What's another example of a crossover using only an analog/causal high pass and low pass section that sums to a transient/phase perfect response ?
Or are you referring to a 3rd driver filler approach or DSP approaches using non causal filters ?
Interesting, I didn't realise that. I don't know enough about that type of filter to disagree with you so I'll take your word on it.
The audibility of a lot of things such as transient perfect square wave reproduction seems to be taken as an article of faith by many but when you do careful testing (or read the research) it's sometimes quite shocking at how "deaf" our ears are to certain classes of defect...
As I was saying in another thread recently, above a certain frequency we have no ability to detect phase at all, (somewhere above about 1-1.5Khz) except by any amplitude summing changes it introduces or perhaps arrival time differences on impulsive sounds if the group delay is really bad.
So if your crossover is well above this frequency and the amplitude response is flat, you're not going to hear the phase shift of the crossover per se, all you might hear is the group delay if it's really bad, or perhaps the filter ringing if it's really bad. A well designed 4th order or lower filter shouldn't have problems with either of these...
Accurate square wave reproduction is one of those things I wouldn't even bother trying to measure or achieve, not when so many sacrifices must be made to come even close to achieving it. Ruining a lot of very audible parameters to optimise one virtually inaudible one is never a good idea IMHO.
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Modeling speakers after instruments is a very, very limiting approach in sound reproduction. What we want to reproduce is a huge variety of aural spaces from bird chirps in a snow covered field to water drops in a huge cave. You have to think "sound field" not "instrument". The latter is only relevant on the production side, not on the consumer side.
There is a hole class of symmetric, 2-way, 2nd order TP crossovers that consists of overlapping HP and LP filters and applying EQ. The 2nd order filter characteristics can be used acoustic targets and the eq is best applied using analog active eq. I wrote an Audio-Xpress article on this back in 2002.
Transient Perfect Designer Two Way
Then there are all the subtractive crossovers that are TP which are defined by HP= 1 - LP. These are generally asymmetric at for any order LP filter the HP section ultimately has a 1st order roll off. The inverse, LP = 1- HP is similar but the LP is ultimately 1st order.
These are all casual and can be implemented using analog or digital approaches though some may not lend themselves to passive realization.
pre-ringing?
isnt pre-ringing due to the type of feedback used in the filter cct? Effectively speeding up the rise time of the attack envelope, u essentially induce resonance? Ive studied this before...to be honest it put me off the whole topology, doesnt it erode the advantage of such filters somewhat?
isnt pre-ringing due to the type of feedback used in the filter cct? Effectively speeding up the rise time of the attack envelope, u essentially induce resonance? Ive studied this before...to be honest it put me off the whole topology, doesnt it erode the advantage of such filters somewhat?
I propose that the microphone captures incomplete information. If the waveform captured is reproduced, we are not guaranteed that it will have the same propagation pattern or behaviour. That is why we need to give it gentle cues about its true form.
The bird in a snow covered field can also be modeled .
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Try binaural recordings with mics inside your ear. The information a microphone can capture IS complete. The only thing inclomplete is the wave field created by the current reproduction chain.
There are a couple of "black holes" where relevant information gets lost and/or distorted. So current channel-based production techniques don't try to recreate a wave field like binaural recordings do because two-speaker stereo is inherently inadequate for such an approach. Even 5.1 or 7.1 multichannel is inadequate.
Do it! Maybe you will succeed where everybody else failed for the last 80 years. Good luck.
If you know what "gentle cues" are needed then do it. Virtually everybody interested in high quality audio reproduction would buy your speaker.
But then again, is it desirable to listen to a symphony orchestra with the acoustic signature of your living room, even if it would sound real? I suppose Beranek would disagree.
But then again, is it desirable to listen to a symphony orchestra with the acoustic signature of your living room, even if it would sound real? I suppose Beranek would disagree.
Should it be capturing the direct sound at all angles? The question may be - would stereo be able to do anything with that? Can you make two speakers project an image and reconstruct the instruments as well? (most instruments are not imaged near the speakers anyway, though that's not even the point). You'd stand a better chance processing the signal beforehand because stereo is stereo, and playing games with your speakers' may just jeopardise the image.
Would you listen to a live performance in an anechoic chamber and not expect everything to sound like what it is, just more dry? The wider sound isn't necessary for us to believe it is what it is. So if we made our listening room anechoic as well and recorded this chamber, the transfer should sound perfect, no?
If this is the limit to stereo, wouldn't you rather embrace it and do it well? What can you do with a manufactured, mixed two channel signal, not much? You could add your own spice.. dipoles maybe? This kind of effect can sound good (I prefer less of it myself) maybe even in dealing with the recordings...recordings (aside from the above) are what we are really working against.
For instance (and this is not unlike many songs I like to listen to), you have a recording where the vocalist has been brought forward by boosting harmonics and reverb has been added to smooth the vocal performance. One instrument has been rolled off to make it sound mellow. One has been pushed back and had reverb added for that far-away sound, and to make the instrument noticeable without being prominent. Some instruments are done in an isolation booth, and each has its own mic with it's own character.
As a result, some parts sound a little harsh as opposed to some sounding mellow which doesn't help you to see them as being in the same space and you can't do a great deal to fix it yourself. The isolation booth parts may sound as if they were recorded separately and added later, dry, not fitting in to the space. A drum kit may have been imaged so that you hear it as being twice as wide as an actual kit, with the drums in the wrong places as if someone just put them where they wanted to.
I think that if you concentrate on reproducing stereo you can focus on the job knowing you're moving forward, seeing past these limitations...until you are relaxing in your chair, cold beverage in one hand but sometimes the equaliser remote in the other hand. Oh well, at least you have a cold beverage in one hand 🙂
In a causal filter such as an analog passive one the output at any moment is dependent on the current and recent inputs, so any ringing in response to an impulse will occur after the impulse, as it can't see into the future.
In a non causal filter the current output is dependant not only on the current and recent input but near future input as well. Something that's nearly impossible to do in the analog domain (unless you had some sort of fancy delay line I suppose) but very easy to do digitally, because you simply delay the output a few samples behind the input then the current input signal becomes your "future" input. With the small penalty of a time delay between input and output your filter can now see into the future and produce an output that is dependant on future input as well as past and present.
So when an impulse arrives at the output the ringing has already preceded it. As to the audibility of pre-ringing, not sure 😛
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Not to be confused with those casual filters. You know, "Maybe I'll respond to the input a little later, or maybe not".
David
David
Of course it SHOULD. But they don't because no one has ever mad anything close to a perfect speaker, perfect mic, perfect anything. It is our quest. We have a very long way to go. We don't even have a perfect idiot, but I have met a few quite close.
i believe so, time domain manipulation of the signal blah blah, i think we move from damped into oscilliatory TF. This belief is what i dont like about these filters. Pre ringing is just as audible as post ringing IF the filter is accurate in its quantisation. Then the question remains, how audible is the pre or post ringing? If it isnt easily audible then i see the advantage of 'messing with time', if it is audible then in my eyes, and maybe only mine, then temporal distortion is just another error component, compounding with the error we would otherwise already have.
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Pre-ringing is vastly more audible than post. That's because there is no sufficient pre-masking in our hearing, but generous post-masking. So anything that introduces pre-ringing to the signal is not desirable.
About them '(non)perfect' mics and soundfields - consider that there are 2 types of environments we wish to reproduce.
Natural environments - bird chirping on a snow field and the cave scenario (thanks Markus) AND any kind of music performances (symphony etc.). If you wish to be aurally transported to the venue then there is already the means. Binaural is easiest, but the lack of head-tracking could destroy the illusion. And WFS, but this is totally different capture/reproduction technique. Analogous to vector graphics.
In regular stereo, we don't need to reconstruct the original sound-field of the venue, but only just a illusion that is constructed by the recording engineer in the studio (or not in the studio, ie. live-feed-from-mic, there is still an 'engineer' positioning the mics etc.). So stereo is an art-form in itself creating a specific illusion, not aspiring to 'reality'
About them '(non)perfect' mics and soundfields - consider that there are 2 types of environments we wish to reproduce.
Natural environments - bird chirping on a snow field and the cave scenario (thanks Markus) AND any kind of music performances (symphony etc.). If you wish to be aurally transported to the venue then there is already the means. Binaural is easiest, but the lack of head-tracking could destroy the illusion. And WFS, but this is totally different capture/reproduction technique. Analogous to vector graphics.
In regular stereo, we don't need to reconstruct the original sound-field of the venue, but only just a illusion that is constructed by the recording engineer in the studio (or not in the studio, ie. live-feed-from-mic, there is still an 'engineer' positioning the mics etc.). So stereo is an art-form in itself creating a specific illusion, not aspiring to 'reality'
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