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Audio Myths and Human Perception - Explored

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John Albright    287
On 7/31/2017 at 8:24 AM, tube fanatic said:

So, as an expert in room acoustics, do you place any value on the role of electronics in producing the desired musical experience?  And, if you do, please elaborate on the parameters of the electronics which have the greatest influence on the sound and explain why solid state equipment, in spite of minuscule distortion and near identical specs, can sound so radically different in a particular system.  I would like to be enlightened since I am only a vocal "amateur"!

 

Maynard

 

Not too many of us are using old Technics receivers, here.  Though I am, to drive KG2s in my music playing room.  ;)

 

However, first things first, if a tube amp has a +/- 3dB response with speaker K, but the room has a +/-20 dB response, which matters more to you?  Most of us surely have made some attempt at room acoustics, heavy drapes, soft furniture and rugs for me, but there is a limit sometimes due to appearance.  The X got the custom engineered theater room in the basement.  

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Marvel    2281
1 hour ago, John Albright said:

The X got the custom engineered theater room in the basement.  

 

I wondered about that... just didn't want to ask.

 

Bruce

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LarryC    233

The knowledge and expertise in this thread are spectacular IMO and well beyond my aging capabilities.  Rather than try to compete with the people here in singing quality, adjusting my Oppo player (205, but I wouldn't know where to start in doing that!), Let me suggest some DVDs that I like and may be a bit out of the mainstream:

  1. Monteverdi Vespers, DG DVD 073 035-9; a true masterpiece, perhaps the greatest before Bach, with an entire textbook devoted to it, by Jeffrey Kurtzman.  Gorgeous views of the interior of San Marco in Venice.  This must be listened to by following all the text and familiarizing yourself with the interesting instruments used in the early Baroque.  Other more advanced "filmings" exist, such as the Royal chapel at Versailles, but this is probably the best and most informative.
  2. Berlioz Damnation de Faust, ArtHaus DVD 102 023.  Great work well-led by Solti if you like his style.  Top-notch singing by von Otter, Keith Lewis, Jose Van Dam, Chicago S Chorus and Orchestra.
  3. Berlioz Les Troyens by the Met, Levine conducting -- spectacular recording and performances, staging, and leadership.  A very great opera from the 19th century, fully as long as Wagner's, and of course finely orchestrated.  DG 00440 073 4310.  Norman, Domingo
  4. Mozart Don Giovanni, conducted by Wilhelm Furtwangler, perhaps the greatest conductor of the 20th century!  He came at the wrong time, with the Nazis, and his most easily available recordings are on Youtube, but the clarity and greatness of his interpretations are unmistakable.  It's well worth sitting down with Youtube for an entire afternoon and evening, and going through one Furtwangler recording after another.  All monophonic 1930's, '40's and '50's German taped sound of course, but the music itself conveys wonderfully well without great concerns for technical reproduction issues.  I more and more listen through such technicalities to pay very close attention instead to THE MUSIC. 
  5. The DVD series of the Beethoven 9, by the Berlin Phil conducted by Claudio Abbado in Berlin and Rome.  This set really keeps your attention on the instruments, and the great, profound, gentility of this beloved conductor, may he RIP.  This is currently my favorite Beethoven 9.

Larry

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Chris A    1571
On 8/3/2017 at 6:01 PM, hsosdrummer said:

If you know of a good Blu-ray recording/performance of Le Sacre I'd greatly appreciate your letting us know about it.

 

On 8/3/2017 at 10:39 PM, Chris A said:

I've found a couple of prospects: one by the Mariinsky Ballet (including the Firebird) (DTS-HD MA) ...

I may go with the Mariinsky due to the ballets also being performed...I'll let you know how it goes when it arrives.

Wow...I'm gobsmacked.

 

I'd put this disc in the top 5 of all BDs that I own in terms of audio quality.  The presence of the recording is extremely high, including the apparent depth and width of the sound stage and the retained dynamics, the closeness of the miking (i.e., it's not like a DG "dark recording"), and will certainly give your setup a workout (note: I'm also referring to extremely deep bass drum transients and very high frequency wind and percussion attacks).  It's also quite riveting to actually see the ballets instead of imagining the stage performance while listening to music only.  It gives meaning to performances that might have led my attention to wander without the ballets.  I have to say that it's never been a disappointment to hear Russians playing a Russian composer's music--it's always a breath of fresh air to hear how they interpret and the strength of their performance--with great balance between power and restraint.  No sentimentality there.

 

The Firebird started with a lot of audience background noise, and you'll not forget at any time that the audience is there...and that's okay since that is the way that it is in real life.  I've never connected with the Firebird music performance like that one.  The ballet is a traditional ballet but performed with power.

 

The Rite of Spring (Le Sacre du printemps) is actually a shorter, earthy piece by comparison that is punctuated by great dynamics and much less emphasis on melodies/dance.  It should be that way because the ballet is a story of spring rituals and human sacrifice by a primitive culture..."earthy".  The camera work would probably work a little better zoomed out a bit but the music is spot on, just like the Firebird performance.  Whomever did the mixing/mastering preserved the power of each performance.  I'm impressed.

 

Chris

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Khornukopia    442
4 hours ago, Chris A said:

the mixing/mastering preserved the power of each performance.  I'm impressed.

I just placed an order for the Blu-ray. I enjoy attending ballet performed with a live symphony, so I am looking forward to bringing this highly recommended recording into my home theater.

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Chris A    1571
On 8/5/2017 at 9:31 AM, etc6849 said:

It would appear as you increase the S/N ratio and improve IMD, it makes the hidden stuff much easier to hear on these types of tracks (e.g. background instruments and the little details still present even on the loud tracks).  I'm not saying these are the only parameters, but it would seem they are far more important than I previously thought if your goal is to get every last detail out of a so called "squashed" track.

Ellery,

 

I actually believe that the major effect is the "clarity" measure as described by Griesinger, above.  While you might be paying close attention to SNR, I personally believe that it's more likely the major effects of decreasing the relative phase distortion at high frequencies, as well as those absorption panels that you had in your listening room that significantly reduce early reflections.  Those two effects together increase the clarity of presentation significantly.  When you look at a time-based envelope of the reproduced track re-recorded in room, the little "spikes" as Griesinger described them are the detail that the human hearing system (specifically the organ of Corti) uses to detect perceived dynamic range.

 

Chris

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hsosdrummer    156
23 hours ago, Chris A said:

I'd put this disc in the top 5 of all BDs that I own in terms of audio quality...

Thanks for your recommendation, Chris! I've added this to my Amazon cart.

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Khornukopia    442

My disc purchase from Amazon arrived already. The Mariinsky Orchestra and Ballet present a spectacular performance of The Firebird and Le Sacre du printemps. I selected DTS HD Master at first and switched over to PCM stereo. The Blu-ray recording is really good either way.

 

IMG_0747.JPG.f37c62449ee263bac364a09828441d88.JPG

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mikebse2a3    437

More on myths and human perception and also how it relates to the use of Equalization.

 

"For a human, frequency is time-dependent"

 

See the story of MR A, MR B, MR C and Mrs D

 

Here is the whole paper which is well worth your time if you are interested in how both the subjective and objective approach is needed to advance in sound reproduction : on_room_correction.pdf

 

I'll also post the pages the story is on for anyone that doesn't want to download the whole pdf but again I suggest reading the whole paper which contains very good information in many areas relating to "faithful sound reproduction" in listening rooms (such as how we perceive reflections differently depending on direction such as front and back walls versus side walls for one example).

 

Bottom line is use equalization responsibly...!!! The key is to learn how it can and can't be used to improve any sound reproduction system in our listening rooms.

 

miketn

 

598e43dc283bb_TheConceptOfFrequencypg4.thumb.jpg.3343775ce6cb417bf795acbdb032af3e.jpg

 

598e47eb40beb_TheConceptOfFrequencypg5....thumb.jpg.17c86343e363665cf9c0c4ee5f20d190.jpg

 

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Chris A    1571

It seems that the wheels fall off on page 11...

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mikebse2a3    437
2 hours ago, Chris A said:

It seems that the wheels fall off on page 11...

 

I would be interested in your thoughts on this.

 

miketn 

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Chris A    1571

It's difficult to single out just one argument out of that paper without leaving the bulk of the readership here behind. 

 

The comment that I made about "pg. 11" is related to its author stating that "early reflections help".   If you like to listen to Bose 901s, then that comment on early reflections will be acceptable to you.  However, if you listen to K-402s in a well-treated room, I think that the author just shot his credibility with that statement.  I see that the real measure of merit in this instance is not envelopment--but rather clarity. 

 

Envelopment is easy to get from the recordings themselves if using loudspeakers with good directivity control--instead of trying to produce envelopment artificially using loudspeakers that splash acoustic energy around the room.  If the recording doesn't have good envelopment, then it will sound thin and piercing.  If the recording engineer took pains to capture envelopment, then when reproduced inside rooms with good directivity control, envelopment will be there. 

 

Inside rooms, clarity always decreases at the frequencies where directivity is lost and starts to reflect off of nearby walls, etc...getting worse as frequencies decrease.  Outside--away from walls or other acoustic reflectors--controlling directivity means little except greater ability to achieve SPL vs. frequency at an area in front of the loudspeakers. It's clarity that determines the bulk of audiophile tastes, in my experience.

 

I've found that small loudspeakers that lose midrange and mid-bass directivity control (i.e., small direct radiating loudspeakers)...some designers will choose the cheap path of "envelopment broadening" by splashing a lot of acoustic energy off of nearby room objects.  The much more difficult path is one that uses directivity control, which preserves clarity and "micro-detail" that most audiophiles are looking for. "Coherence" is the key word.

 

Past that comment on early reflections, I can say that the above Johansson article is a little difficult to follow for the average audio aficionado--unnecessarily so.  I'm not sure what audience that the author was talking to: electrical engineers, audio enthusiasts with a little background in EE concepts of pole-zero plots, and mathematical subjects of infinite/discrete Fourier transforms, or trained acousticians, etc.  The absence of mathematics in the article seems as if the author is wanting to reach a larger audience, but the textual references to pole-zero plots, the more granular differences between minimum phase and linear phase filters, and "information-theoretic perspectives" all tend to lose the broader audience, I think.  Once you lose your audience, they're gone.  So while I could dive into the specifics that the author makes, I'm reasonably sure that the discussion here is going to be limited to perhaps a handful of people that might understand the intersecting concepts that the author is pulling from two or three technical domains.

 

Chris

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Chris A    1571

Perhaps a gloss of the author's text would be useful to some struggling to follow the discussion, i.e., if you already understand the text in its entirety, then this gloss isn't needed to understand...

 

Quote

In this note I discuss some issues in filter design for equalization of sound systems. The emphasis is on rationale, not on experiments, and I will focus on a few common misunderstandings. I will briefly describe the basic concepts used in sound equalization, such as FIR and IIR filters, minimum and linear phase and present basic mathematical facts as well as give a background to the philosophy behind the Dirac Live approach. To limit the length of the text, I assume some basic familiarity with the topics covered. I will however refrain from the popular trend in some engineering periodicals of hiding bad ideas behind complex-looking equations.

Okay, so his article is defending the use of linear+minimum phase filtering in room correction software (or loudspeaker correction alone, as the case might be).  These are typically implemented by the use of IIR (infinite impulse response) and FIR (finite impulse response) filters, respectively, of which analog and digital representations exist for both types.  Analog FIR filters, however, are quite rare nowadays.  For the duration of this article, the author is referring to the digital implementations...so

 

FIR = linear phase and

IIR  = minimum phase

 

...for this discussion.

 

The idea of minimum phase filtering is common among anyone that has dealt with crossover filters--passive or active, analog or digital, and is what is commonly referred to as a "filter".  When these type of filters are applied to loudspeakers or in the signal chain, they not only change the relative amplitude or loudness of certain frequencies, but more importantly, they also change the relative phase of those frequencies, to be either leading or lagging the frequencies just outside of the filter's frequency bandwidth.  For all-pass filters, only phase is affected, not amplitude.

 

Quote

FIR AND IIR FILTERS
The output of an FIR filter is simply a weighted average of the N most recent input samples. The filter order, or length, thus indicates the memory of the filter. An IIR filter does the same thing as an FIR filter, but it adds a weighted average of old output samples to the first average.

This is the described difference between FIR and IIR filters: a IIR filter is recursive (i.e., it uses past historical values in addition to the input value at the moment  to determine their output), while a FIR filter output is irrespective of what has come before--it's like a waterfall--signal comes in, gets filtered around, and exits in some new pattern.  IIR filters can be unstable in their implementation--hence their name: infinite...for infinite response.  The author introduces the term "biquad" to refer to the building block of second-order IIR filters used in digital filtering.

 

Quote

...it is actually impossible to construct an IIR filter with linear phase

An important note: when you use IIR filters (analog or digital)--they change the phase in their region of interest.  For crossover filters, the phase change applies across all frequencies seen by the acoustic driver, either a lead or a lag.

 

Quote

When designing an optimal IIR filter in the sense of minimizing the expected squared error between the filter response and a desired response, the equations turn ugly. It is a non-linear problem, and it becomes mathematically messy to find the right response...

The idea of "optimal" really hasn't been defined here yet--and is left to the reader's imagination for the present.  Suffice it to say that you can design filters to "undo" nonlinearities in the signal chain, including the loudspeakers and room acoustics--as in what this guy is doing for profit: room correction software/firmware.

 

Quote

...On the other hand, if our processor budget only allows a low-order filter (maybe 10 biquads or 20-40 FIR taps) then impulse response correction is not possible, and then it is wiser to focus on the magnitude response and use minimum-phase biquad filters.

So this discussion is now considering the "real time" aspects of digital processing.  DSPs or general-purpose microprocessors have clock speeds and sets of available instructions for the user to program.  If the user programs too much into the processor, it can't complete its filter calculations within each time step, and the processing fails to do what it was intended to do.  So the software/firmware has to be carefully constructed to not overrun the processor's capability to complete all calculations within the time step.  Typically the time step is the inverse of the sampling rate of the digital room correction processor. 

 

The idea of impulse response comes from the convolution integral--a concept from the mathematics of signal processing.  It is also the same technique used by "chirp" or FM radar, and Vibroseis "earth shaker" geophysical vehicles to convert long swept sine wave signals into their equivalent impulse forms by collapsing phase down to zero phase for all frequencies.  It's how REW works, as well as most other room correction software, like Dirac. 

 

It's the idea that you can look at what it takes to form a perfect impulse response from the recorded return from the upsweep, collapse the phase down to an equivalent impulse or spike, and do what is necessary to make is a "perfect" impulse--this is the idea of an optimal (FIR) filter.  It allows for higher effective input of energy into the signal without having to generate and transmit all of that energy all at once.

 

I'll take a break here...

 

Chris

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Ski Bum    307

Chris, that "envelopment broadening" you mentioned a couple posts up is exactly what Toole has extensively covered in his books and articles on reflected sound, perception, and preferences.  Seems that those of us who prefer 'less room' and to let the recordings paint the picture are in the minority.  Toole acknowledges our existence, but doesn't shine the light on the approach as much as he does on the direct radiator types the masses prefer.  And it's kind of ironic, since you're absolutely correct that "envelopment is easy" with the right method.  

 

I still have tons of respect for Toole and his willingness to aggregate all the research.  For the Toole fans, or just those who want to learn more about acoustics, there is a new edition of his book coming out, if it's not out already.  (Seems relevant given the thread title.  Great reference, full of myth-skewering and pragmatic info, even if those of us who use horns and controlled dispersion speakers have to kind of determine what best applies to us.) 

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Chris A    1571
Quote

At this time we must also discuss the concept of a minimum phase system. This is a very peculiar system that has the least energy delay of all systems with a given magnitude response. (In the complex plane a minimum-phase system has all zeros and poles inside the unit circle.) It is very important to emphasize that this does not in any way mean that the impulse response is the shortest of all systems with that magnitude response. It means that it is earliest; the difference is profound. That it is early is a useful feature in applications where latency needs to be kept to a minimum.

This is probably where most people will check out in terms of trying to understand the concept of what the author is saying.  He could have also said the above several different ways, for instance:

  1. If you hit any system with an impulse signal, the signal that you get out of it (assuming this idea of a minimum phase system) is all you need to know about that system in order to correct for its nonlinearities, i.e., the output doesn't look like to the input, and the filters necessary to correct the output of a minimum phase system to look like the input (its input amplitude and phase...except for a constant delay) is the "optimal filter". 
     
  2. When you have a system--like a loudspeaker being measured for SPL response in an anechoic chamber--the behavior of that system under a swept sine wave driving signal can be mathematically reduced down to an equivalent impulse response (i.e., the inverse of the approach seen above).

    If you then test the loudspeaker with an impulse input, the resulting output of the loudspeaker will be equal to the computed impulse response using the swept sine wave input signal to produce the equivalent impulse response, i.e., a minimum phase system. 

    If you however put the loudspeakers in a real room somewhere and perform the same test, the two responses (i.e., computed impulse response and actual impulse response) will not be the same, i.e., a non-minimum phase system.  You have a mixed-phase system, and your equalizer can't be used to correct for all of the more complex behaviors in a simple way. 

 

Latency is always an issue, especially with FIR filters that are trying to correct frequencies below say, 500 Hz.  Now you have to wait at least the period of the acoustic wave (in this example, that's 2 milliseconds) before you get an output from the FIR filter(s) designed to correct for the nonlinearities.  If you try to move that FIR response down to, say 50 Hz, then you would have to wait 20 milliseconds to get any output.  That means that your digital room correction hardware with need at least 20 milliseconds of data and length of FIR filters (number of "taps" equals the sampling rate times the total delay).  That means that you'll need 1920 taps on your FIR filters (assuming 9600 samples/second), and you will also need to delay any synchronized video (TV or projector) by that amount.  On stage in live performances, the musicians or actors will have to deal with audible delays that tend to disrupt their performance.  It's the same problem that cathedral organists must train themselves to ignore--and it's terribly difficult to do that.  Even E. Power Biggs regularly succumbed to the effects of those delays during his own recorded performances. 

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Chris A    1571
1 hour ago, Ski Bum said:

Seems that those of us who prefer 'less room' and to let the recordings paint the picture are in the minority.  Toole acknowledges our existence, but doesn't shine the light on the approach as much as he does on the direct radiator types the masses prefer.

Toole is still selling loudspeakers, it seems, perhaps to assure his retirement pension keeps showing up at the end of the month, perhaps because he preached envelopment over clarity for so many years and doesn't want to invalidate those years of work, or perhaps he actually believes it. 

 

This is where I have always diverged from his writings.  However, Toole has written about the effects of horn-loaded loudspeakers:

 

Quote

(pg. 382--talking about the JBL Everest loudspeaker development):

 

...After EQ adjustments, the loudspeaker sounded as it (finally) looked: very good. If there was a problem, it was a tendency to play it very much louder than is commonplace with consumer loudspeakers. That is one of the seductive characteristics of loudspeakers that do not power compress or distort at high sound levels; they don’t sound loud until they are dangerously loud.

 

"Dangerously loud."

 

I still chuckle at that passage from his book. Toole really couldn't afford to acknowledge that directivity (clarity) was preferred by audiophiles, because he would be throwing darts at his own products if he did. 

 

The same issue (decision bias) also ruled with Villchur and Allison when testing for the audibility of loudspeaker modulation distortion (FM only)--"On the magnitude and credibility of FM distortion in loudspeakers", JAES, 30, No.10, (1982)...which actually got the wrong answer.  See https://www.stereophile.com/reference/1104red/index.html.

 

Chris

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Chris A    1571
Quote

A well-known theorem from complex analysis states that any linear time-invariant system can be factored into two systems: a minimum-phase system and an all-pass system. If we design an equalizer that perfectly inverts the minimum-phase system we have thus made the magnitude response flat...

This is what I was seeing with demastered tracks: as the applied EQ gets the overall time-averaged frequency response flat, so goes the phase response...the phase gets corrected, too, and that means that the clarity of the track goes up (as per Griesinger's presentation).

 

Quote

...but what has happened to the impulse response? On occasion, I have heard competent people say that we have at least not made it worse, since we took care of the minimum-phase part (the inverse of a minimum-phase system is itself a minimum-phase system and since there is only one for any magnitude response we have automatically taken care of it). The fallacy of this statement is obvious: the all-pass system may have a much worse impulse response than the complete system. If you take away the minimum-phase factor you may have taken away the features that removed severe ringing in the all-pass factor. The factorization we chose is completely arbitrary and has no physical interpretation at all. We could just as well have chosen a different factorization.

This is more obscure in my estimation.  I'm not sure why the author inserted this, except to address some argument that is not represented in this paper that deals with different automatic room EQ approaches--using superposition principles to "factor" the calculations.

 

Quote

[The difference between the Fourier representation and the perceptual impression] has some important implications for sound equalization. When we read our magnitude response estimates, we use a very simple estimate of the perceived spectrum, completely suppressing the concept of time. For example, take a minimum-phase impulse response and reverse it. The former starts at time zero and decays with some time constant until it dies out; the latter has instead a substantial pre-ringing but no post-ringing. Now all research that has been carried out on human perception of transients say that pre-echoes and post-echoes sound completely different to us. Yet, the magnitude responses are identical. Of course, this is also manifested if we say something and then play it back in reverse. Both samples have the same Fourier magnitude responses. This illustrates that phase responses, or impulse responses, do affect the perception of sound, even for a monaural source.

No arguments here on this, but note that Fourier transforms (or the discrete FFT as currently used in practice) aren't used for time domain analysis, but rather understanding the frequency response magnitude and phase over a time window.  Why would the arguments include time here, unless pre- and post-ringing is the result of "over-equalization" by room correction software?

 

Quote

There is obviously some threshold of how phase sensitive we are, but the literature on this matter (which is extensive, starting from the 1930’s) has concluded that this threshold, or integration time-constant, is adaptive and varies with what we listen to. What we can say for sure is that we do hear absolute phase but that the higher the frequencies the less sensitive we are (obviously, as the wavelengths become very short and the physics of wave propagation dictates that very little relevant information can be transmitted acoustically at high frequencies due to the chaotic behavior of high-frequency acoustic transfer functions). This implies that a good equalizer needs to consider also phase, not just magnitude.

Now this is a mouthful.  Do I think that humans hear "absolute phase"?  No, I really don't.  I believe that the author has missed the Griesinger concept of relative phase fidelity for clarity, especially at high frequencies.  This text is the reason why I've waded through the author's words up to this point--to highlight the importance of relative harmonic phase and its effects on clarity--which this author has apparently missed...or at least has not chosen to talk about as relevant at this point in room correction software.

 

I also see no real discussions on the use of Hilbert transforms and "excess phase" (i.e., the "all-pass filter in conjunction with a minimum phase filter").  Perhaps there is something that the author believes invalidates this concept, like the discussion on violation of linear superposition of filters.  Admittedly, I have not dived deeply into this subject area, but I can say that excess phase calculations/measurements have been extremely useful and apparently accurate in interpreting REW results.

 

Do I think that room correction software equalizers need to be "linear phase"?  Probably not, if we're really predominantly correcting for minimum phase magnitude errors in loudspeakers at higher frequencies (i.e., well above the Schroeder frequency division of sparse LF mode region and everything else at higher frequencies in room).  [Note that I'm not talking about the type of equalization filters used in mastering music--which is an entirely different animal.]  I'm not sure that the author has identified and distinguished between LF and minimum phase correction of loudspeaker response issues, or if he has done so to a sufficient degree.

 

Overall, I have to say that the author is defending mixed filtering concepts for room correction software, but likely inverted from that which I see is needed looking at REW data: you really want to use FIR filters at low frequency to control phase growth, and minimum phase filters at higher frequencies to correct for minimum phase defects in loudspeaker performance.  It seems to me that the author could be arguing the reverse--if I understand what he is saying correctly.  Using FIR filters at low frequency incurs very large delays, however.

 

Chris

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mikebse2a3    437
10 hours ago, Chris A said:

The comment that I made about "pg. 11" is related to its author stating that "early reflections help".   If you like to listen to Bose 901s, then that comment on early reflections will be acceptable to you.  However, if you listen to K-402s in a well-treated room, I think that the author just shot his credibility with that statement.  I see that the real measure of merit in this instance is not envelopment--but rather clarity. 

 

Here is what the author said:  He has clearly noted this as a "special case, and it is clear that most reflections will simply diminish sound quality."

 

5990a1fbd8f18_reflectioncomments.jpg.287b835b4accbf4eea0942e5658dc1cd.jpg

 

 

 

Chris I do not read the author as you do he is referring to the fact that in "some" instances reflections can help in "speech intelligibility" and in some acoustical environments this has been reported by other respected researchers that I have read. I can tell you I have experienced this personally at work where in large spaces it can be difficult to here the person speaking but if you move around in that space sometimes the reflections can help the mind to fill in the gaps leading to a perceived clarity not heard clearly through the direct path.

 

 

10 hours ago, Chris A said:

Envelopment is easy to get from the recordings themselves if using loudspeakers with good directivity control--instead of trying to produce envelopment artificially using loudspeakers that splash acoustic energy around the room.  If the recording doesn't have good envelopment, then it will sound thin and piercing.  If the recording engineer took pains to capture envelopment, then when reproduced inside rooms with good directivity control, envelopment will be there. 

 

Hmmm... Envelopment from 2 channel reproduction always requires room involvement in my experience. The sound field from 2-channel reproduction is always (with exceptions to some processing manipulation of some recordings) from the plane formed between the 2 loudspeakers with depth perception behind that plane. (ie: a window on the performance). 

 

Developing Envelopment in 2 channel reproduction means to me that I've been able to remove the effects from interfering reflections causing (despite the actual small physical space of the room I'm listening in) the perception/deception of being in a much larger space than I'm physically in and this is perceived as more relaxing/natural experience with the width and depth of the original sound field recorded preserved within this space.

 

The loudspeaker directivity can be used of course to advantage in small rooms especially but that in itself is insufficient to giving the perception of envelopment in my experience.

 

I will also say that the perception of envelopment and clarity are not exclusive of each other when done properly in my experience.

 

I still consider this to be a very good article and my experiences agree with much of what I believe the author is trying to tell us.

 

Wow Chris you hit me with a fire hose of information/thoughts....  :D   and I will try to respond more to all you have posted over time.

 

miketn

 

 

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Chris A    1571
26 minutes ago, mikebse2a3 said:

I can tell you I have experienced this personally at work where in large spaces it can be difficult to here the person speaking but if you move around in that space sometimes the reflections can help the mind to fill in the gaps leading to a perceived clarity not heard clearly through the direct path.

Large spaces, but not small ones...and I understand this comment.

 

26 minutes ago, mikebse2a3 said:

Hmmm... Envelopment from 2 channel reproduction always requires room involvement in my experience.

If you talked with Ellery (etc6849) and looked at his (prior) room, I bet you would be surprised to find that he has basically taken the room out of the running...in other words, his experience is the opposite of what you are reporting.

 

I have personally experienced positive effects of farther-field reflections from the back of the room (40 feet away), but I attribute some of that to sitting in the near field at those frequencies (i.e., low frequencies).

 

26 minutes ago, mikebse2a3 said:

I will also say that the perception of envelopment and clarity are not exclusive of each other when done properly in my experience.

I've found that near-field reflections and clarity are inverses of each other, as described in Griesinger's presentation.  "Properly" seems to be a bit mushy.  Please explain further what you mean.

 

26 minutes ago, mikebse2a3 said:

I still consider this to be a very good article and my experiences agree with much of what I believe the author is trying to tell us.

Well note that I'm not into automatic room correction software in any form because my experiences have been uniformly awful. 

 

Perhaps the software that I've used made extremely poor choices due to the highly directional loudspeakers (Jubilees, as you know) and partial LEDE absorption at the front of the room.  Most consumer loudspeakers and rooms are not arranged in this manner, and therefore the choices made by Audyssey, et al. are better choices than doing nothing at all in those rooms.  But in my room, doing nothing at all is preferred to any of the automatic room correction software.  YMMV.

 

Chris

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