Correction for previous request, I have no idea how the hell this would work, if at all.

Of course, I can see the relevance of summing stereo impulses of individual channels, perhaps for processing each channel with its own HRTF impulse. Of course, that'd be doubly slow for plain stereo... double again for 4.0...

How does one go about converting Sound Forge impulse files (.sfi) to .wav format for loading in FB2K?

Feed that unitpulse2k.wav through it. You can trim it down to the single pulse sample first, as the output will be extended to the full length of the impulse.

MIX ADJUST not work fine here.

Load impulse file, for exemple unitpulse2k
(because problem is very clear)

put slider to 0 %
play
sound isn't different from sound without convolver (fortunately...).

put slider to 100 %
sound no change (logically!)

put slider to 50 %
I can not explain in English what I hears but it is catastrophic!

same problem with all impulses files
foobar 0.667 and 0.7 b22

please, can you confirme if it is a bug or if i have a problem in my system?

Melomane, read whole thread - I had the same "problem". Unitpulse is not really unitpulse (Dirac delta) but unitpulse plus const., so it works as delay filter.

Thank you Messer, now i understand.
i have use file from convolve.zip (0.6 foobar) and all are wrong.

Garf, just wondering...
If the impulse contains response that comes _before_ actual pulse (some pre-echoes etc.), then there's no way to compensate delay by editing impulse file only...

There is also no way to compensate for that on output, unless you want to cut off the pre-echo of the first few samples.

kode54 is correct. It's also reason why the included unitpulse2k is not a true dirac pulse, so effects that influence before the impulse have some 'room'. I noticed that otherwhise most effects aren't able to work well.

If you have an effect with delay and want to have a delayed wet/dry mix, work with the impulse in an external wave editor.

Nah, that's not really a whining, I was just "thinking aloud" about possible situations where adjustable delay could be handy Not that it's worth the hassle...

Garf, in case you need some more whining as motivation

Could you please, pretty please, look into adding crossmixing?

Where can I download the latest foo_convolve plugin?

It's included in the special installer from Case's site or available at official fb2k components site.

Thanks for the links.
I couldn’t find it on http://fb2k-plugins.hydrogenaudio.org/

I was wondering the processing pipeline of FooConvolver?

Is it:-

Impulse converts to 64 bit float and then is processed with what is in the playlist (which is also converted to 64 bit float).

ie:- 64 bit float x 64 bit float regardless of format the originals were in.

all processed using 64 bit float precision?

Also what happens when the samplerates do not match?

All processing is in 32 bit float precision. (There is no quality advantage in 64 bits and it takes up more memory).

When samplerates don't match the samplerate of the output takes priority, and the impulse is treated as having that samplerate (so basically you want to avoid that unless you want to try to get some funny effects).

So does anybody have a good impulse for use with Sennheiser HD580's, an audigy 2 using kernel streaming, and pop/rock music? (weezer, red hot chili peppers, beck, cake, radiohead, etc) (and some classical/techno of course).

Try UnitpulseDirac.wav.

Ahaha, touche!

The HD580 is great without equalization or impulses but sometimes the perfect lineariry does get boring. In those cases I kinda like "Tube Amps/Studer +6dB Updated" for a warmer feeling.

You need at more sophisticated approach for this.

Try DRC program by Dennis SAbrigion. It can be found at Freashmeat

Morten

Is it possible to add support for one impulse response per sample rate ?

I use an impulse response in order to correct the equalization of my speakers, because the equalizer don't have a parametric section. But I have both 44100 Hz and 48000 Hz files in my playlist.

It would be nice if the convolver would use toto44.wav as impulse response when a 44.1 kHz file is played, and switch to toto48.wav when a 48kHz file is played.

Exhibited *very* strange behaviour.

Upon first run with it, attempted to go into prefs, and it foobar crashed as soon as I click "foobar2000".

Removed *all* componets, and it worked fine. So i slowly added them all back in, and now it works fine. lol. *shrugs*

Steve

Shouldn't correct and easier behaviour be to resample the impulse response instead?

I don't know.
If you know as a fact that it works (=produce the desired effect) it would be the easiest solution by far, yes, though some people might prefer having the freedom to choose their own resampler, or to record the impulse again.

That will be true to some extent, since I'd just use foobar's resampler services.

After reading about foo_convolve, I decided to use it to emulate the Winamp built-in MP3 equalizer. The problem is that the impulses are wav files. I'm not good with theory, I mean can I convert the wav impulse to mp3 and feed it through Winamp? If this is not possible, do I have any other option?

Just feed the wav through winamp??

I think Wizard is suggesting that Winamp has a format specific equalizer, that is triggered only when playing that particular format. Hence a wav file won't work if you want to capture the settings used with mp3 files.

I don't have a clue whether it works this way though...

Exactly upNorth! Winamp's in_mp3 has an option "Fast Layer 3 EQ", which is only for MP3 playback and I have a certain preset that I'd like to emulate.

Hi,

Garf, you planned to add cross-channel convolution, have you dropped the idea?
Please, if at all possible, that would be very very good thing.

All those crossfeed plugins are too limited. 2 stage convolution would be ideal to reach any desirable result.


To make decent headphone listening with convolve plugin, you'd need to convolve as you crossmix.

Using crossfeed plugin after convolve does not do it right. You really need to manipulate crossfeed data in both time and frequency response.

http://headwize.com/tech/headrm1_tech.htm

will there pleeease be a 0.9 version when it's final ?

Could Garf or someone else tell me what the functionalities of this convolver are?

Am I right that if you have a stereo signal it expects a stereo convolution and convolves each channel of the original signal with the corresponding channel of the convolution file?

If you had an n channel original and an n channel convolution would that work too?

Presumably in general convolution will take an n channel original and convolve it with an n*m matrix of mono signals to get an m channel result. This convolver doesn't do that I don't suppose?

And are all sample rates supported as long as the original has the same rate as the convolution?

Thanks for any help.

I gave Peter the source code so all new versions should include it by default.

Bump. If this convolver just convolves with a single mono convolution is any convolver more flexible, for foobar or any other playback software?

Bump

Is it still in mono? Would be useful to have it in stereo, due to room asymmetry. Seems like a small change (?)

And what about foobar2000 0.9 support? Because as of now foobar would be as useless as amaroK, which would be better since its a good native Linux player... but I need convolution, and the probably only way to get that is through the insanely difficult BruteFIR... tried many days, couldn't get it working... (though I'm making progress...).

I was wondering if convolver in 0.9 will support 5.1 or more channels in the impulse file.

Any news on 0.9 convolver?

This is the last thing I need to feel at home with 0.9. Any new news?

same here
also waiting

i'm also missing the 0.9 convolver. any plans on updating it?

I would go down on my knees if I have to.

Please, please release a 0.9 ver!

I've been waiting this plugin also and I need it almost desperately.

is there ant plan for the development to support foobar2000 0.9?
I miss this plugin so much

The convolver is not part of the standard foobar install.

Is there any chance that the author for foo_convolve could re-compile it for v0.9 foobar?

Thanks!

it's available now @ http://www.foobar2000.org/components/index.html

Regarding the way Equalization works in Winamp
Winamp's input API allows for Equalizer handling in the input plugin, but you have to go way back to find a Winamp that required the input plugin to Equalize the audio. Most Winamp versions has it's own integrated Equalizer that sits between the DSP and output stages. Nullsoft even improved this Equalizer with new code a while back.

How is this relevant to the discussion? I hope you realize that the last mention of equalization in Winamp is from December 2004.

I did not look at the post date sorry. I will start to look at the post dates from now on.
But why are there so old posts still around here? Time to prune the forum database perhaps?

Clean everything from before 2005 and maybe put really important info in a Wiki.
Wikis are for archival of information while forums are for discussions of up to date topics.

Thank you!

Now I wonder how to learn how to make Impulse response files from scratch using Adobe Audition.