Is ASR right

Camverton

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Why can't Dirac correct high frequencies?
It can, but does require care in taking the measurements and creating the filter. If the sound from left and right channels is different due to room irregularities than dirac can correct this with the result that imaging is more precise and stable across the frequency range. Get it wrong and the sound is likely to be worse in that it is likely to sound rather dull with life being stripped from the music. 

It is much easier to use in the range up to 250 Hz and is very effective at dealing with room nodes.

 

Lawrence001

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Whereas I think it might be doing it well but I just prefer the effect of the room on the soundstage. Plus I've built my system around the room and it probably wouldn't be the same system if I had used Dirac from the start. Same speakers and probably amp, but I'd have ended up with different cables and DAC.

 

Camverton

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As for the SHD and fwiw, I think it is a very good value for money, very versatile item. It is not the finest quality sounding piece of equipment but then again that is hardly surprising in that it acts as DAC, preamp, crossover and has full range dirac included.  By and large the benefits, particularly if integrating a sub, far outweigh a small degradation in sound quality. I personally didn’t like the sound from its DAC, sounding rather unnatural, but feeding it into a Chord Qutest solved that for me.

On changing speakers and DAC it’s shortcomings, a small degradation in cleanliness of sound (assessed when no filters or dirac enabled) became a little bit more noticeable to me and I now use a sub, not crossed over but with built in adjustments to counteract a 60 Hz room mode.

In practice, it’s quality will depend on the partnering equipment, the expectations of the user and how necessary the included features are. Does the ASR review take these factors into account and quantify them?

 

Camverton

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Whereas I think it might be doing it well but I just prefer the effect of the room on the soundstage. Plus I've built my system around the room and it probably wouldn't be the same system if I had used Dirac from the start. Same speakers and probably amp, but I'd have ended up with different cables and DAC.
Absolutely, dirac isn’t an automatic panacea and it’s value will depend on a user’s circumstances. 

 

andreweast

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From the 'reviews' I've read on ASR, I can't see any details about the pscyho-acoustic correlative to their measurements, without which they are meaningless. They focus so much on THD, without having established the criteria for the audibility of distortion. 

They face something of a philosophical/methodological wormhole here, because you can't establish this criteria without large scale subjective testing, i.e. getting people in to see if this  distortion makes a difference to their aesthetic response.

As it stands, we don't have neurological implants to measure aesthetic response, so we'd have to rely on the thing that so called objectivists abhor - language. Specifically, the language of aesthetic response ('it sounds a bit grainy' etc). So it would a pretty vague and messy mass experiment anyway ('500 participants used the word 'misty' for DAC A, with distortion of X, and 300 participants used the word 'growly' for DAC B'.)

So as far as I can see ASR might be a useful resource for designers, but it can't have any use for aesthetic criticism and response. I'm happy to hear if they're more sophisticated than it seems though.

 
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ChemMan

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Even he seems to prefer too much bass and even a little BBC dip. And his listening room should be bloody awful judging from the level of boundary reflectiveness one sees in the photos...
But those beautiful Revels......... I'm lusting after a pair of the Salons or Studios.

Once I pay for them, I've no idea how I'm going to afford to create a listening environment to use them. :)

 
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andrew s

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From the 'reviews' I've read on ASR, I can't see any details about the pscyho-acoustic correlative to their measurements, without which they are meaningless. They focus so much on THD, without having established the criteria for the audibility of distortion. 
You could start here https://www.******************.com/forum/index.php?threads/audibility-thresholds-of-amp-and-dac-measurements.5734/

Regards Andrew 

 

daccordimark

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It's a shame Amir doesn't do more listening tests because as in the case of his reviews of the Lyngdorf TDAI-3400, the measurements he performed resulted in a less than glowing review, whereas once he actually listened to it and used the RoomPerfect DSP he actually liked the amp. All the negativity around the measurements went out of the window.

He does appear to have some influence over the likes of Topping as they, like car manufacturers with emissions tests, have realised that they can game the system by designing their units to pass his SINAD tests and generate a load of sales off his reviews. Likewise, miniDSP "improved" the SHD after his first review but whether it sounded any better is anyone's guess.

I do admit to spending a bit too much time on ASR sometimes, usually prompted by a friend who seems to have swung from outright subjectivism to hard line objectivism.

Mark.

 

newlash09

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Fair enough  :D .

I'll make three points should you feel any different today  :) .

  1. The ASR measurements were I believe exclusively for the SHD used as a DAC, and so don't apply if DSP is used as well.
  2. The SNR figures posted only apply for the SHD at maximum volume. If the volume is attenuated digitally by a large amount and then fed into a combination of a high gain amp and sensitive speakers then this may make the noise floor audible but doesn't make the the measurements wrong. Using an attenuator at the input to the amp can reduce/solve this problem by allowing higher digital signal levels to be used.
  3. I don't like the effects of applying Dirac Live filters over the full-frequency range, and your soundstage comment in particular read to me like what I find to be one of the negative aspects of doing so. I really hope that you tried only applying Dirac Live below something like 200 Hz, since this is what I think would have most likely given the results that you were looking for.

Where I personally stand on the SHD is that, for me, the improvements brought by DSP to control bass room-modes and subwoofer integration utterly dwarf any potential negative effects. I say this as I think it's very much a pros vs cons type of discussion where DSP is concerned, where different people will of course have different priorities.
Actually I just realised sir, that I had the minidsp SHD shipped to a friend, to hide the purchase from my wife :D

And he did give it a try in his system. I only collected it back after a few months, once I was back home. He did try dirac while it was with him. So I could have probably heard the minidsp SHD with any PEQ filters or dirac correction he might have enabled. My bad sir :(

So will reset it to default factory settings and give it a try. As I really want to love it and use it :D

 

MartinC

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Actually I just realised sir, that I had the minidsp SHD shipped to a friend, to hide the purchase from my wife :D

And he did give it a try in his system. I only collected it back after a few months, once I was back home. He did try dirac while it was with him. So I could have probably heard the minidsp SHD with any PEQ filters or dirac correction he might have enabled. My bad sir :(

So will reset it to default factory settings and give it a try. As I really want to love it and use it :D
Ah 🙂. You may reach the same conclusions but worth a revisit! If you start a thread or reply to the miniDSP owner's thread when you do then I and others may be able to help. 

In terms of a factory reset, this might result in some crossover filters applied. It does on the 2x4 HD but I'm not sure about the SHD.

 
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bencat

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I have never heard or used an SHD but on all my units (DDRC 24 and HD 10x10)) After a full reset I go in to each of the PEQ CR and other options and make sure they are all on by pass before I start on any new session . This then means as you progress through each parameter you are setting your desired levels before you then remove the bypass . So it also means that any other parameters that you perhaps do not use or are not needed in your setup are still in bypass mode and so not adding anything to the audio stream . Perhaps a bit OTT but then if you are on this site then you are most likely to be a bit fussy about things .

 

Lawrence001

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Actually I just realised sir, that I had the minidsp SHD shipped to a friend, to hide the purchase from my wife [emoji3]
And he did give it a try in his system. I only collected it back after a few months, once I was back home. He did try dirac while it was with him. So I could have probably heard the minidsp SHD with any PEQ filters or dirac correction he might have enabled. My bad sir [emoji20]
So will reset it to default factory settings and give it a try. As I really want to love it and use it [emoji3]
Ah if it's got active correction for a different room that may explain the issue!
 
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Digital is one of those things were I like to informally blind test just to validate what I think I’ve heard.  I’ve been surprised a number of times by the result.  
 

In the past I’ve found I couldn’t hear any difference between a fairly expensive Chord dac and a cheap CA cd player, that a almost universally well regarded budget dac ruined my system, I’ve only ever heard transport differences with one dac, and dacs that had subtle flavours all sounded the same blind.  I don’t think I’ve ever heard differences due to output level, and I’ve been told that a massive difference I’ve heard through one amp but not through another could be down to rfi from the dac affecting the amp.  So I’m fully on board with the notion that competent dacs will sound the same, and I’m fully on board with the notion that sometimes they don’t.  I think you need to listen to them in your own system.

 

tuga

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As mentioned often, ASR is not just Amir's measurements and comments but also thrives from the participation of many experts in different fields related to Audio, engineers and designers/manufacturers.

One of them is James D. (jj) Johnston, who later this month will give a Zoom lecture at AES PNW:

https://www.aes.org/sections/pnw/

NOTE: anyone can attend this meeting regardless of their status as an AES member. 
AES MEMBERSHIP IS NOT REQUIRED.

Masking — What is it, and when does it happen?

Lately, there have been discussions about masking involving several topics, from "can I hear this instrument over that instrument," to "can I hear this at all." The answer to that lies in the phenomenon of masking, wherein the cochlear receptors, which have a total dynamic range circa 90dB, are actually 30dB receptors that are effectively gain-ranged. This means that a second signal that is very close in frequency to a stronger signal is most likely gone completely if it's 30dB down, and in some cases, gone at as little as 5.5dB or even 3.5dB lower than the masking signal.

ZOOM MEETING 6:00PM PST (UTC -8)
Tuesday, January 25th, 2022
.
James D. (jj) Johnston is Chief Scientist of Immersion Networks. He has a long and distinguished career in electrical engineering, audio science, and digital signal processing. His research and product invention spans hearing and psychoacoustics, perceptual encoding, and spatial audio methodologies. 

He was one of the first investigators in the field of perceptual audio coding, one of the inventors and standardizers of MPEG 1/2 audio Layer 3 and MPEG-2 AAC. Most recently, he has been working in the area of auditory perception and ways to expand the limited sense of realism available in standard audio playback for both captured and synthetic performances. 

Johnston worked for AT&T Bell Labs and its successor AT&T Labs Research for two and a half decades. He later worked at Microsoft and then Neural Audio and its successors before joining Immersion. He is an IEEE Fellow, an AES Fellow, a NJ Inventor of the Year, an AT&T Technical Medalist and Standards Awardee, and a co-recipient of the IEEE Donald Fink Paper Award. In 2006, he received the James L. Flanagan Signal Processing Award from the IEEE Signal Processing Society, and presented the 2012 Heyser Lecture at the AES 133rd Convention: Audio, Radio, Acoustics and Signal Processing: the Way Forward. In 2021, along with two colleagues, Johnston was awarded the Industrial Innovation Award by the Signal Processing Society "for contributions to the standardization of audio coding technology." 

Mr. Johnston received the BSEE and MSEE degrees from Carnegie-Mellon University, Pittsburgh, PA in 1975 and 1976 respectively.

 
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Lawrence001

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As mentioned often, ASR is not just Amir's measurements and comments but also thrives from the participation of many experts in different fields related to Audio, engineers and designers/manufacturers.
One of them is James D. (jj) Johnston, who later this month will give a Zoom lecture at AES PNW:
https://www.aes.org/sections/pnw/
NOTE: anyone can attend this meeting regardless of their status as an AES member.  AES MEMBERSHIP IS NOT REQUIRED.
Masking — What is it, and when does it happen?
Lately, there have been discussions about masking involving several topics, from "can I hear this instrument over that instrument," to "can I hear this at all." The answer to that lies in the phenomenon of masking, wherein the cochlear receptors, which have a total dynamic range circa 90dB, are actually 30dB receptors that are effectively gain-ranged. This means that a second signal that is very close in frequency to a stronger signal is most likely gone completely if it's 30dB down, and in some cases, gone at as little as 5.5dB or even 3.5dB lower than the masking signal.
ZOOM MEETING 6:00PM PST (UTC -8) Tuesday, January 25th, 2022 . James D. (jj) Johnston is Chief Scientist of Immersion Networks. He has a long and distinguished career in electrical engineering, audio science, and digital signal processing. His research and product invention spans hearing and psychoacoustics, perceptual encoding, and spatial audio methodologies. 
He was one of the first investigators in the field of perceptual audio coding, one of the inventors and standardizers of MPEG 1/2 audio Layer 3 and MPEG-2 AAC. Most recently, he has been working in the area of auditory perception and ways to expand the limited sense of realism available in standard audio playback for both captured and synthetic performances. 

Johnston worked for AT&T Bell Labs and its successor AT&T Labs Research for two and a half decades. He later worked at Microsoft and then Neural Audio and its successors before joining Immersion. He is an IEEE Fellow, an AES Fellow, a NJ Inventor of the Year, an AT&T Technical Medalist and Standards Awardee, and a co-recipient of the IEEE Donald Fink Paper Award. In 2006, he received the James L. Flanagan Signal Processing Award from the IEEE Signal Processing Society, and presented the 2012 Heyser Lecture at the AES 133rd Convention: Audio, Radio, Acoustics and Signal Processing: the Way Forward. In 2021, along with two colleagues, Johnston was awarded the Industrial Innovation Award by the Signal Processing Society "for contributions to the standardization of audio coding technology." 

Mr. Johnston received the BSEE and MSEE degrees from Carnegie-Mellon University, Pittsburgh, PA in 1975 and 1976 respectively.
This is actually quite interesting, and reminds me of something I noticed in my teenage "early audiophile" years. Baroque oboe and violin seen to merge together into a beautiful sounding combination whereas the modern equivalents remain seperate (and not as nice sounding!).

Not directly related to the above issue as they are at similar volume levels, but I assumed the sonic characteristics of the older instruments are much closer than the modern ones.
 

tuga

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This is actually quite interesting, and reminds me of something I noticed in my teenage "early audiophile" years. Baroque oboe and violin seen to merge together into a beautiful sounding combination whereas the modern equivalents remain seperate (and not as nice sounding!).

Not directly related to the above issue as they are at similar volume levels, but I assumed the sonic characteristics of the older instruments are much closer than the modern ones.
It's hard to tell what's causing your perception without measurements but you could be right. Was this with the period instruments vs modern ones playing the same score?

 

tuga

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Different instruments have a different timbre, they produce a different sound envelope even when reproducing the same fundamental note (the ADSR characteristics will be different).

Formas_Onda.png


1: four notes played by a tabla (indian drum) 2: three notes played by a french horn 3: one note made by a flute

 
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