Details

Parent Category: Products

Category: Digital-to-Analog Converter Measurements

Created: 01 June 2024

Link: reviewed by George de Sa on SoundStage! Hi-Fi on June 1, 2025

General Information

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

The Hegel Music Systems D50 was conditioned for 30 minutes at 0dBFS (volume set to 2.64Vrms out) into 200k ohms before any measurements were taken.

The D50 offers two coaxial S/PDIF digital inputs (RCA and BNC), two optical S/PDIF inputs (TosLink), one AES-EBU balanced digital input (XLR), and one USB input. There are two sets of line-level outputs (balanced XLR and unbalanced RCA). Comparisons were made between unbalanced and balanced line-level outputs, but no appreciable differences were seen in terms of THD and noise; however, 1kHz FFTs are provided for both balanced and unbalanced outputs.

The analyzer’s input bandwidth filter was set to 10Hz to 22.4kHz for all measurements, except for frequency response (DC to 1 MHz), FFTs (10Hz to 90kHz), and THD vs. Frequency (10Hz to 90kHz). The latter to capture the second and third harmonic of the 20kHz output signal.

Published specifications vs. our primary measurements

The table below summarizes the measurements published by Hegel for the D50 compared directly against our own. The published specifications are sourced from Hegel’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, 2.64Vrms at the balanced 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 coaxial input and the balanced line-level outputs (unless specified, assume a 1kHz sinewave at 0dBFS, 200k ohms 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 vs. sample rate (16/44.1, 24/96, 24/192)

The plot above shows the D50’s frequency response as a function of sample rate. The blue/red traces are for a 16-bit/44.1kHz dithered digital input signal from 5Hz to 22kHz, the purple/green traces are for a 24/96 dithered digital input signal from 5Hz to 48kHz, and finally orange/pink represents 24/192 from 5Hz to 96kHz. The behavior at low frequencies is the same for the digital input: perfectly flat down to 5Hz. The behavior at high frequencies for all three digital sample rates is brickwall-type filtering right at half the respective sample rates. The -3dB points for each sample rate are: 21, 46, and 92kHz respectively. In the graph above and most of the graphs below, only a single trace may be visible. This is because the left channel (blue, purple or orange trace) is performing identically to the right channel (red, green or pink trace), and so they perfectly overlap, indicating that the two channels are ideally matched.

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

Above are the phase-response plots from 20Hz to 20kHz for a 0dBFS input signal as a function of sample rate. The blue/red traces are for a 16-bit/44.1kHz dithered digital, the purple/green traces are for a 24/96 dithered digital input signal, and finally orange/pink represents 24/192 from 5Hz to 96kHz.  The 16/44.1 stream shows -180 degrees of phase shift at 20kHz, the 24/96 data roughly -10 degrees at 20kHz, and the 24/192 shows no phase shift within the audioband.

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

The graph above shows the results of a linearity test for the coaxial digital input (the optical input performed identically) for both 16/44.1 (blue/red) and 24/96 (purple/green) input data, measured at the balanced line-level output of the D50. For this test, 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 is a straight flat line at 0dB (i.e., the amplitude of the digital input perfectly matches the amplitude of the measured analog output). The 24/96 input data is essentially perfect down to -120dBFS, while the 16/44.1 input data performed well, only over-responding by 2.5/1dB (left/right) at -120dBFS. The 24/96 data yielded such superb results that we extended the sweep down to . . .

. . . -140dBFS. Above we see that even at -140dBFS, the D50 is only over/undershooting by less than 1 dB between -140 and -130dBFS. This is an exemplary linearity test result.

Impulse response

The graph above shows the impulse responses for a looped 24/44.1 test file that moves from digital silence to full 0dBFS (all “1”s) for one sample period then back to digital silence, fed to the coaxial digital input, measured at the balanced outputs, into a 200k ohm-load for the left channel only. We can see that d50 yields an impulse response with essentially no pre-ringing but sustained post-ringing.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone

The plot above shows a fast Fourier transform (FFT) of the D50’s balanced-line level output 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 (purple/green). The steep roll-off above 20kHz in the white-noise spectrum shows the implementation of a brickwall-type reconstruction filter. There are absolutely no imaged aliasing artifacts in the audioband above the -135dBrA noise floor. The primary aliasing signal at 25kHz is highly suppressed at -100dBrA.

J-Test (coaxial RCA input)

The plot above shows the results of the J-Test test for the coaxial digital input measured at the balanced line level output of theD50. 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 (fundamental at 250Hz) down to -170dBrA for the odd harmonics.  The test file can also be used in conjunction with artificially injected sinewave jitter by the Audio Precision, to also show how well the DAC rejects jitter.

The coaxial input shows a strong J-Test result, with peaks visible, but only below the -140dBrA level.

J-Test (optical input)

The optical input shows essentially the same J-Test result as with the coaxial RCA input.

J-Test (coaxial BNC input)

The coaxial BNC input shows essentially the same J-Test result as with the coaxial RCA input.

J-Test (AES-EBU input)

The AES-EBU input shows essentially the same J-Test result as with the rest of the inputs.

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

The plot above shows the results of the J-Test test for the coaxial digital input (the optical input behaved the same) measured at the balanced line-level output, but with an additional 10ns of 2kHz sinewave jitter injected by the APx555, which would manifest as sideband peaks at 10kHz and 14kHz without perfect jitter immunity. The results do show visible sidebands but only below the very low -140dBrA level. This is further evidence of the D50’s strong jitter immunity.

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

The plot above shows the results of the J-Test test for the coaxial digital input (the optical input behaved the same) measured at the balanced line-level output, but with an additional 100ns of 2kHz sinewave jitter injected by the APx555, which would manifest as sideband peaks at 10kHz and 14kHz without perfect jitter immunity. The results do show visible sidebands but only below the low -125dBrA level. This is further evidence of the D50’s strong jitter immunity.

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

The chart above shows THD ratios at the balanced line-level output into 200k ohms (blue/red) and 600 ohms (purple/green) as a function of frequency for a 24/96 dithered 1kHz signal at the coaxial input. The 200k and 600 ohms differ by only about 10dB throughout the audioband, which is an indication that D50’s outputs are robust and can handle loads below 1k ohms with no difficulty. THD ratios into 200k ohms were extraordinarily low, ranging from 0.00003 to 0.00007% from 20Hz to 20kHz. These values are nearing the limits of what the APx555 can measure.

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

The chart above shows THD ratios at the balanced line-level output into 200k ohms as a function of frequency for a 16/44.1 (blue/red) and a 24/96 (purple/green) dithered 1kHz signal at the coaxial input. The 24/96 data (right channel) consistently outperformed the 16/44.1 data by 15-20dB from 20Hz to about 1kHz due to the inherently higher noise floor at 16 bits (i.e., the analyzer cannot assign a THD value below the noise floor). THD ratios with 16/44.1 data ranged from 0.0002% from 20Hz to 1kHz, then down to just below 0.0001% past 10kHz.  24/96 THD ratios ranged from 0.00003-0.00007% from 20Hz to 20kHz.

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

The chart above shows THD ratios measured at the balanced output as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green). Once again, the 24/96 data outperformed the 16/44.1 data, with a THD range from 0.3% at 200uVrms to 0.00005% at 1.5-2.6Vrms, while the 16/44.1 ranged from 3% down to 0.0003% at the maximum output voltage of 2.6Vrms.

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

The chart above shows THD+N ratios measured at the balanced output as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green). The 24/96 outperformed the 16/44.1 data, with a THD+N range from 3% down to 0.0002% at 2.6Vrms, while the 16/44.1 ranged from 30% down to 0.002% at the maximum output voltage of 2.6Vrms.

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 0.06% down to 0.0005% from -15 to 0dBFS.

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 16/44.1. The second (2kHz) and third (3kHz) signal harmonics are barely visible above the noise floor at -130dBrA, or 0.00003%. There are also no power-supply noise peaks to speak of to the left of the main signal peak.

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 24/96. Due to the increased bit-depth, the noise floor is much lower compared to the 16/44.1 FFT at a very low -160dBrA. We see signal harmonics ranging from -130dBrA to below -150dBrA, or 0.00003% to 0.000003%, all the way to 20kHz (and beyond). The second (2kHz) signal harmonic shows a difference between the left (-130dBrA) and right (-140dBrA) channels. Here also, there are no power-supply noise peaks to speak of to the left of the main signal peak.

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the unbalanced output into 100k ohm for the coaxial digital input, sampled at 24/96. Other than a few very low level (-150dBrA) spurious noise peaks to the left of the main signal peak, this FFT is identical to the FFT for the balanced output.

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

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

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 24/961 at -90dBFS. We see the signal peak at the correct amplitude, and one very low-level signal harmonic peak (3kHz) at -160dBrA, or 0.000001%.

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

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced output into 200k ohms for the coaxial input at 16/44.1. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, they would yield 2.6Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz), is not visible above the noise floor at -140dBrA, or 0.00001%, and the third-order modulation products, at 17kHz and 20kHz, are at -125dBrA, or 0.00006%.

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

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced output into 200k ohms for the coaxial input at 24/96. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, they would yield 2.6Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz), is at -150/-135dBrA (left/right), or 0.000003/0.00002%, and the third-order modulation products, at 17kHz and 20kHz, are at -130dBrA, or 0.00003%.

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

Shown above is the FFT of the balanced line-level output of the D50 with the APx 32-tone signal applied to the input. The combined amplitude of the 32 tones is the 0dBFS reference, and corresponds to 2.6Vrms into 200k ohms. The intermodulation products—i.e., the “grass” between the test tones—are distortion products from the amplifier and below the absurdly low -150dBrA, or 0.000003%, level. This is a very clean IMD result.

Diego Estan
Electronics Measurement Specialist

Details

Parent Category: Products

Category: Digital-to-Analog Converter Measurements

Created: 01 April 2024

Link: reviewed by Dennis Burger on SoundStage! Simplifi on April 1, 2024

General Information

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

The CD 50n was evaluated as a fixed-output DAC and conditioned for 30 minutes at 0dBFS (2.3Vrms out) into 100k ohms before any measurements were taken.

The CD 50n offers three digital inputs: one coaxial S/PDIF (RCA), one optical S/PDIF (TosLink), and one USB. There are two sets of unbalanced (RCA) line-level outputs (fixed and variable) and one headphone output (1/4″ TRS). There is an analog volume control for the headphone output.

The CD 50n offers a few features and settings. The following are the default settings used for the coaxial input, unbalanced line-level outputs, using a 0dBFS input, unless otherwise specified:

• Line Out Level: Fixed
• Lock Range: Wide (Medium and Narrow also available)
• Filter: Filter 1 (Filter 2 also available)
• H/P Amplifier Gain: High (Mid and Low also available)

The analyzer’s input bandwidth filter was set to 10Hz–22.4kHz for all measurements, except for frequency response (DC to 1 MHz), FFTs (10Hz to 90kHz). and THD vs. Frequency (10Hz to 90kHz), with the latter to capture the second and third harmonics of the 20kHz output signal.

The CD 50n analog volume control for the headphone outputs appears to be a potentiometer. Channel-to-channel deviation proved typical for this type of volume control implementation.

Volume-control accuracy (measured at the headphone output): left-right channel tracking

Published specifications vs. our primary measurements

The table below summarizes the measurements published by Marantz for the CD 50n compared directly against our own. The published specifications are sourced from Marantz’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 as set at its maximum (DC to 1MHz), assume, unless otherwise stated, assume a fixed 2.34Vrms output (RCA) 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 coaxial input and the single-ended line-level outputs (unless specified, assume a 1kHz sinewave at 0dBFS, 100k ohms loading, 10Hz to 22.4kHz bandwidth):

Our primary measurements revealed the following using the coaxial input and the headphone output (unless specified, assume a 1kHz sinewave at 0dBFS, 300 ohms loading, 10Hz to 22.4kHz bandwidth, gain set to High):

Frequency response vs. sample rate (16/44.1, 24/96, 24/192, Filter 1)

The plot above shows the CD 50n’s frequency response as a function of sample rate. The blue/red traces are for a 16-bit/44.1kHz dithered digital input signal from 5Hz to 22kHz, the purple/green traces are for a 24/96 dithered digital input signal from 5Hz to 48kHz, and finally orange/pink represents 24/192 from 5Hz to 96kHz. The behavior at low frequencies is the same for the digital input—perfectly flat down to 5Hz. The behavior at high frequencies for all three digital sample rates is that of a shallow reconstruction filter (Filter 2 offers brickwall-type filtering). The -3dB point for each sample rate is roughly 17.5, 36.6, and 64.5kHz, respectively. In the graph above and most of the graphs below, only a single trace may be visible. This is because the left channel (blue, purple, or orange trace) is performing identically to the right channel (red, green, or pink trace), and so they perfectly overlap, indicating that the two channels are ideally matched.

Frequency response vs. filter setting (16/44.1)

The plots above show frequency response for a 0dBFS input signal sampled at 44.1kHz for Filter 1 (blue) and Filter 2 (red), into a 100k-ohm load for the left channel only. The graph is zoomed in from 1kHz to 22kHz to highlight the various responses of the filters. We can see that Filter 1 offers soft attenuation around the corner frequency, likely minimizing phase shift, with a -3dB point at 17.5 kHz, while Filter 2 is a brickwall-type filter, with a -3dB point at 20.9kHz. It’s worth pointing out that Filter 1 (the default filter) may be discernible with 16/44.1 content when compared to DACs with ruler-flat frequency responses using brickwall filters, depending on one’s age and high-frequency hearing acuity. The -1dB point is at roughly 12.5kHz.

Phase response (8-ohm loading, line-level input)

Above are the phase-response plots from 20Hz to 20kHz for the coaxial input, measured across at the unbalanced output, using both Filter 1 (blue) and Filter 2 (red) for a 16/44.1 input. We can see that the CD 50n does not invert polarity, with a worst-case phase shift of -80 degrees at 20kHz for Filter 2 (the brickwall filter). What Filter 1 loses in high-frequency response, it gains with zero phase shift in the audioband.

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

The graph above shows the results of a linearity test for the coaxial digital input (the optical input performed identically) for both 16/44.1 (blue/red) and 24/96 (purple/green) input data, measured at the unbalanced line-level output of the CD 50n. The digital input is swept with a dithered 1kHz input signal from -120dBFS to 0dBFS, and the output is analyzed by the APx555. The ideal response is a straight flat line at 0dB (i.e., the amplitude of the digital input perfectly matches the amplitude of the measured analog output). The 24/96 input data is essentially perfect down to -120dBFS, while the 16/44.1 input data performed well, only overresponding by 2.5/2dB (left/right) at -120dBFS. The 24/96 data yielded such superb results that we extended the sweep down to . . .

-140dBFS. Above we see that even at -140dBFS, the CD 50n is only overshooting by 1dB with a 24/96 signal (right channel, the left is still at 0dB). This is an exemplary linearity test result.

Impulse response (24/44.1 data, Filter 1 and Filter 2)

The graph above shows the impulse responses for a looped 24/44.1 test file that moves from digital silence to full 0dBFS (all “1”s), for one sample period then back to digital silence, fed to the coaxial digital input, measured at the unbalanced outputs into a 100k-ohm load for the left channel only. Filter 1 is blue and Filter 2 is red. We can see that Filter 1 is a simple filter with virtually no pre/post ringing. Filter 2 shows almost no pre-ringing, but significant post-ringing.

J-Test (coaxial, Lock Range Wide)

The plot above shows the results of the J-Test test for the coaxial digital input measured at the unbalanced line-level output of the CD 50n, using the Wide Lock Range setting. 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 (fundamental at 250Hz) down to -170dBrA for the odd harmonics.  The test file can also be used in conjunction with artificially injected sinewave jitter by the Audio Precision, to also show how well the DAC rejects jitter.

The coaxial input shows a good J-Test result, with two peaks at the -130dBrA level clearly flanking the 12kHz primary peak. This is an indication that the CD 50n may have good jitter immunity with the Wide setting.

J-Test (coaxial, Lock Range Wide, 100ns)

The plot above shows the results of the J-Test test for the coaxial digital input measured at the unbalanced line-level output, with an additional 100ns of 2kHz sinewave jitter injected by the APx555, which would manifest as sideband peaks at 10kHz and 14kHz without perfect jitter immunity. The results show visible sidebands at 10kHz and 14kHz, but at a very low -130dBrA. This is further evidence of the CD 50n’s strong jitter immunity using the Wide Lock Range setting.

J-Test (coaxial, Lock Range Narrow)

The plot above shows the results of the J-Test test for the coaxial digital input measured at the unbalanced line-level output of the CD 50n using the Narrow Lock Range setting. The result is identical to the Wide Lock Range setting.

J-Test (coaxial, Lock Range Narrow, 100ns)

The plot above shows the results of the J-Test test for the coaxial digital input measured at the unbalanced line-level output, with an additional 100ns of 2kHz sinewave jitter injected by the APx555, which would manifest as sideband peaks at 10kHz and 14kHz, without perfect jitter immunity. The result is very poor, with a significant increase in the noise floor. The Narrow Lock Range setting should not be used with sources that may be prone to jitter.

J-Test (coaxial, Lock Range Medium)

The plot above shows the results of the J-Test test for the coaxial digital input measured at the unbalanced line-level output of the CD 50n using the Medium Lock Range setting. The result is identical to the Wide and Narrow Lock Range settings.

J-Test (coaxial, Lock Range Medium, 100ns)

The plot above shows the results of the J-Test test for the coaxial digital input measured at the unbalanced line-level output, with an additional 100ns of 2kHz sinewave jitter injected by the APx555, which would manifest as sideband peaks at 10kHz and 14kHz, without perfect jitter immunity. The results show visible sidebands at 10kHz and 14kHz, but at an extremely low -145dBrA. This is evidence of the CD 50n’s very strong jitter immunity using the Medium Lock Range setting.

J-Test (optical, Lock Range Wide)

The plot above shows the results of the J-Test test for the optical digital input measured at the unbalanced line-level output of the CD 50n. The optical input is clearly worse than the coaxial input using the Wide Lock Range setting, with peaks as high as -105dBrA flanking the 12kHz fundamental.

J-Test (optical, Lock Range Wide, 100ns)

The plot above shows the results of the J-Test test for the optical digital input measured at the unbalanced line-level output, with an additional 100ns of 2kHz sinewave jitter injected by the APx555, which would manifest as sideband peaks at 10kHz and 14kHz without perfect jitter immunity. The results show visible sidebands at 10kHz and 14kHz, at a significant -80dBrA. This shows that the Wide Lock Range setting on the CD 50n should not be used with the optical input if the source is prone to jitter.

J-Test (optical, Lock Range Narrow)

The plot above shows the results of the J-Test test for the optical digital input measured at the unbalanced line-level output of the CD 50n using the Narrow Lock Range setting. The result is identical to the Wide Lock Range setting for the coax input.

J-Test (optical, Lock Range Narrow, 100ns)

The plot above shows the results of the J-Test test for the optical digital input measured at the unbalanced line-level output, with an additional 100ns of 2kHz sinewave jitter injected by the APx555, which would manifest as sideband peaks at 10kHz and 14kHz without perfect jitter immunity. The result is very poor, with a significant increase in the noise floor. As with the coaxial input, the Narrow Lock Range setting should not be used with sources that may be prone to jitter.

J-Test (optical, Lock Range Medium)

The plot above shows the results of the J-Test test for the optical digital input measured at the unbalanced line-level output of the CD 50n using the Medium Lock Range setting. The result is identical to the wide Lock Range setting for the coax input.

J-Test (optical, Lock Range Medium, 100ns)

The plot above shows the results of the J-Test test for the optical digital input measured at the unbalanced line-level output, with an additional 100ns of 2kHz sinewave jitter injected by the APx555, which would manifest as sideband peaks at 10kHz and 14kHz without perfect jitter immunity. The results show visible sidebands at 10kHz and 14kHz, but at an extremely low -145dBrA. This is evidence of the CD 50n’s very strong jitter immunity using the Medium Lock Range setting. For the optical input, this should be the preferred setting for sources that are prone to jitter.

Wideband FFT spectrum of white noise and 19.1kHz sine-wave tone (coaxial input, Filter 1)

The plot above shows a fast Fourier transform (FFT) of the CD 50n’s unbalanced-line level output 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 (purple/green), using the Filter 1 filter. There is a soft roll-off above 20kHz as the white-noise spectrum shows. There are low-level aliasing artifacts in the audioband at -120dBrA at 6 and 13kHz. The primary aliasing signal at 25kHz is barely suppressed at -10dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sine-wave tone (coaxial input, Filter 2)

The plot above shows a fast Fourier transform (FFT) of the CD 50n’s unbalanced line-level output 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 (purple/green), using the Filter 2 filter. There is a steep roll-off above 20kHz in the white-noise spectrum due to the brick-wall filter. There are absolutely no imaged aliasing artifacts in the audioband above the -135dBrA noise floor. The primary aliasing signal at 25kHz is at -80dBrA.

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

The chart above shows THD ratios at the unbalanced line-level output into 100k ohms (blue/red) and 600 ohms (purple/green) as a function of frequency for a 24/96 dithered 1kHz signal at the coaxial input. The 100k and 600 ohms data are very close throughout the audioband, with only a 5dB increase in THD at 20kHz into the 600-ohm load. The right channel outperformed the left channel by roughly 5dB throughout. THD ratios into 100k ohms (right channel) ranged from 0.0003% from 20Hz to 5kHz, then up to 0.0005% at 20kHz.

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

The chart above shows THD ratios at the unbalanced line-level output into 100k ohms as a function of frequency for a 16/44.1 (blue/red) and a 24/96 (purple/green) dithered 1kHz signal at the coaxial input. THD ratios are roughly equivalent between the 16/44.1 and 24/96 data, roughly between 0.0003 and 0.0005%,  with the same 5dB increase in THD between right and left channels seen in the previous graph.

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

The chart above shows THD ratios measured at the unbalanced output as a function of output voltage for the coaxial input into 100k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green). The 24/96 outperformed the 16/44.1 data at lower levels due to the increased noise floor with the lower 16-bit depth data (the analyzer cannot assign a THD ratio for peaks that do not manifest above the measured noise floor). For the 16/44.1 data, THD ratios ranged from 2% at 200uVrms, down to just below 0.0005% at the maximum output voltage of 2.34Vrms. The 24/96 THD ratios ranged from 0.1% at 200uVrms, down to the same 0.0005% at the maximum output voltage.

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

The chart above shows THD+N ratios measured at the unbalanced output as a function of output voltage for the coaxial input into 100k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green). The 24/96 outperformed the 16/44.1 data due to the increase noise floor with the lower 16-bit depth data. For the 16/44.1 data, THD+N ratios ranged from 20% at 200 uVrms, down to just 0.002% at the maximum output voltage of 2.34Vrms. The 24/96 THD+N ratios ranged from 1% at 200uVrms, down to 0.0005% 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 unbalanced output for 16/44.1 (blue/red) and 24/96 input data (purple/green), from -60dBFS to 0dBFS, for the coaxial input. 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 0.0005% from -10 to -5dBFS, then up to 0.001% at 0dBFS.

FFT spectrum – 1kHz, 16/44.1 at 0dBFS

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the unbalanced output into 100k ohms for the coaxial digital input, sampled at 16/44.1. We see signal harmonics dominate at the second (2kHz) and third (3kHz) position at roughly -110dBrA, or 0.0003%, but also visible at lower levels up to and beyond 20kHz. There is only one small power-supply noise peak—the left channel at 120Hz at -130dBFS, or 0.00003%.

FFT spectrum – 1kHz, 24/96 at 0dBFS

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the unbalanced output into 100k ohm for the coaxial digital input, sampled at 24/96. Due to the increased bit depth, the noise floor is much lower compared to the 16/44.1 FFT at a very low -150 to -160dBrA. We see signal harmonics ranging from -110dBrA to -150dBrA, or 0.0003% to 0.000003%, all the way to 20kHz (and beyond). Here again, the second (2kHz) and third (3kHz) signal harmonics dominate at roughly -110dBrA. We find power-supply-related noise peaks at the second (120Hz), fourth (240Hz), and eighth (480Hz) harmonics, at -130dBrA to -140dBrA, or 0.00003% to 0.00001%.

FFT spectrum – 1kHz, 16/44.1 at -90dBFS

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

FFT spectrum – 1kHz, 24/96 at -90dBFS

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the unbalanced output into 200k ohm for the coaxial digital input, sampled at 24/961 at -90dBFS. We see the signal peak at the correct amplitude, no signal harmonics, and the same power-supply-related noise peaks as seen in the 24/96 0dBFS FFT above.

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

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the unbalanced output into 100k ohms for the coaxial input at 16/44.1. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 2.34Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -125dBrA, or 0.00006%, and the third-order modulation products, at 17kHz and 20kHz, are at -130dBrA, or 0.00003%.

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

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the unbalanced output into 100k ohms for the coaxial input at 24/96. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 2.34Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz), is at -120dBrA, or 0.0001%, and the third-order modulation products, at 17kHz and 20kHz, are at -125dBrA, or 0.00006%.

Intermodulation distortion FFT (coaxial input, APx 32 tone, 24/96)

Shown above is the FFT of the unbalanced line-level output of the CD 50n with the APx 32-tone signal applied to the input. The combined amplitude of the 32 tones is the 0dBFS reference, and corresponds to 2.34Vrms into 100k ohms. The intermodulation products—i.e., the “grass” between the test tones—are distortion products from the amplifier and below the -140dBrA, or 0.00001%, level. This is a very clean IMD result.

Diego Estan
Electronics Measurement Specialist

Details

Parent Category: Products

Category: Digital-to-Analog Converter Measurements

Created: 01 April 2024

Link: reviewed by Phil Gold on SoundStage! Ultra on April 1, 2025

General Information

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

The EMM Labs DV2i was evaluated as a DAC and conditioned for 30 minutes at 0dBFS (volume set to maximum) into 200k ohms before any measurements were taken.

The DV2i is marketed as an integrated DAC because it includes a digital volume control and streamer section. The volume-control knob is on the front panel. The DV2i offers six digital inputs: coaxial S/PDIF (RCA), optical S/PDIF (TosLink), AES-EBU (XLR), USB, a proprietary EMM Labs input, and a network (ethernet) digital input. There are two line-level outputs: balanced (XLR) and unbalanced (RCA). Comparisons were made between unbalanced and balanced line level outputs, where no appreciable differences were seen other than the extra 6dB of signal over the balanced outputs. 1kHz FFTs are nonetheless provided for both balanced and unbalanced outputs.

There are two technologies to make note of: the D2Vi’s proprietary “adaptive” filter, intended to give ideal time-domain or frequency-domain response depending on the characteristics of the incoming signal; and the single-bit (AKA DSD) digital-to-analog converter technology, once again proprietary to EMM Labs. These provide unique performance characteristics.

The analyzer’s input bandwidth filter was set to 10Hz-22.4kHz for all measurements, except for frequency response (DC to 1 MHz), FFTs (10Hz to 90kHz), and THD vs. frequency (10Hz to 90kHz). The latter was to capture the second and third harmonic of the 20kHz output signal.

The DV2i’s digital volume control offers 100 volume steps (the user can also select “dB” instead of “1” to “100” on the display). Volume steps 0 through 12 offer 2dB increments, levels 13 through 40 yield 1dB, and 41 through 100 yield 0.5dB resolution. Channel-to-channel deviation was very good, at around 0.006-0.008dB throughout the range.

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

Published specifications vs. our primary measurements

The table below summarizes the measurements published by EMM Labs for the DV2i compared directly against our own. The published specifications are sourced from EMM Labs’s website, either directly or from the manual available for download, or a combination thereof. Assume, unless otherwise stated, 1kHz at 0dBFS into 200k ohms and a measurement input bandwidth of 10Hz to 22.4kHz:

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

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

The plot above shows the DV2i’s frequency response as a function of sample rate. The blue/red traces are for a 16-bit/44.1kHz dithered digital input signal from 5Hz to 22kHz, the purple/green traces are for a 24/96 dithered digital input signal from 5Hz to 48kHz, and finally orange/pink represents 24/192 from 5Hz to 96kHz. The behavior at low frequencies is the same for all sample rates—perfectly flat down to 5Hz. The behavior at high frequencies for all three digital sample rates is soft filtering at 24/96 and 24/192, and closer to brickwall-type filtering at 16/44.1. For all three sample rates, the responses are at -0.25dB at 20kHz. The -3dB points for each sample rate are roughly 21kHz, 41kHz, and 70kHz, respectively.

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

The graph above shows the results of a linearity test for the coaxial digital input (the optical input performed identically) for both 16/44.1 (blue/red) and 24/96 (purple/green) input data, measured at the balanced line-level output of the DV2i. For this test, 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 is a straight flat line at 0dB (i.e., the amplitude of the digital input perfectly matches the amplitude of the measured analog output). The 24/96 input data is essentially perfect down to -120dBFS, while the 16/44.1 input data performed well, only over-responding by 3dB (left/right) at -120dBFS. The 24/96 data yielded such superb results that we extended the sweep down to . . .

. . . -140dBFS. Above we see that even at -140dBFS, the DV2i is only overshooting by 2-5 dB between -140 and -130dBFS. This is an exemplary linearity-test result.

Impulse response

The graph above shows the impulse responses for a looped 24/44.1 test file that moves from digital silence to full 0dBFS (all “1”s), for one sample period then back to digital silence, fed to the coaxial digital input, measured at the balanced outputs, into a 200k ohm load for the left channel only. We can see that DV2i yields an impulse response with essentially no pre- or post-ringing behaviour, or one that emulates a non-oversampling DAC. This is a surprising result given the extended and near brickwall-type frequency response for a 16/44.1 input signal. The DV2i’s adaptive filters appear to, at least according to our tests, provide the best of both worlds: near perfect time-domain and frequency-domain response.

J-Test (coaxial input)

The plot above shows the results of the J-Test test for the coaxial digital input measured at the balanced line level output of the DV2i. 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 (fundamental at 250Hz) down to -170dBrA for the odd harmonics.  The test file can also be used in conjunction with artificially injected sinewave jitter by the Audio Precision, to also show how well the DAC rejects jitter.

The coaxial input shows a strong J-Test result, with peaks visible, but only below the -140dBrA level centered around the main signal peak.

J-Test (optical input)

The optical input shows effectively the same result as the coaxial input.

J-Test (AES-EBU input)

The AES-EBU balanced input shows basically the same result as the coaxial and optical inputs.

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

The plot above shows the results of the J-Test test for the coaxial digital input (the optical and AES-EBU inputs behaved the same) measured at the balanced line-level output, with an additional 100ns of 2kHz sinewave jitter injected by the APx555, which would manifest as sideband peaks at 10kHz and 14kHz, without perfect jitter immunity. The results show no visible sidebands. This is further evidence of the DV2i’s strong jitter immunity.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone

The plot above shows a fast Fourier transform (FFT) of the DV2i’s balanced-line level output 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 (purple/green). There is a soft roll-off above 20kHz in the white-noise spectrum. This contradicts the brickwall-type results we found in our frequency response plots at 16/44.1. The DV2i, with its adaptive filters, appears to behave differently depending on the type of signal fed to it (we assume that’s the point). Most importantly, there are absolutely no imaged aliasing artifacts in the audioband above the -135dBrA noise floor. The primary aliasing signal at 25kHz is heavily suppressed at -120dBrA. The general rise in the noise floor above 20kHz is likely due to the DV2i’s DSD processing.

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

The chart above shows THD ratios at the balanced line-level output into 200k ohms (blue/red) and 600 ohms (purple/green) as a function of frequency for a 24/96 dithered 1kHz signal at the coaxial input. The 200k and 600 ohms data are identical throughout the audioband, which is an indication that the DV2i’s outputs are robust and can handle loads below 1k ohms with no difficulty. There was an evident difference in THD ratios between the left and right channels, with the left channel outperforming the right by about 5dB from 20Hz to 2kHz. THD ratios (left channel) ranged from 0.00009% from 20Hz to 1.5kHz, then up to roughly 0.001% at 4.5kHz through to 20kHz. The higher THD ratios at higher frequencies were seen in all plots and are due to the rising high-frequency noise floor due to the DSD processing (the analyzer cannot assign a THD value below the noise floor).

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

The chart above shows THD ratios at the balanced line-level output into 200k ohms as a function of frequency for a 16/44.1 (blue/red) and a 24/96 (purple/green) dithered 1kHz signal at the coaxial input. The 24/96 data (left channel) consistently outperformed the 16/44.1 data by 10dB from 20Hz to about 2kHz due to the inherently higher noise floor at 16 bits (i.e., the analyzer cannot assign a THD value below the noise floor). THD ratios with 16/44.1 data ranged from 0.0003% from 20Hz to 2.5kHz, then up to roughly 0.001% at 4.5kHz through to 20kHz. The higher THD ratios at higher frequencies were seen in all plots and are due to the rising high-frequency noise floor because of the DSD-type processing.

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

The chart above shows THD ratios measured at the balanced outputs as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green), with the volume set to maximum. Once again, the 24/96 outperformed the 16/44.1 data, with a THD range from 1% at 300uVrms to 0.0002% at 4Vrms, while the 16/44.1 ranged from 3% down to 0.0003% at the maximum output voltage of 4Vrms.

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

The chart above shows THD+N ratios measured at the balanced outputs as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green), with the volume set to maximum. The 24/96 outperformed the 16/44.1 data, with a THD+N range from 5% down to 0.0005% at 4Vrms, while the 16/44.1 ranged from 30% down to 0.002% at the maximum output voltage of 4Vrms.

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 yielded IMD ratios from 2% down to 0.002% at 0dBFS. The 24/96 data yielded IMD ratios from 0.3% down to 0.0006% from -5 to 0dBFS.

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 16/44.1. We see signal harmonics at 2 and 3kHz. The second (2kHz) harmonic is at -125/-130dBrA, or 0.00006/0.00003%, and the third harmonic (3kHz) is at -130dBrA (right visible only), or 0.00003%. There are also no power-supply noise peaks to speak of to the left of the main signal peak.

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced output into 200k ohms for the coaxial digital input, sampled at 24/96. Due to the increased bit-depth, the noise floor is much lower compared to the 16/44.1 FFT at a low -150dBrA. We see signal harmonics ranging from -120dBrA to -140dBrA, or 0.0001% to 0.00001%, at 2/3/4/5kHz. Here also, there are no power-supply noise peaks to speak of to the left of the main signal peak. The rise in the noise floor above 20kHz is due to the DSD processing.

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the unbalanced output into 100k ohm for the coaxial digital input, sampled at 24/96. It is essentially identical to the FFT above using the balanced inputs.

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

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

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

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

FFT spectrum – 1kHz (digital input, 24/96 data at -80dBFS)

This is not a test we typically do, but because the DV2i has an onboard volume control, we wanted to show if there were differences between lowering the signal level of the analyzer versus using the volume control at its lowest level (-80dBFS). Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 24/961 at -80dBFS with the volume set to maximum but with the analyzer’s signal level reduced to output the correct level. We see the signal peak at the correct amplitude, and essentially no signal harmonics or noise peaks.

FFT spectrum – 1kHz (digital input, 24/96 data at 0dBFS with volume control set to -80dBFS, its lowest level)

This is not a test we typically do, but because the DV2i has an onboard volume control, we wanted to show if there were differences between lowering the signal level of the analyzer versus using the volume control. Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 24/961 at 0dBFS with the volume set to minimum (-80dB). We see essentially the same FFT as above with the volume at maximum and the input signal at -80dBFS. This is good evidence for EmmLabs’s claims of a transparent digital volume control.

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

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced output into 200k ohms for the coaxial input at 16/44.1. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 4Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz), is not visible above the noise floor at -135dBrA, or 0.00002%, and the third-order modulation products, at 17kHz and 20kHz, are at -135dBrA, or 0.00002%. The signals to the right of the 18kHz + 19kHz summed sinewave are presumably the result of aliasing artifacts due to the nature of the adaptive digital filter.

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

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced output into 200k ohms for the coaxial input at 24/96. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 4Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -135/-125dBrA (left/right), or 0.00002/0.00006%, and the third-order modulation products, at 17kHz and 20kHz, are at -130dBrA, or 0.00003%.

Diego Estan
Electronics Measurement Specialist

Details

Parent Category: Products

Category: Digital-to-Analog Converter Measurements

Created: 01 March 2024

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

General Information

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

The Magnetar UDP900 was evaluated as a DAC and was conditioned for 30 minutes at 0dBFS (1.95Vrms RCA out) into 100k ohms before any measurements were taken. But as mentioned below, the headphone output was also measured.

The UDP900 is a universal 4k UHD Blu-ray player. It offers one digital input (asynchronous USB) allowing for its evaluation as a DAC. It is important to note that the Audio Precision (AP) analyzer does not have a dedicated digital-audio output over USB. Audio data over USB to the device under test (DUT) is achieved via a computer (in our case a Lenovo ThinkPad X1 laptop running Windows 11) running the APx software controlling the AP analyzer. The dedicated Magnetar Windows USB driver for the UDP900 was downloaded from the Magnetar website and installed on our laptop. The driver control panel allows for the selection of 16-bit or 24-bit two-channel data. This, along with the APx software controlling the sample rate, allowed for true 16-bit/44.1kHz, 24/96, and 24/192 audio data to be sent to the UDP900 DAC. The UDP900 has seven user-selectable digital-filter options, but for the digital measurements below, the default filter, labeled Brick Wall, was used.

The UDP900 has both balanced (XLR) and unbalanced (RCA) line-level analog outputs. Typically, we find very little performance difference between both types of outputs (other than an extra 6dB of gain over balanced). In the case of the UDP900, however, noticeably more THD was measured over the balanced outputs. Further, the balanced outputs yielded 7.5dB (as opposed to the typical 6dB) more gain than the unbalanced outputs. At 0dBFS (1.95Vrms over RCA and 4.6Vrms over XLR), the balanced outputs yielded 5dB more THD at 1kHz, and a very significant 20dB more at 20kHz (graphs included in this report) than the balanced ouptuts. For this reason, unless otherwise stated, the unbalanced (RCA) analog outputs were used. The UDP900 also offers a ¼ ″ TRS headphone output, which was also evaluated.

The analyzer’s input bandwidth filter was set to 10Hz–22.4kHz for all measurements, except for frequency response (DC to 1 MHz), FFTs (10Hz to 90kHz), and THD vs frequency (10Hz to 90kHz). The latter to capture the second and third harmonics of the 20kHz output signal.

Published specifications vs. our primary measurements

The table below summarizes the measurements published by Magnetar for the UDP900 compared directly against our own. The published specifications are sourced from Magnetar’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, 1.95Vrms output (RCA) 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 USB input and the unbalanced line-level outputs (unless specified, assume a 1kHz sinewave at 0dBFS, 100k ohms loading, 10Hz to 22.4kHz bandwidth):

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

Frequency response (16/44.1, 24/96, 24/192 with Brick Wall filter)

The plot above shows the UDP900’s frequency-response (relative to 1kHz) as a function of sample rate. The blue/red traces are for a 16-bit/44.1kHz dithered digital input signal from 5Hz to 22kHz, the purple/green traces are for a 24/96 dithered digital input signal from 5Hz to 48kHz, and finally orange/pink represents 24/192 from 5Hz to 96kHz. The behavior at low frequencies is the same for all digital sample rates—perfectly flat down to 5Hz. The behavior at high frequencies for all three digital sample rates is as expected, offering sharp filtering around 22k, 48k, and 96kHz (half the respective sample rate). The -3dB point for each sample rate is roughly 20.3, 44 and 81kHz respectively. The ripples (about +/- 0.2dB) in the frequency responses at higher frequencies are real—confirmed with steady-state measurements.  In the graph above and most of the graphs below, only a single trace may be visible. This is because the left channel (blue, purple or orange trace) is performing identically to the right channel (red, green or pink trace), and so they perfectly overlap, indicating that the two channels are ideally matched.

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

The graph above shows the results of a linearity test for the digital input for both 16/44.1 (blue/red) and 24/96 (purple/green) input data, measured at the line-level unbalanced outputs of the UDP900. 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 is a straight flat line at 0dB (i.e., the amplitude of the digital input perfectly matches the amplitude of the measured analog output). The 24/96 input data is perfect down to -120dBFS, while the 16/44.1 input data began to over-shoot significantly below -90dBFS. The 24/96 data yielded such superb results that we extended the sweep down. . .

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

. . . to -140dBFS. Above we see that even at -140dBFS, the UDP900 is only overshooting by 1dB. This is an exemplary linearity test result for the 24/96 data, but somewhat poor for 16/44.1 data (a good result would be flat down to -100dBFS).

Impulse response

The graph above shows the impulse responses for a looped 24/44.1 test file that moves from digital silence to full 0dBFS (all “1”s) for one sample period then back to digital silence, fed to the digital input, measured at the unbalanced analog outputs, for the left channel only. We can see that UDP900 DAC reconstruction filter exhibits symmetrical pre/post ringing as seen in a typical sinc function.

J-Test (USB input)

The plot above shows the results of the J-Test test for the USB input measured at the unbalanced line-level output of the UDP900. 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 (fundamental at 250Hz) down to -170dBrA for the odd harmonics.  The test file can also be used in conjunction with artificially injected sinewave jitter by the Audio Precision, to also show how well the DAC rejects jitter.

The USB digital input shows an average J-Test result, with a few peaks at the -125dBrA level and below, clearly visible both near the primary 12kHz signal peak, and below 1kHz. This is an indication that the UDP900 may be sensitive to jitter.

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

The plot above shows a fast Fourier transform (FFT) of the UDP900’s line-level output 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 (purple/green). There is a steep roll-off above 20kHz in the white-noise spectrum, characteristic of a brickwall-type filter. There are no imaged aliasing artifacts in the audioband above the -130dBrA noise floor. The primary aliasing signal at 25kHz is at -100dBrA.

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

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 a 24/96 dithered 1kHz signal at the USB input. Also shown are THD ratios for the balanced output (pink/orange) into a 200k ohm load. The 100k and 600 ohms data are identical throughout the audioband, which is in indication that UDP900’s outputs are robust and can handle loads below 1k ohms with no difficultly. THD ratios from the unbalanced outputs into 100k ohms ranged from 0.005% from 20Hz to 800Hz, then up to 0.01% at 3kHz, then down to 0.006%at 20kHz. The balanced outputs yielded THD ratios from 0.005% from 20Hz to 100Hz, then a steady rise to 0.06% 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 as a function of frequency for a 16/44.1 (blue/red) and a 24/96 (purple/green) dithered 1kHz signal at the optical input. THD ratios were identical and ranged from 0.005% from 20Hz to 800Hz, then up to 0.01% at 3kHz, then down to 0.006% at 20kHz.

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

The chart above shows THD ratios measured at the line-level unbalanced output as a function of output voltage for the USB input into 100k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green). Also shown are THD ratios for the balanced output (pink/orange) into a 200k ohm load. For the unbalanced output, the 24/96 data outperformed the 16/44.1 data at low output voltages by roughly 20-30dB, with a THD range from 0.1% at 200uVrms to 0.0002% at 0.25Vrms, then up to 0.005% at the maximum output voltage of 1.95Vrms. The 16/44.1 data ranged from 10% down to 0.002% at 0.5 to 1Vrms, then up to 0.005% at 1.95Vrms. The balanced output (at 24/96) yielded slightly higher THD ratios (2-3dB) than the unbalanced output up to about 0.1Vrms. From 0.1 to 1Vrms, THD ratios over the balanced output were as much as 15dB higher than the unbalanced output.

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

The chart above shows THD+N ratios measured at the line-level unbalanced output as a function of output voltage for the USB input into 100k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green). Also shown are THD+N ratios for the balanced output (pink/orange) into a 200k ohm load. For the unbalanced output, the 24/96 data outperformed the 16/44.1 data at low output voltages by roughly 15dB, with a THD+N range from 1% at 200uVrms to 0.0015% at 0.5Vrms, then up to 0.006% at the maximum output voltage of 1.95Vrms. The 16/44.1 data ranged from 10% down to 0.003% at 1Vrms, then up to 0.006% at 1.95Vrms. The balanced output (at 24/96) yielded slightly higher THD+N ratios (2-3dB) than the unbalanced output up to about 0.2Vrms. From 0.2 to 1Vrms, THD+N ratios over the balanced output were as much as 5dB higher than the unbalanced output.

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 line-level unbalanced output into 100k ohm for the USB digital input, sampled at 16/44.1. The third (3kHz) signal harmonic dominates at -85dBrA, or 0.006%. There are also a multitude of low levels peaks from 100Hz to 20kHz just below the -120dBrA, or 0.001%, level.

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 unbalanced output into 100k ohm for the USB digital input, sampled at 24/96. We see the same signal harmonic dominate at 3kHz at -85dBrA, or 0.006%, as is seen in the 16/44.1 FFT above. We find a power-supply-related noise peak at 120Hz at -130dBrA, or 0.00003%.

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced output into 200k ohm for the USB digital input, sampled at 24/96. Compared to the FFT above with the unbalanced output, here we see clearly visible higher odd-ordered signal harmonics (5/7/9/11kHz, etc.) from -95dBRa, or 0.002%, to -130dBrA, or 0.00003%, at 20kHz.

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the line-level output into 100k ohms for the USB digital input, sampled at 16/44.1 at -90dBFS. We see the signal peak at the correct amplitude, with significant odd-ordered signal harmonics at -100dBrA, or 0.001% and below.

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the line-level output into 100k ohms for the USB digital input, sampled at 24/96 at -90dBFS. We see the signal peak at the correct amplitude, no signal harmonics, and even-ordered power-supply-related noise (120/240/360Hz) dominated by the 120Hz peak at -125dBrA, or 0.00006%.

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 line-level unbalanced 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 1% down to 0.005% at -10dBFS, then up to 0.03% at 0dBFS. The 24/96 data yields IMD ratios from 0.1% down to 0.002% at -15dBFS, then up to 0.03% at 0dBFS.

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

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the unbalanced line-level output into 100k ohms for the USB input at 16/44.1. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 1.95Vrms (0dBrA) at the output. We find it difficult to identify the second-order modulation product (i.e., the difference signal of 1kHz) amongst the array of noise peaks just below the -120dBrA, or 0.0001%, level, while the third-order modulation products, at 17kHz and 20kHz, are at -95dBrA, or 0.002%.

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

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the line-level output into 100k ohms for the USB input at 24/96. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 1.95Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz), is at -135dBrA (right channel only), or 0.00002%, and the third-order modulation products, at 17kHz and 20kHz, are at -95dBrA, or 0.002%.

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

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced line-level output into 200k ohms for the USB input at 24/96. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 4.6Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz), is at -130dBrA, or 0.00003%, and the third-order modulation products, at 17kHz and 20kHz, are at -70dBrA, or 0.03%. Again, the result for the balanced output is much worse than for the unbalanced output.

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

Shown above is the FFT of the unbalanced outputs of the UDP900 with the APx 32-tone signal applied to the input. The combined amplitude of the 32 tones is the 0dBrA reference, and corresponds to 1.95Vrms. The intermodulation products—i.e., the “grass” between the test tones—are distortion products and are around the extremely low -120dBrA, or 0.0001%, level.

Diego Estan
Electronics Measurement Specialist

Details

Parent Category: Products

Category: Digital-to-Analog Converter Measurements

Created: 01 December 2023

Link: reviewed by Matt Bonaccio on SoundStage! Hi-Fi on February 15, 2024

General Information

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

The Ferrum Audio Wandla was conditioned for 30 minutes at 0dBFS/4Vrms out (2Vrms unbalanced output) into 200k ohms before any measurements were taken.

The Wandla offers one analog line-level input (RCA) and several digital inputs (S/PDIF over coaxial RCA and optical, AES-EBU, I2S over HDMI, and asynchronous over USB). There are both unbalanced (RCA) and balanced (XLR) outputs. For the purposes of these measurements, unless otherwise stated, the following inputs were evaluated using the balanced XLR outputs: digital coaxial S/PDIF (RCA), analog unbalanced (RCA). Comparisons were made between unbalanced (RCA) and balanced (XLR) line outputs, and lower THD and an extra 6dB of gain were seen using the XLR outputs (FFTs for different configurations can be seen in this report).

Unless noted, the measurements were performed with the stock power supply. Where noted, additional measurements were performed with the optional Ferrum Audio Hypsos power supply.

The Wandla offers two different types of volume control: analog and digital. Unless otherwise stated, the analog volume control was used. Using digital inputs, at the maximum volume, the Wandla can output just over 10Vrms (XLR).

There are also a few options available through the user menu. Unless otherwise stated the following was used by default:

• Filter: HQ Gauss (also available: HQ Apod, HQ Apod MP, HQ short, ESS Lin-Ph)
• Bypass: off
• Theatre bypass: off
• Output balance: C
• Digital input trim: 0dB
• Analog input gain: 0dB (-12dB to 12dB available)

Most measurements were made with a 2Vrms line-level and 0dBFS digital input with the volume set to achieve 4Vrms at the balanced output (100 for analog, 92 for digital).  

Based on the accuracy and random results of the left/right volume channel matching (see table below), the Wandla analog volume control is likely digitally controlled in the analog domain. The Wandla offers 100 volume steps from -93dB to 5.8dB for the line-level analog inputs and balanced outputs. For a 0dBFS digital input, the XLR outputs ranged from 102uVrms (volume 1) to 10.2Vrms (volume 100). All steps are in 1.0dB 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 Ferrum Audio for the Wandla compared directly against our own. The published specifications are sourced from Ferrum’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, 4Vrms output (XLR) 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, 4Vrms output (XLR), 200kohm loading, 10Hz to 22.4kHz bandwidth):

Frequency response (line-level input)

In our measured frequency-response (relative to 1kHz) plot above, the Wandla is perfectly flat within the audioband (0dB at 20Hz and 20kHz). At the extremes, the Wandla is 0dB at 5Hz and 0dB at 205kHz (the maximum allowable frequency by the signal generator). Deviations between 5Hz and 200kHz are within +/-0.05dB. These data corroborate Ferrum’s claim of 10Hz to 200kHz (+/-0.1dB). The Wandla appears to be DC-coupled, as there is no attenuation at low frequencies, even at 5Hz. It is also appropriate to say that the Wandla is a very high-bandwidth audio device. 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 Wandla does not invert polarity and exhibits essentially no phase shift within the audioband.

Frequency response vs. input type

The chart above shows the Wandla’s frequency response (relative to 1kHz) as a function of input type. The dark 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 rate). The 44.1kHz sampled input signal exhibits typical “brickwall”-type behavior, with a -3dB point at 20.3kHz. The -3dB point for the 96kHz sampled data is at 44kHz, and 67kHz for the 192kHz sampled data.

Frequency response vs. filter type (16/44.1; HQ Gauss, HQ Apod, HQ Apod MP)

The chart above shows the frequency response (relative to 1kHz) as a function of filter type for a 16/44.1 0dBFS digital input. The blue trace is for the HQ Gauss filter, red for HQ Apod filter, and green for HQ Apod MP filter. All three behaved the same, with brickwall-type behavior and a -3dB point at 20.3kHz.

Frequency response vs. filter type (16/44.1; HQ short, ESS Lin-Ph)

The chart above shows the frequency response (relative to 1kHz) as a function of filter type for a 16/44.1 0dBFS digital input. The blue trace is for the HQ Short filter, red for ESS Lin-Ph filter. The HQ Short filter exhibits softer attenuation around the corner frequency, with a -3dB point just shy of 20kHz, while the ESS Lin-Ph filter exhibits more brickwall-type behavior with a -3dB point of 21kHz.

Phase response vs. filter type (16/44.1; HQ Gauss, HQ Apod, HQ Apod MP)

Above is the phase response plot from 20Hz to 20kHz as a function of filter type for a 16/44.1 0dBFS digital input. The blue trace is for the HQ Gauss filter, red for HQ Apod, and green for HQ Apod MP. Both the HQ Gauss and HQ Apod filters exhibit essentially no phase shift within the audioband. The HQ Apod MP filter is at -180 degrees at 17kHz.

Phase response vs. filter type (16/44.1; HQ short, ESS Lin-Ph)

Above is the phase-response plot from 20Hz to 20kHz as a function of filter type for a 16/44.1 0dBFS digital input. The blue trace is for the HQ Short filter, red for ESS Lin-Ph. Both the HQ Short and ESS Lin-Ph filters exhibit essentially no phase shift within the audioband.

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 Wandla. 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 +4.5/+2dB (left/right) at -140dBFS. This is an exceptional digital linearity test result.

Impulse response vs. filter type (24/44.1; HQ Gauss, HQ Apod, HQ Apod MP) (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 Wandla. The blue trace is for the HQ Gauss filter, red for HQ Apod and green for HQ Apod MP. The HQ Gauss and HQ Apod filters behave like typical symmetrical sinc functions. The HQ Apod MP filter shows no pre-ringing but long post-ringing.

Impulse response vs. filter type (16/44.1; HQ short, ESS Lin-Ph) (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 Wandla. The blue trace is for the HQ Short filter, red for ESS Lin-Ph. Both filters behave like typical symmetrical sinc functions, but with less pre-/post-ringing compared to the HQ Gauss and HQ Apod filters above.

J-Test (coaxial input)

The chart above shows the results of the “-Test test for the coaxial digital input measured at the line-level balanced output of the Wandla. 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 Wandla shows essentially a perfect J-Test result, with no peaks (other than signal peaks: 12kHz and 250/750Hz) visible above the -160dBrA noise floor.

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 Wandla. The results are the same as the coaxial input.

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 Wandla. The results are the same as the coaxial and optical inputs.

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 Wandla, with an additional 100ns of 2kHz sinewave jitter injected by the APx555. The results are as the pristine J-Test would predict—visible but highly suppressed (-140dBrA) peaks at the 10kHz and 14kHz positions.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (HQ Gauss filter, coaxial input)

The chart above shows a fast Fourier transform (FFT) of the Wandla’s balanced outputs with white noise at -4dBFS (blue/red) and a 19.1 kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1, using the HQ Gauss 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 audio band. The primary aliasing signal at 25kHz is also completely suppressed, while the second and third distortion harmonics (38.2, 57.3kHz) of the 19.1kHz are below -110dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (HQ Apod MP filter, coaxial input)

The chart above shows a fast Fourier transform (FFT) of the Wandla’s balanced outputs with white noise at -4dBFS (blue/red) and a 19.1 kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1, using the HQ Apod filter. We see essentially the same FFT as with the HQ Gauss filter above.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (HQ Apod MP filter, coaxial input)

The chart above shows a fast Fourier transform (FFT) of the Wandla’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 HQ Apod MP filter. We see essentially the same FFT as with the HQ Gauss and HQ Apod MP filters above.

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

The chart above shows a fast Fourier transform (FFT) of the Wandla’s balanced outputs with white noise at -4dBFS (blue/red) and a 19.1 kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1, using the HQ Short filter. Here the roll-off around 20kHz in the white-noise spectrum is shallower compared to the first three filters.  There is also a little bit of attenuation in the main peak at 19.1kHz, not quite reaching 0dBrA. There are no obvious aliased images within the audioband. The primary aliasing signal at 25kHz is also completely suppressed, while the second and third distortion harmonics (38.2, 57.3kHz) of the 19.1kHz are below -110dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (ESS Lin-Ph filter, coaxial input)

The chart above shows a fast Fourier transform (FFT) of the Wandla’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 ESS Lin-Ph filter. We see essentially the same FFT as with the HQ Short filter above.

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) for an analog 2Vrms input as a function of frequency. The 200k ohms and 600 ohms data are close but not identical. THD ratios are very low, from 0.0001% from 20Hz to 1kHz for the 200k ohm load, then rising up to 0.0004% at 20kHz. Into 600 ohms, between 20Hz and 1kHz, THD ratios were roughly 5dB higher.

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

The chart above shows THD ratios at the balanced line-level output into 200k ohms (blue/red) and 600 ohms (purple/green) for a 24/96 0dBFS input as a function of frequency. The 200k ohms and 600 ohms data are close but not identical. THD ratios are extraordinarily low, from 0.00006% to 0.0001% from 20Hz to 1kHz for the 200k ohm load, then rising up to 0.0002% at 20kHz. Into 600 ohms, between 20Hz and 2kHz, THD ratios were roughly 5dB higher.

THD ratio (unweighted) vs. frequency vs. load (digital 24/96, Hypsos power supply)

The chart above shows THD ratios at the balanced line-level output into 200k ohms (blue/red) and 600 ohms (purple/green) for a 24/96 0dBFS input as a function of frequency. This time the Hypsos power supply was used. The pink/orange traces are with the Hypsos using the maximum 30V output, while purple/green is the standard 24V, both into 600 ohms. We find that using the upgraded power supply had no effect on THD ratios into 600 ohms.

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

The chart above shows THD ratios at the balanced line-level output into 200k ohms 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, although this is due to the higher 16-bit noise floor and limited averaging times for each measurement. The long FFTs (below) show that 16/44.1 and 24/96 data yield the same THD results. THD ratios are extraordinarily low for the 24/96 data, from 0.00006% to 0.0001% from 20Hz to 1kHz for the 200k ohm load, then rising up to 0.0002% at 20kHz.

THD ratio (unweighted) vs. output (analog)

The chart above shows THD ratios measured at the balanced outputs of the Wandla as a function of output voltage for the unbalanced line-level input. THD values start at 0.1% at 1mVrms, down to a low of just below 0.0001% at 3Vrms, then a steep rise past 5Vrms 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 Wandla as a function of output voltage for the unbalanced line-level input. THD+N values start at 1% at 1mVrms, down to a low of just below 0.0003% at 5-7Vrms, then a steep rise past 5Vrms 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 Wandla 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 10Vrms, at 0.0003%. For the 24/96 data, THD ratios ranged from 0.2% down to 0.0001% at the maximum output voltage. The 16/44.1 THD ratios were higher, although this is due to the higher 16-bit noise floor and limited averaging times for each measurement. The long FFTs (below) show that 16/44.1 and 24/96 data yield the same THD results.

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 Wandla 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 predictably, reach their low at the maximum output voltage of about 10Vrms, at 0.002%. For the 24/96 data, THD ratios ranged from 1.5% 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 0.003% at 0dBFS. The 24/96 data yields IMD ratios from 0.2% down to 0.0004% near 0dBFS.

FFT spectrum – 1kHz (XLR output,  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 that the signal’s second harmonic, at 2kHz, is just below -120dBrA or 0.0001%, and -130dBrA, or 0.00003%, at the third (3kHz) harmonic. The subsequent signal harmonics are around the very low -140dBrA, or 0.00001%, level. Below 1kHz, we see a hint of a peak at 60Hz, but at -150dBrA, or 0.000003%.

FFT spectrum – 1kHz (RCA output, 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 that the main difference is that the RCA outputs yield higher THD compared to the XLR outputs, with the second (2kHz) harmonic at the -110dBrA, or 0.0003%, 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 balanced outputs for the coaxial digital input, sampled at 16/44.1. We see that both the second (2kHz) and third (3kHz) signal harmonics are just above the -130dBrA, or 0.00003%, level. 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 that both the second (2kHz) and third (3kHz) signal harmonics are just above the -130dBrA, or 0.00003%, level. With the lower noise floor due to the 24-bit depth, we can see higher-order signal harmonics just above and below the -140dBrA, or 0.00001%, level. There as zero noise-related peaks to be seen above the -155dBrA noise floor.

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

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, but using the digital volume control (set to the same value of 92 to achieve 4Vrms at the output). Although this FFT is still relatively clean and similar to the FFT above where the analog volume control was used, it’s not as clean. Signal harmonics here reach almost -120dBrA, or 0.0001%, at 3/5/7kHz. There is no difference in terms of power-supply-related noise.

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the unbalanced outputs for the coaxial digital input, sampled at 24/96. We a similar FFT compared to the 24/96 balanced output FFT, but for a higher second (2kHz) signal harmonic peak here at almost -110dBrA, or 0.0003%.

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

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, but with the external Hypsos power-supply. We can see that this FFT is essentially identical to the 24/96 FFT above using the stock power-supply and balanced output.

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 or noise related peaks above the -155dBrA noise floor.

FFT spectrum – 50Hz (line-level input)

Shown above is the FFT for a 50Hz input sinewave stimulus measured at the balanced outputs for the unbalanced 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%. A small power-supply-related peak can be seen at 60Hz at -140dBrA, or 0.00001%, but this is inherent to the Audio Precision sinewave generator.

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 unbalanced 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 4Vrms at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -120dBRA, or 0.0001%, while the third-order modulation products, at 17kHz and 20kHz, are at around -130dBrA, or 0.00003%. 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 Wandla with the APx 32-tone signal applied to the analog unbalanced input. The combined amplitude of the 32 tones is the 0dBrA reference, and corresponds to 4Vrms. 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 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 -140dBrA, or 0.00001%, noise floor, while the third-order modulation products, at 17kHz and 20kHz, are at -130dBrA, or 0.00003%.

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 -140dBrA, or 0.00001%, while the third-order modulation products, at 17kHz and 20kHz, are at -130dBrA, or 0.00003%. This is an exquisitely clean IMD result.

Square-wave response (10kHz)

Above is the 10kHz squarewave response using the analog line-level input, at roughly 4Vrms at the balanced output. Due to limitations inherent to the Audio Precision APx555 B Series analyzer, this graph should not be used to infer or extrapolate the Wandla’s slew-rate performance. Rather, it should be seen as a qualitative representation of its extremely 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 Wandla’s reproduction of the 10kHz square wave is essentially perfect, with sharp corners and no overshoot/undershoot.

Diego Estan
Electronics Measurement Specialist

Details

Parent Category: Products

Category: Digital-to-Analog Converter Measurements

Created: 15 October 2023

Link: reviewed by Roger Kanno on SoundStage! Simplifi on October 15, 2023

General Information

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

The Bluesound Node X was evaluated as a digital-to-analog converter and conditioned for 30 minutes at 0dBFS (2.1Vrms out) into 100k ohms before any measurements were taken.

The Node X offers one combination digital-optical (S/PDIF) and analog 1/8″ TRS input. There is a 1/4″ TRS headphone output on the front of the unit. There is a digital volume control for the headphone and line-level outputs. There are also tone controls and a subwoofer output that can be turned on using the accompanying BluOS app. The app also offers full bass control with adjustable low/high pass filters. For the analog input, our standard 2Vrms level was replaced with 1Vrms, because at 2Vrms, the Node X’s ADC was nearing overload and random excessive noise was observed at the output. This is consistent with the behavior we have noted with other Bluesound products.

The analyzer’s input bandwidth filter was set to 10Hz–22.4kHz for all measurements, except for frequency response (DC to 1 MHz), FFTs (10Hz to 90kHz), and THD vs. frequency (10Hz to 90kHz). The latter to capture the second and third harmonic of the 20kHz output signal.

The Node X digital volume control ranges from -80 to 0dB, in steps ranging from 1 to 4dB. Channel-to-channel deviation proved excellent, at 0.001dB throughout the range.

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

Primary measurements

Our primary measurements revealed the following using the digitall input and the line-level outputs (unless specified, assume a 1kHz sinewave at 0dBFS, 200k ohms loading, 10Hz to 22.4kHz bandwidth):

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

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

The plot above shows the Node X’s frequency response (relative to 1kHz) as a function of sample rate. The blue/red traces are for a 16-bit/44.1kHz dithered digital input signal from 5Hz to 22kHz, the purple/green traces are for a 24/96 dithered digital input signal from 5Hz to 48kHz, and finally orange/pink represents 24/192 from 5Hz to 96kHz. The cyan plot is for the analog input. It’s obvious from the response that incoming analog signals are sampled at 44.1kHz. There is also a slight roll-off (-0.3dB) from 5–10Hz that is not present for the digital input. The behavior at low frequencies is the same for all digital sample rates—perfectly flat down to 5Hz. The behavior at high frequencies for all three digital sample rates is as expected, offering sharp filtering around 22, 48, and 96kHz (half the respective sample rate), although the 24/192 data shows softer attenuation around the corner frequency. The -3dB point for each sample rate is roughly 21, 46, and 91.5kHz, respectively. In the graph above and most of the graphs below, only a single trace may be visible. This is because the left channel (blue, purple, or orange trace) is performing identically to the right channel (red, green, or pink trace), and so they perfectly overlap, indicating that the two channels are ideally matched.

Frequency response (bass and treble, 24/96)

Above are two frequency-response plots (relative to 1kHz) for the digital input (24/96), measured at the analog outputs, with the treble/balance controls set at both minimum and maximum. They show that the Node X will provide a maximum gain/cut of approximately 6dB at 20Hz and 20kHz.

Frequency response (bass management, 24/96)

Above are two frequency-response plots for the digital input (24/96), measured at the subwoofer output and left/right analog outputs, with the crossover set at 120Hz. The Node X crossover uses a slope of 18dB/octave, and the subwoofer output is flat down to 5Hz.

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

The graph above shows the results of a linearity test for the digital input for both 16/44.1 (blue/red) and 24/96 (purple/green) input data, measured at the line-level outputs of the Node X. 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 is a straight flat line at 0dB (i.e., the amplitude of the digital input perfectly matches the amplitude of the measured analog output). The 24/96 input data is essentially perfect down to -120dBFS, while the 16/44.1 input data performed well, only over-responding by 3/2dB (left/right) at -120dBFS. The 24/96 data yielded such superb results that we extended the sweep down to . . .

. . . -140dBFS. Above we see that even at -140dBFS, the Node X is only overshooting by 1 to 3dB with 24/96 data. This is an exemplary linearity test result.

Impulse response

The graph above shows the impulse responses for a looped 24/44.1 test file that moves from digital silence to full 0dBFS (all “1”s), for one sample period then back to digital silence, fed to the digital input, measured at the analog outputs, for the left channel only. We can see that Node X DAC reconstruction filter exhibits symmetrical pre- and post-ringing as seen in a typical sinc function.

J-Test (optical input)

The plot above shows the results of the J-Test test for the coaxial digital input measured at the analog outputs of the Node X. 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 (fundamental at 250Hz) down to -170dBrA for the odd harmonics. The test file can also be used in conjunction with artificially injected sinewave jitter by the Audio Precision, to also show how well the DAC rejects jitter.

The optical digital input shows an average-to-mediocre J-Test result, with several peaks at the -130dBrA level and below clearly visible throughout the audioband. This is an indication that the Node X may be sensitive to jitter.

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

The plot above shows the results of the J-Test test for the optical digital input measured at the line-level output, with an additional 100ns of 2kHz sinewave jitter injected by the APx555, which would manifest as sideband peaks at 10kHz and 14kHz, without perfect jitter immunity. The results show no visible sidebands at 10kHz and 14kHz, and essentially the same J-Test result as seen above without the injection of jitter. The Node X DAC lost sync with the signal when roughly 600ns of jitter was added to the test signal.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone

The plot above shows a fast Fourier transform (FFT) of the Node X’s line-level output with white noise at -4dBFS (blue/red) and a 19.1 kHz sinewave at 0dBFS fed to the optical digital input, sampled at 16/44.1 (purple/green). There is a steep rolloff above 20kHz in the white-noise spectrum, characteristic of a brick-wall-type filter. There are no imaged aliasing artifacts in the audioband above the -135dBrA noise floor, except for a very small peak at roughly 11kHz at -130dBrA from the left channel. The primary aliasing signal at 25kHz is at -80dBrA.

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

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 a 24/96 dithered 1kHz signal at the optical input. The 100k and 600 ohms data are extremely close throughout the audioband (3-5dB higher for the 600-ohm load at the frequency extremes), which is an in indication that Node X’s outputs are robust and can handle loads below 1k ohms with no difficultly. THD ratios into 100k ohms ranged from 0.0004% from 20Hz to 300Hz, then up to 0.01% at 20kHz.

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

The chart above shows THD ratios at the line-level output into 100k ohms as a function of frequency for a 16/44.1 (blue/red) and a 24/96 (purple/green) dithered 1kHz signal at the optical input. THD ratios were identical and ranged from 0.0004% from 20Hz to 300Hz, then up to 0.01% at 20kHz.

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

The chart above shows THD ratios measured at the line-level output as a function of output voltage for the optical input into 100k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green). The 24/96 data outperformed the 16/44.1 data at low output voltages by roughly 10dB, with a THD range from 0.5% at 200uVrms to 0.0006-0.001% at 0.5 to 2.1Vrms, while the 16/44.1 ranged from 3% down to the same 0.0006-0.001% at 0.5 to 2.1Vrms. The difference in THD ratios is owed to the lower noise floor with 24/96 data—the analyzer cannot measure/assign a THD ratio below the noise floor.

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

The chart above shows THD+N ratios measured at the line-level output as a function of output voltage for the optical input into 100k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green). The 24/96 outperformed the 16/44.1 data throughout by roughly 10dB, with a THD+N range from 6% down to 0.001% at 1–2Vrms, while the 16/44.1 ranged from 20% down to 0.002% at the maximum output voltage of 2.1Vrms.

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

The chart above shows intermodulation distortion (IMD) ratios measured at the line-level 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 0.2% down to 0.001% at 0dBFS. The difference here again is likely due to the lower noise floor with 24/96 data.

FFT spectrum – 1kHz (analog input at 1Vrms)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the line-level output into 100k ohm for the analog input, which is resampled by the Node X ADC at 16/44.1. The second (2kHz) harmonic dominates at nearly -90dBra, or 0.003%, while the third (3kHz) harmonic is at -105dBrA, or 0.0006%. There are very low-level power-supply-related noise peaks to the left of the main signal peak around the -130dBrA, or 0.00003%, level. Also visible are the 43.1kHz and 45.1kHz IMD peaks associated with the 44.1kHz sample rate.

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the line-level output into 100k ohm for the optical digital input, sampled at 16/44.1. The signal harmonic profile is similar but lower in amplitude to the FFT above, which would include artifacts of the Node X’s ADC. The second (2kHz), third (3kHz), and fifth (5kHz) signal harmonics dominate at the -100 to -110dBrA level, or 0.001 to 0.0003%. The noise floor is also lower from 10Hz to 50Hz.

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced output into 100k ohm for the optical digital input, sampled at 24/96. Due to the increased bit-depth, the noise floor is lower compared to the 16/44.1 FFT above, at a very low -150dBrA. We see signal harmonics are essentially the same as the 16/44.1 FFT above. With the lower noise floor, noise-related harmonics are easier to see, and are actually a bit higher than the 16/44.1 FFT above, reaching nearly -120dBrA, or 0.0001%.

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the line-level output into 100k ohm for the optical digital input, sampled at 16/44.1 at -90dBFS. We see the signal peak at the correct amplitude, and no signal harmonics above the noise floor within the audioband.

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the line-level output into 100k ohm for the optical digital input, sampled at 24/961 at -90dBFS. We see the signal peak at the correct amplitude, and extremely low-level signal harmonics (2/3/4kHz) at and below -150dBrA, or 0.000003%. The 60Hz power-supply fundamental peak can be seen at -135dBrA, or 0.00002%.

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

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the line-level output into 100k ohms for the optical input at 16/44.1. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 2.1Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz), is at -120dBrA, or 0.0001%, and the third-order modulation products, at 17kHz and 20kHz, are at -95dBrA, or 0.002%.

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

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the line-level output into 100k ohms for the optical input at 24/96. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 2.1Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz), is at -120dBrA, or 0.0001%, and the third-order modulation products, at 17kHz and 20kHz, are at -95dBrA, or 0.002%.

Diego Estan
Electronics Measurement Specialist

Details

Parent Category: Products

Category: Digital-to-Analog Converter Measurements

Created: 15 September 2023

Link: reviewed by Roger Kanno on SoundStage! Hi-Fi on September 15, 2023

General Information

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

The Technics SL-G700M2 was evaluated as a digital-to-analog converter via the digital inputs and conditioned for 30 minutes at 0dBFS (2.2Vrms out) into 200k ohms before any measurements were taken.

The SL-G700M2 offers three digital inputs: one coaxial S/PDIF (RCA), one optical S/PDIF, and one USB. There are two line-level outputs (balanced XLR and unbalanced RCA) and one headphone output (1/4″ TRS labelled “PHONES”. There is a digital volume control for the headphone and line-level outputs. Comparisons were made between unbalanced and balanced line level outputs, no appreciable differences were seen in terms of THD and noise, but 1kHz FFTs are provided for both balanced and unbalanced outputs.

The SL-G700M2 offers a few features and settings. The following are the default settings used for the coaxial input, balanced line-level outputs, using a 0dBFS input, unless otherwise specified:

• Analog output level: fixed
• MQA processing: off
• Coherent Processing: on (forces a dedicated reconstruction filter)
• Filter: Mode 1, Mode 2, and Mode 3 are available (when Coherent Processing is off). These were evaluated for different parameters, such as phase, frequency, and impulse response (as indicated in the graphs below).

The analyzer’s input bandwidth filter was set to 10Hz to 22.4kHz for all measurements, except for frequency response (DC to 1MHz), FFTs (10Hz to 90kHz), and THD vs frequency (10Hz to 90kHz). The latter to capture the second and third harmonics of the 20kHz output signal.

The SL-G700M2’s digital volume control ranges from -99 to 0dB, in steps of 0.5dB. Channel-to-channel deviation proved average, at around 0.19dB throughout the range.

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

Primary measurements

Our primary measurements revealed the following using the coaxial input and the balanced line-level outputs (unless specified, assume a 1kHz sinewave at 0dBFS, 200k ohms loading, 10Hz to 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 coaxial input and the headphone output (unless specified, assume a 1kHz sinewave at 0dBFS, 300 ohms loading, 10Hz to 22.4kHz bandwidth):

Frequency response vs. sample rate (16/44.1, 24/96, 24/192; Coherent Processing)

The plot above shows the SL-G700M2’s frequency response as a function of sample rate. The blue/red traces are for a 16bit/44.1kHz dithered digital input signal from 5Hz to 22kHz, the purple/green traces are for a 24/96 dithered digital input signal from 5Hz to 48kHz, and finally orange/pink represents 24/192 from 5Hz to 96kHz. The behavior at low frequencies is the same for the digital input—perfectly flat down to 5Hz. The behavior at high frequencies for all three digital sample rates is as expected, offering sharp filtering around 20k, 42k, and 82kHz (less than half the respective sample rate), although the 24/192 data shows softer attenuation around the corner frequency. The -3dB point for each sample rate is roughly 19.2, 41.7 and 81.3kHz respectively. In the graph above and most of the graphs below, only a single trace may be visible. This is because the left channel (blue, purple or orange trace) is performing identically to the right channel (red, green or pink trace), and so they perfectly overlap, indicating that the two channels are ideally matched.

Frequency response vs. filter type (16/44.1)

The plots above show frequency-response for a 0dBFS input signal sampled at 16/44.1 for Mode 1  (blue), Mode 2 (red), Mode 3 (green), and Coherent Processing (pink) filters into a 200k ohm load for the left channel only. The graph is zoomed in from 1kHz to 22kHz to highlight the various responses of the three Mode filters. We can see Mode 1 and Coherent Processing offer essentially the same frequency response, with a -3dB point at 19.2 kHz, while Mode 2 is very close, with a -3dB point at 19.7kHz. It’s worth pointing out that the “knee” for these three filters occurs just past 16kHz, a frequency many audiophiles can no longer even hear. The Mode 3 filter behaves like a typical brickwall-type filter, with a -3dB point at 21.2kHz.

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

Above are the phase-response plots from 20Hz to 20kHz for the coaxial input, measured across at the balanced output, using the Coherent Processing filter setting. The blue/red traces are for a dithered 16/44.1 input at 0dBFS, the purple/green for 24/96, and the orange/pink for 24/192. We can see that the SL-G700M2 does not invert polarity, with a worst-case phase shift of -140 degrees at 20kHz for the 16/44.1 data. Phase shift at 20kHz for the 24/96 and 24/192 input data are inconsequential, at about -5 degrees.

Phase response vs. filter (16/44.1)

Above are the phase response plots from 20Hz to 20kHz for a 0dBFS input signal sampled at 44.1kHz for the Mode 1 (blue), Mode 2 (red), Mode 3 (green), and Coherent Processing (pink) filters into a 200k ohm load for the left channel only. Predictably, the brickwall filter (Mode 3) yields the highest phase shift at around -180 degrees at 20kHz. The Mode 1 and Coherent Processing filters are identical, at -140 degrees at 20kHz, while the Mode 2 filter exhibits no phase shift throughout the audioband.

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

The graph above shows the results of a linearity test for the coaxial digital input (the optical input performed identically) for both 16/44.1 (blue/red) and 24/96 (purple/green) input data, measured at the balanced line-level output of the SL-G700M2. The digital input is swept with a dithered 1kHz input signal from -120dBFS to 0dBFS, and the output is analyzed by the APx555. The ideal response is a straight flat line at 0dB (i.e., the amplitude of the digital input perfectly matches the amplitude of the measured analog output). The 24/96 input data is essentially perfect down to -120dBFS, while the 16/44.1 input data performed well, only over-responding by 4/2dB (left/right channels) at -120dBFS. The 24/96 data yielded such superb results that we extended the sweep down to . . .

. . . -140dBFS. Above we see that even at -140dBFS, the SL-G700M2 is only undershooting by -1 to -3dB. This is an exemplary linearity-test result.

Impulse response vs. filter type (Mode 1, Mode 2, Mode 3, Coherent Processing)

The graph above shows the impulse responses for a looped 24/44.1 test file that moves from digital silence to full 0dBFS (all “1”s), for one sample period then back to digital silence, fed to the coaxial digital input, measured at the balanced outputs, for Mode 1 (blue), Mode 2 (red), Mode 3 (green), and Coherent Processing (pink) filters into a 200k ohm-load for the left channel only. We can see that the Mode 1 and Coherent processing filters are nearly identical, with minimal pre-ringing and some post-ringing. The Mode 3 filter has no pre-ringing, but significant post-ringing, while the Mode 2 filter exhibits symmetrical pre- and post-ringing, as seen in a typical sinc function.

J-Test (coaxial input)

The plot above shows the results of the J-Test test for the coaxial digital input measured at the balanced line level output of the SL-G700M2. 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 (fundamental at 250Hz) down to -170dBrA for the odd harmonics.  The test file can also be used in conjunction with artificially injected sinewave jitter by the Audio Precision, to also show how well the DAC rejects jitter.

The coaxial input shows an average to mediocre J-Test result, with two peaks at the -130dBrA level clearly visible near 11kHz and 13kHz. This is an indication that the SL-G700M2 may be sensitive to jitter.

J-Test (optical input)

The plot above shows the results of the J-Test test for the optical digital input measured at the balanced line-level output of the SL-G700M2. The optical input shows essentially the same result as the coaxial input above.

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

The plot above shows the results of the J-Test test for the coaxial digital input (the optical input behaved the same) measured at the balanced line-level output, with an additional 100ns of 2kHz sinewave jitter injected by the APx555, which would manifest as sideband peaks at 10kHz and 14kHz without perfect jitter immunity. The results show visible sidebands at 10kHz and 14kHz, but at a relatively low -125dBrA. This is further evidence of the SL-G700M2’s average jitter immunity.

J-Test (coaxial input, 2kHz sinewave jitter at 600ns)

The plot above shows the results of the J-Test test for the coaxial digital input (the optical input behaved the same) measured at the balanced line-level output, with an additional 600ns of 2kHz sinewave jitter injected by the APx555. Here sidebands are visible at 10kHz and 14kHz again, but remain relatively low at -110dBrA. With jitter above this level, the SL-G700M2 lost sync with the signal.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (Mode 1)

The plot above shows a fast Fourier transform (FFT) of the SL-G700M2’s balanced-line level output 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 (purple/green), using the Mode 1 filter setting. There is a soft roll-off above 20kHz in the white-noise spectrum. There are absolutely no imaged aliasing artifacts in the audio and above the -135dBrA noise floor. The primary aliasing signal at 25kHz is at -35dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (Mode 2)

The plot above shows a fast Fourier transform (FFT) of the SL-G700M2’s balanced-line level output 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 (purple/green), using the Mode 2 filter setting. There is a soft roll-off above 20kHz in the white-noise spectrum. There are absolutely no imaged aliasing artifacts in the audioband above the -135dBrA noise floor. The primary aliasing signal at 25kHz is at -25dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (Mode 3)

The plot above shows a fast Fourier transform (FFT) of the SL-G700M2’s balanced-line level output 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 (purple/green), using the Mode 3 filter setting. There is a sharp roll-off above 20kHz in the white-noise spectrum showing the implementation of a brickwall-type reconstruction filter. There are absolutely no imaged aliasing artifacts in the audioband above the -135dBrA noise floor. The primary aliasing signal at 25kHz is at -100dBrA.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (Coherent Processing)

The plot above shows a fast Fourier transform (FFT) of the SL-G700M2’s balanced line-level output 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 (purple/green), using the Coherent Processing filter setting. There is a soft roll-off above 20kHz in the white-noise spectrum. There are absolutely no imaged aliasing artifacts in the audioband above the -135dBrA noise floor. The primary aliasing signal at 25kHz is at -35dBrA.

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

The chart above shows THD ratios at the balanced line-level output into 200k ohms (blue/red) and 600 ohms (purple/green) as a function of frequency for a 24/96 dithered 1kHz signal at the coaxial input. The 200k and 600 ohms data are close throughout the audioband (within 10dB from 2kHz to 20kHz), which is in indication that the SL-G700M2’s outputs are robust and can handle loads below 1k ohms with no difficultly. THD ratios into 200k ohms ranged from 0.0002% from 20Hz to 500Hz, then up to 0.002% at 20kHz.

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

The chart above shows THD ratios at the balanced line-level output into 200k ohms as a function of frequency for a 16/44.1 (blue/red) and a 24/96 (purple/green) dithered 1kHz signal at the coaxial input. The 24/96 data consistently outperformed the 16/44.1 data by 5-10dB from 20Hz to about 2kHz due to the inherently higher noise floor at 16 bits (i.e., the analyzer cannot assign a THD value below the noise floor). THD ratios with 16/44.1 data range from 0.0003% from 20Hz to 2kHz, then up to 0.002% at 16kHz. THD ratios with 24/96 data range from 0.0001-0.0002% from 20Hz to 2kHz, up to 0.002% at 20kHz.

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

The chart above shows THD ratios measured at the balanced output as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green). Once again, the 24/96 outperformed the 16/44.1 data, with a THD range from 0.1% at 200uVrms to 0.0001% at 0.5 to 2Vrms, while the 16/44.1 ranged from 2% down to nearly 0.0002%.

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

The chart above shows THD+N ratios measured at the balanced output as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS at 16/44.1 (blue/red) and 24/96 (purple/green). The 24/96 outperformed the 16/44.1 data, with a THD+N range from 1% down to  0.0002% at 1.5-2Vrms, while the 16/44.1 ranged from 20% down to 0.002% at the maximum output voltage of 2.2Vrms.

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 0.1% down to 0.0005% from -10 to 0dBFS.

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 into 200k ohm for the coaxial digital input, sampled at 16/44.1. We see signal harmonics at 2/3/5kHz, with the third harmonic (3kHz) dominating at -120dBra, or 0.0001%. There are also no power-supply noise peaks to speak of to the left of the main signal peak.

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced outputs into 200k ohm for the coaxial digital input, sampled at 24/96. Due to the increased bit-depth, the noise floor is much lower compared to the 16/44.1 FFT at a very low -150 to -160dBrA. We see signal harmonics ranging from -120dBrA to -140dBrA, or 0.0001% to 0.00001%, all the way to 20kHz (and beyond). Here also, there are no power-supply noise peaks to speak of to the left of the main signal peak.

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the unbalanced outputs into 200k ohm for the coaxial digital input, sampled at 24/96. We find small differences in the signal harmonic pattern here compared to the balanced inputs above. Here the second signal harmonic (2kHz) reaches -110dBrA, or 0.0003%, compared to -130dBrA, or 0.00003%, for the balanced inputs. There are also very low-level power-supply-related (or IMD) peaks on the right channel here to the left of the signal peak, from -140 to -150dBrA, that do not show up in the balanced outputs. 

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

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

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 24/961 at -90dBFS. We see the signal peak at the correct amplitude, and extremely low-level signal harmonics (2/3/4kHz) at and below -150dBrA, or 0.000003%.

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

Shown above is an FFT of the intermodulation (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced output into 200k ohms for the coaxial input at 16/44.1. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 2.2Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz), is perhaps barely visible above the noise floor from the right channel at -130dBrA, or 0.00003%, and the third-order modulation products, at 17kHz and 20kHz, are at -120dBrA, or 0.0001%.

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

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced output into 200k ohms for the coaxial input at 24/96. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 2.2Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz), is at -130dBrA (right), or 0.00003%, and the third-order modulation products, at 17kHz and 20kHz, are at -115dBrA, or 0.0002%.

Intermodulation distortion FFT (coaxial input, APx 32 tone, 24/192)

Shown above is the FFT of the balanced line-level output of the SL-G700M2 with the APx 32-tone signal applied to the input. The combined amplitude of the 32 tones is the 0dBFS reference, and corresponds to 2.2Vrms into 200k ohms. The intermodulation products—i.e., the "grass" between the test tones—are distortion products from the amplifier and below the -145dBrA, or 0.000006%, level. This is a very clean IMD result.

Diego Estan
Electronics Measurement Specialist

Details

Parent Category: Products

Category: Digital-to-Analog Converter Measurements

Created: 15 July 2023

Link: reviewed by Phil Gold on SoundStage! Hi-Fi on July 15, 2023

General Information

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

The Musical Fidelity M6x was conditioned for 30 minutes at 0dBFS (4Vrms out) into 200k ohms before any measurements were taken.

The M6x offers five digital inputs: one coaxial S/PDIF (RCA), two optical S/PDIF (TosLink), one AES/EBU (XLR), and one USB. There are two line-level outputs (balanced XLR and unbalanced RCA) and one headphone output (1/4″ TRS). There is a digital volume control that can be engaged for both the line-level outputs and headphone output, but was left in the Fix (fixed) default setting for all measurements, with the exception of the volume tracking table. Comparisons were made between unbalanced and balanced line-level outputs for a 24/96 0dBFS input, and aside from the 6dB extra voltage using balanced, there were no differences in THD+N. In terms of input types (USB, coaxial, optical), there were no differences in terms of THD+N at 24/96.

There is a button labeled Upsample on the M6x, which, when engaged, upsamples incoming PCM data up to 192kHz to 352.8 or 384kHz (using integer multiples of the incoming sample rate). There are eight filter settings labeled 1 through 8. All measurements below, unless otherwise stated, are for the coaxial digital input, and the balanced output, using filter 1. The eight filters are described as follows in the M6x manual:

1. Linear phase fast roll off: most common filter with clean overall suppression and excellent rejection, best for music with large transients. Provides clean, crisp highs.
2. Linear phase slow roll off: low group delay – and symmetrical input response. Less ringing than linear-phase fast roll-off (LPFR). Punchier bass than LPFR, with clean highs.
3. Minimum phase fast roll off: minimal pre-ringing preferred for imaging and soundstage. No aliasing in frequency domain. Stronger bass than linear phase, clean highs.
4. Minimum phase slow roll off: non-symmetrical filter designed to minimize pre-ringing. Strong punchy bass with good transient attacks.
5. Apodizing fast roll off: a version of linear-phase fast roll off filter optimized to improve pre-ringing.
6. Hybrid fast roll off: a combination of linear-phase and minimum phase. Fast transient attack, strong punchy bass, crisp highs.
7. Brick wall: one of the earliest designs, intended for highest suppression possible, with high delay and pre-ringing. Linear phase, crisp clean highs.
8. Oversampling bypass: the oversampling FIR filter, used for the 7 above mentioned presets, is bypassed and source data is up-sampled to 352.8/384kHz.

Note: it seemed clear comparing frequency, phase, and impulse responses for 16-bit/44.1kHz input data, as well as wideband noise FFTs, between filter 8 and any other filter with “Upsample” engaged, yielded the exact same results.

The M6x volume control has no indicator on the front panel. The volume control can be engaged by pressing the Output button on the front panel for 2 to 3 seconds to change from Fix (fixed) to Var (variable) output. When headphones are plugged in, Var is automatically selected. For a 0dBFS 1kHz input signal using the full range of the volume control will yield from a minimum of about 0.1mVrms (-90dB) to 4.1Vrms (0dB) in 1dB steps at the balanced line-level outputs, and the headphone outputs. The volume control operates in the digital domain, as every step was exactly 1dB, and the channel-to-channel deviation was exactly 0.105dB at every step, throughout the range.

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

Published specifications vs. our primary measurements

The table below summarizes the measurements published by Musical Fidelity for the M6x DAC compared directly against our own. The published specifications are sourced from Musical Fidlelity’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, the coaxial digital input (24/96 1kHz sinewave at 0dBFS), the balanced line-level or unbalanced headphone outputs into 200k ohms (line-level) and 300 ohms (headphone) using a measurement input bandwidth of 10Hz to 90kHz, and the worst-case measured result between the left and right channels.

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

Our primary measurements revealed the following using the coaxial input and the headphone output (unless specified, assume a 1kHz sinewave at 0dBFS input, 2Vrms into 300 ohms, 10Hz to 90kHz bandwidth):

Frequency response vs. sample rate (16/44.1, 24/96, 24/192; filter 1)

The plot above shows the M6x frequency response (relative to 1kHz) as a function of sample rate. The blue/red traces are for a 16-bit/44.1kHz dithered digital input signal from 5Hz to 22kHz, the purple/green traces are for a 24/96 dithered digital input signal from 5Hz to 48kHz, and finally orange/pink represents 24/192 from 5Hz to 96kHz. The behavior at low frequencies is the same for the digital input—perfectly flat down to 5Hz. The behavior at high frequencies for all three digital sample rates is as expected, offering sharp filtering around 22, 48, and 96kHz (half the respective sample rate for each). The -3dB point for each sample rate is roughly 21, 45.7, and 70.7kHz, respectively. It is also obvious from the plots above that the 44.1kHz sampled input signal offers the most brickwall-type behavior, while the attenuation of the 96kHz and 192kHz sampled input signals approaching the corner frequencies (48kHz and 96kHz) is more gentle. In the graph above and most of the graphs below, only a single trace may be visible. This is because the left channel (blue, purple or orange trace) is performing identically to the right channel (red, green or pink trace), and so they perfectly overlap, indicating that the two channels are ideally matched.

Frequency response vs. filter type (16/44.1, filters 1 to 4)

The plots above show frequency-response for a 16/44.1 input, for filters 1 through 4, into a 200k ohm-load, for the left channel only. Filter 1 is in red, filter 2 in purple, filter 3 in green, and filter 4 in blue. The graph is zoomed in from 1kHz to 22kH, to highlight the various responses of the filters around the “knee” of the response. At 20kHz, filter 1 is at -0.35dB, filter 2 is at -3.78dB, filter 3 is at -0.38dB, and filter 4 is at -5.33dB.

Note: the filter characteristics are described under the General information section above. Our measured frequency responses match the descriptions provided by Musical Fidelity.

Phase response vs. filter type (16/44.1, filters 5 to 8)

The plots above show frequency-response for a 16/44.1 input, for filters 5 through 8, into a 200k ohm-load, for the left channel only. Filter 5 is in red, filter 6 in purple, filter 7 in green, and filter 8 in blue. The graph is zoomed in from 1kHz to 22kHz to highlight the various responses of the filters around the “knee” of the response. At 20kHz, filter 5 is at -0.15dB, filter 6 is at -12.60dB, filter 7 is at 4.31dB, and filter 8, which is 16/44.1 input data up-sampled to 352.8kHz and the ESS DAC oversampling filter disabled, is at -0.59dB. Of note, filter 5 yields up/down deviations in the frequency response by almost 0.5dB nearing 20kHz.

Note: the filter characteristics are described under General Information section above. Our measured frequency responses generally match the descriptions provided by Musical Fidelity.

Phase response vs. sample rate (16/44.1, 24/96, 24/192; filter 1)

Above are the absolute phase response plots (including group delay) from 20Hz to 20kHz for the coaxial input, measured at the balanced output for the left channel only. The blue/red traces are for a dithered 16/44.1 input at -20dBFS, the purple/green for 24/96, and the orange/pink for 24/192. We can see that the M6x does not invert polarity, with a worst-case phase shift of just under 80 degrees at 20kHz for the 16/44.1, and phase shift just above and below 20 degrees at 20kHz for the 24/96 and 24/192 input data, respectively.

Phase response vs. filter type (16/44.1, filters 1 to 4)

Above are the absolute phase response plots (including group delay) plots from 20Hz to 20kHz, for a 16/44.1 signal at the coaxial input, measured at the balanced output, for filters 1 through 4, into a 200k ohm-load, for the left channel only. Filter 1 is in blue, filter 2 in purple, filter 3 in orange, and filter 4 in green. We find the general trend of better or less severe phase shift for filters that have poorer frequency response (e.g., more attenuation at 20kHz), and vice-versa.

Phase response vs. filter type (16/44.1, filters 5 to 8)

Above are the absolute phase response plots (including group delay) from 20Hz to 20kHz for a 16/44.1 signal at the coaxial input, measured at the balanced output, for filters 5 through 8, into a 200k ohm-load, for the left channel only.  Filter 5 is in blue, filter 6 in purple, filter 7 in orange, and filter 8 in green. We find the general trend of better or less severe phase shift for filters that have poorer frequency response (e.g., more attenuation at 20kHz), and vice-versa.

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

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

. . . that shows the 24/96 data remained within 0.4dB of flat, which is a phenomenal result. The 16/44.1 data results are as expected, showing significant deviations below -120dBFS. It is worth highlighting that a linearity result that is flat down to -100dBFS for 16-bit input data is also exceptional.

Impulse response vs. filter type (24/44.1, filters 1 to 4)

The graph above shows the impulse responses for the first four filter types, 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. Filter 1 is in blue, filter 2 in purple, filter 3 in red, and filter 4 in green.

Note: the filter characteristics are described under General information above. Our measured impulse responses generally match the descriptions provided by Musical Fidelity.

Impulse response vs. filter type (24/44.1, filters 5 to 8)

The graph above shows the impulse responses for the first 4 filter types, 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. Filter 5 is in blue, filter 6 in purple, filter 7 in red, and filter 8 in green.

Note: the filter characteristics are described under General Information above. Our measured impulse responses generally match the descriptions provided by Musical Fidelity.

J-Test (coaxial input)

The plot above shows the results of the J-Test test for the coaxial digital input measured at the balanced line-level output of the M6x. 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 (fundamental at 250Hz) down to -170dBrA for the odd harmonics.  The test file can also be used in conjunction with artificially injected sinewave jitter by the Audio Precision, to also show how well the DAC rejects jitter.

The coaxial SPDIF input shows some of the alternating 500Hz peaks in the audioband but at very low levels, below -140dBrA, with only a few other peaks visible near -150dBrA. This is an indication that the M6x 200M should not be sensitive to jitter.

J-Test (optical input)

The optical S/PDIF input shows essentially the same result as the coax input. This is an indication that the M6x should not be sensitive to jitter. Both the coaxial and optical inputs were also tested for jitter immunity by injecting artificial sinewave jitter at 2kHz on top of the J-Test test file, which would manifest as sidebands at 10kHz and 14kHz without any jitter rejection. Jitter immunity proved exceptional, with no visible sidebands at even 1000ns of jitter level.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (filter 1)

The plot above shows a fast Fourier transform (FFT) of the M6x balanced line-level output with white noise at -4dBFS (blue/red), and a 19.1kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using filter 1. The sharp roll-off above 20kHz in the white-noise spectrum shows the implementation of the brickwall-type reconstruction filter. There are absolutely no imaged aliasing artifacts in the audioband above the -135dBrA noise floor. The primary aliasing signal at 25kHz is at -115dBrA, with subsequent harmonics of the 25kHz peak at or below this level.

Note: the filter characteristics are described under General Information above. Our measured wideband FFT spectrum of white noise and 19.1kHz sinewave tone, generally match the descriptions provided by Musical Fidelity.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (filter 2)

The plot above shows a fast Fourier transform (FFT) of the M6x balanced line-level output with white noise at -4dBFS (blue/red), and a 19.1kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using filter 2. We see a slower roll-off in the white-noise spectrum compared to filter 1, and slight attenuation of the 19.1kHz signal peak. There are absolutely no imaged aliasing artifacts in the audioband above the -135dBrA noise floor. The primary aliasing signal at 25kHz is at -30dBrA, with subsequent harmonics of the 25kHz peak at or below this level.

Note: the filter characteristics are described under General Information above. Our measured wideband FFT spectrum of white noise and 19.1kHz sinewave tone, generally match the descriptions provided by Musical Fidelity.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (filter 3)

The plot above shows a fast Fourier transform (FFT) of the M6x balanced line-level output with white noise at -4dBFS (blue/red), and a 19.1kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using filter 3. The sharp roll-off above 20kHz in the white-noise spectrum shows the implementation of the brickwall-type reconstruction filter. There are absolutely no imaged aliasing artifacts in the audio band above the -135dBrA noise floor. The primary aliasing signal at 25kHz is at -110dBrA, with subsequent harmonics of the 25kHz peak at Ir below this level.

Note: the filter characteristics are described under General information above. Our measured wideband FFT spectrum of white noise and 19.1kHz sinewave tone, generally match the descriptions provided by Musical Fidelity.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (filter 4)

The plot above shows a fast Fourier transform (FFT) of the M6x balanced line-level output with white noise at -4dBFS (blue/red), and a 19.1kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using filter 4. We see a slower roll-off in the white noise spectrum compared to filter 1, and slight attenuation of the 19.1kHz signal peak. There are absolutely no imaged aliasing artifacts in the audioband above the -135dBrA noise floor. The primary aliasing signal at 25kHz is at -30dBrA, with subsequent harmonics of the 25kHz peak at or below this level.

Note: the filter characteristics are described under General Information above. Our measured wideband FFT spectrum of white noise and 19.1kHz sinewave tone, generally match the descriptions provided by Musical Fidelity.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (filter 5)

The plot above shows a fast Fourier transform (FFT) of the M6x balanced line-level output with white noise at -4dBFS (blue/red), and a 19.1kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using filter 5. The sharp roll-off above 20kHz in the white-noise spectrum shows the implementation of the brickwall-type reconstruction filter. There are absolutely no imaged aliasing artifacts in the audio band above the -135dBrA noise floor. The primary aliasing signal at 25kHz is at -100dBrA, with subsequent harmonics of the 25kHz peak at or below this level.

Note: the filter characteristics are described under General Information above. Our measured wideband FFT spectrum of white noise and 19.1kHz sinewave tone, generally match the descriptions provided by Musical Fidelity.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (filter 6)

The plot above shows a fast Fourier transform (FFT) of the M6x balanced line-level output with white noise at -4dBFS (blue/red), and a 19.1kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using filter 6. The sharp roll-off above 20kHz in the white-noise spectrum shows the implementation of the brick-wall type reconstruction filter. There are absolutely no imaged aliasing artifacts in the audioband above the -135dBrA noise floor. The primary aliasing signal at 25kHz is at -110dBrA, with subsequent harmonics of the 25kHz peak at or below this level.

Note: the filter characteristics are described under General Information above. Our measured wideband FFT spectrum of white noise and 19.1kHz sinewave tone, generally match the descriptions provided by Musical Fidelity.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (filter 7)

The plot above shows a fast Fourier transform (FFT) of the M6x balanced line-level output with white noise at -4dBFS (blue/red), and a 19.1kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using filter 7. The sharp roll-off above 20kHz in the white noise spectrum shows the implementation of the brickwall-type reconstruction filter. There are absolutely no imaged aliasing artifacts in the audio band above the -135dBrA noise floor. The primary aliasing signal at 25kHz is at -105dBrA, with subsequent harmonics of the 25kHz peak at or below this level.

Note: the filter characteristics are described under General Information above. Our measured wideband FFT spectrum of white noise and 19.1kHz sinewave tone, generally match the descriptions provided by Musical Fidelity.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (filter 8)

The plot above shows a fast Fourier transform (FFT) of the M6x balanced line-level output with white noise at -4dBFS (blue/red), and a 19.1kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using filter 8, which bypasses the ESS DAC over-sampling filter and engages up-sampling at 352.8kHz. With the up-sampling engaged, high frequency signals exhibit digital clipping at 0dBFS, which explains the all of the harmonics seen in the plot above.

Wideband FFT spectrum of white noise and 19.1kHz sinewave tone (filter 8, -2dBFS)

The plot above shows a fast Fourier transform (FFT) of the M6x balanced line level output with white noise at -4dBFS (blue/red), and a 19.1kHz sinewave at -2dBFS (to avoid digital clipping) fed to the coaxial digital input, sampled at 16/44.1 (purple/green), using filter 8, which bypasses the ESS DAC over-sampling filter and engages up-sampling at 352.8kHz. The very slow roll-off above 20kHz in the white-noise spectrum shows the lack of a reconstruction oversampling filter. There are nonetheless, absolutely no imaged aliasing artifacts in the audio band above the -135dBrA noise floor. The primary aliasing signal at 25kHz is at -20dBrA, with subsequent harmonics of the 25kHz peak at or below this level.

Note: the filter characteristics are described under General Information above. Our measured wideband FFT spectrum of white noise and 19.1kHz sinewave tone, generally match the descriptions provided by Musical Fidelity.

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

The chart above shows THD ratios at the balanced line-level output into 200k ohms (blue/red) and 600 ohms (purple/green) as a function of frequency for a 24/96 dithered 1kHz signal at the coaxial input. The 200k and 600 ohms data are nearly identical from 100Hz to 5kHz, hovering around a very low 0.0003%. At the frequency extremes, THD increased into 600 ohms vs 200k ohms, where we see 0.0005% vs. 0.0003% at 20Hz, and 0.001% vs. 0.0003% at around 20kHz.

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

The chart above shows THD ratios at the balanced line-level output into 200k ohms as a function of frequency for a 16/44.1 (blue/red) and a 24/96 (purple/green) dithered 1kHz signal at the coaxial input. The 24/96 data consistently outperformed the 16/44.1 data by about 3-4dB, up to 2kHz, above which, both data sets performed identically. THD values for the 24/96 data were either just above, or just below, the very low threshold of 0.0002%.

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

The chart above shows THD ratios measured at the balanced output as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS with 16/44.1 (blue/red) and 24/96 (purple/green) dithered input data. Once again, the 24/96 outperformed the 16/44.1 data, with a THD range from 0.2% to nearly 0.0001% at just over 1Vrms, while the 16/44.1 ranged from 5% down to 0.0004% at 4.1Vrms. It’s important to highlight that large discrepancies in THD ratios at lower input levels are due to the inherently lower 24-bit noise floor of the 24/96 data (i.e., when no signal harmonics are measurable in an FFT, a THD value can only be assigned as low as the noise floor permits).

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

The chart above shows THD+N ratios measured at the balanced output as a function of output voltage for the coaxial input into 200k ohms from -90dBFS to 0dBFS with 16/44.1 (blue/red) and 24/96 (purple/green) dithered input data. The 24/96 outperformed the 16/44.1 data, with a THD+N range from 4% down to  0.0004%, while the 16/44.1 ranged from 50% down to 0.002% at the maximum output voltage of 4.1Vrms.

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

The chart above shows intermodulation distortion (IMD) ratios measured at the balanced output as a function of generator input level for the coaxial input into 200k ohms from -60dBFS to 0dBFS with 16/44.1 (blue/red) and 24/96 (purple/green) dithered input data. Here, the SMPTE IMD method is used, where the primary frequency (F1 = 60Hz), and the secondary frequency (F2 = 7kHz), are mixed at a ratio of 4:1. The SMPTE IMD analysis results consider the second (F2 ± F1) through the fifth (F2 ± 4xF1) modulation products. The 24/96 outperformed the 16/44.1 data, with an IMD range from 0.05% down to  0.0005% between -10 and -5dBFS, then up to about 0.0007% at 0dBFS, while the 16/44.1 ranged from 2% down to 0.002% at the maximum output voltage of 4.1Vrms at 0dBFS.

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 16/44.1. We see the third signal harmonic (3kHz) at -115dBrA, or 0.0002%, and subsequent odd harmonics (3, 5, 7, 9kHz) at levels below -120dBrA, or 0.0001%. No even signal harmonics are visible in the audioband above the -135dBrA noise floor. There are also no power-supply noise peaks to speak of to the left of the main signal peak.

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 24/96. Due to the increased bit-depth, the noise floor is much lower compared to the 16/44.1 FFT at a very low -160dBrA. We see signal harmonics ranging from -115dBrA, or 0.0002%, at 3kHz, down to -150dBrA. With the lower noise floor, we can see even order harmonics, for example at 2kHz where the peaks (left/right) are just above and below -140dBrA, or 0.00001%. Here also, there are no power-supply noise peaks to speak of to the left of the main signal peak.

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

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

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sinewave stimulus, measured at the balanced output into 200k ohm for the coaxial digital input, sampled at 24/96 at -90dBFS. We see the main peak at the correct amplitude, and hint of signal harmonic peaks within the audioband at a vanishingly low -160dBrA, or 0.000001%.

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

Shown above is an FFT of the intermodulation distortion (IMD) products for a 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced output into 200k ohms for the coaxial input at 16/44.1. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 4.1Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -130dBRA, or 0.00003%, just barely peaking above the noise floor, while the third-order modulation products, at 17kHz and 20kHz, are slightly higher, reaching -125dBrA, or 0.00006%, for the left channel (the right channel peaks are barely perceptible above the -135dBrA noise floor).

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

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sinewave stimulus tone measured at the balanced output into 200k ohms for the coaxial input at 24/96. The input dBFS values are set at -6.02dBFS so that, if summed for a mean frequency of 18.5kHz, would yield 4.1Vrms (0dBrA) at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is just below -130dBRA, or 0.00003%, while the third-order modulation products, at 17kHz and 20kHz are slightly higher, right around -130dBrA.

Diego Estan
Electronics Measurement Specialist