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Being aware of misbehaviour of the playback chain


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19 hours ago, fas42 said:

Well, at the moment I "trying to get to the bottom" of why your brass doesn't sound, umm, brassy - some setup issues are volume dependent, some are not - the term to bring into one's thinking is ... "troubleshooting".

You are laboring under a misconception, so save your effort. The reason why my brass doesn't sound like real brass has absolutely nothing to do with playback. It's because the sound of real brass cannot be captured. You seem also to be laboring under another misconception. You seem to think that microphones and the rest of the recording chain are perfect. Nothing could be further from the truth. I've used every kind of microphone for recording there is, condenser (both large and small capsule), dynamic and ribbon mikes as well as condenser variations such as RF and electret. I've used about every brand, Sony C37P, C500, ECM-22P; Neumann U-47, U-87, KM-184; AKG-414; RCA 44-BX, 77; Telefunken ELA-M-270, ELA-M-260, AK-47, C-12; Shure SM-33 (AKA The "Johnny Carson Microphone" - but being a Digger, you wouldn't know about that, would you?) etc., etc., etc. I mention all these mikes to illustrate to  you that I know wherefore I speak. NO microphone can capture the true sound of brass,this is especially true of trumpets and coronets, but french horns and trombones cannot be fully, accurately captured either in some registers. Enough is captured so we can identify the instrument's essential sonic signature, of course, but never enough to convince anybody, on playback (and I don't care how good your playback system is) that they are hearing real brass playing in a real space. Can't be done. end of that story.

 

19 hours ago, fas42 said:

At the time I bought the Perreaux it sounded excellent for what I was expecting a quality amp to do, and this was well before my "epiphany" ... I listened to other monsters like Krell, and Mark Levinson at the time  - and they sounded like, well, shit ... my decision was easy ...

 

Back then, all amps were inferior - there was no point in going backwards.
 

Why bring it up? I don't have any experience with early Mark Levinson Power amps, but I do recall that the original Mark Levinson preamp sounded great for its day. I agree about early Krell amps. They sounded very dry (dry? how about arid!) and were at the same time very cold and very clinical sounding. OTOH, I have two amps at the moment; one is a Harman-Kardon HK990, which is my main amp, and the second is a Krell KAV-300i. Both are 150 WPC, and while both are excellent, I have to give the edge to the HK.  

George

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17 hours ago, anwaypasible said:

a laser microphone works by simply using chorus down the beam to rotate back a difference that ultimately depends on the sensitivity of the input sensor.

it's like having a sword & anything that touches it (including the air) is going to cause the chain-linked web to deviate from default & that deviation is audio.

 

..somebody said something about mics being the weak point.

They still are. They are transducers, and like other transducers, phono cartridges, and speakers, none can transduce sound into electricity (or vice versa) without incurring losses. Those losses can be manipulated and minimized, but they cannot be eliminated. As long as the conversion isn't perfect (and it never can be) no transducer will sound exactly like the source.  

George

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19 hours ago, fas42 said:

As it pertains to audio, there was an extremely "weak link" in the phone chain to my house, which still allowed normal operation - this corresponds to competent sound - and finally that link had deteriorated too much, phone line and net linkage were almost unusable - corresponding to less than competent sound. There was a dramatic switch in competence of communications to our home - all dependent on precisely what the chaotic status of that weakness was. Resolution was tracking it down, and fixing the problem.

 

The company's junction box was part of the "equipment" for getting the signal through - whether I could fiddle with it or not is not relevant - one way or another that flaw had to be resolved, for the illusion of good communications to form ... :)

 

That sounds like you may be using FTN, and perhaps your copper pair from the exchange has either a partial s/c or earth leakage problems.

If it was broadband , the coax connector(s)  used on the poles may be the cause. A batch of coax connectors supplied to Optus (SingTel) developed these problems. This can also happen on major roads used to corrosion from engine fumes.

 I had this problem at Ryde in Sydney where there was a State Transit bus depot just up the road a bit.

 

How a Digital Audio file sounds, or a Digital Video file looks, is governed to a large extent by the Power Supply area. All that Identical Checksums gives is the possibility of REGENERATING the file to close to that of the original file.

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1 hour ago, gmgraves said:

You are laboring under a misconception, so save your effort. The reason why my brass doesn't sound like real brass has absolutely nothing to do with playback. It's because the sound of real brass cannot be captured.

 

It's broken record time, again :) ... my contribution to such is, how do you know that? By listening to it through a replay system, or because some magic book somewhere said so? Peter has no problems hearing brass workin' properly - does he live in a magic land where the rules you believe in no longer apply?

 

1 hour ago, gmgraves said:

Why bring it up? I don't have any experience with early Mark Levinson Power amps, but I do recall that the original Mark Levinson preamp sounded great for its day. I agree about early Krell amps. They sounded very dry (dry? how about arid!) and were at the same time very cold and very clinical sounding. OTOH, I have two amps at the moment; one is a Harman-Kardon HK990, which is my main amp, and the second is a Krell KAV-300i. Both are 150 WPC, and while both are excellent, I have to give the edge to the HK.  

 

In those decades nearly all power amps seemed to have poor rejection of self induced noise on their voltage rails - once you learn how to use, say, a musical track to pick this, yes, misbehaviour then it becomes trivial to assess an unknown amp - one doesn't need sophisticated test gear, just the right signal source, and ones' ears.

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19 minutes ago, sandyk said:

 

That sounds like you may be using FTN, and perhaps your copper pair from the exchange has either a partial s/c or earth leakage problems.

If it was broadband , the coax connector(s)  used on the poles may be the cause. A batch of coax connectors supplied to Optus (SingTel) developed these problems. This can also happen on major roads used to corrosion from engine fumes.

 I had this problem at Ryde in Sydney where there was a State Transit bus depot just up the road a bit.

 

Still haven't done the feared NBN to our area yet! Pure, cough cough, copper back to the exchange - the tech said that the extra box on the pole, 30 metres away, didn't make sense - why was it added to the circuit?

 

We're in essentially rural area - a small township in the Blue Mountains ... but the train line is a stone's throw away - the fumes from the 4 header diesel electric freight trains going through, perhaps?

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4 hours ago, fas42 said:

By listening to it through a replay system, or because some magic book somewhere said so? Peter has no problems hearing brass workin' properly - does he live in a magic land where the rules you believe in no longer apply?

 

Prior to reading this, I thought exactly the same but decided not to comment on it. Now it has been mentioned anyway I'd have to say I agree (fully). But what's also on my mind is : to take this out of subjectiveness we should be able to measure it. But then this can only be done theoretically and now we enter the area of again theory that George has to be correct. Theoretically which again is not practice.

 

Err ... :$

 

So while we should easily be able to measure and prove that the squareness of brass (btw I'd say brass isn't the most square of that group of instruments at all BUT it requires a fundament e.g. a trumpet doesn't carry) requires more bandwidth than the microphone can capture and/or the loudspeaker can reproduce.

Easy enough.

Well, not so easy at all because first we must prove that we ourselves require that bandwidth in order to differentiate where differentiation is required. So when is the copper turning into a wooden flute ?

 

The essence of this latter must be understood because it is the difference between "all quite square" and "all quite sine (sinus)" and it is the bandwidth which may change the quite square into quite sine ... of the complete chain and this includes our ears.

So if I can't hear beyond 12KHz I can't differentiate between a violin and a flute (mind you, same thing) ? Well, if it would be about the necessity to (register and) perceive say 40KHz, then yes. But we know it doesn't work like that. So let's say it is about the complexity and how all isn't to end up in a mesh so the violin is smeared into a flute.

And now your DAC is at stake and especially its filtering means ... (but I won't continue on this)

 

I did not say it explicitly in this thread, but I compare with real instruments. So I read that I am accused of being highly subjective just because I listen "on behalf of you" (all) but I am explicitly not. I listen for neutrality and compare with real instruments. And let me try to point out that the Phasure DAC is not that I can read anywhere - "accused" of carrying a sound. Nowhere, anywhere, people are able to describe its sound and I can't myself either. This is because I won't allow any character of devices ... which doesn't prohibit they can improve.

 

So yes, brass is difficult. It also is a measure, but it can't be an explicit measure because in my view I hear too few passing by in my real life. I mean, at the close (like listening room) distance. But this is how a tuba can serve the purpose. And in my view a tuba is as difficult. Not really because of its "copper" nature (it isn't copper as far as I know) but for mentioned fundament it just is. There's parts of the speed just the same, but it isn't about speed (says me). It is about the combination of infinite rise of fundament - which is speed but of a very strange order. Say it is about the slowest frequency and how it can rise in level the fastest. On-off sound.

Envision a periodic low frequency wave of 50Hz but now switch that off for half of the time so a 100Hz infinite speed nature emerges (the on and the off is as square / infinite of frequency as can be) and the fundament is 50Hz.

So when the tuba behaves as I expect (and tuba's do pass by at few feet distance regularly) then the copper will behave too. This is not theory but practice I explicitly test for. Say that it goes like : OK, this tuba Hans Theessink is using there won't get better today, ... hey - yes, that trombone sure sounds more now like it (to my sheer perception which thus is subjective).

 

So using a certain means to tune in the more unknown, is say a daily exercise.

 

Anyway it is hard to state that mine - or anyone's system can represent brass (and beyond) in full, already because of the theory that next year my system will behave better than today just because I think of some whatever hoopla in a few months time that will be aiding again. I mean, it is hard to hold on to "it is THE best" when I prove myself next year that it wasn't.

But between my system and say Frank's there is a whole world of difference already (I am confident) with George's as a third. So anyway all we can do is compare with reality. And give our systems a so-called necessary bandwidth (should be the bandwidth our ears/brain allow for at least but also not really more, I think). And this most certainly also applies to the low end. And add current current and more current to do it for real ("possible" current, for the insiders).

 

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1 hour ago, PeterSt said:

So while we should easily be able to measure and prove that the squareness of brass (btw I'd say brass isn't the most square of that group of instruments at all BUT it requires a fundament e.g. a trumpet doesn't carry) requires more bandwidth than the microphone can capture and/or the loudspeaker can reproduce.

Easy enough.

Well, not so easy at all because first we must prove that we ourselves require that bandwidth in order to differentiate where differentiation is required. So when is the copper turning into a wooden flute ?

 

Not quite sure what you're saying here, Peter - you can "prove that a microphone can't record a trumpet"; is that what you're stating?

 

1 hour ago, PeterSt said:

Anyway it is hard to state that mine - or anyone's system can represent brass (and beyond) in full, already because of the theory that next year my system will behave better than today just because I think of some whatever hoopla in a few months time that will be aiding again. I mean, it is hard to hold on to "it is THE best" when I prove myself next year that it wasn't.

But between my system and say Frank's there is a whole world of difference already (I am confident) with George's as a third. So anyway all we can do is compare with reality. And give our systems a so-called necessary bandwidth (should be the bandwidth our ears/brain allow for at least but also not really more, I think). And this most certainly also applies to the low end. And add current current and more current to do it for real ("possible" current, for the insiders).

 

 

Brass is difficult because of the waveform of the attack of the note - transients are pretty well the Achilles Heel of audio; the "bite your head off" intensity of the sound is usually where they fall short, and always reveal their 'fakeness'.

 

There certainly would be differences between your system and mine - yours is in better 'average' state of tune, on an ongoing basis; it does low bass, which I never go for; it would go louder, because of the components you're using; and for simple music where tonality is everything, it would be more "beautiful". And that's because what I chase is a bit different, and centres on completely invisible speakers, irrespective of recording quality and listening position - because this happens to be an excellent 'metric'.

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10 hours ago, gmgraves said:

They still are. They are transducers, and like other transducers, phono cartridges, and speakers, none can transduce sound into electricity (or vice versa) without incurring losses. Those losses can be manipulated and minimized, but they cannot be eliminated. As long as the conversion isn't perfect (and it never can be) no transducer will sound exactly like the source.  

a laser microphone isn't a transducer.

any microphone that records audio at a higher distortion percentage than the microphone's limit is capturing the signal perfect.

anyways, i've got the way nuclear speakers work on my facebook page - the reverse holds true to build a nuclear microphone & the only reason nuclear speakers are built is because they reach the mecca for more than one room size (speakers are limited to one specific room size).

 

seems odd that you put that much weight towards a situation that can backfire.

such as a microphone element that is magnetically biased to exaggerate all movement (or not) & is held at a gravitational freedom of static bias - because when inertia can meet neutrality, you claimed it isn't possible; but all that really tells me is the word phenomenon holds true to it's typical awe struck glare people give when they experience a phenomenon.

 

i might as well go on to say, why did they ever bother to create microphones & speakers if 'no transducer will sound exactly like the source' .. right? because then the audio we hear out of the speaker doesn't resemble anything like what was supposed to be recorded.

i guess by your way of thinking we all enjoy listening to jibberish noise from the speaker cones.

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56 minutes ago, fas42 said:

Peter - you can "prove that a microphone can't record a trumpet"; is that what you're stating?

 

Not at all.

Instead I try to say that it isn't necessary to record the instrument with all its frequency because you won't be able to perceive it. Besides I am claiming that this all isn't about enormously high frequency. So again but with different wording :

 

If you listen to the live orchestra and the copper section, so you have the idea that there's not sufficient frequency in your ears to perceive the "brass" well ?

So that.

Until what frequency can you hear ?

So that.

 

It isn't about frequency. It is about speed though which to a certain extent *is* frequency, but which it is not when you switch on/off abruptly lower frequencies. Now it is suddenly about transients (indeed) but for the lower frequencies. The mass of the diaphragm, this stiffness at the same time and the current to do it (way fast). The efficiency. The not being held back (hence open baffle).

 

1 hour ago, fas42 said:

intensity of the sound

 

You say it yourself.

 

1 hour ago, fas42 said:

it does low bass, which I never go for

 

So now you contradict somewhat with the notice that if "low bass" (as in really low like 20Hz like) is capable, the regions which matter for sure will be. All is about (audible) distortion and any frequency which can't be set up fast enough (fairly easy for the higher frequencies) will distort by various means. If your woofer section cuts off at any random frequency which is played, you will have distortion. It just can't be avoided. Notice that I refer to a cut off (or less capable) at e.g. 50Hz, that starting to "wobble" (so to speak) and that causing distortion (frequency not correlated to the music) in the upper regions of the driver. So your 100Hz and 200Hz etc. are now worthless too.

 

The biggest problem is the recognition of distortion in the lower regions because we enter the world of not really knowing how the sound (the instrument) was intended. An electric (B) bass of 31Hz is to sound how (knowing that the bass player uses his deliberately distorting amp + woofer to capture the sound by microphone) ? The synth playing at 24Hz ... is that a square ? or does it only sound square because of distortion ? Even that organ pipe producing 32Hz - which should be a sine - may not be utilized like that because of adjacently used resonating other frequencies; you'd have to know the piece. And then again, when you have the idea all sounds nice with the 32Hz, it maybe way too soft because 6dB down (and now the woofer does not distort).

I think we won't find many people in here used to undistorted all the way down at realistic levels. It requires re-learning to listen. Really so.

But it is another world too ...

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If I may point out a few ‘errors of thinking’ here.

 

In order to measure a musical signal, the measurement is subject to the same transformation from pressure wave to electrical impulse as the original recording. So you can’t easily measure the shortcoming of a recording as both are passed through similar transducers.

 

Then there’s live vs. recorded sound. Live sound comes from a single source and is subject to the acoustics of its immediate surroundings. Recorded music comes from 2 sources, already includes the recording venue’s acoustic signature then is subject to the replay venue’s acoustic signature. 

By definition, live music will have a different acoustic signature vs recorded music

 

Finally, there is the influence of the room. In live music, there’s only the single source and its associated acoustic. No mental ‘illusion’ required. In recorder music there’s 2 sources which must be replayed accurately enough to produce the illusion of a single source. As soon as the room interacts with those 2 sources of sound waves, by way of nodes, early reflections etc. It acts to damage or destroy the illusion. Thus, unless the room is highly transparent to the sound it will a. Add its own acoustic and b. Destroy the illusion of a single source. 

 

So you can mod equipment all you like. If rooms are not highly transparent and neutral, you’re still screwed

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58 minutes ago, Blackmorec said:

Finally, there is the influence of the room. In live music, there’s only the single source and its associated acoustic. No mental ‘illusion’ required. In recorder music there’s 2 sources which must be replayed accurately enough to produce the illusion of a single source. As soon as the room interacts with those 2 sources of sound waves, by way of nodes, early reflections etc. It acts to damage or destroy the illusion. Thus, unless the room is highly transparent to the sound it will a. Add its own acoustic and b. Destroy the illusion of a single source. 

 

So you can mod equipment all you like. If rooms are not highly transparent and neutral, you’re still screwed

 

Yes, that's the typical logic appealed to - and at one point I would have agreed to that being a reasonable POV. However, I have the 'evidence' of my  own ears telling me that it doesn't always have to be that way - in high quality replay the acoustic of the recorded event is no longer a minor player compared to the influence of the listening room; subjectively, it "takes over", dominates the reaction of the room - and the ear/brain ignores the contribution of the room. This is why it seems such a "magical" transformation when it occurs - no longer is there just something creating sound in your room; it now becomes you being present at the event when it was recorded.

 

People say that this "taking you to the event" is a characteristic of the recording - in part so; when the acoustic elements are very strongly emphasised during the making of the recording then having a neutral room, etc, helps that happen. And that is a hint of what can occur when the SQ is great enough - no crutches of having a totally suitable listening environment are then required; your mind dismisses the input of the room, the contribution of room reflections, etc, are masked by the acoustic message of the recording only.

 

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2 hours ago, PeterSt said:

It isn't about frequency. It is about speed though which to a certain extent *is* frequency, but which it is not when you switch on/off abruptly lower frequencies. Now it is suddenly about transients (indeed) but for the lower frequencies. The mass of the diaphragm, this stiffness at the same time and the current to do it (way fast). The efficiency. The not being held back (hence open baffle).

 

I could call it speed, but it's the "speed" of the electronics, for me. I can have a speaker which sounds quite mediocre, midfi even - and without doing anything substantive to the speaker itself, can get it to produce "brutal" bass runs, etc, with ease. I have quite a number of recordings which I have never heard another system, no matter how spectacular their nominal bass reproduction capabilities are, get right - bass lines which should be "fast", tight, gut wrenching, are just flabby, with no visceral impact.

 

Classic Boney M. tracks are a good test here - they have an intense "grunt" to the beat, which underpins everything; when missing, so much is lost.

 

2 hours ago, PeterSt said:

The biggest problem is the recognition of distortion in the lower regions because we enter the world of not really knowing how the sound (the instrument) was intended. An electric (B) bass of 31Hz is to sound how (knowing that the bass player uses his deliberately distorting amp + woofer to capture the sound by microphone) ? The synth playing at 24Hz ... is that a square ? or does it only sound square because of distortion ? Even that organ pipe producing 32Hz - which should be a sine - may not be utilized like that because of adjacently used resonating other frequencies; you'd have to know the piece. And then again, when you have the idea all sounds nice with the 32Hz, it maybe way too soft because 6dB down (and now the woofer does not distort).

I think we won't find many people in here used to undistorted all the way down at realistic levels. It requires re-learning to listen. Really so.

But it is another world too ...

 

I've heard other rigs which should do low organ pipe well, but they have been disappointing - so far from conveying the majesty of that instrument that I have not the slightest doubt that it's "wrong"; so how much distortion they're producing at the lowest frequencies doesn't concern me; get the the stuff higher up right first, and then I'll worry about the very bottom! :P

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1 hour ago, fas42 said:

 

Yes, that's the typical logic appealed to - and at one point I would have agreed to that being a reasonable POV. However, I have the 'evidence' of my  own ears telling me that it doesn't always have to be that way - in high quality replay the acoustic of the recorded event is no longer a minor player compared to the influence of the listening room; subjectively, it "takes over", dominates the reaction of the room - and the ear/brain ignores the contribution of the room. This is why it seems such a "magical" transformation when it occurs - no longer is there just something creating sound in your room; it now becomes you being present at the event when it was recorded.

 

People say that this "taking you to the event" is a characteristic of the recording - in part so; when the acoustic elements are very strongly emphasised during the making of the recording then having a neutral room, etc, helps that happen. And that is a hint of what can occur when the SQ is great enough - no crutches of having a totally suitable listening environment are then required; your mind dismisses the input of the room, the contribution of room reflections, etc, are masked by the acoustic message of the recording only.

 

Unfortunately this is really only wishful thinking I’m afraid.  In stereo replay, sound waves that reach the ears are already modified by the room. If the effects of the room are benign, then you’ll hear mostly original signal and its acoustics. But if the effects of the room are profound....room nodes and strong reflections, it modifies the sound waves that reach your ears, changing frequency amplitude and phase, which of course impedes your brain’s ability to construct the illusion of single sound sources in space. 

I don’t for one second dispute that a well preserved signal in a very transparent room can create magical illusions of being present at the original recording (or whatever the sound engineer created),  but the two imperatives are ‘well preserved signal AND very transparent room. 

I remember attending a show in Frankfurt where the sound from one particular room was stunning. Turned out to be quite mediocre gear playing is an acoustically-highly-treated to-achieve-neutrality room. 

The musical message can’t mask anything that fundamentally changes that message and therein lies the problem. The original signal and room effects aren’t 2 separate entities....the room always modulates the musical message....its the nature of sound waves and room boundaries.

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16 hours ago, gmgraves said:

You are laboring under a misconception, so save your effort. The reason why my brass doesn't sound like real brass has absolutely nothing to do with playback. It's because the sound of real brass cannot be captured. You seem also to be laboring under another misconception. You seem to think that microphones and the rest of the recording chain are perfect. Nothing could be further from the truth. I've used every kind of microphone for recording there is, condenser (both large and small capsule), dynamic and ribbon mikes as well as condenser variations such as RF and electret. I've used about every brand, Sony C37P, C500, ECM-22P; Neumann U-47, U-87, KM-184; AKG-414; RCA 44-BX, 77; Telefunken ELA-M-270, ELA-M-260, AK-47, C-12; Shure SM-33 (AKA The "Johnny Carson Microphone" - but being a Digger, you wouldn't know about that, would you?) etc., etc., etc. I mention all these mikes to illustrate to  you that I know wherefore I speak. NO microphone can capture the true sound of brass,this is especially true of trumpets and coronets, but french horns and trombones cannot be fully, accurately captured either in some registers. Enough is captured so we can identify the instrument's essential sonic signature, of course, but never enough to convince anybody, on playback (and I don't care how good your playback system is) that they are hearing real brass playing in a real space. Can't be done. end of that story.

 

Why bring it up? I don't have any experience with early Mark Levinson Power amps, but I do recall that the original Mark Levinson preamp sounded great for its day. I agree about early Krell amps. They sounded very dry (dry? how about arid!) and were at the same time very cold and very clinical sounding. OTOH, I have two amps at the moment; one is a Harman-Kardon HK990, which is my main amp, and the second is a Krell KAV-300i. Both are 150 WPC, and while both are excellent, I have to give the edge to the HK.  

I can confirm this, having been brought up on trombone music (My Mum was lead Trombonist with Ivy Benson's band for a while, I played trombone in local brass bands, no where near as good as my Mum though). Nothing beats the sonic impact of a good brass band live and no playback system I have heard can replicate that.

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9 minutes ago, marce said:

I can confirm this, having been brought up on trombone music (My Mum was lead Trombonist with Ivy Benson's band for a while, I played trombone in local brass bands, no where near as good as my Mum though). Nothing beats the sonic impact of a good brass band live and no playback system I have heard can replicate that.

I agree reproducing brass music well is hard. However, I suspect the problem lies more with the speakers than the microphones. Compared to speakers, even very good ones, microphones have practically zero distortion.

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9 hours ago, Blackmorec said:

Unfortunately this is really only wishful thinking I’m afraid.  In stereo replay, sound waves that reach the ears are already modified by the room. If the effects of the room are benign, then you’ll hear mostly original signal and its acoustics. But if the effects of the room are profound....room nodes and strong reflections, it modifies the sound waves that reach your ears, changing frequency amplitude and phase, which of course impedes your brain’s ability to construct the illusion of single sound sources in space. 

I don’t for one second dispute that a well preserved signal in a very transparent room can create magical illusions of being present at the original recording (or whatever the sound engineer created),  but the two imperatives are ‘well preserved signal AND very transparent room. 

I remember attending a show in Frankfurt where the sound from one particular room was stunning. Turned out to be quite mediocre gear playing is an acoustically-highly-treated to-achieve-neutrality room. 

The musical message can’t mask anything that fundamentally changes that message and therein lies the problem. The original signal and room effects aren’t 2 separate entities....the room always modulates the musical message....its the nature of sound waves and room boundaries.

 

Luckily, no wishing required! ... Just hard work ... :)

 

The mistake you make in your reply is claiming that the brain can't reconstruct the illusion, because reflections modify the sound waves. This is wrong, the brain is capable of extracting what it needs from the incoming signal, even when 'confusing' details are also present. That experience in the show demonstrated how capable the brain is, when the sound waves are 'carefully groomed' so that contradictory sound information is minimised; what I work on is to ensure that the sound waves when leaving the speaker driver surfaces have the least amount of confusing detail present, and only the room interactions have to be dealt with, by my brain.

 

This is well known, the https://en.wikipedia.org/wiki/Cocktail_party_effect ; normally only studied in non-musical settings - but it also works for audio, as a few have found out. The biggest problem is that the ear/brain is very fussy, and requires a very high standard of SQ to do its filtering job - higher than most people try for.

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8 hours ago, mansr said:

I agree reproducing brass music well is hard. However, I suspect the problem lies more with the speakers than the microphones. Compared to speakers, even very good ones, microphones have practically zero distortion.

 

Local library has a set of Oz Naval band recordings, of the usual show stoppers, etc. Recorded in a minimalist way, nice big acoustic, these are fabulous to listen to; tremendous bite and drive, an extravaganza of sound :).

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15 hours ago, anwaypasible said:

a laser microphone isn't a transducer.

Yes it is. By definition, any device that changes sound (or any other vibration) into an electrical waveform that is analogous to that sonic waveform or vibration, is a transducer. It doesn't matter what the methodology is: classic condenser, RF, Electret, ribbon, moving coil magnetic, laser, carbon, piezoelectric, all are transducers. 

15 hours ago, anwaypasible said:

any microphone that records audio at a higher distortion percentage than the microphone's limit is capturing the signal perfect.

I'm sorry. That sentence makes no sense to me. How can a microphone that captures sound that is more distorted than that microphone's limit produce a perfect signal? 

15 hours ago, anwaypasible said:

anyways, i've got the way nuclear speakers work on my facebook page - the reverse holds true to build a nuclear microphone & the only reason nuclear speakers are built is because they reach the mecca for more than one room size (speakers are limited to one specific room size).

Again, I don't know what you are talking about. I've never heard of nuclear speakers or microphones. So I can't comment.

15 hours ago, anwaypasible said:

seems odd that you put that much weight towards a situation that can backfire.

such as a microphone element that is magnetically biased to exaggerate all movement (or not) & is held at a gravitational freedom of static bias - because when inertia can meet neutrality, you claimed it isn't possible; but all that really tells me is the word phenomenon holds true to it's typical awe struck glare people give when they experience a phenomenon.

Are you writing English? I read English words here, but English thoughts don't emerge from from those sentences. In short, to me the above makes no sense. 

15 hours ago, anwaypasible said:

 

i might as well go on to say, why did they ever bother to create microphones & speakers if 'no transducer will sound exactly like the source' .. right? because then the audio we hear out of the speaker doesn't resemble anything like what was supposed to be recorded.

You're being way too literal here! Just because we cannot either capture or playback music exactly as it was performed, doesn't mean that what we do capture is gibberish!

I mean, be reasonable. Music played on hold on the telephone is very limited in frequency range and is very distorted, but you can still recognize it as music! The point many of here are making is that microphones aren't perfect transducers and neither are any speakers. No transducer is perfect. It's a physical impossibility. 

15 hours ago, anwaypasible said:

i guess by your way of thinking we all enjoy listening to jibberish noise from the speaker cones.

This doesn't even warrant a comment. 

George

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1 hour ago, fas42 said:

 

Luckily, no wishing required! ... Just hard work ... :)

 

The mistake you make in your reply is claiming that the brain can't reconstruct the illusion, because reflections modify the sound waves. This is wrong, the brain is capable of extracting what it needs from the incoming signal, even when 'confusing' details are also present. That experience in the show demonstrated how capable the brain is, when the sound waves are 'carefully groomed' so that contradictory sound information is minimised; what I work on is to ensure that the sound waves when leaving the speaker driver surfaces have the least amount of confusing detail present, and only the room interactions have to be dealt with, by my brain.

 

This is well known, the https://en.wikipedia.org/wiki/Cocktail_party_effect ; normally only studied in non-musical settings - but it also works for audio, as a few have found out. The biggest problem is that the ear/brain is very fussy, and requires a very high standard of SQ to do its filtering job - higher than most people try for.

So what you’re saying is that if the sound quality is high enough, there is still enough ;’sensory’ information remaining for the brain to create the illusion of single sources in space, in spite of distortions caused by the room? 

I guess I wouldn’t argue with that, as long as we agree that what we are hearing isn’t pristine, high quality sound, rather its sound with a lot of additional room generated artefacts exactly like following a conversation at a cocktail party. 

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6 minutes ago, Blackmorec said:

So what you’re saying is that if the sound quality is high enough, there is still enough ;’sensory’ information remaining for the brain to create the illusion of single sources in space, in spite of distortions caused by the room? 

I guess I wouldn’t argue with that, as long as we agree that what we are hearing isn’t pristine, high quality sound, rather its sound with a lot of additional room generated artefacts exactly like following a conversation at a cocktail party. 

 

Yes, precisely. The room will add its signature, of course, but the brain discards that as being irrelevant; the acoustic of the musical reproduction has its own integrity, and is the "easiest" to follow - completely unconsciously; no need for an active focus to achieve this.

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10 hours ago, mansr said:

I agree reproducing brass music well is hard. However, I suspect the problem lies more with the speakers than the microphones. Compared to speakers, even very good ones, microphones have practically zero distortion.

It's not the distortion. It's a combination of things, such as the mass of the diaphragm which is a problem in any transducer not just microphones. A perfect transducer would be massless (among other things), the ultrasonic frequency response, the dynamic range of the microphone capsule, etc.  For instance, you need a big diaphragm capsule for good bass response but a small one for high-frequency extension and a free-air resonance that's beyond the audible range, and super fast transient response. (I've often wondered why somebody hasn't marketed microphones designed like speakers with multiple capsules, like a big one for good low frequency response and a very small one for the other required characteristics. Maybe it doesn't work for microphones like it does for speakers ...).

George

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8 minutes ago, gmgraves said:

It's not the distortion. It's a combination of things, such as the mass of the diaphragm which is a problem in any transducer not just microphones. A perfect transducer would be massless (among other things), the ultrasonic frequency response, the dynamic range of the microphone capsule, etc.  For instance, you need a big diaphragm capsule for good bass response but a small one for high-frequency extension and a free-air resonance that's beyond the audible range, and super fast transient response.

The result of those non-ideal properties is distortion.

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