In fact that awesome that I’d like to switch presets via websocket, so I can use my IR or the Rotary Encoder II plugin to toggle presets. Is that already possible somehow, or more likey a feature request?
I’m using the plugin to accommodate for different sitting positions that I use in my listening room. But when I save each location in the presets, it will remember the left/right channel gain settings, but it won’t remember the left/right speaker distance for each seating position, and it will just default back to the last one I entered.
Is it possible to make separate EQ for left and right? If you could do this you should add a button to “make left and right the same” so its easy to duplicate the settings on each channel left to right, and from there, we could make small adjustments at each desired frequency.
Of course.
With graphics Eq, just choose 2x15
With parametric Eq, choose the scope left,right or left+right.
Have a look at the online help FusionDsp How-to
for this time I use an Highpass Filter (60HZ,1db). Because I need a LowCut
around this range.
Now i have a lot of Clipping when there is Music with deep Bass.
What i have to do?
I think i could set Gain in Mixer (CamillaDSP) from -1 to -2, or is there another way to fix it?
You can set it directly in FusionDsp.
First, is auto preamp enabled? It should be enough to avoid clipping.
You can check by setting FusionDsp to ‘pure Camilla gui’ and check vu meter and clipped sample information.
If still clipping, in FusionDsp add attenuation for both channels.
Let me know.
I have to put the Volume in FusionDSP to -4. Then in CamillaDSP there is no more Clipping counting.
So, I am not really confirm with all Filters. I know by some, when you cut a Frequency, there will be a little Mountain before the cut coms. Could be, when I cut at 60HZ, that 65HZ to 80HZ has a litte pusch from 0-3db.
Dont know.
No i don‘t use it.
Yes, i think thst is normaly. And when Q is much higher, the bump will be higher too.
It seems, that auto preamp could not fix clipping for this.
But, it is no problem to set the gain manually.
Automatic preamp use positive values of filters to compensate. In this case, no positive value…
In earlier version of FusionDsp, a défaut -2dB was use to avoid such problems. But as some users complained about it, I removed this behaviour.
So, you have to add a manual attenuation…
Hello all!
I received several question about convolution filters creation for FusionDsp.
I’m preparing a “how-to” but it is not ready yet…
By waiting, some information.
You can fully create filter using REW. You make a measurement, inverse the impulse, apply an house curve and export the result a wav (ok, there is more steps ). Then place it filters folder and use it directly as filter.
You use DRC-FIR in the plugin. First make measurement in REW, export impulse as wav and use it place it in filters-sources folders of FusionDsp. Choose it in DRC-FIR section, choose a target curve, samplerate etc… Generate filter. Once done, load as filter.
A suggestion for everyone benefit might be to have a “step 1” instructions for creating and exporting required files from rew to fusiondsp.
“Step 2” then can be the in depth documentation.
Step 1 gives good sound improvements with only modest knowledge Required. Step two takes the sound to the next level But requires more knowledge in steps.
This has the advantage of giving Relief immediately giving more time to write the complete documentation.
So for step one. We would have to define the exact steps in r e w to create the impulse response file that will be imported into fusion d s p. This could entail some simple operations to tune the impulse response file before exporting to fusion dsp. Such as frequency dependent windowing , smoothing , minimum phase, Sampling resolution and inversion. Some of these may be handled internally within drc so we only need to know rew pre-processing steps that add value and not replicated internally. I could offer to assist writing this step one guide if helpful.
So from a minimalist point here is my workflow subject to advice:
I set microphone in exact equidistance from both speakers.
Record l and r measurement sweeps - for me 96khz and 1m samples.
A third l+r sweep is optional which might get used in averaging in step 2.
Any Important step one operations such as fdw, averaging, inversion need to be defined in detail. If only inversion is required specify exactly that step eg inverting over what?
Export IR in .wav format for fusiondsp. I am ticking off 96khz, 48khz, 44.1khz for 3ea. l and 3ea. r files to be placed in filter_sources folder.
Filenames must have no spaces.
Optionally, if i want to use a house curve other than what is pre-loaded in fusion dsp, in mycase dr. Otoole curve, I strip all the pre and post preamble and make the file the same as the format on the pre-stored curves. Again, i make sure that no spaces in the name.i
I process the ir’s with drc gui which gives me the 3 most common sample rates natively, and when I do play a few 192kz recordings it upsamples. Need to turn on resampling?
I think that a root issue is that there are many approaches to this issue and we need to marry the simplest most effective process Into drc.