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Prospect of Pure Perfect Sound Heralds the Death of the Established Audio Industry

technoian

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The marketing slogan "Pure, Perfect Sound - Forever" was used by Philips in conjunction with the launch of the digital compact disk in the early 1980's. The claim was greeted with scorn but it is only now, some 40 years later, that the other elements of the audio chain are about to get a digital makeover and those early claims may soon be realised.
For a start I think I need to define what I mean by perfect sound. To me a audio component displaying perfect sound is one with sufficiently low distortion so that none of those distortions are detectable by a listener. Currently it is in the field of DACs where were see many new products that are achieving this level of perfection so that a listener should not be able to detect any difference in the sound quality by substitution of any one of these 'top tier' DACS. Moving on to amplifiers it has been suggested that any amplifier exhibiting less than 0.01% THD+N (-80dB) distortion throughout its range of output power can be described as being audibly transparent and can therefore be said to have a perfect sound. There are several amplifiers now displaying this level of performance but they generally come at a premium price. How would you like to see this amplifier performance across all price levels? This prospect is now tantalisingly close with the development of the AX5689, a Class-D audio amplifier controller chip from a company based in the Netherlands named Axign. Axign was founded by former employees of NXP and the Philips Class-D division which was also the former employer of Bruno Putzeys who's subsequent work with Hypex and Purifi has done so much to raise the measured performance of class-D amplifiers. The theory behind the AX5689 chip is that a high speed analogue to digital converter (ADC) is used to sample the output of a class D amplifier and feeds that back to compare against the digital input signal. Any differences are then compensated for in the digital domain so that the amplifier output now closely matches the input signal. Just as a conventional amplifier uses feedback to improve performance this control chip uses a digital control loop to improve the performance of any class-D amplifier. What is especially impressive is that this technology can be used to improve the performance of existing class-D amplifier chips to raise their performance to that of our perfect sound criteria. This technology has been released into its first commercial product in the last month, namely the Harman Kardon Citation Streamer/Dac/Amplifier. We eagerly await its testing in ASR.
So now we have both the DACs and amplifier components exhibiting perfect sound we need to turn our attention to the elephant in the room. That elephant is the loudspeaker and it's performance is critical to overall sound quality but then so is the room itself who's contribution also needs to be addressed. Let's start with the loudspeaker. The loudspeaker is the highest source of distortions in the audio component chain and in a modern digital audio system has the biggest influence on perceived sound quality. The moving coil (dynamic) loudspeaker was invented at the end of the 19'th century and progress over the last 100 years has primarily centred on the use of new materials to tune the weight and stiffness of the drive units. Other loudspeaker technologies have been invented but it is still the dynamic loudspeaker with an number of drive units contained in a box that represents the vast majority of speakers sold today. The room in which the speakers are placed also influence the sound produced which varies with the dimensions and materials of the room itself. Compensation for room effects is now built in to most home cinema amplifiers and for stereo systems the DIRAC system is considered to be the current state of the art. So how might we compensate for the distortions and non-linearities introduced by the combination of speaker and room? What I foresee is an extension of the digital control loop circuit as created by AXIGN to now encompass both these additional components. A calibrated microphone could be used in conjunction with the ADC to feed back to the input of the amplifier which then compares the measured response to the original source signal and adjusts the signal to the amplifier to compensate for any differences. This is certainly not a trivial task with several problems to overcome but the technology needed is all in place today so I predict we will see a development of this coming to the marketplace in the next few years.
So now we can see how the use of digital feedback loop circuits could be key to bringing us perfect sound but how is this going to bring about the death of the current audio industry? I see this happening in two ways. Firstly if every audio system is now producing perfect sound there will be no incentive to upgrade unless you want additional features. Audio sites such as ASR will cease to exist as what is the point in testing anything when it all measures perfectly. Secondly it should be remembered that the digital feedback loop circuit is potentially cheap to implement alongside the existing individual components in the audio chain. The DAC/AMP/Speakers do not have to be individually perfect themselves to achieve a perfect result when subjected to a digital feedback loop. Therefore the technology promises to bring audio perfection at a low overall cost. Boutique audio manufacturers will cease to exist as they will be unable to compete in a market flooded perfect sounding systems all available at low cost.
Now where does the audiophile fit into this new future? Will this be an audio nirvana where we are finally able to concentrate on the music we play rather than on the audio systems we play it on or will an audiophile sub culture form where anything digital is dismissed as sterile sounding and analogue equipment from the 'golden age' is traded at vastly inflated prices to the advocates of analogue?
This is just my vision of the audio future. Would anyone care to comment?
 

Wes

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You haven't read The Collected Works of PT Barnum, have you?
 

BDWoody

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RickSanchez

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Firstly if every audio system is now producing perfect sound there will be no incentive to upgrade unless you want additional features.
This argument does not seem to appreciate the vast spectrum of motivations involved for buyers of audio equipment, especially those motivations that are not particularly logical.

Audio sites such as ASR will cease to exist as what is the point in testing anything when it all measures perfectly.
This argument seems to greatly diminish the importance of ASR and its ability to independently verify (or disprove) manufacturers' published specs. Not to mention the many other contributions that ASR provides to the audio community.
 
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SKBubba

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Dacs, amps, streamers, formats, etc. are pretty much sorted.

The next frontier is transducers, where air gets moved. Air is a funny thing. Way more variables than are even being contemplated at this time.

Ears are funny things, too. The final frontier is direct injection of audio perception into the brain, bypassing air and all this other crap.

After that, assimilation into the primal energy of the Universe.
 

egellings

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Would sound still seem life-like to us if it were injected directly into the brain, bypassing the ears? Would correction be added to this injected signal to add an ear signature to it? Why not just use the ears and get if over with? It'd sure be cheaper!
 

BDWoody

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Would sound still seem life-like to us if it were injected directly into the brain, bypassing the ears? Would correction be added to this injected signal to add an ear signature to it? Why not just use the ears and get if over with? It'd sure be cheaper!

Maybe they can install a socket in your skull to facilitate op-amp or tube rolling...
 

sergeauckland

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When I read the opening post above, I at first thought it was a reprint from 1990, as 'pure perfect sound' was as available then as now. The only things to have improved significantly is power output, as low noise and distortion sufficient for transparency was available easily by then in electronics, and the better loudspeakers of the era were as good as much available today, although I'll accept that the newest DSP active loudspeakers are better than what was commonly available then. Although Meridian had DSP active loudspeakers then, but without integral in-room correction.

Since the late 1980s and early '90s, prices of the high end have got increasingly outrageous with no improvement in quality, as none was possible once transparency was reached. So, it's been style over substance and chasing increasingly small but inaudible technical improvements, 'because we can'!

Is it any wonder Joe Public can't see the point of it any more?

S
 

andreasmaaan

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A calibrated microphone could be used in conjunction with the ADC to feed back to the input of the amplifier which then compares the measured response to the original source signal and adjusts the signal to the amplifier to compensate for any differences. This is certainly not a trivial task with several problems to overcome but the technology needed is all in place today so I predict we will see a development of this coming to the marketplace in the next few years.

The time it takes for the sound to travel to the microphone would be too great for the feedback loop to operate effectively.

However, the idea doesn't sound too dissimilar from other ideas that are already in use: motional feedback and servo control.
 

bloodshoteyed

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Maybe they can install a socket in your skull to facilitate op-amp or tube rolling...


i never realised those were tubes sticking out of his neck...


images.squarespace-cdn.com.png


wonder what he's listening to, tho....The Cure, maybe? or is it Sisters of mercy?
 

egellings

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Once the ears cannot tell a difference anymore, why go further? There'd be no sonic benefit form doing so, even if electrical specs improved.
 

RayDunzl

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The time it takes for the sound to travel to the microphone would be too great for the feedback loop to operate effectively.

Well...

If your coder is tricky enough, the feedback could be to an EQ (phase and frequency amplitude) that could home in on some optimal setting.

Maybe.

A little training might be required.

Establish the basic delay in the loop.

Compare the past source to the current reception.

Calculate and apply some corrective measure.

Make new comparison.

Decide if improved or degraded.

Keep or discard the adjustment.

Repeat.

@pkane could probably whip it up in an hour or two.

Or just do a couple of sweeps and apply the DRC you already have and call it a day.
 
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andreasmaaan

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Well...

If your coder is tricky enough, the feedback could be to an EQ (phase and fequency) that could home in on some optimal setting.

Maybe.

A little training might be required.

Establish the basic delay in the loop.

Compare the past source to the current reception.

Apply some corrective measure.

Make new comparison.

Decide if improved or degraded.

Keep or discard the adjustment.

Repeat.

@pkane could probably whip it up in an hour or two.

Ah, I interpreted the OP to be talking about a system that worked on music in real-time.

Definitely agree this could work when it comes to system/room calibration.
 

RayDunzl

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Paragraphs can be good...

If interested in the topic I'll do it myself as I read it.

I've never seen Grammar Policing work on the public at large

1605484515877.png
 

RayDunzl

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Ah, I interpreted the OP to be talking about a system that worked on music in real-time.

My post was intended to be read as a real-time continuously running loop with music as the source..

Compare what was sent in the past, to what is received in the present, and adjust the correction to be applied for the next iteration.
 
OP
technoian

technoian

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The time it takes for the sound to travel to the microphone would be too great for the feedback loop to operate effectively.

However, the idea doesn't sound too dissimilar from other ideas that are already in use: motional feedback and servo control.
You are right in saying that the time taken (latency) is the is the main problem with introducing the speakers and room into the digital feedback loop. There is the possibility that a series of calibration signals could be used to compensate for speakers and room during setup phase and then the loop just compensate dynamically for the DAC.
 

Putter

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If interested in the topic I'll do it myself as I read it.

I've never seen Grammar Policing work on the public at large

View attachment 93779

The point is communication. Making it easier for people to understand what you're saying. I personally don't see it as policing since there is no way to enforce it. It's more like a strong suggestion. Otherwise you're likely to be either ignored or misunderstood. I'm not big on paragraphs myself, but I do like to at least separate different thoughts and topics by skipping lines.

P.S. Thanks for your 'translation'. I really did skip over the initial post after reading the first few lines.
 

Robin L

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As long as sound goes through transducers turning into electrical waves and vice-versa, it's Xeno's paradox all the way down.
 
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