Stereo Tool

And the fact that you set 'Extra loudness - Allow extra lows (advanced)=103.90461731' implies Allow Louder Lows is still functional? The GUI doesnot feature the control/slider.

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In our program there is lot of tracks with Higher voice over music ... And this irritate me for long time. I know for Clean, but there is no punch then.
I have now new preset that have better control over voice. but there is still problem when bass kicks there is ALWAYS distortion in that moment, Even this new filter "Clip lowest bass" need time to start its job.
That's why i make requst for AGC2 after AGC1 with hope to solve this..
Again i think that will now be simple.
Second AGC need to have different reaction for different freqs. (e.g. 40Hz -8dB, 90Hz - etc)
I know that, because i use 2nd AGC now in preset, with fast down speed and small treshold, to intercept loud sounds from multiband (and bass bost), That AGC reacts always on bass more. More or less 2nd AGC doin' good job, but there is another problem..
Till 1st AGC reacts, multiband is up, and when sudden sound comes now multiband need to go down fast...
That's why i ask for GATE in multiband. Even with that gate, now 1st AGC need to go down fast in that moments..
So, i skipped all this and ask for 2nd agc after 1st with hope to solve that as much as possible. Same as in Loudness window.

And important! Strict limit for bass! It will always left place for other sound! When ANY bass kicks it will kick at 95% and voice shouls stay undistorted.

errors .. I'm sleepy

I know. Actually I think the whole Multiband thing needs a redesign with all kinds of new features in it. But that's for the next release... Same for 2nd AGC.

(Why? Because the current sound is already so much better than that of v5.02, and that's still the version that most people will download. So I really want to update it quickly).

For now, I'll try to add strict bass clipping for Punch tonight. That should help a lot - with Clean there is now almost no distortion left. And I think the same is possible with Punch (although the Punch itself will get reduced a bit)

One small thing:

Would it be complicated to remove this very imprecise rounding (10 Hz steps)
for low frequencies in multiband ?
Could it be replaced with a precise rounding (1 Hz steps ! or better 0.1 Hz if possible ?) for low frequencies ?

For example if you use 64 Hz bass band in .sts then it should be visible in the GUI as 64 Hz and not as 60 Hz (because this is really a big difference = 4 Hz in low bass) !
Until now if you use a preset of another user and see 60 Hz in the GUI it could be 55, 56, 57, 58, 59, 60, 61, 62, 63 or 64 Hz !
Of course I can look into the .sts file to get the precise frequency.
But it would be easier, faster and more elegant to see it in the GUI.

Michi: In practice, the frequency is so unprecise that it really doesn't matter much. At the highest quality setting (4096 samples) at 44.1 kHz, the filter itself has steps of 10.77 Hz (at 2048 samples it's twice that, at 1024 four times etc.). And even then I cannot filter THAT steep without getting horrible artifacts.

Except for that, for now I'm just supporting more precise frequencies for 'expert' users, and I want to keep the display 'clean' for others. That's why I'm rounding them.

wOw i missed a new BETA.... and i am anxious to test it on Monday...
Hans also consider Deep BassBooster...!

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That should still work?

Added a strict bass clipper to Punch.
New slider added at the bottom of the Loudness window; punch bass clipper works only on the lowest frequencies and follows the same dynamics (so no highs -> no clipping).

Set the bass punch clipper at maximum (200%) to turn it off.

PUNCH CLIPPING ONLY WORKS IF 'STRICT' IS ENABLED!

Winamp DSP plugin: http://www.stereotool.com/download/dsp_ ... 510-57.exe
Stand alone version: http://www.stereotool.com/download/ster ... 510-57.exe
VST version: http://www.stereotool.com/download/vst_ ... 510-57.dll
VST version (No SSE2): http://www.stereotool.com/download/vst_ ... 510-57.dll
Command line version: http://www.stereotool.com/download/ster ... 510-57.exe
Linux command line version: http://www.stereotool.com/download/ster ... BETA510-57 (not available)
Linux GUI version: http://www.stereotool.com/download/ster ... BETA510-57 (not available)


Edit: I made a bug that increases the effect of loud 'S' sounds on mid frequencies. And I have the impression that this difference is noticeable.
The good news is that I can also do the opposite and improve this filter. New BETA expected later tonight...

Maybe a stupid question:
So does this mean that it does not matter if I write 55 or 64 Hz to .sts and always get a 60 Hz band ?
WYSIWYG in the GUI (using 4096 samples) ?

Or does "the filter itself has steps of 10.77 Hz" means that all frequencies in a range of 10.77 Hz are processed in the same way ?
(In my own untechnically oriented words) the head of a band is 10.77 Hz broad (using 4096 samples) ?
So do I get with a written 64 Hz in .sts such a band head from 59 to 69 Hz (64 Hz centered) or from 55 to 65 Hz (with rounded 60 Hz as center) ?

And another (maybe again) naive question about steepness:
Besides the given (chosen) bandwith (octave range) are there any other parameters that have an influence to the steepness characteristic ?
So different steepness characteristics (less sharp and more bell-shaped) without a change of the bandwith ?

Hi Michi:
I've checked the code and I see that there are some issues if you attempt to make very small bandwidth bands. Here's what happens:

1. You specify a frequency X, say 100 Hz
2. Then I first determine the top to be 0.9 * freq .. 1.1 * freq, so 90-110 Hz. Let's call 90 'start' and 110 'end'.
3. Next, I draw a straight line from the 'end' frequency of the band below it (amplification 0) to the current band start freq (amplification 1).

So say the previous band was at 45 Hz, then the top would be at 40-50 Hz, and I would draw a line from (50 Hz, 0%) to (90 Hz, 100%)
From there it stays at 100% upto 110 Hz.

4. From the end point, a new line is drawn to the 'start' frequency of the next band.

Each frequency is rounded down to a multiple of 10.77 (more precisely, 44100 / 4096, assuming 44.1 kHz input; at 48 kHz input it would be 48000 / 4096). Very high frequency input levels are downsampled /2 or /4 before the processing start, so 192 kHz is treated here equal to 48 kHz.

So in the end (50 Hz, 0%) - (90 Hz, 100%) - (110 Hz, 100%) - (??? Hz, 0%) results in
(43.08 Hz, 0%) - (86.16 Hz, 0%) - (107.7 Hz, 100%) - (??? Hz, 0%).


In other words: Each frequency that results in the same rounded down end value is treated identically. So whether you write 44 Hz or 48 Hz or 52 Hz doesn't matter (at least not at 44.1 kHz...)


One more thing: If "Flat Frequency response" is enabled, the band is first compressed/limited based on the entire signal (that is, (50%, 100%) - (??? Hz, 100%)), and only afterwards the reduction on the sides is performed.