Link: reviewed by Dennis Burger on SoundStage! Access on June 1, 2022

General Information

All measurements taken using an Audio Precision APx555 B Series analyzer.

The iFi Audio Zen One Signature was conditioned for 30 min at 0dBFS (2.1Vrms out) into 100k ohms before any measurements were taken.

The Zen One Signature offers four digital inputs: one coaxial S/PDIF (RCA), one optical S/PDIF, one USB, and Bluetooth. There are two line-level outputs (balanced 3.4mm TRRS and unbalanced RCA) and one digital output (coaxial, over the same RCA connector used for the coaxial digital input). Comparisons were made between unbalanced and balanced line-level outputs for a 24/96 0dBFS input, and aside from the 6dB extra voltage over balanced, there were virtually no differences in THD+N and dynamic range. In terms of input types (USB, coaxial, optical), there were no differences in terms of THD+N.

Unless otherwise stated, all measurements are with the coaxial digital input and unbalanced outputs.

Published specifications vs. our primary measurements

The table below summarizes the measurements published by iFi Audio for the Zen One Signature compared directly against our own. The published specifications are sourced from iFi’s website, either directly or from the manual available for download or included in the box, or a combination thereof. With the exception of frequency response, where the Audio Precision bandwidth is set at its maximum (DC to 1MHz), assume, unless otherwise stated, the coaxial digital input (24/96 1kHz sine wave at 0dBFS), the unbalanced line-level output into 100k ohms (line-level), using a measurement input bandwidth of 10Hz to 90kHz, and the worst-case measured result between the left and right channels.

Our primary measurements revealed the following using the coaxial input and the unbalanced line-level outputs (unless specified, assume a 1kHz sine wave at 0dBFS, 100k ohms loading, 10Hz to 90kHz bandwidth):

Frequency response (16/44.1, 24/96, 24/192)

The plot above shows the Zen One Signature frequency response as a function of sample rate. The blue/red traces are for a 16-bit/44.1kHz dithered digital input signal from 5Hz to 22kHz, the purple/green traces are for a 24/96 dithered digital input signal from 5Hz to 48kHz, and, finally, orange/pink represents 24/192 from 5Hz to 96kHz. The behavior at low frequencies is the same for the three sample rates—essentially perfectly flat down to 5Hz. The behavior at high frequencies for all three digital sample rates is as expected, offering filtering around 22k, 48k, and 96kHz (half the respective sample rate). The -3dB point for each sample rate is roughly 21.1, 46.2, and 83kHz, respectively. It is also obvious from the plots above that the 44.1kHz sampled input signal offers the most “brick-wall”-type behavior, while the attenuation of the 96kHz and 192kHz sampled input signals approaching the corner frequencies (48kHz and 96kHz) is more gentle. In the graph above and most of the graphs below, only a single trace may be visible. This is because the left channel (blue, purple or orange trace) is performing identically to the right channel (red, green or pink trace), and so they perfectly overlap, indicating that the two channels are ideally matched.

Digital linearity (16/44.1 and 24/96 data)

The graph above shows the results of a linearity test for the coaxial digital input (the optical input performed identically) for both 16/44.1 (blue/red) and 24/96 (purple/green) input data, measured at the unbalanced line-level output. The digital input is swept with a dithered 1kHz input signal from -120dBFS to 0dBFS, and the output is analyzed by the APx555. The ideal response is a straight flat line at 0dB (i.e., the amplitude of the digital input perfectly matches the amplitude of the measured analog output). The 24/96 data is only -1dB at -120dBFS, while the 16/44.1 data was perfect down to -100dBFS, and only +2.5dB (left) at -120dBFS. This is an excellent result.

Impulse response

The graph above shows the impulse response for a looped 24/44.1 test file that moves from digital silence to full 0dBFS (all “1”s) for one sample period then back to digital silence. We see a typical sinc function response.

J-Test (coaxial input)

The plot above shows the results of the J-Test test for the coaxial digital input measured at the unbalanced line-level output of the Zen One Signature. The J-Test was developed by Julian Dunn the 1990s. It is a test signal, specifically a -3dBFS undithered 12kHz square wave sampled (in this case) at 48kHz (24 bits). Since even the first odd harmonic (i.e., 36kHz) of the 12kHz square wave is removed by the bandwidth limitation of the sampling rate, we are left with a 12kHz sinewave (the main peak). In addition, an undithered 250Hz square wave at -144dBFS is mixed with the signal. This test file causes the 22 least significant bits to constantly toggle, which produces strong jitter spectral components at the 250Hz rate and its odd harmonics. The test file shows how susceptible the DAC and delivery interface are to jitter, which would manifest as peaks above the noise floor at 500Hz intervals (e.g., 250Hz, 750Hz, 1250Hz, etc). Note that the alternating peaks are in the test file itself, but at levels of -144dBrA (fundamental at 250Hz) down to -170dBrA for the odd harmonics.  The test file can also be used in conjunction with artificially injected sine-wave jitter by the Audio Precision, to also show how well the DAC rejects jitter.

The coaxial S/PDIF input shows worst case peaks at -130dBrA. This is an indication that the Zen One Signature should not be sensitive to jitter.

J-Test (optical input)

The optical S/PDIF input shows essentially the same result as with the coaxial input. Both the coaxial and optical inputs were also tested for jitter immunity by injecting artificial sine-wave jitter at 2kHz on top of the J-Test file, which would manifest as sidebands at 10kHz and 14kHz without any jitter rejection. Jitter immunity proved exceptional, with no visible sidebands at even 1000ns of jitter level.

Wideband FFT spectrum of white noise and 19.1kHz sine-wave tone

The plot above shows a fast Fourier transform (FFT) of the Zen One Signature unbalanced line-level output with white noise at -4 dBFS (blue/red), and a 19.1 kHz sine wave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green). The sharp roll-off above 20kHz in the white-noise spectrum shows the implementation of the brick-wall-type reconstruction filter. There are a few imaged aliasing artifacts in the audio band, the most predominant at 17kHz at -110dBrA. The primary aliasing signal at 25kHz is at -80dBrA, with subsequent harmonics of the 25kHz peak slightly above this level.

THD ratio (unweighted) vs. frequency vs. load (24/96)

The chart above shows THD ratios at the unbalanced line-level output into 100k ohms (blue/red) and 600 ohms (purple/green) as a function of frequency for a 24/96 dithered 1kHz signal at the coaxial input. There’s a left/right THD imbalance into 100k ohms from 30Hz to 5kHz or so, with the left channel (blue) outperforming the right channel (red) by as much as almost 10dB. Into 600 ohms, the difference in THD between left and right was much smaller, about 3dB. In general, higher THD ratios were observed into 600 ohms, ranging from roughly 0.005% at low frequencies to 0.04% at 20kHz. This compared to the 100k ohm data, which ranged from as low as 0.001% (left channel from 100-300 Hz), to 0.04% at 20kHz.

THD ratio (unweighted) vs. frequency vs. sample rate (16/44.1 and 24/96)

The chart above shows THD ratios at the unbalanced line-level output into 100k ohms as a function of frequency for a 16/44.1 (blue/red) and a 24/96 (purple/green) dithered 1kHz signal at the coaxial input. All data tracked closely, with THD ratios ranging from 0.002% at 20Hz, down to 0.001% from 100-300Hz, then up to 0.04% at 20kHz. The exception is the right channel at 24/96, which yielded THD values almost 10dB higher from 100Hz to 500Hz or so.

THD ratio (unweighted) vs. output (16/44.1 and 24/96) at 1kHz

The chart above shows THD ratios measured at the unbalanced output as a function of output voltage for the coaxial input into 100k ohms from -90dBFS to 0dBFS with 16/44.1 (blue/red) and 24/96 (purple/green) dithered input data. The 24/96 outperformed the 16/44.1 data at lower output levels, with a THD range from 1.5% to nearly 0.0005% at 0.2-0.3Vrms, while the 16/44.1 data ranged from 4% down to 0.001% at 1Vrms. It’s important to highlight that large discrepancies in THD ratios at lower input levels are due to the inherently lower 24-bit noise floor of the 24/96 data (i.e., when no signal harmonics are measurable in an FFT, a THD value can only be assigned as low as the noise floor permits). At output voltages above 0.4Vrms or so, again we see the right channel with higher THD values than the left.

THD+N ratio (unweighted) vs. output (16/44.1 and 24/96) at 1kHz

The chart above shows THD+N ratios measured at the unbalanced output as a function of output voltage for the coaxial input into 100k ohms from -90dBFS to 0dBFS with 16/44.1 (blue/red) and 24/96 (purple/green) dithered input data. The 24/96 outperformed the 16/44.1 data, with a THD+N range from 30% down to  0.003%, while the 16/44.1 ranged from 40% down to 0.004% at the maximum output voltage of 2.1Vrms.

Intermodulation distortion vs generator level (SMPTE, 60Hz:4kHz, 4:1, 16/44.1, 24/96)

The chart above shows intermodulation distortion (IMD) ratios measured at the unbalanced output as a function of generator input level for the coaxial input into 100k ohms from -60dBFS to 0dBFS with 16/44.1 (blue/red) and 24/96 (purple/green) dithered input data. Here, the SMPTE IMD method is used, where the primary frequency (F1 = 60Hz), and the secondary frequency (F2 = 7kHz), are mixed at a ratio of 4:1. The SMPTE IMD analysis results consider the second (F2 ± F1) through the fifth (F2 ± 4xF1) modulation products. The 24/96 outperformed the 16/44.1 data, with an IMD range from 0.5% down to  0.003% between -15 and 0dBFS, although, again, the right channel performed worse (almost 10dB) at these higher generator levels. The 16/44.1 data ranged from 2% down to roughly 0.005% at the maximum output voltage of 2.1Vrms at 0dBFS, with the left channel slightly outperforming the right above -10dBFS.

FFT spectrum – 1kHz (digital input, 16/44.1 data at 0dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the unbalanced output into 100k ohms for the coaxial digital input, sampled at 16/44.1. We see the second signal harmonic (2kHz) at -95dBrA, or 0.002%, and subsequent harmonics (3, 4, 5, 6, 7kHz, etc.) at descending lower levels from -100dBrA, or 0.001%, down to below -120dBrA, or 0.0001%. There are no power-supply noise peaks to speak of to the left of the main signal peak.

FFT spectrum – 1kHz (digital input, 24/96 data at 0dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the unbalanced output into 100k ohms for the coaxial digital input, sampled at 24/96. We consistently see the right channel signal harmonic peaks, 5-10dB higher than the left peaks. The left channel signal harmonic peaks essentially match what was measured at 16/44.1 (shown above). There are no power-supply noise peaks to speak of to the left of the main signal peak.

FFT spectrum – 1kHz (digital input, 16/44.1 data at -90dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the unbalanced output into 100k ohm for the coaxial digital input, sampled at 16/44.1 at -90dBFS. We see the main peak at the correct amplitude, and no signal harmonics above the noise floor within the audio band.

FFT spectrum – 1kHz (digital input, 24/96 data at -90dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the unbalanced output into 100k ohm for the coaxial digital input, sampled at 24/96 at -90dBFS. We see the main peak at the correct amplitude, and a hint of signal harmonic peak within the audio band at 3kHz at a very low -140dBrA, or 0.00001%.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, 16/44.1)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sine-wave stimulus tone measured at the unbalanced output into 100k ohms for the coaxial input at 16/44.1. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 2.1Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is just below -90dBRA, or 0.0003%, while the third-order modulation products, at 17kHz and 20kHz, are slightly higher, reaching -85dBrA, or 0.0006%.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, 24/96)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sine-wave stimulus tone measured at the unbalanced output into 100k ohms for the coaxial input at 24/96. The input dBFS values are set at -6.02dBFS, so that, if summed for a mean frequency of 18.5kHz, would yield 2.1Vrms (0dBrA) at the output. Here again, we find higher distortion peaks for the right channel compared to the left, by about 5dB. Otherwise, the FFT looks essentially the same at the 16/44.1 IMD FFT above.

Diego Estan
Electronics Measurement Specialist

Details

Parent Category: Products

Category: Digital-to-Analog Converter Measurements

Created: 01 December 2021

Link: reviewed by James Hale on SoundStage! Xperience on December 1, 2021

General Information

All measurements taken using an Audio Precision APx555 B Series analyzer.

The EarMen Tradutto was conditioned for 30 minutes at 0dBFS (4.3Vrms out) into 200k ohms before any measurements were taken.

The Tradutto offers four digital-input options: coaxial S/PDIF (RCA), optical S/PDIF (TosLink), USB, and Bluetooth. There are two line-level outputs: balanced 4.4mm TRRS and unbalanced RCA. Comparisons were made between unbalanced and balanced line-level outputs, and aside from the 6dB extra voltage over the balanced connection, there were no appreciable differences observed in terms of THD and noise.

All measurements below, unless otherwise stated, are for the coaxial digital input, and the balanced output. Comparisons were made in terms of THD+N between the coaxial and optical input, and none were found. Between the USB and coaxial input, there was an improvement in THD+N (0dBFS, 24/96), where the USB input measured 0.003%, the coaxial input measured 0.0005%.

Published specifications vs. our primary measurements

The table below summarizes the measurements published by EarMen for the Tradutto compared directly against our own. The published specifications are sourced from Earmen’s website, either directly or from the manual available for download, or a combination thereof. With the exception of frequency response, where the Audio Precision bandwidth is set at its maximum (DC to 1MHz), assume, unless otherwise stated, the coaxial digital input (24/96 1kHz sinewave at 0dBFS), the balanced line-level outputs into 200k ohms, a measurement input bandwidth of 10Hz to 90kHz, and the worst-case measured result between the left and right channels.

Our primary measurements revealed the following using the coaxial input and the balanced line-level output (unless specified, assume a 1kHz sinewave at 0dBFS, 200k ohms loading, 10Hz to 90kHz bandwidth):

Frequency response (16/44.1, 24/96, 24/192)

The chart above shows the Tradutto frequency response as a function of sample rate. The blue/red traces are for a 16-bit/44.1kHz dithered digital input signal from 5Hz to 22kHz, the purple/green traces are for a 24/96 dithered digital input signal from 5Hz to 48kHz, and finally orange/pink represents 24/192 from 5Hz to 96kHz. The behavior at low frequencies is the same for the digital input—perfectly flat down to 5Hz. The behavior at high frequencies for all three digital sample rates is as expected, offering sharp filtering around 22k, 46k, and 96kHz (half the respective sample rates). The -3dB point for each sample rate is roughly 21.5, 45 and 92kHz respectively. All three sample rates offer “brick-wall”-type behavior. In the graph above and most of the graphs below, only a single trace may be visible. This is because the left channel (blue, purple or orange trace) is performing identically to the right channel (red, green or pink trace), and so they perfectly overlap, indicating that the two channels are ideally matched.

Phase response vs. sample rate (16/44.1, 24/96, 24/192)

Above are the phase response plots from 20Hz to 20kHz for the coaxial input, measured at the balanced output. The blue/red traces are for a dithered 16/44.1 input at -20dBFS, the purple/green for 24/96, and the orange/pink for 24/192. The Tradutto does not invert polarity. There’s a worst-case phase shift of around 75 degrees at 20kHz for the 16/44.1 signal, 25 degrees for the 24/96 signal, and roughly 10 degrees for the 24/192 input data.

Digital linearity (16/44.1 and 24/96 data)

The chart above shows the results of a linearity test for the coaxial digital input (the optical input performed identically) for both 16/44.1 (blue/red) and 24/96 (purple/green) input data, measured at the balanced line-level output. The digital input is swept with a dithered 1kHz input signal from -120dBFS to 0dBFS, and the output is analyzed by the APx555. The ideal response is a straight flat line at 0dB (i.e., the amplitude of the digital input perfectly matches the amplitude of the measured analog output). The 24/96 exhibited perfect linearity down to -120dBFS, while the 16/44.1 data was perfect down to -100dBFS, and only +3dB (left) and +1dB (right) at -120dBFS. Because of this DAC’s exceptional linearity performance, a second measurement . . .

. . . was performed, extending down to -140dBFS. The chart above shows linearity performance down to -140dBFS, where the 24/96 remained within 1dB of flat, which is a phenomenal result. The 16/44.1 data results are as expected, showing significant deviations below -120dBFS. But it is worth highlighting that a linearity result that is flat down to -100dBFS for 16-bit input data is also exceptional.

Impulse response

The graph above shows the impulse response for the Tradutto, for a looped 24/44.1 test file that moves from digital silence to full 0dBFS (all “1”s) for one sample period then back to digital silence. We can see that reconstruction filter employed by EarMen is that of a typical sinc function.

J-Test (coaxial input)

The chart above shows the results of the J-Test test for the coaxial digital input (the optical input performed identically) measured at the balanced line-level output. J-Test was developed by Julian Dunn the 1990s. It is a test signal—specifically, a -3dBFS undithered 12kHz square wave sampled (in this case) at 48kHz (24 bits). Since even the first odd harmonic (i.e., 36kHz) of the 12kHz square wave is removed by the bandwidth limitation of the sampling rate, we are left with a 12kHz sinewave (the main peak). In addition, an undithered 250Hz square wave at -144dBFS is mixed with the signal. This test file causes the 22 least significant bits to constantly toggle, which produces strong jitter spectral components at the 250Hz rate and its odd harmonics. The test file shows how susceptible the DAC and delivery interface are to jitter, which would manifest as peaks above the noise floor at 500Hz intervals (e.g., 250Hz, 750Hz, 1250Hz, etc.). Note that the alternating peaks are in the test file itself, but at levels of -144dBrA (fundamental at 250Hz) down to -170dBrA for the odd harmonics.  The test file can also be used in conjunction with artificially injected sinewave jitter by the Audio Precision, to also show how well the DAC rejects jitter.

We can see the primary low-level peak at 250Hz at just below -140dBFS (which is in the test file), some of the subsequent harmonics below -150dBFS, and the worst-case peaks adjacent the main 12kHz peak at a just above -140dBrA. This stellar J-Test FFT is an indication that the Tradutto should not be sensitive to jitter.

J-Test (coaxial input, 100ns of injected jitter)

The chart above shows the results of the J-Test test for the coaxial digital input (the optical input performed identically) measured at the balanced line-level output, with additional 2kHz sinewave jitter using the APx555’s built-in jitter generator. While the Tradutto did not manifest perfect jitter immunity, the characteristic sidebands at 10kHz and 12kHz are at a very low -135dBrA down, or 0.00002%. Even with a very significant 1000us of injected jitter (not shown), not only did the Tradutto not lose sync with the signal, but exhibited 10/12kHz sideband amplitudes of only -115dBrA, or 0.0002%. These are outstanding results.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone

The chart above shows a fast Fourier transform (FFT) of the balanced-line level output with white noise at -4dBFS (blue/red), and a 19.1kHz sinewave at 0dBFS, both fed to the coaxial digital input, sampled at 16/44.1 (purple/green). The sharp roll-off above 20kHz in the white-noise spectrum shows the implementation of the brick-wall-type reconstruction filter. There are no imaged aliasing artifacts in the audioband above the -135dBrA noise floor. The primary aliasing signal at 25kHz is at -110dBrA, or 0.0003%, while the second, third, and fourth distortion harmonics (38.2, 57.3, 76.4kHz) of the 19.1kHz range from -100 to -130dBrA, or 0.001% to 0.00003%.

THD ratio (unweighted) vs. frequency vs. load (24/96)

The chart above shows THD ratios at the balanced line-level output into 200k ohms (blue/red) and 600 ohms (purple/green) as a function of frequency for a 24/96 dithered 1kHz signal at the coaxial input. The 200k and 600 ohms data are very close to one another up until around 5kHz, hovering between 0.0002% and 0.0005%. The right channel also slightly outperformed the right channel by 2-3dB. Beyond 5kHz, THD values peaked for the 600-load data at only 0.0015% at roughly 15kHz.

THD ratio (unweighted) vs. frequency vs. sample rate (16/44.1 and 24/96)

The chart above shows THD ratios at the balanced line-level output into 200k ohms as a function of frequency for a 16/44.1 (blue/red) and a 24/96 (purple/green) dithered 1kHz signal at the coaxial input. The 24/96 and 16/44.1 data are very close to one another, with the same trend of the right channel slightly outperforming the right channel by 2-3dB. THD values are essentially flat from 20Hz to 20kHz, hovering between 0.0002% and 0.0006% across the audioband.

THD+N ratio (unweighted) vs. frequency vs. sample rate (16/44.1 and 24/96)

The chart above shows THD+N ratios at the balanced line-level output into 200k ohms as a function of frequency for a 16/44.1 (blue/red) and a 24/96 (purple/green) dithered 1kHz signal at the coaxial input. Due to the lower noise floor at 24 bits, the 24/96 outperformed the 16/44.1 data throughout, with a fairly constant THD+N value of 0.0005%, compared to the 16/44.1 data at 0.002%.

THD ratio (unweighted) vs. output (16/44.1 and 24/96) at 1kHz

The chart above shows THD ratios measured at the balanced output as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green). The 24/96 outperformed the 16/44.1 data at lower input stimuli, with a THD range from around 0.2% at 200 uVrms down to 0.0001% at 1Vrms (right channel), while the 16/44.1 ranged from 5% at 200uVrms down to 0.0005% at 2-4Vrms (left channel).

THD+N ratio (unweighted) vs. output (16/44.1 and 24/96) at 1kHz

The chart above shows THD+N ratios measured at the balanced output as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green). Due to the lower noise floor at 24 bits, the 24/96 outperformed the 16/44.1 data throughout, with a THD+N range from 5% at 200uVrms down to 0.0005% at 3-4Vrms, while the 16/44.1 varied from 50% at 200uVrms down to 0.002% at the maximum output voltage of 4.3Vrms.

FFT spectrum – 1kHz (digital input, 16/44.1 data at 0dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 16/44.1. The worst-case signal harmonics is at 3kHz at -110dBrA, or 0.0003%. We can also see small power-supply noise peak at 60Hz at -130dBrA, or 0.00003%. When an FFT was collected with a bandwidth of 1MHz (not shown), two distinct peaks at 353.8kHz and 351.8kHz could be seen; these peaks are distinct IMD products of the signal and an 8x oversampling clock (44.1kHz x 8 = 352.8kHz).

FFT spectrum – 1kHz (digital input, 24/96 data at 0dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 24/96. Due to the increased bit depth, the noise floor is much lower compared to the 16/44.1 FFT. The signal harmonics are effectively the same as with the 16/44.1 FFT, with the worst-case peak at 3kHz at -110dBrA, or 0.0003%. The remaining signal harmonics are below -120dBrA, or 0.0001%. We also see power-supply related noise peaks at the fundamental (60Hz) at -130dBrA, or 0.00003%, and at the third harmonic (180Hz) at -140dBrA, or 0.00001%. When an FFT was collected with a bandwidth of 1MHz (not shown), two distinct peaks at 769kHz and 767kHz could be seen; these peaks are distinct IMD products of the signal and an 8x oversampling clock (96kHz x 8 = 768kHz).

FFT spectrum – 1kHz (digital input, 16/44.1 data at -90dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 16/44.1 at -90dBFS. The signal peak has the correct amplitude, and we see no signal harmonics above the noise floor within the audioband.

FFT spectrum – 1kHz (digital input, 24/96 data at -90dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 24/96 at -90dBFS. The signal peak has the correct amplitude, and we see effectively no signal harmonic peaks within the audioband above the -160dBrA noise floor.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, 16/44.1)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced output into 200k ohms for the coaxial input at 16/44.1. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 4.3Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -120dBRA, or 0.0001% for the left channel, and imperceptible above the noise for the right channel, while the third-order modulation products, at 17kHz and 20kHz are at -120dBrA, or 0.0001%.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, 24/96)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced output into 200k ohms for the coaxial input at 24/96. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 4.3Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -120dBRA (left), or 0.0001%, and -135dBrA (right), or 0.00002%, while the third-order modulation products, at 17kHz and 20kHz are at -120dBrA, or 0.0001%. This is a stellar IMD result.

Diego Estan
Electronics Measurement Specialist

Details

Parent Category: Products

Category: Digital-to-Analog Converter Measurements

Created: 01 September 2021

Link: reviewed by Evan McCosham on SoundStage! Hi-Fi on September 1, 2021

General Information

All measurements taken using an Audio Precision APx555 B Series analyzer.

The Terminator-Plus was conditioned for 30 minutes at 0dBFS (4.3Vrms out) into 200k ohms before any measurements were taken.

The Terminator-Plus offers several digital inputs: AES-EBU (XLR), coaxial S/PDIF (RCA and BNC), optical S/PDIF (TosLink), and USB. There are two line-level outputs (balanced XLR and unbalanced RCA). Comparisons were made between unbalanced and balanced line-level outputs, and aside from the 6dB extra voltage over balanced, there were no appreciable differences observed in terms of THD and noise when the output voltage was normalized. Noise was slightly lower over the balanced outputs, yielding a little more than the expected 6dB boost in dynamic range over the unbalanced outputs (129 vs 121dB  with 24/192 data, A-weighted).

There are two filter settings, Sharp and Slow, available in OS (oversampling) mode. There is also a NOS (non-oversampling) mode that can be activated. All measurements below, unless otherwise stated, are for the coaxial digital input, and the balanced output, using the Sharp filter.

Despite Denafrips’ claim that the DAC can be operated in NOS mode, according to our measurements and the explanations provided by the GoldenSound YouTube Channel, it appears that NOS mode is not actually true NOS for the following reasons.

• When wideband (1MHz) FFTs are observed at 44.1kHz and 96kHz sample rates (see graphs further in the report), there is a peak in the spectrum at 352.8kHz (exactly 8x oversampling) in the 44.1kHz spectrum and 384kHz (exactly 4x oversampling) in the 96kHz spectrum, for both NOS and OS modes. No oversampling peaks should be seen.
  
  
• In a NOS DAC, square-wave rise times should be the same regardless of sample rate, however, with the Terminator-Plus in NOS mode, we found the rise time with a 100Hz square wave to be exactly 2x as fast when sampled at 96kHz (12.5us) versus 48kHz (25us).
  
  
• The reproduction of a 15kHz sinewave (not published here) in NOS mode appeared to have linear interpolation applied.
  
  
• The theoretical attenuation for a sine wave at 22.05kHz for a true NOS DAC is -3.9dB, while we measured -4.9dB.
  
  
• Finally, the impulse response of a NOS DAC should be a square, however, the Terminator-Plus in NOS mode exhibited a triangular-shaped impulse response (see graph further below), indicating once again, some form of linear interpolation between samples.

Published specifications vs. our primary measurements

The table below summarizes the measurements published by Denafrips for the Terminator-Plus compared directly against our own. The published specifications are sourced from Denafrips’ website, either directly or from the manual available for download, or a combination thereof. With the exception of frequency response, where the Audio Precision bandwidth is set at its maximum (DC to 1MHz), assume, unless otherwise stated, the coaxial digital input (24-bit/96kHz 1kHz sine wave at 0dBFS), the balanced line-level outputs into 200k ohms using a measurement input bandwidth of 10Hz to 90kHz, and the worst-case measured result between the left and right channel.

Our primary measurements revealed the following using the coaxial input and the balanced line-level outputs (unless specified, assume a 1kHz sine wave at 0dBFS, 200k ohms loading, 10Hz to 90kHz bandwidth, OS with Sharp filter engaged):

Frequency response (16/44.1, 24/96, 24/192, all OS mode with Sharp filter)

The plot above shows the Terminator-Plus frequency response as a function of sample rate. The blue/red traces are for a 16-bit/44.1kHz dithered digital input signal from 5Hz to 22kHz, the purple/green traces are for a 24/96 dithered digital input signal from 5Hz to 48kHz, and finally orange/pink represents 24/192 from 5Hz to 96kHz. The behavior at low frequencies is the same for the digital input types—perfectly flat down to 5Hz. The behavior at high frequencies for all three digital sample rates is as expected, offering sharp filtering around 22k, 48k, and 96kHz (half the respective sample rates for each). The -3dB point for each sample rate is roughly 21.2, 47, and 84kHz, respectively. It is also obvious from the plots above that the 44.1kHz sampled input signal offers the most “brick-wall”- type behavior, while the attenuation of the 96kHz and 192kHz sampled input signals approaching the corner frequencies (48kHz and 96kHz) is more gentle. In the graph above and most of the graphs below, only a single trace may be visible. This is because the left channel (blue, purple, or orange trace) is performing identically to the right channel (red, green, or pink trace), and so they perfectly overlap, indicating that the two channels are ideally matched.

Frequency response (16/44.1 with Sharp and Slow filters, and NOS mode)

The plots above show frequency response for a 0dBFS input signal sampled at 44.1kHz for the Sharp filter (blue), the Slow filter (purple), and NOS mode (orange) into a 200k ohm-load for the left channel only. The graph is zoomed in from 1kHz to 22kHz, and extra sampling points were introduced around the corner frequency (the “knee”) to highlight the various responses of the three filters. We can see that the Sharp filter provides the most “brick-wall”-type response, while the Slow filter shows earlier attenuation but is similar to the Sharp filter. In NOS mode, the behavior is similar to a single-pole analog low-pass filter, with a corner frequency at roughly 15kHz. The -3dB points are at 21.5kHz, 20.9kHz, and 14.5kHz for Sharp, Slow, and NOS respectively. 

Phase response vs. sample rate (16/44.1, 24/96, 24/192, all OS mode with Sharp filter)

Above are the phase response plots from 20Hz to 20kHz for the coaxial input, measured at the balanced output, using the Sharp filter setting. The blue/red traces are for a dithered 16/44.1 input signal at -20dBFS, the purple/green for 24/96, and the orange/pink for 24/192. Additionally the brown/green traces are for the 24/96 data but with the Phase button disengaged, showing that the Phase function works as advertised (180 degrees shift, though the Phase button must be engaged for correct polarity). There’s a worst-case phase shift of around 80 degrees at 20kHz for the 16/44.1 signals, and less than 20 degrees for the 24/96 and 24/192 signals.

Digital linearity (16/44.1 and 24/96 data)

The chart above shows the results of a linearity test for the coaxial digital input (the optical input performed identically) for both 16/44.1 (blue/red) and 24/96 (purple/green) input data, measured at the balanced line-level output. The digital input is swept with a dithered 1kHz input signal from -120dBFS to 0dBFS, and the output is analyzed by the APx555. The ideal response is a straight flat line at 0dB (i.e., the amplitude of the digital input perfectly matches the amplitude of the measured analog output). The 24/96 input data showed a worst-case -1dB deviation just above -120dBFS, and the 16/44.1 data +3dB at -120dBFS. Linearity was also verified below -120dBFS down to -140dBFS, where both 24/96 and 16/44.1 data deviated considerably from the 0dB reference.

Impulse response using Audio Precision Transfer Function measurement (Sharp and Slow filters, and NOS mode)

The chart above shows the impulse response for the two different filter types in OS and NOS modes, for a -20dBFS 16/44.1 dithered input stimulus, measured at the balanced line-level output. The blue plot represents the Sharp filter, the purple represent the Slow filter, and the orange represents NOS mode. We see virtually identical symmetry and behavior in the Sharp and Slow filter settings. NOS mode also yielded a symmetrical impulse response, but with less pre- and post-ringing effects. Of note here is that this is not the shape of NOS DAC impulse response (it should be a square), but that of a typical sinc filter.

Impulse Response using Audio Precision Signal Acquisition measurement with true 0dBFS digital impulse WAV file (Sharp and Slow filters, and NOS mode)

The chart above shows the impulse response for the Terminator-Plus in NOS (blue/red) and OS (Sharp filter, purple/green) modes, for a looped 24/44.1 test file that moves from digital silence to full 0dBFS (all “1”s) for one sample period, then back to digital silence. The previous chart was generated using the Audio Precision Transfer Function measurement, which applies an inverse FFT to a noise signal applied to the DAC. We performed this measurement because we wanted to explore the NOS impulse response further with a true digital impulse applied to the input of the DAC. In OS mode, both this measurement and the previous one show the same typical sinc function response of the oversampling reconstruction filter. In NOS mode however, when zoomed in . . .

. . . we see a stair-stepped triangular-shaped impulse response, where, if the DAC where truly NOS, we would see a square shape. In addition, the frequency of the stair-stepping in the impulse response is exactly 352.8kHz, or 8x oversampling at 44.1kHz.

J-Test (coaxial input with Sharp filter)

The plot above shows the results of the J-Test test for the coaxial digital input measured at the balanced line-level output with the Sharp filter engaged. J-Test was developed by Julian Dunn the 1990s. It is a test signal—specifically, a -3dBFS undithered 12kHz square wave sampled (in this case) at 48kHz (24 bit). Since even the first odd harmonic (i.e., 36kHz) of the 12kHz square wave is removed by the bandwidth limitation of the sampling rate, we are left with a 12kHz sine wave (the main peak). In addition, an undithered 250Hz square wave at -144dBFS is mixed with the signal. This test file causes the 22 least significant bits to constantly toggle, which produces strong jitter spectral components at the 250Hz rate and its odd harmonics. The test file shows how susceptible the DAC and delivery interface are to jitter, which would manifest as peaks above the noise floor at 500Hz intervals (e.g., 250Hz, 750Hz, 1250Hz, etc.). Note that the alternating peaks are in the test file itself, but at levels of -144dBrA (fundamental at 250Hz) down to -170dBrA for the odd harmonics. The test file can also be used in conjunction with artificially injected sine-wave jitter by the Audio Precision, to also show how well the DAC rejects jitter.

We can see the primary low level peak at 250Hz at just below -140dBFS (which is in the test file), some of the subsequent harmonics below -150dBFS, and the worst-case peaks adjacent the main 12kHz peak at a very low -140dBrA. This outstanding J-Test FFT is an indication that the Terminator-Plus should not be sensitive to jitter.

J-Test (coaxial input in NOS mode)

The plot above shows the results of the J-Test test for the coaxial digital input measured at the balanced line-level output, in NOS mode. The FFT is similar but slightly worse that in normal OS (Sharp filter) mode, with the worst-case peaks adjacent the primary 12kHz peak reaching -135dBrA.

We also tried adding additional 2kHz sine-wave jitter using the APx555’s built-in jitter generator, in both OS and NOS modes, however, the Terminator-Plus has essentially perfect jitter immunity. Without any jitter immunity, clear sidebands peaks at 10kHz and 14kHz (12kHz main signal +/- 2kHz jitter signal) manifest in the FFT, however, with the Terminator-Plus, with both the coaxial and optical input (below) in OS and NOS modes, none could be seen even with 1000ns of added jitter. With the maximum allowable jitter by the APx555 (1592ns), the Terminator-Plus did however lose sync with the digital signal.

J-Test (optical input with Sharp filter)

The plot above shows the results of the J-Test test for the optical digital input measured at the balanced line-level output. For this test, the optical input yielded effectively the same, although slightly worse, results compared to the coaxial input, with a few more spurious peaks visible below -140dBrA.

J-Test (optical input in NOS mode)

The plot above shows the results of the J-Test test for the optical digital input measured at the balanced line-level output, in NOS mode. The FFT is basically identical to the NOS coaxial FFT plot above.

Wideband FFT spectrum of white noise and 19.1kHz sine-wave tone (Sharp filter)

The plot above shows a fast Fourier transform (FFT) of the balanced line-level output with white noise at -4dBFS (blue/red), and a 19.1 kHz sine wave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using the Sharp filter setting. The fast roll-off above 20kHz in the white-noise spectrum shows the implementation of the brick-wall type reconstruction filter. There are only very small imaged aliasing artifacts in the audioband in the -125dBrA range. The primary aliasing signal at 25kHz is at -70dBrA, while the second, third, and fourth distortion harmonics (38.2, 57.3, 76.4kHz) of the 19.1kHz range from -95 to-120dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sine-wave tone (Slow filter)

The plot above shows a fast Fourier transform (FFT) of the balanced-line level output with white noise at -4 dBFS (blue/red), and a 19.1 kHz sine wave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using the Slow filter setting. The 19.1kHz FFT is full of distortion, and we are not sure why this occurred, but it was repeatable. Reducing the 19.1kHz signal down to -3dBFS resulted in a clean FFT, which we can see below . . .

Wideband FFT spectrum of white noise and 19.1kHz sine-wave tone (Slow filter, -3dBFS)

At -3dBFS, we can see a slightly slower roll-off above 20kHz in the white-noise spectrum for this Slow filter compared to the Fast filter. There is one more significant imaged aliasing artifact within the audioband at around 13kHz, measuring -115dBrA. The primary aliasing signal at 25kHz is significant at almost -25dBrA, while the second, third, and fourth distortion harmonics (38.2, 57.3, 76.4kHz) of the 19.1kHz tone are down near at -100dBrA or below. The IMD products between the sampling frequency (44.1kHz) and both the 19.1kHz primary tone and 25kHz alias at 63.2kHz and 69.1kHz are also predominant, at around -75dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sine-wave tone (NOS mode)

The plot above shows a fast Fourier transform (FFT) at the balanced line-level output with white noise at -4dBFS (blue/red), and a 19.1 kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using the NOS setting. We see the typical soft roll-off in the white-noise spectrum down to the 44.1kHz sampling frequency, then another rise and fall in the noise spectrum down to a doubling of the sampling frequency (88.2kHz), and so on. The worst-case imaged aliasing artifact in the audioband is at around 13kHz, measuring -105dBrA. Predictably, the primary aliasing signal at 25kHz is significant at -10dBrA, while the second, third, and fourth distortion harmonics (38.2, 57.3, 76.4kHz) of the 19.1kHz tone are in the -110dBrA range. The IMD products between the sampling frequency (44.1kHz) and both the 19.1kHz primary tone and 25kHz alias at 63.2kHz and 69.1kHz are also predominant at around -30dBrA.

THD ratio (unweighted) vs. frequency vs. load (24/96)

The chart above shows THD ratios at the balanced line-level output into 200k ohms (blue/red) and 600 ohms (purple/green) as a function of frequency for a 24/96 dithered 1kHz signal at the coaxial input. The 200k and 600 ohms data are very close to one another. Interestingly, the lower impedance load yielded the lower THD results. Between 20Hz and about 300Hz, there is a 5dB improvement in THD ratios at 600 ohms compared to 200k ohms, and also between right and left channels. Above 1kHz, the difference between all traces is no more than 2-3dB. THD values are as low as 0.0004% between 20 and 200Hz into 600 ohms for the right channel, and at worst, at little over 0.002% at 20kHz for the left channel into 200k ohms.

THD ratio (unweighted) vs. frequency vs. sample rate (16/44.1 and 24/96)

The chart above shows THD ratios at the balanced line-level output into 200k ohms as a function of frequency for a 16/44.1 (blue/red) and a 24/96 (purple/green) dithered 1kHz signal at the coaxial input. The 24/96 and 16/44.1 data are essentially identical, with the same trend of the right channel slightly outperforming the left channel. THD values are a little lower at lower frequencies, but overall, hover between 0.001 and 0.002% across the audioband.

THD ratio (unweighted) vs. output (16/44.1 and 24/96)

The chart above shows THD ratios measured at the balanced output as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green). The 24/96 outperformed the 16/44.1 data at lower input stimuli, with a THD range from 0.6% at 200uVrms down to 0.0004% at 3Vrms (right channel), while the 16/44.1 ranged from 5% at 200uVrms down to 0.0004% at 2Vrms (right channel).

THD+N ratio (unweighted) vs. output (16/44.1 and 24/96)

The chart above shows THD+N ratios measured at the balanced output as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green). Due to the lower noise floor at 24 bits, the 24/96 outperformed the 16/44.1 data throughout, with a THD+N range from 3% at 200uVrms down to just below 0.002% at 3Vrms, while the 16/44.1 varied from 50% at 200uVrms down to 0.003% at the maximum output voltage of 4.3Vrms.

FFT spectrum – 1kHz (digital input, 16/44.1 data at 0dBFS, with Sharp filter)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 16/44.1 in OS mode with the Sharp filter. The worst-case signal harmonic is at 3kHz near -95dBrA, or 0.002%, with other even and odd harmonics at -110dbRA, or 0.0003%, and below. There is a small peak due to power supply noise seen at the mains frequency of 60Hz at -125dBrA. We can also see a clear peak at 352.8kHz, indicating 8x oversampling (i.e., 44.1kHz x 8).

FFT spectrum – 1kHz (digital input, 16/44.1 data at 0dBFS, in NOS mode)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 16/44.1 in NOS mode. Up to 20kHz or so, the FFT is nearly identical to the one above taken in OS mode. Above 20kHz, however, it’s clear that far more noise artifacts can be seen, including the two predominant peaks at -65dBrA due to IMD products between the sampling frequency (44.1kHz) and signal (1kHz) at 43.1 and 45.1kHz. Here we can still see a clear peak at 352.8kHz, indicating that 8x oversampling is likely being applied.

FFT spectrum – 1kHz (digital input, 24/96 data at 0dBFS, with Sharp filter)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the balanced output into 200k ohms for the coaxial digital input, sampled at 24/96 in OS mode with the Sharp filter. Due to the increased bit depth, the noise floor is much lower compared to the 16/44.1 FFT. The signal harmonics are effectively the same as with the 16/44.1 FFT, with the worst-case peak at 3kHz near -95dBrA, or 0.002%. With the lower noise floor, we can not only see the 60Hz peak due to power supply noise at -125dBrA, but also a small peak at 180Hz (third harmonic) near -145dBrA. We can also see a clear peak at 384kHz, indicating 4x oversampling (i.e., 96kHz x 8).

FFT spectrum – 1kHz (digital input, 24/96 data at 0dBFS, in NOS mode)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 24/96 in NOS mode. Up to 20kHz or so, the FFT is nearly identical to the one above taken in OS mode. The FFT looks almost identical to the 24/96 FFT in OS mode above, with the exception of a predominant peak at -85dBrA due to the IMD product between the sampling frequency (96kHz) and signal (1kHz) at 95kHz. Here we can still see a clear peak at 384kHz, indicating that 4x oversampling is likely being applied.

FFT spectrum – 1kHz (digital input, 16/44.1 data at -90dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 16/44.1 at -90dBFS in OS mode using the Sharp filter. The signal peak has the correct amplitude, and we see no signal harmonics above the noise floor within the audioband, and the same power-supply noise peak at 60Hz seen in the 0dBFS FFTs.

FFT spectrum – 1kHz (digital input, 24/96 data at -90dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 24/96 at -90dBFS in OS mode using the Sharp filter. The signal peak has the correct amplitude, and we see several signal harmonic peaks within the audio-band just above and below a very low-140dBrA, or 0.00001%. The same -125dBrA 60Hz noise peak can be seen here again, along with several other noise related peaks, but all are near or below -140dBrA.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, 16/44.1)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sine-wave stimulus tone measured at the balanced output into 200k ohms for the coaxial input at 16/44.1 in OS mode using the Sharp filter. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 4.3Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -110/-125dBRA (right/left), or 0.0003/0.00006%, while the third-order modulation products, at 17kHz and 20kHz, are higher, at around -105dBrA, or 0.0006%.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, 24/96)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sine-wave stimulus tone measured at the balanced output into 200k ohms for the coaxial input at 24/96 in OS mode using the Sharp filter. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 4.3Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -110/-125dBRA (right/left), or 0.0003/0.00006%, while the third-order modulation products, at 17kHz and 20kHz, are higher, at around -105dBrA, or 0.0006%.

Diego Estan
Electronics Measurement Specialist

Details

Parent Category: Products

Category: Digital-to-Analog Converter Measurements

Created: 15 August 2021

Link: reviewed by Mark Phillips on SoundStage! Access on August 15, 2021

General Information

All measurements taken using an Audio Precision APx555 B Series analyzer.

The Aries II was conditioned for 30 min at 0dBFS (4.4Vrms output) in to 200k ohms before any measurements were taken.

The Aries II offers five digital inputs: two coaxial S/PDIF (RCA), two optical S/PDIF (TosLink), and one USB. There are two sets of line-level outputs (balanced XLR and unbalanced RCA). Comparisons were made between unbalanced and balanced line-level outputs, and aside from the 6dB extra voltage using the balanced outputs, there were no appreciable differences observed in terms of THD and noise.

There are two filter settings labeled Sharp and Slow, available in OS (oversampling) mode. There is also a NOS (non-oversampling) mode that can be activated. All measurements below, unless otherwise stated, are for the coaxial digital input and the balanced output, with the Sharp filter engaged.

Despite Denafrips’ claim that the DAC can be operated in NOS mode, according to our measurements and the explanations provided by the GoldenSound YouTube channel, it appears that NOS mode is not actually true NOS for the following reasons:

• When wideband (1MHz) FFTs are observed at 44.1kHz and 96kHz sample rates (see graphs further down in the report), there is a peak in the spectrum at 705.6kHz (exactly 16x oversampling) in the 44.1kHz spectrum and 768kHz (exactly 8x oversampling) in the 96kHz spectrum, for both NOS and OS modes. True NOS would have no oversampling.
  
  
• With a NOS DAC, square-wave rise times should be the same regardless of sample rate; however, with the Aries II in NOS mode, we found the rise time with a 100Hz square wave to be exactly 2x as fast when sampled at 96kHz (12.5us) versus 48kHz (25us).
  
  
• The reproduction of a 15kHz sine wave (not published here) in NOS mode appeared to have linear interpolation applied. With a true NOS DAC, there would be no interpolation.
  
  
• The theoretical attenuation for a sine wave at 22.05kHz for an NOS DAC is -3.9dB, while we measured -4.8dB.
  
  
• Finally, the impulse response of a NOS DAC should be a square; however, the Aries II in NOS mode exhibited a triangular-shaped impulse response (see chart further below), indicating, once again, some form of linear interpolation between samples.

Published specifications vs. our primary measurements

The table below summarizes the measurements published by Denafrips for the Aries II compared directly against our own. The published specifications are sourced from Denafrips’ website, either directly or from the manual available for download, or a combination thereof. With the exception of frequency response, where the Audio Precision bandwidth is set at its maximum (DC to 1MHz), assume, unless otherwise stated, the coaxial digital input (24/96 1kHz sine wave at 0dBFS), the balanced line-level outputs into 200k ohms using a measurement input bandwidth of 10Hz to 90kHz, and the worst-case measured result between the left and right channels.

Our primary measurements revealed the following using the coaxial input and the balanced line-level outputs (unless specified, assume a 1kHz sine wave at 0dBFS, 200k ohms loading, 10Hz to 90kHz bandwidth, Sharp filter, OS engaged):

Frequency response (16/44.1, 24/96, 24/192, all OS mode with Sharp filter)

The chart above shows the Aries II frequency response as a function of sample rate, all in OS mode with the Sharp filter. The blue/red traces (left/right channels) are for a 16-bit/44.1kHz dithered digital input signal from 5Hz to 22kHz, the purple/green traces (left/right) are for a 24/96 dithered digital input signal from 5Hz to 48kHz, and finally orange/pink (left) represent 24/192 from 5Hz to 96kHz. The behavior at low frequencies is the same for all sampling rates—perfectly flat down to 5Hz. The behavior at high frequencies for all three sampling rates is as expected, offering steep filtering around 22, 48, and 96kHz (half the respective sample rate for each). The -3dB point for each sample rate is roughly 21.5, 45, and 86kHz respectively. It is also obvious from the plots above that the 44.1kHz sampled input signal offers the most “brick-wall”-type behavior, while the attenuation of the 96kHz and 192kHz sampled input signals approaching the corner frequencies (48kHz and 96kHz) is more gentle. In the chart above and most of the graphs below, only a single trace may be visible. This is because the left channel (blue, purple, or orange trace) is performing identically to the right channel (red, green, or pink trace), and so they perfectly overlap, indicating that the two channels are ideally matched.

Frequency response (16/44.1 with Sharp and Slow filters, and NOS mode)

The chart above shows the frequency-response for a 0dBFS input signal sampled at 44.1kHz for the Sharp filter (blue), the Slow filter (purple), and NOS mode (orange) into a 200k-ohm load for the left channel only. The graph is zoomed in from 1kHz to 22kHz, and extra sampling points were introduced around the corner frequency (the “knee”), to highlight the various responses of the various modes. We can see that the Sharp filter provides the most “brick-wall”-type response, while the Slow filter shows earlier attenuation but is similar to the Sharp filter. In NOS mode, the behavior is similar to a single-pole analog low-pass filter, with a corner frequency at roughly 15kHz. The -3dB points are, respectively, at 21.5kHz, 20.9kHz and 14.5kHz for Sharp, Slow, and NOS.

Phase response vs. sample rate (16/44.1, 24/96, 24/192, all OS mode with Sharp filter)

Above are the phase-response plots from 20Hz to 20kHz for the coaxial input, measured at the balanced output, using the Sharp filter setting. The blue/red traces (left/right channels) are for a dithered 16/44.1 input signal at -20dBFS, the purple/green (left/right) for 24/96, and the orange/pink (left/right) for 24/192. Additionally, the brown/green traces (left/right) are for the 24/192 data but with the Phase button engaged, showing that the function works as advertised, providing 180 degrees of phase shift (note that the Phase button must be disengaged for correct polarity). There’s a worst-case phase shift of around 80 degrees at 20kHz for the 16/44.1, and less than 20 degrees for the 24/96 and 24/192 input data.

Digital linearity (16/44.1 and 24/96 data)

The chart above shows the results of a linearity test for the coaxial digital input (the optical input performed identically) for both 16/44.1 (blue/red) and 24/96 (purple/green) input data, measured at the balanced line-level output. The digital input is swept with a dithered 1kHz input signal from -120dBFS to 0dBFS, and the output is analyzed by the APx555. The ideal response is a straight flat line at 0dB (i.e., the amplitude of the digital input perfectly matches the amplitude of the measured analog output). The 24/96 input data showed a worst-case of just under -1dB deviation just above -120dBFS, while the 16/44.1 data was +2dB (left) at -120dBFS. Linearity was also verified below -120dBFS down to -140dBFS, where both 24/96 and 16/44.1 data deviated considerably from the 0dB reference.

Impulse response using Audio Precision Transfer Function measurement (Sharp and Slow filters, and NOS mode)

The chart above shows the impulse response for the two different filter types in OS mode and NOS mode for a -20dBFS 16/44.1 dithered input stimulus, measured at the balanced line-level output. The blue plot represents the Sharp filter, the purple represents the Slow filter, and the orange represents NOS mode. We see virtually identical symmetry and behavior in the Sharp and Slow filter settings. NOS mode also yielded a symmetrical impulse response, but with less pre- and post-ringing. Of note here is that this is not the shape of NOS DAC impulse response (it should be a square), but that of a typical sinc-function filter.

Impulse Response using Audio Precision Signal Acquisition measurement with true 0dBFS digital impulse WAV file (Sharp and Slow filters, and NOS mode)

The previous chart was generated using the Audio Precision Transfer Function measurement, which applies an inverse FFT to a noise signal applied to the DAC. The chart above shows the impulse response for the Aries II in NOS (purple) and OS (Sharp filter is blue, Soft filter is orange) modes, for a looped 24/44.1 test file that moves from digital silence to full 0dBFS (all “1”s) for one sample period then back to digital silence. We provided this extra chart because we wanted to explore the NOS impulse response further with a true digital impulse applied to the input of the DAC. In OS mode, this measurement and the previous measurement show the same typical sinc-function response of the oversampling reconstruction filter. The Soft filter impulse response does exhibit a strange notch in the center, which we cannot explain. Of note is that there is also a notch in the frequency response using the Soft filter near 20kHz for a 16/44.1 input signal (see the frequency-response chart above). In NOS mode, however, when zoomed in . . .

. . . we see a stair-stepped triangular-shaped impulse response, where if the DAC were truly NOS, we would see a square shape. In addition, the frequency of the stair-stepping in the impulse response is exactly 705.6kHz, or 16x oversampling at 44.1kHz.

J-Test (coaxial input with Sharp filter)

The plot above shows the results of the J-Test test for the coaxial digital input measured at the balanced line-level output with the Sharp filter. J-Test was developed by Julian Dunn the 1990s. It is a test signal—specifically, a -3dBFS undithered 12kHz square wave sampled (in this case) at 48kHz (24 bit). Since even the first odd harmonic (i.e., 36kHz) of the 12kHz square wave is removed by the bandwidth limitation of the sampling rate, we are left with a 12kHz sine wave (the main peak). In addition, an undithered 250Hz square wave at -144dBFS is mixed with the signal. This test file causes the 22 least significant bits to constantly toggle, which produces strong jitter spectral components at the 250Hz rate and its odd harmonics. The test file shows how susceptible the DAC and delivery interface are to jitter, which would manifest as peaks above the noise floor at 500Hz intervals (e.g., 250Hz, 750Hz, 1250Hz, etc). Note that the alternating peaks are in the test file itself, but at levels of -144dBrA (fundamental at 250Hz) down to -170dBrA for the odd harmonics. The test file can also be used in conjunction with artificially injected sine-wave jitter by the Audio Precision, to also show how well the DAC rejects jitter.

We can see the primary low-level peak at 250Hz at just below -140dBFS (which is in the test file), some of the subsequent harmonics below -150dBFS, and the worst-case peaks adjacent the main 12kHz peak at a just below -130dBrA. This stellar J-Test FFT is an indication that the Aries II should not be sensitive to jitter.

J-Test (coaxial input in NOS mode)

The chart above shows the results of the J-Test test for the coaxial digital input measured at the balanced line level output, in NOS mode. The FFT is nearly identical to that in normal OS (Sharp filter) mode, with the worst-case peaks adjacent the primary 12kHz peak reaching just above -135dBrA.

We also tried adding additional 2kHz sine wave jitter using the APx555’s built-in jitter generator, in both OS and NOS modes; however, the Aries II has essentially perfect jitter immunity. Without any jitter immunity, clear sidebands peaks at 10kHz and 14kHz (12kHz main signal +/- 2kHz jitter signal) manifest in the FFT, but with the Aries II, with both the coaxial and optical inputs (below) in OS and NOS modes, none could be seen even with 1000ns of added jitter.

J-Test (optical input with Sharp filter)

The plot above shows the results of the J-Ttest test for the optical digital input measured at the balanced line-level output. For this test, the optical input yielded effectively the same results compared to the coaxial input.

J-Test (optical input in NOS mode)

The plot above shows the results of the J-Test test for the optical digital input measured at the balanced line level output, in NOS mode. The FFT is basically identical to the NOS coaxial FFT plot above.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (Sharp filter)

The chart above shows a fast Fourier transform (FFT) of the balanced-line level output with white noise at -4dBFS (blue/red) and a 19.1kHz sine wave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using the Sharp filter setting. The steep roll-off above 20kHz in the white-noise spectrum shows the implementation of the brick-wall type reconstruction filter. There are only very small imaged aliasing artifacts in the audioband in the -125dBrA range. The primary aliasing signal at 25kHz is at -70dBrA, while the second, third and fourth distortion harmonics (38.2, 57.3, 76.4kHz) of the 19.1kHz range from -105 to-110dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (Slow filter)

The chart above shows a fast Fourier transform (FFT) of the balanced-line level output with white noise at -4 dBFS (blue/red) and a 19.1 kHz sine wave at -1dBFS (0dBFS caused distortion) fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using the Slow filter setting. We can see a slightly slower roll-off above 20kHz in the white-noise spectrum for this Slow filter compared to the Sharp filter. There is also one more significant imaged aliasing artifact within the audioband at around 13kHz measuring just above -120dBrA (right channel). The primary aliasing signal at 25kHz is significant at almost -25dBrA, while the second, third, and fourth distortion harmonics (38.2, 57.3, 76.4kHz) of the 19.1kHz tone are down near at the -110dBrA and below level. The IMD products between the sampling frequency (44.1kHz) and both the 19.1kHz primary tone and 25kHz alias at 63.2kHz and 69.1kHz are also predominant at around -75dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (NOS mode)

The chart above shows a fast Fourier transform (FFT) at the balanced-line level output with white noise at -4 dBFS (blue/red) and a 19.1 kHz sine wave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using the NOS setting. We see the typical soft roll-off in the white-noise spectrum down to the 44.1kHz sampling frequency, then another rise and fall in the noise spectrum down to a doubling of the sampling frequency (88.2kHz), and so on. The worst-case imaged aliasing artifact in the audio band is at around 13kHz, measuring just below -110dBrA. Predictably, the primary aliasing signal at 25kHz is significant at -10dBrA, while the second, third, and fourth distortion harmonics (38.2, 57.3, 76.4kHz) of the 19.1kHz tone are in the -110 to 120dBrA range. The IMD products between the sampling frequency (44.1kHz) and both the 19.1kHz primary tone and 25kHz alias at 63.2kHz and 69.1kHz are also predominant at around -30dBrA.

THD ratio (unweighted) vs. frequency vs. load (24/96)

The chart above shows THD ratios at the balanced line-level output into 200k ohms (blue/red) and 600 ohms (purple/green) as a function of frequency for a 24/96 dithered 1kHz signal at the coaxial input. The 200k and 600 ohms data are very close to one another; interestingly, the lower impedance load yielded the slightly lower THD results (2-3dB improvement below 15kHz). The left channel also slightly outperformed the right channel by 2-3dB. THD values are as low as 0.0007% between 20 and 2kHz into 600 ohms for the left channel, and, at worst, 0.002% at 20kHz into 600 ohms.

THD ratio (unweighted) vs. frequency vs. sample rate (16/44.1 and 24/96)

The chart above shows THD ratios at the balanced line-level output into 200k ohms as a function of frequency for a 16/44.1 (blue/red) and a 24/96 (purple/green) dithered 1kHz signal at the coaxial input. The 24/96 and 16/44.1 data are very close to one another, with the same trend in the previous chart of the left channel slightly outperforming the right channel by 2-3dB. THD values are essentially flat from 20Hz to 20kHz, hovering between 0.0007 and 0.001% across the audioband.

THD ratio (unweighted) vs. output (16/44.1 and 24/96)

The chart above shows THD ratios measured at the balanced output as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green). The 24/96 outperformed the 16/44.1 data at lower input stimuli, with a THD range from around 1% at 200uVrms down to 0.0007% at 2-3Vrms (left channel), while the 16/44.1 ranged from 5% at 200uVrms down to 0.0007% at 2-3Vrms (left channel).

THD+N ratio (unweighted) vs. output (16/44.1 and 24/96)

The chart above shows THD+N ratios measured at the balanced output as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green). Due to the lower noise floor at 24 bits, the 24/96 outperformed the 16/44.1 data throughout, with a THD+N range from 3% at 200uVrms down to just below 0.002% at 3-4Vrms, while the 16/44.1 varied from 50% at 200uVrms down to 0.003% at the maximum output voltage of 4.4Vrms.

FFT spectrum – 1kHz (digital input, 16/44.1 data at 0dBFS, with Sharp filter)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the balanced output into 200k ohms for the coaxial digital input, sampled at 16/44.1 in OS mode with the Sharp filter. The worst-case signal harmonics are at 2 and 3kHz around -105dBrA, or 0.0006%, with other even and odd harmonics at -115dbRA, or 0.0002%, and below. There are no peaks due to power-supply noise. We can also see a clear peak at 705.6kHz, indicating 16x oversampling (i.e., 44.1kHz x 16).

FFT spectrum – 1kHz (digital input, 16/44.1 data at 0dBFS, in NOS mode)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the balanced output into 200k ohms for the coaxial digital input, sampled at 16/44.1 in NOS mode. Between 20Hz and 20kHz or so, the FFT is nearly identical to the one above taken in OS mode. Above 20kHz, however, it’s clear that far more noise artifacts can be seen, including the two predominant peaks at 70dBrA due to IMD products between the sampling frequency (44.1kHz) and signal (1kHz) at 43.1 and 45.1kHz. Here we can still see a clear peak at 705.6kHz, indicating that 16x oversampling is likely being applied.

FFT spectrum – 1kHz (digital input, 24/96 data at 0dBFS, with Sharp filter)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the balanced output into 200k ohms for the coaxial digital input, sampled at 24/96 in OS mode with the Sharp filter. Due to the increased bit-depth, the noise floor is much lower compared to the 16/44.1 FFT. The signal harmonics are effectively the same as with the 16/44.1 FFT, with the worst-case peak at 2 and 3kHz near -105dBrA, or 0.0006%. Even with the lower noise floor, we still cannot see any peaks due to power supply noise. We can also see a clear peak at 768kHz, indicating 8x oversampling (i.e., 96kHz x 8).

FFT spectrum – 1kHz (digital input, 24/96 data at 0dBFS, in NOS mode)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the balanced output into 200k ohms for the coaxial digital input, sampled at 24/96 in NOS mode. Between 20Hz and 20kHz or so, the FFT is nearly identical to the one above taken in OS mode, with the exception of a predominant peak at -80dBrA due to the IMD product between the sampling frequency (96kHz) and signal (1kHz) at 95kHz. Here we can still see a clear peak at 768kHz, indicating that 8x oversampling is likely being applied.

FFT spectrum – 1kHz (digital input, 16/44.1 data at -90dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the balanced output into 200k ohms for the coaxial digital input, sampled at 16/44.1 at -90dBFS in OS mode using the Sharp filter. The signal peak has the correct amplitude, and we see no signal or power-supply noise harmonics above the noise floor within the audioband.

FFT spectrum – 1kHz (digital input, 24/96 data at -90dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 24/96 at -90dBFS in OS mode using the Sharp filter. The signal peak has the correct amplitude, and we see several signal harmonic peaks within the audioband below a very low-130dBrA, or 0.00003%. Here we can see power-supply noise peaks just above the noise floor at 60Hz and 180Hz (-150dBrA, or 0.000003%, for the right channel).

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, 16/44.1)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sine-wave stimulus tone measured at the balanced output into 200k ohms for the coaxial input at 16/44.1 in OS mode using the Sharp filter. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 4.4Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -110dBRA, or 0.0003%, while the third-order modulation products, at 17kHz and 20kHz are at roughly the same level.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, 24/96)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sine-wave stimulus tone measured at the balanced output into 200k ohms for the coaxial input at 24/96 in OS mode using the Sharp filter. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 4.4Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -110dBRA, or 0.0003%, while the third-order modulation products, at 17kHz and 20kHz, are at roughly the same level.

Diego Estan
Electronics Measurement Specialist

Details

Parent Category: Products

Category: Digital-to-Analog Converter Measurements

Created: 15 June 2021

Link: reviewed by Mark Phillips on SoundStage! Access on June 15, 2021

General Information

All measurements taken using an Audio Precision APx555 B Series analyzer.

The DacMagic 200M was conditioned for 30 minutes at 0dBFS (4Vrms out) into 200k ohms before any measurements were taken.

The DacMagic 200M offers five physical digital input connections plus Bluetooth: two coaxial S/PDIF (RCA), two optical S/PDIF (TosLink), one USB Type B. However, the coaxial and optical inputs are “paired,” so only one of each can be used at a time, limiting the total number of inputs to three physical inputs plus Bluetooth.

There are two line-level outputs (balanced XLR and unbalanced RCA) and one headphone output (1/4″ TRS). There is a digital volume control for the headphone output, which can also be engaged for the line-level outputs, but was left in the fixed default setting (disabled) for all measurements. Comparisons were made between unbalanced and balanced line-level outputs, and aside from the 6dB extra voltage of unbalanced over balanced, the only other difference was a small variance in terms of noise and dynamic range. The unbalanced outputs offered slightly lower noise levels, but not enough to make up for the 6dB lower output voltage, which, with 24-bit/96kHz data, resulted in 3dB more dynamic range measured over the balanced output. With 16/44.1 input data, dynamic range measurements were the same over balanced and unbalanced outputs. In terms of input types (USB, coaxial, optical), there were no differences between each input type measured at both 16/44.1 and 24/96.

There are three filter settings labeled: Fast, Slow, and Short Delay. All measurements below, unless otherwise stated, are for the coaxial digital input, and the balanced output, using the Fast filter.

The DacMagic 200M volume control has no level indicator on the front panel. For a 0dBFS 1kHz input signal, using the full range of the volume control will yield from a minimum of about 50uVrms (-96dB) to 3.1Vrms (0dB) at the headphone output, in 47 steps. The lowest range of the volume control offers 6 to 3dB increments, the middle of the range 2dB steps, and the top of the range 1dB steps. The right channel was consistently 0.3dB higher in level than the left, which is also seen when the volume is fixed over the line-level outputs (see tables below).

Volume-control accuracy (measured at line-level outputs): left-right channel tracking

Published specifications vs. our primary measurements

The table below summarizes the measurements published by Cambridge Audio for the DacMagic 200M compared directly against our own. The published specifications are sourced from Cambridge Audio’s website, either directly or from the manual available for download, or a combination thereof. With the exception of frequency response, where the Audio Precision bandwidth is set at its maximum (DC to 1MHz), assume, unless otherwise stated, the coaxial digital input (24/96 1kHz sinewave at 0dBFS); the balanced line-level or unbalanced headphone outputs into, respectively, 200k ohms and 300 ohms; a measurement input bandwidth of 10Hz to 90kHz; and the worst-case measured result between the left and right channels.

*results for left/right channels shown due to 0.3dB difference

Our primary measurements revealed the following using the coaxial input and the balanced line-level outputs (unless specified, assume a 1kHz sinewave at 0dBFS, 200k ohms loading, 10Hz to 90kHz bandwidth):

Our primary measurements revealed the following using the coaxial input and the headphone output (unless specified, assume a 1kHz sinewave at 0dBFS output, 300 ohms loading, 10Hz to 90kHz bandwidth):

Frequency response (16/44.1, 24/96, 24/192)

The chart above shows the DacMagic 200M’s frequency response as a function of sample rate. The blue/red traces are for a 16-bit/44.1kHz, dithered input signal from 5Hz to 22kHz, the purple/green traces are for a 24/96, dithered input signal from 5Hz to 48kHz, and finally orange/pink represents 24/192 from 5Hz to 96kHz. The behavior at lower frequencies is the same for all signals: it is perfectly flat down to 5Hz. The behavior at high frequencies for all three digital sample rates is as expected, offering sharp filtering around 22, 48, and 96kHz (half the respective sample rate). The -3dB point for each sample rate is roughly 21, 46 and 91kHz, respectively. It is also obvious from the plots above that the 44.1kHz sampled input signal offers the most “brick-wall” type behavior, while the attenuation of the 96kHz and 192kHz sampled input signals approaching the corner frequencies (48kHz and 96kHz) is more gentle. In the graph above and most of the graphs below, only a single trace may be visible. This is because the left channel (blue, purple or orange trace) is performing identically to the right channel (red, green or pink trace), and so they perfectly overlap, indicating that the two channels are ideally matched.

Frequency response (16/44.1 with Fast, Slow, Short Delay fiters)

The plots above show frequency-response for a 0dBFS input signal sampled at 44.1kHz for the Fast filter (blue), the Slow filter (purple), and the Short Delay filter (orange) into a 200k ohm-load for the left channel only. The graph is zoomed in from 1kHz to 22kHz, and extra sampling points were introduced around the corner frequency (the “knee”) to highlight the various responses of the three filters. We can see that the Fast filter provides the most “brick-wall” type response, the Slow filter shows the earliest attenuation (-1dB at 17.4kHz), and the Short Delay filter is very similar to the Fast filter. The -3dB points for all three filters are close to 21kHz.

Phase response vs. sample rate (16/44.1, 24/96, 24/192 with Fast filter)

Above are the phase-response plots from 20Hz to 20kHz for the coaxial input, measured across the balanced output, using the Fast filter setting. The blue/red traces are for a dithered 16/44.1 input at 0dBFS, the purple/green for 24/96, and the orange/pink for 24/192. We can see that the DacMagic 200M does not invert polarity, with a worst-case phase shift of just under 40 degrees at 20kHz for the 16/44.1 and 24/96 input data, and nearly zero phase shift for the 24/192 input data.

Phase response vs. filter type (16/44.1)

The chart above shows the phase responses for the three different filter types, for a -20dBFS 16/44.1 dithered input stimulus, measured at the balanced line-level output of the DacMagic 200M. The blue plot represents the Fast filter, the purple the Slow filter, and the orange represents the Short Delay. As per Cambridge Audio’s descriptions of the filters’ behaviors, the Short Delay filter does show more phase shift at high frequencies compared to the other two filters.

Digital linearity (16/44.1 and 24/96 data)

The graph above shows the results of a linearity test for the coaxial digital input (the optical input performed identically) for both 16/44.1 (blue/red) and 24/96 (purple/green) input data, measured at the balanced line-level output of the DacMagic 200M. The digital input is swept with a dithered 1kHz input signal from -120dBFS to 0dBFS, and the output is analyzed by the APx555. The ideal response is a straight flat line at 0dB (i.e., the amplitude of the digital input perfectly matches the amplitude of the measured analog output). The 24/96 input data exhibited a strange, significant under-response below -100dBFS. This was confirmed with an FFT with a 24/96 -105dBFS input signal that resulted in a peak at around -150dBFS. The 16/44.1 input data performed well, only overresponding by a few dB at -120dBFS. From -100 to 0dBFS, both input data performed flawlessly, with output amplitudes perfectly matching the input stimulus.

Impulse response (Fast, Slow, Short Delay filters)

The chart above shows the impulse responses for the three different filter types, for a -20dBFS 16/44.1 dithered input stimulus, measured at the balanced line-level output of the DacMagic 200M. The blue plot represents the Fast filter, the purple the Slow filter, and the orange represents the Short Delay. Cambridge Audio describes each filter as follows: “The Fast setting consists of a sharp linear phase filter designed to ensure a very clean spectrum and minimize 'out-of-band' noise. The Slow setting lowers phase distortion and improves temporal characteristics by significantly reducing pre- and post-impulse ringing, however the anti-aliasing effect will be less effective. The Short Delay setting minimizes pre-impulse ringing and has quick temporal response. The anti-alias filtering is excellent on this setting, but at the expense of a minor phase distortion.” In the impulse responses shown above, we find the descriptions provided by Cambridge Audio to be valid. We see symmetry in the Fast and Slow filter settings, with more amplitude in the peaks and longer decay for the Fast filter, suggesting more taps in the DSP filter were used to achieve a greater “brick-wall” effect. The Short Delay filter is also as described, with minimized pre-impulse ringing.

J-Test (coaxial input)

The chart above shows the results of the J-Test test for the coaxial digital input measured at the balanced line-level output of the DacMagic 200M. The optical and USB inputs yielded the same results. The J-Test was developed by Julian Dunn the 1990s. It is a test signal, specifically a -3dBFS, 24-bit, undithered, 12kHz square wave sampled (in this case) at 48kHz. Since even the first odd harmonic (i.e., 36kHz) of the 12kHz square wave is removed by the bandwidth limitation of the sampling rate, we are left with a 12kHz sinewave (the main peak). In addition, an undithered, 250Hz square wave at -144dBFS is mixed with the signal. This test file causes the 22 least significant bits to constantly toggle, which produces strong jitter spectral components at the 250Hz rate and its odd harmonics. The test file shows how susceptible the DAC and delivery interface are to jitter, which would manifest as peaks above the noise floor at 500Hz intervals (e.g., 250Hz, 750Hz, 1250Hz, etc). Note that the alternating peaks are in the test file itself, but at levels of -144dBrA (fundamental at 250Hz) down to -170dBrA for the odd harmonics.  The test file can also be used in conjunction with artificially injected sinewave jitter by the Audio Precision, to also show how well the DAC rejects jitter.

The coaxial input shows some of the alternating 500Hz peaks in the audio band but at very low levels; below -140dBrA, or 0.00001%. This is an indication that the DacMagic 200M should not be sensitive to jitter.

J-Test (coaxial input, 2kHz sinewave jitter at 100ns)

The chart above shows the results of the J-Ttest test for the coaxial digital input measured at the balanced line-level output, with an additional 100ns of 2kHz sinewave jitter injected by the APx555. The results are very clear, as we see the sidebands at 10kHz and 14kHz (12kHz main signal +/-2kHz jitter signal) manifest at a very low -135dBrA. This is a clear indication that the DacMagic 200M has good jitter immunity. For this test, the optical input yielded effectively the same results. It is also worth noting, however, that at jitter levels exceeding 200ns, the DacMagic 200M would routinely lose sync with the signal entirely.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (Fast filter)

The plot above shows a fast Fourier transform (FFT) of the DacMagic 200M’s balanced-line level output with white noise at -4dBFS (blue/red), and a 19.1kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using the Fast filter setting. The sharp roll-off above 20kHz in the white noise spectrum shows the implementation of the brick-wall type reconstruction filter. There are absolutely no imaged aliasing artifacts in the audio band above the -135dBrA noise floor. The primary aliasing signal at 25kHz is at -115dBrA, or 0.0002%, while the second, third, and fourth distortion harmonics (38.2, 57.3, 76.4kHz) of the 19.1kHz tone are at around the same level. These data also corroborate Cambridge Audio’s description of the Fast filter: “very clean spectrum and minimize 'out-of-band' noise.”

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (Slow filter)

The plot above shows a fast Fourier transform (FFT) of the DacMagic 200M’s balanced-line level output with white noise at -4dBFS (blue/red), and a 19.1kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using the Slow filter setting. The slower roll-off above 20kHz in the white noise spectrum shows the implementation of a softer reconstruction filter compared to the Fast filter. There is one visible imaged aliasing artifact within the audioband at around 13kHz and -120dBrA, 0r 0.0001%. The primary aliasing signal at 25kHz is significant at almost -10dBrA, or 30%, while the second, third and fourth distortion harmonics (38.2, 57.3, 76.4kHz) of the 19.1kHz tone are down near the -120dBrA level, or 0.0001%. Also visible are other IMD products between the primary 19.1kHz tone and the 25kHz aliasing peak, such as the peak at 30.9kHz. These data also corroborate Cambridge Audio’s description of the Slow filter: “however the anti-aliasing effect will be less effective.”

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (Short Delay filter)

The plot above shows a fast Fourier transform (FFT) of the DacMagic 200M’s balanced-line level output with white noise at -4dBFS (blue/red), and a 19.1kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using the Short Delay filter setting. The sharp roll-off above 20kHz in the white noise spectrum shows the implementation of the brick-wall type reconstruction filter. There are absolutely no imaged aliasing artifacts in the audio band above the -135dBrA noise floor. The primary aliasing signal at 25kHz is at -110dBrA, while the second, third and fourth distortion harmonics (38.2, 57.3, 76.4kHz) of the 19.1kHz tone are at around the same level. These data also corroborate Cambridge Audio’s description of the Short Delay filter: “the anti-alias filtering is excellent on this setting, but at the expense of a minor phase distortion.” That phase distortion can be seen in a chart above.

THD ratio (unweighted) vs. frequency vs. load (24/96)

The chart above shows THD ratios at the balanced line-level output into 200k ohms (blue/red) and 600 ohms (purple/green) as a function of frequency for a 24/96, dithered 1kHz 0dBFS signal at the coaxial input. The 200k and 600 ohms data are nearly identical from 100Hz to 1kHz, hovering around a very low 0.00015%. At the frequency extremes, THD increased into 600 ohms vs 200k ohms, where we see 0.0006% vs 0.0002% at 20Hz, and 0.001% vs 0.0005% at around 15kHz.

THD ratio (unweighted) vs. frequency vs. sample rate (16/44.1 and 24/96)

The chart above shows THD ratios at the balanced line-level output into 200k ohms as a function of frequency for a 16/44.1 (blue/red) and a 24/96 (purple/green) dithered 1kHz 0dBFS signal at the coaxial input. The 24/96 data consistently outperformed the 16/44.1 data by about 5dB across the audioband. Above about 3kHz, the left channel outperformed the right channel by about 2-3dB, and above 5kHz, the THD ratios rose slightly and were identical for the 16/44.1 and 24/96 data at 0.0005% (right) and 0.0003% (left) at 12-15kHz. Across most of the audioband, THD values are very low, around 0.0003% (16/44.1) and 0.00015% (24/96).

THD ratio (unweighted) vs. output (16/44.1 and 24/96)

The chart above shows THD ratios measured at the balanced output as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green). Once again, the 24/96 outperformed the 16/44.1 data, with a THD range from 0.2% to  0.00015%, while the 16/44.1 ranged from 6% down to 0.0003%.

THD+N ratio (unweighted) vs. output (16/44.1 and 24/96)

The chart above shows THD+N ratios measured at the balanced output as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green). The 24/96 outperformed the 16/44.1 data, with a THD+N range from 4% down to  0.0003%, while the 16/44.1 ranged from 50% down to 0.002% at the maximum output voltage of 4Vrms.

FFT spectrum – 1kHz (digital input, 16/44.1 data at 0dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 16/44.1. We see signal harmonics at -125dBrA, or 0.00005%, and below at the odd harmonics of 3, 5, and 9kHz. No even-signal harmonics are visible in the audioband above the -135dBrA noise floor. There are also no power supply noise peaks to speak of to the left of the main signal peak.

FFT spectrum – 1kHz (digital input, 24/96 data at 0dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 24/96. Due to the increased bit-depth, the noise floor is much lower compared to the 16/44.1 FFT at a very low -160dBrA. We see signal harmonics ranging from -125dBrA, or 0.00005%, down to -150dBrA, or 0.000003%. With the lower noise floor, we can see even order harmonics, for example at 2kHz where the peaks are at -135/145dBrA (right/left), or 0.00002/0.000006%. Here also, there are no power supply noise peaks to speak of to the left of the main signal peak.

FFT spectrum – 1kHz (digital input, 16/44.1 data at -90dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 16/44.1 at -90dBFS. We see no signal harmonics above the noise floor within the audioband.

FFT spectrum – 1kHz (digital input, 24/96 data at -90dBFS)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 24/96 at -90dBFS. We only see two signal harmonic peaks within the audioband above the very low noise floor, one at 3kHz for the left channel at -155dBrA, or 0.000002%, and another at 7kHz just above -150dBrA, or 0.000002%, for the right channel.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, 16/44.1)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced output into 200k ohms for the coaxial input at 16/44.1. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 4Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -130dBRA, or 0.00003%, just barely peaking above the noise floor, while the third-order modulation products, at 17kHz and 20kHz are higher, at around -120dBrA, or 0.0001%.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, 24/96)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced output into 200k ohms for the coaxial input at 24/96. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 4Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is just below -130dBRA, or 0.00003%, while the third-order modulation products, at 17kHz and 20kHz, are higher, at around -120dBrA, or 0.0001%.

Diego Estan
Electronics Measurement Specialist