XTA TechBlog

Both of them fleeting, both of them British, both of them brilliant.  One of them owned my me and in my studio and , despite what you might have assumed, it’s not the NXBoB – it’s the MPC.

A heady mix of barely 8 bit digital control and noisy analogue filters and oscillators and even a couple of noise sources, the MPC is capable of synthesising a real drum kit to the same startling levels as I am able to play a real drum kit – that is – not very well.  It honks and hisses like an angry goose, and restarts in the middle of a sequence, but despite all this I love it.  I don’t love it enough to actually sync it up any more, to be fair, so these days it’s samples of loops that make it into tracks, not real time bongs

Now let’s move forward nearly 30 years to the other product in question – the NXBoB.  After undertaking development work for MC2 for the Ti Series amplifiers, breakout boxes and processors, we did a lot of listening tests with the TiBoBs, being fed 96k networked audio, up against a DP448, and a Lake processor, also getting a networked audio feed.

Initially, the TiBoB didn’t quite seem to be on a par with the other two processors, but this was a slightly unfair test as the BoB was running at 48k, as opposed to 96k.  Blind tests were always able to pinpoint the BoB – not that it sounded in any way poor – just that it could be chosen meant that it wasn’t quite as transparent.  A software update later, to allow to run at 96k, and things became a lot closer.

Based on these results, the decision was made to rebadge this as an XTA NX product – and the NXBoB was born.  However, this was to be a fleeting product, due to the fact that we didn’t have, at the time of release, any method of getting audio onto a network and until this was resolved, we did not want to release “half” a solution.

The good news is that in the MC2 portfolio, the TiBoBs fitted in much better with the entire Ti product range, as there are amplifiers and a processor to compliment it.  The TiBoB also has a few other tricks up its sleeve as it can monitor any connected Ti amps to check they are present/working/connected correctly to their speakers, and report this info back without the need of a connected computer – a bonus for installations where keeping it simple for operators is a priority.

The TiBoBs can also operate at 48k and 96k (switchable at start-up and remembered at power down).  The unit can be configured via iCore for its amplifier babysitting duties, but if you don’t need that, it will work straight out of  the box as a multichannel soundcard or network breakout, configuring the streams via the requisite Cobranet or Dante software.

If you want to find out more then head over to MC2′s site or download a datasheet directly here!

Making its debut at ProLight+Sound next month will be the analogue to digital option card for our new DS8000 Audio Distribution System.  The addition of a couple of extra connectors on the rear panel opens up a wealth of new facilities and possibilities for the DS8000, and the best part is they are all in addition to what’s already there – so you get even more splitting and distro options!

The micro-controlled card provides 8 channels of 24-bit XTA grade analogue to digital conversion with selectable sample rate up to 192kHz.  Sample rate is adjusted via links on the board and indicated using a tri-colour LED next to the innocuous looking 25 pin D-type on the rear panel.  This socket (wired to accept any Tascam format AES breakout cable) actually has another trick up its sleeve – each AES stream (of a pair of channels) is also split and offered on ‘A’ and ‘B’ positions, transformer isolated from each other.

So, each input now can split to 2 x transformer balanced analogue outputs, 2 x electronically balanced outputs AND 2 x AES outputs – and 8 channels of this!

But that’s not the end of the story…there is also a BNC socket for word clock sync.  This can be used as an input, syncing from 32kHz up to 192k automatically, with an associated LED showing when the card has successfully locked to the incoming word clock signal.  It can also be switched to provide a master clock output (adjustable via link on card) at 48/96/192k, to allow one unit in a system to synchronise any other DS8000s (or other digital gear).

We will have a unit at the show connected up to the max with AES outputs feeding one external system directly, as well as to an AES-MADI system for direct to disk recording, and of course the standard analogue outputs – come along and see it in action…we’re in Hall 8.0 with Atlantic Audio on stand F60.

We have yet to try this out, and would love some feedback from you about it, but Jordi Vandekerkhof posted this on our Facebook page recently – he has ported AudioCore over to the mac!  Looking at the download  size (it’s over 100M – Windows install is only just over 2M! ) I can only imagine it runs within a Wine shell giving Windows emulation on the fly, but if it works for you then that’s great!

Anyway, you can find it here:

http://www.gearslutz.com/board/live-sound/807579-xta-audiocore-mac-recoded-jordi-vandekerkhof.html

Let us (and Jordi!) know what you think…

Not the pithiest title there, but one that is easily searchable…even through the blizzard conditions here at XTA this morning!
OK so currently that might be  a slight exaggeration, but it’s coming down thick and fast now and the offices may well be deserted by lunchtime

Bundled in with the FIR filtering additions to the 4 Series, we have added some very useful remote functionality to the range, and this has also been included in DC1048 firmware.  If any of you have used Crestron systems (or similar)  or had reason to need remote control of our gear outside AudioCore or iCore, we added (some years ago now) a layer of simple remote control protocol to give basic adjustment of key parameters, without the “framing and packeting” that the full remote access requires.

This simple system allows changes to input and output gains, mutes, and memory recalls.  The only slight flaw in this approach was that the gain adjustments were all absolute so, unless you were prepared to start from a known gain value and overwrite the current setting(s) in a unit, gain controls remotely could cause jumps in level.  This is not peculiar to our units – it is just a consequence of “blind” remote access (same thing would occur with for example MIDI control – any system where the starting point (current value) of a control has not been queried or cannot be determined.

Obviously AudioCore’s full protocol queries a unit for all its settings so adjustments made are from a known starting point, but this simple system is “one-way” and so cannot get the settings first.

To circumvent this problem, we have added (not replaced  - absolute gain settings are still possible) an increment/decrement command to the simple protocol that allows adjustments to be trimmed about the current value of any channel, with step size and limits also available.

There’s no need for this post to go into the details of how it works as it’s all in a TechNote on the website  here:

http://www.audiocore.co.uk/tech-support-docs/simple_serial_control.pdf

This  document replaces the previous one and expands greatly on how to construct the simple messages so even if you don’t need the new facility, it might be worth checking it out if you need to work out how to recall a preset remotely for example…

So just to be clear – this protcol enhancement has been added to all 4 Series in Version 2.20, and to the DC1048 in version 1.20 (and the MC2 version, the Ti1048).

A couple of nice little extras have also gone into the DC1048 – there is now an extra step in the LED Timeout parameter – choose for the button LEDs and LCD backlight to stay on for 5 to 90 seconds (as before) or choose “Off”  - this leaves all the lights on permanently.

This was added after some customer comments that they loved the way the LEDs and backlight faded up and down when the unit was accessed, but when the lights went off, in the absence of any audio (no if no meters showing), it was hard to tell  from  a distance if the device was actually powered up!

Also added is a virtual “COMMS” LED in the form of  the LCD backlight.  Now (assuming the LED Timeout hasn’t been set to its new “Off” position!) if the unit is accessed remotely, the LCD backlight will fade up for 5 seconds to show it’s been addressed.  If you have more than one unit, you’ll be able to see if any aren’t responding on the network – no backlight on!

All firmware is available to download in a zip file, bundled with the loader app here:

http://www.audiocore.co.uk/software/upgrdxta.zip

Now I’m off for a snowball fight in the car park!

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[i] 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 at

http://www.audiocore.co.uk/pdfs/lib_mngr_manual_v2p30.pdf

 

 

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:

http://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.