why don't we use predistortion more in the audio world?

I hope you are right! Could they also synthesise the physical sensation of the bass drum sound hitting your chest etc.? I imagine the 'trouser flap' might be a difficult one to do, though.
I think you haven't yet fully taken off the blidners ;) Think about generalizing this fully: what happens when someone says that their disability is their IQ or learning problems and they need a prosthesis for that? Note that rat hippocampus model was already duplicated in silicon nearly a decade ago. We're talking brain-segment prostheses here.

The sci-fi authors of old were wrong. It's not outer space that science and engineering are making the most staggering progress in the medium term; it's inner space.
 
I hope you are right! Could they also synthesise the physical sensation of the bass drum sound hitting your chest etc.? I imagine the 'trouser flap' might be a difficult one to do, though.

Different interfaces for different prices, depending on insurance. I can imagine waiting list to a surgeon for an interface upgrade. :D

Also, re-sellers of illegal devices that being plugged into legal interfaces produce illegal sensations. :D
 
Knowing my own inability to not hear what I expect to hear when it comes to subtle differences, I am doubtful whether I would be able to meaningfully try this myself. To me, a CD recording of an LP sounds exactly like the LP, but maybe that's just because that is what I expect to hear. I am looking for confirmation of digital's transparency (to an arbitrary degree) from an arch-objectivist who actually understands fully the characteristics of these systems..! Otherwise, there must always be an audiophile question mark over any correction system based on digital.

The Garrard is not a typical turntable . It is like a big V8 engined car . The grin factor is off the scale . The 301 even more so . Differences become obvious . My 501 was reviewed in Germany . The reviewer said suddenly it's the 1960's again and I am listening to the Beatles for the first time on the radio at home . Was he imagining it or was it really that sound that only Garrard's and EMT's make that got through the radio ? Absolutely right .


I was listening to Elvis on Long Wave , my other half said whose that ? Elvis , to which she said " are you sure " . Yes I said however it is CD and lacks the chest harmonics . This was on French radio and at a great distance . I can say with certainty that it would be that . I heard also Garrard 301 Heath valve amp and stacked Quad 57's playing Frank Sinatra live . Asked as to how it sounded I said I didn't like it because it frightened me . Why ? Because he is dead and somehow he is not dead on this system . Everyone in the room nodded . Sinatra seems to always suffer the De-aging when played via CD ( he sounds 18 when 40 ) . Now on FM 13 bit I never noticed that ! The BBC used 401's ( SP10's , 301's ) . I wonder if BBC digital was somehow better and by a big margin ? I did read all the standards for early digital systems . Surprisingly some of it was to do with digital optical film sound tracks and editing . Maybe that caused some problems ?
 
. I heard also Garrard 301 Heath valve amp and stacked Quad 57's playing Frank Sinatra live . Asked as to how it sounded I said I didn't like it because it frightened me . Why ? Because he is dead and somehow he is not dead on this system . Everyone in the room nodded .

Something similar happened when I brought my prototypes of tube amp and concert line arrays to the restaurant where my wife's colleagues celebrated Christmas. I designed speakers to work effectively from 60 Hz only, so took 1/3 octave graphics EQ to boost lows. The night before I tested on fiber-glass cones of speakers mix of Plasticine and lacquer to damp surface resonances, it was still wet. Louis and Ella sounded alive, like here and now. We thought the whole staff of that restaurant served in our room. They were coming to hear live dead artists... Their boss asked where to buy such a system, I said that it was a prototype, not for sale. And was right: when thin layer of damping material on cones dried out few days later the magic dissappeared.
 
DF said BBC 13 bit is basically NICAM . This quote might be the key ? I also suspect the discreet circuitry of early BBC digital might be a factor ( get at-able ) ? OP-AMP's and DAC'S pull rabbits out of hats , or do they ?

Quote Wiki .

NICAM sampling is not standard PCM sampling, as commonly employed with the Compact Disc or at the codec level in MP3, AAC or Ogg audio devices. NICAM sampling more closely resembles Adaptive Differential Pulse Code Modulation, or A-law companding with an extended, rapidly-modifiable dynamic range.

http://downloads.bbc.co.uk/rd/pubs/reports/1972-31.pdf

http://www.freescale.com/files/app_specific_stand_prod/doc/data_sheet/MC44C404.pdf
 
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Perhaps you could put your zero-feedback amp into the feedback loop of a second, zero distortion, amplifier. Record your music in this configuration, and then play it back through your amp as usual. This might illustrate the point that feedback is an inevitable part of error correction.

Like this :)

Top graphs: amp without predistortion
Lower graphs: amp with predistortion

input: bottom left
amp: bottom right
distortion engine: bottom centre
 

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digital, in general ? or a certain spec? because digital is merely a representation. How accurate it is, is variable (and can be pretty effin' accurate).

I know that, and you know that, but on these forums there are those who say that there is something indefinably bad about (practical realisations of) 'digital' and good about analogue, despite what the 'conventional' measurements say.

For a while I thought I had found someone with impeccable academic and objectivist credentials who would confirm that there is no reason why a digitally processed or recorded version of vinyl cannot sound like vinyl. However, you will notice that despite my very best efforts I cannot get him to say it..!
 
Obviously yes. The trick is, using multiple position of microphone in the room eliminate as many as possible variations of the sound field dependent on listening position, and equalize speakers only, as best as it is possible. As I said before, Audissey does that quite well.

Still thinking about this. The question is: which is more 'correct'? A flat loudspeaker in a room, or a loudspeaker equalised to eliminate variations of the sound field dependent on listening position in the room? Is it the same answer for a pair of stereo loudspeakers? Is it the same answer for all types of recording?

It occurs to me that a real musician in a room is not in any way equalised, so a solo musician recorded in 'dry' acoustics at a far extreme of the stereo field (on one speaker only) could certainly sound as though they are 'here' if left unequalised, whatever the characteristics of the room or listening position. As a producer that might be something I am aiming for. By playing with the playback EQ in any way, I would be damaging that illusion because I would be adding unnatural peaks and troughs to the spectrum of the solo instrument. Maybe that's a very rare and artificial situation, however, and normally we are being invited to listen to the musician 'there', in another acoustic.

However, I think we are pretty good at assessing the acoustics of a recording even when listening to it in our own 'live' room, as though we naturally 'hear through' the acoustics of our room - to some extent at least. Undoubtedly, though, when we want to listen to something in detail we automatically reach for the headphones, in order to eliminate the smearing of detail that the live room causes.

There's also that point about short, isolated, transients (say 5ms) reaching our ears before any reflections. Any basic frequency-dependent EQ to correct 'room response' will mess that up, so the 'bite' of the first few milliseconds of a snare drum is compromised. The same will be true of any music whose content is changing rapdly: it simply does not suffer from permanent peaks and troughs in the response that can be eliminated by frequency-dependent EQ.

I can't help but think that frequency-dependent EQ is a waste of time and is bound to sound unnatural no matter how the settings are derived. The alternative is to go with an impulse response correction that aims to give you the headphone sound at your listening position - but we know that doesn't work if you move your head by a millimetre. Seems to me that the best compromise will be what most people say: cut down the reflections with acoustic treatments and only apply time- and frequency-diminishing impulse response correction at the lowest frequencies.
 
I can't help but think that frequency-dependent EQ is a waste of time and is bound to sound unnatural no matter how the settings are derived. The alternative is to go with an impulse response correction that aims to give you the headphone sound at your listening position - but we know that doesn't work if you move your head by a millimetre. Seems to me that the best compromise will be what most people say: cut down the reflections with acoustic treatments and only apply time- and frequency-diminishing impulse response correction at the lowest frequencies.[/QUOTE]


As John Deans a friend says you can't EQ time domain problems . However it helps to try .
 
As John Deans a friend says you can't EQ time domain problems.

And I suppose it depends what you think of as a problem. A bit of 'live-ness' in a listening room does not sound wrong, even if the steady state frequency response looks truly terrible. Listening to stereo in an anechoic chamber might be theoretically correct, but I bet it would feel horrible.

Anyway, although EQ is still pre-distortion by certain definitions, it's off topic for this thread! (But I feel better now I've got it off my chest).
 
Oxford town hall is where I did many recordings . I could get a sound infinitely better than the actual sound . It took me years to ignore the sound of the building . One day I noticed some unoccupied more expensive seats where I knew it would be better . I have to say the sound difference when forced to take my proper seat was a more dramatic change than a new set of speakers . I fell out with the safety man at the town hall who considered my microphones to be a safety risk ( idiot ) . I decided to call it a day . On returning some years later I found I could still process the sound in my head and enjoy it . That surprised me .

Our local Littlewoods store used a string quartet to celebrate the enlargement of the store . They were playing in the section that sells coats . It was completely anechoic . Hayden sounded dreadful . I asked if they had any Ravel . They were very snooty and said only a piece of 20 bars . It was total magic . It seems Ravel somehow built echoic qualities into his music . It is better played by slightly detached professionals as it needs no schmaltz . They were perfection .
 
That predistortion will have the same effect as NFB, in that it will sharpen the eventual clipping - there is no free lunch! However, it is a good example of applying predistortion where NFB cannot be applied (at least, not easily). In that case NFB would require somehow detecting the response of the tape head or even the tape. The same issue does not arise with amplifiers, although it could perhaps arise with loudspeakers.
 
Black's 1st pass idea was feedforward error correction, famous audio implementation by Quad "Current Dumping" Amp

the "non-interacting" power output combiner is a difficulty

the L,R output combiner could be seen as a XO, the low frequency branch thru the output series L is driven by the main power amp

the fast correction amp path to the output through the resistor would be a 1st order highpass

but since the correction needs to add up to unity at the output you need a 1st order roll off for the correction amp's signal - hence the integrator

people like to draw the Quad Dumper circuit as a AC 4-arm bridge, the parallel circuit shown has a similar correction amp signal frequency shaping


I'm not yet really sure of the utility Black's Feedforward Error Correction scheme for audio power amps - except as a "band-aid" for being stuck with a poor performing main power amp
since you still have to accurately measure the error it seems like negative feedback around a better power amp would just be better

In the example of the BobAmp circuit I think doubling up ouput Q to avoid Beta droop and using 5x faster RET would let you get similar distortion improvement with 2-pole compensation

the Vanderkooy paper did suggest Black's Feedforward Error Correction as a way to help out Class D main power stages
 
practically I'd expect the distortion residual of a high feedback amp to be highly variable with operating point, thermal history – and very, very small

which would make it hard to accurately model, predict to maintain the cancellation with a predistoriton approach

to predict the Q die temp in the amp is a multi-physics problem = thermal radiation, convection, conduction
how much instrumentation are you going to afford - air, heatsink, transistor tab temperature sensors?, output current, power supply V and Re deltaV to get electrical power dropped in the output Q? - what about the driver Q now?

To be motivated to try predistortion in audio power amps over really good implementations of high negative feedback in audio amplifiers I'd like to see better evidence of “the problem” with negative feedback

sighted, uncontrolled listening impressions by the crew that failed the Carver Challenge aren't doing it for me


with their own source, speakers, in their own listening room Stereophile's "Golden Eared" reviewers couldn't tell Bob Carver's $600 SS amp from their own choice of "SOTA" tube amp after Bob tweaked the SS amp frequency response, output impedance for a deep null with the tube amp
 
practically I'd expect the distortion residual of a high feedback amp to be highly variable with operating point, thermal history – and very, very small

which would make it hard to accurately model, predict to maintain the cancellation with a predistoriton approachto predict the Q die temp in the amp is a multi-physics problem = thermal radiation, convection, conduction
how much instrumentation are you going to afford - air, heatsink, transistor tab temperature sensors?, output current, power supply V and Re deltaV to get electrical power dropped in the output Q? - what about the driver Q now?

If I was minded to try this experiment, I would try to maintain the amp's temperature at a fixed value in order to reduce that dimension at least, but I could feed the output of various sensors into the processor, too. The question would be mainly one of repeatability: I imagine I could learn to pre-distort optimally using previous and future samples as inputs to my neural-style network, but would it work the same two weeks later?

how much instrumentation are you going to afford

I don't think you are really into this audiophile thing! A few temperature sensors, extra heating transistors, even air conditioning for the whole room are as nothing compared to what these people spend on a mains cable. I want to do this for the audiophiles..!