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    Digital Vinyl: Temporal Domain

    Note : The following article contains information that has been deemed incorrect by leading digital audio engineers. I attempted to corroborate the findings of this article by asking several digital audio experts. I was unable to find anyone who could back up the statements made, with any scientific data or theory. Consider the following article retracted.

     

    I am leaving the text of this article up on CA because it has enabled a good discussion to take place. By leaving it up, people can read what was claimed and read the followup arguments that the prove it incorrect. To remove the article completely only opens up a space for this to happen again, and again, and again.

     

     

    I take full responsibility for the publishing of this article. I should have had a technical editor check it before publication. I apologize to the CA Community for the error in judgement.

     

    - CC.

     

     

     

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    Temporal Domain of Signal, or What is More Important for Listening to Music, Static or Dynamic Characteristics of the Sound Signal?

     

    Every time my audiophile friends, who do not have an analog setup (TT), come to me and see huge piles of expensive, rare LPs, they get puzzled. They wonder, how can it be that LP lovers spend huge amounts of money on their "analog" hobby, while suffering such discomforts when listening to music. They say this method of listening in the 21st century is absolutely impractical. In addition, there are signal distortion and limitations in many of the technical aspects of vinyl.

     

    In response, I always say the same thing in support of analog - it's mainly because of the time domain signal. We (fans of analog audio) are willing to make these sacrifices and inconveniences for much better performance in a time aspect, the so-called dynamic characteristics. Static characteristics, those belonging to the spectral and dynamic domains (Dynamic Range, THD + N, Frequency Response, etc.) certainly are important for high-quality sound, but when it comes to listening to music in real time, in my opinion, it is the dynamic characteristics that matter most

     

     

    Often, in response to my comments, people react with skepticism. They say they are used to trusting technical information that can be measured and compared and what I say is very subjective and ephemeral.

     

    Also viewing comments here on СA, especially those connected with the current topics such as MQA, I have noticed that some members react rather skeptically to the arguments about MQA's improvements of characteristics in the time-domain. And, some even question the very existence of such improvements.

     

    Here it is shown that "High-resolution in temporal, spatial, spectral, and dynamic domains together determine the quality value of perceived music and ,sound and that temporal resolution may be the most important domain perceptually". Temporal resolution, is actually what I would like to briefly discuss with you.

     

    There's a deeply rooted opinion that frequency above 10 kHz, and moreover above 20 kHz, contains a small amount of music information. And yet research shows that, for example transients from cymbals contain significant frequency components extending even above 60 kHz. The trumpet playing fortissimo has transients components up through 40 kHz, and in the case of the violin even temporary frequency of 100 kHz occurs.

     

    As you can see quite a lot of music information is contained in frequencies above 20 kHz. Of course, immediately a question is raised: "Are we able to hear it?". To answer this, it is worth mentioning some rarely discussed issues. Commonly cited audibility up to 20 kHz frequency is derived from conventional hearing tests, which are based on the audibility of simple sounds. But there is an alternative look at the issue from the more "dynamic" side. This is the temporal resolution of the ear, not the "static" harmonic content and audibility of pure sinusoidal tones.

     

    This may be more appropriate in a case of music signals than the prospect of simple tones. The actual music signals have a very complex structure as a result of the imposition of the attack and decay of many instruments. More importantly, their frequency spectrum is very different between the short period of the initial attack, or the rise of sound, eg. as a result of pulling a string or striking a key of a piano, and the subsequent, much longer sound decay.

     

    There is a large group of instruments, which are characterized by a very "transient," dynamic nature of the initial attack phase of the sound. Xylophone, trumpet, cymbals and striking a drum achieve dynamic levels in between 120 and 130 dB within 10 ms or less. One thing we can say for sure, it is not possible for a CD-quality sample scattered at 22.7ms to have an opportunity to correct the commissioning attack phase of musical instruments, which are half the distance between two consecutive samples.

     

    And the attack phase is very important for audio reception. In experiments, in which the samples were of wind instruments dissected in a way that combined the short attack phase of one instrument with a longer sound decay of another one, listeners identified the sound that of the instrument with the short fragment attack, not the longer decay sound.

     

     

    image2.png

     

    The sound wave graph from a cymbal being struck by a stick. The sound increase is nearly instantaneous, followed by a long sustain of a rather uniform nature. - from highfidelity.pl

     

     

    When viewed from the hearing mechanism perspective, you can find information indicating that the signals which have pulsing character (i.e., generally transients), in contrast to simple tones, activate significantly larger areas of hearing cells than pure sinusoidal tones (which, in nature are almost non existent). In the case of pulses, the possible temporal resolution of the human ear may be up to 10 microseconds, corresponding to frequencies of 100 kHz.

     

    This information is also confirmed in the opinion of recognized practitioners. Art Dudley from "Stereophile" magazine, in an interesting interview from The Editors cycle, is of the opinion that the Nyquist frequency does not apply while there are working decimation and reconstruction filters of complex music signals. In his opinion, two samples may be used to describe a single frequency, but do not provide sufficient density samples to describe the speed at which the signal increases or decreases. This is crucial to distinguishing between music and ordinary sound.

     

    Also I would like to quote, in the context of the above information, an excerpt from my correspondence with Dr. Rob Robinson:

     

    "My thoughts are that with extended frequency response you are not capturing "audible" frequencies but rather preserving the critical time relationships in the music at all frequencies. Human hearing might not be able to "detect" sounds above 15 - 20 kHz or so, but on the other hand hearing, in conjunction with the brain, is very sensitive to temporal information. It's been reported that the human auditory system is capable of discerning temporal differences of tens of microseconds or less (and note, at 192 kHz the time between samples is 5 microseconds). This temporal discrimination is the reason we are able to accurately discern directional / spatial cues. Hearing evolved so that the location of threats, e.g., the cougar about to pounce, could be determined accurately, as key to survival. The spatial information comes not only from amplitude, but the time difference between the same sound arriving at each ear. And the more sensitive hearing is to temporal information, the more accurately that spatial cues can be located.

     

    A CD format brickwall filter will affect time relationships, part of the reason that CD format digital audio may sound less "natural" than analog (or live sound). Preserving temporal information is key to preserving lifelike sound and imaging. While all digital audio will affect temporal information, the influence diminishes the higher the sample rate, because the antialiasing and reconstruction filters are operating at ultrasonic frequencies. So, by using higher sample rates, even though we may be recording sounds that are inaudible, we have better preservation of the temporal information in the signal, which conveys a more lifelike presentation of the music. Besides using a high sample rate to capture the signal, we also have the ultra wide 5,000 kHz bandwidth (five thousand kilohertz, as contrasted with "just" 20 kilohertz as the generally accepted audible upper frequency limit) of the Seta preamplifier which again faithfully preserves temporal relationships in the music signal (internally, the front end circuitry has a risetime of less than 50 nanoseconds)." - Dr. Rob Robinson

     

    If we take into consideration the typical technical parameters of audio, which is mainly bandwidth and dynamics (signal-to-noise ratio), we can easily come to the conclusion that, omitting the variables associated with the physiology of hearing, audiophile devices should not differ from each other, and moreover sonically stand out in relation to the audio devices from the mass market.

     

    And yet, there are people willing to pay much higher prices for equipment and the typical specifications are often similar or even slightly worse than the cheaper devices of the mass segment.

     

    Most importantly, in many cases audiophiles agree on the description of the main attributes of the sound of the given device, although expressed in a specific descriptive dictionary, and not in strict technical parameters.

     

    This raises a difficult to challenge conclusion that if some audiophile characteristics are consistently perceived by a large number of people there's a good chance that behind this stands specific physical phenomena, though their nature can be complicated and can be difficult to express in simple numerical parameters, eg. dynamic range or frequency response.

     

    What may these phenomena be? If the key to the mystery lies not in the parameters of the frequency domain (frequency response) or dynamics (noise at a low level), then a single area remains, and that's phase issues, or timing aspects of the sound. In fact, these are the most fundamental parameters of the sound signal, because they underlie its creation, what a sound wave actually looks like in the time domain. The question is how much of the sound wave graph corresponds to the wave reaching the microphone registering this recording.

     

    The nuances of the tonal colors, to the greatest extent, are shaped by the sound wave characteristic from each instrument. And, it's not just a simple analysis of the contents of the so-called harmonics but more of dynamic aspects, mainly the so-called attacks, or the rising of sound at the moment of its creation. It is not difficult to imagine that the course of the rise in amplitude of the sound will be quite different for wind, string and plucked instruments. It's a very fine structure of transients, which over a very short period time, this new tone of a musical instrument provides the bulk of information about its color and texture. Studies show that the human ear is most sensitive to the initial part of the pulse of a new musical sound.

     

    Any disturbance or contamination of this sensitive time structure leads to a noticeable loss of sound quality from the perspective of people sensitive to audiophile aspects, such as nuances in fidelity transmission of all the colors of musical instruments.

     

    In other words, the time domain signal (issues phase, or timing aspects of the sound). In fact, these are the most fundamental parameters of the sound signal, because they lie in its creation - thus what a sound wave in the time domain actually looks like.

     

    So, one of the main advantages of vinyl is the lack of restrictions of temporal resolution in LP. One of the key challenges for us in the Pure Vinyl Club was to find a way (technology, a method of recording) the equipment to maintain a maximum level of temporal resolution from the LP while recording in digital. This does not mean that we were going to compromise or neglect other characteristics which are also important for the sound.

     

    Paweł Piwowarski in his article "PLIKI HI-RES - niezbędny krok do nirwany czy nadmiarowy gadżet?" on High Fidelity.pl in the October 2016 issue to which I referred above, noted that "The trumpet playing fortissimo contains transients of 40 kHz". I invite you to watch this little video using our LP rip, which clearly shows that transients of the trombone can get higher than 50 kHz, and trumpet reaches almost 70kHz!

     

    Later, in one of the following articles, which might be called "What is actually recorded on LP" I will showcase many interesting videos and screenshots, which clearly show that in many musical instruments transients exceed the 40-50 kHz threshold, and among them will be some unexpected ones (contrabass and sibilance of the human voice).

     

    Also, many audiophiles have prejudices about the LPs Dynamic Range. Here's a screenshot of the DR of an album's full side (Duration: 24:07, RAW Record).

     

     

     

    screnshot-DR.jpg

     

     

     

    I will focus on these and other interesting LP aspects in more detail in in the next articles of the Digital Vinyl series.

     

     

    Thank you,

     

    Igor

     

     

     

     

     

    Sound Samples

     

     

    Trippin (Kenny Drew – Trippin (1984, Japan) Promo WL, Baystate (RJL-8101))

    Official DR Value: DR13, Gain Output Levels (Pure Vinyl) – 14.00dB, Edit “Click Repair” – yes

     

    192 kHz / 24 bit (103MB)

     

     

     

    Play Fiddle Play (Isao Suzuki Quartet + 1 – Blue City (1974, Japan) Three Blind Mice (TBM-24))

    Official DR Value: DR13, Gain Output Levels (Pure Vinyl) – 8.02dB, Edit “Click Repair” – yes

     

    192 kHz / 24 bit (113MB)

     

     

     

    Make Someone Happy (Carmen McRae – Live At Sugar Hill San Francisco (1964, USA) Time Records (S/2104))

    Official DR Value: DR14, Gain Output Levels (Pure Vinyl) – 7.23dB, Edit “Click Repair” – yes

     

    192 kHz / 24 bit (99MB)

     

     

     

    Early In The Morning (John Henry Barbee, 1963

    VA – The Best Of The Blues (Compilation) (RE 1973, West Germany) Storyville (671188))

    Official DR Value: DR14, Gain Output Levels (Pure Vinyl) – 10.63dB, Edit “Click Repair” – yes

     

    192 kHz / 24 bit (75MB)

     

     

     

    La Cumparsita (Werner Müller And His Orchestra – Tango! (1967, USA) London Records (SP 44098))

    Official DR Value: DR11, Gain Output Levels (Pure Vinyl) – 0.00dB, Edit “Click Repair” – yes

     

    192 kHz / 24 bit (104MB)

     

     

     

    Wild Is The Wind (The Dave Pike Quartet Featuring Bill Evans – Pike’s Peak 1962 (RE 1981, USA) Columbia (PC 37011))

    Official DR Value: DR12, Gain Output Levels (Pure Vinyl) – 10.31dB, Edit “Click Repair” – yes

     

    192 kHz / 24 bit (109MB)

     

     

     

    People Are Strange (The Doors – 13 (1970, USA) Elektra (EKS-74079))

    Official DR Value: DR11, Gain Output Levels (Pure Vinyl) – 0.00dB, Edit “Click Repair” – yes

     

    192 kHz / 24 bit (82MB)

     

     

     

    Let’s Groove (Earth, Wind and Fire – Raise! (1981, Japan) CBS/Sony (25AP 2210))

    Official DR Value: DR15, Gain Output Levels (Pure Vinyl) – 7.89dB, Edit “Click Repair” – yes

     

    192 kHz / 24 bit (108MB)

     

     

     

    Smooth Operator (Sade – Smooth Operator (1984, Single, 45rpm, Japan) Epic (12・3P-581))

    Official DR Value: DR13, Gain Output Levels (Pure Vinyl) – 7.15dB, Edit “Click Repair” – yes

     

    192 kHz / 24 bit (114MB)

     

     

     

    Fernando (Paul Mauriat – Feelings (1977, 45rpm, Japan) Philips (45S-14))

    Official DR Value: DR14, Gain Output Levels (Pure Vinyl) – 5.76dB, Edit “Click Repair” – yes

     

    192 kHz / 24 bit (112MB)

     

     

     

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    Don't know if that was the reason for skipping CD or not. The SACD has a redbook layer.

     

    As for remastering, well the original was stereo DAT tape or according to some a Nakamichi PCM machine. With nothing done just put straight onto the CD. So any "remastering" is likely to be a step away from pure honest fidelity. Record two channels and distribute two channels is pure. Remastering means maybe EQ, maybe compression maybe lots of stuff. All of which will degrade pure fidelity. There was talk when this was announced they would reclock it for remastering. Reclocking can't fix original clocking problems at all. Reclocking has no place unless you first convert to analog for remastering processes. Adding AD/DA steps along with processing again steps you that much further away from simple direct fidelity. If this was recorded onto two track Nakamichi PCM at 16/44 that is as good as it can get. Every other version involves some additional steps none of which can improve fidelity.

     

    I'm not saying it's the case here, but if the recording device has, say, a non-flat frequency response, that can be corrected digitally. It might also be possible to remove some noise.

     

    Likewise, a touch of EQ can compensate for acoustic issues in the room such as an unwanted resonance. Of course this will deviate from what was actually heard during the recording, but it may well sound better.

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    As for the Nyquist-Shannon sampling theorem, it states that as long as the sampling frequency is more than double that of the bandwidth, the digital signal will contain ALL the information with NO loss and NO distortion. In the case of CDs, it means it will contain ALL the information up to 22 kHz, although in practice, it usually starts to roll off around 21 kHz. Yes, some vinyl records may contain actual music a tad higher in frequency than this, but the level will be so low that's inaudible - not to mention that it's already inaduble at those frequencies. In practice, most vinyl records are cut with a low-pass filter that rolls off frequencies above 15 or 18 kHz. Some go even lower. Listen to an original copy of Grand Funk's "We're an American Band". It seems to roll off everything above 5 kHz!

    The only "problem" with digital is the conversion, but as for instance Ethan Winer's loop-back test on his website shows, even going through 10 to 20 generations of conversion can be inaudible.

     

    Yes and nice summary.

     

    I have still seen NO evidence that people can hear a difference between an analogue master tape (or a vinyl record) and a properly digitized copy at 44.1 kHz/16 bit.

     

    You would agree that if someone were to present a "properly" digitized 44.1 vs 96 or 192 or even DSD256 and your were able to detect an audible difference *and* assuming there is > 22 kHz info in the hi-res recording, then that would be evidence of ultrasonic effects on hearing?

     

    Problem is definition of "properly" digitized. May not be easy.

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    I'm not saying it's the case here, but if the recording device has, say, a non-flat frequency response, that can be corrected digitally. It might also be possible to remove some noise.

     

    Likewise, a touch of EQ can compensate for acoustic issues in the room such as an unwanted resonance. Of course this will deviate from what was actually heard during the recording, but it may well sound better.

     

    Just wondering if anyone knows - could something like the Plangent process (correction for tape speed variation) be used for DAT, or is that irretrievably baked into the original digital file?

     

     

    Sent from my iPhone using Computer Audiophile

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    Yes and nice summary.

     

     

     

    You would agree that if someone were to present a "properly" digitized 44.1 vs 96 or 192 or even DSD256 and your were able to detect an audible difference *and* assuming there is > 22 kHz info in the hi-res recording, then that would be evidence of ultrasonic effects on hearing?

     

    Problem is definition of "properly" digitized. May not be easy.

     

     

    The bigger problem is side effects of ultrasonics. These are headaches, fatigue, and discomfort at industrial workstations. Count me among those who find the side effects of them unpleasant and when I'm around them my balance is affected as well.

     

    So why do we want them in our music? A nice summary is JOSE 2013, Volume 19, No. 2, Pages 195-202

     

    My overall feelings are summarized in Listening 165.

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    The bigger problem is side effects of ultrasonics. These are headaches, fatigue, and discomfort at industrial workstations. Count me among those who find the side effects of them unpleasant and when I'm around them my balance is affected as well.

     

    So why do we want them in our music? A nice summary is JOSE 2013, Volume 19, No. 2, Pages 195-202

     

    My overall feelings are summarized in Listening 165.

     

    Besides this, there are the possibilities of intermodulation distortion when ultrasonic frequencies are included.

     

    But people always, always, ALWAYS want more. So, in five years it will be 48 bit and 7 gazillion kHz sampling rates, because "bigger numbers are better" if you don't understand the technology. And that's the main problem: People don't understand the Nyquist sampling theorem, so they really do think that a higher sample rate more closely approximates an analogue sound wave. But how can a signal that already includes everything contain more than everything?

    The pictures we usually see are misleading and confusing. The only things that can exist between "the steps" on a digital signal are information at or above the Nyquist frequency (in the case of CDs, information at or above 22.05 kHz).

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    By the way: The Kunchur study I mentioned claimed the EXACT same thing as this article: That CDs can encode small enough increments. If you do the calculation I mentioned it will come to something like 1 nanosecond. That is the number you should look at!

    Another by the way: I was vinyl only for 15 years, but then compared almost 800 albums on vinyl and CD, then sold a lot of record, bought some of them again and now live and enjoy both, but I feel like I understand the technology behind digital now :-).

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    Besides this, there are the possibilities of intermodulation distortion when ultrasonic frequencies are included.

     

    But people always, always, ALWAYS want more. So, in five years it will be 48 bit and 7 gazillion kHz sampling rates, because "bigger numbers are better" if you don't understand the technology. And that's the main problem: People don't understand the Nyquist sampling theorem, so they really do think that a higher sample rate more closely approximates an analogue sound wave. But how can a signal that already includes everything contain more than everything?

    The pictures we usually see are misleading and confusing. The only things that can exist between "the steps" on a digital signal are information at or above the Nyquist frequency (in the case of CDs, information at or above 22.05 kHz).

     

    More than everything, I want it! ;)

     

    One other possibility with higher resolutions is intermodulation from ultrasonics that are actually signal rather than noise (e.g., cymbals, some of which have ~40% of their acoustic energy at ultrasonic frequencies). But that assumes a lot of good recording practices, and that playback systems could/would properly reproduce the intermodulation.

     

     

    Sent from my iPhone using Computer Audiophile

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    I'm so happy I can pick from a genius's brain !!

    What happens, what does the sampling theory say when the signal is no more than 20K and sampled at 44.1 but varies every 5 microseconds ?

    You can imagine a continuous decay but what would truly be worth of your genius would be bursts emitted every microsecond randomly in a 20_20K range sampled at 44.1

     

    I understand the SN sampling theory with a continuous frequency but just can't see how events faster than 1/44 100 second can merely be "flashed"

     

    You either take the time to study the maths, or you trust those who have.

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    I'm so happy I can pick from a genius's brain !!

    What happens, what does the sampling theory say when the signal is no more than 20K and sampled at 44.1 but varies every 5 microseconds ?

    You can imagine a continuous decay but what would truly be worth of your genius would be bursts emitted every microsecond in a 20_20K range sampled at 44.1

     

    A signal that is "no more than 20k" cannot by definition change faster than the maximum slope of a 20KHz sine wave. Any burst, pulse, square wave, with a vertical or near-vertical slope contains frequencies higher than 20KHz, so naturally a sample rate of approximately twice that would be insufficient.

     

     

    Sent from my iPhone using Computer Audiophile

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    I'm so happy I can pick from a genius's brain !!

    What happens, what does the sampling theory say when the signal is no more than 20K and sampled at 44.1 but varies every 5 microseconds ?

    You can imagine a continuous decay but what would truly be worth of your genius would be bursts emitted every microsecond randomly in a 20_20K range sampled at 44.1

     

    I understand the SN sampling theory with a continuous frequency but just can't see how events faster than 1/44 100 second can merely be "flashed"

    Such a signal has a bandwidth greater than 20 kHz.

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    More than everything, I want it! ;)

     

    One other possibility with higher resolutions is intermodulation from ultrasonics that are actually signal rather than noise (e.g., cymbals, some of which have ~40% of their acoustic energy at ultrasonic frequencies). But that assumes a lot of good recording practices, and that playback systems could/would properly reproduce the intermodulation.

     

     

    Sent from my iPhone using Computer Audiophile

     

    I believe you're right here. Also I remember that several tests have been done in the past (e.g. by Philips) to be able to determine the actual effects of ultrasonics. If I remember correctly ultrasonics themselves (naturally) may not be audible, but they do have an influence on the audible frequency range, and thus on the 'sound' of it. Probably it's some sort of intermodulation indeed. To me it does sound credible that music related ultrasonics in this way contribute to a different and probably more 'natural' sound.

     

    Sent from my HTC One_M8 using Computer Audiophile mobile app

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    Just wondering if anyone knows - could something like the Plangent process (correction for tape speed variation) be used for DAT, or is that irretrievably baked into the original digital file?

     

     

    Sent from my iPhone using Computer Audiophile

     

     

    That process uses the high frequency tape bias signal to determine when during playback scrape and flutter and wow has occurred. With some DSP help it can fix or mostly fix it using the high frequency bias as a sort of servo-mechanism.

     

    I don't know what you would use for the better clock with digital. Of course if your playback is clocked better in current gear there is some benefit vs playback with poorly clocked gear from back when. That isn't fixing any timing issues on the original however, just making sure you aren't adding to it. But perhaps some really smart people could think of something.

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    More than everything, I want it! ;)

     

    One other possibility with higher resolutions is intermodulation from ultrasonics that are actually signal rather than noise (e.g., cymbals, some of which have ~40% of their acoustic energy at ultrasonic frequencies). But that assumes a lot of good recording practices, and that playback systems could/would properly reproduce the intermodulation.

     

    Exactly -- ultrasonics -- just like regular sonics could be good or bad depending.

     

    The point is that if the are audible then if the violin at the symphony or the cymbal at the jazz club or the reed on the sax etc emit > 20 kHz then I want it in my recording.

     

    Likewise at work I don't want blasts of ultrasonics or drones of 8 kHz...

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    A signal that is "no more than 20k" cannot by definition change faster than the maximum slope of a 20KHz sine wave. Any burst, pulse, square wave, with a vertical or near-vertical slope contains frequencies higher than 20KHz, so naturally a sample rate of approximately twice that would be insufficient.

     

     

    Sent from my iPhone using Computer Audiophile

     

    Are you thinking for instance of some cymbal signals at 40 khz and 46 khz that intermodulate in the air into 6 khz tones? If so, if in the air, the 20 khz and below recording will capture that part without having to have response to 46 khz.

     

    It also is probably unlikely. Though some cymbals have 40% of energy in ultrasonics it is spread over as much as a dozen harmonics making each harmonic not that high in level. So any intermodulation is going to be even lower in level. Maybe not zero, but not much.

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    Besides this, there are the possibilities of intermodulation distortion when ultrasonic frequencies are included.

     

    But people always, always, ALWAYS want more. So, in five years it will be 48 bit and 7 gazillion kHz sampling rates, because "bigger numbers are better" if you don't understand the technology. And that's the main problem: People don't understand the Nyquist sampling theorem, so they really do think that a higher sample rate more closely approximates an analogue sound wave. But how can a signal that already includes everything contain more than everything?

    The pictures we usually see are misleading and confusing. The only things that can exist between "the steps" on a digital signal are information at or above the Nyquist frequency (in the case of CDs, information at or above 22.05 kHz).

     

    Not much evidence to support people want more. Hi-res is having a very tough go in the market. If people actually wanted it there would be more than under 16,000 albums currently available.

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    I'm so happy I can pick from a genius's brain !!

    What happens, what does the sampling theory say when the signal is no more than 20K and sampled at 44.1 but varies every 5 microseconds ?

    You can imagine a continuous decay but what would truly be worth of your genius would be bursts emitted every microsecond randomly in a 20_20K range sampled at 44.1

     

    I understand the SN sampling theory with a continuous frequency but just can't see how events faster than 1/44 100 second can merely be "flashed"

     

    I'll give another answer. Not really different than Jud or mansr. Just a different description because when I have seen this explained to other people it is hard for them to see sometimes. Often it takes two or three or four varieties of answers until one of them clicks with the questioner.

     

    If you are thinking of a signal that appears and departs each microsecond that would be a 1 mhz signal.

     

    Perhaps you are thinking of a burst of noise with 5 khz fundamental tone which appears and disappears each microsecond. It is easy to think that could happen. Many asking this question have that in their mind (don't know if you do or not). The answer is no 5 khz tone can come and go in one microsecond. In the air a 5khz would be a wave of pressure with a length of 2.7 inches (6.9 cm). If such a wave began, but then went away in one microsecond it can't be a 2.7 inch long wave it would be about 1/100th of an inch long. Such a wave isn't 5 khz, it is 1 megahertz. By starting and stopping the sound in that period of time it can't travel far enough to be a 5 khz wave. So you can't have a 1 microsecond portion of a 5 khz wave in the air.

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    The bigger problem is side effects of ultrasonics. These are headaches, fatigue, and discomfort at industrial workstations. Count me among those who find the side effects of them unpleasant and when I'm around them my balance is affected as well.

     

    So why do we want them in our music? A nice summary is JOSE 2013, Volume 19, No. 2, Pages 195-202

    Assuming the ultrasonics we are discussing are created by musical instruments and are present when listening to live music, there is no reason to believe that these specific ultrasonics are harmful in any way. The very fact that you can perceive ultrasonics at all suggests that their absence from a recoding may form part of the difference between hearing the performance "live" vs "recorded".

     

    I don't know anyone who gets headaches when listening up front and close to a quartet who likes classical music, same for jazz for people who like jazz. Perhaps its the ultrasonics which sends the "chill down your spine" when listening up front and close to live music?

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    Assuming the ultrasonics we are discussing are created by musical instruments and are present when listening to live music, there is no reason to believe that these specific ultrasonics are harmful in any way. The very fact that you can perceive ultrasonics at all suggests that their absence from a recoding may form part of the difference between hearing the performance "live" vs "recorded".

     

    I don't know anyone who gets headaches when listening up front and close to a quartet who likes classical music, same for jazz for people who like jazz. Perhaps its the ultrasonics which sends the "chill down your spine" when listening up front and close to live music?

     

    Ultrasonics contribute very little energy to the sound of most instruments, so my guess (it's only that) is *if* our experience is in any way diminished due to lack of ultrasonics, it would be because it is lacking just that last tiny bit to match the pattern that for us is the experience of a live violin, for example. (See my discussion of pattern matching in the other thread - was it the one about cables?). It would be something like CGI modeling of hair flowing and bouncing: so very, very close to perfect, but just not that last infinitesimal step to "real."

     

     

    Sent from my iPhone using Computer Audiophile

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    thank you. I'll sleep on your answers and maybe figure out. I did not have in mind that the bursts would stop and start in 1 microsecond. Just in mind the idea that the global envelope of an orchestra's sound is continuously changing and that in the context of this thread this should be represented by bursts of whatever audible frequencies emitted at a faster pace than the sampling rate. They don't have to stop and cycle in 1 microsecond : simply there's an event changing the global sound every microsecond. At the end of the day this might be as simple as the belief that more gives a better description but on the other hand why non quantic physic should be counterintuitive?

    I'll give another answer. Not really different than Jud or mansr. Just a different description because when I have seen this explained to other people it is hard for them to see sometimes. Often it takes two or three or four varieties of answers until one of them clicks with the questioner.

     

    If you are thinking of a signal that appears and departs each microsecond that would be a 1 mhz signal.

     

    Perhaps you are thinking of a burst of noise with 5 khz fundamental tone which appears and disappears each microsecond. It is easy to think that could happen. Many asking this question have that in their mind (don't know if you do or not). The answer is no 5 khz tone can come and go in one microsecond. In the air a 5khz would be a wave of pressure with a length of 2.7 inches (6.9 cm). If such a wave began, but then went away in one microsecond it can't be a 2.7 inch long wave it would be about 1/100th of an inch long. Such a wave isn't 5 khz, it is 1 megahertz. By starting and stopping the sound in that period of time it can't travel far enough to be a 5 khz wave. So you can't have a 1 microsecond portion of a 5 khz wave in the air.

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    thank you. I'll sleep on your answers and maybe figure out. I did not have in mind that the bursts would stop and start in 1 microsecond. Just in mind the idea that the global envelope of an orchestra's sound is continuously changing and that in the context of this thread this should be represented by bursts of whatever audible frequencies emitted at a faster pace than the sampling rate. They don't have to stop and cycle in 1 microsecond : simply there's an event changing the global sound every microsecond. At the end of the day this might be as simple as the belief that more gives a better description but on the other hand why non quantic physic should be counterintuitive?

     

    When you say that something can change the sound every microsecond, it depends by how much. When you see an orchestra's sound captured by mics and shown on an oscilloscope, the sound waves have a slope. In other words, the waves move a certain distance horizontally as they move vertically, and the ratio of these - how steep the steepest wave is - can't be steeper than a 20KHz sine wave, or a sample rate of around 44.1kHz won't be adequate to reproduce it.

     

    A 20KHz sine wave won't move vertically very much at all in one millionth of a second horizontal distance on the oscilloscope, since the wave will take 1/20,000 of a second to complete. That is 50 microseconds, so only 1/50th of the total vertical path of the wave will be complete in a microsecond.

     

     

    Sent from my iPhone using Computer Audiophile

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    Not much evidence to support people want more. Hi-res is having a very tough go in the market. If people actually wanted it there would be more than under 16,000 albums currently available.

    I'm not just talking about audio, but human nature in general. If people had been taught to be content with what they have, hi-res would never have been launched, nor would CDs for that matter. We are brought up on dreams and being told that we can achieve anything we set our minds to. So, this attitude of "wanting more" is what made us evolve from half-monkeys banging bones in caves to what we are today. Wanting more makes "losers" turn their lives around and become part of the Fortune 500, etc.

    But specifically in the realm of audiophilea, people usually want to try new speakers, new amps, new this, new that, in the hopes of getting closer to perfection. I've met people who've considered paying $35,000 for speakers and still said they had never had a stereo system they were content with. That says something about always wanting more. Buying a nice stereo system made me the same way. Before that I had tolerated three reasonable systems for 20+ years, but now I have to look at new things constantly :-/.

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    I believe you're right here. Also I remember that several tests have been done in the past (e.g. by Philips) to be able to determine the actual effects of ultrasonics. If I remember correctly ultrasonics themselves (naturally) may not be audible, but they do have an influence on the audible frequency range, and thus on the 'sound' of it. Probably it's some sort of intermodulation indeed. To me it does sound credible that music related ultrasonics in this way contribute to a different and probably more 'natural' sound.

     

    Sent from my HTC One_M8 using Computer Audiophile mobile app

    And the effect ultrasonics may have on the content in the audible spectrum is, of course, already included in the audible signal to begin with.

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    I'm so happy I can pick from a genius's brain !!

    What happens, what does the sampling theory say when the signal is no more than 20K and sampled at 44.1 but varies every 5 microseconds ?

    You can imagine a continuous decay but what would truly be worth of your genius would be bursts emitted every microsecond randomly in a 20_20K range sampled at 44.1

     

    I understand the SN sampling theory with a continuous frequency but just can't see how events faster than 1/44 100 second can merely be "flashed"

     

    You probably didn't see this, but I mentioned a few posts earlier that the calculation is not 1/44 100 second, but rather this: 1/(20000*2*pi*65536)

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    thank you. I'll sleep on your answers and maybe figure out. I did not have in mind that the bursts would stop and start in 1 microsecond. Just in mind the idea that the global envelope of an orchestra's sound is continuously changing and that in the context of this thread this should be represented by bursts of whatever audible frequencies emitted at a faster pace than the sampling rate. They don't have to stop and cycle in 1 microsecond : simply there's an event changing the global sound every microsecond. At the end of the day this might be as simple as the belief that more gives a better description but on the other hand why non quantic physic should be counterintuitive?

     

    Okay think of this. A 1 khz sine wave recorded by a microphone is changing all the time. 44.1 khz systems can sample and fully reconstruct that wave. If the recorded wave were 1 volt peak at peak maximum it will be 1 volt and one microsecond later it will be .99998 volts approximately. Our little 44.1 khz sample rate has managed to create this signal which is changing all the time. A continuously varying signal even at the smallest time intervals.

     

    Now a constraint of discontinuously sampling a continuous waveform means above some frequency our sampling and reconstructing methods would not work correctly. So we are limited to a bit under half the sample rate. For all signals not filtered out however any possible waveform with no higher frequencies is reconstructed to change all the time. Even if the signal changes are in between samples even if new signals start or stop between samples.

     

    In the case of our ears or microphones and such they also change continuously with the continuously changing waveform even at the smallest intervals of time. Due to the flexibility, weight, physical impedance to the air and damping our ear drums or microphone diaphragm at some high frequency will no longer couple to the air and can't change fast enough to mirror some part of the signal. They still change continuously, but for instance a 200,000 hz portion of the signal changes so fast our eardrums never move in response to that signal. So were it there in isolation or added to lower frequencies our ears drums aren't responding to it.

     

    So if our discontinuously sampled digital recording has enough sample rate to record and reconstruct frequencies equal to or exceeding our ear's response higher frequencies will not matter even though filtered out. The signal that is reconstructed will vary rapidly enough our ear could not detect it any faster.

     

    Maybe this helps instead of muddying the water.

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    And the effect ultrasonics may have on the content in the audible spectrum is, of course, already included in the audible signal to begin with.

     

    No. The effect of the ultrasonics would be on the nervous system to modulate the audible input to the nervous system.

     

    There should be no assumption that there exists an equivalent purely audible signal that produces the same effect in the nervous system.

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