limits on hearing...
One thing's indisputable... KBK's never short on words ..,.
I recall when Mr. Boyk published that work, I had some questions on usenet way back when about the audibility of said ultrasonic signals and, among other things, their impact on reproduced sound. It occurred to me that if a synthesized difference signal between the harmonic content out of band and the in-band signal was responsible for the perception of the ultrasonics, most likely that difference signal would occur in-band and hence be available in the digital reproduction.
For some reason, he could not (or would not) acknowledge or understand what i was asking, instead started shouting me down acting as though I was trying to somehow disprove his research to prove that analog was better than digital or something...
It takes all kinds, i guess
John L.
One thing's indisputable... KBK's never short on words ..,.
I recall when Mr. Boyk published that work, I had some questions on usenet way back when about the audibility of said ultrasonic signals and, among other things, their impact on reproduced sound. It occurred to me that if a synthesized difference signal between the harmonic content out of band and the in-band signal was responsible for the perception of the ultrasonics, most likely that difference signal would occur in-band and hence be available in the digital reproduction.
For some reason, he could not (or would not) acknowledge or understand what i was asking, instead started shouting me down acting as though I was trying to somehow disprove his research to prove that analog was better than digital or something...
It takes all kinds, i guess
John L.
(answering several posts from different people in this one post)
It was mostly noise. I could hear a faint background of high frequencies, rather grainy, but only with the playback gain turned way up from normal (the software lets you boost up to 75dB). Not zero difference, but enough to impress me with how little it was. Different ears might judge differently, though, but I'd hate to be a saleperson pitching 2496 audio with that as a demonstration.
The original 96kHz file was recorded at true 96kHz. What was subtracted from it was a 44.1kHz 16 bit file that had been sample converted to 96kHz. In order to subtract, there has to be the same sample spacing in each track. One track was pure recorded 24bit 96kHz. The other started life as 16bit 44.1kHz (but was recorded simultaneously from the same mastering panel), but was then upsampled.
No, not in any stereo recordings I am aware of. All mic-ing techniques I am aware of also will give an interchannel amplitude variation with source position, not just timing. Spread mic technique picks up mostly amplitude differences. Single point (crossed cardiod) technique is dominated by amplitude difference, there being little difference between mic positions. Hearing is known to be sensitive to fractions of a decibel differences, those can't be neglected. If you wanted to do a real test, take a mono source and process it into two channels so that ONLY the relative delay varies. Then you might be able to show what kind if delay sensitivity an ear has (though, like most such tests, you still won't be able to show a limit for all ears and all sound sources).
With the amps being made from parts that have tolerances and with tubes that have transconductance that changes with time, and little feedback to minimize effects of transconductance change, how do the amps have the same delay at the high frequencies? I'm sure you are aware of and can calculate the phase shifts that occur with rolloff in even a simple one pole RC circuit (only a DIGITAL filter can roll off response without a frequency dependent delay variation!!). The 17kHz rolloff can easily be 18kHz in one channel and 17.0kHz in the other. HF delay will vary accordingly.
As have I, with likely more years put in, but my conclusions differ from yours. I'd never argue with the other guy's preference, but I can with the technical reasoning used to convince others. I'd take the digital crossover any day (why don't you trade me yours for an old Fisher tube amp I've got stashed out in my garage?).
It was mostly noise. I could hear a faint background of high frequencies, rather grainy, but only with the playback gain turned way up from normal (the software lets you boost up to 75dB). Not zero difference, but enough to impress me with how little it was. Different ears might judge differently, though, but I'd hate to be a saleperson pitching 2496 audio with that as a demonstration.
The original 96kHz file was recorded at true 96kHz. What was subtracted from it was a 44.1kHz 16 bit file that had been sample converted to 96kHz. In order to subtract, there has to be the same sample spacing in each track. One track was pure recorded 24bit 96kHz. The other started life as 16bit 44.1kHz (but was recorded simultaneously from the same mastering panel), but was then upsampled.
No really. The amplitude of the bell changes with distance from the listener (6dB every time the distance doubles). Even more significant, the reflection and constructive/destructive interference pattern changes radically as the bell moves across the room. Amplitude variations are still quite significant. Just listen to your own voice as you move closer and further from nearby wall to experience this (the delay of the direct path from mouth to ear could be assumed to be pretty constant in that case, I'd think).
No, not in any stereo recordings I am aware of. All mic-ing techniques I am aware of also will give an interchannel amplitude variation with source position, not just timing. Spread mic technique picks up mostly amplitude differences. Single point (crossed cardiod) technique is dominated by amplitude difference, there being little difference between mic positions. Hearing is known to be sensitive to fractions of a decibel differences, those can't be neglected. If you wanted to do a real test, take a mono source and process it into two channels so that ONLY the relative delay varies. Then you might be able to show what kind if delay sensitivity an ear has (though, like most such tests, you still won't be able to show a limit for all ears and all sound sources).
With the amps being made from parts that have tolerances and with tubes that have transconductance that changes with time, and little feedback to minimize effects of transconductance change, how do the amps have the same delay at the high frequencies? I'm sure you are aware of and can calculate the phase shifts that occur with rolloff in even a simple one pole RC circuit (only a DIGITAL filter can roll off response without a frequency dependent delay variation!!). The 17kHz rolloff can easily be 18kHz in one channel and 17.0kHz in the other. HF delay will vary accordingly.
As have I, with likely more years put in, but my conclusions differ from yours. I'd never argue with the other guy's preference, but I can with the technical reasoning used to convince others. I'd take the digital crossover any day (why don't you trade me yours for an old Fisher tube amp I've got stashed out in my garage?).
I can always see a difference, I'm more concerned with what I might hear. I can see differences in most anything that are really different in some way, just use equipment that is sensitive enough. I can see differences in any two "identical" cables you give me if I can use a 40GHz network analyzer (I have to work with RF systems where cable matching requirements can be really tricky).
The differencing test is more intended to let a person experience how strong the differences SOUNDS like. When you listen to only the difference (separated from the signals that are the same) it can be a pretty convincing by-ear calibration of just what the significance of some of the things that get argued about is likely to be.
The test software is free, try yourself if you want to hear this kind of result -- there is a good demo of what I'd call a insignificant difference resulting from using the famed "green pen treatment" on a CD.
http://libinst.com/Audio%20DiffMaker.htm
bwaslo said:
Suffice it to say, anything that one says can be attacked or commented upon, with a null or negative result in the attempt of communication. I always hope that one can apply intellect and reason to another's musing and conclusions, other than arguing the lack of full detailed explanations. No-one has a desire to write entire discourses here on the forum. Nor do I expect you to do so.
In essence, I don't think your objection to my points raised are entirely valid or reflect the points I'm attempting to get across. They merely seem to represent a misunderstanding or misinterpretation, real or deliberate, of what I'm saying.
I'm a big fan of high gain difference tests, though I've only done them in analog, not digital. The nice thing is you don't have to worry about putting a name on whatever differences you find, only whether you can hear them. What I've found is that amplifier channels usually have quite measurable differences in both gain and phase at the ends of the audio band. I tend to be very sensitive to balance issues and IMO, matching components to get the same phase shifts (and thus rolloffs) is a good thing to do. Thus, I wouldn't give an automatic pass to any amp, ss or tube, without verifying what it really does. More on-topic, I remember hearing about a scheme where two focused modulated ultrasonic beams can be used to beam an audio message directly at an individual, so there's no question as to whether out-of-band signals have an effect within the audio band. The frequency response of my hearing drops off like a stone at about 14.5kHz. In the old days, with LPs, when I could hear to 20kHz, I could barely hear the effect of switching in the 15kHz filter on my Dynaco preamp. It simply made no significant difference. Yet, if I listen to a digital recording (44.1) from one of my LPs, I believe I can still hear a slight difference if I drop out everything above 15kHz using a digital filter. There's obviously something going on here, though one would be foolish to call it a huge effect. Next I'll try it with some 96kHz sampled stuff.
I don't doubt that instruments produce significant content above 20kHz, in fact it's completely unsurprising.
As far as I have ever read or witnessed human hearing drops off sharply at 15kHz. About half the time somebody claims to hear the tone above around 13-14KHz they can still mysteriously hear the tone when the function generator is turned off, but the amplifier is on. I have always attributed this to noise in the amplifier.
As far as I have ever read or witnessed human hearing drops off sharply at 15kHz. About half the time somebody claims to hear the tone above around 13-14KHz they can still mysteriously hear the tone when the function generator is turned off, but the amplifier is on. I have always attributed this to noise in the amplifier.
soongsc said:
This is very true, and there are a number of clues involving
perception and signals with different pitch, amplitude, and phase.
Biosonar was kind enough to point me to the book "The
Psychology of Music" by Diana Deutsch and a summary of some of
these phenomena in her article "Paradoxes of Musical Pitch" in
the August '92 issue of Scientific American.
BTW, as pure sine, triangle, and square waves have no 2nd
harmonic, you would be looking to the 3rd harmonic in a
comparison of their audibility. If you can hear the difference at
7 KHz, this implies that you can hear 21 KHz.
but people often make the mistake of thinking they can just push the button on their function generator to do the switching - and then they do hear a diff between them
in fact the amplitude of the sine wave fundamental of the tri and sq waves is not simply the peak values of those waveforms
you must correctly adjust the amplitude of the the sine ref in a ab/x test to the value of the fundamental component in the test waveform before going off half cocked and crowing about your >25KHz ultrasonic hearing
in fact the amplitude of the sine wave fundamental of the tri and sq waves is not simply the peak values of those waveforms
you must correctly adjust the amplitude of the the sine ref in a ab/x test to the value of the fundamental component in the test waveform before going off half cocked and crowing about your >25KHz ultrasonic hearing
Here are some examples of Diana Deutsch's work:
http://philomel.com/musical_illusions/
http://philomel.com/phantom_words/
and some other audio illusions:
http://www.noah.org/science/audio_paradox/
http://philomel.com/musical_illusions/
http://philomel.com/phantom_words/
and some other audio illusions:
http://www.noah.org/science/audio_paradox/
I have not read any of Diana Deutsch's work before, but it seems very interesting. Ideally, we would want the acoustic signal to be a perfect replica of the original signal, but when we cannot acheive it, then better understanding of human perception provides good information on what the tradeoff should be. Many people have testified that gradual roll-off at the top frequency is better than sharp roll-off at higher frequencies. Since various frequencies will also impact our perception of the other spectrum end, it's fun to just keep an open mind and try things out.
One thing that I seemed to notice when listening to and measuring various capaictors and speaker internal wires is that whenever I heard an improvement in the high frequencies, the measured improvement seemed to be in the low frequency range.
One thing that I seemed to notice when listening to and measuring various capaictors and speaker internal wires is that whenever I heard an improvement in the high frequencies, the measured improvement seemed to be in the low frequency range.
Re: Re: limits on hearing...
According to THE authority WikiPedia, at least in one case, the mixing occurs in the air. Of course, this doesn't support harmonics at lower power levels directly, but maybe hints at the effect?
"A transducer can be made to project a narrow beam of modulated ultrasound that is powerful enough, (100 to 110 dBSPL) to change the speed of sound in the air that it passes through. The air within the beam behaves nonlinearly and extracts the modulation signal from the ultrasound, resulting in sound that can be heard only along the path of the beam, or that appears to radiate from any surface that the beam strikes."
http://en.wikipedia.org/wiki/Sound_from_ultrasound
John L.
rdf said:
According to THE authority
WikiPedia, at least in one case, the mixing occurs in the air. Of course, this doesn't support harmonics at lower power levels directly, but maybe hints at the effect?"A transducer can be made to project a narrow beam of modulated ultrasound that is powerful enough, (100 to 110 dBSPL) to change the speed of sound in the air that it passes through. The air within the beam behaves nonlinearly and extracts the modulation signal from the ultrasound, resulting in sound that can be heard only along the path of the beam, or that appears to radiate from any surface that the beam strikes."
http://en.wikipedia.org/wiki/Sound_from_ultrasound
John L.
Nelson Pass said:
BTW, as pure sine, triangle, and square waves have no 2nd
harmonic, you would be looking to the 3rd harmonic in a
comparison of their audibility. If you can hear the difference at
7 KHz, this implies that you can hear 21 KHz.
not too sure what constitutes a "pure" square wave??? The sum of an infinite # of sines??? How would you produce it??? Do tell...
... maybe has more to do with Gibb's phenomena?? Or maybe invoke Heaviside??
"An ideal square wave requires that the signal changes from the high to the low state cleanly and instantaneously. This is impossible to achieve in real-world systems, as it would require infinite bandwidth."
http://en.wikipedia.org/wiki/Square_wave
John L.
Our hearing is non-linear. The physical structure of the ear actually produces distortion.
http://www.isvr.soton.ac.uk/SPCG/Tutorial/Tutorial/Tutorial_files/Web-hearing-difference.htm
http://www.sciencedirect.com/scienc...serid=10&md5=decc09cbd6d5fb687a54fe9144a3cfc3
http://cat.inist.fr/?aModele=afficheN&cpsidt=14641728
http://bmb.oxfordjournals.org/cgi/content/full/63/1/121
http://www.mp3-tech.org/programmer/docs/non-linear_human_hearing.pdf
As near as I can tell, mixing occurs both inside and outside the ear. It can happen in the electronics, the transducer, in the air. (If you can get close to a hot soprano section, you can experience the IM – not nice, but sounds good out in the hall).
But decent conventional speakers operated within design constraints tend to produce mostly even order components and these tend to fall off very quickly. This does leave them vulnerable to producing audible non-linear artifacts from other components in the system.
soonsc's comment is applicable:
LF electronic grunge was modifying HF sound. At least, given the energy content of LF and HF signals, that's my conclusion.
Our hearing is outrageously sensitive in a range of about 1 – 5 kHz and remarkably subtle differences in this area seem to have a large effect on reproduction of things like violin harmonics, singers’ formants, symphonic and choral ensemble detail, etc.
My question is what sort of artifacts might be transposed/inserted into this range by using that brickwall filter?
Is it just that it doesn’t roll off steeply enough?
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