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i'm the type of person to do the whole frequency response rather than only a small section.
i dont know if it is because of the techno and hiphop that i listen to.. or if it is because i would feel too silly to have one portion sounding a whole lot better than a different portion of the frequency response, ALL FOR THE SAME SONG.

but then again, i would probably prefer a full and flat frequency response over a higher fidelity midrange any day.
once i heard the full sound of a rather full frequency response, i just dont want to go back.
my system was playing from 20hz to 16khz as flat as i could get it with a calibrated microphone.
there was more audio information there than a 2-way set that plays down to 40hz
and i know there are quite...

anwaypasible

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i suppose some of us here do, and i am feeling free to talk about it.
for starters, the highest quality speaker wont sound as good as it could without help.

only three things need to be considered to make a leap.
1. calibrate the time alignment for the distance between each speaker to the listening position.
2. get your wild wall reflections tame.
3. seperating the impulse from the reflections is going to allow you to hear clean air when the audio dies out quickly like it should.

number 1 can help anybody with any speaker.
sometimes number 2 isnt easy to do, but people need to do it.. doesnt matter if there are small speakers or big ones.
if the soundwave can make it to the wall, that reflection can either be used or an attempt to attenuate it with software is an option.
this leads me to number 3 .. sometimes the space is big enough to cause no problem to the listener.
if there are noises and artifacts by the wall or in a corner, it doesnt matter as long as you are far enough away to not hear it.
but to be truthful, soundwaves are pressure.. and sometimes the room can build up with pressure without actually hearing any echoes or reflections.
it is simply wind resistence when the air goes past your ears, hits the wall, and bounces back.
you might have diffusors up, and that might manipulate the phase.. but it doesnt mean all of the waves havent been transformed into a 3hz soundwave (or even an absolutely solid 'gust' of wind).
if you want to overcome that backpressure, you would turn up the volume.. and that would increase the backpressure, as two wrongs dont make it right.
backpressure in the form of wind (its air colliding with the air coming from the speakers) has the same effect as putting your finger on the speaker cone while it is trying to play audio.
the effects arent as strong, but they exist.
the situation might not always be a perfect bounce back to the speakers, and that means sometimes yes and sometimes no .. creating listening fatigue.

since air is a gas, and gases in a room are seen as fields.. you can realize this yourself by listening to the change in the sound when the room is in the 80-90 degree F temperature area , and then again in the 60-70 degree F temperature area.
the heat means the air atoms (and other molecules) are already bouncing around more rapidly
as the air cools down, the atoms are slower and move when asked by the soundwave more easier.
the difference is like trying to hammer a nail into wood, or trying to hammer a nail into a piece of metal.
heat can act like diffusion when the atoms moving rapidly attack and murder the soundwaves.. but they are also very stubborn to get in uniform and form the soundwave (which is why the soundwave can be destroyed by the air, to a small extent)

sound panels can sometimes alter the angle of the wall reflection.
if the sound panel is really aborbing all of the soundwaves that touch it, then the new reflection would be either in front of it or behind it.
speaker cones are round.. and that means you should expect the soundwave to travel in the exact opposite direction.
if you had a spare speaker cone, you would hold the cone up to the speaker and trace each angle that way.
speakers are dipole, meaning when the top of the speaker pushes out and down (because it is bent down) .. the bottom of the speaker will push out and up (because the bottom of the cone is bent upwards)
they smash into eachother, creating a 'polarity' response.
normally these would be perfectly circle, but cones bend and flex .. creating the unique shape.
the information above has a conflict.. one states the soundwave is like half of a globe and goes in all directions.
the other statement says the cone presses the soundwave into a circle.
well both are true, because as the compressed circle touches the air in front of the speaker (a few inches away) the air will react to the movement in a uniform manner.. much like watching a bullet break the sound barrier (or an airplane or jet)
the circles arent rings, they are solid like a plate or dish.. but the dust cap being in opposite shape will push the soundwave in an opposite direction of the compressed circle.. certifying the 'polarity' shape of the output.
at lower speaker cone travel.. the soundwave will be that flat plate, but with protrusions in the middle.
as the speaker cone travels more, the flat plate can break into a ring.
depending on the air and the speaker, i dont see why the opposite cant also be said (stating that i might have one backwards, so i might as well say 'em both)

you need the speaker cone and the dustcap to get a visualization of the soundwave leaving the speaker.
the cone compresses the soundwave into a flat dish, and the dustcap pushes the middle outwards.
inverted dust caps wont push the middle of the soundwave outwards anymore, instead the soundwave will be a mirror to the cone diameter and angle of the cone.
some cones are flatter than others.. and some speakers have a plate to keep them totally flat.
the size of the dustcap and how much the dustcap sticks outwards will affect the soundwave emitted from the speaker.
if the ratios are correct, the output can form a cone with a point (the point facing away from the speaker)
so that is why you get a speaker cone and dust cap, then flip it around to opposite of the speaker cone.
you picture that shape, and realize that the air will touch that shape and go in that direction.
how many times you measure the soundwave's directivity is dependent on the user.
you could do every square inch (center the cube with a string and some tape)
or
you could do every square millimeter.
that is why a single sound absorption panel on the wall can be foolish.
but
sometimes electricity is emitted from the coil, and if that energy is directional.. it can tend to creating a void in the area of soundwave output.
as the electricity 'electrifies' the air, it can cause the atoms to either rush into that space or run away from it.
again, this can manipulate the shape of the polarity response.. and it is usually dependent on how much power is going into the coil.

you might put a sound panel on the floor in front of the speaker, but soundwaves are ripples.. and you might have blocked that ripple, but the next ripple is sure to touch the ground again.
the only way to avoid it is to have all of the walls (including the floor and ceiling) to be covered in sound panels.
with a hole cutout for the speaker to be shown to the air (usually an in-wall speaker setup)

since covering the entire room with sound panels isnt practical for most people, it puts major emphasis on the digital sound processors to recreate the sound of a room full of sound panels.
not everybody is going to reduce their reflections to 0ms
and that is totally okay.. because the distance between the click from the speakers and the click heard from the walls is the important part.
lots of treble is attenuated because of the impulse response, leading me to think the manufacturers secretly apply DSP technology to add enough negative attack to the treble to keep it from being attenuated.

if you know how attack and sustain works, negative attack will soften the sound when the room is refusing to shut up.
it makes it possible for a soundwave to exit the speaker and travel a distance of only a few feet before the DSP throws out an accelerated opposing phase at twice the speed (or faster, as this calibrates the distance allowed) to cancel out the soundwaves already in the air.

but..
soundwaves are fast!
so the likelyhood of you throwing out an opposing soundwave to cancel out the first one before you hear it is hard on the equipment.
processing needs to be fast, the amplifier needs to be fast, and the speaker needs to be fast.
again, the speaker would need to have a quality coil to play two seperate 'audio tracks' at once.
while it is saying something of the present, it has to send out an opposing phase of the past.
this means dual work, but also dual phase.

sound panels need to be a thing of the past, as people dont want sound panels all over their room.
some people despise installing carpet as it gets dirty over time and isnt a sanitary solution.
it would be absolutely no different with sound panels.
as rubber and latex, and most materials used, would capture and hold bacteria.
they are also not stain resistant.
you cant spill a beverage on a sound panel like you can a tile floor.
above all of that, walking on the sound panels would probably (eventually) degrade the sound panels.

these digital sound processors need to make their way to home theater receivers all over the board, so that everybody can appreciate the entertainment of the movie and audio industry.
until these 'negative attack' processors are easily available, people will have to program their own and sell 'em to customers.

attack and sustain are not the same thing that the room does.
there is a whole lot of sustain, with coincidental collisions of opposite phase causing some silence for that soundwave (but not all of 'em)
and the same is true for attack, there isnt any really .. but when two soundwaves of the same phase collide, the re-energize eachother and whatever is the result will be a re-energized boost of nonsense.
this is why people put subwoofers in a corner facing the wall.. because it creates sustain and slurring moments of attack.

i'm not a genius persay, but i do believe it is possible that the math equation used to create the negative attack could affect the purity of the sound.
a simple delete is not the same as an attenuation.
all softare equalizers are the quality of the math used to boost or cut a frequency.
most times common/casual people will produce a result of attenuation that changes the purity of the audio.
and once the changes of 'cleanliness' are in order, then there are still phase differences.

gotta stop messing around with attenuation and start mastering the cancellation of soundwaves using opposite soundwaves.
if the room had no reflections, you wouldnt need it.
but rooms do have walls that bounce the soundwaves.. and those reflections will ruin the attempt to cancel out the soundwave with an opposing soundwave.
mainly because the soundwaves in the room are not the exact same as the opposing soundwave ment to cancel out the original soundwave sent out.
there are attenuation losses because of age.. there are attenuation changes because of colliding soundwaves (some serious cuts, and some serious boosts of nonsense)
that is why it is best to know the room you are dealing with and emphasize the frequencies that the room emphasizes.
if you remove all of the peaks in the room.. does that introduce a release on the hold of the natural cuts?
you'll have to put emphasis on the boosts, then have a look at what is left.
the chip would be programmed custom for the room, or would be pre-programmed for each height/width/length
the next stage would be the ability to record and analyse the recording.
the next step from that would be selling your product to a home theater receiver manufacturer.

but be warned.. you are essentially programming reverb.
when you input the length from speaker to first reflection, that gives a variable.
and when you input the distance from the first reflection to the second reflection.. you have simply given the reverb your exact room width and length measurements IF the speaker is against the wall and pointed towards the corner.
turn it away from the corner and things do get more difficult (as the distances are the most simple form of algebra and are hard to argue with)
but, this doesnt say anything about the height of the ceiling.. and it is just as important as the distance from speaker to first reflection.
it is easier to simply program the DSP for each square foot of space.
then give a toggle switch for rectangled rooms (with the option of longways or the thin part of the rectangle).
this isnt perfect though, because tall ceilings wont sound the same as short ceilings.
its simply a matter of x,y,z to some.
others will tell you how ugly things can sound when the speaker is too close to a side wall.. and that those reflections need to be tamed.
square footage as a preset will keep some improvement, but they will also not decrease the impulse response as low as they could.
the increase in impulse response will be for every speaker layout option available that doesnt meet the current layout.
putting a speaker in the middle of a wall and asking for square foot optimization is alright.
putting the speaker right next to a side wall will require a custom negative attack.
as you are trying to cancel out soundwaves, you have to know what to expect.. and once you do that, your impulse response times will severely drop.

but dont get me wrong, it is still worth it to program a DSP to send out the opposite phase as fast as possible.
even if that means connecting two soundcards and forcing them to play in opposite phase.
you get a wire adaptor to connect both soundcards to the receiver.
realize that each soundcard playing opposites will be double the power, and when two opposite soundwaves are forced to be together.. it is a direct constant voltage.
some receiver inputs might be more tolerant to this than others.
but then you get yourself some time delay and start one soundcard with 1ms difference and listen to the sound start to come out of the speakers (if your equipment hasnt died yet from receiving DC voltage)

this is a rich man's way to something beautiful.. but could prove annoying when preamp inputs are fussy.
maybe some receivers have a protection circuit to immediately shutoff the receiver when DC voltage is 'sensed'
and you would have to increase the delay for one of the soundcards until the protection stopped tripping.

i'd do it myself.. but i dont have two working soundcards.
plus, i dont know of any program/software that allows for the phase to be flipped 180 degrees.
i also dont know of a program/software that would allow for two sets of front and left to be delayed seperately.
you could use the plugin to copy the front speakers and add delay to the rear speakers, then use the rear speaker outputs of the soundcard with the front outputs of the second soundcard.
asking the program to play to two different soundcards at once would also prove to be difficult.
the processing time to send data to the second card might already be like 10ms or more.
but theoretically, if your soundcard can ASIO with 8ms .. both soundcards would be 8ms .. meaning both are the same.
getting a program like 'virtual audio cable' to output the audio to the second soundcard instantly might be hard.
asking the chipset on the motherboard to send data to both soundcards equally might add some time too.
getting it setup might prove to be whatever milliseconds of delay between the soundcards.
but say if it is 10ms of delay between the soundcards.. if your room's impulse response was in the 20ms range .. you would still drop the impulse response time to half with 10ms of delay being removed from the opposite phased soundwave.

could be a whole lot of work.. might be less than a week for some.
the result is worth it if the receiver/amplifier doesnt break from the extra DC current.
and the only problem i could forsee would be when two analog waves come into contact with eachother.
that could produce anomolies, degrading the purity of the audio.
this can be alleviated with a solid wave in the first place.. if the wave is solid, then it is stubborn to change.
it would be more of a sonogram type view than an analog wave view.
as i believe it is already true that some analog waves already collide at intersections.
there should be a term for this when it happens on a single left or right channel.. i just dont know what that term is.
and it would amount to how the amplifier circuits of the soundcard 'talk' to eachother when they are both connected.
i believe if the output resistance is high, then the signal wouldnt allow for any communication between the two soundcards.
because once the wave is in the preamp cord, it is on its way to the amplifier without anything else to worry about.
it is when the electricity tries to creep inside the second soundcard instead of going down the preamp cord.

perhaps ugly and long to make the room sound like it is full of carpet and sound panels.
but you will hear more when the impulse response time is reduced to drastic numbers like half.

lots of reverb simply lack the amount of negative attack to eliminate the echoes.
all of the controls are there, minus ceiling height and distance to the floor.
as the reverb only deals with the horizontal plane, and should deal with the horizontal and verticle plane.
 

ien2222

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My response would be that high end audio is simply accurately reproducing the recording (with all of flaws of said recording) in such a way that the equipment gets out of the way and doesn't add anything to it. For high quality recordings, it means that you can close your eyes and you wouldn't be able to tell the difference between the recording, or the live performance. As such, it's all about the ear.

If you mean in relation to the posts on this sub-form of high end audio, I'd say 99+% of the threads don't belong here.
 

MEgamer

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well there will always be a didffernece, again i second that its all about teh ear... cos speakers will never be gd as the real shizz :) for example in a live performace (say classical) there are many sources ( each individual instruments), where as for speakers u are just hearing 2 points sources, a high quality (the best) will be able to render any material in its most realistic way, however it will still not sound thhe 'same'.

rendering realistically and sounding the same are 2 very very VERY differnt things.
 

MEgamer

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corresponding factors that relates to the factors of a high end car, high end PC, high end laptop, high end ram stick, high end CPU, high end soundcard... and so on.
 

MEgamer

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since air is a gas, and gases in a room are seen as fields.. you can realize this yourself by listening to the change in the sound when the room is in the 80-90 degree F temperature area , and then again in the 60-70 degree F temperature area.
the heat means the air atoms (and other molecules) are already bouncing around more rapidly
as the air cools down, the atoms are slower and move when asked by the soundwave more easier.
the difference is like trying to hammer a nail into wood, or trying to hammer a nail into a piece of metal.
heat can act like diffusion when the atoms moving rapidly attack and murder the soundwaves.. but they are also very stubborn to get in uniform and form the soundwave (which is why the soundwave can be destroyed by the air, to a small extent)

"heat can act like diffusion when the atoms moving rapidly attack and murder the soundwaves.. "

well not really it will only help the sound to mvoe faster, even though it is less dense, ther molevules are moving faster, so energy tranfer is quicker.

diffusion of sound in air is pretty much minimal, even wind.
 

anwaypasible

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i'm loving this post.
except that you wont have to close your eyeballs to get the feelig of realness.
you might have to shut your brain down a notch, because your brain 'knows' you arent really there.. but after that, it is as good as hearing it with your own two ears.

bringing me to megamers post..
no matter how many sound sources in an orchestra, your body only has two ears.
all sounds should be picked up by two microphones and those two microphones would take the place of your ears.. making it as realistic as if you were standing there yourself.
video certainly helps put the person there.. as i am stubborn and require video to get me the satisfaction of realism.
otherwise, i need quite a long time of listening before my brain settles down and lets in the fantasy.
 

anwaypasible

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yes, that is what i was saying about cold air.. the density can be confusing, and the person needs to remember that energy needs to transfer.
so to say what allows the energy to transfer faster (and what allows the energy to transfer slower) can be worth a temperature change in the room.

i wouldnt argue that any diffusion of sound in air is pretty minimal.. but i also wouldnt second-guess the power of the air pressure making the speakers attempts at pouring out soundwaves more dull.
if you think about it, midranges dont always move a whole lot... sometimes less than a millimeter.
more amplifier power can help, as well as more force from the magnet.
but there are speakers that play sounds as light as a feather, and keeping the pressure inside of the room like a sealed box is only going to cause some adverse affects to those speakers (and even distant soundwaves, as they lose some energy when they travel).
this example can be easily seen with a car subwoofer (and maybe a home theater system with enough subwoofer power).
when the windows are up, the pressure is high and the notes can be felt tighter.
but
if you open a window, the bass gets louder and the feeling of the notes grow loose.. even though, sometimes you can clearly tell the subwoofer has more freedom to jump from lows to highs.
i think the example of the freedom given should be enough to make taming some wild pressure on the list of 'to do' for a listening room.
that might mean getting speakers that are capable of performing in a pressurized space.
or
that might mean releasing the pressure to see what your speakers can do with the extra freedom.
if there is an air vent, this might prove to need some tweaking to offer a perfect blend of pressure that is not too much or too little.. making the speakers sound their best.
i wouldnt say it is absolutely necessary for the bouncing reflections to be calibrated for midranges and tweeters only.. because those speakers usually dont provide enough pressurizing of the room to begin with.
but how hot/cold the air is will make the soundwaves transfer to the air faster or slower, and it could help some 2-way owners.

as i live in the midwest, i can open a window just about any time of the year except the hot days of summer, to hear an improvement in the energy transfer.
hot stale air in the room isnt really friendly.
it is fatiguing to the person, and fatiguing to the audio.

you could calibrate your equalizer for hot or cold weather, then see the calibration get ruined because the air changed in the room.
like going from warm and stuffy to opening a window, i always hear an increase in the treble.
i've noticed that the reverb software that comes with the creative x-fi soundcards give the option to adjust for the different air temperatures.
if a person cant afford an air conditioner, maybe they can open the window at night and let the room fill up with cold air.. then close the windows at sunrise (or shortly after) and seal up the cold air for the remainder of the day.
i did this last year in the hot summer weather... but i didnt like how hot my computer was getting, and the quality of the air was still fatiguing.
i've spent the last 3-4 years in a house of some sort with 80+ degree temperature.. i dont necessarily need to go down to 72, but 74 would help.
there were times when my digital thermometer was saying 90's inside with windows blocking the sunlight.
tried a dehumidifier and that brought the temperature up into the 80's from the 70's .. brought it back and got an air conditioner.
the air conditioner doesnt remove the water half as much as the dehumidifier did.. and i would probably want both to try and achieve a really really comfortable living space.
but
i cant afford to get too selfish at the moment.

i tried to find the difference in weight between air and water vapor.
i think water vapor is more of a heavier solid than air, at least to say it is stubborn in terms of soundwave transfer.
and that water in the air isnt helping you breathe air, i'd want to get rid of it.
i've had nose bleeds in the past when it was dry.. but i wonder if it was really dry or if the water was boiled from the furnace.
seems like when my nose would bleed, the blood was like an envigorating splash of wet in a coincidentally 'dry' environment.

i wonder if it is the plug and play devices, or the lack of information as to how to calibrate something correctly, that is making a way with motivation of people.
 

musical marv

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High end audio is similar what you mentioned it is closest to the real thing real music.It involves expensive equipment mostly and a decent sound room will help.The most important aspect of the system is the preamp which controls the main system of amplifier, turntable or cd and speakers.
 

ien2222

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I more so stated the goal, not necessarily something that was achievable in it's current state. But if you ever get a chance to hear an extremely good setup, you would swear that it's possible. In some cases you actually have to make it a point to critique the sound to notice the differences because it's so close.

Hmm...on the other hand, it does suck to know exactly what you are missing because you can't afford it.
 

anwaypasible

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i can imagine 'test subjects' who have been given the privilege to listen to such extremely good setups.
and based on their response, the room would be altered to try and manipulate the sound characteristic to what was requested.
that might mean re-organizing things on a counter (if recorded in a kitchen or dining room) to try and re-shape the diffusion/reflections on the sides.
 

MEgamer

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No matter how gd it is... you are listening to 2 point sources... EVERY instrument in....say a classical ochestra -has an DIFFERNET "almost omnidirectional" nature of its own... i never said that the sound was different,... sure there are probably speakers out there that can play almost exactly the way as it should sound... but is that sound travelling out fomr the speaker towards u in the same manner as a REAL INSTRUMENT??? no of course not. and that is the main differnence.

This difference can be even more noticable at louder volumes. you can try getting rid of room modes by playing a source outdoors... but heck its a speaker its a point source or a line souce(planar speakers) at its best.

plus all spekaers have boundaries(edge of soudstage in this case).. u can only say there isnt one, when u are in an illusion (a very very very gd setup needed.)
 

anwaypasible

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you still only have two ears.
if the microphone isnt quality enough to capture a 180 degree path for its side of the head.. other microphones would be needed to capture the full atmosphere.
reason being why i said something about distance and using some mics far away to help capture the sense of distance.

seems like this could easily be said to be a matter of angles, meaning a lack of omnidirectional functionality.
if your mic can only capture 90 degrees at the tip, you would need two mics for each ear to re-create the 180 degrees.
and even then.. our right ear can still hear sounds from the left side.
but as long as the speakers are filling in the voids of the airspace, it would then be the recording process causing a foul.
i dont think many people care to test their microphones ability to capture omnidirectional sound.. and i also dont think many of them test the distance that can be recorded.
anybody can put a parabolic dish on a microphone to help record distant sounds.. but the nature of the dish's shape will reduce the near field, causing a 'one or the other' situation of choice.
hard to place two microphones in the same location without capturing some reflections off of the other mic.
might be sound if the layout was really bad, but it could bend/break the breeze in the air too.

recording techniques arent too difficult, and i dont think it requires college courses if you keep an open mind to your layout .. and know your options to capture what it is you are trying to reproduce.
but
i think eventually we need to see events where all of the recording layouts are generally the same, with no real attempts being made to zoom in on anything more than omnidirectional sound and distance of sound.
the camera's perspective is the final say, and hearing two people whisper @ 80dB from 20ft away is childish and disrespectful to 'natural brain function'
people might start to really develop some kind of mental or emotional disorder, based on the listening experience they have grown accustom to.
might seem far-fetched to some, but people build an attitude and different views on life every day.
maybe it is a lack of appetite, or maybe it is how the person feels they have been mistreated.
i think all of us can be upset about the massive amount of dynamic range used in movies nowadays.
jumping from 60dB to 90dB is stupid.
it will come to a point where people will stop watching action movies because they are tired of dealing with the volume knob.
nobody wants to witness audio too low to hear, and putting a limiter on things much louder isnt going to kill the experience.
we dont enjoy standing 30-40ft from an explosion every time we watch an adrenaline rushing movie.
they want the camera to be close to capture the money they spend on the explosion, and that leads to us having to hear the extra amplitude - because the camera is close.

nobody in their right mind needs to re-create a bomb going off inside their living space.
it isnt practical, and it isnt humorous to anybody.
there are only so many times you will play back a loud gunshot in a movie (or some actor/actress screaming for help) and you giggle because if anybody heard it, they might think the actions on the screen really happened in the living space.

we live in an era of technological revolution, people have had their chance to listen to silence and then *BOOM* all of a sudden.
the only jumping we do is to grab for the volume knob and get upset.
lots of science fiction, or other exhilarating chances to show the human brains power to process ideas is needed to stay ahead.
since the 1990's there has been action and violence and horror.
been seeing constant movies in these genres with much dissapointment.
mysteries are hard to create that will make the viewer feel as if they have seen an original film.
the economy is selling all of these toys, and nobody is going to want to continue playing with them.
i believe that is called an uprising, as it could lead to lots of extra people out wandering the streets.
i can certainly testify that the broadcasting companies arent doing their part to keep us entertained.
cable companies should have their equipment on a high priority security watch, as people might try to destroy some hardware to get back at all of the lost money they spent on monthly fees.

science fiction and standup comedy are the two biggest voids in the entertainment industry right now.
havent seen a lot of mysteries myself, but then again.. i havent been impressed by them in the past.
i would expect to see a lot more family films being made, as these are extremely easy to create since they are usually teaching the basics of being responsible.
could take that one scenario and create lots of different storylines, as we have all learned from pixars toy story and wall-e .. the lessons learned are only as good as the storyline/plot/theme/etc.
filled with a dozen 'one-liners' to make the adults laugh and the whole family is happy.
i know it is kinda sad to be in this technological revolution and these science fiction people are hiding their ideas because they dont want anybody to cast away on the principle making money.

i think there is a real fear in the movie making business, as we dont want to give all of it away .. to preserve some 'freshness' to fill voids.
but as far as i can tell, we only live one life.. and if they poured out everything for each 'lifetime' - people would die and their children would be watching the re-makes.
besides, i have watched some re-makes myself.
i dont care to do it every 5 years.. but give it some time and i'll thoroughly scan the plot for change, even though the entire principle is the same.
new actors and actresses help a marginal percentage.
look at all of the action films with jet li and jackie chan and whoever else was in the mix.
i couldnt look myself in the mirror proudly if i went out and rented each one of those movies.
the matrix movie treated me right.. and i would hope to see more movies drawn out over the course of 3-4 films more often.
episodes are nothing new to us consumers, as we have watched many 'broadcast movies series' in the past.
even when a sitcom says at the end of the program 'to be continued' we seem to get excited about the next episode.
a lot of fun for us to sit back and wait for the next sequel to come out.
no fun at all when we realize there isnt anything else worth recording with the DVR.

the economy itself needs to get it together.
i am thinking they have run their course of 'we bought a robot factory to build things.. NOW BUY 'EM'
get it home and it breaks because we arent worthy to upgrade the furniture with a glaze of protective coating.
as an economy, we expect things to simply 'fall on our lap' when we are paying, or otherwise involved, in the subcription.
 

ien2222

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Anywaypasible sort of covered that already. We only have two ears, we pinpoint direction by the difference in time that the sound waves hit each one. We neither have multiple ears, nor can we 'sense' the sound waves throughout the room around us. Given these limitations in our hearing, you can realistically recreate a forward facing soundstage with 2 points, assuming it is planar(no height), in a given listening area.

I had the opportunity to listen to a Genesis setup. When listening to orchestral music, you actually could point in specific directions as to where the instruments were being played. It was so precise you could even tell the difference between 1st, 2nd, and 3rd violin despite the fact that there is some overlap between them in a standard arrangement.

The problem for most people usually comes in the form of not having a large enough room. This listening room was around 25-30' by 35-40' and roughly 10' high ceiling. Granted, this is more the ideal listening room, but for some reason you do have people who think they can put a good $10+k setup in a 15'X15'x7' room full of furniture and expect it to sound incredible. That just won't happen.

The only problem with a 2 point source recreating a soundstage is that there is a 'echo' of sorts. There's 2 waves hitting both ears. But, the good thing is, is that we are presented with echos everywhere, off walls, objects, floor, ceiling, ground, etc. We are already conditioned to disregard certain echos unless you are specifically trained to listen for them, such as being deaf and using them for echo location for example. So for most people, it isn't a problem, especially if you are just causally listening to it.
 

stillerfan15

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Wow Marv, what have you started here? :D
I agree that the goal of a true high end audio system is to recreate the source at the highest level to sound like live music. The problem is that recorded music is so processed that even if you have the best source LPs or CDs on the best source equipment and the best preamp and power amp and the best speakers set up perfectly in your listening room you still have to deal with the problems at the recording level. That said, I would say that the best way to assemble a high end audio system is to start with source (cds and lps) components and work your way downstream. I know I'm sounding like a Linn Soundeck TT add.
Dave
 

musical marv

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The closest sound I heard to real music in high end was my friend's system which costs with no exaggeration over $150,000.Mostly all Mark Levinson equipment and custom made speakers which alone the bass woofers weighed over 400 pounds each. When you heard a vocal sound it sounded like the singer was in the room really.fantastic sound.The system was tri amped completely and it was a dedicated sound room.
 

musical marv

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Source is essential especially a good table like Linn or VPI.
 

MEgamer

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u sure ??? im pretty sure ppl have a hard time hearing each individual insturment in the REAL THING!!!. i mena ya... i can hear it.. but pinpoint is another thing...
 

ien2222

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I said 1st, 2nd, and 3 violin, those are sections, ranging from 3-7 players each. And yes, depending on the music, you can hear individual instruments, like solos, and pinpoint them.
 

stillerfan15

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Not recommending a $2000 TT or CD player with $200 worth of electronics and speakers - you have to have some balance. But if you don't get the signal right at the beginning you can't fix it later. A digital source seems much less dependent on cost than vinyl or tape. Thanks for the thread,
Dave
 

musical marv

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You are talking about pin point imaging which can happen with Maggies, ML and perhaps Wilson speakers and some dynamic speakers.But this is still an illusion.
 

anwaypasible

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it does seem a little strange that microphones arent always the best to be recording in the first place.
but to say that the microphone is good and it was recorded with analog, eventually that analog will be transfered to digital.
if there is no transfer to digital, the audio wont be mass produced to all of the consumers.
this would mean you are stuck with the few select friends in a group.. but that doesnt mean you cant send an analog tape to a professional music 'reviewer'
but
it does mean the person will inevitably tell you that the transfer to digital will have to happen for millions of people to enjoy it.
(and they might say thousands of people wont be able to enjoy it)

i dont remember ever hearing a consumer-level tape sounding any better than cd.
i didnt have the speakers for it, and i dont recall any increase in anything such as noise floor.
i think the dynamic range with cd's are higher than those tapes.
the next question would be slew rate.
somebody said the slew rate is how much of the audio is output from the speakers based on how fast and sensitive the playback device is (if the data is there in the first place)

doesnt seem like much of an upgrade when you can simply increase the sample rate to get a higher slew rate.
the data would be there, but would the digital to analog convertor and amplifier circuit play back the extra audio information?

a whole lot of emphasis was placed on the noise floor and dynamic range.
not much was said about the slew rate and transients.

an example..
my creative x-fi elite pro (now dead waiting for some repair info) has a low noise floor.. but the realtek HD audio soundcard integrated onto my motherboard has a faster slew rate.
yes, the higher noise floor is annoying.. but the faster slew rate helps pour out more audio.
since the noise floor is too high, transients are less pleasing and harder to hear.

it seems like a compromise of noise floor to more audio data.
and i think the lower noise floor is less annoying, but knowing that there is some audio data i still dont hear is also annoying.
clean with missing audio
or
dirty with more audio
seems like the choice is already made for us with the lower noise floor being the winner.

i miss the silence, but i feel left out.
 

anwaypasible

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awww c'mon.. time isnt an illusion.
all of the ability to hear direction isnt the timing, as amplitude is also a function.
getting the timing out of the speaker is one thing.
getting each speaker to line up with eachother to present the timing and amplitude is what most people fail to do by not using the time alignment and other room optimization digital sound processor functions.

gotta have speakers capable of the time.
gotta have the wild soundwaves in the room tame and organized.
if either one is not met, the chance of pinpointing things with high accuracy is non-existant.
that is why some of us laugh at people who buy really expensive speakers without doing the room optimization from software.
maybe some old timers who laugh at people using software room optimizers because they build a room that tries to do the same thing.
those construction workers might get close to some of the sound processing of the 1980's and early 1990's .. but not coming close to software digital sound processing that tailors to room optimizing.
sometimes they try to get the time alignment right.
sometimes they try to get the room reflections down.. could be sound absorbing panels or simpling grabbing those extra soundwaves and putting them into some kind of 'junk drawer'
but at the end of the day.. all rooms can be made to sound better with software digital sound processing, even those rooms that have been custom built for audio.
seems like the more those construction workers mess with the room.. the harder it is to find some audio software that will make the room sound better.
square and rectangled rooms are the average, and that is why those pieces of software are designed for them.

custom audio optimization software for the room can go a long long long way if the programmer knows what is what.
sometimes the optimizing software can be designed with the furniture, and if you move or replace any furniture .. the whole thing sounds different.
 
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