Arcam SA30 Integrated Amplifier-DAC

Link: reviewed by Roger Kanno on SoundStage! Hi-Fi on October 15, 2021

General Information

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

The SA30 was conditioned for 1 hour at 1/8th full rated power (~15W into 8 ohms) before any measurements were taken. All measurements were taken with both channels driven, using a 120V/20A dedicated circuit, unless otherwise stated.

Since the SA30 relies heavily on software and software updates, the following are the versions displayed through the System Info SA30 menu, as tested: Software version: v1.72, Net version: v1206, ARC version: v1.7, ARC Rx: v1.0.0.

The SA30 offers several RCA line-level analog inputs; two pairs of RCA phono inputs, for moving magnet (MM) and moving coil (MC) cartridges; RCA preamp outputs; two S/PDIF coaxial (RCA) and two S/PDIF optical (TosLink) inputs; one HDMI input; one USB digital input; Bluetooth support; two pairs of speaker-level outputs (left and right channel); and one headphone output over a 1/8” TRS connector. For the purposes of these measurements, the following inputs were evaluated: digital coaxial, analog line-level, as well as MM and MC phono.

Most measurements were made with a 2Vrms line-level analog input, 5mVrms MM input, 0.35mVrms MC input, or 0dBFS digital input. The volume control is variable from 0 to 99. The following volume settings yielded 10W into 8 ohms: 43 for analog line-level input, 62 for MM and MC inputs, and 42 for digital input. The signal-to-noise ratio (SNR) measurements were made with the same input signal values but with the volume set to achieve the rated output power of 130W into 8 ohms. For comparison, on the line-level input, a SNR measurement was also made with the volume at maximum, where only 0.19Vrms was required to achieve 130W into 8 ohms.

The SA30 will digitize incoming analog input signals by default, in order to perform Dirac Live room EQ. This can be bypassed by engaging the Analogue Direct mode. Unless otherwise specified, all analog input measurements were performed in the Analogue Direct mode. The SA30 provides seven different digital filters; unless otherwise stated, the default (apodizing) filter was used for all digital measurements.

Based on the accuracy of the left- and right-channel volume-control matching (see table below), the SA30 volume control is likely digitally controlled but in the analog domain. The SA30 offers 4dB volume steps ranging from 1 to 9, 2dB volume steps ranging from 10 to 20, 1dB volume steps ranging from 21 to 50, and 0.5dB volume steps ranging from 51 to 99. Overall range is -65dB to +44.6dB (line-level input, speaker output).

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

Published specifications vs. our primary measurements

The table below summarizes the measurements published by Arcam for the SA30 compared directly against our own. The published specifications are sourced from Arcam’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, 10W into 8 ohms and a measurement input bandwidth of 10Hz to 90kHz, and the worst-case measured result between the left and right channel.

Our primary measurements revealed the following using the line-level analog input and digital coaxial input (unless specified, assume a 1kHz sine wave at 2Vrms or 0dBFS, 10W output, 8-ohm loading, 10Hz to 90kHz bandwidth):

For the continuous dynamic power test, the SA30 was able to sustain 198W into 4 ohms using an 80Hz tone for 500ms, alternating with a signal at -10dB of the peak (19W) for 5 seconds, for 5 continuous minutes without inducing a fault or the initiation of a protective circuit. This test is meant to simulate sporadic dynamic bass peaks in music and movies. During the test, the top of the SA30 was warm to the touch, but did not cause discomfort to the touch. There was also a slight buzz that could be heard and felt during the high-power bursts.

Our primary measurements revealed the following using the MM phono-level input (unless specified, assume a 1kHz sine wave, 10W output, 8-ohm loading, 10Hz to 90kHz bandwidth):

Our primary measurements revealed the following using the MC phono-level input (unless specified, assume a 1kHz sine wave, 10W output, 8-ohm loading, 10Hz to 90kHz bandwidth):

Our primary measurements revealed the following using the analog input at the headphone output (unless specified, assume a 1kHz sine wave, 2Vrms output, 300 ohms loading, 10Hz to 90kHz bandwidth):

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

In our frequency-response plots above, measured across the speaker outputs at 10W into 8 ohms, the blue and red (left/right channels) traces are for Analogue Direct mode (i.e., the input is not digitized), while the purple and green (left/right channels) are with Analogue Direct mode disengaged (i.e., the input is digitized). In the chart above and most of the charts 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.

In Analogue Direct mode for the chart above, the SA30 is nearly flat within the audioband (20Hz to 20kHz). At the audioband extremes, the SA30 is 0dB at 20Hz and -0.2dB down at 20Hz. These data corroborate Arcam’s claim of 20Hz to 20kHz (+/-0.2dB). The -3dB point is just past 80Hz. However, when the analog input signal is digitized, the frequency response is brick-walled before 20kHz. As a result, Arcam’s claim of 20Hz to 20kHz with a +/-0.1dB deviation is not corroborated, since we measured -32dB at 20kHz and -3dB at 16kHz. There are also two +0.1dB bumps in the response around 4kHz and 9kHz.

Because of this unusual high-frequency behavior with the analog input when fed through the SA30 and digitized, especially when considering Arcam’s claim that the ADC operates at 32bit/192kHz, we brought it to the attention of the Arcam engineers. They acknowledged the issue and provided us with a fix (net version 1306), which, as of this writing, is not yet available to the public at large. With the fix, the frequency response was re-measured and yielded the following results:

Here we find the same frequency response in Analogue Direct mode, but when this mode is disengaged (i.e., the signal is digitized), the frequency response now extends well past 20kHz (-0.4dB at 20kHz), with a -3dB point at around 55kHz.

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

Above is the phase response plot from 20Hz to 20kHz for the line-level input, measured across the speaker outputs at 10W into 8 ohms. The SA30 does not invert polarity and exhibits, at worst, 20 degrees (at 20kHz) of phase shift within the audioband. 

Frequency response vs. input type (8-ohm loading, left channel only)

The chart above shows the SA30’s frequency response as a function of input type measured across the speaker outputs at 10W into 8 ohms. The green trace is the same 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, and the purple trace is for a 24/96 dithered digital input signal from 5Hz to 48kHz, while the pink trace is for a 24/192 dithered digital input signal from 5Hz to 96kHz. The behaviors for the 24/96 and 24/192 input data are identical to the digitized analog signal from the chart above: -3dB at 16kHz. Interestingly, the 16/44.1 input data yielded slightly higher bandwidth, with brick-wall behavior right at 20kHz.

This unusual high frequency behavior with 96/192kHz sample rates was also brought to the attention of the Arcam engineers. They acknowledged the issue and provided us with a fix (net version 1306), which as of this writing, is not yet available to the public. With the fix, the frequency response was re-measured and yielded the following results:

Here we find effectively the same frequency response for a 16/44.1 sampled digital input before the update; however, for 24/96 and 24/192 sampled data, the high-frequency response has been greatly improved. We now find -3dB points at 20.3kHz, 43.9kHz, and 62.7kHz for, respectively, the 16/44.1, 24/96 and 24/192 input data.

Frequency response (8-ohm loading, MM phono input)

The chart above shows the frequency response for the MM phono input with maximum deviations of about -2.5/-0.1dB (20Hz/20kHz) from 20Hz to 20kHz. What is being displayed is the deviation from the RIAA curve, where the input signal sweep is EQ’d with an inverted RIAA curve supplied by Audio Precision (i.e., zero deviation would yield a flat line at 0dB).

Frequency response (8-ohm loading, MC phono input)

The chart above shows the frequency response for the MC phono input, with maximum deviations of about -2.5/-0.2dB (20Hz/20kHz) from 20Hz to 20kHz for the left channel, while the right channel was -0.5dB at 20kHz. As you can see from the tiny ripples, there were small (0.1-0.2dB) channel-to-channel deviations through the audioband, which shouldn’t be audible.

Phase response (MM/MC input)

Above is the phase response plot from 20Hz to 20kHz for the MM phono input (the MC input performed identically), measured across the speaker outputs at 10W into 8 ohms. The SA30 does not invert polarity. For the phono input, since the RIAA equalization curve must be implemented, which ranges from +19.9dB (20Hz) to -32.6dB (90kHz), phase shift at the output is inevitable. Here we find a worst case of about -60 degrees at 200Hz and 5kHz.

Digital linearity (24/44.1 data, filters 1-7)

The chart above shows the results of a linearity test for the coaxial digital input for both 16/44.1 (blue/red channels) and 24/96 (purple/green channels) input data, measured at the line-level output of the SA30. 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. The 24/96 input data yielded near perfect results all the way down to -120dBFS, while the 16/44.1 input data were perfect from -100dBFS to 0dBFS. At -120dBFS, the 16/44.1 data were only +2dB (left) and +3dB (right) above reference.

Impulse response (16/44.1 and 24/96 data)

The plots above and below show the impulse responses for the SA30’s various digital filters, fed to the coaxial digital input, measured at the line-level output of the SA30, 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.

Above is the Apodizing (default) filter (blue), described by Arcam as: “A compromise between phase, frequency response and ringing. Its main advantage is that it removes most of the ringing that has been introduced upstream in the recording process when the original material was recorded and mastered.” Next in purple is the Linear Phase Fast Roll Off filter, described by Arcam as: “Higher and equal levels of pre and post ringing compared with linear phase slow roll off. No phase shifts and with minimal high frequency aliasing compared with slow roll off.” Next in green is the Linear Phase Slow Roll Off filter, described by Arcam as: “Low and equal levels of pre and post ringing. No phase shifts but can introduce high frequency aliasing at a higher level than linear phase fast roll off. Very high frequencies will be slightly attenuated.” In the chart below . . .

. . . in blue is the Minimum Phase Fast Roll Of filter, described by Arcam as: “No pre-ringing and the phase response varies at higher frequencies. There are significantly higher amounts of post ringing compared with the linear phase filter options.” Next in purple is the Minimum Phase Slow Roll Off filter, described by Arcam as: “No pre-ringing artefacts but can introduce phase shifts at higher frequencies. It has less post ringing than the Minimum Phase Slow Roll Off, but this is still higher than the linear phase filter options. Very high frequencies in the last half octave of the filter pass band will be slightly attenuated.” Next in green is the Corrected Minimum Phase Fast Roll Of filter, described by Arcam as: “Low pre-ringing and the phase response varies at higher frequencies. There is more post ringing compared with linear phase and apodizing filters.” And finally . . .

. . . in purple above, is the Brick Wall filter, described by Arcam as: “No phase shift, but introduces both pre and post ringing artefacts.” Our measured impulse responses of the seven filters generally corroborate Arcam’s descriptions for each filter.

J-Test (coaxial)

The chart above shows the results of the J-Test test for the coaxial digital input measured at the line-level output of the SA30. J-Test” was developed by Julian Dunn the 1990s. It is a test signal, specifically a -3dBFS undithered 12kHz square wave sampled (in this case) at 48kHz (24 bits). Since even the first odd harmonic (i.e., 36kHz) of the 12kHz square wave is removed by the bandwidth limitation of the sampling rate, we are left with a 12kHz sinewave (the main peak). In addition, an undithered 250Hz 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 sine-wave jitter by the Audio Precision, to show how well the DAC rejects jitter.

The coaxial input exhibits low-level peaks in the audioband, at -110dBrA and below. This is a reasonably good J-Test result, indicating that SA30 DAC will yield good jitter immunity.

J-Test (optical)

The chart above shows the results of the J-Test test for the optical digital input measured at the line-level output of the SA30. The optical input exhibits low-level peaks in the audioband, at -120dBrA and below. Both the coaxial and optical inputs were also tested for jitter immunity by injecting artificial sine-wave jitter at 2kHz, which would manifest as sidebands at 10kHz and 14kHz without any jitter rejection. Even with 1000ns of jitter level, which is high, no further peaks, or worsening of existing peaks, were observed.

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

The chart above shows a fast Fourier transform (FFT) of the SA30’s line-level output with white noise at -4dBFS (blue/red), and a 19.1 kHz sinewave at 0dBFS fed to the coaxial digital input, sampled at 16/44.1. The steep rolloff around 20kHz in the white-noise spectrum shows the behavior of the SA30’s default reconstruction filter (Apodizing). There are small aliased images within the audioband, at around -115dBrA between 5kHz and 10kHz. The primary aliasing signal at 25kHz is at -60dBrA, while the second and third distortion harmonics (38.2 and 57.3kHz) of the 19.1kHz tone are at -90 and -95/105dBrA (left/right channels) respectively.

RMS level vs. frequency vs. load impedance (1W, left channel only)

The chart above shows RMS level (relative to 0dBrA, which is 1W into 8 ohms or 2.83Vrms) as a function of frequency, for the analog line-level input swept from 5Hz to 100kHz. The blue plot is into an 8-ohm load, the purple is into a 4-ohm load, the pink plot is an actual speaker (Focal Chora 806, measurements can be found here), and the cyan plot is no load connected. The chart below . . .

. . . is the same but zoomed in to highlight differences. Here we find that there’s a total deviation of about 0.2dB from no load to 4-ohm loading, which is an indication of a high damping factor, or low output impedance. The maximum variation in RMS level when a real speaker was used as a load is less, deviating by just over 0.1dB within the flat portion of the curve (20Hz to 10kHz), with the lowest RMS level, which would correspond to the lowest impedance point for the load, exhibited around 200Hz, and the highest RMS level, which would correspond to the highest impedance point for the load, at around 2kHz.

THD ratio (unweighted) vs. frequency vs. output power

The chart above shows THD ratios at the output into 8 ohms as a function of frequency for a sine-wave stimulus at the analog line-level input. The blue and red plots are for left and right channels at 1W output into 8 ohms, purple/green at 10W, and pink/orange at 100W. The power was varied using the volume control. The 10W and 1W data exhibited close to the same THD values, between 0.0005% at 20Hz and up to 0.002% at 20kHz. At 100W, THD values were the lowest between 50 and 200Hz (0.005-0.006%), then increased to 0.006-0.008% at 20kHz.

THD ratio (unweighted) vs. frequency at 10W (MM and MC phono inputs)

The chart above shows THDs ratio as a function of frequency plots for the MM and MC phono inputs measured across an 8-ohm load at 10W output. The MM configuration is shown in blue/red (left/right channels), and MC in purple/green (left/right channels). The input sweep is EQ’d with an inverted RIAA curve. For both data sets, the right channel outperformed the right by 5-10dB from 300Hz to about 7kHz. The THD values for the MM configuration vary from around 0.05% (20Hz) down to 0.002% (2kHz, right channel), then up to 0.003% (20kHz). The MC THD values were higher, ranging from around 0.3% (20Hz, left channel) down to 0.01% (300Hz to 20kHz, right channel), and down to 0.003% (20kHz, left channel).

THD ratio (unweighted) vs. output power at 1kHz into 4 and 8 ohms

The chart above shows THD ratios measured at the output of the SA30 as a function of output power for the analog line level-input, for an 8-ohm load (blue/red for left/right) and a 4 ohms load (purple/green for left/right). The 8-ohm data generally outperformed the 4-ohm data by 2-5dB, with the 8-ohm data ranging from 0.003% down to 0.0001% (10-25W), then up to 0.003% at the first “knee”, then up again at the second “knee” just past 100W. The 4-ohm data first “knee” occurs at around 40W, where THD value move from 0.0005% to 0.005%, and the second “knee” is at around 150W. The 1% THD mark for the 8-ohm data is at 136W, and 200W for the 4-ohm data. Of note is that this dual “knee” behavior in the plots above may seem unusual compared to typical Class AB amps, but is characteristic of Class G amplifier topology. Class G amps provide more than one power-supply rail at different voltages and switch between them as the signal output approaches each level, optimizing power efficiency at the output transistors. Before the first “knee” is the THD behavior for the first, lower power-supply rail, above the first “knee” is the THD behavior for the second, higher voltage power-supply rail. Even above the first “knee”, at 30W to 100W (8 ohms), THD values are consistently low (~0.002%) by any amplifier standard.

THD+N ratio (unweighted) vs. output power at 1kHz into 4 and 8 ohms

The chart above shows THD+N ratios measured at the output of the SA30 as a function of output power for the analog line-level-input, for an 8-ohm load (blue/red for left/right channel) and a 4-ohm load (purple/green for left/right channels). Overall, THD+N values for the 8-ohm load before the second “knee” ranged from around 0.04% (50mW) down to about 0.002%. The 4-ohm data was similar, but 2-5dB worse.

THD ratio (unweighted) vs. frequency at 8, 4, and 2 ohms (left channel only)

The chart above shows THD ratios measured at the output of the SA30 as a function of load (8/4/2 ohms) for a constant input voltage that yielded 10W at the output into 8 ohms (and roughly 20W into 4 ohms, and 40W into 2 ohms) for the analog line-level input. The 8-ohm load is the blue trace, the 4-ohm load the purple trace, and the 2-ohm load the pink trace. We find increasing levels of THD from 8 to 4 to 2 ohms, with about a 5dB increase each time the load is halved, except around 300 to 500Hz, where the differences are smaller. Overall, even with a 2-ohm load at roughly 40W, THD values were quite low and ranged from as low as 0.0007% (300 to 500Hz) up to 0.007% at 20kHz.

FFT spectrum – 1kHz (line-level input)

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the output across an 8-ohm load at 10W for the analog line-level input. We see that the signal’s second harmonic, at 2kHz, is at -110dBrA, or 0.0003%, and around the same level at the first odd harmonic (3kHz), although the left channel is above -110dBrA, and the right channel below this level. Below 1kHz, we see very few peaks from power-supply noise, with the second (120Hz), fourth (240Hz) and sixth (360Hz) noise harmonics visible at or around -130dBrA, or 0.00003%.

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the output across an 8-ohm load at 10W for the coaxial digital input, sampled at 16/44.1. We see that the signal’s second and third harmonic, at 2 and 3kHz, like for the analog input, are around -110dBrA, or 0.0003%. Below 1kHz, because of the slightly elevated noise floor due to the 16-bit input data, power-supply noise peaks are not really noticeable.

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

Shown above is the fast Fourier transform (FFT) for a 1kHz input sine-wave stimulus, measured at the output across an 8-ohm load at 10W for the coaxial digital input, sampled at 24/96. We see effectively the same signal noise peaks as with the 16/44.1 input FFT above. With the lowered noise floor due to the higher bit depth, small power-supply related peaks can be seen at -130dBrA, or 0.00003%.

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 sine-wave stimulus at the coaxial digital input, measured at the output across an 8-ohm load. We see the 1kHz primary signal peak, at the correct amplitude, and no signal or power-supply related noise peaks.

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

Shown above is the FFT for a 1kHz -90dBFS dithered 24/96 input sine-wave stimulus at the coaxial digital input, measured at the output across an 8-ohm load. We see the 1kHz primary signal peak, at the correct amplitude, and no signal or power-supply related noise peaks.

FFT spectrum – 1kHz (MM phono input)

Shown above is the FFT for a 1kHz input sine-wave stimulus, measured at the output across an 8-ohm load at 10W for the MM phono input. We see the second signal harmonic (2kHz) dominating at around -90dBrA, or 0.003%, where the left channel is just above this level, and the right channel just below. The rest of the spectrum is dominated by noise peaks, at -75dBrA, or 0.02%, and below.

FFT spectrum – 1kHz (MC phono input)

Shown above is the FFT for a 1kHz input sine-wave stimulus, measured at the output across an 8-ohm load at 10W for the MC phono input. We see the second signal harmonic (2kHz) dominating at around -70/-80dBrA, or 0.03/0.01% (left/right channels). The rest of the spectrum is dominated by noise peaks, at -60dBrA, or 0.01%, and below.

FFT spectrum – 50Hz (line-level input)

Shown above is the FFT for a 50Hz input sine-wave stimulus measured at the output across an 8-ohm load at 10W for the analog line-level input. The X axis is zoomed in from 40Hz to 1kHz, so that peaks from noise artifacts can be directly compared against peaks from the harmonics of the signal. The most predominant (non-signal) peaks are that of the signal’s second (100Hz) and third (150Hz) harmonic at around -110dBrA, or 0.0003%. Power-supply related peaks, such as the second harmonic at 120Hz, are very low, at -130dBrA, or 0.00003%.

FFT spectrum – 50Hz (MM phono input)

Shown above is the FFT for a 50Hz input sinewave stimulus measured at the output across an 8-ohm load at 10W for the MM phono input. The X axis is zoomed in from 40Hz to 1kHz, so that peaks from noise artifacts can be directly compared against peaks from the harmonics of the signal. It is difficult to make any distinctions between non-signal peaks, as there are many. They range in level from -80dBrA, or 0.01%, and below.

FFT spectrum – 50Hz (MC phono input)

Shown above is the FFT for a 50Hz input sine-wave stimulus measured at the output across an 8-ohm load at 10W for the MC phono input. The X axis is zoomed in from 40Hz to 1kHz, so that peaks from noise artifacts can be directly compared against peaks from the harmonics of the signal. It is difficult to make any distinctions between non-signal peaks, because, as with the MM chart above, there are many. They range in level from -60dBrA, or 0.1%, and below.

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

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sine-wave stimulus tone measured at the output across an 8-ohm load at 10W for the analog line-level input. The input RMS values are set at -6.02dBrA, so that, if summed for a mean frequency of 18.5kHz, would yield 10W (0dBrA) into 8 ohms at the output. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is very low at -115dBRA, or 0.0002%, while the third-order modulation products, at 17kHz and 20kHz, are around the same level for the left channel, and right around -110dBrA, or 0.0003%, for the right channel.

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 sine-wave stimulus tone measured at the output across an 8-ohm load at 10W for the digital coaxial input at 16/44.1. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is at -110dBRA, or about 0.0003%, while the third-order modulation products, at 17kHz and 20kHz, are at -105dBrA, or 0.0006% for the right channel, and below -120dBrA, or 0.0001%, for the left channel. We also see the main aliased peaks at 25.1kHz and 26.1kHz around -65dBrA and their IMD products.

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 sine-wave stimulus tone measured at the output across an 8-ohm load at 10W for the digital coaxial input at 24/96. We find that the second-order modulation product (i.e., the difference signal of 1kHz) is virtually non-existent at -135dBRA, or about 0.00002% (right channel), while the third-order modulation products, at 17kHz and 20kHz, are at -120dBrA, or 0.0001%, and below. It’s important to note that the main reason for the lower results here is the SA30’s attenuation of signals at 18/19kHz (almost -20/-30dB down) when fed a 96kHz sampled digital signal (see frequency-response charts above).

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

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sine-wave stimulus tone measured at the output across an 8-ohm load at 10W for the MM phono input. The second-order 1kHz peak is just above -90dBrA, or 0.003%, while the third-order peaks are at -115/-100dBrA (left/right channels), or 0.0002/0.001%.

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

Shown above is an FFT of the intermodulation distortion (IMD) products for an 18kHz + 19kHz summed sine-wave stimulus tone measured at the output across an 8-ohm load at 10W for the MC phono input. The second-order 1kHz peak is just above -70/-75dBrA (left/right channels), or 0.03/0.02%, while the third order peaks are at -95/-85dBrA (left/right channels), or 0.002/0.006%.

Square-wave response (10kHz)

Above is the 10kHz square-wave response using the analog line-level input, at roughly 10W into 8 ohms. Due to limitations inherent to the Audio Precision APx555 B Series analyzer, this graph should not be used to infer or extrapolate the SA30’s slew-rate performance. Rather, it should be seen as a qualitative representation of its relatively extended bandwidth. An ideal square wave can be represented as the sum of a sine wave 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 SA30’s reproduction of the 10kHz square wave is clean, with some softening of the edges.

Damping factor vs. frequency (20Hz to 20kHz)

The final chart above is the damping factor as a function of frequency. Both channels show relatively constant damping factors across the audioband, with the left channel slightly outperforming the right. The left channel measured from around 75 down to 65 at 20kHz, while the right measured around 70 down to 60 at 20kHz.

Diego Estan
Electronics Measurement Specialist