What Are DSPs in Speakers?

The use of Digital Signal Processors (DSP) in studio monitors has become a common occurrence, with many considering DSPs to represent inevitable progress in speaker technology while others remain suspect of its potential for introducing degradation into the signal path.

At HEDD Audio we have engaged with both sides of the argument: Our original MK1 line of monitors featured analogue circuitry and we eventually embraced the digital approach with our MK2 line, which features DSP-enabled innovations such as the HEDD Lineariser and Closed or Ported.

In this article, HEDD founders Klaus Heinz and Frederik Knop draw from their combined decades of experience in speaker manufacturing and R&D to offer their technical perspective on what DSPs can do for the listening experience and why they chose to rely on DSPs for the MK2 line.

A Primer on Digital Signal Processors

DSPs are specialised microprocessor chips, with architecture optimised to convert analog signals in order to measure, filter or compress them. The first thing to bear in mind with DSPs is that their inclusion in a speaker requires the signal path to be altered. Signals can be sent to speakers either digitally or analogue and if DSPs are used then additional conversion is needed in the signal path: either once for digital signals (using a D/A converter) or twice in the case of analogue signals (using A/D and D/A converters). Therefore, regardless of the design quality of the monitor and/or converter it is always likely that the signal quality, and thus sound quality, of a monitor will be affected to some degree by the inclusion of DSPs.

This said, DSPs can do things that a purely analogue monitor setup cannot. And this has been a big part of their appeal.

We can generalise the main tasks of a DSP in a monitor’s input section to be:

• to form a frequency dividing network
• to apply equalisers to boost or lower certain areas within the frequency spectrum
• to insert limiters
• to provide different voltage gains to the individual bands and the overall amplification

These functions are made possible by the use of filters and many DSPs make use of Infinite Impulse Response (IIR) filters, which enable all of the above and, in some rare cases, can also apply a delay to the tweeter to compensate for the earlier start of its signal compared to the woofer output to align their time behaviour. It’s important to note that the characteristics of IIR filters are the same as those found in analogue circuitry.

One of the most popular and practical advantages of DSPs is that they offer an easy way to input filter or EQ values while also giving users the possibility to apply very steep filters to the signal, up to 96 dB/octave (16th order). However, while these filters seem impressive, in practice it becomes evident that a 24 dB/oct (or 4th order) filter, the maximum available in analog filters, is good enough to achieve results without compromise. Our R&D experience shows no real benefit from higher filter steepness.

The TYPE07 MK2 backplate in production, featuring some of the DSP-enabled controls available on the MK2 range.

Finite Impulse Response and Phase Alignment

One of the most theoretically interesting approaches to DSP filters is the use of Finite Impulse Response (FIR) filters, also known as linear phase filters. These filters allow linear phase behaviour for the entire monitor independent of its frequency response, something that IIR filters cannot do. Phase response describes the time relationship between the different frequencies that travel through the unit. In short, different frequencies take different amounts of time to travel within a monitor, the simplest example being that lower frequencies take longer to travel than higher ones. This is an inevitable aspect of speaker design but largely unwanted in practical situations. FIR filters make it possible to correct the phase response and align all frequencies in time.

It is now time to discuss the Fourier Transformation, which delivers the fundamental relationship between the time and frequency domains of waves or vibrations in general. The Fourier Transformation allows to calculate and display the interdependencies between these two domains. In practical measurements, Fast Fourier Transformation (FFT) plays an outstanding role in many technical fields, among them frequency/time responses for loudspeakers and modal analysis of cones or cabinets. One cannot influence the frequency response of speakers or other electroacoustical devices without changing their phase response and vice versa. To deliver both linear frequency and phase response together is not possible in the analogue world but is possible with FIR filters.

This type of filter control was not possible during the first century or so of loudspeaker design and its development in the late 20th century came as a complete, but welcome, surprise. This is why today, FIR has become one of the two primary types of filters available in DSPs alongside IIR.

In order to control linear frequency and phase response, FIR filters allow for the storing of parts of the signal until everything is in the timely order needed before sending it to the output. The price to be paid for this linear phase response is a delay in the overall signal, with practical numbers between two and 500 milliseconds, depending on various parameters. This is no problem in a pure reproduction chain, but may cause issues in a recording situation which is why FIR filters are best used in specific studio situations.

We believe that the main attraction of FIR filters is the correct time behaviour that becomes audible in rather perfect impulse and phase responses of the speaker under test. In turn, we also believe that these technical improvements do translate into the listening experience, especially when it comes to the perception of the stereo field and the details it contains.

SHARC DSP boards for HEDD MK2 monitors being assembled and tested in our Berlin warehouse.

The HEDD Lineariser - A More Accurate Sonic Experience

For us at HEDD, it was the innovation afforded by FIR filters that convinced us to integrate DSPs into our MK2 line up. Our DSPs are powered by a SHARC processor, an industry standard, and they provide various key functionalities for the range including three desktop and two shelving filters, to adjust the monitors to the space, bass extension and complete linear phase Satellite-Subwoofer setup with the BASS subwoofers, digital AES and analog XLR connections, Closed or Ported modes, and, the HEDD Lineariser, an optional FIR filter that allows users to control the linear phase response of their MK2 monitor to create a fascinating stereo image and sonic depth.

The Lineariser allows users to flatten the speaker phase so that the exact timely relationships of the incoming signal are reproduced. To do this we record impulse responses of every model, which show us frequency delays occurring within the speaker. Those impulses are then coded into the Lineariser section of the DSP so that the FIR filter can correct the delays.

We often think of what the Lineariser does as akin to lifting a curtain over the audio signal. It helps tighten the audio signal and it improves the localisation of objects and panning information. When it comes to transient-rich audio material, the Lineariser will provide you with a stunning, more accurate sonic experience.

Pros and Cons of DSPs in Monitors

Ultimately, the inclusion of DSPs in a monitor design should not be the sole reason for choosing it. Sound quality, while measurable to a degree, remains primarily subjective and there are great sounding monitors in both the analogue and digital realms, just as there are poor sounding ones. We always recommend taking time to listen and compare different monitor designs and approaches in order to find the one that is right for you.

DSP Pros:

• Quick and easy to set and apply filter and equaliser settings
• Piece to piece consistency for the electrical filter values
• User interface (not all DSPs offer one)
• FIR filters allow linear phase behaviour of the complete monitor, at the cost of signal delay
• Room correction as a possible extension
• Steeper filter characteristics up to 96 dB/oct, however the effect of filters past 24 dB/oct (the maximum for analog filters) is not relevant

DSP Cons:

• Requires one or two A/D or D/A conversions which will affect sound quality
• Potential for small distortion and noise issues compared to high-end analogue designs
• FIR filters introduce time delays between two and 500ms which are not acceptable in live recording situations