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 Post subject: Re: Stereo Tool 6.00
PostPosted: Wed Jan 05, 2011 11:56 am 
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I tried this latest beta with 1024 latency and sounds good... but when i change latency to 2048 sound like a mess and i think 2048 sounds better.. :(

The trick is that you first select 4096 and then 2048 to sound good.

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 Post subject: Re: Stereo Tool 6.00
PostPosted: Wed Jan 05, 2011 12:14 pm 
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Quote:
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FM Output latency. That's important! And there is one thing there. Standalone.
At buffer size i can't set it under 0.02sec using Juli@ and Asio. 0.02sec mean 20ms. Is it really 20ms? I know that asio can go almost to 2ms (4ms safe).
If i try to set under 0.02sec stuttering occures. Just checked again at transmitter site 0.017sec is minimal what i can set, currently it is at 0.018.
With what latency setting is this?
Processing takes some CPU time. If the latency is higher, more data needs to be processed - in bursts.
If this occurs at latency 512 etc., something is really wrong. At latency 4096 I would expect something like this. (Basically, the minimum setting would be 4096 samples / 2 * CPU load, so for a CPU load of 50% (at the core where ST runs), that would be 2048 / 44100 * .5 = 23 ms.
If the latency is 512 samples, the minimum ASIO latency would drop to 256 / 44100 * .5 = 3 ms.
Then it's OK! Cpu Load is about 30-35%. I am using 4096 for FM. So i guess it's good that i can make it stable at 17/18ms.(0.018sec)
I Checked 512 mode. True! I can set lower buffer size, at 0.004sec (4ms), safe 5ms.
btw, Will be good if you can make buffer size to be in 3 decimals.


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 Post subject: Re: Stereo Tool 6.00
PostPosted: Wed Jan 05, 2011 12:31 pm 
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Is that new in this new BETA (005)? If so, are you using a higher (> 48 kHz) input sampling rate? I don't fully trust my new upsampling filter yet...
Yes it's the new beta and It doesn't matter what I set the sample rate to (I usually use 44.1 kHz). I also tried Bojchas addiction v11 preset and got the same result.

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 Post subject: Re: Stereo Tool 6.00
PostPosted: Wed Jan 05, 2011 3:36 pm 

Joined: Sun Dec 12, 2010 2:26 pm
Posts: 885
Quote:
Quote:
Quote:
camclone:
1: No, afterwards!
2: It just tries a little less hard to avoid bass distortion. You can definitely use it for FM broadcasts (or anything else). Some people like the effect, others don't.
In regards to his question, and partially to do with mine that's unanswered about the clipping in multiband:

Does the "Maximum boost from" slider then take into account the setting in bandpass and shift the boost start point up to the highpass setting? In other words, I have a highpass at 35 Hz, which is supposedly the bottom-end of my Logitech Z-2200s. If I leave the Maximum boost slider at 0, is the max boost at 0 or 35?
First any audio below 35 Hz is removed (well, theoretically. The filter isn't that good. It does get very strict when you select 'non-phase linear' filtering.

Then it just processes the audio. At that point it doesn't matter whether there was audio below 35 Hz in the input signal or not: It's not there anymore so nothing is done with it.

So audio below 35 Hz 'does not exist' as far as the Bass Boost filter is concerned. And it doesn't matter why it doesn't exist (wasn't there in the first place, or was filtered out).

Hope this answers your question...
Not really.

My question is concerning the slider "Maximum Boost From" in Bass Boost. That value is a decaying value that seems to want 100 Hz of operational room. My issue is I'd rather not boost any higher than about 135 Hz due to some potential crossover bleed-through. I'm not sure if the subwoofer is crossed at 120 or 150. I had been trying to tinker around with figuring out the crossover point, but the people I live with have extreme fits when I'm playing only sounds that low at high volumes, so my testing is rather slow going. Again, only by ear here, no meters. :(

So, if I set a Maximum Boost from 0 to 60, your other end of that slider wants to go to 160. What I'm wondering about is if it was any way feasible to have the Bass Boost take any value of a highpass as its' starting point.

In other words, if I set a highpass value of 60, I'd like to leave the Maximum Boost From 0 to x slider set at 0 and have Bass Boost know that a highpass had already been set ahead of it and automagically use the highpass value as where it starts applying boost. This would make the boost decay (from 60 to 100) happen at a much more rapid pace, but it would mean that I wouldn't have to have boost in the area from 100-150Hz, or if I figure out the cross is indeed at 120 Hz, I wouldn't be feeding any boosted signal to the satellites and could continue bass boost until 120. They are 3 inch drivers and, in my opinion really can't reproduce 130-150 that well, so that's why I'm trying to avoid boosting that range.

Alternatively, being able to set the "Drop to no boost" at less than 100 Hz delta between those two values would work too. In your current software terms, I'd like to have:

Maximum Boost from 0 up to 60 Hz
Then drop to no boost at 100 Hz

Or 120Hz if I discover that works ok

Is there a reason why you stick to 100Hz ranges for Bass Boost? Percentage calculation is smoother, or is it an actual auditory difference if you don't provide that much room for the decay?


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 Post subject: Re: Stereo Tool 6.00
PostPosted: Wed Jan 05, 2011 4:07 pm 
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Ok, I understand what you mean now.

The reason for the 100 Hz space was to avoid artifacts due to too steep filtering (as in Multiband etc.). Since filtering artifacts have improved lately, it might be possible to reduce the space a bit now.

Except for that: I'm now sure if what you want to do is really possible. Bass Boost boosts the bass inside the specified frequency range by adding harmonics at higher frequencies. So if you have bass at 60 Hz, Bass Boost will add frequencies at (among others) 120, 180, 240, etc Hz. The 2nd filter then filters out the ones that are too high (and hence annoying).

So actually if your speakers don't function too wel at 120 Hz, then if there's a bass there it will be made louder, but you'll also get harminics at - among others - 240 Hz, which could be useful in this case.


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 Post subject: Re: Stereo Tool 6.00
PostPosted: Wed Jan 05, 2011 4:21 pm 

Joined: Sun Dec 12, 2010 2:26 pm
Posts: 885
Quote:
Ok, I understand what you mean now.

The reason for the 100 Hz space was to avoid artifacts due to too steep filtering (as in Multiband etc.). Since filtering artifacts have improved lately, it might be possible to reduce the space a bit now.

Except for that: I'm now sure if what you want to do is really possible. Bass Boost boosts the bass inside the specified frequency range by adding harmonics at higher frequencies. So if you have bass at 60 Hz, Bass Boost will add frequencies at (among others) 120, 180, 240, etc Hz. The 2nd filter then filters out the ones that are too high (and hence annoying).

So actually if your speakers don't function too wel at 120 Hz, then if there's a bass there it will be made louder, but you'll also get harminics at - among others - 240 Hz, which could be useful in this case.
What I'm doing is -2.5 to -3.0 dB of reduction at 125 Hz in my X-Fi's equalizer to help cut some of the bleed. It's a fine-tuning thing for my particular setup, but I have to imagine someone else out there will have a similar situation where they're just trying to tweak the sound to their particular speakers, be that home use (me), or a DJ...

I also understand the harmonics issue. A long time ago, I was going to go into Nuclear Physics, but I haven't used advanced math for over 20 years, so a lot of this is rusty / way-over-my-head, but not that part.

Anyway, that's why I'm asking about this and about appropriate amounts of clipping in multiband. The space from 35-480ish is a concern for what I'm wanting to do. Right now I'm just leaving it set for 0 to 100, then harmonics from 100 to 200. What I'd like is to be able to bass boost from 35 to 100 or 120, with the maximum boost happening from 35 (not 0) to 60. I'd worry about harmonics after that. I may not be able to really hear them once all the clipping and loudness filtering happens though. Also, I'm a lowly home user, so the harmonics aren't going to cost me listeners or make people run out of a club with their ears bleeding from a harmonic resonance wave ;)


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 Post subject: Re: Stereo Tool 6.00
PostPosted: Wed Jan 05, 2011 10:14 pm 
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@ BETA005: LOL, I left a line in there that I used for testing... :oops: Oddly, the error only affects Hard Limit - which I had turned off during my final test.

Next version will contain a fix.


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 Post subject: Re: Stereo Tool 6.00
PostPosted: Wed Jan 05, 2011 10:56 pm 
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Joined: Tue Sep 28, 2010 10:56 pm
Posts: 326
Location: Usa
O M G... OMG... whats the problem Hans... i did nt notice so far, any sound problem.
What this line can cause???

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 Post subject: Re: Stereo Tool 6.00
PostPosted: Thu Jan 06, 2011 12:13 am 
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Building update now. Only a fix for the issues in BETA005, no further changes.


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 Post subject: Re: Stereo Tool 6.00
PostPosted: Thu Jan 06, 2011 1:43 am 
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BETA601-006: Fix for the issues in BETA601-005. No further changes.

Winamp DSP plugin: http://www.stereotool.com/download/dsp_ ... 01-006.exe
Stand alone version: http://www.stereotool.com/download/ster ... 01-006.exe
VST version: http://www.stereotool.com/download/vst_ ... 01-006.dll
VST version (No SSE2): http://www.stereotool.com/download/vst_ ... 01-006.dll
Command line version: http://www.stereotool.com/download/ster ... 01-006.exe
Linux command line version: http://www.stereotool.com/download/ster ... ETA601-006 [not available]
Linux GUI version: http://www.stereotool.com/download/ster ... ETA601-006 [not available]


Next step will be to do the same with the downsampling at the start of the processing, which will reduce the latency further for input sample rates above 60 kHz.


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