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Parent Category: Products

Category: Preamplifier Measurements

Created: 01 May 2025

Link: reviewed by AJ Wykes on SoundStage! Simplifi on May 1, 2025

General Information

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

The WiiM Ultra was conditioned for 30 minutes at 2Vrms in/out into 100k ohms before any measurements were taken.

The Ultra offers one set of line-level analog inputs (RCA), one set of moving-magnet (MM) phono-level analog inputs (RCA), one S/PDIF digital input (optical), one HDMI (ARC) input, as well as streaming over ethernet or WiFi and Bluetooth inputs. Available outputs are a set of analog line-level (RCA), S/PDIF digital over optical (TosLink) and coaxial (RCA), as well as a single subwoofer output (configurable with internal bass management). There is also a ⅛″ TRS headphone output jack on the front panel. For the purposes of these measurements, unless otherwise stated, the following inputs were evaluated: digital optical S/PDIF (RCA), analog line-level (RCA), as well as phono MM (RCA).

Most measurements were made with a 2Vrms line-level and 0dBFS digital input with the volume set to achieve 2Vrms at the output (volume at maximum). Of note is that the Ultra digitizes all incoming analog signals to perform volume, EQ, and bass management. The ADC’s bit depth and sample rate are user-selectable (unless otherwise stated, 24/192 was used). As a line-level analog preamp, the Ultra does not offer any gain (-0.05dB at max volume), and the ADC will clip with a 2.15Vrms input signal. As a phono preamp, the Ultra was set to maximum gain (51.6dB at max volume), which yielded an output of roughly 1.9Vrms for a 5mVrms input level.

The Ultra also offers seven different reconstruction filters for the DAC, they are:

1. Brick Wall Filter (labelled Filter 1 in this report)
2. Corrected Minimum Phase Fast Roll-Off (labelled Filter 2 in this report)
3. Apodizing Fast Roll-Off (labelled Filter 3 in this report)
4. Minimum Phase Slow Roll-Off (labelled Filter 4 in this report)
5. Minimum Phase Fast Roll-Off (labelled Filter 5 in this report)
6. Linear Phase Slow Roll-Off (labelled Filter 6 in this report)
7. Linear Phase Fast Roll-Off (labelled Filter 7 in this report and the default filter unless otherwise stated)

The Ultra’s volume control operates in the digital domain, evidenced by our left-right channel tracking table below, which shows identical deviations (dB) at each sample volume level. Overall volume ranges from -59.6dB to -0.05dB (analog line-level in/out). Volume steps range from 1.5dB (lower volume levels) to 0.3dB (levels 50 through 100).

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

Published specifications vs. our primary measurements

The table below summarizes the measurements published by WiiM for the Ultra compared directly against our own. The published specifications are sourced from WiiM’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 was set at its maximum (DC to 1MHz), unless otherwise stated, assume a 1kHz sinewave at 0dBFS (24/96) at the input, 2Vrms at the output into 100k ohms, and a measurement input bandwidth of 10Hz to 22.4kHz, and the worst-case measured result between the left and right channels.

Our primary measurements revealed the following using the balanced line-level analog input and digital coaxial input (unless specified, assume a 1kHz sinewave at 2Vrms or 0dBFS, 2Vrms output, 100kohm loading, 10Hz to 22.4kHz bandwidth):

Our primary measurements revealed the following using the MM phono-level input (unless specified, assume a 1kHz sinewave at 5mVrms, 1.9Vrms output, 100kohm loading, 10Hz to 22.4kHz bandwidth):

Our primary measurements revealed the following using the analog input at the headphone output (unless specified, assume a 1kHz sinewave, 0dBFS (24/96) input, 2Vrms output, 300 ohms loading, 10Hz to 22.4kHz bandwidth):

Frequency response (line-level input)

In our measured frequency-response (relative to 1kHz) plot above, the Ultra is near flat within the audioband (-0.2dB at 20Hz and 20kHz). At the extremes, the Ultra is -2.5dB at 5Hz, and brickwall filtering just below 96kHz (due to the 24/192 ADC sampling). In the graph above and most of the graphs below, only a single trace may be visible. This is because the left channel (blue or purple trace) is performing identically to the right channel (red or green trace), and so they perfectly overlap, indicating that the two channels are ideally matched.

Frequency response vs. ADC sample rate (line-level analog input)

In our measured frequency-response (relative to 1kHz) plot above, the blue/red traces are sampled at 44.1kHz, purple/green at 48kHz, pink/orange at 96kHz, and finally the two green traces are sampled using the default 192kHz (seen in the first graph above but limited to 80kHz). At high frequencies, there is nothing unusual, as we see brickwall type filtering around half the sample rate for all data. At low frequencies, strangely, we see different low-frequency extensions. The best results are with the 44.1kHz and 48kHz sample rates (-0.5dB at 5Hz), then 96kHz (-1.0dB at 5Hz), and finally the worst performer, the 192kHz sample rate (-2.5dB at 5Hz).

Frequency response with bass management (line-level analog input)

In our measured frequency response (relative to 1kHz for main output, relative to 20Hz for the subout) plot above, the blue trace is the subout, whereas the red trace is the main left output. The crossover frequency was chosen at 80Hz. We see 12dB/octave slopes, and a crossover point at the correct selected frequency.

Phase response (line-level analog input)

Above is the phase-response plot from 20Hz to 20kHz for the line-level input. The Ultra does not invert polarity, but because the signal is digitized, there is a significant amount of phase shift (350000 degrees of shift at 20kHz) due to the delay involved in performing the sampling.

Phase response (line-level analog input, excess only)

Above is the phase response plot from 20Hz to 20kHz for the line-level input, only showing excess phase shift (beyond the timing delays due to ADC sampling). We see just shy of +20 degrees at 20Hz, and no phase shift at 20kHz.

Frequency response vs. input type

The chart above shows the Ultra’s frequency response (relative to 1kHz) as a function of input type. The green trace is the same (but limited to 80kHz) analog input data from the previous graph (sampled at 24/192). The blue trace is for a 16-bit/44.1kHz dithered digital input signal from 5Hz to 22kHz using the coaxial input, the purple trace is for a 24/96 dithered digital input signal from 5Hz to 48kHz, and finally pink is 24/192 from 5Hz to 96kHz. The behavior at low frequencies is the same for all the digital sample rates, perfectly flat down to 5Hz, as opposed to the -2.5dB at 5Hz response from the sampled analog input. The behavior at high frequencies for all three digital sample rates is as expected, offering brickwall type filtering around 22, 48, and 96kHz (half the respective sample rate). Also as expected, the analog input (sampled at 24/192), follows the exact same high frequency response as the 24/192 input data.

Frequency response vs. digital filter type (16/44.1)

The chart above shows the Ultra’s frequency response (relative to 1kHz) for three filter types with a 16/44.1 input signal. The blue trace is the Brick Wall filter, red is Corrected Minimum Phase Fast Roll-Off, and green is Apodizing Fast Roll-Off. The Brick Wall filter has a -3dB point at 19.9kHz and is near 0dB at 19kHz. The Corrected Minimum Phase Fast Roll-Off filter has a -3dB point at 19kHz and is at -0.5dB at 18kHz. The Apodizing Fast Roll-Off filter has a -3dB point just past 20kHz and is near 0dB at 20kHz, though shows a ripple response (+/-0.2dB) from 10kHz to 20kHz.

The chart above shows the Ultra’s frequency response (relative to 1kHz) for four filter types with a 16/44.1 input signal. The pink trace is the Minimum Phase Slow Roll-Off filter, blue is Minimum Phase Fast Roll-Off, purple is Linear Phase Slow Roll-Off, and orange is Linear Phase Fast Roll-Off (the default filter). The Minimum Phase Slow Roll-Off filter has a -3dB point at 19.1kHz and is at -0.5dB at 17kHz. The Minimum Phase Fast Roll-Off filter has a -3dB point at 21.1kHz and is near 0dB at 19kHz. The Linear Phase Slow Roll-Off filter has a -3dB point just past 19.7kHz and is at -0.5dB at 17.3kHz. The Linear Phase Fast Roll-Off filter has a -3dB point at 21.1kHz and is near 0dB at 19kHz, essentially identical to the Minimum Phase Fast Roll-Off filter.

Phase response vs filter type (16/44.1 input, excess only)

Above is the phase-response plot (excess) from 20Hz to 20kHz for the first three filter type with a 16/44.1 input signal. The blue trace is the Brick Wall filter, red is Corrected Minimum Phase Fast Roll-Off, and green is Apodizing Fast Roll-Off. The Brick Wall and Apodizing Fast Roll-Off filters have identical behaviour at -45 degrees at 20kHz. The Corrected Minimum Phase Fast Roll-Off filter shows a bump from 10-15kHz (-20 to -10 degrees), then a steep phase roll-off (-180 degrees at 20kHz).

Above is the phase response plot (excess) from 20Hz to 20kHz for the next four filter types with a 16/44.1 input signal. The pink trace is the Minimum Phase Slow Roll-Off filter, blue is Minimum Phase Fast Roll-Off, purple is Linear Phase Slow Roll-Off, and orange is Linear Phase Fast Roll-Off (the default filter). The Minimum Phase Slow Roll-Off and Minimum Phase Fast Roll-Off filters have very close behaviours, right around -180 degrees at 20kHz. The Linear Phase Slow Roll-Off and Linear Phase Fast Roll-Off filters are identical at -45 degrees at 20kHz.

Frequency response (MM input)

The chart above shows the frequency response (relative to 1 kHz) for the MM phono input. What is shown is the deviation from the RIAA curve, where the input signal sweep is EQd with an inverted RIAA curve supplied by Audio Precision (i.e., zero deviation would yield a flat line at 0dB). The result shows relatively small maximum deviations within the audioband: about +0.2dB at 50kHz and +0.5dB at 20kHz. At low frequencies we see -1dB at 20Hz, and the -3dB point is at 15Hz.

Phase response (MM phono input)

Above is the phase response plot from 20Hz to 20kHz for the MM phono input. The Ultra does not invert polarity, but because the signal is digitized, there is a significant amount of phase shift (400000 degrees of shift at 20kHz) due to the delay involved in performing the sampling.

Phase response (MM phono input, excess only)

Above is the phase response plot from 20Hz to 20kHz for the phono MM input only showing excess phase shift (beyond the timing delays due to ADC sampling). For the phono input, since the RIAA equalization curve must be implemented, which ranges from +19.9dB (20Hz) to -32.6dB (90kHz), phase shift at the output is inevitable. We see typical phase shifts associated with the implementation of RIAA filters: +100 degrees at 20Hz, 0 degrees at 200Hz and 5-6kHz, +20 degrees at 1kHz.

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 outputs of the Ultra. For this test, 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 would be a straight flat line at 0dB. At -120dBFS, the 16/44.1 data overshot by only 1-2dB, while the 24/96 remained perfect. To verify how well the 24/96 data would perform down to -140dBFS, we extended the sweep in the chart below.

Here we can see that the 24/96 data only missed the mark by +1/-1dB (left/right) at -140dBFS. This is an exceptional digital-linearity test result.

Impulse response (24/48 data)

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, measured at the outputs of the Ultra. The blue trace is the Brick Wall filter, red is Corrected Minimum Phase Fast Roll-Off, and green is Apodizing Fast Roll-Off. Both the Brick Wall and Apodizing Fast Roll-Off filters show typical symmetrical sinc functions with pre/post ringing. The Corrected Minimum Phase Fast Roll-Off filter minimizes pre-ringing but shows extensive post-ringing.

This chart shows the remaining four filters. The pink trace is the Minimum Phase Slow Roll-Off filter, blue is Minimum Phase Fast Roll-Off, purple is Linear Phase Slow Roll-Off, and orange is Linear Phase Fast Roll-Off (the default filter). The Minimum Phase Slow Roll-Off filter shows no pre-ringing and minimum post-ringing. The Minimum Phase Fast Roll-Off filter shows no pre-ringing but significant post-ringing. The Linear Phase Slow Roll-Off filter is closest to an idealized impulse response: symmetrical with very little pre- and post-ringing. The Linear Phase Fast Roll-Off filter is another example of a typical symmetrical Sinc function with pre/post ringing.

J-Test (optical input)

The chart above shows the results of the J-Test test for the optical digital input measured at the line-level output of the Ultra. TheJ-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 sinewave (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 and below.  The test file can also be used in conjunction with artificially injected sinewave jitter by the Audio Precision, to show how well the DAC rejects jitter.

The optical SPDIF input of the Ultra shows a strong J-Test result, with spurious peaks at the -130dBrA and below level.

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

The chart above shows the results of the J-Test test for the optical digital input measured at the line-level output of the Ultra, but with an additional 100ns of 2kHz sinewave jitter injected by the APx555. The results are essentially the same as the J-Test result without additional jitter, with no visible peaks at the 10kHz and 14kHz positions. Further evidence of the Ultra’s strong jitter immunity.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (optical input, Filter 1)

The chart above shows a fast Fourier transform (FFT) of the Ultra’s outputs with white noise at -4 dBFS (blue/red) and a 19.1 kHz sinewave at 0dBFS fed to the optical digital input, sampled at 16/44.1, using the Brick Wall filter. The steep roll-off around 20kHz in the white-noise spectrum shows the use of a brickwall-type reconstruction filter. There are no obvious aliased images within the audioband. The primary aliasing signal at 25kHz is highly suppressed at -100dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (optical input, Filter 2)

The chart above shows a fast Fourier transform (FFT) of the Ultra’s outputs with white noise at -4 dBFS (blue/red) and a 19.1 kHz sinewave at 0dBFS fed to the optical digital input, sampled at 16/44.1, using the Corrected Minimum Phase Fast Roll-Off filter. The steep roll-off around 20kHz in the white-noise spectrum shows the use of a brickwall-type reconstruction filter. There are no obvious aliased images within the audioband. The primary aliasing signal at 25kHz is highly suppressed at -110dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (optical input, Filter 3)

The chart above shows a fast Fourier transform (FFT) of the Ultra’s outputs with white noise at -4 dBFS (blue/red) and a 19.1 kHz sinewave at 0dBFS fed to the optical digital input, sampled at 16/44.1, using the Apodizing Fast Roll-Off filter. The steep roll-off around 20kHz in the white-noise spectrum shows the use of a brickwall-type reconstruction filter. There are no obvious aliased images within the audioband. The primary aliasing signal at 25kHz is highly suppressed at -100dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (optical input, Filter 4)

The chart above shows a fast Fourier transform (FFT) of the Ultra’s outputs with white noise at -4 dBFS (blue/red) and a 19.1 kHz sinewave at 0dBFS fed to the optical digital input, sampled at 16/44.1, using the Minimum Phase Slow Roll-Off  filter. The slow roll-off around 20kHz in the white-noise spectrum matches the description of the reconstruction filter. There are no obvious aliased images within the audioband. The primary aliasing signal at 25kHz is barely suppressed at -30dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (optical input, Filter 5)

The chart above shows a fast Fourier transform (FFT) of the Ultra’s outputs with white noise at -4 dBFS (blue/red) and a 19.1 kHz sinewave at 0dBFS fed to the optical digital input, sampled at 16/44.1, using the Minimum Phase Fast Roll-Off filter. The steep roll-off around 20kHz in the white-noise spectrum shows the use of a brickwall-type reconstruction filter. There are no obvious aliased images within the audioband. The primary aliasing signal at 25kHz is highly suppressed at -110dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (optical input, Filter 6)

The chart above shows a fast Fourier transform (FFT) of the Ultra’s outputs with white noise at -4 dBFS (blue/red) and a 19.1 kHz sinewave at 0dBFS fed to the optical digital input, sampled at 16/44.1, using the Linear Phase Slow Roll-Off filter. The slow roll-off around 20kHz in the white-noise spectrum matches the description of the reconstruction filter. There are no obvious aliased images within the audioband. The primary aliasing signal at 25kHz is barely suppressed at -30dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (optical input, Filter 7)

The chart above shows a fast Fourier transform (FFT) of the Ultra’s outputs with white noise at -4 dBFS (blue/red) and a 19.1 kHz sinewave at 0dBFS fed to the optical digital input, sampled at 16/44.1, using the Linear Phase Fast Roll-Off filter. The steep roll-off around 20kHz in the white-noise spectrum shows the use of a brick-wall type reconstruction filter. There are no obvious aliased images within the audioband. The primary aliasing signal at 25kHz is highly suppressed at -110dBrA.

THD ratio (unweighted) vs. frequency vs. load (analog)

The chart above shows THD ratios at the line-level output into 100k ohms (blue/red) and 600 ohms (purple/green) as a function of frequency for the analog inputs. The 100k-ohm THD data are identical to the 600-ohm data, an indication that the Ultra will have no issues with sub-1k ohm amplifier input impedances. THD data ranged from 0.05% at 20Hz, down to 0.003% from 200Hz to 4kHz, then up to 0.006% at 20kHz.

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

The chart above shows THD ratios at the line-level output into 100k ohms for a 16/44.1 (blue/red) dithered 1kHz signal at the optical input and a 24/96 (purple/green) signal, as a function of frequency. The 16/44.1 THD ratios were higher (around 0.0002%) than the 24/96 THD ratios due to the increased noise floor from the lower bit-depth (the analyzer cannot assign a THD ratio for harmonic peaks it cannot see above the noise floor). 24/96 THD ratios were lower (by about 5dB) for the right channel from 30Hz to 1kHz, ranging from 0.00003% to 0.00005%. At 20kHz, all THD ratios measured 0.0002%.

THD ratio (unweighted) vs. frequency (MM phono input)

The graph above shows THD ratio as a function of frequency plot for the phono input. The input sweep is EQ’d with an inverted RIAA curve. The THD values ranged from around 0.3% (20Hz) down to around 0.003% (200Hz to 300Hz), then up to 0.06% at 20kHz.

THD ratio (unweighted) vs. output (analog)

The chart above shows THD ratios measured at the outputs of the Ultra as a function of output voltage for the analog line-level input, with the volume control set to maximum. THD values start at 0.2% at 1mVrms, down to a low of 0.001% at 0.2-0.5Vrms, then a rise to 0.003% nearing 2Vrms, then the 1% THD mark, due to ADC clipping, at 2.15Vrms.

THD+N ratio (unweighted) vs. output (analog)

The chart above shows THD+N ratios measured at the outputs of the Ultra as a function of output voltage for the analog line-level input, with the volume control at maximum. THD+N values start at 3% at 1mVrms, down to a low of 0.003% at 1-1.9Vrms, then a steep rise to the 1% THD mark, due to ADC clipping, at 2.15Vrms.

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

The chart above shows THD ratios measured at the outputs of the Ultra as a function of output voltage for the digital optical S/PDIF input, swept from -90dBFS to 0dBFS, with the volume control at maximum. Blue/red traces are for 16/44.1 data, and purple/green for 24/96. For the 16/44.1 data, THD values start at 2%, and predictably, reach their low at the maximum output voltage of just over 2Vrms, at 0.0002%. For the 24/96 data, THD ratios ranged from 0.2% down to 0.00005% at the maximum output voltage.

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

The chart above shows THD+N ratios measured at the outputs of the Ultra as a function of output voltage for the digital optical S/PDIF input, swept from -90dBFS to 0dBFS, with the volume control at maximum. Blue/red traces are for 16/44.1 data, and purple/green for 24/96. For the 16/44.1 data, THD+N values start at 20% and reach their low at the maximum output voltage of just over 2Vrms, at 0.002%. For the 24/96 data, THD+N ratios ranged from 1.5% down to 0.0002% at the maximum output voltage.

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 output for 16/44.1 (blue/red) input data and 24/96 input data (purple/green), from -60dBFS to 0dBFS. 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 16/44.1 data yields IMD ratios from 2% down to 0.003% at 0dBFS. The 24/96 data yields IMD ratios from 0.15% down to 0.0005% at 0dBFS.

FFT spectrum – 1kHz (analog line-level input)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the outputs for the analog line-level input. We see that the signal’s second (2kHz), third (3kHz) and fifth (5kHz) harmonics dominate, at -90/-100/-115dBrA, or 0.003/0.001/0.0002%. Subsequent signal harmonics can be seen at and below the -120dBrA, or 0.0001%, level. Below 1kHz, we see two small peaks from power-supply noise artifacts at 60Hz and 180Hz, around the -130dBrA, or 0.00003%, level.

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 outputs for the optical digital input, sampled at 16/44.1. We see the third (3kHz) signal harmonic just above the noise floor at -130dBrA, or 0.00003%. The noise floor is much higher due to the 16-bit depth limitation.

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 outputs for the optical digital input, sampled at 24/96. We see the second (2kHz), third (3kHz) and fifth (5kHz) signal harmonics at the very low levels of -140/-130/-140dBrA, or 0.00003% to 0.00001%. Power-supply-related noise peaks can be seen at and below the very low level of -135dBrA, or 0.00002% at 60/180/300Hz.

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

Shown above is the FFT for a 1kHz -90dBFS dithered 16/44.1 input sinewave stimulus at the optical digital input, measured at the analog outputs. We see the 1kHz primary signal peak, at the correct amplitude, and no signal- or noise-related harmonic peaks above the -135dBrA noise floor.

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

Shown above is the FFT for a 1kHz -90dBFS dithered 24/96 input sinewave stimulus at the optical digital input, measured at the analog outputs. We see the 1kHz primary signal peak, at the correct amplitude, and signal- and noise-related peaks at around and below -140dBrA, or 0.00001%.

FFT spectrum – 1kHz (digital input, 24/96 data at -0dBFS by putting volume control at minimum)

Shown above is the FFT for a 1kHz -0dBFS dithered 24/96 input sinewave stimulus at the optical digital input, measured at the analog outputs, with the volume set to minimum, which yields a signal at -60dBrA. This FFT can be compared to the FFT below, which shows . . .

FFT spectrum – 1kHz (digital input, 24/96 data at -60dBFS by reducing analyzer level but with volume control at maximum)

. . . a 1kHz -60dBFS dithered 24/96 input sinewave stimulus but wth the volume set to maximum. We find that both FFTs are essentially the same, indicating that the digital volume control, even at the lowest levels, is transparent.

FFT spectrum – 1kHz (MM phono input)

Shown above is the FFT for a 1kHz input sinewave stimulus, measured at the analog outputs for the MM phono input. The dominant signal related harmonics can be seen at 2/3/4/5/6/7/10kHz, at -80dBrA to -100dBrA, or 0.01% to 0.001%. The main noise-related peak can be seen around 80Hz at -70dBrA, or 0.03%.

FFT spectrum – 50Hz (line-level input)

Shown above is the FFT for a 50Hz input sinewave stimulus measured at the outputs for the line-level input. The X axis is zoomed in from 40 Hz to 1kHz, so that peaks from noise artifacts can be directly compared against peaks from the harmonics of the signal. The most predominant (non-signal) peaks are that of the signal’s second (100Hz) harmonic at -80dBrA, or 0.01%, and the third signal harmonic (150Hz) at -90dBrA, or 0.003%. A small power-supply-related noise peak can be seen at 180Hz at -130dBrA, or 0.00003%.

FFT spectrum – 50Hz (MM phono input)

Shown above is the FFT for a 50Hz input sinewave stimulus measured at the outputs for the MM phono input. The most predominant (non-signal) peaks are that of the signal’s second (100Hz) and third (150Hz) harmonics, as well as two noise-related peaks clustered just past 80Hz. All these peaks are around the -70dBrA level, or 0.03%.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, line-level input)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the outputs for the analog line-level input. The input RMS values are set at -6.02dBrA so that, if summed for a mean frequency of 18.5kHz, it would yield 10W (0dBrA) into 8 ohms 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 the same level.

Intermodulation distortion FFT (line-level input, APx 32 tone)

Shown above is the FFT of the outputs of the Ultra with the APx 32-tone signal applied to the analog input. The combined amplitude of the 32 tones is the 0dBrA reference, and corresponds to 2Vrms. The intermodulation products—i.e., the “grass” between the test tones—are distortion products from the preamplifier and are around the -100dBrA, or 0.001%, level from 20Hz to 100Hz, and below the -120dBrA, or 0.0001%, level from 300Hz to 20kHz.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, coaxial digital input, 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 outputs for the digital optical input at 16/44.1. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is difficult to distinguish above the -135dBrA noise floor, while the third-order modulation products, at 17kHz and 20kHz, are at -125dBrA, or 0.00006%. We also see the main aliased peaks at 25.1kHz and 26.1kHz at around -120dBrA.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, coaxial digital input, 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 outputs for the digital coaxial input at 24/96. 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 at -125dBrA, or 0.00006%.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, MM phono input)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the outputs for the phono MM input. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -95dBrA or 0.002%, while the third-order modulation products, at 17kHz and 20kHz, are at -60dBrA, or 0.1%.

Square-wave response (10kHz)

Above is the 10kHz squarewave response using the analog line-level input, at roughly 2Vrms. Due to limitations inherent to the Audio Precision APx555 B Series analyzer, this graph should not be used to infer or extrapolate the Ultra’s slew-rate performance. Rather, it should be seen as a qualitative representation of its limited bandwidth to the 24/912 digital sampling. An ideal squarewave can be represented as the sum of a sinewave and an infinite series of its odd-order harmonics (e.g., 10kHz + 30kHz + 50kHz + 70kHz . . .). A limited bandwidth will show only the sum of the lower-order harmonics, which may result in noticeable undershoot and/or overshoot, and softening of the edges. The Ultra’s reproduction of the 10kHz square wave is relatively clean for a digitized input, but due to the limited bandwidth, rippling can be seen in the plateaus of the squarewave.

Diego Estan
Electronics Measurement Specialist

Details

Parent Category: Products

Category: Preamplifier Measurements

Created: 01 November 2024

Link: reviewed by Roger Kanno on SoundStage! Simplifi on November 1, 2024

General Information

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

The Eversolo Audio DMP-A8 was conditioned for 30 minutes at 2Vrms in/out into 200k ohms before any measurements were taken.

The DMP-A8 offers a multitude of inputs, both digital and analog (balanced and unbalanced), and line-level analog outputs over balanced XLR and unbalanced over RCA. For the purposes of these measurements, unless otherwise stated, the following inputs were evaluated: digital coaxial S/PDIF (RCA) and analog balanced (XLR). Comparisons were made between unbalanced (RCA) and balanced (XLR) line inputs and outputs, and no appreciable differences were seen in terms of gain and THD+N (FFTs for different configurations can be seen in this report).

Most measurements were made with a 2Vrms line-level and 0dBFS digital input with the volume set to achieve 2Vrms at the output. The signal-to-noise ratio (SNR) measurements were made with the same input signal values, and, for comparison, on the line-level input, a SNR measurement was also made with the volume at maximum, but with a lower input voltage to achieve the same 2Vrms output.

The DMP-8 offers a wide range of DSP functions that can only be applied to the digital inputs. All DSP functions were turned off for these measurements. There are also six different DAC reconstruction filters that can be selected (unless otherwise noted, the Sharp Roll-off filter was used for these measurements):

1. Sharp Roll-off
2. Slow Roll-off
3. Short Delay Sharp Roll-off
4. Short Delay Slow Roll-off
5. Super Slow Roll-off (emulation of NOS)
6. Low Dispersion Short Delay

The DMP-A8 also offers a range of volume step settings: 0.5/1/2/3dB. Unless otherwise stated, the 0.5dB setting was used for all measurements. Based on the accuracy and random results of the left/right volume channel matching (see table below), the DMP-A8 volume control is likely digitally controlled in the analog domain. The overall range is from -89.3dB to 9.9dB for the line-level inputs.

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

Published specifications vs. our primary measurements

The table below summarizes the measurements published by Eversolo for the DMP-A8 compared directly against our own. The published specifications are sourced from Eversolo’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 was set at its maximum (DC to 1MHz), unless otherwise stated, assume a 1kHz sinewave at 2Vrms or 0dBFS at the input, 2Vrms at the output into 200k ohms, and a measurement input bandwidth of 10Hz to 22.4kHz, and the worst-case measured result between the left and right channels.

Our primary measurements revealed the following using the balanced line-level analog input and digital coaxial input (unless specified, assume a 1kHz sinewave at 2Vrms or 0dBFS when a sample rate is specified, 2Vrms output, 200k-ohm loading, 10Hz to 22.4kHz bandwidth):

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

Frequency response (line-level input)

In our measured frequency response (relative to 1kHz) plot above, the DMP-A8 is near perfectly flat within the audioband (0dB at 20Hz, -0.05dB at 20kHz). At the extremes, the DMP-A8 is 0dB at 5Hz, -0.6dB at 100kHz, and -5dB just before 200kHz. The DMP-A8 appears to be DC-coupled, as there is no attenuation at low frequencies, even at 5Hz. In the graph above and most of the graphs below, only a single trace may be visible. This is because the left channel (blue or purple trace) is performing identically to the right channel (red or green trace), and so they perfectly overlap, indicating that the two channels are ideally matched.

Phase response (line-level input)

Above is the phase response plot from 20Hz to 20kHz for the balanced line-level input. The DMP-A8 does not invert polarity and exhibits, at worst, less than -10 degrees (at 20kHz) of phase shift within the audioband.

Frequency response vs. input type (left channel only, Sharp Roll-off filter)

The chart above shows the DMP-A8’s frequency response (relative to 1kHz) as a function of input type. The green trace is the same (but limited to 80kHz) analog input data from the previous graph. The blue trace is for a 16-bit/44.1kHz dithered digital input signal from 5Hz to 22kHz using the coaxial input, the purple trace is for a 24/96 dithered digital input signal from 5Hz to 48kHz, and finally pink is 24/192 from 5Hz to 96kHz. The behavior at low frequencies is the same for all the digital sample rates, as well as the analog input—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 rates). The 44.1kHz sampled input signal exhibits typical “brick-wall”-type behavior, with a -3dB point at 20.9kHz. The -3dB point for the 96kHz sampled data is at 45.5kHz, and 68.9kHz for the 192kHz sampled data.

Frequency response vs. filter type (16/44.1, Sharp Roll-off, Slow Roll-off, Super Slow Roll-off)

The plots above show frequency-response for a -3dBFS input signal sampled at 16/44.1 for the Sharp Roll-off filter (blue), the Slow Roll-off filter (red), and the Super Slow Roll-off filter (green) 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 Sharp Roll-Off  filter provides a “brick-wall”-type response, and the Slow Roll-Off filter shows gentle attenuation (-1dB at 12.8kHz), and the Super Slow Roll-Off filter shows the gentlest attenuation (-1dB at 11kHz). The -3dB points for all three filters are: 20.9kHz (blue), 15.8kHz (red), and 18.7kHz (green).

Frequency response vs. filter type (16/44.1, Short Delay Sharp Roll-off, Short Delay Slow Roll-off, Low Dispersion Short Delay)

The plots above show frequency-response for a -3dBFS input signal sampled at 16/44.1 for the Short Delay Sharp Roll-off filter (blue), the Short Delay Slow Roll-off filter (red), and the Low Dispersion Short Delay filter (green) 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 both the Short Delay Sharp Roll-Off and Low Dispersion Short Delay filters provide a “brick-wall”-type response very similar to the default Sharp Roll-off filter, while the Short Delay Slow Roll-off filter shows gentle attenuation (-1dB at 12.8kHz). The -3dB points for all three filters are: 20.9kHz (blue), 15.8kHz (red), and 20.8kHz (green).

Phase response vs. filter type (16/44.1, all filters)

Above are the phase response plots from 20Hz to 20kHz for a -3dBFS input signal sampled at 16/44.1 for the Sharp Roll-off filter (blue), the Slow Roll-off filter (red), the Short Delay Sharp Roll-off filter (green), the Short Delay Slow Roll-off filter (pink), the Super Slow Roll-off filter (purple), and the Low Dispersion Short Delay filter (orange) into a 200k ohm-load for the left channel only. We find that only the two short-delay filters show any phase shift within the audioband: -180 and -80 degrees at 20kHz for the Sharp and Slow Roll-off short-delay filters, respectively.

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 outputs of the DMP-A8. The digital input was swept with a dithered 1kHz input signal from -120dBFS to 0dBFS, and the output was analyzed by the APx555. The ideal response would be a straight flat line at 0dB. At -120dBFS, the 16/44.1 data overshot by only 1dB, while the 24/96 remained perfect. To verify how well the 24/96 data would perform down to -140dBFS, we extended the sweep . . .

. . . in the chart above. Here we can see that the 24/96 data only undershot the mark by -1dB (left) at -140dBFS. This is an exceptional, and essentially perfect, digital-linearity test result.

Impulse response (24/48 data)

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, measured at the balanced outputs of the DMP-A8. The blue plot is for the Sharp Roll-off filter, red for Slow Roll-off, and green for Short Delay Sharp Roll-off. The Sharp Roll-off filter shows a typical symmetrical sinc function response. The Slow Roll-off filter also shows a symmetrical response but with much less pre- and post-ringing. The Short Delay Sharp Roll-off filter shows no pre-ringing but extensive post-ringing.

Impulse response (24/48 data)

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, measured at the balanced outputs of the DMP-A8. The blue plot is for the Short Delay Slow Roll-off filter, red for Super Slow Roll-off, and green for Low Dispersion Short Delay. The Short Delay Slow Roll-off filter shows no pre-ringing and minimized pots-ringing. The Super Slow Roll-Off filter shows a near ideal response, close to what would be expected from an NOS DAC—a sharp impulse with no pre- and post-ringing. The Low Dispersion Short Delay filter shows minimized pre-ringing with more post-ringing.

J-Test (coaxial input)

The chart above shows the results of the J-Test test for the coaxial digital input measured at the line-level output of the DMP-A8. J-Test was developed by Julian Dunn in 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 sinewave (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 and below.  The test file can also be used in conjunction with artificially injected sinewave jitter by the Audio Precision, to show how well the DAC rejects jitter.

The coaxial S/PDIF input of the DMP-A8 shows a near-perfect J-Test result, with only a few small peaks in the audioband at a extraordinarily low -155dBrA.

J-Test (optical input)

The chart above shows the results of the J-Test test for the optical digital input measured at the balanced outputs of the DMP-A8. The results here are very similar to the result from the coaxial input above.

J-Test (coaxial, 2kHz sinewave jitter at 10ns)

The chart above shows the results of the TJ-Test test for the coaxial digital input measured at the line level output of the DMP-A8, with an additional 10ns of 2kHz sinewave jitter injected by the APx555. The results are essentially the same as the J-Test result without additional jitter. The same was true for the optical input.

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

The chart above shows the results of the J-Test test for the coaxial digital input measured at the line-level output of the DMP-A8, with an additional 100ns of 2kHz sinewave jitter injected by the APx555. The results are essentially the same as the J-Test result without additional jitter. The same was true for the optical input. This is an exceptional result.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (coaxial input, Sharp Roll-off filter)

The chart above shows a fast Fourier transform (FFT) of the DMP-A8’s balanced outputs with white noise at -4 dBFS (blue/red), and a 19.1 kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1, using the Sharp Roll-off filter. The steep roll-off around 20kHz in the white-noise spectrum shows that this filter is of the brick-wall-type variety. There are no obvious aliased images within the audioband. The primary aliasing signal at 25kHz is highly suppressed at -120dBrA, while the second and third distortion harmonics (38.2, 57.3kHz) of the 19.1kHz signal are at similar levels.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (coaxial input, Slow Roll-off filter)

The chart above shows a fast Fourier transform (FFT) of the DMP-A8’s balanced outputs with white noise at -4 dBFS (blue/red) and a 19.1 kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1, using the Slow Roll-off filter. The slow roll-off around 20kHz matches Eversolo’s description for this filter. There is one aliased image within the audioband, around 13kHz at -130dBrA. The primary aliasing signal at 25kHz is barely suppressed at -20dBrA, while the second and third distortion harmonics (38.2, 57.3kHz) of the 19.1kHz signal are much lower at -120dBrA and below.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (coaxial input, Short Delay Sharp Roll-off filter)

The chart above shows a fast Fourier transform (FFT) of the DMP-A8’s balanced outputs with white noise at -4 dBFS (blue/red, and a 19.1 kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1, using the Short Delay Sharp Roll-off filter. The steep roll-off around 20kHz in the white-noise spectrum shows that this filter is of the brick-wall-type variety. There are no obvious aliased images within the audioband. The primary aliasing signal at 25kHz is completely suppressed, while the second and third distortion harmonics (38.2, 57.3kHz) of the 19.1kHz signal are at and below -110dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (coaxial input, Short Delay Slow Roll-off filter)

The chart above shows a fast Fourier transform (FFT) of the DMP-A8’s balanced outputs with white noise at -4 dBFS (blue/red) and a 19.1 kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1, using the Short Delay Slow Roll-off filter. The slow roll-off around 20kHz matches Eversolo’s description for this filter. There is one aliased image within the audioband, around 13kHz at -130dBrA. The primary aliasing signal at 25kHz is barely suppressed at -20dBrA, while the second and third distortion harmonics (38.2, 57.3kHz) of the 19.1kHz signal are much lower at -120dBrA and below.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (coaxial input, Super Slow Roll-off filter)

The chart above shows a fast Fourier transform (FFT) of the DMP-A8’s balanced outputs with white noise at -4 dBFS (blue/red) and a 19.1 kHz sinewave at -1dBFS fed to the coaxial digital input, sampled at 16/44.1, using the Super Slow Roll-Off filter. The very gentle roll-off around 20kHz and repeating nature in the noise spectrum at intervals of the 44.1kHz sample rate are indicative of a filter emulating a NOS DAC. There are obvious aliased images within the audioband at 13.2/7.3/1.4kHz at -80/-100/-115dBrA. The primary aliasing signal at 25kHz is barely suppressed at -10dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (coaxial input, Low Dispersion Short Delay filter)

The chart above shows a fast Fourier transform (FFT) of the DMP-A8’s balanced outputs with white noise at -4 dBFS (blue/red) and a 19.1 kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1, using the Low Dispersion Short Delay filter. The medium-steep roll-off around 20kHz in the white noise spectrum shows that this filter is of the brick-wall type variety. There are no obvious aliased images within the audioband. The primary aliasing signal at 25kHz is at -50dBrA, while the second and third distortion harmonics (38.2, 57.3kHz) of the 19.1kHz signal are at and below -110dBrA.

THD ratio (unweighted) vs. frequency vs. load (analog)

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 the analog balanced inputs. The 200k-ohm THD data are lower than the 600-ohm data by roughly 5dB. The 200k-ohm data range from 0.00003% at 20Hz, down to an astonishingly low 0.00002% at 40Hz to 1kHz, then a rise to 0.0002% at 20kHz. It’s important to note that these THD values are less than twice as high as the AP’s signal generator, thereby pushing the limits of the analyzer capabilities.

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 for a 16/44.1 (blue/red) dithered 1kHz signal at the coaxial input and a 24/96 (purple/green) signal, as a function of frequency. The 16/44.1 THD ratios were higher (0.0003% to 0.0002%) than the 24/96 THD ratios (0.00006% to 0.0001%) from 20Hz to 6kHz, due to the increased noise floor from the lower bit-depth (the analyzer cannot assign a THD ratio for harmonic peaks it cannot see above the noise floor). At 20kHz, all THD ratios measured 0.0005%.

THD ratio (unweighted) vs. output (analog)

The chart above shows THD ratios measured at the balanced outputs of the DMP-A8 as a function of output voltage for the balanced line level-input, with the volume control at maximum. THD values start at 0.02% at 1mVrms, down to a low of 0.00003/0.00005% (left/right) at 1-3Vrms, then a steep rise past 12Vrms to the 1% THD mark at 16.9Vrms.

THD+N ratio (unweighted) vs. output (analog)

The chart above shows THD+N ratios measured at the balanced outputs of the DMP-A8 as a function of output voltage for the balanced line level-input, with the volume control at maximum. THD+N values start at 0.2% at 1mVrms, down to a low of 0.0001% at 2-10Vrms, then a steep rise past 12Vrms to the 1% THD mark at 16.9Vrms.

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

The chart above shows THD ratios measured at the balanced outputs of the DMP-A8 as a function of output voltage for the digital coaxial S/PDIF input, swept from -90dBFS to 0dBFS, with the volume control at maximum. Blue/red traces are for 16/44.1 data, and purple/green for 24/96. For the 16/44.1 data, THD values start at 2%, and predictably, reach their low at the maximum output voltage of about 4.3Vrms, at 0.0002%. For the 24/96 data, THD ratios ranged from 0.05% down to 0.00008% at the maximum output voltage.

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

The chart above shows THD+N ratios measured at the balanced outputs of the DMP-A8 as a function of output voltage for the digital coaxial S/PDIF input, swept from -90dBFS to 0dBFS, with the volume control at maximum. Blue/red traces are for 16/44.1 data, and purple/green for 24/96. For the 16/44.1 data, THD+N values start at 20% and reach their low at the maximum output voltage of about 4.3Vrms, at 0.002%. For the 24/96 data, THD ratios ranged from 0.5% down to 0.00015% at the maximum output voltage.

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) input data and 24/96 input data (purple/green), from -60dBFS to 0dBFS. 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 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 around 0.0005% near 0dBFS.

FFT spectrum – 1kHz (XLR line-level input)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced outputs for the balanced line-level input. Because both the noise and signal related peaks are so vanishingly low for the DMP-A8, we must directly compare against the loopback (AP signal generator’s outputs connected to the analyzer’s inputs using the same XLR cables used with the DUT) FFT spectrum below. For the signal related harmonics, we find roughly the same levels at the second (2kHz) position: -150/-140dBrA (left/right), or 0.000003/0.00001%. We therefore attribute these peaks to the signal generator and not the DMP-A8. At the third (3kHz) position, we find that the signal generator contributes -150dBrA (0.000003%) of the -140dBrA (0.00001%) levels seen above. Subtracting one from the other yields a level of 0.000007% distortion at the 3kHz position for the DMP-A8. The only significant signal-related harmonic peak remaining for both channels is at the fifth (5kHz) harmonic position at -150dBrA, or 0.000003%. This means that the true THD ratio for the DMP-A8 is at the absurdly low 0.00001% (-140dBrA) level. For the noise-related peaks, we can ignore the peaks at the 60 and 120Hz positions, as these are also seen in the loopback FFT below. What we are left with are a few spurious peaks (mostly right channel) at near and below -150dBrA, or 0.000003%. This is an exceptionally quiet device. In fact, the residual rms noise (volume at minimum, A-weighted) we measured from the DMP-A8 (0.55uVrms for RCA output, 0.75uVrms XLR) essentially equals the lowest noise we have ever seen in a device (RME ADI-2 DAC IEM headphone output). These levels are right around the noise levels inherent to the AP’s signal generator (0.66 uVrms, A-weighted). A truly phenomenal result, which translates to a signal-to-noise ratio (A-weighted, relative to 2Vrms) of 128dB!

FFT spectrum – 1kHz (XLR line-level input)

Shown above is the fast Fourier transform (FFT) for a 1kHz 2Vrms input sinewave stimulus, measured by directly connecting the AP signal generator to the analyzer using our balanced XLR interconnects.

FFT spectrum – 1kHz (RCA line-level input)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the unbalanced outputs for the unbalanced line-level input. We see close to the same results as with the balanced input FFT above. The main difference is the higher 2kHz signal harmonic peak at -130dBrA instead of -140dBrA.

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 outputs for the coaxial digital input, sampled at 16/44.1. We see the second (2kHz) and third (3kHz) signal harmonics at roughly -130dBrA, or 0.00003%, and -120dBrA, or 0.0001%, respectively.  The noise floor is much higher due to the 16-bit depth limitation.

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 outputs for the coaxial digital input, sampled at 24/96. We see power-supply-related noise peaks but only near and below the -150dBrA, or 0.000003%, level. The second (2kHz) and third (3kHz) signal harmonics dominate at -130dBrA, or 0.00003%, and -120dBrA, or 0.0001%. Higher signal harmonics can be seen at -140dBrA, or 0.00001%, and below.

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

Shown above is the FFT for a 1kHz -90dBFS dithered 16/44.1 input sinewave stimulus at the coaxial digital input, measured at the balanced outputs. We see the 1kHz primary signal peak, at the correct amplitude, and no signal or noise-related harmonic peaks above the -140dBrA noise floor.

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

Shown above is the FFT for a 1kHz -90dBFS dithered 24/96 input sinewave stimulus at the coaxial digital input, measured at the balanced outputs. We see the 1kHz primary signal peak, at the correct amplitude, and no signal-related harmonic peaks. We see power-supply-related noise peaks but only near and below the -150dBrA, or 0.000003%, level.

FFT spectrum – 50Hz (line-level input)

Shown above is the FFT for a 50Hz input sinewave stimulus measured at the balanced outputs for the balanced line-level input. The X axis is zoomed in from 40 Hz to 1kHz, so that peaks from noise artifacts can be directly compared against peaks from the harmonics of the signal. The most predominant (non-signal) peak is that of the signal’s third (150Hz) harmonic at -135dBrA or 0.00002%. Power-supply-related peaks can be seen but only near and below the -150dBrA, or 0.000003%, level.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, line-level input)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced outputs for the balanced line-level input. The input RMS values are set at -6.02dBrA so that, if summed for a mean frequency of 18.5kHz, would yield 10W (0dBrA) into 8 ohms at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -125dBRA or 0.00006%, while the third-order modulation products, at 17kHz and 20kHz, are at -120dBrA or 0.0001%. This is a very clean IMD result.

Intermodulation distortion FFT (line-level input, APx 32 tone)

Shown above is the FFT of the balanced outputs of the DMP-A8 with the APx 32-tone signal applied to the input. The combined amplitude of the 32 tones is the 0dBrA reference, and corresponds to 2Vrms. The intermodulation products—i.e., the “grass” between the test tones—are distortion products from the preamplifier and are around the extremely low -150dBrA, or 0.000003%, level.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, coaxial digital input, 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 outputs for the digital coaxial input at 16/44.1. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -130dBrA (left channel), or 0.00003%, while the third-order modulation products, at 17kHz and 20kHz, are at -120dBrA, or 0.0001%.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, coaxial digital input, 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 outputs for the digital coaxial input at 24/96. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -130/140dBrA (left/right), or 0.00003/0.00001%, while the third-order modulation products, at 17kHz and 20kHz, are at -120dBrA or 0.0001%.

Square-wave response (10kHz)

Above is the 10kHz squarewave response using the analog line-level input, at roughly 2Vrms. Due to limitations inherent to the Audio Precision APx555 B Series analyzer, this graph should not be used to infer or extrapolate the DMP-A8’s slew-rate performance. Rather, it should be seen as a qualitative representation of its extended bandwidth. An ideal squarewave can be represented as the sum of a sinewave and an infinite series of its odd-order harmonics (e.g., 10kHz + 30kHz + 50kHz + 70kHz . . .). A limited bandwidth will show only the sum of the lower-order harmonics, which may result in noticeable undershoot and/or overshoot, and softening of the edges. The DMP-A8’s reproduction of the 10kHz square wave is very clean, with only extremely mild softening in the edges.

Diego Estan
Electronics Measurement Specialist

Details

Parent Category: Products

Category: Preamplifier Measurements

Created: 01 October 2024

Link: reviewed by Aron Garrecht on SoundStage! Ultra on October 1, 2024

General Information

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

The Simaudio Moon 891 was conditioned for 30 minutes at 2Vrms in/out into 200k ohms before any measurements were taken.

The 891 offers a multitude of inputs, both digital and analog (balanced and unbalanced), as well as line-level analog balanced outputs over XLR and unbalanced over RCA. For the purposes of these measurements, unless otherwise stated, the following inputs were evaluated: digital coaxial S/PDIF (RCA), analog balanced (XLR), and phono RCA, configured using the default settings for moving-magnet (MM) and moving-coil (MC) cartridges. Comparisons were made between unbalanced (RCA) and balanced (XLR) line inputs and outputs, and no appreciable differences were seen in terms of gain and THD+N (FFTs for different configurations can be seen in this report).

Most measurements were made with a 2Vrms line-level analog and 0dBFS digital input with the volume set to achieve 2Vrms at the output. For the phono input, a 5mVrms MM level and 0.5mVrms MC level was used to achieve 1Vrms at the output. The signal-to-noise ratio (SNR) measurements were made with the same input signal values and, for comparison, on the line-level input, a SNR measurement was also made with the volume at maximum, but with a lower input voltage to achieve the same 2Vrms output.

The 891 also offers a range of gain settings (40 in total) by using the Offset feature in the configuration menu. The menu allows for a setting between -10dB to +10dB, in 0.5dB steps, individually assignable to each input. Note that this changes the gain for the input, it does not offset the volume level. Also of note, despite the -10dB to +10dB in the menu, actual gain varies from roughly -6dB to +14dB. The default setting in the menu is +6dB (+10dB of actual gain), which is, unless otherwise stated, what was used for these measurements.

The 891 offers two different volume steps in the setup menu: 0.5dB and 0.1dB. Based on the accuracy and random results of the left/right volume channel matching (see table below), the 891 volume control is likely digitally controlled but operating in the analog domain.

The 891 offers an incredible 620 (using the 0.1dB setting) volume steps from -69dB to 9.7dB for the line-level inputs. The first 20 steps (0 to 20dB) are in 1dB increments, and then the 20dB to 80dB volume positions can be changed in 0.1dB increments. Turning the volume knob quickly will increase the volume step sizes. It is also worth highlighting the superb channel matching in the table below. The worst-case deviation seen was 0.006dB, and very often throughout the volume range, left/right channel matching was measured at 0.000dB. This superb channel matching, along with the 620 steps and 0.1dB resolution, coupled with the volume knob’s silky smooth “feel” (also replicated on the remote control), add-up to what is arguably the finest volume control on any consumer audio device anywhere.

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

Published specifications vs. our primary measurements

The table below summarizes the measurements published by SimAudio for the 891 compared directly against our own. The published specifications are sourced from SimAudio’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 was set at its maximum (DC to 1MHz), unless otherwise stated, assume a 1kHz sinewave at 2Vrms or 0dBFS at the input, 2Vrms at the output into 200k ohms, and a measurement input bandwidth of 10Hz to 22.4kHz, and the worst-case measured result between the left and right channels.

Our primary measurements revealed the following using the balanced line-level analog input and digital coaxial input (unless specified, assume a 1kHz sinewave at 2Vrms or 0dBFS, 2Vrms output, 200kohm loading, 10Hz to 22.4kHz bandwidth):

*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 phono-level input, MM configuration (unless specified, assume a 1kHz sinewave at 5mVrms, 1Vrms output, 200kohm loading, 10Hz to 22.4kHz bandwidth):

Our primary measurements revealed the following using the phono-level input, MC configuration (unless specified, assume a 1kHz sinewave at 0.5mVrms, 1Vrms output, 200kohm loading, 10Hz to 22.4kHz bandwidth):

Frequency response (line-level input)

In our measured frequency response (relative to 1kHz) plot above, the 891 is near perfectly flat within the audioband (0dB at 20Hz, -0.05dB at 20kHz). At the extremes, the 891 is 0dB at 5Hz, -0.8dB at 100kHz, and -3dB just past 200kHz. These data corroborate SimAudio’s claim of 2Hz to 200kHz (0/-3dB). The 891 appears to be DC-coupled, as there is no attenuation at low frequencies, even at 5Hz. In the graph above and most of the graphs below, only a single trace may be visible. This is because the left channel (blue or purple trace) is performing identically to the right channel (red or green trace), and so they perfectly overlap, indicating that the two channels are ideally matched.

Phase response (line-level input)

Above is the phase response plot from 20Hz to 20kHz for the balanced line-level input. The 891 does not invert polarity and exhibits, at worst, less than -10 degrees (at 20kHz) of phase shift within the audioband.

Frequency response vs. input type (left channel only)

The chart above shows the 891’s frequency response (relative to 1kHz) as a function of input type. The green trace is the same (but limited to 80kHz) analog input data from the previous graph. The blue trace is for a 16-bit/44.1kHz dithered digital input signal from 5Hz to 22kHz using the coaxial input, the purple trace is for a 24/96 dithered digital input signal from 5Hz to 48kHz, and, finally, pink is 24/192 from 5Hz to 96kHz. The behavior at low frequencies is the same for all the digital sample rates, as well as the analog input: flat down to 5Hz. The behavior at high frequencies for all three digital sample rates is as expected, offering filtering around 22kHz, 48kHz, and 96kHz (half the respective sample rate). The 44.1kHz sampled input signal exhibits typical “brick-wall”-type behavior, with a -3dB point at 21kHz. The -3dB point for the 96kHz sampled data is at 46kHz, and 91kHz for the 192kHz sampled data.

Frequency response (MM input)

The chart above shows the frequency response (relative to 1 kHz) for the phono input (MM configuration). What is depicted is the deviation from the RIAA curve, where the input signal sweep is EQd with an inverted RIAA curve supplied by Audio Precision (i.e., zero deviation would yield a flat line at 0dB).  The result shows extremely small maximum deviations within the audioband: about +0.1dB from 5kHz to 20kHz and -0.05dB at 20Hz. This is an example of excellent RIAA tracking implemented in the analog domain.

Frequency response (MC input)

The chart above shows the frequency response for the phono input (MC configuration). We see essentially the same result as with the MM configuration.

Phase response (MM and MC phono inputs)

Above is the phase response plot from 20Hz to 20kHz for the phono input (MM and MC configurations behaved identically). For the phono input, since the RIAA equalization curve must be implemented, which ranges from +19.9dB (20Hz) to -32.6dB (90kHz), phase shift at the output is inevitable. Here we find a worst case of about -60 degrees at 200Hz and -90 degrees at 20kHz.

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 outputs of the 891. In this test, 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 would be a straight flat line at 0dB. At -120dBFS, the 16/44.1 data overshot by only 1-2dB, while the 24/96 remained perfect. To verify how well the 24/96 data would perform down to -140dBFS, we extended the sweep, shown in the chart below.

Here we can see that the 24/96 data only overshot the mark by +1dB (left/right) at -140dBFS. This is an exceptionally good digital-linearity test result.

Impulse response (24/48 data)

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, measured at the balanced outputs of the 891. We can see that the 891 utilizes a reconstruction filter that favors no pre-ringing and significant post-ringing.

J-Test (coaxial input)

The chart above shows the results of the J-Test test for the coaxial digital input measured at the line-level output of the 891. 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 sinewave (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 and below.  The test file can also be used in conjunction with artificially injected sinewave jitter by the Audio Precision, to show how well the DAC rejects jitter.

The coaxial S/PDIF input of the 891 shows a near-perfect J-Test result, with only two very small peaks on either side of the 12kHz fundamental at an extraordinarily low -155dBrA.

J-Test (optical input)

The chart above shows the results of the J-Test test for the optical digital input measured at the balanced outputs of the 891. The results here are similar but not quite as pristine as the coaxial input above. Here, the peaks adjacent to the 12kHz fundamental reach -145dBrA.

J-Test (AES-EBU input)

The chart above shows the results of the J-Test test for the AES-EBU digital input measured at the balanced outputs of the 891. The results here are essentially identical to those found on the coaxial input.

J-Test (coaxial, 2kHz sinewave jitter at 10ns)

The chart above shows the results of the J-Test test for the coaxial digital input measured at the line-level output of the 891, with an additional 10ns of 2kHz sinewave jitter injected by the APx555. The results are essentially the same as the J-Test result without additional jitter. The same was true for the optical input (not shown).

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

The chart above shows the results of the J-Test test for the coaxial digital input measured at the line-level output of the 891, with an additional 100ns of 2kHz sinewave jitter injected by the APx555. This time, the tell-tale peaks at 10kHz and 12kHz can be seen; however, they are very small in amplitude, just below -120dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (coaxial input)

The chart above shows a fast Fourier transform (FFT) of the 891’s balanced outputs with white noise at -4 dBFS (blue/red) and a 19.1 kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1. The steep roll-off around 20kHz in the white-noise spectrum shows that the 891 uses a brick-wall-type reconstruction filter. There are no obvious aliased images within the audioband. The primary aliasing signal at 25kHz is highly suppressed at -95dBrA, while the second and third distortion harmonics (38.2, 57.3kHz) of the 19.1kHz tone are even lower.

THD ratio (unweighted) vs. frequency vs. load (analog)

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 the analog balanced inputs. The 200k-ohm THD data are lower than the 600-ohm data by 5-10dB. The 200k-ohm data range from 0.0004% at 20Hz, down to an astonishingly low 0.00003% at 300Hz to 1.5kHz (left channel). It’s important to note that these THD values are only about twice as high as the AP’s signal generator, so they are pushing the limits of the analyzer’s capabilities. The right channel did exhibit about 5dB higher THD compared to the left channel above 1kHz. At 20kHz, the left channel into 200k ohms was at 0.0002%, while the right channel was at 0.0003%. The right channel into 600 ohms was at 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 for a 16/44.1 (blue/red) dithered 1kHz signal at the coaxial input and a 24/96 (purple/green) signal, as a function of frequency. The 16/44.1 THD ratios were slightly higher (0.0003% to 0.0002%) than the 24/96 THD ratios (0.00007% to 0.0002%) from 20Hz to 2kHz, due to the increased noise floor from the lower bit-depth (the analyzer cannot assign a THD ratio for harmonic peaks it cannot see above the noise floor). At 20kHz, all THD ratios measured 0.001%.

THD ratio (unweighted) vs. frequency (phono input, MM and MC)

The graph above shows THD ratio as a function of frequency plots for the phono input. The MM configuration is shown in blue/red (left/right) and MC in purple/green (left/right). The input sweep was EQ’d with an inverted RIAA curve. The THD values for the MM configuration vary from around 0.004% (20Hz) down to around 0.0002% (1kHz to 10kHz), then up to 0.0003% at 20kHz.  The MC THD values were higher, although these are limited by the higher noise floor (the analyzer cannot assign a THD ratio for harmonic peaks it cannot see above the noise floor). THD ratios ranged from 0.04% (20Hz), down to 0.0006% (8kHz), then up to 0.0015% at 20kHz.

THD ratio (unweighted) vs. output (analog)

The chart above shows THD ratios measured at the balanced outputs of the 891 as a function of output voltage for the balanced line-level input. THD values start at 0.05% at 1mVrms, down to a low of 0.00004% at 2-3Vrms, then a steep rise past 10Vrms to the 1% THD mark at 20Vrms.

THD+N ratio (unweighted) vs. output (analog)

The chart above shows THD+N ratios measured at the balanced outputs of the 891 as a function of output voltage for the balanced line level-input, with the volume control at maximum. THD+N values start at 0.4% at 1mVrms, down to a low of 0.00015% at 5Vrms, then a steep rise past 10Vrms to the 1% THD mark at 20Vrms.

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

The chart above shows THD ratios measured at the balanced outputs of the 891 as a function of output voltage for the digital coaxial S/PDIF input, swept from -90dBFS to 0dBFS, with the volume control at maximum. Blue/red traces are for 16/44.1 data and purple/green for 24/96. For the 16/44.1 data, THD values start at 3%, and predictably, reach their low at the maximum output voltage of about 6.5Vrms, at 0.0002%. For the 24/96 data, the right channel outperformed the left by 5dB, and THD ratios ranged from 0.2% down to 0.0001% at the maximum output voltage.

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

The chart above shows THD+N ratios measured at the balanced outputs of the 891 as a function of output voltage for the digital coaxial S/PDIF input, swept from -90dBFS to 0dBFS, with the volume control at maximum. Blue/red traces are for 16/44.1 data, and purple/green for 24/96. For the 16/44.1 data, THD+N values start at 30% and reach their low at the maximum output voltage of about 6.5Vrms, at 0.002%. For the 24/96 data, THD ratios ranged from 1% down to 0.0002% at the maximum output voltage.

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) input data and 24/96 input data (purple/green), from -60dBFS to 0dBFS. 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 16/44.1 data yields IMD ratios from 2% down to 0.003% at 0dBFS. The 24/96 data yields IMD ratios from 0.15% down to 0.0005% to 0.001% near 0dBFS.

FFT spectrum – 1kHz (XLR line-level input)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced outputs for the balanced line-level input. We see that the signal’s second harmonic, at 2kHz, is at -130dBrA or 0.00003%, and lower at -135dBrA or 0.00002% at the third (3kHz) harmonic. A few other signal harmonics are at a vanishingly low -140dBrA, or 0.00001%, and below. Below 1kHz, we see two small peaks from power-supply noise artifacts at 60Hz and 120Hz. These are extraordinarily low at around -150dBrA, or 0.000003%, and it must be highlighted that they are actually from the Audio Precision’s sinewave generator, and not inherent to the 891. We can say this with confidence for two reasons: these same peaks can be seen at roughly the same amplitudes when the Audio Precision’s sinewave generator is connected directly to the inputs of its analyzer (loopback), and these peaks are non-existent in the digital 24/96 FFT below, where the Audio Precision’s DAC generator is used instead of its analog sinewave generator. It should also be stressed how extraordinarily low the 891’s noise floor is—quite possibly the quietest preamp we’ve ever measured. The 891 does not seem to have any correlated power-supply-related noise (60Hz and harmonics related, which we would describe as “hum”). The residual A-weighted noise from the 891 (volume control set to minimum with no signal) due to non-correlated thermal noise (what we would describe as “hiss”) was measured at 0.97uVrms, compared to the analyzer’s self-noise of 0.66uVrms. Even with the volume set to the reference position (70.5dB) for our measurements and with a 2Vrms output signal present (notched out by the analyzer), A-weighted noise was measured at 2.2uVrms. It’s important to highlight here that Simaudio have managed to reduce both THD and noise when compared to their 791 model preamp, which measures extraordinarily well. In terms of THD, the main improvement is in the third harmonic, where the 791 yielded -115dBrA (0.0002%), compared to the 891’s -135dBrA (0.00002%). In terms of residual noise, the 791 measured an already astounding 1.23uVrms, only to be outdone by the 891’s 0.97uVrms! We can only commend Simaudio for producing what must be one of the most transparent, if not the most transparent, analog audio products in existence.

FFT spectrum – 1kHz (RCA line-level input)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the unbalanced outputs for the unbalanced line-level input. We see close to the same results as with the balanced input FFT above, except for a slightly higher 3kHz signal harmonic peak at -130dBrA instead of -135dBrA.

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 outputs for the coaxial digital input, sampled at 16/44.1. We see the second (2kHz) and third (3kHz) signal harmonics at roughly -120dBrA, or 0.0001%. The second harmonic peak for the right channel is at -130dBrA, or 0.00003%. The noise floor is much higher due to the 16-bit depth limitation.

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 outputs for the coaxial digital input, sampled at 24/96. We see no power-supply-related noise peaks above the -160dBrA noise floor, and the second (2kHz) and third (3kHz) signal harmonics at roughly -120dBrA, or 0.0001%. The second harmonic peak for the right channel is at -130dBrA, or 0.00003%. Higher signal harmonics can be seen at -130dBrA, or 0.00003%, and below.

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

Shown above is the FFT for a 1kHz -90dBFS dithered 16/44.1 input sinewave stimulus at the coaxial digital input, measured at the balanced outputs. We see the 1kHz primary signal peak, at the correct amplitude, and no signal- or noise-related harmonic peaks above the -140dBrA noise floor.

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

Shown above is the FFT for a 1kHz -90dBFS dithered 24/96 input sinewave stimulus at the coaxial digital input, measured at the balanced outputs. We see the 1kHz primary signal peak, at the correct amplitude, and signal related harmonic peaks at 2kHz and 4kHz at around -140dBrA, or 0.00001%. Other signal related harmonics can be seen but at the extremely low -150dBrA, or 0.000003%, level.

FFT spectrum – 1kHz (MM phono input)

Shown above is the FFT for a 1kHz input sinewave stimulus, measured at the balanced outputs across for the phono input, configured for MM (default 40dB of gain in phono stage). The dominant signal-related harmonic can be seen at 2kHz, at -115dBrA, or 0.0002%. Power-supply-related noise peaks can be seen at the -95dBrA, or 0.002%, level at 60Hz, and at 180Hz at -105dBrA (right channel), or 0.0006%.

FFT spectrum – 1kHz (MC phono input)

Shown above is the FFT for a 1kHz input sinewave stimulus, measured at the balanced outputs across for the phono input, configured for MC (default 60dB of gain in phono stage). There are no visible signal-related harmonic peaks above the -110 to -120dBrA noise floor. Power-supply-related noise peaks can be seen at the -75dBrA, or 0.02%, level at 60Hz, and at 180Hz at -85dBrA (right channel), or 0.006%.

FFT spectrum – 50Hz (line-level input)

Shown above is the FFT for a 50Hz input sinewave stimulus measured at the balanced outputs for the balanced line-level input. The X axis is zoomed in from 40 Hz to 1kHz, so that peaks from noise artifacts can be directly compared against peaks from the harmonics of the signal. The most predominant (non-signal) peaks are that of the signal’s second (100Hz) harmonic at -115dBrA or 0.0002%, and the third signal harmonic (150Hz) at -125dBrA, or 0.00006%. Power-supply-related peaks can be seen at 60Hz (-140dBrA or 0.00001%) and 120Hz (-150dBrA or 0.000003%), but as discussed above, these are inherent to the Audio Precision sinewave generator.

FFT spectrum – 50Hz (MM phono input)

Shown above is the FFT for a 50Hz input sinewave stimulus measured at the balanced outputs for the phono input configured for MM. The most predominant (non-signal) peaks are that of the signal’s second (100Hz) harmonic at -110dBrA or 0.0003%, and the primary (60Hz) power-supply-noise harmonic at -95dBrA, or 0.002%.

FFT spectrum – 50Hz (MC phono input)

Shown above is the FFT for a 50Hz input sinewave stimulus measured at the balanced outputs for the phono input configured for MC. The most predominant (non-signal) peak is that of the primary (60Hz) power-supply-noise harmonic at -75dBrA, or 0.02%. No signal harmonic peaks can be seen above the noise floor.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, line-level input)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced outputs for the balanced line-level input. The input RMS values are set at -6.02dBrA so that, if summed for a mean frequency of 18.5kHz, would yield 10W (0dBrA) into 8 ohms at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -125dBrA or 0.00006%, while the third-order modulation products, at 17kHz and 20kHz, are at -120dBrA or 0.0001%. This is a very clean IMD result.

Intermodulation distortion FFT (line-level input, APx 32 tone)

Shown above is the FFT of the balanced outputs of the 891 with the APx 32-tone signal applied to the input. The combined amplitude of the 32 tones is the 0dBrA reference, and corresponds to 2Vrms. The intermodulation products—i.e., the “grass” between the test tones—are distortion products from the preamplifier and are around the extremely low -150dBrA, or 0.000003%, level.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, coaxial digital input, 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 outputs for the digital coaxial input at 16/44.1. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -130dBrA (left channel), or 0.00003%, while the third-order modulation products, at 17kHz and 20kHz, are at -120dBrA, or 0.0001%. We also see the main aliased peaks at 25.1kHz and 26.1kHz at around -100dBrA.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, coaxial digital input, 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 outputs for the digital coaxial input at 24/96. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -130dBrA (left), or 0.00003%, and -150dBrA (right), or 0.000003%, while the third-order modulation products, at 17kHz and 20kHz, are at -120dBrA, or 0.0001%.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, MM phono input)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced outputs across for the phono input configured for MM. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -105dBrA, or 0.0006%, while the third-order modulation products, at 17kHz and 20kHz, are at -120dBrA, or 0.0001%.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, MC phono input)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced outputs across for the phono input configured for MC. We find that the second-order modulation product (i.e., the difference signal of 1kHz) for the left channel is just barely noticeable above the noise floor at -105dBrA, or 0.0006%, while the third-order modulation products, at 17kHz and 20kHz, are at around -120dBrA, or 0.0001%, but only visible at 20kHz.

Square-wave response (10kHz)

Above is the 10kHz squarewave response using the analog line-level input, at roughly 2Vrms. Due to limitations inherent to the Audio Precision APx555 B Series analyzer, this graph should not be used to infer or extrapolate the 891’s slew-rate performance. Rather, it should be seen as a qualitative representation of its extended bandwidth. An ideal squarewave can be represented as the sum of a sinewave and an infinite series of its odd-order harmonics (e.g., 10kHz + 30kHz + 50kHz + 70kHz . . .). A limited bandwidth will show only the sum of the lower-order harmonics, which may result in noticeable undershoot and/or overshoot, and softening of the edges. The 891 reproduction of the 10kHz square wave is very clean, with only extremely mild softening in the edges.

Diego Estan
Electronics Measurement Specialist

Details

Parent Category: Products

Category: Preamplifier Measurements

Created: 01 October 2024

Link: written about by Jason Thorpe on SoundStage! Ultra on October 1, 2024

General information

The Sonic Frontiers SFL-2 was released in about 1995. The SFL-2 under test here was owned by SoundStage! writer Jason Thorpe. He bought it in the late 1990s and had used it in his system until late 2023. He’s since sold it to a consumer. But before he sold it, we wanted to measure it to gauge its current level of performance. It was measured in March 2024. Jason reports that the tubes at time of measurements had under 2000 hours on them, which he felt meant they had plenty of life left since they were rated for about 10,000 hours.

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

The SFL-2 was conditioned for 30 minutes at 2Vrms at the output before any measurements were taken. All measurements were taken with both channels driven.

The SFL-2 offers five sets of line-level unbalanced (RCA) inputs, one set of line-level balanced (XLR) inputs, and two sets each of unbalanced and balanced outputs. The balanced outputs offer 6dB more gain than the unbalanced outputs. The volume control is a stepped attenuator with 23 positions (including the minimum position, which is grounded). Based on the accuracy and repeatable nature of the channel deviation (table below), the volume control is in the analog domain but must have been, at one time, carefully level matched between channels at each volume position due to the high level of accuracy. The stepped attenuator offers, for the most part, 3dB increments throughout the entire range. There’s an additional 0dB/-1.5dB switch that can enable 1.5dB of attenuation, for a finer adjustment.  

There is a significant difference in terms of THD between unbalanced and balanced inputs and outputs in the SFL-2 (see both the main table and FFTs below). The lowest THD configuration measured was balanced in/balanced out, although the left channel exhibits much higher THD than the right channel. Unless otherwise stated, measurements were made with the volume set to unity gain, using the XLR inputs and outputs, with a 2Vrms input.

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

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Primary measurements

Our primary measurements revealed the following using the balanced line-level inputs (unless otherwise specified, assume a 1kHz sinewave, 2Vrms input and output into 200k ohms load, 10Hz to 22.4kHz bandwidth):

Frequency response

In our measured frequency-response plot above, the SFL-2 is essentially perfectly flat within the audioband, and 0dB at 5Hz and -1.5dB at 200kHz. The SFL-2 appears to be DC-coupled, as it yielded 0dB of deviation at 5Hz. In the graph above and most of the graphs below, only a single trace may be visible. This is because the left channel (blue or purple trace) is performing identically to the right channel (red or green trace), and so they perfectly overlap, indicating that the two channels are ideally matched.

Phase response

Above is the phase-response plot from 20Hz to 20kHz. The blue/red traces are with the phase switch set to 0 degrees, the purple/green traces are with the phase switch set to 180 degrees. The SFL-2 does not invert polarity (except when the phase switch is set to 180 degrees), and it yielded a worst-case 5 degrees or so of phase shift at 20kHz.

THD ratio (unweighted) vs. frequency

The plot above shows THD ratios at the output as a function of frequency (20Hz to 20kHz) for a sinewave input stimulus. The blue and red plots are for left and right channels into 200k ohms, while purple/green (L/R) are into 600 ohms. As previously mentioned, there are gross differences in THD ratios between the left and right channels into 200k ohms. The right channel ranged from 0.003% at 20Hz, down to 0.0004% at 1 to 2kHz, then up to 0.003% at 20kHz. It should be noted that because this is a tube-based preamp, noise levels are higher than what would be seen in a well-designed solid-state preamp. As such, the limiting factor in assigning THD ratios for the right channel into 200k ohms is largely due to the noise floor (the analyzer cannot assign a THD ratio below the noise floor). THD ratios for the left channel into 200k ohms were relatively constant through the audioband, just above 0.01%. The 600-ohm THD ratio were considerable higher, ranging from 0.03/0.1% (left/right) at 20Hz, then up to 0.3/0.4% (left/right) from 200Hz to 20kHz. The SFL-2 is significantly impacted by a lower impedance load, due to the high output impedance (about 800 ohms) tube outputs. This is also evidenced by the maximum output signals (at 1% THD) measured at the balanced outputs into 200k ohms and 600 ohms: a staggering 90Vrms versus 1.5Vrms.

THD ratio (unweighted) vs. output voltage

The plot above shows THD ratios measured at the output of the SFL-2 as a function of output voltage into 200k ohms with a 1kHz input sinewave. At the 10mVrms level, THD values measured around 0.1% at 10mVrms, dipping down to around 0.0003% at 2Vrms for the right channel and 0.003% at 0.4Vrms for the left channel. Between 1 and 30Vrms at the output, we find a 10 to 25dB difference in THD between the left and right channels. The 1% THD point is reached at around 90Vrms. It’s also important to mention that anything above 2-4Vrms is not typically required to drive most power amps to full power.

THD+N ratio (unweighted) vs. output voltage

The plot above shows THD+N ratios measured at the output of the SFL-2 as a function of output voltage into 200k ohms with a 1kHz input sinewave. At the 10mVrms level, THD+N values measured around 1.5%, dipping down to around 0.003% at 5-7Vrms for the right channel, and 0.01% at 1-2Vrms for the left channel. Between 3 and 30Vrms at the output, we find a 10 to 25dB difference in THD+N between the left and right channels.

FFT spectrum – 1kHz (balanced in, balanced out)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the output into a 200k-ohm load, for the balanced inputs and outputs. We see that the signal’s second harmonic, at 2kHz, is at around -80dBrA, or 0.01%, for the left channel, and only -120dBrA, or 0.0001% for the right channel. The third harmonic, at 3 kHz, is at -115dBrA, or 0.0002%, for both channels. There are no power-supply-related noise peaks above the relatively high noise floor, which varies from -100dBrA at low frequencies, down to -130dBrA at 20kHz.

FFT spectrum – 1kHz (unbalanced in, balanced out)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the output into a 200k-ohm load, for the unbalanced inputs and balanced outputs. We see that the signal’s second harmonic, at 2kHz, is at around -70/-75dBrA (left/right), or 0.03/0.02%. The third harmonic, at 3kHz, is at -110dBrA, or 0.0003%, for both channels. There are no significant power-supply-related noise peaks above the relatively high noise floor, but for a very small -110dBrA (left channel), or 0.0003%, peak at 120Hz.

FFT spectrum – 1kHz (unbalanced in, unbalanced out)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the output into a 200k-ohm load, for the unbalanced inputs and outputs. We see that the signal’s second harmonic, at 2kHz, is at around -70/-60dBrA (left/right), or 0.03/0.1%. The third harmonic, at 3 kHz, is at -100dBrA, or 0.001%, for both channels. There are no significant power-supply-related noise peaks above the relatively high noise floor, except for the very small -110dBrA (right channel), or 0.0003%, peak at 120Hz.

FFT spectrum – 1kHz (balanced in, unbalanced out)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the output into a 200k-ohm load, for the balanced inputs and unbalanced outputs. We see that the signal’s second harmonic, at 2kHz, is at around -60dBrA, or 0.1%. The third harmonic, at 3kHz, is at -100dBrA, or 0.001%, for both channels. Using the unbalanced outputs appears to yield the worst-case THD ratios. Again, there are no significant power-supply-related noise peaks above the relatively high noise floor, except for a very small -110dBrA (right channel), or 0.0003%, peak at 120Hz.

FFT spectrum – 50Hz

Shown above is the FFT for a 50Hz input sinewave stimulus measured at the output into a 200k-ohm load. The X axis is zoomed in from 40Hz to 1kHz, so that peaks from noise artifacts can be directly compared against peaks from the harmonics of the signal. The only significant non-signal peak is from the signal’s second harmonic (100Hz) at -80/-105dBrA (left/right), or 0.01/0.0006%. The third signal harmonic (150Hz) is at -115dBrA, or 0.0002%. As above, there are no significant-power-supply related noise peaks above the relatively high noise floor, except for the very small -110dBrA (left channel), or 0.0003%, peak at 120Hz.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the output into a 200k-ohm load. The input RMS values are set at -6.02dBrA so that, if summed for a mean frequency of 18.5kHz, would yield 2Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -85/-95dBrA (left/right), or 0.006/0.002%, while the third-order modulation products, at 17kHz and 20kHz, are at -115dBrA, or 0.0002%.

Intermodulation distortion FFT (line-level input, APx 32 tone)

Shown above is the FFT of the balanced outputs of the SFL-2 with the APx 32-tone signal applied to the analog balanced input. The combined amplitude of the 32 tones is the 0dBrA reference, and corresponds to 2Vrms. The intermodulation products—i.e., the “grass” between the test tones—are distortion products from the amplifier and are around the -115dBrA level for the left channel and the -130dBrA level for the right channel (below 15kHz or so).

Square-wave response (10kHz)

Above is the 10kHz squarewave response at the output into 200k ohms. Due to limitations inherent to the Audio Precision APx555 B Series analyzer, this graph should not be used to infer or extrapolate the SFL-2’s slew-rate performance. Rather, it should be seen as a qualitative representation of its very high bandwidth. An ideal squarewave can be represented as the sum of a sinewave and an infinite series of its odd-order harmonics (e.g., 10kHz + 30kHz + 50kHz + 70kHz . . .). A limited bandwidth will show only the sum of the lower-order harmonics, which may result in noticeable undershoot and/or overshoot, and softening of the edges. The SFL-2’s reproduction of the 10kHz squarewave is essentially perfect, with sharp corners and no ringing.

Diego Estan
Electronics Measurement Specialist

Details

Parent Category: Products

Category: Preamplifier Measurements

Created: 01 September 2024

Link: reviewed by Roger Kanno on SoundStage! Hi-Fi on September 1, 2024

General Information

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

The NAD Masters M66 was conditioned for 30 minutes 2Vrms in/out into 200k ohms before any measurements were taken.

The M66 offers a multitude of inputs, both digital and analog (balanced and unbalanced), and line-level/subwoofer analog balanced outputs over XLR and unbalanced over RCA. Also offered is a ¼″ TRS headphone output on the front panel. For the purposes of these measurements, unless otherwise stated, the following inputs were evaluated: digital coaxial S/PDIF (RCA), analog balanced (XLR), as well as phono moving magnet (MM) and moving coil (MC). Comparisons were made between unbalanced (RCA) and balanced (XLR) line inputs and outputs, other than the 6dB of extra gain over the balanced outputs, there were no appreciable differences in terms THD+N (FFTs for different configurations can be seen in this report). Unless otherwise stated, the Analog Direct mode was used for the analog inputs.

Most measurements were made with a 2Vrms line-level and 0dBFS digital input with the volume set to achieve 2Vrms at the output. For the phono input, a 6.4mVrms MM level and 0.5mVrms MC level was used to achieve 1Vrms at the output. Of note is the low preamp gain in Analog Direct mode (-3.6/2.7dB RCA/XLR out). Using the MM input over the balanced output, 1Vrms could not be achieved with the volume at maximum with a standard 5mVrms input. Also noteworthy is the early onset of clipping when Analog Direct is turned off (which is required to use DSP functions). The ADC clipped with a 1.92Vrms input (RCA and XLR). The signal-to-noise ratio (SNR) measurements were made with the same input signal values and for comparison, on the line-level input, a SNR measurement was also made with the volume at maximum, but with a lower input voltage to achieve the same 2Vrms output.

The M66 offers 80 volume steps in 1dB increments, from -76dB to +2.7dB (line-level, XLR in/out). Based on the random and non-repeatable channel deviation values observed below, the M66 utilizes a digitally controlled volume control operating in the analog domain.

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

Published specifications vs. our primary measurements

The table below summarizes the measurements published by NAD for the M66 compared directly against our own. The published specifications are sourced from NAD’s website, either directly or from the manual available for download, or a combination thereof. Except for frequency response, where the Audio Precision bandwidth was set at its maximum (DC to 1MHz), unless otherwise stated, assume a 1kHz sinewave at 2Vrms or 0dBFS at the input, 2Vrms at the output into 200k ohms, and a measurement input bandwidth of 10Hz to 22.4kHz, and the worst-case measured result between the left and right channels.

Our primary measurements revealed the following using the balanced line-level analog input and digital coaxial input (unless specified, assume a 1kHz sinewave at 2Vrms or 0dBFS, 2Vrms output, 200k-ohm loading, 10Hz to 22.4kHz bandwidth):

*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 phono-level input, MM configuration (unless specified, assume a 1kHz sinewave at 6.4mVrms, 1Vrms output, 200k-ohm loading, 10Hz to 22.4kHz bandwidth):

Our primary measurements revealed the following using the phono-level input, MC configuration (unless specified, assume a 1kHz sinewave at 0.5mVrms, 1Vrms output, 200k-ohm loading, 10Hz to 22.4kHz bandwidth)

Our primary measurements revealed the following using the analog input at the headphone output (unless specified, assume a 1kHz sinewave, 2Vrms input/output, 300-ohm loading, 10Hz to 22.4kHz bandwidth, Analog Direct Mode on):

Frequency response (line-level input)

In our measured frequency-response (relative to 1kHz) chart above, the M66 is essentially perfectly flat within the audioband (0dB at 20Hz and 20kHz). At the extremes, the M66 is 0dB at 5Hz, and -0.2dB at 200kHz. These data corroborate NAD’s claim of ±0.2dB (20Hz-80kHz). With Analog Direct mode, the M66 can be considered an extremely wide-bandwidth audio device. The M66 also appears to be DC-coupled, as there is no attenuation at low frequencies, even at 5Hz. In the graph above and most of the graphs below, only a single trace may be visible. This is because the left channel (blue or purple trace) is performing identically to the right channel (red or green trace), and so they perfectly overlap, indicating that the two channels are ideally matched.

Frequency response (line-level input, subwoofers active)

Above are the frequency-response (relative to 1kHz and 20Hz) plots for the M66 balanced line-level outputs (red/blue) with two subwoofers (purple/green) engaged and the crossover set to 120Hz. In order to implement bass management, Analog Direct must be turned off to enable DSP functions. The line-level outputs show sharp attenuation just past 30kHz, with a -3dB point just past 40kHz, suggesting that the M66 samples incoming analog signals at 96kHz. The high- and low-pass slopes appear to be second-order (12dB/octave), with the crossover point at -6dB. The subwoofer outputs are perfectly flat down to 5Hz.

Phase response (line-level input)

Above is the phase response plot from 20Hz to 20kHz for the balanced line-level input. The M66 does not invert polarity and exhibits, at worst, less than -5 degrees (at 20kHz) of phase shift within the audioband.

Frequency response vs. input type (left channel only)

The chart above shows the M66’s frequency response (relative to 1kHz) as a function of input type. The green trace is the same (but limited to 80kHz) analog input data from the first frequency- response graph. The blue trace is for a 16-bit/44.1kHz dithered digital input signal from 5Hz to 22kHz, the purple trace is for a 24/96 dithered digital input signal from 5Hz to 48kHz, and finally pink is 24/192 from 5Hz to 96kHz—all using the coaxial input. The behavior at low frequencies is the same for all the digital sample rates, as well as the analog input: 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). All three digital data plots show “brick-wall”-type behavior, with-3dB points at 21.8kHz (16/44.1), 47kHz (24/96), and 94kHz (24/192).

Frequency response (MM input)

The chart above shows the frequency response (relative to 1 kHz) for the MM phono input. What is shown is the deviation from the RIAA curve, where the input signal sweep is EQ’d with an inverted RIAA curve supplied by Audio Precision (i.e., zero deviation would yield a flat line at 0dB). The chart shows extremely small maximum deviations within the audioband: 0dB at 20Hz and about +0.2dB at 20kHz. This is an example of exceptional RIAA tracking implemented in the analog domain (Analog Direct is turned on). At the extremes, there is sharp attenuation at low frequencies (-3dB at 5Hz) and a rise at high frequencies (+0.5dB at 50kHz).

Frequency response (MC input)

The chart above shows the frequency response for the MC phono input. We see essentially the same result as with the MM configuration.

Phase response (MM and MC phono inputs)

Above is the phase-response plot from 20Hz to 20kHz for the phono input (MM and MC behaved identically). For the phono input, since the RIAA equalization curve must be implemented, which ranges from +19.9dB (20Hz) to -32.6dB (90kHz), phase shift at the output is inevitable. Here we find a worst case of about -60 degrees at 200Hz and -90 degrees at 20kHz.

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 outputs of the M66. 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 would be a straight flat line at 0dB. At -120dBFS, the 16/44.1 data overshot by only 1-2dB, while the 24/96 remained perfect. To verify how well the 24/96 data would perform down to -140dBFS, we extended the sweep in the chart below.

Here we can see that the 24/96 data only undershot the mark by 2dB (left/right) at -140/-130dBFS. This is an exceptional digital linearity-test result.

Impulse response (24/48 data)

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, measured at the balanced outputs of the M66. We can see that the M66 utilizes a reconstruction filter that favors no pre-ringing.

J-Test (coaxial input)

The chart above shows the results of the J-Test test for the coaxial digital input measured at the line-level pre-outputs of the M66 where 0dBFS is set to 1Vrms. 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 sinewave (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 and below.  The test file can also be used in conjunction with artificially injected sinewave jitter by the Audio Precision, to show how well the DAC rejects jitter.

The coaxial S/PDIF input of the M66 shows a strong but not perfect J-Test result, with small peaks in the audioband at a low -135dBrA and below level.

J-Test (optical input)

The chart above shows the results of the J-Test test for the optical digital input measured at the balanced outputs of the M66. The results here are similar but not quite as strong as the coaxial input above. Here, the peaks reach the -125dBrA level.

J-Test (AES-EBU input)

The chart above shows the results of the J-Test test for the AES-EBU balanced digital input measured at the balanced outputs of the M66. The results here are very similar to those found on the coaxial input.

J-Test (coaxial, 2kHz sinewave jitter at 10ns)

The chart above shows the results of the J-Test test for the coaxial digital input measured at the line-level output of the M66, with an additional 10ns of 2kHz sinewave jitter injected by the APx555. The results are close to the same as the J-Test result without additional jitter. The same was true for the optical input. No peaks can be seen at the 10kHz and 14kHz positions.

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

The chart above shows the results of the J-Test test for the coaxial digital input measured at the line-level output of the M66, with an additional 100ns of 2kHz sinewave jitter injected by the APx555. The tell-tale peaks at 10kHz and 14kHz still cannot be seen. This is a strong result showing that the M66 DAC has very good jitter immunity.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (coaxial input)

The chart above shows a fast Fourier transform (FFT) of the M66’s balanced outputs with white noise at -4 dBFS (blue/red) and a 19.1 kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1. The steep roll-off around 20kHz in the white-noise spectrum shows that the M66 uses a brick-wall-type reconstruction filter. There are no obvious aliased images within the audioband. The primary aliasing signal at 25kHz is highly suppressed at -70dBrA, while the second and third distortion harmonics (38.2, 57.3kHz) of the 19.1kHz tone are even lower.

THD ratio (unweighted) vs. frequency vs. load (analog)

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 the analog balanced inputs. The 200k-ohm THD data are lower than the 600-ohm data by 5-10dB from 20Hz to 200Hz. The 200k-ohm data range from 0.00015% at 20Hz, down to an astonishingly low 0.00003% at 200Hz, then a steady climb to 0.002% near 20kHz. It’s important to note that these THD values are only about twice as high as the AP’s signal generator, thereby pushing the limits of the analyzer’s capabilities.

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 for a 16/44.1 (blue/red) dithered 1kHz signal at the coaxial input and a 24/96 (purple/green) signal, as a function of frequency. The 16/44.1 THD ratios were slightly higher (0.0003% to 0.0002%) than the 24/96 THD ratios, which were more of a constant 0.0002% across most of the audioband.

THD ratio (unweighted) vs. frequency (phono input, MM and MC)

The graph above shows THD ratio as a function of frequency plots for the phono input. The MM input is shown in blue/red (left/right channels) and MC in purple/green (left/right channels). The input sweep is EQ’d with an inverted RIAA curve. The THD values for the MM configuration vary from around 0.02% (20Hz) down to around 0.0003% (2kHz to 3kHz), then up to 0.0005% at 20kHz. The MC THD values were higher, although these are limited by the higher noise floor (the analyzer cannot assign a THD ratio for harmonic peaks it cannot see above the noise floor). THD ratios ranged from 0.3% (20Hz), down to 0.002% (2kHz to 20kHz). It should be pointed out that the higher THD ratios at low frequencies are also likely limited and driven by the higher noise floor.

THD ratio (unweighted) vs. output (analog)

The chart above shows THD ratios measured at the balanced outputs of the M66 as a function of output voltage for the balanced line-level input, with the volume control at maximum. THD values start at 0.03% at 1mVrms, down to a low of 0.00004% at 2-3Vrms, then a steep rise past 10Vrms to the 1% THD mark at 15.5Vrms.

THD+N ratio (unweighted) vs. output (analog)

The chart above shows THD+N ratios measured at the balanced outputs of the M66 as a function of output voltage for the balanced line level-input, with the volume control at maximum. THD+N values start at 0.4% at 1mVrms, down to a low of 0.00015% at 4-5Vrms, then a steep rise past 10Vrms to the 1% THD mark at 15.5Vrms.

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

The chart above shows THD ratios measured at the balanced outputs of the M66 as a function of output voltage for the digital coaxial S/PDIF input, swept from -90dBFS to 0dBFS, with the volume control at maximum. Blue/red traces are for 16/44.1 data, and purple/green for 24/96. For the 16/44.1 data, THD values start at 2%, and predictably, reach their low at the maximum output voltage of about 5Vrms, at 0.0003%. For the 24/96 data, THD ratios ranged from 0.1% down to 0.0001% near the maximum output voltage.

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

The chart above shows THD+N ratios measured at the balanced outputs of the M66 as a function of output voltage for the digital coaxial S/PDIF input, swept from -90dBFS to 0dBFS, with the volume control at maximum. Blue/red traces are for 16/44.1 data, and purple/green for 24/96. For the 16/44.1 data, THD+N values start at 20% and reach their low at the maximum output voltage of about 5Vrms, at 0.002%. For the 24/96 data, THD ratios ranged from 0.5% down to 0.0002% near the maximum output voltage.

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 for 16/44.1 (blue/red) input data and 24/96 input data (purple/green), from -60dBFS to 0dBFS. 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 16/44.1 data yields IMD ratios from 2% down to 0.003% at 0dBFS. The 24/96 data yields IMD ratios from 0.1% down to 0.0005% at 0dBFS.

FFT spectrum – 1kHz (XLR line-level input)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced outputs for the balanced line-level input. We see that the signal’s third harmonic, at 3kHz, is at -130dBrA, or 0.00003%. There are a multitude of high-order signal harmonics, but they are all at the extremely low -140dBrA, or 0.00001%, level. Below 1kHz, we see two small peaks from power-supply noise artifacts at 60Hz and 120Hz. These are extraordinarily low at around -150dBrA, or 0.000003%, and it must be highlighted that they are actually from the Audio Precision’s sinewave generator, and not inherent to the M66. We can say this with confidence for two reasons: these same peaks can be seen at roughly the same amplitudes when the Audio Precision’s sinewave generator is connected directly to the inputs of its analyzer (loopback), and these peaks are non-existent in the digital 24/96 FFT below, where the Audio Precision’s DAC generator is used instead of its analog sine wave generator. It should also be stressed how extraordinarily low the M66’s noise floor is. The M66 does not seem to have any correlated power-supply-related (60Hz and harmonics) noise (what we would describe as “hum”). The residual A-weighted noise from the M66 (volume control set to minimum with no signal) due to non-correlated thermal noise (what we would describe as “hiss”) was measured at 1.7uVrms, compared to the analyzer’s self-noise of 0.66 uVrms. FFTs don’t get much cleaner than this!

FFT spectrum – 1kHz (RCA line-level input)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the unbalanced outputs for the unbalanced line-level input. We see essentially the same result as with the balanced input FFT above.

FFT spectrum – 1kHz (XLR line-level input, Analog Direct off)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced outputs for the balanced line-level input, with Analog Direct turned off. Here we see signal harmonics at higher levels compared to when the signal path is purely analog. There are a multitude of signal harmonics ranging from -120dBrA, or 0.0001%, down to -140dBrA, or 0.00001%. We also see the IMD peaks between the 1kHz signal and the ADC’s 96kHz sample rate at 95 and 97kHz.

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 outputs for the coaxial digital input, sampled at 16/44.1. We see the third (3kHz) signal harmonic dominate at roughly -120dBrA, or 0.0001%.

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 outputs for the coaxial digital input, sampled at 24/96. We see no power-supply-related noise peaks above the -155dBrA noise floor, and the third (3kHz) signal harmonic dominates at roughly -120dBrA, or 0.0001%. A multitude of signal harmonics can also be seen below this level down to -140dBrA, or 0.00001%.

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

Shown above is the FFT for a 1kHz -90dBFS dithered 16/44.1 input sinewave stimulus at the coaxial digital input, measured at the balanced outputs. We see the 1kHz primary signal peak, at the correct amplitude, and no signal or noise-related harmonic peaks above the -140dBrA noise floor.

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

Shown above is the FFT for a 1kHz -90dBFS dithered 24/96 input sinewave stimulus at the coaxial digital input, measured at the balanced outputs. We see the 1kHz primary signal peak, at the correct amplitude, and signal related harmonic peaks at 2kHz and 4kHz at and below -140dBrA, or 0.00001%.

FFT spectrum – 1kHz (MM phono input)

Shown above is the FFT for a 1kHz input sinewave stimulus, measured at the balanced outputs, for the MM phono input. It is very difficult to see any signal-related harmonics above the noise floor and the myriad of noise-related peaks, at -80dBrA, or 0.01%, and below. Given how clean the M66’s line-level analog FFT is, the MM phono FFT is not the cleanest and somewhat disappointing by comparison.

FFT spectrum – 1kHz (MC phono input)

Shown above is the FFT for a 1kHz input sinewave stimulus, measured at the balanced outputs, for the MC phono input. Again, it is very difficult to see any signal-related harmonics above the noise floor and the myriad of noise-related peaks, at -60dBrA, or 0.1%, and below. Once again, given how clean the M66’s line-level analog FFT is, the MC phono FFT is not the cleanest and is disappointing by comparison.

FFT spectrum – 50Hz (line-level input)

Shown above is the FFT for a 50Hz input sinewave stimulus measured at the balanced outputs for the balanced line-level input. The X axis is zoomed in from 40 Hz to 1kHz, so that peaks from noise artifacts can be directly compared against peaks from the harmonics of the signal. The most predominant (non-signal) peaks are that of the signal’s second (100Hz) harmonic at -120dBrA, or 0.0001%, and the third signal harmonic (150Hz) at -130dBrA or 0.00003%. Power-supply-related peaks can be seen at 60Hz (-140dBrA, or 0.00001%) and 120Hz (-150dBrA, or 0.000003%), but as discussed above, these are inherent to the Audio Precision sinewave generator.

FFT spectrum – 50Hz (MM phono input)

Shown above is the FFT for a 50Hz input sinewave stimulus measured at the balanced outputs for the MM phono input. There are a myriad of noise-related peaks at -80dBrA, or 0.01%, and below.

FFT spectrum – 50Hz (MC phono input)

Shown above is the FFT for a 50Hz input sinewave stimulus measured at the balanced outputs for the MC phono input. There are, again, numerious noise-related peaks at -60dBrA, or 0.01%, and below.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, line-level input)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced outputs for the balanced line-level input. The input RMS values are set at -6.02dBrA so that, if summed for a mean frequency of 18.5kHz, would yield 10W (0dBrA) into 8 ohms at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -125dBRA, or 0.00006%, while the third-order modulation products, at 17kHz and 20kHz are at roughly the same level. This is a very clean IMD result.

Intermodulation distortion FFT (line-level input, APx 32 tone)

Shown above is the FFT of the balanced outputs of the M66 with the APx 32-tone signal applied to the input. The combined amplitude of the 32 tones is the 0dBrA reference and corresponds to 2Vrms. The intermodulation products—i.e., the “grass” between the test tones—are distortion products from the preamplifier and are around the extremely low -150dBrA, or 0.000003%, level.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, coaxial digital input, 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 outputs for the digital coaxial input at 16/44.1. We find that the second-order modulation product (i.e., the difference signal of 1kHz) cannot be seen above the -135dBrA noise floor, while the third-order modulation products, at 17kHz and 20kHz, are at -125dBrA, or 0.00006%. We also see the main aliased peaks at 25.1kHz and 26.1kHz at around -80dBrA.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, coaxial digital input, 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 outputs for the digital coaxial input at 24/96. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -135dBrA, or 0.00002%, while the third-order modulation products, at 17kHz and 20kHz, are at -125dBrA or 0.00006%.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, MM phono input)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced outputs across for the MM phono input. We find it difficult to distinguish the second-order and third-order IMD peaks amongst all the noise-related peaks.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, MC phono input)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced outputs across for the MC phono input. Once again, we find it difficult to distinguish the second-order and third-order IMD peaks amongst all the noise-related peaks.

Square-wave response (10kHz)

Above is the 10kHz squarewave response using the analog line-level input, at roughly 2Vrms. Due to limitations inherent to the Audio Precision APx555 B Series analyzer, this graph should not be used to infer or extrapolate the M66’s slew-rate performance. Rather, it should be seen as a qualitative representation of its extended bandwidth. An ideal squarewave can be represented as the sum of a sinewave and an infinite series of its odd-order harmonics (e.g., 10kHz + 30kHz + 50kHz + 70kHz . . .). A limited bandwidth will show only the sum of the lower-order harmonics, which may result in noticeable undershoot and/or overshoot, and softening of the edges. The M66 reproduction of the 10kHz square wave is very clean, with no softening in the edges.

Diego Estan
Electronics Measurement Specialist

Details

Parent Category: Products

Category: Preamplifier Measurements

Created: 15 January 2024

Link: reviewed by Phil Gold on SoundStage! Hi-Fi on January 15, 2024

General information

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

The PRE was conditioned for 30 minutes at 2Vrms at the output before any measurements were taken. All measurements were taken with both channels driven.

The PRE offers two sets of line-level unbalanced (RCA) inputs, one set of line-level balanced (XLR) inputs, one set each of unbalanced (RCA) and balanced (XLR) outputs (both always on). The PRE offers a maximum of 6dB of gain from input to output for the same input type. That is to say, if the volume is set to unity gain, an input of 2Vrms will yield 2Vrms at the output for the unbalanced input/output scenario, and the balanced input/output scenario. For the unbalanced in/balanced out scenario, 12dB gain is available. For the balanced in/unbalanced out scenario, 0dB of gain is available.

Based on the accuracy and non-repeatable nature of the channel deviation (table below), the volume control is in the analog domain, but digitally controlled. It offers between 2 and 3dB step increments for the first 12 volume steps. From steps 12 to 22, 1dB steps were measured. Beyond level 22 up to 100, the volume control offers 0.5 dB steps.  Overall gain was measured at -68.7dB for volume step one, up to +6dB at the maximum position (100). Volume channel tracking proved exquisite, ranging from 0.000dB to 0.008dB.

There is a difference in terms of THD and noise between unbalanced and balanced signals in the PRE (see both the main table and FFTs below). The balanced outputs have about 6dB more uncorrelated thermal noise, whereas using the balanced inputs yields about 10dB less THD compared to the unbalanced inputs. Unfortunately, the lower distortion is only apparent in the FFTs, because they allow averages over multiple data runs, which averages out and lowers the noise floor, making the very low distortion peaks visible. During normal real-time THD measurements, the analyzer is set to measure for 2-3 seconds (maximum) and cannot assign a THD value below the measured noise floor. This explains why in the primary table below, THD appears lower for the unbalanced input/output compared to the balanced input/output. The true THD ratio figure for the balanced configuration, based on the balanced input/output FFT, is an astounding 0.00002% (about -135dB), compared to the 0.00007% (about -123dB) or so for the unbalanced input.

Unless otherwise stated, balanced input and output was evaluated, with an input and output of 2Vrms into a 200k ohm-load, with the analyzer’s input bandwidth filter set to 10Hz to 22.4kHz (exceptions include FFTs and THD vs frequency sweeps where the bandwidth is extended to 90kHz, and frequency and squarewave response where the bandwidth is extended from DC to 1MHz).

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

Published specifications vs. our primary measurements

The table below summarizes the measurements published by Meitner for the PRE compared directly against our own. The published specifications are sourced from Meitner’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 was set at its maximum (DC to 1MHz), assume, unless otherwise stated, a 1kHz sinewave, 2Vrms input and output into 200k ohms load, 10Hz to 22.4kHz bandwidth, and the worst-case measured result between the left and right channels.

*SNR measured with unbalanced in/out = 115.3dB
*SNR calculated with residual noise (volume at 0) and unbalanced in/out = 118.6dB

Our primary measurements revealed the following using the balanced line-level inputs (unless otherwise specified, assume a 1kHz sinewave, 2Vrms input and output into 200k ohms load, 10Hz to 22.4kHz bandwidth):

Frequency response

In our measured frequency-response plot above, the PRE is perfectly flat within the audioband (0dB at 20Hz and 20kHz). The PRE appears to be DC-coupled, as it yielded 0dB of deviation at 5Hz. The PRE can certainly be considered an extended-bandwidth audio device, as it is only 0.14dB down at 200kHz. In the graph above and most of the graphs below, only a single trace may be visible. This is because the left channel (blue or purple trace) is performing identically to the right channel (red or green trace), and so they perfectly overlap, indicating that the two channels are ideally matched.

Phase response

Above is the phase-response plot from 20Hz to 20kHz. The PRE does not invert polarity, and exhibited zero phase shift within the audioband.

THD ratio (unweighted) vs. frequency

The plot above shows THD ratios at the output as a function of frequency (20Hz to 20kHz) for a sinewave input stimulus. The blue and red plots are for the left and right channels into 200k ohms, while purple/green (L/R) are into 600 ohms. THD values were flat across most of the audioband at 0.0001% into 600 ohms and 200k ohms, with a small rise to 0.0002% at 20kHz. This shows that the PRE’s outputs are robust and would yield identical THD performance feeding an amplifier with either a high or low input impedance.

THD ratio (unweighted) vs. output voltage

The plot above shows THD ratios measured at the output of the PRE as a function of output voltage into 200k ohms with a 1kHz input sinewave. At the 10mVrms level, THD values measured around 0.03%, dipping down to around 0.00006% at 6-8Vrms, followed by a rise to 0.0003% at around 18Vrms. The 1% THD point is reached at 20.2Vrms. It’s also important to mention that anything above 2-4Vrms is not typically required to drive most power amps to full power.

THD+N ratio (unweighted) vs. output voltage

The plot above shows THD+N ratios measured at the output of the PRE as a function of output voltage into 200k ohms with a 1kHz input sinewave. At the 10mVrms level, THD+N values measured around 0.2%, dipping down to around 0.0003% at 10-18Vrms.

FFT spectrum – 1kHz (balanced in, balanced out)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the output into a 200k-ohm load, for the balanced inputs and outputs. We see that the signal’s second harmonic, at 2kHz, is extremely low at around -135dBrA, or 0.00002%, and subsequent signal harmonics are not visible above the -145dBrA noise floor. Below 1kHz, we can see very small peaks at 60, 120, 148, 180, and 300Hz. These peaks are all below the -130dBrA, or 0.00003%, level. This is a very clean FFT.

FFT spectrum – 1kHz (unbalanced in, balanced out)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the output into a 200k-ohm load for the unbalanced inputs and balanced outputs. The main difference here compared to the FFT above is the higher second signal harmonic, at -125dBRa, or 0.00006%, versus the -135dBrA 2kHz peak seen when the balanced inputs are used. Noise peaks left of the signal peak are at similar levels as the FFT above.

FFT spectrum – 1kHz (unbalanced in, unbalanced out)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the output into a 200k-ohm load for the unbalanced inputs and outputs. The same distortion profile with the higher 2kHz peaks can be seen here as with the FFT above. The common denominator is the use of the unbalanced inputs. The overall noise floor is at its lowest here, at -150dBrA.

FFT spectrum – 1kHz (balanced in, unbalanced out)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the output into a 200k-ohm load for the balanced inputs and unbalanced outputs. The same distortion profile with the lower 2kHz peaks can be seen here as with the first FFT above. The common denominator is the use of the balanced inputs.

FFT spectrum – 50Hz

Shown above is the FFT for a 50Hz input sinewave stimulus measured at the output into a 200k-ohm load. The X axis is zoomed in from 40Hz to 1kHz, so that peaks from noise artifacts can be directly compared against peaks from the harmonics of the signal. Signal-related peaks can be seen at the second (100Hz) and third (150Hz) harmonics, at an extremely low -140dBrA, or 0.00001%. Noise-related peaks are all below -135dBrA, or 0.00002%.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the output into a 200k-ohm load. The input RMS values are set at -6.02dBrA so that, if summed for a mean frequency of 18.5kHz, would yield 2Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -125dBrA, or 0.0006%, while the third-order modulation products, at 17kHz and 20kHz, are at roughly the same level. This, like the 1kHz FFTs, is an extremely clean result.

Intermodulation distortion FFT (line-level input, APx 32 tone)

Shown above is the FFT of the output of the PRE into 200k ohms with the APx 32-tone signal applied to the input. The combined amplitude of the 32 tones is the 0dBrA reference, and corresponds to 2Vrms into 200k ohms. The intermodulation products—i.e., the “grass” between the test tones—are distortion products from the amplifier. Distortion products are at a vanishingly low -140dBrA, or 0.00001%. Thus, even with a complex input signal, the PRE does not add any audible coloration to the input signal.

Square-wave response (10kHz)

Above is the 10kHz squarewave response at the output into 200k ohms. Due to limitations inherent to the Audio Precision APx555 B Series analyzer, this graph should not be used to infer or extrapolate the PRE’s slew-rate performance. Rather, it should be seen as a qualitative representation of its relatively high bandwidth. An ideal squarewave can be represented as the sum of a sinewave and an infinite series of its odd-order harmonics (e.g., 10kHz + 30kHz + 50kHz + 70kHz . . .). A limited bandwidth will show only the sum of the lower-order harmonics, which may result in noticeable undershoot and/or overshoot, and softening of the edges. The PRE’s reproduction of the 10kHz squarewave is extremely clean with sharp corners.

Diego Estan
Electronics Measurement Specialist

Details

Parent Category: Products

Category: Preamplifier Measurements

Created: 01 December 2023

Link: reviewed by Doug Schneider on SoundStage! Hi-Fi on December 1, 2023

General Information

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

The Simaudio Moon North Collection 791 was conditioned for 30 minutes with 2Vrms in/out into 200k ohms before any measurements were taken.

The 791 offers a multitude of inputs, both digital and analog (balanced and unbalanced), and line-level analog balanced outputs over XLR and unbalanced over RCA. For the purposes of these measurements, unless otherwise stated, the following inputs were evaluated: digital coaxial S/PDIF (RCA), analog balanced (XLR), as well as phono (RCA), configured both using the default settings for moving-magnet (MM) and moving-coil (MC) cartridges. Comparisons were made between unbalanced (RCA) and balanced (XLR) line inputs and outputs, and no appreciable differences were seen in terms of gain and THD+N (FFTs for different configurations can be seen in this report).

Most measurements were made with a 2Vrms line-level and 0dBFS digital input with the volume set to achieve 2Vrms at the output.  For the phono input, a 5mVrms MM level and 0.5mVrms MC level were used to achieve 1Vrms at the output. The signal-to-noise ratio (SNR) measurements were made with the same input-signal values and, for comparison, on the line-level input, a SNR measurement was also made with the volume at maximum, but with a lower input voltage to achieve the same 2Vrms output.

The 791 also offers a range of gain settings (40 in total) by using the Offset feature in the onscreen menu system. The menu allows for a setting between -10dB to +10dB, in 0.5dB steps, individually assignable to each input. Note that this changes the gain for the input, it does not offset the volume to level. Also of note, despite the -10dB to +10dB in the menu, actual gain varies from roughly -6dB to +14dB. The default setting in the menu is +6dB (+10dB of actual gain), which is, unless otherwise stated, what was used for these measurements.

Based on the accuracy and random results of the left/right volume channel matching (see table below), the 791 volume control is likely digitally controlled but operating in the analog domain. The 791 offers 140 volume steps from -69dB to 9.8dB for the line-level inputs. The first 20 steps (0 to 20dB) are in 1dB increments, and then the 20dB to 80dB volume positions can be changed in 0.5dB increments.

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

Published specifications vs. our primary measurements

The table below summarizes the measurements published by Simaudio for the 791 compared directly against our own. The published specifications are sourced from Simaudio’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 was set at its maximum (DC to 1MHz), assume, unless otherwise stated, assume a 1kHz sinewave at 2Vrms or 0dBFS at the input, 2Vrms at the output into 200k ohms, and a measurement input bandwidth of 10Hz to 22.4kHz, and the worst-case measured result between the left and right channels.

Our primary measurements revealed the following using the balanced line-level analog input and digital coaxial input (unless specified, assume a 1kHz sinewave at 2Vrms or 0dBFS, 2Vrms output, 200kohm loading, 10Hz to 22.4kHz bandwidth):

*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 phono-level input, MM configuration (unless specified, assume a 1kHz sinewave at 5mVrms, 1Vrms output, 200kohm loading, 10Hz to 22.4kHz bandwidth):

Our primary measurements revealed the following using the phono-level input, MC configuration (unless specified, assume a 1kHz sinewave at 0.5mVrms, 1Vrms output, 200kohm loading, 10Hz to 22.4kHz bandwidth):

Frequency response (line-level input)

In our measured frequency response (relative to 1kHz) plot above, the 791 is near perfectly flat within the audioband (0dB at 20Hz, -0.05dB at 20kHz). At the extremes the 791 is 0dB at 5Hz, and -0.8dB at 100kHz, and -3dB just past 200kHz. These data corroborate Simaudio’s claim of 2Hz to 100kHz (0/-3dB). The 791 appears to be DC-coupled, as there is no attenuation at low frequencies, even at 5Hz. In the graph above and most of the graphs below, only a single trace may be visible. This is because the left channel (blue or purple trace) is performing identically to the right channel (red or green trace), and so they perfectly overlap, indicating that the two channels are ideally matched.

Phase response (line-level input)

Above is the phase response plot from 20Hz to 20kHz for the balanced line-level input. The 791 does not invert polarity and exhibits, at worst, less than -10 degrees (at 20kHz) of phase shift within the audioband.

Frequency response vs. input type (left channel only)

The chart above shows the 791’s frequency response (relative to 1kHz) as a function of input type. The green trace is the same (but limited to 80kHz) analog input data from the previous graph. The blue trace is for a 16bit/44.1kHz dithered digital input signal from 5Hz to 22kHz, the purple trace is for a 24/96 dithered digital input signal from 5Hz to 48kHz, and finally pink is 24/192 from 5Hz to 96kHz (using the coaxial input). The behavior at low frequencies is the same for all the digital sample rates, as well as the analog input—flat down to 5Hz. The behavior at high frequencies for all three digital sample rates is as expected, offering filtering around 22, 48, and 96kHz (half the respective sample rate). The 44.1kHz sampled input signal exhibits typical “brick-wall”-type behavior, with a -3dB point at 21kHz. The -3dB point for the 96kHz sampled data is at 46kHz, and 91kHz for the 192kHz sampled data.

Frequency response (MM input)

The chart above shows the frequency response (relative to 1 kHz) for the phono input (MM configuration) and shows extremely small maximum deviations within the audioband of about +0.05 (100-200Hz) and -0.1dB (20kHz). What is shown is the deviation from the RIAA curve, where the input signal sweep is EQ’d with an inverted RIAA curve supplied by Audio Precision (i.e., zero deviation would yield a flat line at 0dB). This is an example of exceptionally accurate RIAA tracking.

Frequency response (MC input)

The chart above shows the frequency response for the phono input (MC configuration). We see essentially the same result as with the MM configuration.

Phase response (MM and MC phono inputs)

Above is the phase response plot from 20Hz to 20kHz for the phono input (MM and MC configurations behaved identically). For the phono input, since the RIAA equalization curve must be implemented, which ranges from +19.9dB (20Hz) to -32.6dB (90kHz), phase shift at the output is inevitable. Here we find a worst case of about +40 degrees at 20Hz and +20 degrees at 1kHz.

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 outputs of the 791. For this test, 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 would be a straight flat line at 0dB. At -120dBFS, the 16/44.1 data overshot by only 1-2dB, while the 24/96 remained perfect. To verify how well the 24/96 data would perform down to -140dBFS, we extended the sweep in the chart below.

Here we can see that the 24/96 data only overshot the mark by +2/+1dB (left/right) at -140dBFS. These tests show exceptional digital-linearity results for 16/44.1 and 24/96 data.

Impulse response (24/48 data)

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, measured at the balanced outputs of the 791. We can see that the 791 utilizes a reconstruction filter that favors no pre-ringing.

J-Test (coaxial input)

The chart above shows the results of the J-Test test for the coaxial digital input measured at the line-level output of the 791. 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 sinewave (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 and below.  The test file can also be used in conjunction with artificially injected sinewave jitter by the Audio Precision, to show how well the DAC rejects jitter.

The coaxial S/PDIF input of the 791 shows a near-perfect J-Test result, with only two very small peaks on either side of the 12kHz fundamental at a vanishingly low -155dBrA.

J-Test (optical input)

The chart above shows the results of the J-Test test for the optical digital input measured at the balanced outputs of the 791. The results here are similar but not quite as pristine as the coaxial input above. Here, the peaks adjacent to the 12kHz fundamental reach -145dBrA.

J-Test (coaxial, 2kHz sinewave jitter at 10ns)

The chart above shows the results of the J-Test test for the coaxial digital input measured at the line-level output of the 791, with an additional 10ns of 2kHz sinewave jitter injected by the APx555. The results are essentially the same as the J-Test result without additional jitter. The same was true for the optical input.

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

The chart above shows the results of the J-Test test for the coaxial digital input measured at the line-level output of the 791, with an additional 100ns of 2kHz sinewave jitter injected by the APx555. This time, the tell-tale peaks at 10kHz and 12kHz can be seen; however, they are very small in amplitude, just below -120dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (coaxial input)

The chart above shows a fast Fourier transform (FFT) of the 791’s balanced outputs 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. The steep roll-off around 20kHz in the white-noise spectrum shows that the 791 uses a brick-wall-type reconstruction filter. There are no aliased images within the audioband. The primary aliasing signal at 25kHz is highly suppressed at -95dBrA, while the second and third distortion harmonics (38.2, 57.3kHz) of the 19.1kHz tone are even lower.

THD ratio (unweighted) vs. frequency vs. load (analog)

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 the analog balanced inputs. The 200k and 600 ohms data are identical throughout the audioband, which is in indication that the 791’s outputs are robust and can handle loads below 1k ohms with no difficultly. THD ratios are very low, from 0.0003% to 0.0002%.

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 for a 16/44.1 (blue/red) dithered 1kHz signal at the coaxial input and a 24/96 (purple/green) signal, as a function of frequency. The 16/44.1 THD ratios were slightly higher than the 24/96, at 0.0005% to 0.0002%.  The 24/96 data ranged from 0.0003% down to 0.00015%.

THD ratio (unweighted) vs. frequency (phono input, MM and MC)

The graph above shows THD ratio as a function of frequency plot for the phono input. The MM configuration is shown in blue/red (left/right channels), and MC in purple/green (left/right channels). The input sweep is EQ’d with an inverted RIAA curve. The THD values for the MM configuration vary from around 0.003% (20Hz) down to just above and below 0.0002% (1kHz to 2kHz), then up to 0.0005% at 20kHz. The MC THD values were higher, although these are limited by the higher noise floor (the analyzer cannot assign a THD ratio for harmonic peaks it cannot see above the noise floor). THD ratios ranged from 0.03% (20Hz) down to 0.002% (2kHz to 20kHz).

THD ratio (unweighted) vs. output (analog)

The chart above shows THD ratios measured at the balanced outputs of the 791 as a function of output voltage for the balanced line-level input. THD values start at 0.05% at 1mVrms, down to a low of 0.00005% at 3Vrms, then a steep rise past 10Vrms to the 1% THD mark at 20Vrms.

THD+N ratio (unweighted) vs. output (analog)

The chart above shows THD+N ratios measured at the balanced outputs of the 791 as a function of output voltage for the balanced line level-input, with the volume control at maximum. THD+N values start at 0.4% at 1mVrms, down to a low of 0.00015% at 5Vrms, then a steep rise past 10Vrms to the 1% THD mark at 20Vrms.

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

The chart above shows THD ratios measured at the balanced outputs of the 791 as a function of output voltage for the digital coaxial S/PIF input, swept from -90dBFS to 0dBFS, with the volume control at maximum. Blue/red traces are for 16/44.1 data, and purple/green for 24/96. For the 16/44.1 data, THD values start at 3%, and predictably, reach their low at the maximum output voltage of about 6.5Vrms, at 0.0002%. For the 24/96 data, the right channel outperformed the left by 5-10dB, and THD ratios ranged from 0.2% down to 0.0001% at the maximum output voltage.

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

The chart above shows THD+N ratios measured at the balanced outputs of the 791 as a function of output voltage for the digital coaxial S/DPIF input, swept from -90dBFS to 0dBFS, with the volume control at maximum. Blue/red traces are for 16/44.1 data, and purple/green for 24/96. For the 16/44.1 data, THD+N values start at 20% and reach their low at the maximum output voltage of about 6.5Vrms, at 0.002%. For the 24/96 data, THD ratios ranged from 1% down to 0.0002% at the maximum output voltage.

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 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 16/44.1 data yields IMD ratios from 2% down to 0.002% at 0dBFS. The 24/96 data yields IMD ratios from 0.15% down to 0.0004% near 0dBFS.

FFT spectrum – 1kHz (XLR line-level input)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced outputs for the balanced line-level input. We see that the signal’s second harmonic, at 2kHz, is at -130dBrA, or 0.00003%, and around -115dBrA, or 0.0002%, at the third (3kHz) harmonic. The subsequent signal harmonics are at a vanishingly low -140dBrA, or 0.00001%, and below. Below 1kHz, we see two small peaks from power-supply noise artifacts at 60Hz and 120Hz. These are extraordinarily low at around -150dBrA, or 0.000003%, and it must be highlighted that they are actually from the Audio Precision’s sinewave generator, and not inherent to the 791. We can say this with confidence for two reasons: these same peaks can be seen at roughly the same amplitudes when the Audio Precision’s sine wave generator is connected directly to the in inputs of its analyzer (loopback), and, these peaks are non-existent in the digital 24/96 FFT below, where the Audio Precision’s DAC generator is used instead of its analog sine wave generator. It should also be stressed how extraordinarily low the 791’s noise floor is—quite possibly the quietest preamp we’ve ever measured. The 791 does not seem to have any correlated power-supply (60Hz and harmonics) related noise (what we would describe as “hum”). The residual A-weighted noise from the 791 (volume control set to minimum with no signal) due to non-correlated thermal noise (what we would describe as “hiss”) was measured at 1.2 uVrms, compared to the analyzer’s self-noise of 0.66uVrms. Even with the volume set to the reference position (70.5dB) for our measurements and with a 2Vrms output signal present (notched out by the analyzer), A-weighted noise was measured at 2.4uVrms. Given all of the digital circuitry inside the 791, this is an impressive feat accomplished by Simaudio.

FFT spectrum – 1kHz (RCA line-level input)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced outputs for the unbalanced line-level input. We see effectively the same results as with the balanced input FFT above, except for a slightly lower 2kHz signal harmonic peak for the left channel (-140dBrA instead of -130dBrA).

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 outputs for the coaxial digital input, sampled at 16/44.1. We see similar results in terms of the second (2kHz) and third (3kHz) signal harmonics compared to the FFTs above. The noise floor is much higher due to the 16-bit depth limitation.

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 outputs for the coaxial digital input, sampled at 24/96. We see essentially the same signal harmonic profile within the audioband as with the balanced analog FFT above. There are zero noise-related peaks to be seen above the -160dBrA noise floor.

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

Shown above is the FFT for a 1kHz -90dBFS dithered 16/44.1 input sinewave stimulus at the coaxial digital input, measured at the balanced outputs. We see the 1kHz primary signal peak, at the correct amplitude, and no signal- or noise-related harmonic peaks above the -140dBrA noise floor.

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

Shown above is the FFT for a 1kHz -90dBFS dithered 24/96 input sinewave stimulus at the coaxial digital input, measured at the balanced outputs. We see the 1kHz primary signal peak, at the correct amplitude, and signal-related harmonic peaks at 2kHz (left, -135dBrA, or 0.00002%) and 4kHz (-140dBrA, or 0.00001%). Other signal-related harmonics can be see but at the extremely low -150dBrA, or 0.000003%, level.

FFT spectrum – 1kHz (MM phono input)

Shown above is the FFT for a 1kHz input sinewave stimulus, measured at the balanced outputs across for the phono input, configured for MM (default 40dB of gain in phono stage). The dominant signal-related harmonic can be seen at 2kHz, at -115dBrA, or 0.0002%. Power-supply-related noise peaks can be seen at the -100dBrA, or 0.001%, level at 60Hz and 180Hz.

FFT spectrum – 1kHz (MC phono input)

Shown above is the FFT for a 1kHz input sinewave stimulus, measured at the balanced outputs across for the phono input, configured for MC (default 60dB of gain in phono stage). There are no visible signal-related harmonic peaks above the -110 to -120dBrA noise floor. Power-supply-related noise peaks can be seen at the -80dBrA, or 0.01%, level at 60Hz and 180Hz.

FFT spectrum – 50Hz (line-level input)

Shown above is the FFT for a 50Hz input sinewave stimulus measured at the balanced outputs for the balanced line-level input. The X axis is zoomed in from 40 Hz to 1kHz, so that peaks from noise artifacts can be directly compared against peaks from the harmonics of the signal. The most predominant (non-signal) peaks are that of the signal’s second harmonic (100Hz) at -120dBrA, or 0.0001%, and the third signal harmonic (150Hz) nearing -110dBrA, or 0.0003%. Power-supply-related peaks can be seen at 60Hz (-140dBrA or 0.00001%) and 120Hz (-150dBrA or 0.000003%), but as discussed above, these are inherent to the Audio Precision sinewave generator.

FFT spectrum – 50Hz (MM phono input)

Shown above is the FFT for a 50Hz input sinewave stimulus measured at the balanced outputs for the phono input configured for MM. The most predominant (non-signal) peaks are that of the signal’s second harmonic (100Hz) at -110dBrA, or 0.0003%, and the primary (60Hz) and third (180Hz) power-supply-noise harmonics at -100dBrA, or 0.001%.

FFT spectrum – 50Hz (MC phono input)

Shown above is the FFT for a 50Hz input sinewave stimulus measured at the balanced outputs for the phono input configured for MC. The most predominant (non-signal) peaks are that of the signal’s second harmonic (100Hz)  at -90dBrA, or 0.003%, and the primary (60Hz) and third (180Hz) power-supply noise-harmonics at -80dBrA, or 0.01%.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, line-level input)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced outputs for the balanced line-level input. The input RMS values are set at -6.02dBrA so that, if summed for a mean frequency of 18.5kHz, would yield 10W (0dBrA) into 8 ohms at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -125dBRA, or 0.00006%, while the third-order modulation products, at 17kHz and 20kHz, are at -120dBrA, or 0.0001%. This is a very clean IMD result.

Intermodulation distortion FFT (line-level input, APx 32 tone)

Shown above is the FFT of the balanced outputs of the 791 with the APx 32-tone signal applied to the input. The combined amplitude of the 32 tones is the 0dBrA reference, and corresponds to 2Vrms. The intermodulation products—i.e., the “grass” between the test tones—are distortion products from the amplifier and are around the extremely low -150dBrA, or 0.000003%, level.

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

Shown above is an FFT of the intermodulation (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced outputs for the digital coaxial input at 16/44.1. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -130dBrA, or 0.00003%, while the third-order modulation products, at 17kHz and 20kHz, are at around the same level. We also see the main aliased peaks at 25.1kHz and 26.1kHz at around -100dBrA.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, coaxial digital input, 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 outputs for the digital coaxial input at 24/96. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -130dBrA, or 0.00003%, while the third-order modulation products, at 17kHz and 20kHz, are at around the same level.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, MM phono input)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced outputs across for the phono input configured for MM. 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 around -130dBrA, or 0.00003%.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus, MC phono input)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced outputs across for the phono input configured for MC. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -100dBrA, or 0.001%, but only visible for the left channel, while the third-order modulation products, at 17kHz and 20kHz, are at around -105dBrA, or 0.0006%, but only visible at 20kHz.

Square-wave response (10kHz)

Above is the 10kHz squarewave response using the analog line-level input, at roughly 2Vrms. Due to limitations inherent to the Audio Precision APx555 B Series analyzer, this graph should not be used to infer or extrapolate the 791’s slew-rate performance. Rather, it should be seen as a qualitative representation of its extended bandwidth. An ideal squarewave can be represented as the sum of a sinewave and an infinite series of its odd-order harmonics (e.g., 10kHz + 30kHz + 50kHz + 70kHz . . .). A limited bandwidth will show only the sum of the lower-order harmonics, which may result in noticeable undershoot and/or overshoot, and softening of the edges. The 791 reproduction of the 10kHz squarewave is very clean, with only extremely mild softening in the edges.

Diego Estan
Electronics Measurement Specialist

Details

Parent Category: Products

Category: Preamplifier Measurements

Created: 15 November 2023

Link: reviewed by Jason Thorpe on SoundStage! Ultra on November 15, 2023

General information

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

The Angela-Gilbert Yeung C312 was conditioned for 30 minutes at 2Vrms at the output before any measurements were taken. All measurements were taken with both channels driven.

The C312 under test offers three sets of line-level unbalanced (RCA) inputs, two sets of line-level balanced (XLR) inputs, one set of unbalanced outputs, two set of balanced outputs, and a set of fixed line-level unbalanced outputs. There was no difference in terms of gain between unbalanced and balanced inputs, while there was a 6dB increase in terms of gain for the balanced outputs compared to the unbalanced outputs. There was effectively no difference in terms of THD and noise between balanced and unbalanced inputs and outputs; however, 1kHz FFTs are included in this report with all four i/o combinations for comparison purposes. The volume control does not have a numerical display. Based on the accuracy and non-repeatable nature of the channel deviation (table below), the volume control is a potentiometer operating in the analog domain.

The C312 is a very unusual preamp, as it offers three different adjustments on the front panels via three dials. These are labeled Warm, Tube S, and SS. Unless otherwise stated, measurements were made with the volume set to unity gain, using the XLR inputs and outputs, with a 2Vrms input, and the three control dials set to the same positions as were used by the reviewer Jason Thorpe (for the most part): Warm and Tube S at the 10 o’clock position (about 1/3 of full deflection), and SS at the 9 o’clock position (about ¼ of full deflection). The short description as to what these dials do is to control the gain of various stages in the preamp. If the dials are set to minimum, there is no usable output from the preamp with the volume at maximum, while the total gain measured from the preamp with all dials at maximum is an astonishing 52dB (in order to avoid clipping, a very small input signal of 10mVrms was applied). At the end of this report, an attempt was made to characterize the measured difference (if any) to the output signal that the dials have when adjusted.

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

Primary measurements

Our primary measurements revealed the following using the balanced line-level inputs (unless otherwise specified, assume a 1kHz sine wave, 2Vrms input and output into 200k-ohm load, 10Hz to 90kHz bandwidth):

Frequency response

In our measured frequency response (relative to 1kHz) plot above, the C312 is essentially flat within the audioband (0dB at 20Hz, less than -0.1dB at 20kHz). The C312 appears to be AC-coupled, as it yielded about -0.2dB of deviation at 5Hz. In the graph above and most of the graphs below, only a single trace may be visible. This is because the left channel (blue or purple trace) is performing identically to the right channel (red or green trace) and so they perfectly overlap, indicating that the two channels are ideally matched.

Phase response

Above is the phase response plot from 20Hz to 20kHz. The C312 does not invert polarity, and it yielded a worst-case -20 degrees of phase shift at 20kHz.

THD ratio (unweighted) vs. frequency

The plot above shows THD ratios at the output as a function of frequency (20Hz to 20kHz) for a sinewave input stimulus. The blue and red plots are for left and right into 200k ohms, while purple/green (L/R) are into 600 ohms. THD values range from 0.0003-0.0005% from 20Hz to 200Hz, then up to 0.02% at 20kHz into 200k ohms. Into a 600-ohm load, THD ratios were nearly identical, but 2-3dB higher through most of the frequency sweep.

THD ratio (unweighted) vs. output voltage

The plot above shows THD ratios measured at the output of the C312 as a function of output voltage into 200k ohms with a 1kHz input sinewave, with the volume set to maximum. At the 10mVrms level, THD values measured around 0.1%, dipping down to around 0.0004% at 5-6Vrms, followed by a rise to 0.0007% at the “knee,” at around 12Vrms. The 1% THD point is reached at 13.5Vrms. It’s also important to mention that anything above 2-4Vrms is not typically required to drive most power amps to full power.

THD+N ratio (unweighted) vs. output voltage

The plot above shows THD+N ratios measured at the output of the C312 as a function of output voltage into 200k ohms with a 1kHz input sinewave, with the volume set to maximum. At the 10mVrms level, THD+N values measured around 1%, dipping down to around 0.0015% at 12Vrms.

FFT spectrum – 1kHz (balanced in, balanced out)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the output into a 200k-ohm load, for the balanced inputs and outputs. We see that the signal’s second harmonic, at 2kHz, is at around -100dBrA, or 0.001%, while the third harmonic, at 3kHz, is much lower at -125dBrA, or 0.00006%. Higher order harmonics are non-existent above the -130dBrA noise floor. Below 1kHz, we can see power-supply-related noise peaks at the fundamental (60Hz) and second harmonic (120Hz) at -110dBrA, or 0.0003%, and higher harmonics at -115dBrA, or 0.0002%, and below.

FFT spectrum – 1kHz (unbalanced in, balanced out)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the output into a 200k-ohm load for the unbalanced inputs and balanced outputs. The FFT is essentially identical to the balanced-in/balanced-out FFT above.

FFT spectrum – 1kHz (unbalanced in, unbalanced out)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the output into a 200k-ohm load for the unbalanced inputs and outputs. Again, the FFT is essentially identical to the balanced-in/balanced-out FFT above.

FFT spectrum – 1kHz (balanced in, unbalanced out)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the output into a 200k-ohm load for the balanced inputs and unbalanced outputs. Yet again, the FFT is essentially identical to the balanced in/balanced out FFT above.

FFT spectrum – 50Hz

Shown above is the FFT for a 50Hz input sinewave stimulus measured at the output into a 200k-ohm load. The X axis is zoomed in from 40Hz to 1kHz so that peaks from noise artifacts can be directly compared against peaks from the harmonics of the signal. The most predominant non-signal peaks are from the power-supply-related noise peaks at 60/120Hz at -110dBrA, or 0.0003%. The second (100Hz) and third (150Hz) signal harmonics are very low at -125dBrA, or 0.00006%.

Intermodulation distortion FFT (18kHz + 19kHz summed stimulus)

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the output into a 200k-ohm load. The input RMS values are set at -6.02dBrA so that, if summed for a mean frequency of 18.5kHz, would yield 2Vrms (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 -120/-115dBrA, or 0.0001/0.0002%. This is a clean IMD result.

Intermodulation distortion FFT (line-level input, APx 32 tone, 24/96)

Shown above is the FFT of the speaker-level output of the C312 with the APx 32-tone signal applied to the input. The combined amplitude of the 32 tones is the 0dBrA reference, and corresponds to 10W into 8 ohms. The intermodulation products—i.e., the “grass” between the test tones—are distortion products from the preamplifier and below the -120dBrA, or 0.0001%, level. This is another clean IMD result. The peaks that reach the -110dBrA level at lower frequencies are not IMD products but power-supply-related noise peaks.

Square-wave response (10kHz)

Above is the 10kHz squarewave response at the output into 200k ohms. Due to limitations inherent to the Audio Precision APx555 B Series analyzer, this graph should not be used to infer or extrapolate the C312’s slew-rate performance. Rather, it should be seen as a qualitative representation of its relatively high bandwidth. An ideal squarewave can be represented as the sum of a sinewave and an infinite series of its odd-order harmonics (e.g., 10kHz + 30kHz + 50kHz + 70kHz . . .). A limited bandwidth will show only the sum of the lower-order harmonics, which may result in noticeable undershoot and/or overshoot, and softening of the edges. The C312’s reproduction of the 10kHz squarewave is clean, with only mild softening in the corners.

What do the Warm, Tube S, and SS control dials do?

Each dial controls the gain in different stages of the preamp. The Warm dial provides the most significant changes in gain: from -42dB to +27.8dB (with the other two dials held at the 9 o’clock position). Both the Tube S and SS dials varied the gain from about -10dB to +15dB (in each case with the other two dials held at the 9 o’clock position). The effects of changing each dial, while maintaining the other two dials at the 9 o’clock position, were explored. When varying the dial positions, we found no appreciable changes in terms of: frequency response, phase, crosstalk, and output impedance. Because each dial affects gain, we predictably found changes in terms of noise and distortion (and IMD). In terms of the dials yielding differences in noise and distortion from one dial to the other, we found the effects of varying Tube S and SS to be essentially identical, while Warm yielded more distortion, with more high frequency harmonics.

We first explored changing the dials while maintaining low distortion, and below are the 1kHz FFTs with each dial at the 3 o’clock position, while maintaining the other two dials at the 9 o’clock position. In each case, an input voltage of 1Vrms was maintained and an output of 2Vrms (using the volume control). We found that with the Warm dial set to 3 o’clock, there was more distortion with a clear peak at the third harmonic (3kHz) at -110dBrA, or 0.0003%, that was not there when Tube S and SS were set to the same position. Having the Warm dial set to the 3 o’clock position did yield less noise compared to when Tube S and SS were set to the same position; however, this may be due to having the overall volume set to a lower position. There was absolutely no difference in the 1kHz FFTs between Tube S and SS set to the 3 o’clock position.

FFT spectrum—1kHz (Warm at 3 o’clock)

FFT spectrum—1kHz (Tube S at 3 o’clock)

FFT spectrum—1kHz (SS at 3 o’clock)

We then explored changing the dials to achieve high distortion (~5% THD). This was done with a baseline of maintaining all dials at the 12 o’clock position with a 2Vrms input, and 2Vrms output, then adjusting one dial at a time to achieve 5% THD, all the while adjusting the overall volume to maintain 2Vrms at the output. We also included a scope capture to display the shape of the 1kHz waveform. In addition, we show an FFT and scope capture for the scenario where Warm is set to maximum. We found that the Warm dial yields “harder” clipping, which can be seen in the distorted peaks of the sinewaves compared to when Tube S and SS were adjusted to yield 5% THD. Once again, no differences were seen between Tube S and SS in the 5% THD scenario.

FFT spectrum—1kHz (all dials at 12 o’clock—the baseline)

FFT spectrum—1kHz (Warm causing 5% THD)

Scope—1kHz (Warm causing 5% THD)

FFT spectrum—1kHz (TUBE S causing 5% THD)

Scope—1kHz (TUBE S causing 5% THD)

FFT spectrum—1kHz (SS causing 5% THD)

Scope—1kHz (SS causing 5% THD)

FFT spectrum—1kHz (Warm at maximum)

Scope—1kHz (Warm at maximum)

Because adjusting Tube S and SS seemed to yield identical results, we explored this further by maintaining Warm at the 10 o’clock position and alternating between Tube S at maximum with SS at minimum, and vice versa, while maintaining the input at 1Vrms and the output at 2Vrms. Below you will find FFTs for a 1kHz sinewave, IMD (CCIF, 18+19kHz, 1:1) and 32-tone, as well as frequency response. We also explored (not shown) IMD (SMPTE, 60Hz+7kHz, 4:1), crosstalk, phase, and output impedance. With the exception of a very small difference in frequency response, there were absolutely no differences between Tube S and SS adjustments. With the Tube S set to maximum, there was a small dip at very low frequencies (-0.2dB at 5Hz), whereas with SS set to maximum, we measured 0dB at 5Hz. This would be inaudible.

FFT spectrum—1kHz (Tube S maximum, SS minimum)

FFT spectrum—1kHz (Tube S minimum, SS maximum)

FFT spectrum—IMD (Tube S maximum, SS minimum)

FFT spectrum—IMD (Tube S minimum, SS maximum)

FFT spectrum—32-tone (Tube S maximum, SS minimum)

FFT spectrum—32-tone (Tube S minimum, SS maximum)

Frequency response (Tube S maximum, SS minimum)

Frequency response (Tube S minimum, SS maximum)

Diego Estan
Electronics Measurement Specialist