Audio Input Level too Low

The issues found when using the Input/Output Manager panel
maxbax0808
Posts: 1
Joined: Tue Aug 30, 2016 1:16 pm
Real Name: Max xx

First off: I really like this Software, it´s Simple and get´s the Job done very well!
But there is one major Problem I have. And that is that the Input Level of my "What you hear" Input from my Soundblaster Card is too quiet (only 0% to 1%) with maximum Windows Volume.
I tried everything I know of but it just won´t work.

QLC+ Version: 4.10.4
Operating System: Windows 10 64bit
Soundcard: Creative Soundblaster Z
toob
Posts: 1
Joined: Wed Sep 28, 2016 5:43 pm
Real Name: toobs

Same here. Windows mixer all up full on every channel and still on 3% max seen in QLC+ Please add a booster gain control within the program?
starwolf73
Posts: 5
Joined: Tue Jun 14, 2016 5:07 am
Real Name:

Hi,
Great software but same here, I would like to use the audio trigger but for some reason the audio is extreme low around 3-5%. If I put everything to 100% I get around 20% but that included lots of distortion (power speaker need to be low down for this test case).

As request a GAIN option will be very welcome.

I use a Presonus Studiolive 16.4.2 with the latest drivers. I even try on another laptop with a different build-in sound card and same thing. So I doubt it is related to the Presonus drivers.

Anyone had success using this widget? And on which OS?

OS: Windows 7 64bits
QLC+: 4.10.5b

Regards,
Eric
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GGGss
Posts: 2662
Joined: Mon Sep 12, 2016 7:15 pm
Location: Belgium
Real Name: Fredje Gallon

I think I might be on something...
@mcallegari
in https://github.com/mcallegari/qlcplus/b ... apture.cpp
on line 230 you are averaging the audio levels...
If we want to have a clue about the mood of the music - averaging is a way

BUT (!) we are triggering things based on peak levels. Ergo no averaging of the signal.
I think this explains why audio levels in the spectra-meters are so low - they are the average of levels seen.

Please change the line 198 from

Code: Select all

            pwrSum += m_audioBuffer[i];
to

Code: Select all

            if(m_audioBuffer[i] > pwrMax) pwrMax=m_audioBuffer[i];
and line 230 from

Code: Select all

 m_signalPower = pwrSum / m_captureSize;
to

Code: Select all

  m_signalPower = pwrMax; 
of course pwrMax should be declared so change line 184 to

Code: Select all

quint64 pwrMax = 0;
Actually we should do a RMS measurement but I don't have a clue how to.

I hope this will settle this issue...

Kind regards,

PS: reminder for myself: http://www.gaussianwaves.com/2015/11/in ... formation/
All electric machines work on smoke... when the smoke escapes... they don't work anymore
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mcallegari
Posts: 4462
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Location: Italy
Real Name: Massimo Callegari
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You got it wrong.
I don't change lines of code on demand. You change the lines and you test, then report if you obtained any improvement.
The average of the SUM of the FFT results provides the signal magnitude, so the global audio volume. I din't invent it. It's a well known formula you can find online.

This audio volume thing has been reported a million of times and still I don't see it.
Most likely it depends on the audio card, cause my stupid microphone on Linux (gain @70%) shows this
Screenshot_20170106_172707.png
Screenshot_20170106_172707.png (3.02 KiB) Viewed 6776 times
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GGGss
Posts: 2662
Joined: Mon Sep 12, 2016 7:15 pm
Location: Belgium
Real Name: Fredje Gallon

I will try to compile my own version as suggested... This will be the first time I do something like that - so bare me a minute or two (read: many hours)
Or if someone else is more experienced compiling please do help me and report back.

Sorry to say Mr. Callegari - whatever I try I'm stuck with these results (see image in annex)
From the same source (line-in != microphone) I recorded a small sample.
If I use audacity to analyse the results (see image2), you'd expect to get peaks in the -3dB range (almost clipping so to say)
fft.png
QLC+ is not reflecting that. Audacity & qlc+ are using the same audio source at the time of recording...
audio-levels.png
audio-levels.png (11.4 KiB) Viewed 6769 times
In answer to your point where you reveal that the code is general knowledge - I have knowledge about Hanning, Blackman et al. The FFT needs to be averaged to reflect the amplitude of the sample in measuring. Yes you are right (see my PS: above - this was my source)
And again: since we are triggering on peaks in levels - we cannot use average values in amplitude - we need max. values.

I'll try it out in the next days and report back.

I don't want to be rude or anything (as a newby on this forum). I simply try to add my experience and knowledge to your project. Rest assure that I'm admiring your work with great enthousiasm ;-) As a former Chamsys operator finding and using QLC+ has been the best thing since ... well 20+ years being a light-engineer.

Thanks for your feedback.
Enjoy your weekend,
Fred
All electric machines work on smoke... when the smoke escapes... they don't work anymore
msikorsk
Posts: 7
Joined: Tue Sep 20, 2016 8:30 pm
Real Name: Marcin Sikorski

Hi,
same issue here. OSX with audio Loopback. Very low signal level.
Wouldn't be a simple fix to add signal gain control? As you said, different sound cards, different signals. Shouldn't be a big deal to amplify it before midget master control (for example at audio input setup)


Have a good day,


Marcin Sikorski
CalebGrayson
Posts: 12
Joined: Tue Apr 14, 2015 4:27 pm
Real Name: Caleb Grayson

the audio for me is also very low — always has been from versions maybe 3 to current 4.11.0.
it own't take internal sound, either, as i set input to be my Kontrol S8 audio card out and nothing happens. i have to rout our the booth out of the Kontrol S8 mixer into the line in on my MacBookPro and only with the booth out maxed into a maxed MBP line in preamp do i get really good level readings.

also, i can't figure out a way to set a MIDI controller to change the masterfader on the Audio widget.
wersonnst
Posts: 1
Joined: Mon Jul 24, 2017 3:19 am
Real Name:

Had the same issue with low level.

I tried the code from GGGss which cause in an variable mismatch error
so I done a few simple changes.

The best results I have with

Code: Select all

196           pwrSum += -1 * m_audioBuffer[i] * 15;
197     else
198           pwrSum += m_audioBuffer[i] * 15;
in https://github.com/mcallegari/qlcplus/b ... apture.cpp

With that the input level controls the Audio Trigger from very gentle to mostly all flicker. Depends on the show were to use what.
The implementation of an Extra Gain or boost level would be nice.

@ mcallegari
I don,t find the word's how to thank you for this piece of nice software and all your outstanding work , so we can work with QLC+ and enjoy the world's people with the best light show's and make them smile. :)
Thank YOU.
yokosuna
Posts: 204
Joined: Tue Mar 22, 2016 9:07 am
Real Name:

Is this fixed in 4.11.2 ? Cause I´m too having this problem ever since ... (Windows 7/Asus Xonar)
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sandinak
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Real Name: Branson Matheson
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+1 on a boost or trim adjustment..
+1 on level marks for settings .. would help alot.

with the standard audio output on the Mac.. I am also in the 5-15% at full blast. I havn't tried on my pi's yet .. but that's coming.
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andres robles
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Joined: Tue May 17, 2016 7:41 am
Location: Spain
Real Name: Andres Robles

Hello, I can think of some solutions, I do not know if it's valid in windows or mac, in Linux it probably works,
The solution would be to interpose a plugin amplifier between microphone and qlc ..
example of use in linux
I open jack, I open audacity, I open QLC +, and I open some plugin amplifier, (calf plugin ... or another)
in jack connect, from micro to audacity, Ready
from micro to plugin amplifier and from plugin amplifier to pulseaudio
I say in configuration of I put audio that qlc take the sound of qjack
Adjust desired volume for QLC +
Ready
I do not know if in Windows it works qjack or there are tools for the routing of audio connections so that you can interpose plugin between different audio input software to be able to adjust independent volumes to each software from the same micro, I do not know if I explain and I hope that will help you find some effective solution
a greeting
BillDoE
Posts: 31
Joined: Fri Apr 05, 2019 6:23 am
Real Name: Bill K

I'm having the same issue in Win 10 QLC 1.12.2 GIT. All my levels are cranked but QLC Audio input level hits about 25% tops.
Using Foobar2000 cranked through VB Audio Virtual Cable. I even tried making an amplifier with Purr Data.
My Audio gets so loud Its distorted but qlcplus audio level moves up like 2%.
Cranked-qlc-Audiio2019-07-11-2.jpg
RPi4b 4GB
BillDoE
Posts: 31
Joined: Fri Apr 05, 2019 6:23 am
Real Name: Bill K

I've tried this now with.
Raspberry Pi4 or Windows 10 x64
qlc 4.12.2 IMG
& 2 different external USB sound cards.
Focusrite Scarlett 2i2.
or E-MU.
Recording levels in Audacity are peaking at 100% on the same Pi machine using the same settup.

I get 100% Audio levels on the sound cards from the mixing board and the levels top out at 55% in QLC+.
Sure would like to solve this!
RPi4b 4GB
BillDoE
Posts: 31
Joined: Fri Apr 05, 2019 6:23 am
Real Name: Bill K

Using a Focusrite Scarlett 2i2 (Gen 1) usb card and XLR inputs on raspberry pi4.
I have no idea what the actual problem is, but this seems to solve my issue for now.

On my Raspberry pi4. I was stuck at 50% Peak in QLC+
I changed pwerSum to (pwrSum * 2) on line 229 in audiocapture.cpp and recompiled.

Code: Select all

229   m_signalPower = (pwrSum * 2) / m_captureSize;
I'm getting good peaks with better dynaminc range.

I do not claim to know what I'm doing. Use at your own risk.
RPi4b 4GB
richard
Posts: 1
Joined: Thu Feb 13, 2020 12:32 am
Real Name: Richard

I bought a Movo MC1000 Conference USB Microphone that has a 20 foot range pickup for $50 US from Amazon that solved my input level problems on my Windows 10 PC. I too was having the same issues as everyone else. I did turn up the sensitivity in Windows on the mic to 100% and it's working great with QLC to the point I need to turn down the sensitivity control on the QLC sound widget. It works well for conferences when I need it as well!
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edogawa
Posts: 567
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Real Name: Edgar Aichinger

I tried the power_sum*2 thing and it improved things, but then again I also thought we're looking for peaks, not average power in the main volume and bands.

As this is such a long-standing issue and I was hit by it myself a year ago when I needed it, I thought I'd go over to freenode IRC channel #lad (linux audio developers) and ask for an extra eye (having no notable fft or c++ knowledge myself).

I pointed them to this thread and damo22 (Damien Zammit, developer of the Zam audio plugin suite) answered this:
<damo22> edo_lap: i had a quick look, i *think* he is calculating the power spectrum wrong
<damo22> i think he needs to take the average of the transformed values, not the raw signal
<damo22> edo_lap: i think you need to modify the AudioCapture::fillBandsData() function to compute the average as well as the maximum and return both
<damo22> and dont use pwrSum because its wrong
I doubt though to be able to implement the change he proposes... so I hope this makes sense to Massimo or whoever can code well enough, and helps to resolve the issue.
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GGGss
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Real Name: Fredje Gallon

3 years or so ago - I came to the same conclusion.
The math behind the audio levels is off (not wrong).
Massimo pointed me to 2, 3 related articles defending his programming decisions.
I thanked him pointing these to me and went for another solution.
All electric machines work on smoke... when the smoke escapes... they don't work anymore
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edogawa
Posts: 567
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Real Name: Edgar Aichinger

Damien seems to cook up something for us:
https://github.com/mcallegari/qlcplus/c ... io:fix-fft
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mcallegari
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Real Name: Massimo Callegari
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QLC+ can surely use the help of an audio expert.
However I think I've seen an error in the change he made. He's ignoring the number of bars the user wants.
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