How does it "know" the volume unless you are using it as a volume control?
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Device for Equal Loudness/Fletcher Munson Curve? Do Any Speakers Adapt to This?
- Thread starter stevenswall
- Start date
It takes in the pre-amp out signal or a variable line signal.
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i‘ve had a DEQ2496 for many years too. Got is soon after my first son was born and he is 20. DEQ 2496 is a wonderful toy. Have mostly used it for headphone crossfeed & EQ, and AD & DA conversion. It’s right here at my bedside. Also have used it to play around with concepts of compression, limiting, rotation of audio imaging, playing around with effects of changing certain frequency ranges, etc. Did not know you could do something approximating dynamic loudness control with it. Thanks to all who pointed it out and fleshed out concept here!
Gotcha.
OP
- Thread Starter
- #25
Does that mean the plugin would work on a MiniDSP with Volumio like some of them have?
Edit: Looks like it can!
My ADI-2 has dynamic loudness dependent on the volume setting. However you still have to adjust the impact because the unit does not know the sensitivity of your speakers or the size of the room. With inefficient speakers in a large room it obviously needs more boost. But once you have adjusted for that the rest is automatic.
It may work but as Volumio on MiniDSP devices is a custom Volumio installation you probably won't get any support from MniDSP nor from Volumio and you may as well loose warranty. If you choose to do it I strongly suggest you first make a backup of your original Volumio MiniDSP installation so you may restore it back if things go wrong way.
Yamaha integrated amp: https://au.yamaha.com/en/products/audio_visual/hifi/a-s301/index.html
https://au.yamaha.com/en/products/audio_visual/hifi/a-s301/features.html#product-tabs
* dynamic loudness control
* Inbuilt DAC
* Phono stage
https://au.yamaha.com/en/products/audio_visual/hifi/a-s301/features.html#product-tabs
* dynamic loudness control
* Inbuilt DAC
* Phono stage
Hi,
I did install your Plugin and am exploring the interface. "DIRAC PULSE" is selected, eh, how would I definde those filters? I have a NanoAVR Dirac, is there a way to use those…? Any Integration with REW?. Thanks for the great Plugin and the instructions how to install it
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NanoAVR Dirac is a proprietary miniDSP solution with Dirac integration. You can use it a standalone room EQ solution if you create room EQ filters with Dirac.
Volumio DRC (ex BruteFIR) plugin can use EQ filters created with REW, procedure is described here.
P.S. I am not developer of this plugin, I helped in it's feature creation and testing, but I appreciate your thanks.
digitalfrost
Major Contributor
Just found out that EqualizerAPO can do Loudness Correction, however I have no idea how to use it. There's no documentation and the calibration doesn't work for me at all.
e: Playing around with it, I can't hear any difference, no matter what I set the reference volume to. Tried it with pink noise, it doesn't do anything at all.
e: Playing around with it, I can't hear any difference, no matter what I set the reference volume to. Tried it with pink noise, it doesn't do anything at all.
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Don Hills
Addicted to Fun and Learning
Also see this earlier discussion:
https://www.audiosciencereview.com/forum/index.php?threads/equal-loudness-contour-loudness-dsp-for-playback.5955/#post-133594
https://www.audiosciencereview.com/forum/index.php?threads/equal-loudness-contour-loudness-dsp-for-playback.5955/#post-133594
Not a physical device but Equalizer APO has this functionality!
digitalfrost
Major Contributor
Can you tell me how to use it.
Can I install this on the minidsp shd which uses volumio?
MRC01
Major Contributor
Natural acoustic instruments have a different timbre at different volume levels, due not only to the non-linear response of our hearing being discussed here, but also being played louder or software changes the instrument's frequency structure or timbre. It seems that a loudness curve would have to know:
1. The absolute level at which the music is being played (what SPL would a full scale amplitude be)
in addition to knowing
2. The relative level at which the music is at every moment in time (something like a 3 second running average RMS amplitude).
The loudness curve should apply only to the first, not to the second, else it would squash or mask the timbre changes with musical dynamics that is an essential and natural part of the music.
Regarding (1), you would have to graph your system's SPL at listener position versus volume control knob position (e.g. preamp voltage), in order to tune the loudness contour to the SPL. This of course depends on the room size, speaker efficiency, and power amp gain.
PS: on 2nd read what I said sounds confusing. So here's the point: any loudness curve should be based on the actual SPL level of the loudest musical peaks (or average levels). If the music is dynamic and reaches a quiet spot, or a loud spot, the loudness curve should not compensate for that. This is natural and the change in timbre or FR is part of the musical event. Once calibrated the curve should depend on volume position only.
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flaviowolff
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How does this analog variable loudness from Yamaha work?
tuga
Major Contributor
Is this of any interest?
Adaptive loudness compensation in audio reproduction
Leonardo Fierro
This work involves the study of the psychoacoustic phenomenon of nonlinear and frequency dependent loudness perception, its modeling, and the use of digital filters to introduce an adaptive compensation based on the reproduction level.
Music and sound are mixed and mastered at a particular loudness level, which is usually louder than the level they are commonly played at. This implies a change in the perceived spectral balance of the audio source, which is largest in the bass and sub-bass ranges. As the volume setting in music reproduction is decreased, a loudness compensation filter can be designed to introduce an appropriate boost, so that the low frequencies are still heard well and the perceived spectral balance is preserved.
This thesis describes a loudness compensation function derived from the standard equal-loudness level contours and its implementation via a digital first-order shelving filter, and it documents a formal listening test, designed and conducted to validate the accuracy of such a method.
Adaptive loudness compensation in audio reproduction
Leonardo Fierro
This work involves the study of the psychoacoustic phenomenon of nonlinear and frequency dependent loudness perception, its modeling, and the use of digital filters to introduce an adaptive compensation based on the reproduction level.
Music and sound are mixed and mastered at a particular loudness level, which is usually louder than the level they are commonly played at. This implies a change in the perceived spectral balance of the audio source, which is largest in the bass and sub-bass ranges. As the volume setting in music reproduction is decreased, a loudness compensation filter can be designed to introduce an appropriate boost, so that the low frequencies are still heard well and the perceived spectral balance is preserved.
This thesis describes a loudness compensation function derived from the standard equal-loudness level contours and its implementation via a digital first-order shelving filter, and it documents a formal listening test, designed and conducted to validate the accuracy of such a method.
MRC01
Major Contributor
I've always suspect that one reason most rock/pop recordings sound so bright (whether midrangy or tinny) is because the engineers mastering it do so at a zillion decibels and I'm playing it back at lower levels.
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