Measuring QSC SPA4-100 Amplifier and Understanding Driving Modes of Speakers

As I had mentioned a couple of times (see this and this posts), I drive my DIY LXmini speakers from QSC SPA4-100 power amplifier. I had chosen it because of its compact form-factor (1U half rack) and power capabilities (4 x 100W channels) that fit perfectly the LXmini use case. Finally, I've got time to do some measurements on it. While I'm very much satisfied with the sound I'm getting from this amp + speakers, there are a couple of questions I want to get an answer for:

1. What is the difference in output between the cases when unbalanced or balanced inputs are used with this amplifier.
2. Does the bridged output mode of the amp provide any improvements in THD compared to single ended mode (the effect that I've seen with Monoprice Unity amplifier).
3. How a more expensive Class D amplifier (QSC) stands in measurements against a less expensive one (Monoprice).

I decided to measure the amp in 4 Ohm output mode driving 4 Ohm and 8 Ohm loads. This corresponds to the nominal impedances of LXmini's full range driver (SEAS Prestige FU10RB H1600-04) and woofer (SEAS Prestige L16RN-SL H1480). For the loads I used wire-wound resistors attached to massive heat sinks.

Single Ended Mode

Output Power

Below is the table of results obtained by driving one channel of the amp with a 1 kHz sine signal from QuantAsylum QA401. The voltage was measured over the load using Agilent U1252B TrueRMS multimeter:

Load, Ohm	Input, dBV	Output, Vrms	Power, W
8	0, unbal	16.74	35
8	-4, bal	20.88	54.5
4	0, unbal	16.55	68.5
4	-4, bal	20.58	105.6

Trying to go above -4 dBV for a balanced input was tripping the input limiter. This is consistent with the manufacturer's specification for the input sensitivity which is +4 dBu = 1.78 dBV ~ -4 dBV of balanced input (doubling of logarithmic voltage is approx. +6 dBV increase). The gain of the amplifier for unbalanced input is 24.5 dB. In balanced mode it's slightly above 30 dB.

Output power figures are also consistent with the manufacturer's specification. Maximum output power is achieved when maximum allowed input is provided. It can also be seen that the maximum is not achievable when using unbalanced output as the input voltage is limited. This is important as miniDSP 2x4 HD only has unbalanced outputs. They are specified as having 2 Vrms = +8 dBu maximum level, so it's possible to hit the limiter when setting the output gain on miniDSP too high.

I must say that the resulting sound power from LXminis together with the subwoofer so far was enough for playing quite loud in my living room. But it's good to know that power output can be increased if I switch to balanced inputs on the amplifier.

Distortion and Frequency Response

For these measurements I hooked up QuantAsylum QA401 in parallel to resistive load. There is a caution in the amplifier manual warning against connecting any output to the ground. I suppose, trying to do that will trip the short circuit detection circuit in the amplifier. So I used differential connection instead, leaving probes ground connectors floating.

The lowest THD was achieved while driving an 8 Ohm load in 4 Ohm output mode (the picture was taken while using balanced input to the amplifier):

Note that there are two small "spikes" around the test frequency which look surprisingly similar to jitter peaks from DAC tests. I suppose, it's totally possible with Class D amplifiers as they effectively sample the input signal. Thus, small variations in the frequency of the triangle wave generator used for sampling can cause some samples to be off by a small amount. Although, there isn't much worry about that as these spikes are below -110 dB from the main signal, so they are inaudible. Harmonics and aliases also look very small compared to the main signal.

Testing IMD shows more severe distortion and strong aliases at about 60 kHz:

Looks like the antialiasing filter is "slow". Indeed, we can see that from the FR graph:

I also saw similar weak filtering on Monoprice's Class D amplifier and at that point decided that it's because it's a rather cheap model. But now I'm seeing the same on a more expensive amp. Looks like manufacturers decided to use a weak filter to avoid compromising power output. Out of curiosity I also tried measuring the frequency response with a real speaker load, hoping that the inductivity of the speaker would act as a low pass filter, but instead I've got absolutely the same graph. It's good to be aware of this issue.

Driving a 4 Ohm load in 4 Ohm mode yields slightly higher distortion figures. If for an 8 Ohm load we have THD+N 0.0074%, for a 4 Ohm load it becomes 0.0115%.

Balanced Mode

This is where things get pretty interesting. I looked up in the manual how to enable balanced mode, and found that this amplifier doesn't have a switch for that. Instead, the manual says "drive both inputs at the same level, connect the positive terminal of Output 1 and the negative terminal of Output 2 to the load":

This forced me to pause and think a bit about what does that mean for Channels 2 and 4 in non-bridged mode. Since there is no switch for the bridged mode, the amplifier always works the same way regardless of whether we use it for driving two channels in single ended mode, or one channel in bridged mode. For Channels 1 & 3 this doesn't cause any issue—the positive wire of the output gets driven by the amplifier. But what about Channels 2 & 4? It seems that they must be driven via the negative wire of the output and in an inverted phase. Is that true? To answer that, first I connected QA401 left and right inputs to both ends of the load connected to Channel 1, L (blue) to "+", R (red) to "-":

We see a natural voltage drop across the resistive load confirming that the amplifier only drives the "+" wire. What about Channel 2 (connections are done the same was as for Channel 1):

Yes, it's completely opposite—the "-" wire is active! For checking signal phase I connected the left input of QA401 to "+" of Channel 1, and the right input to "+" of Channel 2. Since the positive wire of Channel 2 receives attenuated signal, I adjusted the attenuator on Channel 1 to make the levels to be similar:

In time domain, we can see that Channel 1 and Channel 2 are driven in opposite phases.

Wait, does it mean that Channels 2 and 4 have inverted polarity when the amplifier is used in single ended mode? Actually no, because speakers are differential devices. I'll talk about this later. Just in order to verify that the polarity is correct, I connected two identical speakers the same way to Channel 1 and Channel 2, placed a microphone between them and ran Acourate's "Microphone Alignment" procedure:

As we can see, both speakers are in phase, no need to worry. Let's continue to measurements.

Output Power

I ran a couple of measurements into 4 Ohm load in bridged mode from an unbalanced input.

Load, Ohm	Input, dBV	Output, Vrms	Power, W
4	0, unbal	32.77	268.5
4	-4, unbal	20.63	106.4
4	-10, unbal	10.33	28.4

As we can see, the voltage gain in bridged mode from unbalanced input is the same as for single ended mode from balanced input—30 dB. Doubling the output voltage allows for almost 4x increase in the output power—compare 68.5 W into 4 Ohm from 0 dBV that we have seen for the single ended mode vs. 268.5 W from the same input in bridged mode. Nice! But what about distortion?

Distortion

Unfortunately, distortion doesn't look good. I had to lower the input level to -10 dBV to avoid clipping on the input of QA401, and distortions graph from a 1 kHz input looks like this:

Two tone distortion produces high levels of ultrasonic noise (from same -10 dBV level):

And remember, that's for 10 Vrms output (28.4 W power). In single ended mode even 20 Vrms output produced much less distortion. Clearly, the bridged mode of this amplifier is designed for something like PA applications, not for high fidelity.

Conclusions on the QSC SPA4-100 Amplifier

Answering the questions I've stated in the beginning of this post. We can see that this QSC amplifier is way more linear in its best operating mode (single ended) than cheaper Monoprice in its best mode (bridged)—just take another look at the graphs in the post about Monoprice.

Also, QSC's capabilities are specified much closer to real measurements than what Monoprice had specified. And clearly, higher price point of QSC is fully justified.

The bridged mode produces higher distortion even at lower input signal levels. This can be explained by the fact that each driving amplifier in this case "sees" twice less load. As we have observed on the 4 Ohm vs 8 Ohm load, distortion in this amplifier increases as the load impedance decreases. I suppose, it increases even more with 4 Ohm load gets divided in half by bridging.

What is common for both amplifiers is that there are some visible ultrasonic artefacts that are not filtered out even when using a real inductive speaker load. So actually, driving some sensitive speaker at high output level may overload and even damage it due to excessive high frequency energy.

Speaker Driving Modes

We can see that the audio engineers at QSC are very creative. As we have observed, the same speaker can be driven by this amplifier in 3 modes:

• from the "+" terminal in positive phase;
• from the "-" terminal in inverted phase;
• from both terminals.

Does it make any difference to the speaker? In fact, no because what speaker "sees" is the difference of potentials between its "+" and "-" terminals. Say, we have 1 V (relative to some arbitrary reference point) applied to "+" terminal, and 0 V applied to "-" terminal. The speaker "sees" 1 V - 0 V = +1 V voltage. This voltage drives the cone forward (if enough current is supplied by the amplifier).

What if we apply 0 V to the "+" terminal and -1 V to the "-" terminal? The speaker "sees" 0 V - (-1 V) = +1 V voltage. This voltage drives the cone forward. Now, what if we apply 0.5 V to the "+" terminal and -0.5 V to the "-" terminal? Absolutely the same thing.

This is why it's possible to drive a speaker from the "-" terminal using an inverted signal. The speaker will behave the same as if driven from the "+" terminal using the signal in the original phase. Same thing happens if we drive the speaker from both sides. The only participant for which bridging matters is the amplifier. After internalizing all this stuff, I've re-read the post from Benchmark Media about myths of balanced headphone connections and this time I understood every word from it. Practicing with amplifiers helps to understand the theory!

Case Study of LXmini in Our New Living Room

This summer we moved into a new rented house and finally I got some time to set up LXmini in this new environment. I've learned a lot while doing this and hope that sharing this experience could be useful for other people.

Initially I was planning to recreate my old 4.1 surround setup with two pairs of LXminis as front and surround speakers + KRK 10s subwoofer (only for LFE channel). However, I tried watching a couple of movies on a temporary stereo setup of LXminis and decided that stereo image they create is immersive enough and I don't want to complicate the setup with another pair.

The challenges I faced while getting the stereo setup right were different from what I had in our old apartment. First, we have bought a tall wide console for the computer and XBox, and I learned that the console creates strong reflections if speakers are put too close to it. On the other hand if I set the speakers further from the console, they get either too close to the couch or to the side wall. Second, this time I decided to use the subwoofer as a low frequency extension for LXminis but didn't want to compromise their excellent output.

Minimizing Reflections

This is a schematic drawing of the room. Note that the ceiling is quite high and sloped. This reduces vertical room modes significantly. The bad news is that the listening space is asymmetric and narrow. Below are views from the top and from the side, all lengths are in meters:

Blue circles represent the positions of the speakers in my temporary setup. The orange circles is the final setup. I've spent some time looking for the best placement and used a number of "spatially challenging" test tracks:

• tom-tom drum naturally panned around (track 28 "Natural stereo imaging" from "Chesky Records Jazz Sampler & Audiophile Test Compact Disc, Vol. 3");
• LEDR test—HRTF-processed rattle sound (track 11 "LEDR" from "Chesky Records Jazz Sampler & Audiophile Test Compact Disc, Vol. 1");
• phantom center test files from Linkwitz Lab page.

When the speakers were placed too close to the console, LEDR was sounding smeared and so were the phantom center tests. ETC curves were also showing some strong early (< 6 ms) reflections:

I moved the speakers further from the console and placed them wider, so they didn't get too close to the couch. Though, the right speaker was now too close to the right wall. Fortunately, the reflections from the wall can be defeated by rotating the speaker appropriately. The hint that I've read in the notes of S. Linkwitz was to put a mirror to the wall and ensure that from the listening position I see the speaker from the side. Since LXmini is a dipole speaker there is a null at the side, thus the most harmful reflection from the nearby wall is minimized. We can see that on the ETC graphs from the new position (the graphs from the initial position are blended in for comparison):

For the left speaker, instead of the two reflections above -20 dB within the first 6 ms there is now one of a bit lesser power. For the right speaker, the overall level of reflections arriving during the first 6 ms are significantly reduced, and its ETC graph resembles more the ETC of the left speaker.

Playing the test tracks has also confirmed the improvement—now I can feel the rattle sound in LEDR moving in vertical and front-back directions clearly. Also, by avoiding creating strong reflections for the right speaker, I've made it essentially equal to more "spacious" left speaker placement, thus the asymmetry of the listening space doesn't matter anymore. However, the resulting "aggressive" toeing in of the right speaker has narrowed the listening "sweet spot". Apparently, it's not easy to achieve a perfect setup under real life conditions.

Equalizing Speakers

From my previous measurements I knew that the quality of the speaker drivers used in LXminis make them well matched. However, my initial measurements has shown some discrepancy which I wanted to correct:

I'm not a fan of excessive equalization—I believe that our brains are a much more powerful computers than our audio analyzers. But adding a couple of filters to correct for speaker placement seems reasonable here. In this case, I reduced the amplitude of one of the notch filters in LXmini equalization and added a couple more filters:

Note that I didn't do anything below 50 Hz because I plan to use the subwoofer with the crossover frequency at 45 Hz.

Then I adjusted KRK 10s to inhibit its output in the range of 30–60 Hz to "boost" its output at 20 Hz. Here I used filters suggested by Room EQ Wizard for the listening position:

Subwoofer Alignment in Time Domain

This was the most challenging part. I connected subwoofer using a cascaded miniDSP 2x4 HD in the following way:

Additional processing delay, phase shifts, and asymmetric positioning together create a framework which is challenging to analyze. Instead, I decided to apply the approach suggested by the author of Acourate software Dr. Ulrich Brüggemann. The procedure consists of the following steps:

1. Capture the impulse response of the main speaker using Acourate without the subwoofer.
2. Capture the impulse response of high-passed main speaker plus the subwoofer. The high frequency part of the response allows Acourate to align these IRs in time.
3. Convolve both impulse responses with a sine wave from the overlapping region.
4. By comparing the mutual offsets of the resulting sine waves in the initial transient moment and during sustained period deduce time delay and possibly phase inversion.

As I've learned from my experience, aligning based on a single frequency in the Step 3 may not provide the best results as at low frequencies the phase and the group delay of speakers may fluctuate severely. So instead of using a single sine wave I used a log sweep range in the bass region. This doesn't provide data for aligning initial transients, but for bass frequencies I think the sustained stage is much more important.

Here is how convolutions with a log sweep from 40 to 100 Hz were looking initially for the left and right speaker:

The left graph is mostly aligned, while the right one shows a delay of the main speaker for 2.5 ms. It can be seen that even on the left speaker, the alignment in the low bass region is poorer than at higher frequencies. I don't consider that to be a problem because there the contribution of LXminis is negligible. It's much more important to time align the region where both the sub and the LXminis can be heard together. It's also easy to see that if we attempt to use the crossover frequency (45 Hz) as the anchor point for time alignment, the speakers would be out of phase for higher frequencies which will result in "sagged" frequency response.

To avoid compromising the alignment of the left speaker, I decided to delay the sub for 1.25 ms which improves alignment for the right speaker, but doesn't degrade it too much for the left one. Below are the graphs of LXminis filtered with Linkwitz-Riley 24 dB/oct crossover at 45 Hz and with added subwoofer:

Definitely we can see extended bass range. You can also feel it :) I think, setting the crossover point low allows to get the maximum fidelity from the LXminis + subwoofer combination.

With all this laborious setup done, it's time to enjoy music!