You can alter the width of a stereo signal and add ambience in analogue. I've no idea what the others you mention are - except they seem to be named to confuse what they do. ;-)
It usually sound worse. Much worse. Trying to make a silk purse out of a sow's ear never did work despite clever tricks. Better to start out with silk.
If you're interested, SRS labs (now called DTS) had early commercial success with those techniques. 3D provided extra wide stereo imaging which is now commonplace. It also had other tweaks such as one which allegedly makes the sound appear as if it was coming from the same height as your head.
TruBass worked well. It tricked your ears into hearing a bass note by playing only the upper harmonics (or something like that).
Point is that these are digital techniques and are almost impossible to replicate with analogue methods.
what is easy with digital is stuff using time delay. That was very very hard with 'just' analogue electronics which is why we had spring and plate reverbs, - essentially you need a mechanical device to delay sound waves.
I cant actually think of anything I did with analogue that couldn't be done easier* with digits, with the sole proviso that you needed a lot of computation per audio sample. Especially for that dreaded reverb.
And there are a few things - like frequency division and reverberators that are WAY easier in digits.
It was the realisation that a modern bog standard INTEL CPU and sound card had more than enough grunt to do what back then needed a very expensive DSP chip, that set me wondering how easy it would be to do an amp in digits.
Of course the likes of VOX already do this, so there is no commercial potential, but its fun to play...
Don't confuse that fact of digital compression with the fact of digital signal processing.
A full blown floating point processed 44KHz 16 bit (or higher) sampled sound is in fact what I assume (or even more) modern recording studios are using.
Everything is easier if you have the computing power.
And no introduced hum and noise once you are in digits.
*easier in terms of hardware: obviously you still have to write basic code for a tone control etc.
Most of them seem to revolve around trying to 'cure' the results of poor speakers and poor speaker placement. Exactly as you'd find on a TV. Place the speakers too close together and facing the wrong way? Good way to sell a sound bar. Which also has the speakers too close and of poor quality. So you try and fiddle it to sound better.
You can't change the basic laws of physics.
And pretty well anything that can be done in the digital domain as regards signal processing can be done in analogue. But would obviously cost more.
The only real killer feature of digital audio wise is the ability to do lossless recordings. And of course to do some forms of processing more cheaply.
Err, do you not know the difference between delay and reverberation?
Surely cost is key because nowadays consumer products can include signal processing (such as tv sound enhancement which you may or may not like) which would be unaffordable using analogue techniques.
I don't always like the results of modern sound enhancements but sometimes it works very well such as the audio on my Samsung telly. It's a relatively cheap set at ?350 with nothing more than slim down firing speakers but, with sound enhancements, it has surprisingly acceptable audio. Apart from the bass frequencies and maximum volume, my separate AV amp provides little improvement. If I'm not watching a film or a concert then I often turn the AV amp off (it has an audible fan) with little loss of audio enjoyment.
Perhaps I've just been vry lucky to get such a set. However it's audio certainly benefits from its onboard sound enhancements.
You don't like Bob carver?
he doesnt like all that goes before.
Hmm. Easily pleased. ;-)
I don't give a shit about Bob Carver. You included 108 lines of text in order to add one line of your own.
Oh I see. Sorry about that. I usually trim to context but the convention in this group seems to be to quote most of the post.
If it annoys you, most newsreaders have a command not to show quoted text.
There are idiots in this group. You don't have to copy them :-)
Floating point?
I do all my processing in 24-bit fixed point. Mind the gear I have, although better than domestic, isn't top end studio stuff.
Andy
I can't say I blame you for 'quitting' in this case. BTDTGT(Bloody)TS (& the Celebratory Mug) when I felt impelled to set him straight, a year or two back, on the basics of balanced line transmission circuits as used by the telephony industry and applied elsewhere in things like firewire and SATA interfaces to name but two examples from a much wider remit.
He appears to largely understand the physics of the real world but with the odd misconception thrown in here and there which, despite all reasoned arguments designed to dispel such misconceptions, he seems to prefer to ignore or side track rather than re-appraise. His attitude appears to be at odds with his chosen nom de plume.
As you pointed out, being wrong (of itself) is not a sin. Only when one is being *wilfully* wrong does it *then* become a sin. I'm afraid TNP can be rather sinful at times. :-( It's a pity really because he's generally right about most things technical most of the time and he *does* offer reasoned arguments against the railings of the eco-bollicks, anti-nuclear brigade, a character trait I wholly support. :-) There are plenty of others who post to this news group with far worse character flaws than getting stuck on an erroneous or irrelevant "technicality".
====snipped====
Nicely observed! :-)
I'm sure he does. I think the point he was making was that there was no compact electronic method to delay audio signals in the milliseconds range in order to produce reverb effects without resorting to mechanical means involving springs and/or large metal plates. Being able to electronically delay audio signals is key to the creation of a completely electronic reverberation (or echo) unit.
ISTR at the time, 30 odd years or so ago, that experimental electronic delay lines using bucket brigade devices (courtesy of MOS technology) had been built and tried but I don't think they were quite good enough for commercial use due, IIRC, noise issues. Were any such BBD based reverb units ever successfully marketed?
====snip====
And to even greater extremes without introducing the gross distortion products inherent in the analogue methods (limiting to FSD without clipping for example - not usually beneficial to a musical performance though Radio Bulgaria[1] would have given their right arm for such a processor - they had to make do with an RF clipping based processor).
[1] A short wave cold war propaganda radio station famously noted for its harshly compressed news and propaganda announcements that had been processed to within a millimetre of its life in order to cut through the fading and QRM. So famous in fact as to lend its name to the effect generated by the RF based speech clipping/compressors used by radio hams when working DX on RT (SSB and, in rare cases since SSB became the modulation method of choice over forty years ago, AM transmissions).
ANY SPEAKER ARRAY AS LONG AS IT IS THE CORRECT OHM-AGE FOR YOUR OUTPUT TRAN SISTORS IN YOUR AMPLIFICATION ARE AS GOOD AS ANY OTHERS. ALL SPEAKERS DO I S SHIFT AIR THE BIGGER THE MAGNETS ARE THE MORE AIR THEY SHIFT .. QUALITY OF AMPLIFICATION AND AMPLIFICATION,STABILITY UNDER LOAD ARE MORE OF AN IMPO RTANT FACTOR .. IF ITS VOLUME YOU ARE AFTER PUSHING AMPS HARD CAUSES DISTOR TION.... SO YOU COMPROMISE QUALITY FOR LOUDNESS ITS BETTER TO RUN AN AMP AT ABOUT 50-70% OF ITS POTENTIAL LIKE DRIVING A CAR YOU RACE THE BALLS OUT OF IT IT GOES FAST BUT YOU GET A QUALITY RIDE AT AN AVERAGE SPEED .... YOU WA NT 1000 WATTS RMS RUN A 2000 WATT AMP AT HALF VOLUME. YOU WILL GET 1000 WAT TS QUALITY DEPENDING ON YOUR PRE AMP...... YOU ONLY GET OUT THE QUALITY YOU PUT IN .... 1000. WATS RUN FLAT OUT EVEN THROUGH THE BEST SPEAKERS IN THE WORLD WILL SOUND SHIT. EVEN IF THEY ARE RATED AT 2000 WATTS RMS.. BEST RE COMMENDATION IS TO USE POWERED SPEAKERS {NO LOSS OF SIGNAL DUE TO CABLES}US E GOOD PRE AMPS AND DAISY CHAIN YOU SPEAKERS TOGETHER IE LINE IN LINE OUT.. .....
{{curtsey}} :-)
I'm afraid you're both imagining a non existent problem. Whilst it's true that the art of creating a loudspeaker from drive units and carefully crafted boxes is more an arcane art than a science (at least when it comes to creating something with a passingly good frequency response and not too many unwanted resonances), assembling an array of nominally identical drive units to make an even bigger more powerful 'super speaker' is going to add almost no additional detriment to such a loudspeaker's already compromised performance as a high fidelity sound reproducer.
Assuming the enclosure has been scaled up to allow each drive unit its own *allowance* of breathing space and the enclosure properly constructed to handle the extra sound power (suitably braced and strengthened to compensate for the larger panel areas and so on), a 2 by 2 or 3 by 3 drive unit array will work just like a single drive unit of 4 or 9 times the power rating and (if wired series/parallel) of the same nominal impedance without any unwanted interactions worthy of note between the drive units. Also, they will remain equally well damped as if each were connected directly to a low impedance amplifier.
The "Damping Factor"(tm) 'figure of merit' still being quoted (apparently) is pure marketing bullshit which has very little to do with the reality of the electrodynamic damping effect of a bass driver's fundamental resonance other than it's a good idea to keep the amplifier's output impedance below the 1 or 2 ohm mark. Any values below this are kind of sabotaged by the presence of the 8 ohms of voice coil
*resistance* effectively in series with the amplifier's output impedance as far as the electrodynamic damping mechanism is concerned.HomeOwnersHub website is not affiliated with any of the manufacturers or service providers discussed here. All logos and trade names are the property of their respective owners.