Stereo Tool

Part 1:
The VLevel latency mystery

The low latency using VLevel with a buffer length of 2.0 s (default setting) inside of foobar2000 is really mysterious.
Why is there almost no delay (latency) when you click and step from one track to another up and down the foobar2000 playlist ?

You would expect that (using the default 2.0 s buffer length) VLevel should add this 2 seconds to the overall latency.
That is just the way it works with StereoTool 3.40 using Hans' settings: This adds 1.60 seconds to the latency (click a new track you have to wait 1.60 seconds before you can hear it).

But not with VLevel.
Even when you set the buffer length to 5.0 seconds.
There is no extra delay (latency).
It does not make any difference if you use VLevel or not for the overall latency of foobars audio output.

So how could this lookahead (default 2.0 seconds) work without extra playback latency ?

Any idea ?


Part 2:
VLevel - the dark side of the moon

I have done some serious listening tests with VLevel.
It does compress (with default settings) !
Technically: the only way to avoid any compression would recommend a lookahead as long as the processed track - 2 or 5 seconds lookahead automatically leads to compression of the dynamic range of the whole track.
Though this is not a compression in the traditional meaning of audio engineering (using a compressor), because it should work as lookahead AGC only -> so without any change in the (temporary) sound signature.
I thought it should work this way with VLevel.
But it does not.
Using the default settings for VLevel there must be some kind of "real" compression applied, too.

Try some older tracks with wide dynamics, long fade ins and fade outs and little noise.
VLevel amplifies this noise in a very ugly and annoying way.
I have compared it with Stereo Tool and simple Pre Amp settings to get the same
loudness as with VLevel.
Using Stereo Tool the noise is a little bit louder (dependant on the Pre Amp Gain)
than the unprocessed original, but not that annoying as with VLevel default processing on similar percepted loudness (VLevel vs. Stereo Tool 5).

The only way to avoid this extraordinary noise gain is to change the default settings for VLevel.

Instead of (default):

strenght: 0.80
max multiplier: 25 (+28.0 dB)
buffer length: 2.0 s

Use this:

strenght: 1.00
max multiplier: 3 (+9.5 dB)
buffer length: 0.1 s

This should be to 99 % very close to the default characteristic, but without that extreme noise gain.
Ironically, though this works with a very low buffer length (compared to the default 2.0 s setting) it leads to more consistent results.
I had some older tracks with originally quiet segments - using the default VLevel settings - that became louder than the originally loud segments (I think Bojcha knows this side effect, too ?).

Based on this experience with VLevel that the lookahead buffer can be relative low I have tried to use lower lookahead buffer lengths with ST 3.40, too.
Instead of 1.60 seconds I tried 0.80, 0.40 and 0.20.
0.20 s was too low, but 0.40 s works.
But ST 3.40 standard (low) latency plus 0.40 s is still too much overall latency.

But this gave me an idea that there might be another way (without big lookahead
buffers) to emulate VLevel characteristic.
This is were
Stereo Tool 5 VLevel Emulation Preset
story starts:
viewtopic.php?p=5107#p5107

Michi,
You are missing one important thing..
Winamp, foobar .. are Players. StereoTool, breakaway, Oprimod, Orban, Are Audio processors.
What that mean.

I want you to figure out yourself.

Try This!

Start Winamp and StereoTool standalone, Use VAC with WaveOut in winamp, so input for ST is VAC1 and output fow winamp is same VAC1. Use Normal output in ST. Play some track. Try to play/pause winamp you will notice "latency" !
Now go to winamp Wave output and change "Buffer Lenght" from default 2000ms to 200ms.
Again hit play/pause.

Latency is lower ?


Try Then same Winamp without ST, and try ANY "Buffer Lenght"

Realtime always ?

Bojcha and Michi,

I still haven't checked Vlevel yet, but I'm guessing here: One of the shortcomings in the Winamp plugin interface is that it's not possible to tell Winamp that I'm causing latency. If, in Foobar, that IS possible, the whole latency issue could - although it's still there - be completely hidden by Foobar.

(Say Vlevel tells Foobar that the latency is 2 seconds. Then Foobar could - upon start - immediately send 2 seconds worth of data to Vlevel before it starts playing anything. And then just ignore the first 2 seconds. Or, alternatively, Vlevel could reply 'I have no data' until it has received 2 seconds worth of audio.


Note: The latter is actually also possible in Winamp, but:

1. Other applications don't implement it properly and may start acting strangely. Example: If you're using Direct Soundcard Output, you can select to send 'nothing' back to Winamp. That uses this feature! But it doesn't always work well - even in Winamp itself... And for example SAM sometimes hangs if I do this. Even weirder: I can even send back more data than I have received. But I have to use the same buffer - so I'm writing in 'random' memory, which could fail at any time. Also not very useful.

2. I thought that this would be useful when playing movies, but if I return no data at the start Winamp skips the start of the movie, so the audio is still not synchronized.

WOWWW ....
Let me get things clear for the man with Vlevel compression .
This is not a compression Ok? This is symply normalizing the volume automatically with ....let`s say " a point of maximum level " , that`s all , not compression not everything , only auto normalizing . Don`t mess Hans`s thoughts ?!!!

This is the theory.
We wish it should work that way.
But in practice that extreme noise gain cannot be explained by a simple (2 second default) normalisation.
And this extreme noise gain is a definite indication for real compression !

For example try it (if you have this) with Earth and Fire - Weekend (I have it on a sampler called "Disco Night Fever") !

Look , In VLEVEL when bass stops and volume normalizes ........ when the bass come back the volume normalizes again before the bass kicks , In Stereotool this is opposite when the bass come back first kicks are distorted then the volume is normalized ?!? Understand ?

DJ-DOGGY, please try the latest BETA version, with default settings (hit RESET, then turn AGC on). The burst of too loud output when a loud sound kicks in is greatly reduced there. And you can also use the new gating function to stop the volume from rising when there is no or little sound at the input, for example between tracks.