It's hardly fair to ask people not to trust their ears and then provide arguments like this. Shame on you (just kidding!)
But would you say that about Wikipedia if Wikipedia had agreed with your own views?
I am not against your conclusions, necessarily.
Darren
Sent from my HTC Sensation Z710e using Tapatalk
Results 51 to 60 of 149
Thread: 192kHz considered harmful
-
2012-03-07, 11:35 #51Senior Member
- Join Date
- Mar 2007
- Location
- UK
- Posts
- 1,074
Last edited by darrenyeats; 2012-03-07 at 11:41.
Check it, add to it! http://www.dr.loudness-war.info/
http://www.amazon.co.uk/gp/richpub/l...606506-5721503.
SB Touch
-
2012-03-07, 11:40 #52Senior Member
- Join Date
- Mar 2007
- Location
- UK
- Posts
- 1,074
Check it, add to it! http://www.dr.loudness-war.info/
http://www.amazon.co.uk/gp/richpub/l...606506-5721503.
SB Touch
-
2012-03-07, 11:46 #53Senior Member
- Join Date
- Jan 2007
- Posts
- 734
-
2012-03-07, 11:57 #54--------------------------------------------------------------------
Main hifi: Touch + CIA PS +MeridianG68J MeridianHD621 MeridianG98DH 2 x MeridianDSP5200 MeridianDSP5200HC 2 xMeridianDSP3100 +Rel Stadium 3 sub.
Bedroom/Office: Boom
Kitchen: Touch + powered Fostex PM0.4
Misc use: Radio (with battery)
iPad1 with iPengHD & SqueezePad
(in storage SB3, reciever ,controller )
server HP proliant micro server N36L with ClearOS Linux
http://people.xiph.org/~xiphmont/demo/neil-young.html
-
2012-03-07, 11:58 #55Senior Member
- Join Date
- Apr 2005
- Location
- Buckinghamshire, England
- Posts
- 9,983
It is trivial to show that the error inherent in the process is below the range of detection.
A computer-generated sine wave dataset (I.e. not recorded via an ADC) can be passed through a DAC and null-compared. I've done this and the result should be no surprise to anyone.
PS don't try this with a nos DAC!
The human ear/brain also operates on a sampling basis (there's nothing "analogue" about the ear by the way) and uses a reconstruction filter just like a DAC to integrate the discrete samples into something we can understand.
What some people are misunderstanding here is that it is the reconstruction filter that recovers the analogue signal, not the DAC. The DAC simply presents the filter with a set of voltages over time. It is within the filter that the sinc function becomes manifest and this is indeed an infinite series - the mathematical definition of a filter is a continuous function over time. A filter is not a step function!
To be clear on this, what comes out of the filter IS a mathematically perfect sine wave... All the way up to the Nyquist frequency. This is both predicted by the theorem and demonstrable in practice. There is no known way to differentiate between a 1khz sine wave sampled at 44.1 or 192. In every conceivable way of " measuring" or analysing that sine wave it they will be indistinguishable.
What is true for one sine wave is also of course true for any combination of sine wAves (or as we usually refer to it... Music).
Or do we need to have a conversation about Fourier transforms and the practical implications of mathematical infinite series as well?
Now there IS a way to break this model. Try a perfect square wave!. This has to be done using a mathetmatically generated waveform/data set because it is impossible to generate or record a perfect square wave with infinite rise/fall times in the analogue domain. The filter will introduce non-linear ringing.
This is all predict by the theorem because an infinitely fast slope requires an infinite number of samples... I'm sure you get the idea.
Bottom line is this; for real world sound distribution 44.1 is fine ... Which is why there are many many fine sounding red book CD's... And why there is no published evidence that stands scrutiny to support the idea that anyone can tell the difference between the same master distributed and played back at 192 or down sampled to 44.1.Last edited by Phil Leigh; 2012-03-07 at 12:00.
You want to see the signal path BEFORE it gets onto a CD/vinyl...it ain't what you'd call minimal...
Touch(wired/W7)+Teddy Pardo PSU - Audiolense 3.3/2.0+INGUZ DRC - MF M1 DAC - Linn 5103 - full Aktiv 5.1 system (6x LK140's, ESPEK/TRIKAN/KATAN/SEIZMIK 10.5), Pekin Tuner, Townsend Supertweeters,VdH Toslink,Kimber 8TC Speaker & Chord Signature Plus Interconnect cables
Stax4070+SRM7/II phones
Kitchen Boom, Outdoors: SB Radio, Harmony One remote for everything.
-
2012-03-07, 12:47 #56
Just trying to educate myself here...
Does this mean that there could be audible distortion introduced due to ringing in a recording that has clipped waveforms? Taken to an extreme, clipping could start to approximate a square wave.
BTW, I'm finding this discussion to be very interesting...I'm learning a lot.Last edited by maggior; 2012-03-07 at 12:48. Reason: I meant clipped waveforms, not samples.
Rich
---------
Setup: 2 SB3s, 4 Booms, 1 Duet, 1 Receiver, 1 Touch, iPeng on iPod Touch, SqueezeCommander, OrangeSqueeze, and SqueezePlayer on Xoom and Galaxy Player 4.2. CentOS 6.3 Server running LogitechMediaServer 7.7.2 and SqueezeSlave.
Current library stats: 40,810 songs, 3,153 albums, 582 artists.
http://www.last.fm/user/maggior
-
2012-03-07, 12:59 #57Senior Member
- Join Date
- Mar 2007
- Location
- UK
- Posts
- 1,074
Check it, add to it! http://www.dr.loudness-war.info/
http://www.amazon.co.uk/gp/richpub/l...606506-5721503.
SB Touch
-
2012-03-07, 13:05 #58
Hmm thinking about it there can not really be any invalid combinations of samples ?
Whatever gets coded inside the 16/44.1 code space or whatever must be recoverable you might not want to listen to it and the square wave is approximate as you can't have infinite slope and it do ring .
But afaik the ringing are above >20kHz .--------------------------------------------------------------------
Main hifi: Touch + CIA PS +MeridianG68J MeridianHD621 MeridianG98DH 2 x MeridianDSP5200 MeridianDSP5200HC 2 xMeridianDSP3100 +Rel Stadium 3 sub.
Bedroom/Office: Boom
Kitchen: Touch + powered Fostex PM0.4
Misc use: Radio (with battery)
iPad1 with iPengHD & SqueezePad
(in storage SB3, reciever ,controller )
server HP proliant micro server N36L with ClearOS Linux
http://people.xiph.org/~xiphmont/demo/neil-young.html
-
2012-03-07, 14:30 #59Senior Member
- Join Date
- Jan 2007
- Location
- Midlands, England
- Posts
- 618
We do have to distinguish properly between CD at 44.1k sampling with 16bits and 44.1k - 24 bits. The 24 bits do have an effect. They allow a significantly greater dynamic range, assuming the mastering engineer does't get into loudness wars compression.
It may be for some people that they compare 96/24 with CD and it is the 24bit resolution not the sampling frequency differences that are being heard.
Also if its old music, my late 60s and 70s youth, any new releases will be from analogue tape remastered. This remastering may (and I know several cases of did) produce entirely different mix from early CD versions. IMHO not always better.
Phil Leigh would have more input on these recording and mastering issues.
Today with the freedom to use anti-aliasing and reconstruction filters that are low order and relatively benign, due to very high frequency oversampling techniques, I believe we have an almost untainted record / replay chain.
Dave
-
2012-03-07, 14:33 #60
The article linked in the OP covers bit depth too.
The incorrect '96dB' figure ignores the spectral power density of a signal. 16 bit audio can go considerably deeper than 96dB, and deeper yet with proper use of dither. Handled correctly, the dynamic range of 16 bit audio reaches 120dB in practice [10], more than twenty times deeper than the 96dB claim.
That's greater than the difference between a mosquito somewhere in the same room and a jackhammer a foot away.... or the difference between a deserted 'soundproof' room and a sound loud enough to cause hearing damage in seconds.
16 bits is enough to store all we can hear, and will always be enough.


Reply With Quote
bonus point if they used thier cryogenic treated cables in the process ( that was not a joke they do that, amazing ?)

