Stuart has written papers that show that 52 kHz / 11 bits are the absolute minimum PCM channel using noise shaping, "capable of replicating the information received by the ear". 14 bits "ought to be adequate" to offer enough headroom. That would be 18.2 bits for a rectangular channel (TPDF, no noise shaping).

Quote from: 2Bdecided on 2014-11-21 17:21:35

Quote from: amirm on 2014-11-21 17:11:16

Tell me again why you guys are arguing.

Again?! Isn't 18 pages of it sufficient?

I would think so and I am looking at ending my participation.  It is a shame though that I can't get people to agree with the simple business principal that high resolution downloads are happening.  And by downloading such bits I am free of worrying whether downgrading them is audible or not.

Quote from: amirm on 2014-11-21 17:52:42

And by downloading such bits I am free of worrying whether downgrading them is audible or not.

There is some logic in there. There is also some logic in capturing the microphone output as losslessly as possible, even if we can't hear it. DXD looks like a safe format for that, since most microphones don't have a bandwidth of more than 100 kHz. Higher-res downloads are more expensive and the claim is that the higher the bitrate, the better the sound quality. From older papers I got the impression that Bob Stuart seems to think that audible differences disappear above 24/96. Besides the satisfaction of "having the studio quality", what would be the point of buying higher formats ?

Some prices for an album in euro, from 2L.no , the provider of the samples of this paper:

stereo:
mp3      09.00
16/44.1   15.00
24/96   21.00
24/192      23.00
DSD64      28.00
DSD128    33.00
DXD          37.00
multichannel:
24/96        25.00
DSD64      33.00

(sorry, can't get tabs to work)

Quote from: amirm on 2014-11-21 17:52:42

And by downloading such bits I am free of worrying whether downgrading them is audible or not.

There is some logic in there. There is also some logic in capturing the microphone output as losslessly as possible, even if we can't hear it. DXD looks like a safe format for that, since most microphones don't have a bandwidth of more than 100 kHz. Higher-res downloads are more expensive and the claim is that the higher the bitrate, the better the sound quality. From older papers I got the impression that Bob Stuart seems to think that audible differences disappear above 24/96.

Besides the satisfaction of "having the studio quality", what would be the point of buying higher formats ?

stereo:
mp3      09.00
16/44.1   15.00
24/96   21.00
24/192      23.00
DSD64      28.00
DSD128    33.00
DXD          37.00
multichannel:
24/96        25.00
DSD64      33.00

(sorry, can't get tabs to work)

Quote from: amirm on 2014-11-21 17:52:42

Quote from: 2Bdecided on 2014-11-21 17:21:35

Quote from: amirm on 2014-11-21 17:11:16

Tell me again why you guys are arguing.

Again?! Isn't 18 pages of it sufficient?

I would think so and I am looking at ending my participation.  It is a shame though that I can't get people to agree with the simple business principal that high resolution downloads are happening.  And by downloading such bits I am free of worrying whether downgrading them is audible or not.

I have answered this to my satisfaction, if not yours, already.

There are any number of inaudible objective improvements one could make to the recording and reproduction process. Most could fall into the "they can't make it worse so why not" category. Add them together, and you have spent all your money and all your life chasing numerical improvements with little or no audible benefit. Meanwhile the real improvements have passed you by. That is why we have to let the inaudible improvements pass us by, and go and chase audible improvements instead.

So, why should we care about THIS inaudible imrovement?

Good morning amir,

Something about this little story has been bugging me a bit, perhaps you could enlighten us. 

I am asking why people here and other forums go crazy, absolutely crazy at the idea that someone goes and downloads higher resolution bits.  Why?  How is bothering them?  Read Krab's posts.  Or AJ's.  Or Arny's.  Or Xnor.  Or even our moderator.  These are not logical reactions to me going buying high resolution audio.  I know all of their arguments yet they keep repeating them.  As you do here .  Do they think I or others haven't hear them?  Why keep repeating it and of all people tell me about it?

Can you answer the psychology of this?  I mean I have this new guy Xnor in my life who is determined to say I have cheated.  Cheated!  How on earth did he decide it is appropriate to wake up one morning and decide his mission in life is to demonstrate that.  And to what effect?  What is driving him?  You know him better than I.

Culturally it is pretty far over the line to go there.  I would think he would do that if he thought I had done him some personal harm.  But to advocate getting the bits prior to conversion to 16/44.1?  That is all it took for him to write 12-quote replies now?  How did that become a priority for him?

I mean isn't it stupid of all people to take me on?  I take double blind tests.  I have built a career based objective science.  I am an engineer.  I talk about science and technology of audio.  And this is the person that Krab says is going to put on "ignore" list?  What is left to read then if it is not what I just explained.

Quote from: amirm on 2014-11-21 17:52:42

And by downloading such bits I am free of worrying whether downgrading them is audible or not.

There is some logic in there.

Some prices for an album in euro, from 2L.no , the provider of the samples of this paper:

stereo:
16/44.1   15.00
24/96   21.00

Is there any record anywhere, of you demonstrating this elite aural athleticism, at say, the fully documented listening Olympics, supervised, to your peers?

Quote from: Steve Burke link=msg=0 date=

Even Amirm has said that he cannot make this claim:

Surely you wouldn't imply that 640k tracks are indistinguishable from TrueHD, would you?

Amirm: "No I wouldn't"

Amir is also a golden-ears audio compression expert.

Transparency can be achieved by any of the right compression, content, and audience .

Digital Media Technology Insider with Microsoft

My compression blog[/color]

Or is the very first demo to the world earlier this year (2014), when you the ex-MS exec, sat at your Windows pc (possibly with your occasionally barking dog as the sole witness) and "passed" Arnie's corrupt ABX windows computer file ABX online test? And all similar subsequent ones of course.
Just curious, is all. Thanks.

Why do you come here AJ?

Quote from: amirm on 2014-11-16 21:20:35

FYI, I passed all of my double blind ABX tests using my laptop and headphones.

You probably think you're getting a hard time here, but you should take it as a compliment that no one has challenged you on this. The assumption must be that, given your employment history, you must know what you're doing. Anyone else who arrived on HA and made this claim would get a really hard time, due to a long history of drivers, sound cards, and transducers that don't take kindly to ultrasonic content. ABXing hi-res vs 16/44.1 on such crappy equipment is easy, and it's not because hi-res is audibly better.

Cheers,
David.

Are there some details on the system amir used? Which operating system/version, which headphones, which soundcard, which sound API, which DirectSound common sampling rate was configured (if applicable) ... ?

Quote from: ajinfla on 2014-11-21 14:42:15

Is there any record anywhere, of you demonstrating this elite aural athleticism, at say, the fully documented listening Olympics, supervised, to your peers?

I imagine you have done special things in your work that are not public and can be verified this way, right? 
But yes, there are hundreds of people at Microsoft that know about my listening abilities many of them first hand. Vast majority of them are not active on forums though.
I didn't think there would have been a need years later to prove that on some forum.  I would have gotten notarized statements from witnesses otherwise.

This being the Internet, I trust you accept Internet type evidence.  I did a search on AVS and quickly landed on a post by Ben Waggoner.  Ben is quite a famous compressionist.  You can google his name to find more.  Anyway.....

I tell you what AJ.  I will make a deal with you. I will fly to Florida to see you.  You pay half of my expenses and I pay the other half.  I will bring just my laptop and headphones and repeat the same tests I have post here.  If I pass, you pay the other half of my expenses plus another $1,000 I give to charity.  If I lose, I will do the reverse and you can decide what to do with the extra $1,000.  What say you?
Alternatively you fly to Seattle and I take the test here in front of you and we only wager the $1,000.

I didn't answer you xnor because all of this has been hashed out on AVS Forum.  And mzil gave you part of it.

So that you know what he is referring to, my team at Microsoft was responsible for entire Audio/Videos tack in Windows including DS, mixing and resampling pipeline, etc.  I championed ditching the horrible one in XP and creating a proper one.  I hired JJ to head that activity and the results were a major step up from XP.

Most of my testing was done on my rather new HP Zbook 14.  I take these tests casually in our family room with TV on and such (hence the dogs barking).  I like to use my Etymotic headphones because they seal so well.  I was challenged on that so I repeated the test with my Shure IEM.  To really put that argument to bed, I also ran a test with my uncomfortable Paradigm IEM.  I have a stock audio stack in my laptop and have Foobar running in default configuration.  Since all of the tests have been at 24-bit/96 Khz, I have set that as the audio property of the sound card.  There is still dither added on the way in/out of the kernel stack but seeing how I had positive results, I didn't need to go with WASAPI or ASIO to eliminate that.
[...]
Edit: Forgot to mention the OS.  It is Windows 7 Pro.

New version of Foobar ABX spits out the audio stack selection so here is David's test I ran:

What say you?

No, you are confused. I'm not talking about ripple but ringing. I never even mentioned ripple.

Adding to the ringing post above:



The background signal (blue) shows a sweep from 20 kHz to 23 kHz.
The rest show this signal filtered with the following pass band edges:
magenta = 20.5 kHz
cyan = 21.591 kHz
red = 22 kHz

Stop band starts at 22.05 kHz in all cases.

Here the same graph when subtracting the filtered signals from the original:


Here's what happens when you mess up the filtering (see cyan line which doesn't null anymore with the original signal): png

Quote from: amirm on 2014-11-21 23:32:20

Why do you come here AJ?

Why did I bait you into coming here? I thoroughly enjoy our conversations, because your lack of answers, two-stepping, etc. do provide answers. 
What do I also enjoy? Turning the tables on the elitists, bullies, illogical condescending snobs and highly self deluded "audiophiles" that permeate my audio hobby . Giving them healthy doses of their own medicine.

Nothing you could be cognizant of, figure out, much less worry about.

Oh and BTW, the thread is about the BS test, not "What Amir Bolt wants with audio files".

Quote from: amirm on 2014-11-22 00:07:54

What say you?

Contact this guy in WA.

cheers,

AJ

I don't want to wind up on his list of idiots by doing so.

You can contact him though since you have so much passion in this.

AJ I have been meaning to say this but keep forgetting.  Your picture in your avatar has the wrong aspect ratio.  It is squashed horizontally.  Are you not even aware of such obvious flaws?  Good grief man. I have not even seen the original but I am pretty sure it is wrong.

On page 2 I find two rather conflicting statements and I'm not sure what conclusion the authors draw:

Quote

It has further been suggested [22, 23, 24] that listeners can discriminate timing differences of the order of 5 µs and below, which, if correct, would require a Nyquist frequency of 32 kHz or higher...

And below that:

Quote

However, it can not be assumed that the auditory system cannot extract time differences that are shorter than the periods between successive samples, even when convolved with a sinc function in D/A conversion (assuming a sufficient signal-to-noise ratio).

It is my understanding (references to JJ, David Griesinger e.a.) that "In Physics, the accuracy of timing is not determined by the bandwidth, but roughly by the product of the bandwidth and the signal to noise ratio".
Am I correct that the authors say that the time resolution argument based on sample rate and excluding SNR is not valid ?

On page 2 I find two rather conflicting statements and I'm not sure what conclusion the authors draw:

Quote

It has further been suggested [22, 23, 24] that listeners can discriminate timing differences of the order of 5 µs and below, which, if correct, would require a Nyquist frequency of 32 kHz or higher...

And below that:

Quote

However, it can not be assumed that the auditory system cannot extract time differences that are shorter than the periods between successive samples, even when convolved with a sinc function in D/A conversion (assuming a sufficient signal-to-noise ratio).

It is my understanding (references to JJ, David Griesinger e.a.) that "In Physics, the accuracy of timing is not determined by the bandwidth, but roughly by the product of the bandwidth and the signal to noise ratio".
Am I correct that the authors say that the time resolution argument based on sample rate and excluding SNR is not valid ?

You may want to read this thread, starting from around the 12th or 13th post or so.

To also accurately reproduce changes in a signal’s frequency spectrum with a temporal resolution down to 6 microseconds, the sampling rate of a digital audio system must operate at a minimum of the reciprocal of 6 microseconds = 166 kHz.
...
table 504: Main decision parameters for the selection of a digital audio system’s sample rate

Audio quality issues:  desired temporal resolution 

48 kHz    20 ?S - high quality
96 kHz    10 ?S - very high quality
192 kHz    5 ?S - beyond human threshold

OK, finally I get to this technical point:
...

The above is also the reason I say that if CD had picked 48 Khz as the sampling, I would have no beef with it.  We have plenty of room to implement our filter above 20 Khz. But by picking 44.1, it leaves us a small margin forcing sharper filters and more time domain ringing.