[EA Support]

EA Support

19 February 2022 at 7:19 am #427

Hi Sunil,

Version 2.2.x introduced Marani IIR filters and a “FIR output sample rate” parameter, which is different to the channel sample rate used for IIR and gain & delay. The “FIR output sample rate” parameter broke the Wavelet view for cases where the “FIR output sample rate” is different to the channel sample rate. We are currently working on a fix. (The filters are correct; it’s just the Wavelet display that is broken.)

I’d suggest confirming that the channel sample rate and “FIR output sample rate” parameter are the same. If they are the same and you still see problems with the Wavelet view, for now please downgrade to the most recent 2.1 release; 2.1.11. If you don’t have access to the downloads page, email us at the info address on the https://eclipseaudio.com/contact/ page.

And if you have time, can you email us your system (fds) and project (fdp) files.

Best regards,
EA Support

• This reply was modified 4 years, 8 months ago by EA Support.
• This reply was modified 4 years, 8 months ago by EA Support.

[EA Support]

EA Support

19 February 2022 at 6:45 am #425

Thanks for letting us know. We’ll take a look at this. In the meantime we’ll email you a direct link.

Regards,
EA Support

[EA Support]

EA Support

18 February 2022 at 6:15 am #421

>> I use FIR Filters for Hifi Equipment. Do you have an good idea for me which
>> FIR Equalizer works fine to build an active crossover with more than 1024 taps
>> for each channel?. Balanced Outputs desired if possible. Mini DSP2x4HD is fine,
>> but I would wish in first order more channels and more taps if possible. Also
>> better sound quality should be fine..

Sorry, we don’t have the time to keep up-to-date on the latest processors. MiniDSP have a few multi-channel products that may run longer filters, and a number of their products support balanced IO.

Regards,
EA Support

[EA Support]

EA Support

18 February 2022 at 6:06 am #420

In short, Power Averaging averages the magnitude response only and ignores phase. Complex Averaging includes phase, and so frequency regions with substantially different phase will partially cancel and result in lower level in the averaged result.

When combining room measurements to EQ a single loudspeaker or loudspeaker cluster, Power Averaging is usually more appropriate. When measuring drivers and designing crossovers, Complex Averaging may be more useful.

Regards,
EA Support

[EA Support]

EA Support

16 February 2022 at 7:51 am #414

Hi David,

The second and fourth images give a clue.

The second image shows the “Design sample rate” set to 48 kHz and the lowest frequency response peak, in the FIR filter, is close to 1 kHz. The fourth image shows the same magnitude response shape but the lowest frequency peak is close to 2 kHz.

The MiniDSP2x4HD runs at 96 kHz and so it’s treating the FIR coefficients as if they have been designed for 96 kHz. In FIR Creator, change the “Design sample rate” to 96 kHz, then export and import the FIR filter again.

Best regards,
EA Support

BTW Thank you for providing clear screenshots and a very clear description. This really helps.

• This reply was modified 4 years, 8 months ago by EA Support.

[EA Support]

EA Support

5 February 2022 at 8:03 am #406

>> Why no Auto Mag and Phase?

If Auto Mag and/or Auto Phase are enabled, then when the measurement is loaded, the Auto tabs change the filtering to push the measurement towards target. At which point you’re no longer looking at your FIR filter applied to the measurement; rather you are looking at the measurement pushed towards your FIR filter which is interpreted as a Target.

By approximating the FIR filter first, using the Mag Adjust and Phase Adjust, before loading the measurement, the filtering doesn’t change when the measurement is loaded.

These steps are only a crude approximation for what you want. Ideally we need to implement a view that shows a FIR filter directly applied to a measurement.

Best,
EA Support

[EA Support]

EA Support

12 January 2022 at 11:15 am #397

Hi Lukas,

To the right of “Auto Mag Correction” and “Auto Phase Correction” there are “Enable” checkboxes. Unchecking these will save you having to turn off each individual frequency band, however these enable checkboxes are per-channel, and would need to be toggled for each channel.

We understand what you are describing w.r.t. performance, especially if you are using mouse control of filters (which triggers continual updating of everything). We’ll consider a global enable/disable idea. Thanks.

Shortcuts, also a great idea. We’ll look into it.

Best regards,
EA Support

[EA Support]

EA Support

5 January 2022 at 6:53 am #394

Hi Lukas,

Our apologise we haven’t kept the Target file formats matched. We’ll remedy that in the next update.

Best regards,
EA Support

[EA Support]

EA Support

21 December 2021 at 8:23 pm #391

The best placed to start is the “FIR Designer 3 Overview” video on https://eclipseaudio.com/fir-filter-design-tutorial/. After that, take a look at the “FIR Designer M (Multi-way) Overview” video.

Best regards,
EA Support

• This reply was modified 4 years, 9 months ago by EA Support.

[EA Support]

EA Support

4 December 2021 at 6:14 am #387

The required number of taps increases for lower frequencies and with higher Q filtering. Consequently LF drivers will need longer filters than HF drivers. There isn’t any clear rule of thumb. Just increase the filter length until the processing works for your design.

Best,
EA Support

[EA Support]

EA Support

3 December 2021 at 10:05 pm #385

Your original 48 kHz filter was 27 ms long. In the most recent screenshots your filter is only 6.25 ms long. I think you’ll need to increase your filter length. (See my previous comment above about doubling the tap length, when changing from 48 kHz to 96 kHz, to maintain the filter length in ms.)

Regards,
EA Support

[EA Support]

EA Support

3 December 2021 at 11:47 am #379

Could you send your System and Channel files to our “info” address and we’ll take a look.

In 2.2.3 we fixed a bug where the top “Enable” checkboxes on some tabs were sometimes not refreshing correctly when a channel project file was loaded, but we haven’t seen this with Systems.

Best regards,
EA Support

[EA Support]

EA Support

3 December 2021 at 6:21 am #375

No you don’t need to remeasure. (The sample rate of the measurement does not need to match the FIR Designer sample rate.) Just change the sample rate in FIR Designer to 96 kHz and take a look at the Export tab – you’ll probably need to double the filter delay and length settings to match the time intervals at 48 kHz – then export the filter again.

Regards,
EA Support

[EA Support]

EA Support

2 December 2021 at 5:27 pm #373

>> i was doing 48khz sample rate while the xta is native at 96k.
>> Do i need to redo my measurement as well in 96k?
No, just change the sample rate in FIR Designer to 96 kHz and check the Export tab – you’ll probably need to double the filter delay and length settings to match the time intervals at 48 kHz – then export the filter again.

>> …. are these correct?

Yep, I think that’s a good starting point. We have tutorial showing a 2-way design with only FIR at

Designing FIR Crossover Filters for 2-Way Loudspeaker

This is just one approach to doing a FIR 2-way. Like I said above, another way is to do a regular IIR crossover design, then measure the whole cabinet and create a single FIR correction filter for the whole speaker. Some processors can only do a single FIR before crossover. Some can run a FIR for every output.

Best,
EA Support

[EA Support]

EA Support

2 December 2021 at 2:43 pm #371

Since you’re trying to flatten the phase from about 200 Hz upwards, I think it’s ok to leave the 100 Hz HPF engaged.

The bigger issue is the crossover and the relative physical placement of the drivers in the cabinet. Generally you can’t ignore these and expect to linearise the phase of the LF driver from 200 Hz up to crossover. The crossover filtering itself and the relative physical placement of the drivers both impart phase change through the crossover region. And so you need to measure the system as a whole. And like I said above, the manufacturer supplied crossover would have been designed with dispersion characteristics in mind.

The only time it may be possible to linearise the drivers separately is if you use a linear-phase brick-wall type crossover (like in the Lake processors). However you’ll likely still have a narrow frequency phase jump at crossover.

The 300 Hz versus 600 Hz issue could be that you’re designing the filter for 48 kHz sampling but the processor is running it at 96 kHz. Try changing the sample rate (in FIR Designer) to 96 kHz and see what happens.

Regarding question 3, FIR filters can be minimum phase (with no coefficients prior to the peak), linear phase (with coefficients symmetrically either side of the peak), maximum phase (with the peak at the end) or a mixture of all. Phase processing, as you are doing, generally results in a mixed phase filter. The “Filter delay” parameter enables the peak to be placed to minimise the error between the ideal FIR filter, and the filter windowed and truncated to the number of taps available (e.g. 1300). For more details, check out the links and tutorials referenced above.

When phase linearising using a FIR filter, the net effect is that the loudspeaker will have a bulk delay of approximately this “Filter delay” value. And if you are doing this on just one driver, the other driver will need to be delayed by the same amount. However see my previous comment about measuring the speaker as a whole.

If you measure the loudspeaker as a whole, you can still design the FIR for, say, 200 Hz to 1 kHz. The important part here is that you’ve measured WITH the crossover in the measurement/s.

Best,
EA Support