I hope I'll be able to judge with headphones? My son has a vacation from School this next week, that will limit my listening opportunities.
I'm still puzzled by my (positive) listening results with a (small) positive spike at ~0.27 ms. I can't explain it. To count as anti cross talk a negative spike there should have the bigger benefit.
When I find the time I should really get the microphone out and measure at that spot in space and time. You know, the ear position, to look for clues to get me further.
I tried listening with and without that small positive spike (targeted solely at phantom sounds) and each time I prefer to listen with the spike.
My HT mix now has it too, it does seem to increase the amount of intelligibility and level out the tonal balance across the 'screen'. (too bad my TV is too small
)
If I use any signal changes between 0.27 ms and 3 ms sounds get worse fast! Longer than 3ms it's already less detrimental, but certainly does change the tonal balance, most apparent on vocals. This makes me think about the disagreement between the teachings of Toole and Geddes. Toole being an advocate of "a certain level of early reflections" while Geddes likes to avoid them completely, however Geddes seems to be "ok" with reflections starting at ~15 ms. I tend to side with Geddes on this point, although I do understand what Toole is getting at. I'd like to "see" the early reflections in a measurement (IR) of one of those Harmon listening rooms, they simply don't look much like my room to begin with.
The later these reflections are timed, the less the tonal shift in vocals seems to be. However the perception does still change and it seems to smooth the overall perception. Even the tonality seems to get smoothed by these late reflections. Provided it's frequency content is smooth enough over a large frequency spectrum.
The most spectacular part of those late timed reflections is the holographic rendition they can present. That 3D feel of what's in front of you. Space around the performers yada yada yada... It's all there... The higher the level of that late return, the less 3D-ness though. A little goes a long way.
Having conversations in different rooms with people you know have all these same effects on tonal balance, if you listen for it. Our brain tends to alter that perception for us (automatically) but when one really tries, it's easy to hear. (or just record it and play back on headphones for a more obvious result)
Almost all POP music already has ( a lot of) coloration on vocals, more often than not a deliberate choice. Making it much harder to judge how it's supposed to sound.
Anyway, I'm rambling on again...
I'm still puzzled by my (positive) listening results with a (small) positive spike at ~0.27 ms. I can't explain it. To count as anti cross talk a negative spike there should have the bigger benefit.
When I find the time I should really get the microphone out and measure at that spot in space and time. You know, the ear position, to look for clues to get me further.
I tried listening with and without that small positive spike (targeted solely at phantom sounds) and each time I prefer to listen with the spike.
My HT mix now has it too, it does seem to increase the amount of intelligibility and level out the tonal balance across the 'screen'. (too bad my TV is too small
)If I use any signal changes between 0.27 ms and 3 ms sounds get worse fast! Longer than 3ms it's already less detrimental, but certainly does change the tonal balance, most apparent on vocals. This makes me think about the disagreement between the teachings of Toole and Geddes. Toole being an advocate of "a certain level of early reflections" while Geddes likes to avoid them completely, however Geddes seems to be "ok" with reflections starting at ~15 ms. I tend to side with Geddes on this point, although I do understand what Toole is getting at. I'd like to "see" the early reflections in a measurement (IR) of one of those Harmon listening rooms, they simply don't look much like my room to begin with
.The later these reflections are timed, the less the tonal shift in vocals seems to be. However the perception does still change and it seems to smooth the overall perception. Even the tonality seems to get smoothed by these late reflections. Provided it's frequency content is smooth enough over a large frequency spectrum.
The most spectacular part of those late timed reflections is the holographic rendition they can present. That 3D feel of what's in front of you. Space around the performers yada yada yada...
It's all there... The higher the level of that late return, the less 3D-ness though. A little goes a long way.Having conversations in different rooms with people you know have all these same effects on tonal balance, if you listen for it. Our brain tends to alter that perception for us (automatically) but when one really tries, it's easy to hear. (or just record it and play back on headphones for a more obvious result)
Almost all POP music already has ( a lot of) coloration on vocals, more often than not a deliberate choice. Making it much harder to judge how it's supposed to sound.
Anyway, I'm rambling on again...
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Filter is in situation transducer dependent so with a track convolution test it has to be played back on towers although take your time, but in its only a couple of tracks your son can probably forgive us also i can convolute a AC/DC track so he can get happy and maybe help with his vote. So far used a couple of hours preparing convolution filter for left tower, it looks as below Z-phase curve in reverse and can you feel how it splits ones objective hat in tranducer is transfered to a non minimum device in amplitude domain so would love some serious ears and feedback, well back to life plus some work prepare insane filter for right tower
.Attachments
By this I suppose you mean in an impulse response such as that used by the shuffler? What's the amplitude of the spike?
As for the WE15A simulations, I could not quite follow your flow chart, but the description made it clear. The only experience I've had with the 15A was as a low mid in a 4-way system. There was about 5-7 meters behind the horns, damped with heavy velour curtains. The sound depth I don't particularly remember, it was the over all realism of the different spaces reproduced that was so amazing. Easily going from "they are here" to "you are there" with different recordings. In my lava cave the wall behind the speakers was about 10 meters back. That gave an amazing sense of depth, if the recording had it. Maybe a diffuse echo at ~55-75ms helps depth.
Yes, the amplitude is about 18 dB below the mains. The shuffler had a varying left and right part, these are both the same, used on the (L+R) only with subtraction of (L-R).
Pano said:
Let me first state that I'm envious here
. I would have loved to have experienced something similar. I bet it beat the Munich Hifi Silbatone setup by far. A late 'reflection like' (meaning broad FR spectrum) return will possibly give us that realism, the indescribable 'space' around performers. Holographic...
I had a diffusive tail (reverb) added after that 'fake horn' reflection, that longer tail might have helped in/or created the depth part.
The ambient channels I use have a very similar effect, but with the added draw back of only being able to use it up to ~3.5 KHz (12 dB slope) for side panned sounds or it will become obvious/noticeable.
I also use a downwards slope toward high frequencies on any ambient sound, but having it mixed in with mains I can let it run free up higher without it ever being obvious. It just sounds "lighter" that way.
A diffuse tail running past ~150 ms can give us a strong sense of envelopment, but it's much harder to get it to sound natural or obvious. Great fun to play with though! Makes that "you are there" sense even stronger. That only needs lower midrange content, no high frequencies.
Kind of simulating the reverberation that we just don't get/have in small rooms, down to 100-200 Hz or even lower.
I actually fooled around with real 'as measured' IR's from great halls, it actually does work, but also adds the same signature to everything. I got tired of it quickly.
The Lexicon "Random Hall" is the most convincing for me.
I'll go back to that reverb soon and continue these experiments. To much fun not to! I removed the 'fake horn reflection' from the mains, but kept the reverb tail that followed and put the Haas Kicker back into the ambient channels again.
That has been the most fun to listen to, engaging and 3D. While being completely transparent. All it does (or seems to do anyway) is get rid of my walls, while it doesn't attract any attention to itself. Up next I'll try to get back some envelopment as that always made me feel warm and fuzzy inside
.
Filter is in situation transducer dependent so with a track convolution test it has to be played back on towers although take your time, but in its only a couple of tracks your son can probably forgive us also i can convolute a AC/DC track so he can get happy and maybe help with his vote. So far used a couple of hours preparing convolution filter for left tower, it looks as below Z-phase curve in reverse and can you feel how it splits ones objective hat in tranducer is transfered to a non minimum device in amplitude domain so would love some serious ears and feedback, well back to life plus some work prepare insane filter for right tower
Which one are you meaning to inverse? Red or black? I'll await patiently, take your time
.
I'm glad to see that the ramblings and tweaking are still going on...
I'm still here... following. Some of the stuff I'd like to implement, and some is way out there...
Now, I know I could get a quick answer here, you guys are always on top of things, and I have trouble wrapping my head around this.
So, if I may get you guys distracted for a couple of posts....
I've seen someone say this is a great step response... It's a 3-way. Inverted tweeter, mid-woofer and bass woofers.
I thought a great response would be time aligned and coherent, which, to my eyes, this is not.
Why would someone say this step response is perfect?
I'm still here... following. Some of the stuff I'd like to implement, and some is way out there...
Now, I know I could get a quick answer here, you guys are always on top of things, and I have trouble wrapping my head around this.
So, if I may get you guys distracted for a couple of posts....
I've seen someone say this is a great step response... It's a 3-way. Inverted tweeter, mid-woofer and bass woofers.
I thought a great response would be time aligned and coherent, which, to my eyes, this is not.
Why would someone say this step response is perfect?
Attachments
Which one are you meaning to inverse? Red or black? I'll await patiently, take your time
Was probably too fast there, red is phase from your left speakers Z curve plot long time ago when we played around with passive motional networks, black is my reverse work so left tower convolution filter is ready and i'm about third way into right towers insane filter
.
@ perceval: Too much fun to not go on, even as an ever lasting learning experience...
I think that's a pretty good STEP response, for a 3 way, passively filtered. The tweet/mid etc. hand off pretty good to the next driver.
See this thread: Group Delay Questions and Analysis for an excellent example of a STEP, as measured at the listening spot of a Troels designed speaker with second order crossover in a more than great room.
If you look at an APL_TDA plot of that speaker you'll see what the wave front does over time.
That graph was taken with a STEP that was showing an excellent hand over with second order slopes. The STEP you showed will have the graph even more stretched out, the lower you go in frequency.
There's a lot of debate if this is actually audible or not. The ideal transducer would look like this:
(which is an actual measurement of my DAC)
My speaker's plot looks like this at the LP:
But just look at that room from Jim! I have all kinds of reflections showing up in the area after the wave front, he has a clear blue see of silence. My bass is quicker though!
(also note his Haas Kicker in action)
So the speaker in your graph would have an even bigger group delay, I do think that it would matter even more what the room would do with/to that result. It's one thing to have a speaker with an excellent STEP, the next hurdle would be the in room result.
I still believe it matters, to approach the ideal STEP response. Preferably at the listening spot. I don't believe (for one minute) you can test something like that reliably on headphones to sense this difference. It is a hear/feel experience. Your body listens too! Think of someone dropping something heavy beside you (without you knowing). That sound will be instant! So should a reproduction of such an event be that instant!
I think single full range drivers (or a large horn that plays most of the telephone band) are popular for a reason. But what do I know...
As far as a conventional speaker goes, it still is a fine example of a STEP. First order speakers are almost impossible, the Harsch filter does not give you complete transients as it does influence FR response. The filler driver is hard to do as well, a Synergy horn could pull it off, and DSP setups... pick your poison!
I think that's a pretty good STEP response, for a 3 way, passively filtered. The tweet/mid etc. hand off pretty good to the next driver.
See this thread: Group Delay Questions and Analysis for an excellent example of a STEP, as measured at the listening spot of a Troels designed speaker with second order crossover in a more than great room.
If you look at an APL_TDA plot of that speaker you'll see what the wave front does over time.
That graph was taken with a STEP that was showing an excellent hand over with second order slopes. The STEP you showed will have the graph even more stretched out, the lower you go in frequency.
There's a lot of debate if this is actually audible or not. The ideal transducer would look like this:
(which is an actual measurement of my DAC
)My speaker's plot looks like this at the LP:
But just look at that room from Jim! I have all kinds of reflections showing up in the area after the wave front, he has a clear blue see of silence. My bass is quicker though!
(also note his Haas Kicker in action)
So the speaker in your graph would have an even bigger group delay, I do think that it would matter even more what the room would do with/to that result. It's one thing to have a speaker with an excellent STEP, the next hurdle would be the in room result.
I still believe it matters, to approach the ideal STEP response. Preferably at the listening spot. I don't believe (for one minute) you can test something like that reliably on headphones to sense this difference. It is a hear/feel experience. Your body listens too! Think of someone dropping something heavy beside you (without you knowing). That sound will be instant! So should a reproduction of such an event be that instant!
I think single full range drivers (or a large horn that plays most of the telephone band) are popular for a reason. But what do I know...
As far as a conventional speaker goes, it still is a fine example of a STEP. First order speakers are almost impossible, the Harsch filter does not give you complete transients as it does influence FR response. The filler driver is hard to do as well, a Synergy horn could pull it off, and DSP setups... pick your poison!
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Because tests have proved especially if they controlled and blind that real many can't notice phase, but then again some can.
Now a advice that hope is alright give before you start discussions with these guys over there, else please slam me i can take it
noticed many times when you and xrk971 talk about you heard a time alligned system and what it stands for please get down on earth about what is what because those systems compromises also add pleasing and interesting distortions that could be mis concluded as what time allignment and transient perfect system really stand for, and it shouldn't be so hard to prove simply listen to some smooth head phones or visit Netherlands to hear wesayso's smooth towers
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Was probably too fast there, red is phase from your left speakers Z curve plot long time ago when we played around with passive motional networks, black is my reverse work so left tower convolution filter is ready and i'm about third way into right towers insane filter
I recognised that shape
. But I still do have that compensation network in place.... In fact, my FIR filter is created with that network in there. Would that not counteract what you would try to demonstrate?There are small differences in FR (and phase) if I take the compensation out.
The only thing I did remove was the high frequency zobel network. So my electrical phase curve still looks like this:
(that's what the Goldmund clone sees anyway)
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Do you think there are any distortion mechanisms that can result in a perceived increase in height ? The reason I ask is that I had a discussion with XRK971 here Question regarding phase differences of amps. where he an a couple of others thought that they could hear more vertical placement with phase accurate amps and I wondered if there was something else going on.Because tests have proved especially if they controlled and blind that real many can't notice phase, but then again some can.
Now a advice that hope is alright give before you start discussions with these guys over there, else please slam me i can take it
I recognised that shape
Good spottet
have been around it myself and best i can see is network only help motional flyback current to be dampned or shorted out much better and before than without a network where a higher current potential will reach back to power amp and via amps correction negative feedback network it probably would reach sensitive input area too, in this parallell situated position passive network doesn't change phase in reverse therefor resonance in reverse is into convolution filter in hope you get a replicate sound of what i hear up here. TC9 is a exemplarish smooth performer in impedance domain and beat 10F there so phase change from filter is not so much as for my 10F and they also both exemplarish in LeX deviation, well 10F in 4 ohms version looks really perfect over at ZappAudio and TC9 looks also very good so this stock filter should work for them else if there was LeX deviation for Xmax inward/outward verse rest position then we could add same filter using latests version of Fabfilter in it has got dynamic adjustable EQ features and if we cascade a IIR that takes care of phase plus amplitude followed by a FIR filter that inverse amplitude back to neutral we end a sum where only phase is changed but with the add on feature to have dynamic settings to dial on.Not shure but if there is small response difference then network is probably set a tiny squizze too tight down at DC point area you know lowering the natural RE number a very little bit, if that is good or bad i don't know, could imagine its a good brake to stop motion and get tight low end impulses and sure your new amp don't care it see a tiny brake here and there
have a Duelund paper somewhere about resonance and inductance networks and if remember right he state very serios with usual passion how important it is to have both networks in place no matter non linearity area is outside passband and also how important it is filter is precision with correct component values.Hope to get filter finished and covoluted few tracks tonight or first thing tomorrow.
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Ok, thanks for the replies, guys.
I know we can get stuck on measurements sometimes and forget other things.
But I was believing that the best STEP would be something like a Dunlavy speaker, phase and time coherent.
So, if most people do not hear phase inversions from XOs, than why is it the motto for full range speakers, where the claim is no phase issues induced by XO?
I know we can get stuck on measurements sometimes and forget other things.
But I was believing that the best STEP would be something like a Dunlavy speaker, phase and time coherent.
So, if most people do not hear phase inversions from XOs, than why is it the motto for full range speakers, where the claim is no phase issues induced by XO?
I, for one have never voted against the importance of phase coherency.
My phase plot follows the band passed frequency curve as if it were one big full range source. This is not linear phase but it is phase coherency.
As I said in the example in my previous post, real sounds come at you all at once. That's what I want from my reproduction chain too. If I were running speakers with crossovers I'd add DSP compensation for them. (and still fix the room)
I don't care if a large part of the community thinks otherwise. I do like the view Dunlavy had and agree with this view. A deviation in time is still a form of distortion of the signal.
From his paper Loudspeaker Accuracy:
This bottom example shows a true STEP as measured from a DAL speaker.
To get my own bandpass behaviour at the listening spot I correct phase below 500 Hz with DRC-FIR and even do some smaller adjustments in RePhase. Making sure it works in more than that single sweet spot(!) For that to happen the room has to work with the speaker.
Here's another recap on STEP I found online:
Loudspeaker step response measurement and explanation – Audio Judgement
My goal didn't end with the speaker, I wanted to get that at the listening seat.
The lower your system can play, the longer the STEP plot would 'fly' above the "0" line. Most plots of the phase coherent published STEP plots on Stereophile only show 3 to 4 ms.
My own plots "fly higher" a little longer:
And that shape is largely dependant on the frequency target curve. With a straight line target, so no tilted down higher frequencies etc. it flies above the zero line up to 15 ms.
(only true in the sweet spot with both stereo speakers playing, something I hope to fix with the subs)
Is it important? To me? Yes. But I regard things like tonal balance and a reasonable tracking of the frequency curve between the left and right channel and even the room response itself as more urgent matters to fix. The phase coherency is only one small part of it all, but I don't disregard it. I just wanted to get as many of these little things right as I can as a base goal. Meaning you have to look at what the room does to be able to get plots like that out at the listening spot (and beyond). You can't just fix the room problems with DSP and expect the next seat measurement to look the same. So I investigated each wiggle in the IR that was making the STEP/IR result out of shape (in more than one single spot). The speaker has to have low diffraction as well, if it needs to hold up similar behaviour off-axis.
Due to having an array of drivers, moving up and down has no influence. Think about a woofer and tweeter on a regular baffle and move the microphone up and down. Could it hold that STEP shape? No way(*). That's why I said only a few concepts even have a shot at getting this right over a larger (listening) area. A single driver can do it over it's pass band, coaxials might be able to do it, Dunlavy's WMTMW, Danley's Synergy...
Floor and ceiling reflections etc. would determine how much of that shape holds up. You've got to be at the right spot too. A Dunlavy speaker is meant to be coherent at listening distance. On this page you see it at 3m distance.
See how the room would become more and more important at these longer distances?
So what does it do? Why would I even want it? It has a tiny little advantage in being more "immediate", more 'life' like. Hear and feel the impact, even in the midrange (on parts of your body like your eyelids). Keeping excellent harmonic structures over a large part of the frequency curve making these harmonics sounds sweet and full bodied, floating even. Almost makes it bloom for lack of a better explanation, believable. Though even here the room still plays a big part too. And probably things like the amplifier could have an influence, one that has second order clearly above third would probably sound more sweet.
Speaker drivers differ there too, in sound and harmonics. Look at subjective reports on speaker drivers. Drivers that exhibit higher third order are often praised for their detailed sound. Others find them clinical or harsh. But drivers with mainly second order are often called musical. To me, getting the phase right is just that: musical.
In all honesty, going into this project I expected it to be very important. That view certainly has changed for me. It's part of it, but not the most vital part to hunt for. The sweetness I attribute partly to phase. But it's not as obvious on every recording. The room response and every other little thing probably has even more influence on what we perceive. But I still think it helps get that sweet realness trough to our ears, if everything else supports it too. Part of the magic.
The hole in the Stereo perception, due to cross talk is definitely fighting its importance.
Is this helping at all? I've heard so many 'sounds' in recordings that startled me with their sense of realism. Be it in voices or percussive sounds etc. Effects in movies etc. Even complete songs that took me for a ride. It's much harder to get that suspense of disbelieve every time you listen. Keeping my eyes on every detail seems to work to get me close enough to that ideal. Phase is just one more factor to get right for a natural sounding reproduction. But it's not as vital by itself as I once believed.
I use whatever works (for me) to get closer to that goal of realism. That's what all the experiments are about.
(*) = Unless the sources are less than a 1/4 wave length apart at their crossover frequency. Then they would hold up it's phase relationship for longer over the vertical axis as well. Think of the Synergy design, it keeps that relationship over any angle the horn covers! Other speakers like the Dunlavy big towers get it trough symmetry at a certain distance. Arrays are dependant on distance too, I fix my IR at a listening distance, moving towards or away from the speaker does change the overall relationship.
Just not as much as a single tweeter/mid combo. It will be more gradual.
My phase plot follows the band passed frequency curve as if it were one big full range source. This is not linear phase but it is phase coherency.
As I said in the example in my previous post, real sounds come at you all at once. That's what I want from my reproduction chain too. If I were running speakers with crossovers I'd add DSP compensation for them. (and still fix the room)
I don't care if a large part of the community thinks otherwise. I do like the view Dunlavy had and agree with this view. A deviation in time is still a form of distortion of the signal.
From his paper Loudspeaker Accuracy:
This bottom example shows a true STEP as measured from a DAL speaker.
To get my own bandpass behaviour at the listening spot I correct phase below 500 Hz with DRC-FIR and even do some smaller adjustments in RePhase. Making sure it works in more than that single sweet spot(!) For that to happen the room has to work with the speaker.
Here's another recap on STEP I found online:
Loudspeaker step response measurement and explanation – Audio Judgement
My goal didn't end with the speaker, I wanted to get that at the listening seat.
The lower your system can play, the longer the STEP plot would 'fly' above the "0" line. Most plots of the phase coherent published STEP plots on Stereophile only show 3 to 4 ms.
My own plots "fly higher" a little longer:
And that shape is largely dependant on the frequency target curve. With a straight line target, so no tilted down higher frequencies etc. it flies above the zero line up to 15 ms.
(only true in the sweet spot with both stereo speakers playing, something I hope to fix with the subs)
Is it important? To me? Yes. But I regard things like tonal balance and a reasonable tracking of the frequency curve between the left and right channel and even the room response itself as more urgent matters to fix. The phase coherency is only one small part of it all, but I don't disregard it. I just wanted to get as many of these little things right as I can as a base goal. Meaning you have to look at what the room does to be able to get plots like that out at the listening spot (and beyond). You can't just fix the room problems with DSP and expect the next seat measurement to look the same. So I investigated each wiggle in the IR that was making the STEP/IR result out of shape (in more than one single spot). The speaker has to have low diffraction as well, if it needs to hold up similar behaviour off-axis.
Due to having an array of drivers, moving up and down has no influence. Think about a woofer and tweeter on a regular baffle and move the microphone up and down. Could it hold that STEP shape? No way(*). That's why I said only a few concepts even have a shot at getting this right over a larger (listening) area. A single driver can do it over it's pass band, coaxials might be able to do it, Dunlavy's WMTMW, Danley's Synergy...
Floor and ceiling reflections etc. would determine how much of that shape holds up. You've got to be at the right spot too. A Dunlavy speaker is meant to be coherent at listening distance. On this page you see it at 3m distance.
See how the room would become more and more important at these longer distances?
So what does it do? Why would I even want it? It has a tiny little advantage in being more "immediate", more 'life' like. Hear and feel the impact, even in the midrange (on parts of your body like your eyelids). Keeping excellent harmonic structures over a large part of the frequency curve making these harmonics sounds sweet and full bodied, floating even. Almost makes it bloom for lack of a better explanation, believable. Though even here the room still plays a big part too. And probably things like the amplifier could have an influence, one that has second order clearly above third would probably sound more sweet.
Speaker drivers differ there too, in sound and harmonics. Look at subjective reports on speaker drivers. Drivers that exhibit higher third order are often praised for their detailed sound.
Others find them clinical or harsh. But drivers with mainly second order are often called musical. To me, getting the phase right is just that: musical.In all honesty, going into this project I expected it to be very important. That view certainly has changed for me. It's part of it, but not the most vital part to hunt for. The sweetness I attribute partly to phase. But it's not as obvious on every recording. The room response and every other little thing probably has even more influence on what we perceive. But I still think it helps get that sweet realness trough to our ears, if everything else supports it too. Part of the magic.
The hole in the Stereo perception, due to cross talk is definitely fighting its importance.
Is this helping at all?
I've heard so many 'sounds' in recordings that startled me with their sense of realism. Be it in voices or percussive sounds etc. Effects in movies etc. Even complete songs that took me for a ride. It's much harder to get that suspense of disbelieve every time you listen. Keeping my eyes on every detail seems to work to get me close enough to that ideal. Phase is just one more factor to get right for a natural sounding reproduction. But it's not as vital by itself as I once believed.I use whatever works (for me) to get closer to that goal of realism. That's what all the experiments are about.
(*) = Unless the sources are less than a 1/4 wave length apart at their crossover frequency. Then they would hold up it's phase relationship for longer over the vertical axis as well. Think of the Synergy design, it keeps that relationship over any angle the horn covers! Other speakers like the Dunlavy big towers get it trough symmetry at a certain distance. Arrays are dependant on distance too, I fix my IR at a listening distance, moving towards or away from the speaker does change the overall relationship.
Just not as much as a single tweeter/mid combo. It will be more gradual.
Attachments
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So how was that for a pretty long winding answer?
I have even more to add. What I find strange is that a lot of the objective crowd, the ones that really take to measuring, saying what goes in must come out etc. disregard the phase relationship with great ease.
Isn't it just another deviation from the input signal? Only this time it's primary part is time distortion? I've mentioned it before, in music I think timing is a very vital part of musicality. That's why I won't ever deny it has it's role.
When a band is enjoying themselves on stage, they start to move along to the rhythm/beat. Not because they want to, it's a feel thing and they react. Tapping your feet to the music. A little phase turn isn't going to upset the enjoyment though. A lot of music will probably have been mixed on speakers that didn't have the coherency.
How coherent are the speakers in an amplified concert? The room will be as or even more important. Take up an acoustic guitar and make it sing, I'll admit I'm not even that good at it, give it to a competent player and he'll get out the harmonics that are supposed to come out. Yet I do see lots of players that lack that skill, while making a living from it.
Maybe it's not that important for all of us... lots of people could live with mp3. I love to drown in the music, get lost in it. I love what it does to me on an emotional level. That's why I'm in on this hobby. If I go see an amplified concert, I do enjoy it more if the room seems to carry the tune. At moments like that I'm not worried about phase.
I have even more to add. What I find strange is that a lot of the objective crowd, the ones that really take to measuring, saying what goes in must come out etc. disregard the phase relationship with great ease.
Isn't it just another deviation from the input signal? Only this time it's primary part is time distortion? I've mentioned it before, in music I think timing is a very vital part of musicality. That's why I won't ever deny it has it's role.
When a band is enjoying themselves on stage, they start to move along to the rhythm/beat. Not because they want to, it's a feel thing and they react. Tapping your feet to the music. A little phase turn isn't going to upset the enjoyment though. A lot of music will probably have been mixed on speakers that didn't have the coherency.
How coherent are the speakers in an amplified concert? The room will be as or even more important. Take up an acoustic guitar and make it sing, I'll admit I'm not even that good at it, give it to a competent player and he'll get out the harmonics that are supposed to come out. Yet I do see lots of players that lack that skill, while making a living from it.
Maybe it's not that important for all of us... lots of people could live with mp3. I love to drown in the music, get lost in it. I love what it does to me on an emotional level. That's why I'm in on this hobby. If I go see an amplified concert, I do enjoy it more if the room seems to carry the tune. At moments like that I'm not worried about phase
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It was okSo how was that for a pretty long winding answer?
Phase coherence in speakers seems a good idea, obviously getting them to marry well together, I recently time aligned my subs with my Eikonas, and I believe it sounded better, isn't that all that matters, might have been just the feel good factor? Plenty has gone on that we can't control in the whole process, doesn't mean it's not a good idea to control what we can, and it's simpler now with the processing power we have.
That's not easy to answer... I do know certain frequency balance tricks can make height queues more obvious. Tweeter territory.
But phase accurate amps? The sweet sounding amps could make the bloom effect I mentioned a bit more pronounced? I loved a class A amp on my Vifa XT tweeters in my car. Even though the tweeters themselves were mounted a bit above knee height (Porsche 911) the stage never seemed to be low.
I think that had little to do with phase accurate amps, just clever EQ adjustments and a properly time aligned crossover.
Guess try ask xrk971 and AKSA if they have high precision lab gear measurements that really prove their amps have accurate phase that follow amplitude and their competitors or all the great Japanese amps from 80-90ties have not, and what is accurate phase well in my eyes its minimum phase same as wesayso call coherent phase and reason is nature or enviroment is of that kind. Seen xrk971 and AKSA promote dial in compensation network by ear, network that normal is for stabilization but in amp is super overstable they use it for kind of inverse phase as normal would need power from a FIR filter, well is that phase accurate and how could a man sit and dial in that magic point over his own replay system and call this natural and phase accurate
what we get from that situation is the illusion from their ears over that particular system at that particular time, how can we know their actual reference system is not miles away from the real reference. Thumb up passion about audio there but the high numbers of designs and statements mostly from subjective conclusions is a big worry in my view, next week they learn about a new schematic or layout/implementation then it starts all over again with new statements and hardware release.That said if xrk971 say to hear height or phase that must be it although chance there its a illusion is also at place, phase can be audiable and push things but we cant repair in recordings materials that first device the microphone plus our ears are two cascaded passband limitations so we can forget recreate stuff below or above that passband.
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Thank you Ricky, you get it, and your grasp of the English language is exemplary Perhaps I should have given up sooner, only I wanted to get to the bottom of it, and X appeared genuinely interested and open-minded, as for AKSA I'm not sure why he got so upset, anyway, I can only conclude there's nothing to it other than what is already well known.
Perhaps I should have given up sooner, only I wanted to get to the bottom of it, and X appeared genuinely interested and open-minded, as for AKSA I'm not sure why he got so upset, anyway, I can only conclude there's nothing to it other than what is already well known.