Cambridge Audio DacMagic 200M Digital-to-Analog Converter Measurements

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

General Information

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

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

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

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

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

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

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

Published specifications vs. our primary measurements

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

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

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

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

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

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

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

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

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

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

Phase response vs. filter type (16/44.1)

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

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

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

Impulse response (Fast, Slow, Short Delay filters)

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

J-Test (coaxial input)

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

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

Diego Estan
Electronics Measurement Specialist