24DB Low Pass Filtering - SOUND / AUDIO Processing

[some1uk03]

Hi Schmidt/Olaf,

Thanks for the update, but I'm still getting same results ? Still have a black notch in my spectrum and noise on the wav ? just like the images I posted before?

I'm only using the ButterWorth class, so I'm using it by:

Dim kk As New cButterWorth
kk.FilterType = FilterLowPass
kk.Freq = 1000     'EditWAV.SamplingFreq / 4
kk.SampleRate = EditWAV.SamplingFreq
kk.Order = 4

kk.Filter EditWAV.WavData, True

refreshData True

[Schmidt]

I'm only using the ButterWorth class, so I'm using it by:

 
kk.Filter EditWAV.WavData, True

In case you are using the updated cButterworth-Class now, the IsMono-Param is
the last argument, Optional - and by default *False* (meaning Stereo-Mode).

As I wrote in the other post - I changed that in comparison to your own variant,
to not break that much stuff in my older Demo-App.

So, the above line which finally calls the new Filter-method should explicitely pass a True in the last param:

kk.Filter EditWAV.WavData, True, True

Olaf

[some1uk03]

I'm not infront of my PC now to confirm %100 but i do remember well that i changed your updated function to...

Optional isMono As Boolean = True

..so by default its True, without me having to pass it here:

kk.Filter EditWAV.WavData, True, True

EDIT:
Just realised that in my original kk.Filter call the True goes out to the FIRSTbuff... what does that do exactly ? Splits the audio to few buffers ? havent really analysed properly...

[Schmidt]

Just realised that in my original kk.Filter call the True goes out to the FIRSTbuff... what does that do exactly ? Splits the audio to few buffers ? havent really analysed properly...

The FirstBuff-param is needed, to properly initialize the surrounding Helper-Arrays, which store the current filter-states.

If you want to play e.g. a song from a 25MB-Wav-File, then you have to do it in chunks -
now when you have the BW-Filter in place, you have to signalize when a new "streaming begins" -
and on your very first buffer you're sending to the Class, you can tell the Filter-method about that -
... and all following buffers shall have to "connect seamlessly" to the last values in the previous
buffers (and use the derived coefficients from those last buffer-values), so with Firstbuff = False
you can signalize "continuation of the current state of calculations".

If your MonoBuffers come in complete chunks, containing the whole sound - then FirstBuff should be
always True (so, False is only thought for following Buffers after the first one in a streaming-scenario).

Olaf

[some1uk03]

Ok, After further testing today... the LP / Mono is working, BUT I still get NOISE depending on the Freq / Hertz...

The spectrum this time only show a black boundary not in the MIDDLE, but TOP area only... what ever the Freq limit is set to... i.e upto 11000
as in there is No Falloff...
As far as I can calculate, there needs to be a loop which decreases the SET Freq depending on TIME...to get a side slope just like in the GREEN Mid graph i posted earlier..

Does that seem to make any sense?

[Schmidt]

Ok, After further testing today... the LP / Mono is working, BUT I still get NOISE depending on the Freq / Hertz...

Now (and from what you wrote in a PM) I begin to understand that you're looking for a Noise-Cancelling solution.

The spectrum this time only show a black boundary not in the MIDDLE, but TOP area only... what ever the Freq limit is set to... i.e upto 11000

That's as it should be with a "normal" Filter (a LowPass in this case).

A LowPass is letting (by definition) the frequencies below a certain "cut-off-frequency" through...
The frequencies above the cut-off will be "dampened" (in case of a Butterworth-Filter of the 4th order with 24dB per Octave)...
And the latter (increased) dampening is, what's causing the "blacked out area" in your frequency-spectrum-visualization.

So, there's nothing wrong with the Butterworth-Filter - its one of the better suited algorithms for audio-processing - and it behaves as it is designed to do in your case.

Noise (White Noise to be precise) spreads by definition: http://en.wikipedia.org/wiki/White_noise
its energy across the whole spectrum, so cutting-off above a certain frequency is not a sufficient
"noise-cancelling-approach", since the lower-frequency-bands (below the cut-off) will still contain
a lot of "noise-energy" - it just doesn't sound all that "aggressive" anymore, when at least the
higher frequency-bands are dampened.

What you need is a better suited algorithm which works across the whole frequency-spectrum,
cleaning up (or flattening) "the ripple" in each of the frequency-bands - a single Low-Pass
(no matter how strong its dampening is adjusted) will not work as good as such a specialized algo.

Not sure, what you need this for - if this is e.g. for a de-noising of radio-transmitted audio-signals, then maybe this link here
is helpful for a general overview: http://digitalcommons.calpoly.edu/cg...context=theses

Olaf

[some1uk03]

Thanks for coming back to this Olaf.

The butterworth-Filter in itself, is fine no problem. But yes, it seems I need some form of a Noise Cancelling solution.

From what I can figure out, is that some type of a GRADUAL Frequency Cut off ? That's what I'm evaluating from the Spectrum Graphs I posted before, but been researching and have No idea how to implement this.

Not sure, what you need this for..

I have Audio Sample Wavs that require some of this Noise Removal treatment.