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PostPosted: Fri Jun 19, 2009 12:26 am 
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Now when is version 4 is finished. i have some questions about this subject.

Couple times i try to calibrate stereo separation. I almost never can do anything from calibrating phase differences.
But .. very big effect i have when i start set 38kHz slider (from step 1).

So if i put ST FM output directly to mpxtool i need to set that slider to 1.03-1.04 .. for best separation.
For Transsmiter and tuner .. i have to put that slider to 1.08 - 1.24 (depend what tuner). but i have peeks over 110% in that case.
So question .. what else can be done here .. because even 38kHz slider at 1.04 makes peeks over 100%.

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PostPosted: Fri Jun 19, 2009 1:32 am 
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Quote:
Now when is version 4 is finished. i have some questions about this subject.

Couple times i try to calibrate stereo separation. I almost never can do anything from calibrating phase differences.
But .. very big effect i have when i start set 38kHz slider (from step 1).

So if i put ST FM output directly to mpxtool i need to set that slider to 1.03-1.04 .. for best separation.
For Transsmiter and tuner .. i have to put that slider to 1.08 - 1.24 (depend what tuner). but i have peeks over 110% in that case.
So question .. what else can be done here .. because even 38kHz slider at 1.04 makes peeks over 100%.
It's very odd that you have to put it at 1.03-1.04 for best separation in MPXtool..... You are using a VAC connection? If that's the case there might be a bug somewhere......

About the tuner: If it differs per tuner, it's not reliable at all... For the (good quality) tuners that I have I have to use exactly the same settings.

About the peaks: How do you see those peaks? If you measure the signal on the PC before it leaves the sound card and transmitter, it makes sense that you are seeing peaks. However, if calibrated properly, the signal that leaves the transmitter should have NO peaks at all. If it does, the calibration is off.

Note: I am assuming that you are using "FM Volume Overdrive" set to at least 105%, otherwise occasional peaks up to about 5% do occur. That's exactly what "FM Volume Overdrive" is for: To constantly use the extra space instead of having occasional peaks.


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PostPosted: Fri Jun 19, 2009 2:04 am 
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actualy i must put it to 1.03 because i must lower 10kHz in step2 .. for mpx tool.
i'll reset and try all again tonoght.

mistake not 10kHz ... 19kHz (step2)

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Last edited by Bojcha on Fri Jun 19, 2009 11:38 am, edited 1 time in total.

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PostPosted: Fri Jun 19, 2009 9:30 am 
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This is really odd. The calibration is there to correct sound card/cable/transmitter problems; if none of those are present (as is the case with a direct link to MXPtool), you should not need it, and anything you do SHOULD make the stereo separation worse.

If you use another application to generate an FM signal, is the stereo separation then better than with Stereo Tool with no calibration? If so, there might be something wrong with Stereo Tool, otherwise it might be MPXtool...


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PostPosted: Fri Jun 19, 2009 11:42 am 
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First i'll must do some things.

- wait to come another Juli@ for transmitter
- change cable from sound card to transmitter (but it's alredy half meter long)
- change cable from my tuner to soundcard (that cable is bad)

then i'll see this again.

btw Hans what tuner do you have?

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PostPosted: Sun Jun 28, 2009 4:35 pm 
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I replaced cable from my tuner to soundcard .. here was part of problem.

but now when i have Juli@ at transmitter i set asio buffer for output curently to 0.06. What is safe buffer to set?
yesterday i set it to 0.05 but today droping started.

And interesting is that always drops when buffer is between ~0.10 and 0.20.

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PostPosted: Sun Jun 28, 2009 5:43 pm 
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I replaced cable from my tuner to soundcard .. here was part of problem.

but now when i have Juli@ at transmitter i set asio buffer for output curently to 0.06. What is safe buffer to set?
yesterday i set it to 0.05 but today droping started.

And interesting is that always drops when buffer is between ~0.10 and 0.20.
Usually, bigger is better. But it's odd that you're getting MORE hiccups at higher buffer sizes.

You might need to configure the Juli@ latency properly. Go to the Juli@ config app, click on Config -> Latency, and there choose something big enough.
(For example, the minimum is 48 samples... Now at 192 kHz that's REALLY short. Basically, ASIO calls Stereo Tool for each block of samples to provide those samples. So at the minimum setting that would mean that Stereo Tool is called 4000 times per second - which your pc probably won't be able to handle...

By the way, I usually set the latency to 0.020 or so and that has always worked fine (even on my old single core PC). So dropouts at 0.050 is very unexpected. What does the buffer filling display in the Configure Sound Cards window show?

In ASIO mode, you're using the same sound card for input and output, so - except for temporal performance problems on your PC due to other programs or Windows services that use a lot of CPU - there's no reason why you should get dropouts after running for a while.


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PostPosted: Sun Jun 28, 2009 9:19 pm 
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By the way, I usually set the latency to 0.020 or so and that has always worked fine (even on my old single core PC). So dropouts at 0.050 is very unexpected. What does the buffer filling display in the Configure Sound Cards window show?
In ASIO mode, you're using the same sound card for input and output, so - except for temporal performance problems on your PC due to other programs or Windows services that use a lot of CPU - there's no reason why you should get dropouts after running for a while.
That is problem (i think) i must use VAC for input.
Juli@ latency is at 256 samples and that is default.
Today i set it to 0.06 and this dropouts starts much erlier. i'll set it now to 0.02 (that is minimum).
buffer filling is at first 90% green and after some time it's gone ad this dropouts starts.
if this don't work shuld i try to set Juli@ latency to 128 or 512 ?

i'll be happy if i can use Juli@ for input and output but i don't know how :S

So what i need to make.
Player on transmitter PC from what sound from studio comes to use Juli@ but same that sound to be for ST input. and Asio output to transmitter.

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PostPosted: Sun Jun 28, 2009 11:04 pm 
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Quote:
Quote:
By the way, I usually set the latency to 0.020 or so and that has always worked fine (even on my old single core PC). So dropouts at 0.050 is very unexpected. What does the buffer filling display in the Configure Sound Cards window show?
In ASIO mode, you're using the same sound card for input and output, so - except for temporal performance problems on your PC due to other programs or Windows services that use a lot of CPU - there's no reason why you should get dropouts after running for a while.
That is problem (i think) i must use VAC for input.
Juli@ latency is at 256 samples and that is default.
Today i set it to 0.06 and this dropouts starts much erlier. i'll set it now to 0.02 (that is minimum).
buffer filling is at first 90% green and after some time it's gone ad this dropouts starts.
if this don't work shuld i try to set Juli@ latency to 128 or 512 ?

i'll be happy if i can use Juli@ for input and output but i don't know how :S

So what i need to make.
Player on transmitter PC from what sound from studio comes to use Juli@ but same that sound to be for ST input. and Asio output to transmitter.
In that case VAC is indeed the problem. But you're lucky: You don't need VAC if you have a Juli@!

The Juli@ driver contains its own virtual cable implementation. Read the manual, and look for DirectWire. That's basically the same as VAC...


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PostPosted: Sun Jun 28, 2009 11:14 pm 
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That is what i need.
i wait now to start NightProgram ... to start wireing :)

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