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The flaws of blind listening tests


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By FAR, the biggest variation in my system (very small, limited -- only for technical reasons -- not for casual listening) is my hearing/ears.  The variation is based upon BOTH hearing and my ears.  Hearing is dependent on mental state and mental fatigue, and ears have lots of variable imperfections also.

Controlling the audio listening experiments is incredibly tedious -- because hearing doesn't just vary in the way that something sounds better or worse, but can sound better or worse in MANY ways.

Even though we attempt to measure the experimental results of my project -- sometimes listening is the only way to determine success/failure (quality.)  It is the variable of human hearing that is so difficult to control.

Even simple A/B can result in confusing results.   IMO, getting proper/maximally correct results requires statistical measurement because there are so darned many variables.  If someone believes that there are aspects of their systems that might have variability, then those issues can also be mitigated by statistical measurement.  (The only places that I believe might be a little bit variable are the transducers and whatever electronics poorly designed/drifting.  Sometimes EMI/RF and other enviornmental issues can also cause problems.)   A good example of 'electronics' issues confusing the stats might be homebrew designs that don't properly account for environmental changes.  (Back when I worked at Bell Labs, we had to design electronics that worked over enormous temperature/humdity ranges -- and the required design discipline is much wider ranging than doing a simple room temperature design.)   A minor example that often happens in the meat of audio design might be power amplifier biasing -- but professional quality designs would account for the entire circuit working at specification over VERY MINIMALLY 0 to 50C, and hopefully  -10 to 70C.   Such criteria helps with actual operation in the extended environments, and gives additional margin for reliability.

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7 hours ago, Jud said:

In this discussion, it is not necessary to reinvent the wheel. There are a wealth of academic peer reviewed papers available on the relevant topics.  I personally am aware of dozens, and I'm sure there are at least hundreds if not thousands.  So instead of everyone opining, we would all likely learn more if people brought up relevant papers and discussed them.

I VERY MUCH agree with your sentiments, seems that there is sometimes too much navel contemplation in similar situations, but as long as the learning progresses, and people don't get tied up in useless opinons, I believe that the exercise can be beneficial.

I do worry that there might start being too much polarization like in the 'stair step' and 'jitter' religions.  Not everyone who develops an opinon has the math/engineering/signal processing background for the opinion to have meaning in the real world. (This probably comes from a distrust of 'experts' -- in some cases, rightfully so!!!) It takes lots of discipline from EVERYONE to make the discussion progress and come to coherent conclusion without prematurely developing opinions.

It would be nice if someone actually reads a few of the papers (with references) give a synopsis and point the discussion to those papers (hopefully available.)   Probably wont' happen, and the navel contemplation will ensue  :-).

 

John

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4 hours ago, Jud said:

 

Although for this particular topic, the background, or at least curiosity and desire for education, should be in the human auditory system and associated areas.

Actually'flaws in blind listening tests' include not using statistics thereby getting noisy results.  For example,  is also helpful to know  that the environment can also cause things like transducers to vary in characteristics.  The various aspects of human hearing ARE important, but there are lots of really interesting areas in that field which are not important.  There are other fields of study which will also help to get the best possible results.   I wouldn't spend all of my time resource on  human 'hearing', but also study the areas of the appropriate fields which will help to get/process good end results.  BTW -- even the general idea of 'collecting data about human hearing' isn't sufficent to test the listening characteristics of audio equipment (but cannot hurt to already know as much as possible.)

 

When reading papers (studying for a work purpose), sometimes a certain focus is needed.  Much of the time, we EEs, DSP, and software people (I am all three) cannot be basic experts in all of the areas where we must work, so for the sake of efficiency, studying appropriate  areas with a focus on the needed information (basically 'getting to the point') is critical.

In this instance, trying to determine and fix the flaws in listening tests, understanding the techniques needed to remove as much bias as possible, while STRUCTURING the test is just as important as knowing the loudness curves or (for example) problems with hearing in old age (and that problem IS causing me troubles -- with rather frustrating changes in my hearing from day to day!!!)   Also -- appropr iate for the N hemisphere right now --  the big change in humidity in homes due to outside air temperature can signifcantly change the characterstics of certain kinds of transducers.  Also, home brew equipment is likely to be 'eccentric' relative to lots of factors (power supply variations/etc), so making sure to consider those factors might also be important.   There are very many (external) things that can mistakenly help to cause more error than really need be.  Lots of strange external factors can creep in -- then the results can have more error than need be.

 

I don't think that focusing primarily on the characteristics of human hearing is best allocation of resources, but some background in hearing, stats, engineering, and attempt to avoid personal bias (yes -- I know -- really cannot eliminate personal bias, but we can try) will make it possible to get usable AND CONVINCING results. 

 

BTW -- I don't basically disagree with you that it is INFINITELY more useful to do a bit of research before giving personal opinons.  If I hadn't done the 'data collection and presentation' thing many times, I probably wouldn't have written much about this.

 

John

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13 hours ago, Allan F said:

 

Another "pablum" generality completely devoid of any substantive content. Why don't you tell us how great it is to breathe air? Bye Bye.

I agree with the idea that it is frustrating to actually communicate when someone is trying to express themselves  without the engineering/math/scientific expertise.  Sometimes a provincial language emerges that is used to communicate amongst the non technical individuals.  I don't know the answer to the matter other than 1) a semi-permanent division between those who really understand and those who THINK that they understand  OR 2) an attempt to communicate and inform/educate those who are coming up to speed. (Maybe a combo of the two choices -- there might be more choices, just giving examples.)

 

I do have a somewhat 'soft spot' for those who are missing some technical background - being a very flexible and quick learning engineer, then I sometimes have a LOT to learn in specialty fields.  Sometimes I am clumsy when communicating with the experts in a specialty.

Even though I am a fairly competent DSP/EE/and software engineer/developer, there are areas where I have limited knowledge.  Just happens is that I am currently learning a little bit about REAL professional mastering/recording, etc.   I am NOT an expert in the field at all, but my software project is a tool that such individuals might need (probably the first fully functional software DolbyA decoder that REALLY sounds like a DolbyA.)   Just because I wrote the software, developed the rather sophisticated algorithms to emulate rather eccentric hardware design, doesn't mean that I really know how to professionally use the device :-).

Just trying to explain -- I do have toleration for those with a bit of pseudo-knowledge, as long as those people are willing to learn something new.  In fields other than my own real expertise and experience, I might seem to be full of 'pseudo-knowhow' also!!!

However -- I do not like to deal with people who have lots of pseudo-knowledge and NOT have an open mind to learn.

 

Just my opinion....

John

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1 hour ago, fas42 said:

 

The limitation in what you're saying is that the psychoacoustic element is largely ignored, or inadequately understood. Only the most recent research into human hearing is getting a better handle on how the brain operates, with explanations for "how" tied up in the whole 'mystery' of human awareness.

 

What my first "competent" rig did was to throw up a completely convincing sound field - it was literally impossible for me to make myself register that the speakers were the source of what I was hearing - the 'illusion' was rock, solid. Now, this exact same setup could 10 minutes later fall off this high perch, and sound just like an ordinary hifi, just another pretty decent audio combo - and nothing obvious had altered in this time frame. So, what the hell was going on here?!! ... and to this day I still haven't got a fully comprehensive answer.

 

Part of the answer is that the mind "fills the gaps"; when the sound is good enough, the brain adds the extra needed to complete the picture, and a 'mirage' fully forms. But to comprehensively explain that, with a fully technical explanation, is still some time off ...

You gotta realize that the 'psychoacoustical' issues are also able to be expressed in technical terms.  All of this isn't rocket science nowadays -- it might have been 50-70yrs ago, but not today.

Part of an 'educated ear' is to be educated as to how to talk about what you are hearing.  This aspect of 'listening' is one reason why my DolbyA decoder works so well...  I can often interpret what I hear -- even with my 63yr old ears, and determine what kind of technical remedy is needed.  Sometimes (more often than I'd like), I run into problems that I don't know how to interpret -- and that is motivation to learn something new.   It is best not to use diffuse or obscure language, or to obfuscate by bringing in new topics.  NONE OF THIS IS ROCKET SCIENCE ANYMORE!!!  If I thought in obscure terms or push something off to 'psycho acoustical' when it isn't necessary, then nothing would have gotten done...  It is also not to push-off (for example, my DolbyA project) as simple -- it has over 50FIR filters, over 20IIR filters (probably 2-3X those numbers -- havent counted), some of them dynamically change their Q/FC on the fly.  The DolbyA expansion attack/decay times are necessarily dynamically calculated because of the unfolding of the original feedback to feedfoward design.   When someone talks about a defect in a simple amplifier, preamp, RIAA curve this or that, noise, stereo image, or whatever -- I wonder why they cannot express such simple matters.   Think about the complexity of something that is over 50x more complex with10x that in interactions.  THAT is complicated, and never been as successfully done before.

Do you remember that old phrase 'KISS' (and I don't mean to be disrespectful -- just reminding people to keep things as simple as possible!!!

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8 minutes ago, sandyk said:

 

 

That is not the impression that I get from reading most of your lengthy replies. ;)

 Please remember also, that this is an Audiophile site, where members should not be required to learn " Engineer Speak" in order to convey their messages/thoughts to qualified Electronics Engineers.

 However, quite a few members do have some experience in this area, despite not being qualified E.E.s , and a few like myself also have friends who are qualified E.E.s

EEs cannot read minds -- they can try to understand rather than blow-off someone as silly, just like an 'audiophile' shouldn't  feel threatened because there are people who REALLY know what is going on in the digital processing/circuitry/physics (e.g. acoustics), and human hearing (or at least know enough necessary to deal with the various audio issues.)  The subject of audio is multi-disciplinary, but doesn't require all that much in-depth knowledge in detailed medical or neurological fields.  Statistics for gathering and processing information -- for problem solving/quality testing -- is usually MUCH MORE OFTEN important than things like the Fletcher-Munson curve (it is sometimes helpful to know those things also, however.)

It is best for ANY of US to avoid the notion that they are some kind of pioneer in audio -- a lot of people have 'been there/done that'.  It is definitely more of an issue of enjoying the listening EXPERIENCE, and if there is something wrong that really bugs you -- then get help from people (or vendors) who are not going to take cruel advantage of you.

Trying to work together rather than 'pulling' back is important to solve the perceived problems.  For example (not to digress) but for example, the matter of 'jitter' often described as some kind of transfer of FM/PM noise through a digital system that is resynched at every step is 'nonsense', but the HONEST EE understanding that there might be a real problem somewhere is really necessary to solve the problem.  (BTW, most of the time the problem is analog ground/circuit noise -- but that is neither here nor there.)

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19 minutes ago, Jud said:

 

I feel, at least from my reading, that there's a bit of a gap between the leading psychoacoustic research and the understanding of psychoacoustics by audio electronics engineers.  But I'd be happy to see where I'm wrong.

 

One other area where I feel we're not quite all the way there yet is modeling the behavior of full audio systems made up of individual audio components and connections, analogously to the way we model audio circuits made up of individual circuit components and connections.

The real problem that you are talking about has to do with the environment and transducers, less so the digital/analog signal processing.  There are aspects of conversion of spatial representations (to improve perception), and there has been a lot of work in that area -- including software to implement some of the ideas.  However, there is a huge amount of silliness in the transducer arena, and I don't trust their accuracy  at all for my work .  This is why, when all else fails, use statistical techniques to zero in on the best solution.   For my own purposes, I cannot even think about using speakers in a room -- not accurate enough.   My headphones aren't perfect, but give me repeatable results.  About human hearing -- my ears are the most frustrating variable in my work.  Even though I take PERFECT care of my ears, I still have to deal with various kinds of fatigue, blood flow issues, tinnitus/etc.   Very frustrating indeed, but I have learned to remove part of those errors from my process.

Still as an absolute -- I know that my hearing is far from perfect, but since I use what I have intelligently, I can gather super useful information from it.

 

So -- I agree with you from the standpoint transducers and rooms -- proper coupling (from both the physical tranduction and spatial representation) to the human hearing system is not well understood by very many people.   That is outside of the scope of my work, and I tend to utilize whatever resources that I can get from the transducer manufacturers.  As an absolute, they cannot be trusted, however.

Used to be, you couldn't trust the accuracy of amplifiers either (with lots of distortion -- including the beloved even order warmth), but nowadays true near-perfection is avaiable at reasonable cost.

 

 

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1 minute ago, fas42 said:

 

I agree 100% with this, but not the earlier part of your post, :). Unfortunately, that "neither here nor there" is just about everything - if one wants the best standard of playback ...

Do you mean that EEs can really read minds?   It is the responsibility for both parties to be mature and understand their limitations.  This can make it easier to communicate.  However, that might be too much to ask when people have bought-in to an emotional position.

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49 minutes ago, fas42 said:

 

Auditory Scene Analysis (ASA) is very recent, only 20 years old, and vigorously being investigated - in part, it's about how humans look for patterns in what they hear, and if what they hear matches well to stored patterns, then the sound "gets a tick". Precisely how close the match has to be is the yet to be answered question ....

When doing an audio system -- the neurological 'pattern's, hearing structures need not be understood -- that is the purpose of the production/recording/playback mechanisms to provide transparency of the original arti'sts intent.  How it works inside of the brain/body is outside of the scope of anyone but researchers.  As it is, we have a stereo/quad/10channel or whatever signal with the intentional design of an artist/recording engineer/God or whomever :-).

 

We know that correctly designed/utilized  electronics are now perfect, the digital signal processing as a tool (not always how it is used) is perfect, so we need to know where problems need to be solved.   What else is left?  the coupling of the signal to the hearing system.

 

Right now, we have headphones (which can provide a perfect 1:1 relationship), and we have speakers (which are only loosely and imperfectly coupled.)

The other things include how to deal with the issue of spatial relationships -- do you trust the artist/recording engineer to produce a proper stereo (or other) signal, or do you want to somehow modify it from the original artists design?   That kind of thing is YOUR choice for what makes you 'feel good' :-).   No-one can read anyone's mind -- gotta communicate the desired changes to the original audio, or perhaps describe the defects in the 1:1 mapping of the artists stereo signal to the headphones.  (For larger numbers of channels, there needs to be a specification of the mapping -- e.g. where the speakers are, how to map the signal to the headphones, etc.)   Those are the perview of the originating source (which might include Dolby with their fancy encoding systems.)   They (for example) will prescribe how to utilize their signals.

 

More often than not -- when looking at the matter of some of the desire for audiophile perfection -- some of this stuff seems more like an extreme extension of 'tone controls'.  First, just 'bass', the 'bass/treble', then 'loudness' controls, then paremetric or multiband equalizers,  then 3BX, 4BX, 5BX, then messing with spatial relationships/etc.  Forever and ever.  Guess what? perfection is impossible, but it can be fun to tweak!!! :-).

 

John

 

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On 1/10/2019 at 6:59 PM, mansr said:

"For every expert, there is an equal and opposite expert." 

I like your statement.  That is why I TRY not to take strong positions and equivocate.  It isn't being wishy-washy, but too often, appearing to be 'too absolute' will cause a shutdown of communications.  In these kinds of discussions it is best to TRY to be mature (not always successful) and altruistic (don't try to make money from people.)

 

John

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21 hours ago, March Audio said:

 

So what if the wine taster refuses to accept what the engineer tells them? 

 

For example, "we don't know everything yet" or “there's a mechanism unknown to science going on that you can't measure which explains my subjective response". 

I agree with your concerns on that matter.  Maturity is important, and controlling ones own ego is also helpful.  i learned a long time ago in my field -- I am so damned good at what I do, until I realize that I am not so good :-).   I finally gave up on trying to be 'right' all of the time, and as a reflex action, step back and try to solve the REAL problem (whatever that might be), ignoring my ego.  My ego sucks, and I know it :-).

 

Now, my being/intention is based on solving the problem, helping to make someone feel better rather than just simply winning the war about who is right.

This maturity (and competency)  problem DOES exist in both the consumer & in the engineering communities.  However, it seems (just my impression) that as some consumers becomes more entrenched into their hobby -- some become more ego-bound and narrow minded.  A SUCCESSFUL (read: SUCCESSFUL, not just experienced) engineer will tend to become less and less ego-bound, and try to solve problems for people.

Maturity is critical, and open, honest and trusting communication is important.  The REAL problem for ALL of us are the snake-oil salespeople!!!   These money grabbers will dyseducate honest but vulnerable people with their profitable song+dance.

We engineers (who are really honest, and TRY to be helpful) can only offer help, and TRY to be kind.

 

Sometimes it can be VERY HURTFUL and DIFFICULT when some self assured and miseducated person (often by snake oil people) makes a misguided claim about the honest person's integrity or competency.  All we can do is to try to keep on doing the best that we can.  IT IS GOOD TO LISTEN, but sometimes gotta filter out the noise.

 

Anecdote:  I have been told over and over again that doing a DolbyA decoder in software is impossible, and that I am an incompent liar of some kind.  However, I and my collaborators have existence proof otherwise.  I am not trying to TELL someone that they need the results of my project, as I'll never make any money on it (far far from recouping the 2000+ person hours on the admittedly seemingly  impossible project.)  However, in some people's eyes, because of their dyseducation -- I am the 'bad' guy.

 

I have given up on trying to win little arguments about right and wrong -- gotta just do the best/right thing to help.

 

John

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7 hours ago, sandyk said:

 There does not appear to much point to your software unless it's for your own personal use, where you have identified quite a few CDs in your collection that were sloppily mastered for release , AND you play regularly.

 However, if there was made available a list of many of the sloppily mastered CDs that could be a different matter, with some people prepared to pay for the software.

However, being able to compile a list like that sounds a bit like " Mission Impossible"  too . ;)

Firstly, there is a demand for a superior quality DolbyA decoder for historical archives.  Much material is DolbyA encoded, and the normal DolbyA HW splats distortion all over the place (the well known 'softening' of the sound.)

Additionally, and more important to the consumer, a lot of that 'NOT QUITE RIGHT'  sounding digital material is actually DolbyA encoded -- as a consumer, you just dont' realize it.  So, if you got that horrible sounding ABBA, Carpenters (even from HDtracks) or whomever recording from before the 1990s, and want a master tape quality copy -- just use my DolbyA decoder.  IT IS A REAL ISSUE.

Refer to my repository -- the files with 'DAencoded' are DolbyA encoded, and files with 'DHDA', 'mastered' or whatever like that have been passed through my decoder.  THE PROBLEM OF LEAKED-INTO-CONSUMER-SPACE  DOLBYA IS REAL.  The sound of DolbyA tends to be fairly subtle on compressed/limited material -- but without decoding, the quality is inferior.

 

Repository:

https://spaces.hightail.com/space/yDG3L339Rn

 

Note that I dont' expect or intend to make any money on the project -- however, well known gov't historical institutions and grammy award winning engineers have shown interest in our project.  The formal name is DHNRDS, and there is a very primitive WWW site (not quite ready for prime time yet) for the project.  (I don't control the PR side of the project -- my recording engineer friend IS in control of the marketing/informational side...  I am just the author/owner of the software -- my engineer friend and myself own the project together. He has full usage/distribution rights to the software so that he can make decisions withiout asking me.)  He has been a WONDEROUS help in keeping me straight on what is needed...

 

john

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On 1/11/2019 at 10:57 PM, Jud said:

 

Actually, I was thinking more about things like ground and noise currents in actual systems in people's homes.  Unlike SPICE for circuit components, we don't have good software for modeling system topology: Whether you'll get less noise by connecting your DAC, amp, preamp, etc., with cables that have electrical characteristics A, or cables that have electrical characteristics B; with the power cords plugged in with arrangement Y or arrangement Z; etc.

There is software that is helpful for circuit topology -- but as you say, it isn't good for the application.  Such software is very often used by RF/microwave engineers - I have used such software.   Some free stuff is also available.   The best solution for normal audio work is a set of common knowlege about making sure that voltages don't develop because of inductance/resistance and ground currents.  This is one reason for the old mention of 'running all grounds to a single point'.  In some cases, that might not be best, but such noise is almost always because of  'current*impedance' developing evil voltages.

This syndrome of developing errant voltages is probably 90% of the misguided thinking that somehow clock jitter propagates through circuitry that is  reclocked/resynchronized.  The counterargument that I give about the 'jitter' issue is that internet jitter can be on the order of milliseconds or seconds, how can someone listen to music with so many long delays?  Answer:  buffering/resynchronization - and that happens in any reasonably competent recent design nowadays.

CDjitter does NOT propagate to the audio output unless there is a ground current causing voltage disturbance or equivalent analog issue.  Jitter due to clock edges on clock D flip-flip type schemes  (the usual explanation) can have some effect, but not usually in a competent design with good quality componentry.   Ground noise is the biggest bugaboo by far.

 

John

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2 hours ago, Jud said:

 

What I've seen on the forums isn't so much the notion of jitter propagating, but of (1) ground noise getting into clocking circuitry and causing jitter; and/or (2) poor signal integrity causing input circuitry in the DAC to be electrically noisy, resulting in electrical noise getting into the clock circuitry and causing jitter.  Whether and how much these matter I don't have the technical knowledge to say.

You are hitting 'the nail on the head.'  When I mentioned 'ground noise' -- that is most often the cause of many of the effects that you are mentioning.  Being more complete -- not only ground noise, but also other kinds of conducted noise coming from currents, radiated noise effects and capacitance (which are kind of two different things -- radiation is a 'bigger' process than just capcitance or inductive coupling.)  (probably a few more basic sources that I didn't think of off the top of my head.)  (Just a note -- simple capcitive & inductive coupling fall off very quickly in their effect, while radiation goes MUCH further.)

 

Very very very small amount of the 'jittering' phenomenon (various kinds of analog noise more generally) can maybe caused a LITTLE BIT by timing variations on a clocked flipflop, but there are so many bigger sources (which you also describe the specifics) that cause troubles.  When I hear about a new 'jitter free' this or that, I cringe -- hopefully the vendor is secretly doing the right thing -- basically ignoring jitter, and simply fix the noise problems!!!


John

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20 minutes ago, Jud said:

 

Have you read any of the discussions surrounding Dolby processing (don't know what letter) and Steely Dan's "Katy Lied"?  Supposedly the sound quality of the master was particularly stellar until someone ran it through Dolby processing.

 

Eidt: Looks like my memory was wrong, it was dbx noise reduction that at least partially/initially messed things up.

Both DolbyA and DBX I have their evils.   A lot of recording engineers really hate DolbyA because it creates distortion.  DBX is worse when it comes to noise modulation.  I am trying to mitigate some of the DolbyA evils.

 

John

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53 minutes ago, sandyk said:

 

Judging by the marked improvement with " ABBA-DayBeforeYouCame-DHDA" with a great deal of success !

Thank you!!!  My project partner and myself have already made some more major improvements since those demos.  It is my position and driving motivation that ANY distortion of any kind is not acceptable.  My ego isn't in the 'program', my ego is entirely in the results and making my project partner and the (eventual) users happy with the results.

 

Without having a real recording engineer working with me -- and several others commenting on the various issues, the decoder would have been a total piece of cr*p.  I know DSP and programming, but the real goal is the audio -- and it aint easy!!!  I have been told over and over again that a good software DolbyA decoder is impossible -- and I think that we have an existence proof that claim is wrong :-).

 

We do have a VERY NEW VERY LIMITED web site -- barely getting started, but it is DHNRDS.com.  Serious questions should go there -- however, I can always answer questions unofficially (not as a project spokesperson.)  There is some seriously innovative DSP code in the decoder (especially avoiding the worst of the modulation sidebands when doing the fast gain control -- interesting and tricky subject!!!)  My project partner is the official interface, but again -- I can talk tech talk (e.g. describe how to make really good compressors/expanders.)

 

IN FACT -- I think that the latest play version of my 3 band RMS compressor (source include) resides on the same WWW site as the demos.  It works pretty well -- might even fool people if it is working or not (it is very subtle), and uses the same kind of RMS calculations as DBX and THATcorp (but further optimized in dynamic attack/decay, etc.)  File name: simplecompMB-V1.1B-23dec2018.zip.  I don't maintain it, but if someone needs help, I am happy to make suggestions on how to use it (there are some postings on Hydrogenaudio for simple use) and how to modify it.

 

Take care, and thanks again!!

John

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  • 2 weeks later...
9 minutes ago, gmgraves said:

I can't explain it, but then, I'm no psychoacoustician. Who knows what subconscious factors are at work in any individual at any one time over any given issue? I mean, there are Americans who still think that Trump is doing a good job as POTUS!

Well being precise -- excellent at being an executive, even as President, but not the expected type of statesman, uses poor restraint  when communicating, and irritates the embedded parts of the government.  Is that good or bad?   In some ways, very good -- in other ways, not so good.  We have a system, the election spoke the will of the electorate per the Constitution, and lets get on with it.  It is better to do what is good for the country, not change ones stance because of hate (e.g. some people wanted the needed layer of security before Trump, and now don't want it because of Trump.)  Too much hate...  Trump might have problems, but the hate trumps Trump.

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1 hour ago, gmgraves said:

I used to have  a Dolby-A unit that I used for all my analog recordings. I recorded a pro symphony orchestra for 5 seasons using it as well as various jazz venues (including the San Francisco Jazz Festival) for the NPR "Jazz Alive" series. Aaron Copland, Hubert Laws, Dizzy Gillespie, and Stepan Grappelli are among the luminaries that I recorded. I used 15 ips (38cm/sec) and half Track stereo using Dolby-A for all of these. I'm pretty sensitive to distortion and what I found is that if the operator hasn't been scrupulous with matching the levels with Dolby-A on both the record and playback cycles, it will introduce distortion. My opinion is that this is fairly straightforward to do, but it is time consuming and I can see many technicians getting sloppy when they have to adjust separate Dolby units for as many as 48 tracks!. I'm going to venture a guess and say that this is probably where the notion of Dolby-A (and possibly DBX with which I have no experience) introducing distortion came from. I was only recording in two channel, and therefore took the time to make sure that my R-to-R tape decks were properly and meticulously set up and paired to my Dolby-A unit. 

Gotta say, when digital recording came along, I was happy to put those laborious chores behind me!

You are definitely right about DolbyA creating distortion when not calibrated correctly and I cringe if someone has multiple channels improperly calibrated trying to line everything up.

However, there is a deeper form of distortion that is created by any fast compressor or expander.  The DolbyA has the potential of a 1msec attack time at frequencies about 3kHz-20kHz (actually, the 3kHz band includes frequencies down to about 1kHz plus or minus.)  The 20-80 and 80-3k bands can have as fast as 2msec attack time.  Those fast attack times in cases of intense material will produce rather large amounts of transient distortion possbily all the way up to 30-40kHz.  In fact, if you look at the spectgram before and after a DolbyA encoding operation, you might notice a significant increase of above 20kHz material.  Much of that increase of 'splats' as I call them result from the mixing of the 1msec attack (with lots of harmonics) with the multiplication (gain control) of the audio signal.  This creates mixing products that are distortion.(The mixing is similar to the action of an RF mixer as used in radio recievers.)

So, when you get that old recording that had been DolbyA encoded/decoded (or from many other HW noise reduction systems -- not just DolbyA), and you see all of that 'nice' above 20kHz energy....  Not all of that energy comes directly from the music, but at least some (if not most) comes from the noise reduction system.

The kind of distortion that is created by DolbyA appears to be a softening of the audio (that is the sense of the distortion.)  The 'splats' don't help with the compatibility with equipment that doesn't deal well with excess HF energy.

 

John

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2 hours ago, gmgraves said:

While I do not doubt you in the least, I must say that I have never noticed this distortion on any of the recordings I made using Dolby-A. However the kind of distortion that you speak of; the "splats" as you call them, are very noticeable with FM broadcast compressors The one-time ubiquitous CBS Automax compressor (and the Fairchild compressors before it) had this sound in spades. Bob Orban of Orban Associates worked extremely hard when developing his original "OptiMod" compressor/limiter/multiplex stereo generator. He found (IIRC) that a fast attack and slow release cut down on this type of distortion (which in FM, is , of course, a single-ended compressor, and not a compander like the Dolby or DBX noise reduction systems). So, I am aware of the kind of distortion that you are talking about

The DolbyA distortion doesn't really sound 'bad', but tends to soften the sound -- and that is one reason why it hasn't been very aggressively fixed.  DolbyA encoded/decode CAN cause a lost 'precision' in the sound (gawd -- hate to use terms like that :-)).


Wrt simple anti-distortion mechanisms...  R Dolby acutally used a rather ingenius and simple one on the DolbyA (but it still becomes too fast), and that is that crazy diode network that also doubles as a rectifier.  (If you look at the 361/cat22 schematic and see the 4 JFET compressors -- the fancy diode network is on the right side of the compressor circuit.)   That circuit should be looked at SUPER carefully, because he actually made the attack slow for small increases in signal level, then speed up rapidly for fast increases.  It was a cool design with his very careful way that he finessed the log/exponential characteristics of the diodes, when the know-how was a little less common in 1965.

 

* One manifestation of DolbyA distortion is when there might be a chorus of two, three or four people, and somehow their distinct voices are softened into a chorus (it really isn't 3-4 clearly distinguishable voices anymore.)  That is one commonly manifested kind of distortion occuring on DolbyA.  The effect is similar to tape compression (which is a form of distortion), where it softens the sound in a mysterious way.  A good group to check for the distortion might be Simon & Garfunkel when they are singing clearly without much accompanyment.

 

WRT Orban -- I think that he is the one with the phasing patent.  I use a similar technology (but super different design) in my decoder to cancel out practically all of the distortion (it really doesn't get every last bit, but it does amazing things.)   I always avoid infringing on patents, however I really didn't consider his approach at all for what I have been doing.  (His thing is really cool also.)  Using his phasing concept can be helpful in hiding some of the IMD from fast limiters (and it actually implements part of the fast limiter.)  Doing what Orban (I do hope he is the person with that patent) did back in the 1970's was pretty good forward thinking also.

 

I am working very hard right now, trying to get a semi-final release ready.  Needed a break, so did some forum trolling (hopefully not in a negative way!!!!)

 

John

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