IME (admittedly not spanning 30 years in digital audio yet) its down to the preponderance of fairly pants implementations
Could be. IME, it's most often the source material.
But we should keep in mind that Zwicker/Fastl talks about thresholds with single tones.
And that it is based on a sample of 100 participants, in which the spread is already quite high (note that the lines correspond to 5%-95% of the sample).
It is not mentioned if the sample was constructed as a representative one, but we could construct a statistic with estimations based on this sample.
Straight forward statistical analysis shows that several million people are most probably outside of the range shown in the graph (of course to both sides).
Exaggeration it surely is wrt to the whole population but if somebody does percept something it is not.
It seems to be established nowadays that the temporal resolution is higher than the upper bound normally would predict.
I guess that is related to Meyer/Moran.
While the results might be true, it is hard to tell because it was mainly an example of not so well done scientific work.
(And i am still wondering why this article could make it through the review process as it did not fulfill the basic requirements of any engineering report)
Does someone guarantee that no `illegal` data occurs during sampling/editing or digital audio work in general?
(illegal means digital data that could not be existent if a signal is quantized after a proper antialiasing filter)
It is the same problem as with intersample overs; it would not happen if everybody .....
Applying this line of reasoning to the size of people, we should have 5 meter tall giants.
Temporal resolution works on the basis of phase shifts. As you may know, < 3.500 Hz or so, the nerve fibers connected to you inner ear fire at a precise point of the incoming signal, that is, they are phase locked. Hence, with for example a 1000 Hz tone (1mS wide), the temporal resolution may be < 10uS.
Therefore, you can't argue that, since the temporal resolution is < 10uS, we need >100KHz bandwith. They are not related. As a matter of fact, for higher frequencies, where the nerve fibres are no longer phase locked, the concept of temporal resolution looses its validity.
vac
No, nobody makes any such guarantee. If you test a lot of equipment out there, you'll find plenty of aliasing in the digital outputs, and some amount of aliasing in the A/D. That's because, even if there are anti-alias filters, they might only reduce the 'illegal' frequencies by a couple of digits of dBs, but that still doesn't mean that they've been eliminated completely. Cost cutting in the circuits ends up allowing just a bit of aliasing in both recording and playback.
By the way, if you're using an audio interface for such analysis, it helps to set the frequency to linear rather than logarithmic. The latter is needed for creative tone sculpting (mastering), but when you're looking for aliasing or other errors the linear graph will show it more clearly as a reflection in the frequency domain. I often record at 48 kHz for DVD, and when musicians use digital equipment on stage it's quite common to see a mess of aliased frequencies between 22.05 kHz and 24 kHz. These are just aliased artifacts that are produced by budget digital gear running at 44.1 kHz without adequate reconstruction filters. Technically, there should be nothing but noise between 22.05 kHz and 24 kHz, but there's quite a bit more than that, and it's clearly signal dependent.
I don't understand the point you're trying to make. Anti-aliasing filters work. Are you saying that digital media sometimes have errors? That seems quite unexceptional.
I don't follow your reasoning. If the recording system does not preserve 10 us events, then both periodic and nonperiodic detail is lost. We might search forever to find someone who can prove that they hear 100 kHz or even 50 kHz, but once you bandlimit to 20 kHz you've lost those 10 us events.
The current reasoning that I understand is that it's all about preserving the impulse response, not about preserving ultrasonic tones that are continuous. The clincher is that preserving 10 us impulses happens to also preserve 100 kHz tones. It's not that you need the 100 kHz tones, but you get them anyway if you preserve the 10 us impulse response.
Well we may have had horrible digital music for the last 30 years (not my view I have to say PMA!) but I don''t know that there's that much pristine information above 20KHz on an LP either. There may be a case for a wideband digital audio format (SACD was supposed to be it, but got heavily criticized for the noise shaping and ultimately the high levels of u/sonic noise - but most of the reviews I read seemed to give the format a big + for sound quality) but whether it will ever become a reality . . . I doubt it.
I have to agree with SY here as well - I find huge differences in recordings and I think this is where most of the issues lie. I have a Fleet Foxes album ("Helplessness Blues') and the music is just wonderful . . . but the recording has a shrill sound to it, almost like there's a few sharp peaks in the repsonse up in 1-2 KHz region that make my ears ring and it can become quite tiresome after a while. Then I put another CD on (say Eliane Elias's 'Something for You') and that issue is not there. I put a Paul Simon CD on ('Rythym of the Saints') - no problem - very relaxed recording. So, where does that leave us?
I have to agree with SY here as well - I find huge differences in recordings and I think this is where most of the issues lie. I have a Fleet Foxes album ("Helplessness Blues') and the music is just wonderful . . . but the recording has a shrill sound to it, almost like there's a few sharp peaks in the repsonse up in 1-2 KHz region that make my ears ring and it can become quite tiresome after a while. Then I put another CD on (say Eliane Elias's 'Something for You') and that issue is not there. I put a Paul Simon CD on ('Rythym of the Saints') - no problem - very relaxed recording. So, where does that leave us?
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Would the sine sum generated 1kHz triangle be linear even in its middle third?
The 22kHz brickwall filtering is unacceptable for highest fidelity music reproduction. No one has ever proven that this sharp frequency cut-off has no influence to music perception and sound quality. If the roll-off were slow, then there would be no big problem. As the slow roll-off is impossible for the reason of aliases, we get what we get and we can complain to eternity. "We" means probably 0.00000000001% of population, and of course the rest counts for the consumer music industry.
The best DAC I've ever heard uses a first order 22kHz filter.
John
Thorsten,
Limited?
Not every discussion is an argument that has to be won at all costs. There has been a very large market for highish end portable recorders for a long time. John will remember the Nagra, IIRC retailed for more than a Revox or Ampex. Several models were specifically designed for the professional motion picture industry.
I have a folk LP from 1965 where one cut sounded far better than all the rest, when the CD came out in 1999 that cut was missing. Upon inquiry it turns out that cut was an audience tape made on a Nagra and was lost.
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Howard,
Quite good.
ECC83 aka 12AX7 A bit of a common audio tube.
2N107 GE's killer app germanium transistor
Ua703 Fairchild's IF amplifier chip
Ul923 Fairchild's RTL Flip Flop Probably the first widely used logic family.
Reasonably close! But it is intended for just audio and actually doesn't go quite low enough in frequency for audio and not high enough for where I think I want to look. Thanks!
Thanks Demian,
I have 1 to 1000 Mhz covered with my RF gear. (We install moderate size cable TV systems 1000+ drops.)
I guess I'll find out what injecting noise does to conventional measurements. I suspect I will see widening of the test tone FFT width at the baseline, what else should be interesting.
There is something there when combined with reproduction approaches does show up as differences in player quality. Maybe that error correction really does count!
Hi,
Maybe not. Both AES and ASA tend to be suspiciously US focused and dominated.
For example much of the research published by the IRT in Germany (which is generally excellent) does not make it to the AES (maybe the Researchers fear the AES thought police, or maybe they cannot be bothered to translate to English?)
The Naim of the Gentleman in question is Gerard Slot and he appears dutch. The books is a translation from the dutch. Published by Drake New York. The gentleman also seems to have written other treatises with interesting titles, especially those that appear to be listed only in german (wohl derm der deutsch kann).
This is what I was able to find to on the title by highlighting what John had typed and right clicking "google"...
Audio quality
Audio quality: requirements for high quality audio equipment
by G. Slot
Publisher: New York : Drake, 1971.
ISBN: 0877490678
Notes: Translation of Geluidskvaliteit.
Includes bibliographical references.
Book Details:
Language: eng
Physical Description: 154 p. : ill. ; 23 cm.
Ciao T
Maybe not. Both AES and ASA tend to be suspiciously US focused and dominated.
For example much of the research published by the IRT in Germany (which is generally excellent) does not make it to the AES (maybe the Researchers fear the AES thought police, or maybe they cannot be bothered to translate to English?)
The Naim of the Gentleman in question is Gerard Slot and he appears dutch. The books is a translation from the dutch. Published by Drake New York. The gentleman also seems to have written other treatises with interesting titles, especially those that appear to be listed only in german (wohl derm der deutsch kann).
This is what I was able to find to on the title by highlighting what John had typed and right clicking "google"...
Audio quality
Audio quality: requirements for high quality audio equipment
by G. Slot
Publisher: New York : Drake, 1971.
ISBN: 0877490678
Notes: Translation of Geluidskvaliteit.
Includes bibliographical references.
Book Details:
Language: eng
Physical Description: 154 p. : ill. ; 23 cm.
Ciao T
I'd love to work with such a rig. Gary Pimm and I did some testing of an AC line filter a couple of years ago using his very cool HP spectrum analyzer. But we were not injecting precise signals, just looking at what was there and what got by the filter. The problem was that line noise is forever changing.
FWIW, we found that the filter was better at keeping noise IN, than OUT. It went into the junk pile.
Yes, RF carriers are meant to be demodulated. Scaling these effects directly to audio is a stretch, but you are welcome to.
Thanks for the 'legwork' Thorsten. I picked up the book in London, in 1976 at either Foyles or Modern Book Co. and it is a hardcover edition.
It was also published by Philips Paperbacks, this is part of their description:
"This book deals with the problem of obtaining the quality of sound reproduction that will give the maximum satisfaction to those listening to it. The author considers the whole problem objectively. A chapter spent on discussing the 'problem' is followed by one on technical specifications. ..." A good book, one of the best on this subject.
It was also published by Philips Paperbacks, this is part of their description:
"This book deals with the problem of obtaining the quality of sound reproduction that will give the maximum satisfaction to those listening to it. The author considers the whole problem objectively. A chapter spent on discussing the 'problem' is followed by one on technical specifications. ..." A good book, one of the best on this subject.
Quote without comment: Slot P.52.
"The effect of the response sounding brighter or shriller than when the characteristic is straight (freq resp) starts when the roll-off is more than 6 dB/octave.
Unfortunately we know of no complete explanation of the phenomen nor of an exact quantitative analysis. In part it can perhaps be explained by assuming that cutting off an important part of the residue will to some extend reduce our certainty about the pitch of of tone, particularly when its fundamental is not very strong. Another explanation can be that a sharp cut-off in the frequency characteristic of a pick-up, amplifier or loudspeaker has the consequence that part of the ear that is reacting to the cut-off frequency will react abnormally to the mutilated signal [18]. "
[18] Franssen, N.V. 'Some Considerations on the mechanism of directional hearing. Thesis Delft 1960'
"The effect of the response sounding brighter or shriller than when the characteristic is straight (freq resp) starts when the roll-off is more than 6 dB/octave.
Unfortunately we know of no complete explanation of the phenomen nor of an exact quantitative analysis. In part it can perhaps be explained by assuming that cutting off an important part of the residue will to some extend reduce our certainty about the pitch of of tone, particularly when its fundamental is not very strong. Another explanation can be that a sharp cut-off in the frequency characteristic of a pick-up, amplifier or loudspeaker has the consequence that part of the ear that is reacting to the cut-off frequency will react abnormally to the mutilated signal [18]. "
[18] Franssen, N.V. 'Some Considerations on the mechanism of directional hearing. Thesis Delft 1960'
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