HAM6.80DSP

INITIAL RECORDING, DESIGN FEATURES

The amplifier is compact - the body is only 10.6 x 4.2 x 19 cm (plus connectors at the ends).

Mounting - with corners on the long sides.

The kit includes, as usual, gloves.

The inputs are as follows

4 channel high level input

2-channel linear (AUX)

Digital optical

Bluetooth audio module (sold separately)

The power supply, amplifier outputs, and high level inputs are located in different connectors. On many compact processors, the same remote can be seen in the same block as the high-level inputs, which is not very convenient: if you do not intend to use them, you still have to connect the "tail" and leave the wires "hanging". There is no such drawback here - we only connect what is needed.

All 8 outputs of the processor are implemented as linear ones, i.e. Hellion HAM6.80DSP can be used not only as a processor (the first 6 channels go to the built-in amplifier), but also simply as a processor plus separate, more powerful amplifiers. By the way, it will be possible to use both the linear outputs and the built-in amplifier at the same time - I'll tell you more about this later.

The processor is configured from a computer connected via USB. A long cable is included.

An optional front panel can be purchased.

Now for some filling. The whole device is almost completely assembled on one board - only a small part with connectors is placed on the "second floor".

Math - on the old friend Cirrus Logic CS47048C-DQZ. The chip is by no means new, but well deserved; it could be seen, for example, in the first Audison Prima.

A special feature of the power supply is that it can withstand significant voltage drops. The amplification section shuts down when the on-board voltage drops to 9V, and the processor stays on until the voltage drops to 5V. Unlike traditional amplifiers, the power supply is unipolar.

Each of the 6 amplification channels is built as a bridge, so the amplifier actually has 12 channels combined in pairs.

The amplification part is mounted on TPA3116 power chips (if I'm not mistaken, they're also used in the Helix V Twelve DSP).

The chips are pressed through an aluminum strip to the case. Their efficiency is high, so the amp is warm, but not hot.

HOW TO SET IT UP AND WHAT IT CAN DO

 

The software starts without installation, the connected processor sees it immediately.

It seems that everything is familiar, but there are some peculiarities, and some of them are quite important: from the possibility to work on two systems at the same time to the automatic configuration, which I already mentioned in passing.

I'll start with the simple things. First of all, you can immediately switch to Russian, the Russianization is correct. Channel routing (what from which input channel goes to which output) can be set directly from the main window - a plus for simplicity and clarity.

During the initial setup, you can set the type of output channel while the filters are set to certain initial values and the system can be safely turned on. For example, if you select Twitter as the channel type, the high-pass filter will automatically be turned on, and you will definitely not burn anything. If you select Subwoofer, the low pass filter will be turned on. And so on. In principle, not new, but convenient.

How to build two systems on one processor. First, if you plan to use only the processor part, then the built-in amplifier can be disabled by software. Moreover, this does not just remove the signal from it, D-class chips actually go into "sleep". This can be seen at least by the decrease in power consumption during shutdown.

Second, the presets remember not only the filter, equalizer, and delay settings, but also the selected input and amp mode (on or off), i.e., the entire process state.

These two features make it easy to build two systems at the same time. The first is "everyday" for working in the car, connecting the speakers to the built-in amplifier. The second is a loud pop system with separate, more powerful amplifiers connected to the line outputs. And you can switch between them by simply selecting the desired processor preset (if you have a remote, this usually takes a few seconds). The most you have to do manually is turn the remote on or off at the amps. However, the developers promise to add this feature in future versions of the software.

Automatic processor tuning. You will need to purchase an optional microphone for this. The first one you come across here will not work, but this problem can be solved with the official distributor of Hellion (let me remind you, this is AAC Distribution).

So, what can auto-tuning do? We connect a microphone to the processor, go to the auto-tuning section and first indicate which channels are used in the system. Then we specify what exactly needs to be adjusted: delays, signal phase, or equalizer (actually, it's better all of the above). If you wish, you can set the target frequency response you want to achieve (the same buttons for Rock, Classic, etc.).

It is important to note that the crossover filters are not automatically adjusted, i.e. they still have to be set in advance.

The only thing left to do is to attach the microphone to the headrest (and if it is a stage system, then outside the car at the listening point) and press "Start". Protseus himself turns on the noise signal and "catches" it with a microphone. At first at a low volume, he gradually increases the volume to the desired level. Then he starts another series of signals and thinks about something. When it is done, it displays the measured frequency response and equalization curve on the screen.

 

I tried this algorithm on my PSB Synchrony One B bookshelf speakers. They have the ability to be bi-wired, so I simulated a common channel. I won't describe the entire process because I experimented a lot - with two columns and with one. Here are the conclusions:

The delays are set correctly - they can be used as a basis for further, more precise settings.

The processor detects the mixed phase between the left and right channels, but it is better to check the phase of the tweeters relative to the mid-bass speakers yourself (if it is a two-channel channel).

If the overall tonal balance is satisfactory, automatic equalization can be omitted. The fact is that the microphone "listens" to the signal over a fairly wide angle, and during measurements it not only "hears" the speakers themselves, but also actively catches the reflection, which greatly affects the measurement results. This means that it may try to apply correction where it is not really necessary. It only makes sense to turn on the EQ if you have obvious peaks and dips caused by the loudspeaker's characteristics that cannot be compensated by anything other than an equalizer (neither by combining the bands with a crossover, nor by phasing).

CHECKING STANDARD DSP FUNCTIONS

Since cheap DSPs often have problems with the correct operation of filters and equalizers, I checked how the Hellion works. The signal from the Audiomatica Clio measurement complex was applied to the linear inputs, and the output signal was not taken from the amplifier outputs, but from the linear ones, in order to exclude the influence of the amplification part on the result.

Thus, the signal processing time by the processor is approximately

Check the operation of the delays. Maximum delays are up to 15 ms (equivalent to "moving" the speakers about 5 meters). In principle, there are no comments on this part. The pulse shape remains constant (in some processors it changes when delays are added). Good.

Check the operation of the filters. A typical three-stripe is simulated. By default, all filters have a slope of 24 dB/octave and a filter type of Linkwitz. So far, the fit is good:

When switching to a Butterworth filter, the curves of the frequency response are a bit sharper, as expected, and with the Bessel type they are smoother. Everything is the same as in the textbook:

Checking the operation of the equalizer. Since everything was fine with the previous points, no catch was expected here. But some nuances appeared, although I wouldn't say they were critical.

For example, this is how the adjustment to plus and minus actually happens (the volume was reduced to 42 units out of 66 to ensure that the signal does not clip):

From a formal point of view, there are small errors at low Q settings (a wide control band), and small artifacts in the frequency response at positive settings. From a practical point of view, however, this is not so important. So we just follow the general advice - try not to turn the equalizer all the way to positive, and everything will be fine.

CHECKING THE OPERATION OF THE AMPLIFIER SECTION

 Since each channel in Hellion is designed as a bridge, the usual way of analyzing a signal with Audiomatica Clio is to "dance with tambourines". I can only note a good signal-to-noise ratio - with a useful signal of about 1 W, the background noise "splashes" somewhere above -90 dBV (at least in the audio range).

For more clarity, I decided to do it another way - I connected a generator to the input and used an oscilloscope to see what was happening at the output (load - 4 ohms). First of all, the standard frequency for such cases is 1 kHz:

The characteristic high-frequency ripples are almost absent (for the D-class, this sometimes becomes a real problem). The limit occurs at an amplitude of about 17 V (this is about 12 V RMS). We take out a calculator and calculate how much power it is at a 4 ohm load:

Power parameters of the Hellion HAM6.80DSP (measured at 1 kHz):

Operating at 4 ohms, undistorted output power - 36 W

Operating at 4 ohms, power reserve of the power supply - 72 W

If you move away from the generally accepted frequency of 1 kHz and look at the operation of the amplifier at low and high frequencies, you will notice that the amplifier reaches its limit in a slightly different way. In general, however, the power reserve remains constant. Oscillogram of a 100 Hz signal:

At the upper end of the range (20 kHz frequency) there is a little more high frequency ripple, but overall the amplifier works quite well. The smoothing of the peaks is the result of the D-class output filter:

HOW DOES IT SOUND?

And the Hellion sounds surprisingly happy and playful. The TPA3116 chips themselves are interesting, and here their capabilities seem to be used quite successfully. When choosing the acoustics, it is better to avoid overly heavy mid-bass drivers, as the power is still not prohibitive. The Hellion is also good at handling normal and even more sensitive acoustics: the D-class bridge rules in this respect.

If you do not neglect this advice, the amp will give you a light and dynamic sound with clear vocals and quite decent detail for its price. All in all, it plays quite transparently and comfortably. And what is important - without the dryness or causticity at high frequencies characteristic of the D-class, even when listening with acoustics of an obviously higher class. In general, the sound impressions are very pleasant considering the price of the device.

BRIEFLY ABOUT THE CASE. FOR WHICH TASKS SHOULD I CHOOSE?

First of all, the Hellion HAM6.80DSP is a good choice for an inexpensive but decent "everyday" system. The main thing is to choose acoustics that are not too heavy and you will get a system with a very comfortable sound.

Second, it will fit well into a variety system. Or rather, double systems - pop and everyday. For example, we connect speakers in standard places to the built-in amplifier - this will be a system for work in the cabin. And from the linear outputs we build a system with powerful amplifiers and pop-up acoustics, for example in the trunk. Because presets remember not only the current settings, but the entire state of the processor, you can literally switch between configurations in a matter of seconds. Each system can even be powered by its own source.

All in all, if you're on a budget but still want something that sounds decent, the Hellion is definitely worth considering.