Eversolo Audio DMP-A8 Streaming Preamplifier Measurements

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