Would these passive components help with the back EMF, as seen by the amplifier?
That has been my motivation to include it, even though I have an active setup.
Most comments I got were: "any modern amp should be able to handle an impedance like that" etc. But my motivation has been, what if it helps make the amp's job a little easier, or the influence of back EMF a little smaller.
I kept the components in place all these years, noticing the changes between amplifiers have made me think there should be something to this theory.
That has been my motivation to include it, even though I have an active setup.
Most comments I got were: "any modern amp should be able to handle an impedance like that" etc. But my motivation has been, what if it helps make the amp's job a little easier, or the influence of back EMF a little smaller.
I kept the components in place all these years, noticing the changes between amplifiers have made me think there should be something to this theory.
Its for sure that impedance correction doesn't hurt. No doubt it helps some amps more than others, through some second order effect perhaps other than the output impedance of the amp.
But why doesn't the Z matter more with only active filters?
A driver apparently can be modeled as a voltage driven device. Sound level output is proportional to voltage input which makes it independent of Z; not power input which would make it a depend on Z. Does anyone know the physical reason why this is so?
But why doesn't the Z matter more with only active filters?
A driver apparently can be modeled as a voltage driven device. Sound level output is proportional to voltage input which makes it independent of Z; not power input which would make it a depend on Z. Does anyone know the physical reason why this is so?
I'd look at the speaker and amp as a system. What one amp can handle easily another cannot. Hard to point a finger at whos at "fault". The speaker or the amp?
I don't think the amplifier I use right now would have a problem driving the arrays without impedance correction.
Circuits are modeled after the Goldmund Telos 400.
6 pairs of Mosfets per channel making 350 watt into 8 ohm/channel.
My preference was determined when I was still using my trusty (but old) Pioneer A757 MKII which was about 100 watt into 8 ohm.
I've had that one since the early 90's when I bought it new. I still use a Pioneer A-447 for my ambient channels.
The bigger question is: is it beneficial to use impedance compensation? The arrays don't have an unusual or straining curve.
It would not be considered a difficult load.
Circuits are modeled after the Goldmund Telos 400.
6 pairs of Mosfets per channel making 350 watt into 8 ohm/channel.
My preference was determined when I was still using my trusty (but old) Pioneer A757 MKII which was about 100 watt into 8 ohm.
I've had that one since the early 90's when I bought it new. I still use a Pioneer A-447 for my ambient channels.
The bigger question is: is it beneficial to use impedance compensation? The arrays don't have an unusual or straining curve.
It would not be considered a difficult load.
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While being active on other array threads where the emphasis keeps focussing on whether or not this line array principle can work I'm still enjoying songs with the revisited mid/side EQ.
Still need to work out the bass management as the next step. Not as critical with the more potent amplifier but definitely worth it.
Still need to work out the bass management as the next step. Not as critical with the more potent amplifier but definitely worth it.
LOL!! That is hilirious..
Some ought to listen to arrays or at least pay attention to those who have heard them .
Concerning the impedance correction, I don´t think it is so much a matter of driving power, but more how amplifiers with a feedback loop handles back EMF. I have experienced that it is audible , and for the better, when you use an impedance correction network with a lot of very different amplifiers.
I'm still surprised with the presentation of the arrays. Boggles the mind all that high frequency stuff can sound coherent.
A little OT maybe:
Just as wesayso I have a pre-eq before the convolver in Jriver, as I use the a pre-eq before I let Audiolense make the FIR filter to room correction/final EQ.
Now my question is: Can the pre-eq be combined with the convolution file to make one convolution file ?
The reason I ask is , I want to try HQplayer because of its better handling of DSD direct and HQplayer has no parametric equalyzer , only a convolver..
Just as wesayso I have a pre-eq before the convolver in Jriver, as I use the a pre-eq before I let Audiolense make the FIR filter to room correction/final EQ.
Now my question is: Can the pre-eq be combined with the convolution file to make one convolution file ?
The reason I ask is , I want to try HQplayer because of its better handling of DSD direct and HQplayer has no parametric equalyzer , only a convolver..
Yes they can. You can use REW to combine two impulse response using the maths functions on the All SPL tab. Select the files you want to combine together in the A and B boxes then choose the type of function A*B is what I used from memory as multiplication is the same as convolution when used in this way. I think the level is increased from the multiplication which can be offset with a 120dB reduction in level or whatever you need to rebalance. It's easy to test using two simple EQ files to see if the result is as expected.
Create an impulse response of your parametric EQ to make sure it is the same as there are many different ways to describe the Q that are not compatible with each other.
Sorry for my ignorance, but how exactly do I create an impulse response from the parametric eq settings?
Another way would be to set the output of JRiver to "Disk Writer".
Next feed it with a Stereo Dirac pulse and your complete chain of convolver and EQ will be recorded to a wave file.
The dirac should have enough silence before and after the pulse due to the delay of the convolution. The wave file can than be cut to length or you can import it into REW.
I often use this last one to "see" my total correction.
If you have no IIR, wouldn't you miss the mid/side EQ functionality?
Next feed it with a Stereo Dirac pulse and your complete chain of convolver and EQ will be recorded to a wave file.
The dirac should have enough silence before and after the pulse due to the delay of the convolution. The wave file can than be cut to length or you can import it into REW.
I often use this last one to "see" my total correction.
If you have no IIR, wouldn't you miss the mid/side EQ functionality?
One can create it with RePhase, but as Fluid said: interpretations of "Q" often are a little different between programs. The reasons of the many presets in REW in the Equalizer tab.
Come to think of it, one could also use REW to create the impulse on the EQ tab.
Arrgghh bummer. You are right! It would be far easier if I could place Jriver in front of HQ player in the playback chain.
Convolve the Mid side EQ into some test tracks and use those to evaluate HQPlayer. If you want to use your own reconstruction filter you can upsample PCM and roll your own filter. Unless your DAC has complete delta sigma modulator bypass I can't see how it (HQPlayer) can possibly be worth the hassle myself.
As wesayso said before the easiest way to get your EQ as an impulse is to send a dirac pulse through it and record the output, Jriver does it and Audacity can do it too.
As wesayso said before the easiest way to get your EQ as an impulse is to send a dirac pulse through it and record the output, Jriver does it and Audacity can do it too.
But it is exactly the point. I am experimenting with a no-dac where the DSD stream is just lowpass filtered and the magnitude of different filters and modulators in HQplayer is much better than in Jriver. I have been using Jriver up until now to convert all playback to DSD on the fly and the sound coming out of this , except for some noise, is very analog like and like nothing I have heard before from a digital source, so I want to optimize and reduce the noise.
Do the Parametric equalizer and mid/side eq work when Jriver is dealing with DSD? Dosen't Jriver have to convert DSD to PCM in order to make any DSP changes? I know the volumen control does not work in DSD..
Maybe I could use something like this:
LoopBeAudio - A Virtual Audio Loopback Device
Then I could use jriver first and then loop back into HQplayer?
True, I forgot about that...
You could try if it works... an hour of play with the free evaluation version. (and after every reboot)