Author: XTA

How to use the FIR filtering within the DP448

Before explaining our implementation of FIR filtering, please remember that this feature is designed to be used by speaker manufacturers and designers, and acousticians.  It is NOT designed as an end user adjustable addition to the DP448.  As such, end users interested in hearing the difference  between FIR filtering in place of standard IIR filter implementations should contact the manufacturer of their speaker system(s) and ask for preset file(s) containing the FIR versions.

As mentioned in the introduction, FIR Filtering is not available on all DP448 units – for information about how to check if a unit will accept the update and be able to use it, please see the FAQs at the end of this document.

The FIR processing blocks are available on each output channel on the DP448, and utilise a combined pool of processing allowing different channels to have differing amounts of filtering (or no FIR filtering).  The total number of taps available is 2900, or 4000 if the graphic equalisers are disabled.

FIR filter data is imported into a unit alongside preset data and as such, Library Manager is used to include it in preset files.  When designing a preset, all the “traditional” filters associated with normal output memories are available, with the addition of an extra FIR data tab:

 

To include FIR filtering coefficient data on any particular output, the “Load New” button is pressed on the corresponding output.  Coefficient data must be in a comma separated variable (.csv) formatted file, although the coefficients will still be recognised as long as each one is separated by a carriage return (so each one is on a new line).

A typical data file might look like this:

-0.02012136176742655
-0.05843583195045424
-0.061166134623015594
-0.010897364468132498
0.05127863696321711
0.03318844462546994
-0.056622474759205677
-0.08572325032853853
0.06337939281116886
0.3109413381355739
0.4345629035910811
0.3109413381355739
0.06337939281116886
-0.08572325032853853
-0.056622474759205677
0.03318844462546994
0.05127863696321711
-0.010897364468132498
-0.061166134623015594
-0.05843583195045424

Loading this file into an output will immediately update the channel with the number of taps used, and the remaining reserve.  If the number of taps exceeds 2900, a warning is also displayed to remind you that the input graphic equalisers will be disabled when this preset is used.

As explained earlier in the document, there is a time penalty to be paid when using FIR filtering and this varies dependant on the complexity of the filtering (more taps = longer delay).  As each set of filter data is loaded into an output’s FIR processing block, Library Manager automatically calculates the processing delay that will be introduced on that particular channel, and adds a compensation delay across all other channels.

In the above example we can see that two different data sets are being used – one with 1344 taps, and one with 912 taps.  Library manager calculates the longest filter’s delay and this is applied to all outputs.  The delay applied to the output with the shorter filter is the difference between its inserted processing delay and the longest delay, so aligning it with all the other outputs.

In other words – 1344 taps causes 7.000mS of delay, and 912 taps causes 4.75mS of delay, so the output with the 912 tap filter needs an additional 2.25mS (7 – 4.75) of delay to keep it aligned.

These delays can be added automatically using the standard output delay lines in the preset, by ticking “Enable Auto Delay Correction”.

On the delays tab, these additional delay times can also be displayed by ticking the “Show FIR Delay Correction” (note that times shown are not related to those in the above example)

 If the correction delay is shown, you will not be able to adjust the output delay times to 0mS.

Once the filter data has been added and all other adjustments have been made to limiters, and if necessary standard IIR filters, the preset file is assembled and built as normal.

Description of this process is covered in the Library Manager manual, available on-line here

So where does the coefficient data come from?

The implementation of FIR filtering on the DP448 has been designed with loudspeaker manufacturers and acoustic engineers in mind.

It is not recommended that end users should attempt to utilise this facility.

FIR filtering coefficient data is available as an output from either DSP filter design software, or from audio analysis software.

As an example, a simple package that has been used to test the system is available on-line here:

https://t-filter.appspot.com/fir/index.html

As stated previously, coefficient data must be in a comma separated variable (.csv) formatted file, although the coefficients will still be recognised as long as each one is separated by a carriage return (so each one is on a new line).

 

FIR filtering in AudioCore

 

All FIR filtering data is stored within the presets that are sent to the unit via out loader software, in the same manner as a standard non-FIR preset file.  Note that a preset file may contain a combination of FIR only presets (all IIRS not used), FIR + IIR filtering, or standard IIR filters.  This allows easy comparison between standard crossover methods and an FIR based corrective method.

With this in mind, when a preset is recalled in AudioCore, which contains any FIR filter sections, whilst AudioCore does NOT have access to the data, it is aware of the FIR filters, and shows this by colouring the corresponding output channel’s X-Over section on the device window in green:

Clicking on output 2 in the last example will jump directly to the output’s EQ edit screen – note the FIR filter message beside the device name, and the number of taps tip in the top right of the frequency response curve:

 

Note that the frequency response does NOT include the effect of the FIR filtering.  This is also true of the global view.
Frequently asked questions

How do I know if my DP448 is compatible with this upgrade?

As this feature only became available after a hardware upgrade, and XTA reserve the right to improve the specification of the DP448 at any time, only units with a serial numbers in the range 3760-3798 and then from 3960 onwards will support the update.

Additionally, you can check when turning the unit on – if the unit is running the latest firmware (currently V1.22) then the display will show

 += AudioCore DP448 =+
 === Software V1.22 ===

if the compatible DSP platform is fitted or

== AudioCore DP448 ==
== Software V1.22 ===

if the incompatible older DSP platform is fitted.  The “+” at each end of the top line instead of an “=” is the signifier of the newer DSP.

You can also check in the System Status through the System Sub-Menu – one of the scrolling messages will either show

System Status
= DSP TYPE=2 Number=1 =

if the compatible DSP platform is fitted or

System Status
= DSP TYPE=1 Number=2 =

if the incompatible older DSP platform is fitted.

What happens if I use the new firmware with an older version of AudioCore?

FIR filter data will simply be ignored and the unit will operate as normal otherwise.
We do not recommend this of course, as it may lead to confusing operation – see the FAQ below about graphic equaliser behaviour.

 

How does Copy and Paste work with FIR Filtering Data?

Copying and pasting either an output or a device will NOT copy the FIR filtering data.  A warning will be shown to highlight this:

Pasting in data to a channel with FIR settings (even from another channel with FIR settings) will delete the FIR filtering on ALL outputs.

How are memories handled when there is FIR Filtering Data involved?

Recalling a preset including FIR data, adjusting the standard filtering (crossovers and PEQs on output) and saving this in a user memory location WILL “include” the FIR filter data.  What actually happens is that the link from the user memory to the FIR data is preserved so when the user memory is recalled, the associated FIR data is also updated.

Recalling a user memory or preset which does NOT include FIR data will delete the FIR filtering associated with ALL outputs.  Saving a user memory (based on a preset containing FIR filtering data) via the front panel menus will also preserve the links to any output FIR data.

How can I tell if a preset contains FIR filter Data?

AudioCore does not know until a preset is recalled, if there is FIR filtering data associated with it.  Remember that FIR filter data is associated with Crossover memories only.  We would advise you to name memories containing FIR filtering data with an “F” at the end of the name to remind you.

On the DP448 itself, indication of FIR filtering being in use is on an output channel’s GAIN adjustment screen – an “F” will be displayed as below on any output which has FIR filtering.

Out2    Output A   Gain
Output Gain = +6.0dB   F

What happens if I delete or replace the preset file in the DP448?

Downloading a different preset file will clear any links to previous FIR filtering data.  Subsequently recalling any crossover settings stored in user memories that were previously linked to FIR filter data will still recall the presets, but the FIR filtering will of course not be present.

Where has my Graphic Equaliser gone!?

Using FIR filtering with more than 2900 total taps will bypass the input graphic equalisers on ALL channels.
They will not appear in the input editing list via the unit’s front panel, and clicking on them in AudioCore within the device window will jump instead to input PEQ editing (and the GEQ tab will be missing).

Recalling a preset or user memory with less than 2900 taps or with no FIR filtering data will reinstate the GEQs with their last used settings.

Time for some proper technical stuff on the techblog! 
Not too technical, but think of it as a layman’s guide to this somewhat misunderstood topic – FIR 101 if you like…

Our actual implementation will be covered in a subsequent article.

Overview

With the introduction of a more powerful DSP platform in 2010, newer DP448 units[i] now benefit from increased power which we have used to implement additional processing.  Finite impulse Response (FIR) filter technology has been in the spotlight in recent years as another tool in the armoury of filters, delays and dynamics used to manage speakers as effectively as possible.  However, as with the majority of technological advances, there are advantages and disadvantages to be aware of when using FIR filtering.

FIR Filtering NOT FIR Filters

Whilst traditional filter responses such as parametric EQ bands, crossover filters of any shape, or shelving filters can all be achieved through FIR filtering, in many cases, the technology is NOT used for this purpose.  There are two main reasons for this.

Real-time vs. Off-line Adjustments

Real-time adjustment of a traditional filter modelled using FIR filtering is not a practical scenario.  The implementation of any filtering using FIR topology is the result of digital data being passed through a series of processing stages traditionally known as filter taps.  Each tap can be thought of as a computational component and different filtering scenarios will require more or less calculation stages, or taps.  As would be expected, for every tap, or calculation, there will be a time penalty involved – more taps = longer time.  Designing a filter to operate at a certain frequency and with a certain level of attenuation may produce x  number of taps, but adjustment of a turnover frequency or required level of attenuation can often result in the number of taps changing and so the computational delay associated with the filtering also changing.

Whilst this delay difference is unlikely to very large (sub-millisecond in modern systems sampling at high rates with typical filtering), even small changes can affect alignment delays between drivers in a multi-way system, introducing side effects that are both difficult to determine and account for.

Processing Delay Penalty

All digital signal processing relies on the manipulation of sampled data using calculations.  The major difference between the implementation of Infinite Impulse Response (IIR) filtering and FIR filtering is that the calculations applied to the incoming sample stream will involve feedback of data back into the filter “maths” for an IIR filter, whereas FIR filtering is a linear process with no feedback.

Considering a basic audio analogy, traditional analogue filtering nearly always utilises feedback to reduce system complexity and provide a wider range of adjustment with fewer components.  This method is analogous to IIR filter topologies which utilise feedback of data as mentioned above.  Care needs to be taken with the analogue filtering and IIR filter design as, where there’s feedback, there’s also the possibility of instability and unexpected behaviour as a result of this. 

FIR filters, with their linear processing topology, are inherently stable and cannot be forced in oscillation (no feedback).  No input will result in no output!  Designers strive to achieve this with IIR filtering as well but, as any audio engineer knows, just stopping talking into a microphone won’t always result in silence – the slow build-up of oscillation caused by too much feedback causing instability is a constant enemy!

So… it would seem like FIR filters are the best solution, but this is not always true due to the simple fact that their linear operation means more steps of calculation that are cascaded one after the other, resulting in more time processing = delay.  Even in a modern audio digital signal processor, this delay can become quite significant, especially when processing low frequency signals at high resolution.  The reasons for this are beyond the scope of this document, but these delays can be in the order of 5 to 15 mS which would be deemed unsuitable in many live performance situations such as music concerts and theatre.  This doesn’t matter quite so much when dealing with installed sound, such as studios, clubs and even voice evacuation or announcement systems.

Negative negative negative…

After all that doom and gloom, are there any advantages to using FIR filtering?  Yes – of course there are, and the one that makes the most news for audio is the phase response characteristic.

So what is “phase response”?  Simply put, the phase response defines the relationship between different frequencies present in a signal relative to each other in time.  The timbre or characteristics that can help identify a particular instrument or sound are heavily influenced not just by the frequencies present, but also by how they relate to each other in time.

Traditionally, we manipulate the frequency response of speaker system by altering the balance of certain bands of frequencies – feeding low frequencies to subs, and filtering off the LF before it gets to the more delicate high frequency drivers.  This is the nature of crossover filters.  Various corrective filters may also be applied to individual drivers to reduce resonances or boost areas lacking, in an attempt to achieve the perfect “flat” response – everything from the lowest lows to the highest highs being output at the same relative level.

However, as anyone who has ever used a digital crossover will know, this isn’t quite the end of the story.  Within speaker cabinets, improvements can be made to the final output if care is taken to correct driver alignment delays.  This method is used to align the outputs of the various drivers in time with each other as their actual centres of output will be physically different within the box.  An HF driver’s voice coil will most likely sit tens of mm further forward than the voice coil of a 12” bass driver, resulting in the HF output reaching the listener typically hundreds of microseconds ahead of the bass driver’s output. 

This would not be too objectionable, were it not for the fact that speakers and crossovers are not perfect and their bands of useful output always overlap to some extent.  It is in the areas of overlap that we are most concerned about the time differences.  If the signals from the LF and HF have a phase difference (or a time delay difference) then they will start to interfere, both constructively and destructively, resulting in peaks and troughs in the response.  This is occurring about the crossover point of the drivers.  Adding a delay to the HF driver output to ensure the sound it emits arrives at the same time as the LF driver’s output has the effect of these signals summing “more correctly” resulting in a smoother response.

Going back to the timbre of a particular instrument, the timing relationships between various frequency components are part of what gives the instrument its “sound”.  Preserving these relationships goes a long way towards improving the clarity, and purity of any system.

Traditional analogue filters and IIR filters can have an adverse effect on the phase relationships present in a signal, “smearing” them so different frequencies arrive at subtly different times to others.  This needn’t be disastrous however – careful filter design can result in a system with minimal phase distortion and very good characteristics.

FIR filtering does not introduce phase distortion – it can exhibit a linear phase response, meaning none of the “smearing” of frequencies in time occurs, no matter how “steep” the filtering is, so allowing the use of higher order crossover filters.  Traditional analogue and IIR filters with very steep or sharp slopes have a greater adverse effect on the phase response of the system.  However, as mentioned in the introduction of this article, modelling traditional designs using FIR filtering is only one approach…
Throw away thoughts of traditional filters!

A bold statement, but one that encapsulates the main design ethos of our implementation of FIR filtering.

As explained in the previous section, traditional approaches to loudspeaker system design would involve high and low pass filters to split bands up, and “corrective” filters (shelving and PEQs, possibly notch filters) to “iron out” any problems.  Digital signal processing normally uses IIR filters, set up as models of the analogue equivalents, to achieve this.

There is another approach to this that involves a more holistic method.  In a similar way to measuring a frequency response and attempting to apply an inverse correction to fix problems in the frequency domain, a system’s impulse response can be measured and a corrective response calculated that will account for any frequency time smearing that is present in the system.

The details of how this works are beyond the scope of this document, but the basic principle is that instead of applying a range of frequencies one after the other (as in a low to high sweep) and measuring the various resulting output levels (so creating a frequency vs. level response), an “impulse” is applied to the system.  A perfect impulse (an infinitely short duration “spike”) will contain all frequencies and is impossible to achieve in reality, but what we are trying to get to it a measure of the time alignments of all frequencies that the system needs to reproduce.

The output of an impulse response measurement will display level vs. time, not level vs. frequency.  Applying a mathematical method known as a Fast Fourier Transform will supply the frequency vs. level version of the information.  However, what can also more usefully be derived from the impulse response is a holistic corrective response filter that is effectively an “inverse” response to the system’s impulse response.  This filter is NOT made up of discrete “bands” and traditional filters – it is a mathematical calculation that works as a whole on the system.