High-Resolution Audio Listening Event at CE Week 2014 - Page 7 - AVS Forum
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post #181 of 257 Old 07-06-2014, 08:35 PM
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Originally Posted by arnyk View Post
The upper wave is not properly dithered, and is therefore represents very substandard performance as related to modern digital hardware.

The lower waveform shows the beneficial effects of dithering which is a standard component of every properly-designed contemporary digital system.
They are both the same hardware Arny so if one is broken, the other is too. And both signals are undithered as it clearly says below them:



The only difference is that the top graph is 16 bits and the lower 24 bits. The signal is down to the last 1.5 bits of 16 bit signal so it shows almost a binary/square wave as there is essentially no resolution left in the system. The lower graph of course has 256 levels so it is able to show the sine wave.

We see one of the problems with digital in that people are so unfamiliar as to how its performance is characterized. Understanding requires a ton of signal processing knowledge which is not something you pick up by googling or reading forum posts.

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post #182 of 257 Old 07-06-2014, 09:28 PM
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Quote:
Originally Posted by krabapple View Post
If people are going to cite Fielder re dynamic range requirements, can they *at least* do the due diligence of citing his *more recent* (as in from 1995, not from the *start* of the digital era, for pete's sake) review of the topic?

http://www.aes.org/e-lib/browse.cfm?elib=7948

It's even available in full text on the web, if you google it.
Didn't realize you had joined my side of the argument Krab . Since everyone can read that abstract, here are two snippets from the conclusion section of the paper:



As you see, he has actually up the numbers from the older papers.

I think Krab is hanging his hat on the abstract that says current gear falls short of this dynamic range. But read the last line and you see that as of 1995, we were 5 db shy of meeting that spec. I am OK with that even if there has been no improvements in 19 years since .

The paper by the way which I have quoted extensively in the past, is super detailed and leaves no stone unturned. It goes through every part of the recording and playback chains and identifies their perceptual performance range. It directly disputes all the contrary claims in this thread on equipment noise, dynamic range of music, speaker distortion, etc., etc.

Last but not least, if by googling Krab means that awful poorly formatted version of the paper without graphs, then I suggest not bothering to find it. Pay the few dollars and get the actual copy.

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post #183 of 257 Old 07-07-2014, 12:37 AM
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Quote:
Originally Posted by Scott Wilkinson View Post
The event coincided with an announcement that the CEA, DEG, The Recording Academy, Sony Music Entertainment, Universal Music Group, and Warner Music Group had developed a formal definition of high-resolution audio (HRA)—"lossless audio that is capable of reproducing the full range of sound from recordings that have been mastered from better than CD quality music sources." In addition, four "descriptors" were introduced to specify the provenance of HRA or so-called "master-quality" recordings:

MQ-P
From a PCM master source 48 kHz/20-bit or higher (typically 96/24 or 192/24 content)

MQ-A
From an analog master source

MQ-C
From a CD master source (44.1 kHz/16-bit content)

MQ-D
From a DSD/DSF master source (typically 2.8 or 5.6 MHz content)
This is good to hear.

It would have been good to have some academics present at this event, even if it was just the Princeton University physicist who was directly involved in the research.

Also, where were Neil Young, Jimmy Iovine and Dr Dre?!
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post #184 of 257 Old 07-07-2014, 12:52 AM
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Originally Posted by imagic View Post
I'd love to hear some... Lil' Wayne... while sitting in a fine studio.
Can you imagine 'A Milli' and 'Lollipop' on those big speakers?
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post #185 of 257 Old 07-07-2014, 12:54 AM
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Quote:
Originally Posted by amirm View Post
They are both the same hardware Arny so if one is broken, the other is too.
Too bad! Apparently you've never seen a piece of hardware or software that was sophisticated enough to have a switch for turning dither on and off.

I don't know about the version of Audition you have, but the version of Audition that I have has such a (software) switch, for example:



If I take an approximate -96 dB 24/96 signal and convert it to 16 bits without dither I obtain this:



and if I do it with dither:



Notice that the square wave appearance of the waveform is gone.
Attached Images
File Type: png cool edit conver sample type.png (10.0 KB, 121 views)
File Type: png undithered 2496 to 1696.png (28.2 KB, 123 views)
File Type: png dithered 2496 to 1696.png (33.4 KB, 127 views)
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post #186 of 257 Old 07-07-2014, 04:59 AM
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Attached you see the noise levels in the 24bit mosaic file from Scotts comparison challenge.
Mosaic1 shows the actual noise levels Mosaic2 shows that the later samples are nulled...





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File Type: jpg mosaic-noise1.jpg (211.4 KB, 125 views)
File Type: jpg mosaic-noise2.jpg (212.3 KB, 123 views)

Last edited by Frank Derks; 07-07-2014 at 05:18 AM. Reason: attach pictures
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post #187 of 257 Old 07-07-2014, 05:43 AM
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Quote:
Originally Posted by Frank Derks View Post
Attached you see the noise levels in the 24bit mosaic file from Scott's comparison challenge.
There is a fade out. The low amplitudes are 100% artificial.

I find it interesting that the corresponding segment of the 16 bit file stops well short of going down to -93 to 96 dB (16 bit digital black), and the noise floor has a kinda odd looking spectrum.

If I do the conversion from 24 bits to 16 bits I obtain the expected result - digital black @ -96 dB.
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post #188 of 257 Old 07-07-2014, 06:22 AM
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Quote:
Originally Posted by arnyk View Post
Too bad! Apparently you've never seen a piece of hardware or software that was sophisticated enough to have a switch for turning dither on and off.
Good morning Arny. As you said the topic of dither is not well understood.

Dither is applied *before* conversion of a higher resolution signal to lower (e.g. going from 24 bits to 16). If you add dither after the fact, you are just adding noise. It does nothing else. Its magic only comes from randomizing the bits (e.g. 24 bits) prior to conversion to lower bit depth (e.g. 16).

In the case of the test files that were posted, the incoming data was undithered on purpose. You can have any hardware you want but you cannot add noise to those signals and think you are applying dither. You are not.

The setting in the hardware you used likely had a dither option for its internal processing such *it* doing the conversion from 24 to 16. Or having digital volume control/mixer that would then need dither to work right. Such a feature would have absolutely nothing to do with incoming file that is preconstructed to have no dither.

Please post the make/model of the hardware you are talking about and I am confident I can show the above being true.

Quote:
Originally Posted by arny
In this case.

I don't know about the version of Audition you have, but the version of Audition that I have has such a (software) switch, for example:

That is because inside Audition you can keep your resolution high and then perform dither prior to conversion to lower bit depths. This is NOT the same as having a "dither switch" in hardware. If I feed the hardware 16 bit data, there is no high resolution version for it to add dither prior to reducing the bit depth.

Quote:
Originally Posted by arny
If I take an approximate -96 dB 24/96 signal and convert it to 16 bits without dither I obtain this:

Which is precisely what I explained in my post where I said the waveform in 16 bits is almost a square wave due to the fact that the difference between 98 db and 90 db test signal is just over one bit.

Quote:
Originally Posted by arny
and if I do it with dither:



Notice that the square wave appearance of the waveform is gone.
It better not be gone Arny or you would have lost your original signal! Fortunately if you step back some you can still see the waveform as it was without dither in your first post. What dither has done is to randomize the distortion. That is what the up and down jumping is showing. That randomization is limited such that the original signal is still preserved. The randomization is proportional to amount of distortion, not the amount of signal (i.e. music) we have.

So I am afraid as with the prior post I was responding to, we are seeing yet again the difficulty of understanding the digital system. It seems that digital is simple with "ones and zeros" but it is anything but. It is only simple when you have the files on your computer. How they came about and how they become analog audio is a deep and complicated topic full of mathematics.

So summarizing:

1. Both test signals that Ray had shown were without dither. This was clearly labeled below the graphs so there should be no argument about it. The files were intended that way as to create the precise waveforms intended for testing. Adding dither would have changed the waveform (i.e. randomize some of it) which would then make the stimulus unpredictable.

2. No hardware can have a switch to "add dither" to a signal that is already truncated to lower bit depth. Dither is just noise and adding it in the manner you are implying will simply raise your noise floor and do nothing to reduce distortion.

3. Any "hardware switch" for jitter would be for processing that the hardware itself performs, not for dealing with truncation of the file up stream. Once you truncate the bits without adding dither prior to that conversion, you have in essence "burned in" the distortion in the data stream. There is no way to now find and remove it and certainly not with a "dither switch."

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post #189 of 257 Old 07-07-2014, 06:32 AM
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Adding on, the type of confusion I was addressing is the precise reason why I like to see the original high resolution masters. If Arny is having a hard time understanding what dither is and how it is applied you can imagine what Joe music recorder/mastering person would know. They may easily think that dither is "adding noise" and hence is bad. Here is a quick search on Gearslutz which is the forum for pros in recording space: http://www.gearslutz.com/board/maste...-any-type.html

"So, I just started using Reaper this year. It is the first time I have found myself in a situation where I can apply dither as I listen and toggle it on and off.

The dither sounds like poo.

The sound stage just gets so much smaller."


There are countless threads like these. The result is that folks would release truncated files with no dither.

So all this talk about "dither" and "noise shaping" as a practical matter is moot. You have no idea who created your music and their level of knowledge to have created the "right" truncated bits. By getting the 24-bit files prior to mutilation, we are assured of the person not having added distortion due to misconceptions or improper use of the tool. If you think you know this topic well then you can get the 24 bit files and perform your own conversions down to 16 but the rest of will play it as is .

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post #190 of 257 Old 07-07-2014, 07:50 AM
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Quote:
Originally Posted by amirm View Post
So all this talk about "dither" and "noise shaping" as a practical matter is moot. You have no idea who created your music and their level of knowledge to have created the "right" truncated bits.
Interesting argument for higher bit rate - technical incompetence. Can't argue too strongly against that. I can relate some surprising stories involving major players, albeit in the video industry.
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post #191 of 257 Old 07-07-2014, 08:33 AM
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Quote:
Originally Posted by amirm View Post

2. No hardware can have a switch to "add dither" to a signal that is already truncated to lower bit depth. Dither is just noise and adding it in the manner you are implying will simply raise your noise floor and do nothing to reduce distortion.
Obviously Amir due to your lack of hands-on experience with digital audio, you don't know that the above is not what I am talking about. A knowledgeable person would know the appropriate steps to take to produce a proper word-length reduced file. The statement above is a false claim.

The proper steps for producing a word-length reduced file are:

(1) Add appropriate amounts of dither to the larger word length file.
(2) Then truncate it to the shorter word length.

Pretty easy, eh?

That's why they pay me the big bucks! ;-)

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Originally Posted by amirm View Post
3. Any "hardware switch" for jitter would be for processing that the hardware itself performs, not for dealing with truncation of the file up stream.
So what? At this point it is clear that a very simple point has been missed. I tried!

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Originally Posted by amirm View Post
Once you truncate the bits without adding dither prior to that conversion, you have in essence "burned in" the distortion in the data stream. There is no way to now find and remove it and certainly not with a "dither switch."
That's all completely true, and hardly any kind of news flash to those who are well-informed about digital audio processing. Circumventing that problem is really pretty easy as I showed above.
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post #192 of 257 Old 07-07-2014, 08:42 AM
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Originally Posted by amirm View Post
So all this talk about "dither" and "noise shaping" as a practical matter is moot.
Well noise shaping is moot for 16 bit files due to the lack of real world program material that exceeds the capability of 16 bits with flat spectrum TPDF dither. But you seem to be in denial about that, so I'll play along with your little kid's game.

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Originally Posted by amirm View Post
You have no idea who created your music and their level of knowledge to have created the "right" truncated bits.
That is true of any recording in so many ways that it is a near-infinite rabbit hole.

If you choose to go down that rabbit hole Amir, be my guest. Its the end of the discussion.


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Originally Posted by amirm View Post
By getting the 24-bit files prior to mutilation, we are assured of the person not having added distortion due to misconceptions or improper use of the tool.
The only way to have the kind of control you seem to want over the process is to do it yourself. That's common sense, which come to think of it is a dangerous proposition to bring up in any discussion with you, Amir. ;-)

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Originally Posted by amirm View Post
If you think you know this topic well then you can get the 24 bit files and perform your own conversions down to 16 but the rest of will play it as is .
Better than that, for about 12 years I had free access to as many live performances as I could stand to record. I've tried just about every permutation of processing. I applied technical tests to the products I produced at many steps along the way. In the end I found that will a little skill in terms of level setting, 16 bits was entirely sufficient to get the best results that were obtainable.
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post #193 of 257 Old 07-07-2014, 09:11 AM
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Quote:
Originally Posted by krabapple View Post
For those who worry about the miniscule distortion introduced by non-integer rate conversion (so called 'arbitrary sample rate conversion')

Why do you suppose DSD's SR was specified as an integer multiple of Redbook in the first place?
As 44.1's Nyquist limit has been suggested by some to be so close to audible frequencies, the commonly used 48khz gives a bit more breathing room. 88.2khz is an option that many appear to use but 96khz seems more popular. Some mention less than universal support for 88.2 as a drawback. As iMagic suggested, this may be a good time to stop using 44.1khz for file distribution in favor of 48 or 96 which of course are integer multiple related. As for the probable need to convert to 44.1 for CD distribution, from what I've read integer multiple sample rate conversions no longer have a real advantage by use of an intermediate upconversion (also mentioned in this thread) which can also allow for jitter reduction. Moving to 48 would also help unify music and Film/TV standards.
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post #194 of 257 Old 07-07-2014, 03:17 PM
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Originally Posted by TVOD View Post
As 44.1's Nyquist limit has been suggested by some to be so close to audible frequencies, the commonly used 48khz gives a bit more breathing room.
Fortunately, the above speculations are not true. With complex music signals the limit of audiblity is more like 16 KHz due to masking, and that is more than plenty far enough away from 44.1 KHz with modern digital filters.
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post #195 of 257 Old 07-07-2014, 04:10 PM
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Originally Posted by arnyk View Post
Fortunately, the above speculations are not true. With complex music signals the limit of audiblity is more like 16 KHz due to masking
Is that to say 16khz audio bandwidth is sufficient?
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post #196 of 257 Old 07-07-2014, 04:45 PM
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Quote:
Originally Posted by arnyk View Post
Obviously Amir due to your lack of hands-on experience with digital audio, you don't know that the above is not what I am talking about. A knowledgeable person would know the appropriate steps to take to produce a proper word-length reduced file. The statement above is a false claim.
Hi Arny. I can only go by what you are posting and this was what I was responding to:

Quote:
Originally Posted by arnyk View Post
Too bad! Apparently you've never seen a piece of hardware or software that was sophisticated enough to have a switch for turning dither on and off.
As I explained having hardware that has a dither switch does you no good at all. Yet this is what you said above.

Quote:
Originally Posted by arny
The proper steps for producing a word-length reduced file are:

(1) Add appropriate amounts of dither to the larger word length file.
(2) Then truncate it to the shorter word length.

Pretty easy, eh?
If you know what you are doing it can be easy but the hardware played no role in that conversion. Ergo your comment about hardware having a dither switch as to somehow aid with this was incorrect.

Quote:
Originally Posted by arny
That's all completely true, and hardly any kind of news flash to those who are well-informed about digital audio processing. Circumventing that problem is really pretty easy as I showed above.
The reference to hardware dither switch implied otherwise Arny. Since we all agree that you are a genius when it comes to digital signal processing, I thought it was important to help correct that misstatement as to not let anyone think you don't know the ABCs of signal processing/dither.

I also wanted to make sure there is no "sophisticated hardware with dither switch" that people need to go and get. Hardware that has such a dither switch as you implied would be the sign of someone who doesn't know what they are designing. It has happened by the way. Folks think dither is noise and it is the same whether you add it before or after conversion (the latter is what you implied). No one needs a dither switch in a DAC. Any implication that such a feature is necessary is sign of clear incompetence. Present company excluded of course .

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post #197 of 257 Old 07-07-2014, 04:48 PM
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Quote:
Originally Posted by TVOD View Post
Is that to say 16khz audio bandwidth is sufficient?
Earl Geddes (gedlee.com) I believe said that what happens in a speaker past 17 kHz or so is irrelevant. Not everyone shares his opinion. Risking an appeal to authority, I believe he's an AES fellow and presents at various technical conferences, those that know him say he's a jolly good fellow that nobody can deny.

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post #198 of 257 Old 07-07-2014, 05:04 PM
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Originally Posted by arnyk View Post
Fortunately, the above speculations are not true. With complex music signals the limit of audiblity is more like 16 KHz due to masking, and that is more than plenty far enough away from 44.1 KHz with modern digital filters.
Ah yes. I recall you saying this before:

Quote:
Originally Posted by arnyk View Post
I've done experiements with sliding brick wall filters down to lower and lower frequencies. Usually, they start being barely audible around 16 KHz.
Can you please outline those tests? Were they double blind or sighted? And was the person performing the tests had his hearing tested to make sure they could hear above 16 Khz?

And a reference to masking of everything above 16 KHz would also make for great bedtime reading for me .

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post #199 of 257 Old 07-07-2014, 05:19 PM
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Already finished 50 Shades of Grey?

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post #200 of 257 Old 07-07-2014, 05:20 PM
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Quote:
Originally Posted by TVOD View Post
Is that to say 16khz audio bandwidth is sufficient?
Yes. Take the best audio system you can find. Take the best recordings you can find - recordings that sound great and also have significant content > 20 KHz, even > 35 KHz. Switch a 16 KHz brick wall filter in and out of the signal path. Nobody notices nuttin'

People say: "But I can hear pure tones at 21 KHz". Probably true. But that is without content at other frequencies masking it. Music is composed of many tones at many different frequencies. Masking in the upward direction frequency-wise is very strong.
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post #201 of 257 Old 07-07-2014, 05:30 PM
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Quote:
Originally Posted by amirm View Post
Ah yes. I recall you saying this before:

Quote:
Originally Posted by arnyk
I've done experiments with sliding brick wall filters down to lower and lower frequencies. Usually, they start being barely audible around 16 KHz.
Can you please outline those tests?
I did what I said in the previous post. I made recordings of live musicans in an exceedingly quiet and non-reverberent room @24/96 using 1/4" measurement mics that had strong content > 20 KHz, even 30 KHz.

I set up an ABX between the 24/96 files with a 16 KHz brickwall filter, and with full bandpass. I used speakers and amps with strong response > 30 KHz and put the listeners on axis of their supertweeters.

Quote:
Originally Posted by amirm View Post
Were they double blind or sighted?
both

The training sequence was files brick walled at lower frequencies such as 9 KHz, and working up in logical steps.

Quote:
Originally Posted by amirm View Post
And was the person performing the tests had his hearing tested to make sure they could hear above 16 Khz?
Yes.

Quote:
Originally Posted by amirm View Post
And a reference to masking of everything above 16 KHz would also make for great bedtime reading for me .
I don't think its said as such very often. However it is well known among many lossy encoder writers that they can use a sharp roll off > 16 KHz without detracting from the transparency of their products. It must be, because that is how their products perform!
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post #202 of 257 Old 07-07-2014, 05:34 PM
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Quote:
Originally Posted by amirm View Post
Hi Arny. I can only go by what you are posting and this was what I was responding to:



As I explained having hardware that has a dither switch does you no good at all. Yet this is what you said above.


If you know what you are doing it can be easy but the hardware played no role in that conversion. Ergo your comment about hardware having a dither switch as to somehow aid with this was incorrect.


The reference to hardware dither switch implied otherwise Arny. Since we all agree that you are a genius when it comes to digital signal processing, I thought it was important to help correct that misstatement as to not let anyone think you don't know the ABCs of signal processing/dither.

I also wanted to make sure there is no "sophisticated hardware with dither switch" that people need to go and get. Hardware that has such a dither switch as you implied would be the sign of someone who doesn't know what they are designing. It has happened by the way. Folks think dither is noise and it is the same whether you add it before or after conversion (the latter is what you implied). No one needs a dither switch in a DAC. Any implication that such a feature is necessary is sign of clear incompetence. Present company excluded of course .
Back in the day there were audio interfaces with switchable dither and I owned several. As I pointed out in another post, that became less feasible with the transition to Sigma-Delta converters because in their usual configuration, the dither function is tightly integrated with the whole Sigma Delta package, and its not switchable.
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post #203 of 257 Old 07-07-2014, 09:52 PM
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I got bored with the Arny/Amirm show so I've started pestering Mark Waldrep about *his* NTS (No True Scotsman) argument

(It goes like this: 'Of course hi rez sounds markedly better than CD! Even my wife/kids/sheep can hear it! What's that you say? Downconverted SACDs and DVDAs are *really hard* to tell apart from the sources, if at all, in a controlled comparison? Well, that's because those SACDs and DVDAs weren't *TRUE* high rez recordings to being with! What's that you say? Audiophiles and high-end magazine reviewers have been raving about the 'high rez' sound of those SACDs and DVDa since day 1? Well, erm, ah, hmmm....')


http://www.realhd-audio.com/?p=3107#comment-139221

Last edited by krabapple; 07-08-2014 at 07:56 AM.
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post #204 of 257 Old 07-08-2014, 02:13 AM
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Quote:
Originally Posted by krabapple View Post
I got bored with the Arny/Amirm show so I've started pestering Mark Waldrep about *his* NTS (No True Scotsman) argument

(It goes like this: 'Of course hi rez sounds markedly better than CD! Even my wife/kids/sheep can hear it! What's that you say? Downconverted SACDs and DVDAs are *really hard* to tell apart from the sources, if at all, in a controlled comparison? Well, that's because those SACDs and DVDAs weren't *TRUE* high rez recordings to being with! What's that you say? Audiophiles and high-end magazine reviewers have been raving about the 'high rez' sound of those SACDs and DVDa to 'high rez' since day 1? Well, erm, ah, hmmm....')


http://www.realhd-audio.com/?p=3107#comment-139221

In the cited piece Waldrep pats himself on the back:

"The first track that I plugged in to the ABX Tester was the “On the Street Where You Live” by Steve March Torme. This track has a tremendous dynamic range and lots of ultrasonics. I know that the last 40 seconds contains the highest dynamic range so I concentrated on that section as I did the test. I listened to all 5 “X” items and made my choices."


So, is the final 40 seconds of “On the Street Where You Live” by Steve March Torme some kind of killer track?

Here is a comparison of my idea of the hottest segment of Street versus a segment of Sound Liason's (a competitive HD tracks site) version of Batik's "The Defeat":



The green lines show the spectral content of "The Defeat" and the red lines show the spectral content of Waldrep's Street. Above 5 KHz, "Street" gets "Defeat"-ed. ;-)
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Last edited by arnyk; 07-08-2014 at 07:47 AM.
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post #205 of 257 Old 07-08-2014, 05:05 AM
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Not to get lost in the Arny/Amir show, but as an aside, Sound Liason has some great sounding stuff.
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post #206 of 257 Old 07-08-2014, 08:37 AM
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Not to get lost in the Arny/Amir show, but as an aside, Sound Liason has some great sounding stuff.
But is that because it's 'high rez' specifically , or because its just well-played, well-recorded and well-mastered generally?
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post #207 of 257 Old 07-08-2014, 09:51 AM
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I have 3 of their recordings. The music is excellent, the performers excellent and the sound superb. Check out A Thousand Shades of Blue. There have that song as free download. Check it out for for yourself. I don't have a CD copy to compare the 24/96 file to. Don't care really, the 24/96 sounds that good and I didn't have to pay any shipping or wait to get it.
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post #208 of 257 Old 07-08-2014, 10:05 AM
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I gave a template for the No True Scotsman argument; here it is in practice -- Waldrep's reply. Emphases mine.



Quote:
Originally Posted by Mark Waldrep
AdminJULY 8, 2014 AT 8:36 AM


I has already substantiated my thoughts on this matter through numerous posts. I have clearly stated the criteria that I believe are necessary to qualify as high-resolution audio. The BAS or you or anyone else can obviously establish your own and measure things differently. Every item in the list above (which I’ve referenced previously) is not high-resolution based on the provenance or production path used to create them. Every single SACD can be ruled out because they were all done using DSD 64, which is very good in the “audio band” only as stated by the folks at SONY. This means that they have the same frequency response as a good CD. They may sound different but they are no better (in stereo) than a traditional CD as far a fidelity goes. The few items on the list that are DVD-Audio release ALL came from analog tape. This is the limiting factor with regards to the PCM tracks used.

You may not agree with my definition…and you are certainly entitled to your own opinion…but the works cited above guarantee that a realtime downconversion to CD quality would be no different than the “high-resolution” sources. And that’s why the BAS research is meaningless.

CD can do a really terrific job of capturing audio fidelity. In fact, I agree that the dynamic range is sufficient to meet the needs of virtually every recording available. There are some (I possess a number of them) that exceed those specs. I believe that the added octave and dynamic range is important and can make an audible difference in the sound of recording. I’m hoping to do a proper study and establish that fact. We’ll see.

Steely Dan's Two Against Nature DVDA, btw, was an all-digital recording. (Analog tapes were* also* made during production, for possible future release on 'vinyls' to....*audiophiles* ). On it there's obvious spectral content up to 24 kHz (suggesting it was a 48kHz SR recording) , and low-level noise all the way up to the 48 kHz.
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Last edited by krabapple; 07-08-2014 at 10:30 AM.
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post #209 of 257 Old 07-08-2014, 12:57 PM
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Take the best audio system you can find. Take the best recordings you can find - recordings that sound great and also have significant content > 20 KHz, even > 35 KHz. Switch a 16 KHz brick wall filter in and out of the signal path. Nobody notices nuttin'
Perhaps a third file with 16khz bandwidth should be created to AB with the other two.
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post #210 of 257 Old 07-08-2014, 06:06 PM
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I have read read the full thread, twice. Thank you for the informative/instructive depictions, Arny.

...And Amir for the valiant effort.
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