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Topic: foo_dsp_convolver (Read 38427 times) previous topic - next topic
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Re: foo_dsp_convolver

Reply #50 –
Hi,
I am trying to use the plugin with a four channel impulse, build with one stereo impulse duplicated with Audacity in a 4 channel wave file, but in the end, after loading this in foobar, only the first and fourth output channels are working. the 2 middle ones are 'muted'.
any info to help me to solve this ?
thanks,
Laurent

Re: foo_dsp_convolver

Reply #51 –
Quote from: aquascop on
I am trying to use the plugin with a four channel impulse, build with one stereo impulse duplicated with Audacity in a 4 channel wave file, but in the end, after loading this in foobar, only the first and fourth output channels are working. the 2 middle ones are 'muted'.
Does audio signal that you are trying to process with convolver has 4 channels too?

Re: foo_dsp_convolver

Reply #52 –
Hi,
Yes. my DSP processing chain is :
Resampling to 96khz
Foo_dsp_convolver,  with stereo impulse for phase control
Stereo to 4 channels
Foo_dsp_convolver,  with quad impulse for activ crossover

When i remove the second convolver, i have my 4 channels on my soundcard (Digiface) (L/R/L/R)
As soon as i insert it, i have only the 1st and the 4th activ.

Could it be linked to a 24/32bits conversion somewhere that i could have done wrong ? I would like to test with a 'blank' 4 channels impulse (like a passthrough), but how can i build it ?

thanks,

Laurent

Re: foo_dsp_convolver

Reply #53 –
Upload your quad impulse somewhere, and I'll take a look.

Re: foo_dsp_convolver

Reply #54 –
Quote from: aquascop on
I would like to test with a 'blank' 4 channels impulse (like a passthrough), but how can i build it ?
In you not have blank 4 channels impulse, how did you build your own 4-channels impulse at all? Blank impulse with samplerate 96 kHz in attachment.

Re: foo_dsp_convolver

Reply #55 –
Hi,
Thanks for your feedbacks. Following the idea of the blank impulse ( i never realized i could do this simply with Rephase), I played with various formats of blank impulses, simply at 44.1khz in order to simplify my processing chain :

Stereo blank impulse created by Rephase
With Audacity, I copy paste the impulse to create a quad impulse

And in foobar, my DSP chain is just
Stereo to 4 channels
Foo_dsp_convolver with various blank quad impulses

i played with the number of taps (from 8192 to 524288) , and the format (24bPCM to 32bfloat)

I succeeded to make this working with my four channels ... but to a certain limit of size and format of the pulse : After many many (...) trials, i realized that with big pulses, when i load the pulse into foo_dsp_convolver, sometimes, the Sample length is totally wrong. Please note that my goal is to use quite big impulses ( 131 000 taps or more).

I therefore came back to stereo, and compared stereo pulse loadings between convolver(gapless) and the original convolver. Conclusion is that with the original convolver, the number of samples is always good, whereas with convolver(gapless), sometimes the number is crazy ( like even 1 sample, 2 channels, 44.1kHz).

it looks to me there is a problem, or a limit somewhere in the pulse loading that is unclear to me ...

Any idea ?

Re: foo_dsp_convolver

Reply #56 –
Quote from: aquascop on
whereas with convolver(gapless), sometimes the number is crazy ( like even 1 sample, 2 channels, 44.1kHz)
This convolver automatically detects excessive digital silence in impulse and ignores it.

As lvqcl already said: share your actual problematic impulses files.

Re: foo_dsp_convolver

Reply #58 –
Something gone wrong with this impulse Open it in Audacity and see yourself. Peak level in 2nd and 4th channels is -40 dB. So, yes, these channels are almost muted.
.

As for convolver, it works as expected.

EDIT: Sorry, they are not muted of course. I can't reproduce your problem. Using Convert stereo to 4 channels and then convolver with your impulse, no channels in output are muted.

https://youtu.be/xjy8jPuwxCw

Re: foo_dsp_convolver

Reply #59 –
Quote from: aquascop on
After many many (...) trials, i realized that with big pulses, when i load the pulse into foo_dsp_convolver, sometimes, the Sample length is totally wrong. Please note that my goal is to use quite big impulses ( 131 000 taps or more).

I therefore came back to stereo, and compared stereo pulse loadings between convolver(gapless) and the original convolver. Conclusion is that with the original convolver, the number of samples is always good, whereas with convolver(gapless), sometimes the number is crazy ( like even 1 sample, 2 channels, 44.1kHz).

it looks to me there is a problem, or a limit somewhere in the pulse loading that is unclear to me ...

Any idea ?

The impulse file that you uploaded has 524288 samples. To optimize performance, this plugin cuts too quiet samples from the beginning and the end of an impulse. That's why it shows "2917 samples" - that's all that was left after trimming.

Besides that, the plugin works as expected, no channels are muted.

BUT: the order of channels in foobar2000 is Front Left, Front Right, Back Left, Back Right. With your impulse, channels 2 and 4 (i.e. Front Right and Back Right) are lowpassed. Probably that's not what you want, so you need to swap channel 2 and channel 3 in your impulse file.

Re: foo_dsp_convolver

Reply #60 –
first of all, thanks lvqcl and Rollin for your time spent on my questions. I really appreciate.

So this convolver is not a straight convolver. It works 'on its own' on the input impulse, i did'nt know that. Does it also works on the input signal to adapt/simplify the convolution ? i Thought that the 'no delay convolution' was obtained by a clever buffer inside the dsp, but it seems not to be the case.

Do you also know the process used to simplify the impulse ? because i find it a little bit stange on the 2 blank pulses bellow. one in 24bit PCM, the other in 32bit PCM. Exactly the same file (from rephase) in audacity, but 2 ways of saving it. the first one, is read with 16k sample, what i would expect. the other one ... 1 sample .... stange no ?

However, if i go back to my initial question with my quad impulse file : it is also strange, because i clearly confirm that the quad impulse file i transferred give me a complete mute on channel 2 and 3. I am using foobar 1.6.5 on a w10 64b system.

>With your impulse, channels 2 and 4 (i.e. Front Right and Back Right) are lowpassed. Probably that's not what you want,

No, that's exactly what i want. Channel 2, is in fact Right low band, Channel 3 Left High Band, and CHannel 4 Left low band. that is how it is connected after on my multichannel DA.

Re: foo_dsp_convolver

Reply #61 –
Quote from: aquascop on
because i find it a little bit stange on the 2 blank pulses bellow. one in 24bit PCM, the other in 32bit PCM. Exactly the same file (from rephase) in audacity, but 2 ways of saving it. the first one, is read with 16k sample, what i would expect. the other one ... 1 sample .... stange no ?
They are not the same. 24 bit file has noise. You cannot see it in Audacity with default settings but it is there. Samples with noise are not silent samples, so they aren't trimmed. Also, phase in 32 bit file is inverted, but this is irrelevant in this case. See spectrograms for noise.
24 bit file:


32 bit file:

If in Audacity you will set Range for spectrograms to 180 dB, this noise will be visible too.

Quote from: aquascop on
However, if i go back to my initial question with my quad impulse file : it is also strange, because i clearly confirm that the quad impulse file i transferred give me a complete mute on channel 2 and 3. I am using foobar 1.6.5 on a w10 64b system.
What happens if you use only two DSPs - Convert stereo to 4 channels and Convolver with your quad impulse? What happens if you use only one DSP - Convert stereo to 4 channels?

Re: foo_dsp_convolver

Reply #62 –
Quote from: Rollin on
/quote]What happens if you use only two DSPs - Convert stereo to 4 channels and Convolver with your quad impulse? What happens if you use only one DSP - Convert stereo to 4 channels?
Hi,
With only stereo to 4, i have all my 4 channels. and with the addition of dsp convolver with this particular filter, channel 2 and 3 are muted. But i build another quad filter, and this last one works well, with 4 channels ok. So there is something wrong with my first quad impulse, but what ?

Thanks also for your analysis of the 2 Audacity recording : conversion to 24bPCM is surprisingly creating low backgroung noise ... I will avoid this conversion and stay in 32bit float.

Tonight, i tested carefully the audio quality, with a stereo impulse (phase shift compensation), btw standard convolver, and dsp_convolver, and the advantage is clear for standard convolver. Is it normal if both convolver work from the same impulse ? how do you explain this ?

thanks,

Re: foo_dsp_convolver

Reply #63 –
Precision of processing is higher than human perception in both convolvers. But standard convolver has some limitation. So, most probably, standard covolver does not what your think it does in this case.
Show your DSP chains and settings of both convolvers, and share impulse. So i can tell you what causes differences.

Re: foo_dsp_convolver

Reply #64 –
[quote
Quote from: aquascop on
Tonight, i tested carefully the audio quality, with a stereo impulse (phase shift compensation), btw standard convolver, and dsp_convolver, and the advantage is clear for standard convolver. Is it normal if both convolver work from the same impulse ? how do you explain this ?

Sorry for this, there was a gain bias btw the two settings, so the difference in quality is finally not so clear ! i need to take time to listen and compare. There is definitely a difference btw the 2 processes. so i added a VST altenativ with convolverVST to compare and also measure the resulting impulse and phase correction. I'll report the results.

thanks for reading

Re: foo_dsp_convolver

Reply #65 –
Can someone recompile it for foobar2000 v2 and 64 bit?

Re: foo_dsp_convolver

Reply #68 –
+1, I would really appreciate a 64-bit version.
I cannot use the stereo convolved as I need 8 channels.

Re: foo_dsp_convolver

Reply #69 –
Just as an FYI: Foobar has a very nice VST host component.
So it's possible to use convolution VST effects, including free ones like MConvolutionEZ or Freeverb.

I used the convolver in Foobar for headphone correction, but I then switched to using Equalizer APO and I like it better this way.