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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)

     

     

     

    1-Pixel.png




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    Regardless of any merit the article may have, one has to be suspicious when Art Dudley is used as a reference for a technological discussion.

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    Ah- there's the disconnect.

     

    This article is about the Temporal domain. Not the time domain.

     

    The spatial and temporal domains have been used for imagery resolution for 70+ years.

     

    If you are willing to accept certain simplifications, such as all sound is only sinusoidal plane waves. That is a proposition.

     

    Sheffield Labs did a good job 35+ years ago demonstrating that 50kHz clearly demonstrated a number of audiophonic differences. On just stereo platters. Open fields vs closed halls vs anechoic chambers. Placing microphones on instruments vs placing microphones at a listener's distance and location.

     

    MQA starts off with a simple dismissal that above 30 kHz contributions for phase and harmonics are not "requirements". That's a sales proposition.

     

    Since Fourier's time- our understanding of sound and perception has changed greatly. So you will need a tool kit larger than just the one. Which in part is the the thesis of this article's author.

     

    There are now at least 6 attributions. (IMHO- at least a 7th, soundscape, should be a peer to the other six.)

     

    Even when one tries to do the cepstral domain using Fourier (the cepstrum is defined as the inverse DFT of the log magnitude of the DFT of a signal) has given rise to these other attribution sets. {Cepstral domain was an attempt to get a signal that looks more like what our ears would actually hear it as opposed to how it actually looks according to the electronic sensors.}

     

    The flag flown in our face ought to be the recording industry's own representations. I am putting a picture here of the encoders. Note the psychoacoustic model box. Another estimation (not authentic or mathematical certainty) of the original sound. Unless "signal" for a radio transceiver is the same as the perceived sound.

     

    And this author starts from a position that his cognition isn't satisfied by that simplification.

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    Ah- there's the disconnect.

    This article is about the Temporal domain. Not the time domain.

    ...

     

    And the difference is... ?

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    considering the definition of temporal is either "of relating to time" or "concerning present life or this world" in this context how could the time domain not also be the temporal domain.

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    the cepstrum is defined as the inverse DFT of the log magnitude of the DFT of a signal

     

     

    OK, found this portion of the copy-and-paste from a Texas A&M University Intro to Speech Processing course. Wonder where the rest came from?

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    Since Fourier's time- our understanding of sound and perception has changed greatly. So you will need a tool kit larger than just the one. Which in part is the the thesis of this article's author.

     

    The Fourier transform is still valid. It's a proven mathematical fact, and as such it will always be true.

     

    Nothing of what you say makes any sense.

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    Moreover, those mathemagicians were concerned with mathematics. Not sound, not audio, not perception.

    And when they wanted some music they could go out and attend a WAM gig.

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    Moreover, those mathemagicians were concerned with mathematics. Not sound, not audio, not perception.

    And when they wanted some music they could go out and attend a WAM gig.

     

    Hey! I've been to a WHAM gig, Good times!

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    Regardless of any merit the article may have, one has to be suspicious when Art Dudley is used as a reference for a technological discussion.

     

    This article is specifically written in a clear, easy form, which would be 90% of the readers on the CA, which are not interested in all this higher mathematics and who just want to know what they need for good listening to your favorite music.

    It was not meant for technological discussion. And only a small passage contains some technical information and is the cause of all this disagreement and criticism (due to misunderstanding or unwillingness to understand that in fact the article is written about something else).

    Maybe it's my fault, because I do not know good English and I use Google translator.

     

    Quotation from an interview with Art Dudley was used precisely because it is directly related to the material of this article. And it is to this passage. Or is Art Dudley not good enough to quote? To make it clear what exactly Art Dudley wanted to say, and in the context of which he said this, I provide a larger excerpt from this interview. Sorry, but the interview is only in Polish and I again have to use Google translator:

     

    "Wojciech Pacuła: What are then the greatest sins of modern audio?

    Art Dudley: In my opinion the two sins are the worst:

    1) We wrangled down, with every movement, by those who point to this or that element of design or construction, and say, "It does not matter." I have one answer for them: "Bullshit, EVERYTHING is important!"

    We hear, however, again and again. Manufacturers claim that it does not matter what material performed amplifier housing. From engineers to remasteringiem who think it does not matter that the board LP was incised with digital tape (or using a digital delay). From the people, by which high resolution is not important, because the Nyquist frequency for CD 44.1 kHz is sufficient.

    The latter is particularly problematic when we realize that the Nyquist frequency does not apply to work and reconstruction filters decymacyjnych composite signal. Indeed, the two samples may be used to describe a single frequency, but do not provide a sufficient density of the samples to describe the speed with which the signal increases or decreases - and this is a key distinction between the music and mere sound."

     

    High Fidelity

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    This article is specifically written in a clear, easy form,

     

    Clear, easy, and 100% wrong.

     

    which would be 90% of the readers on the CA, which are not interested in all this higher mathematics and who just want to know what they need for good listening to your favorite music.

    It was not meant for technological discussion. And only a small passage contains some technical information and is the cause of all this disagreement and criticism (due to misunderstanding or unwillingness to understand that in fact the article is written about something else).

    Maybe it's my fault, because I do not know good English and I use Google translator.

     

    Quotation from an interview with Art Dudley was used precisely because it is directly related to the material of this article. And it is to this passage. Or is Art Dudley not good enough to quote? To make it clear what exactly Art Dudley wanted to say, and in the context of which he said this, I provide a larger excerpt from this interview. Sorry, but the interview is only in Polish and I again have to use Google translator:

     

    "Wojciech Pacuła: What are then the greatest sins of modern audio?

    Art Dudley: In my opinion the two sins are the worst:

    1) We wrangled down, with every movement, by those who point to this or that element of design or construction, and say, "It does not matter." I have one answer for them: "Bullshit, EVERYTHING is important!"

    We hear, however, again and again. Manufacturers claim that it does not matter what material performed amplifier housing. From engineers to remasteringiem who think it does not matter that the board LP was incised with digital tape (or using a digital delay). From the people, by which high resolution is not important, because the Nyquist frequency for CD 44.1 kHz is sufficient.

    The latter is particularly problematic when we realize that the Nyquist frequency does not apply to work and reconstruction filters decymacyjnych composite signal. Indeed, the two samples may be used to describe a single frequency, but do not provide a sufficient density of the samples to describe the speed with which the signal increases or decreases - and this is a key distinction between the music and mere sound."

     

    High Fidelity

     

    Bullshit. That quote is a perfect example of why Art Dudley is not to be trusted on technical matters. He talks a lot, but he clearly has no clue whatsoever.

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    This article is specifically written in a clear, easy form, which would be 90% of the readers on the CA, which are not interested in all this higher mathematics and who just want to know what they need for good listening to your favorite music.

    "Wojciech Pacuła: What are then the greatest sins of modern audio?

    Art Dudley: In my opinion the two sins are the worst:

    1) We wrangled down, with every movement, by those who point to this or that element of design or construction, and say, "It does not matter." I have one answer for them: "Bullshit, EVERYTHING is important!"

    We hear, however, again and again. Manufacturers claim that it does not matter what material performed amplifier housing. From engineers to remasteringiem who think it does not matter that the board LP was incised with digital tape (or using a digital delay). From the people, by which high resolution is not important, because the Nyquist frequency for CD 44.1 kHz is sufficient.

    The latter is particularly problematic when we realize that the Nyquist frequency does not apply to work and reconstruction filters decymacyjnych composite signal. Indeed, the two samples may be used to describe a single frequency, but do not provide a sufficient density of the samples to describe the speed with which the signal increases or decreases - and this is a key distinction between the music and mere sound."

     

    High Fidelity

    I think it's a great shame that rather than sticking to "I like the sound of this", people have to grasp for pseudo-technical explanation for their preferences. AS with Mr Dudley's interview, the article in the OP produces entirely spurious arguments for the preference for vinyl. This is painfully obvious, but for some reason it is a persistent mistake. The heart of the matter does not lie in physics, maths or information theory but in psychoacoustics. Human beings (even audiophiles) are not sound quality measuring machines; or if they are they are not good ones. The experience of hearing depends only partially on fluctuations in air pressure in the region of the ears which we call "sound": it depends to a very large extent on layers of processing of the neural information generated by those sound pressure waves and other sensory information (including sight). People sometimes like distortion and noise even if they think that they are enjoying something else.

     

    It is easy to explain why (some) people prefer vinyl (sometimes) in terms which do not require one to mangle, trivialise or distort any laws of physics, maths or information theory, or to imagine that one can hear sounds over 20khz when one in fact probably can't hear over 15khz. The only difficult thing about it is giving up the notion that what one loves is fidelity/accuracy etc in any objective sense.

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    Or is Art Dudley not good enough to quote?

     

     

    [/i]High Fidelity

     

    Exactly. He is not qualified to discuss the science or mathematics involved in sound reproduction.

     

    By using the quote you did, you are implying that he is an expert in the technology of digital recording. In fact, he as no scientific or mathematical qualifications and has never written anything that would lead one to believe otherwise. His forte is discussing what he hears, not the technology behind it.

     

    Quote him about what he hears, not why.

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    I think it's a great shame that rather than sticking to "I like the sound of this", people have to grasp for pseudo-technical explanation for their preferences. AS with Mr Dudley's interview, the article in the OP produces entirely spurious arguments for the preference for vinyl. This is painfully obvious, but for some reason it is a persistent mistake. The heart of the matter does not lie in physics, maths or information theory but in psychoacoustics. Human beings (even audiophiles) are not sound quality measuring machines; or if they are they are not good ones. The experience of hearing depends only partially on fluctuations in air pressure in the region of the ears which we call "sound": it depends to a very large extent on layers of processing of the neural information generated by those sound pressure waves and other sensory information (including sight). People sometimes like distortion and noise even if they think that they are enjoying something else.

     

    It is easy to explain why (some) people prefer vinyl (sometimes) in terms which do not require one to mangle, trivialise or distort any laws of physics, maths or information theory, or to imagine that one can hear sounds over 20khz when one in fact probably can't hear over 15khz. The only difficult thing about it is giving up the notion that what one loves is fidelity/accuracy etc in any objective sense.

     

    It seems, this whole thing has also just degenerated into yet another "CD vs. vinyl" religious war. You always accuse me of opposing vinyl and digital. Show me a place in my two articles and comments, where I praise LP and disparage DIGITAL. All just the opposite. There is no TT in any of my sound systems. And I do not have any of my LPs, none at all! And never will be!

     

    TT, on which I make rips with LP is in the studio and it is integrated into professional equipment. And even in the studio, I never hear an "analog" sound from the LP, only DIGITAL. Because the phono preamp that I use for recording does not have a RIAA curve corrector. Only the "flat" XLR output and RIAA are superimposed in the digital domain. I always hear from the LP only a digital sound and I really like what I hear.

     

    This is the essence of this project - to make the most effective TT and with the help of high-class professional equipment to make the most effective LP rip. And that's all! Farewell to LP! Put it on the shelf in the old closet and forget how terrible a dream, all those inconveniences and problems with LP. Just this! It is the high quality of the DIGITAL equipment and software that has allowed to achieve very good results. Now you can write down everything that is on LP, with great reserve, absolutely all the information.

     

    And yes, I really like high fidelity when listening to music. In the studio, I listen to professional monitors. But at home I do not have any tube amplifiers. Only active pro monitors of different sizes (4", 5", 6", 8"). And my main setup is the Grimm Audio LS-1, which many consider the most high fidelity system in the world.

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    It seems, this whole thing has also just degenerated into yet another "CD vs. vinyl" religious war. You always accuse me of opposing vinyl and digital. Show me a place in my two articles and comments, where I praise LP and disparage DIGITAL. All just the opposite. There is no TT in any of my sound systems. And I do not have any of my LPs, none at all! And never will be!

     

    TT, on which I make rips with LP is in the studio and it is integrated into professional equipment. And even in the studio, I never hear an "analog" sound from the LP, only DIGITAL. Because the phono preamp that I use for recording does not have a RIAA curve corrector. Only the "flat" XLR output and RIAA are superimposed in the digital domain. I always hear from the LP only a digital sound and I really

     

    "So, one of the main advantages of vinyl is the lack of restrictions of temporal resolution in LP. "

    I can't help it if you can't follow your own argument. There is no such advantage. If you want to capture everything on an LP., make sure you record at least 10/32

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    "Wojciech Pacuła: What are then the greatest sins of modern audio?

    Art Dudley: In my opinion the two sins are the worst:

    1) We wrangled down, with every movement, by those who point to this or that element of design or construction, and say, "It does not matter." I have one answer for them: "Bullshit, EVERYTHING is important!"

    We hear, however, again and again. Manufacturers claim that it does not matter what material performed amplifier housing. From engineers to remasteringiem who think it does not matter that the board LP was incised with digital tape (or using a digital delay). From the people, by which high resolution is not important, because the Nyquist frequency for CD 44.1 kHz is sufficient.

    The latter is particularly problematic when we realize that the Nyquist frequency does not apply to work and reconstruction filters decymacyjnych composite signal. Indeed, the two samples may be used to describe a single frequency, but do not provide a sufficient density of the samples to describe the speed with which the signal increases or decreases - and this is a key distinction between the music and mere sound."

     

    High Fidelity

     

    What? If your two samples are sampling a large signal the sample values would be higher. If a lower signal it will be lower. If it goes from low to high rapidly sample values will increase rapidly. If the signal changes too much between samples you have clipping (from too large a signal). If the signal changes one way and then diminishes in the other direction it by definition is a frequency above the sample rate. In competent gear that will have been filtered out so as not to happen.

     

    Another name for the speed at which a signal changes is slew rate. Slew rate is not just about frequency. People think 30 khz is fast the rate of change is so rapid. Yet a 3000 hz tone at max level has the same max slew rate as 30,000 hz at -20 db. Even cymbals have their ultrasonic level down more than 20 db versus the max level at lower frequencies. The speed of change is not even close to a problem. If it is CD then it will have been filtered out, but the rate of change is not stressing the digital sampling.

     

    Here is a page with cymbal reviews by John E Johnson Jr.

    Cymbal Reviews with Spectral Analysis - DRUMMERWORLD OFFICIAL DISCUSSION FORUM

     

    Better still here are some files recorded with an Earthworks microphone calibrated to 30 khz. It likely responds pretty nicely beyond 30 khz, but that is the calibrated bandwidth. Recordings were at 176 khz/24.

     

    Welcome Secrets of Home Theater and High Fidelity

     

    The other deceptive picture people have in their heads is things like guitar strings and cymbals get plucked or smacked and go straight to high frequency max level. This however is not the case. It takes a few cycles to build to max level.

     

    Here is a picture of the fastest beginning transient I found on that page with the recorded cymbals. I didn't look at all of them, but this is one of the faster ones. Notice it takes something like 8 cycles to go from silent to maximum sound level after it is struck. I have seen similar behaviour from stringed instruments, and brass instruments during 'sharp' transients. After reaching max level it goes into approximately an exponential decay.

     

    initial transient mega power ride ping.png

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    Another name for the speed at which a signal changes is slew rate. Slew rate is not just about frequency. People think 30 khz is fast the rate of change is so rapid. Yet a 3000 hz tone at max level has the same max slew rate as 30,000 hz at -20 db.

     

    Slew rate is proportional to frequency times amplitude. This is why opamps are usually specified with a gain-bandwidth product.

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    Here you can see the absence of temporary errors in vinyl and DSD:

     

    13.gif

     

    Here in the DAC's NOS:NOS

     

    There should be two categories of recordings:For home: DSD 5.6MHz and at least 30dB dynamic range - or, if preferred PCM, 24/192 and at least 30dB dynamic range. In this case recommend the use of DAC's NOS.For the car: DSD 2.8MHz and at least 10dB dynamic range - or 16/48 and at least 10dB dynamic range.The DSD does not even need a DAC to be played:http://Http://www.diyaudio.com/forums/digital-line-level/273474-best-dac-no-dac.htmlDynamic range problems have been reported for years by Recording Engineers, and those of temporary errors by such well-known engineers as Bob Stuart of Meridian.Let us imagine that we are talking about the fraud of the VW emissions, which was kept secret, and when it was discovered the legislators legislated in favor of the consumer, the injured, and against the manufacturer, the fraudster.

    The situation in Audio is as if the legislators had punished to the consumers to contaminate in the case of the emissions. Record companies and musicians have seriously degraded the quality of the recordings, applying more and more compression, which currently tends towards the 4dB of dynamic range, but charging them as if they were products of the highest quality, and legislators have punished the Consumers for not wanting to pay for substandard products.Lawmakers should oblige record companies and musicians to respect the quality standards I have exposed, since they are the fraudsters, not the consumers, and see how the situation changes and consumers pay for good products, the proof is in the vinyl.And there would be a rather simple way of eliminating, if it would seriously hinder piracy, but it is something that I certainly will not talk about as long as this consumer fraud situation persists. Fraud that lasts for more than 35 years, is enough.Greetings.

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    Here you can see the absence of temporary errors in vinyl and DSD:

     

    13.gif

     

    Here in the DAC's NOS:NOS

     

    There should be two categories of recordings:For home: DSD 5.6MHz and at least 30dB dynamic range - or, if preferred PCM, 24/192 and at least 30dB dynamic range. In this case recommend the use of DAC's NOS.For the car: DSD 2.8MHz and at least 10dB dynamic range - or 16/48 and at least 10dB dynamic range.The DSD does not even need a DAC to be played:http://Http://www.diyaudio.com/forums/digital-line-level/273474-best-dac-no-dac.htmlDynamic range problems have been reported for years by Recording Engineers, and those of temporary errors by such well-known engineers as Bob Stuart of Meridian.Let us imagine that we are talking about the fraud of the VW emissions, which was kept secret, and when it was discovered the legislators legislated in favor of the consumer, the injured, and against the manufacturer, the fraudster.

    The situation in Audio is as if the legislators had punished to the consumers to contaminate in the case of the emissions. Record companies and musicians have seriously degraded the quality of the recordings, applying more and more compression, which currently tends towards the 4dB of dynamic range, but charging them as if they were products of the highest quality, and legislators have punished the Consumers for not wanting to pay for substandard products.Lawmakers should oblige record companies and musicians to respect the quality standards I have exposed, since they are the fraudsters, not the consumers, and see how the situation changes and consumers pay for good products, the proof is in the vinyl.And there would be a rather simple way of eliminating, if it would seriously hinder piracy, but it is something that I certainly will not talk about as long as this consumer fraud situation persists. Fraud that lasts for more than 35 years, is enough.Greetings.

     

    Really pitiful post. Apparently your first as well.

     

    The analog fiction in the graphic, I hope you realize your analog is not really going to look like that. At least not on tape or on LP. That signal as an input would have a bandwidth of several hundred kilohertz. So no surprise that a band limited digital recording doesn't reproduce it completely. Equally fictitious is the DSD representation. DSD incorporates an ultrasonic filter at the output and with it in place it won't look like the fiction in the graphic when it comes out the other end.

     

    People do love fiction.

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    The graphic is from a Professional Audio company, it is not fiction. Fiction is the so-called 'perfection' of digital audio.---The correct link: The Best DAC is no DAC - diyAudio

     

    Yes, it is fiction. Yes the graph was drawn up by a commercial audio company, and it still is fiction. If there is something in that other forum directly relevant it would be good to link to it. Not likely to read 177 pages for whatever is related to your post.

     

    Try running the impulse of the graph thru a tape machine or across the cutter head for an LP and let me know how it comes out. Do you think your eardrum can respond like the impulse?

     

    Digital audio is not perfect. It is very good, and it does what it is claimed to do. Pretending otherwise doesn't help matters. But that lacks a self-satisfying narrative.

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    It is worth pointing out that a nos dac demonstrably produces the wrong value at all points between the sampling instants. These are gross time domain errors and occur at all frequencies, unlike those of the pcm system which are tiny errors occurring only at the transition zone of the anti imaging/ anti aliasing frequencies. Of course in order to understand that one has to be interested in the facts not Sony marketing brochures.

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    only at the transition zone of the anti imaging/ anti aliasing frequencies.

    that should read "only in the transition band of the anti imaging/ anti aliasing filters" [why can't I edit the post?]

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    Here you can see the absence of temporary errors in vinyl and DSD:

     

    13.gif

     

    Here you are slightly incorrectly pointing to vinyl, although in fact here it is said about ANALOG (that is what a best microphone can register). Vinyl will never be able to repeat this result without loss - too much interference will occur in the way of the LP creation process. And even more distortion will be when trying to play LP on your TT.

     

    In fact, this is a popular picture with a chart showing ANALOG (signal from the best microphone), as a reference standard. And the loss (or lack of loss, as they represent -))), in the case of DSD) in the temporal domain of the impulse response and energy when trying to register and then reconstruct this signal with the help of various digital standards (48, 96,192 and DSD).

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    Here you can see the absence of temporary errors in vinyl and DSD:

     

    13.gif

     

    Here you are slightly incorrectly pointing to vinyl, although in fact here it is said about ANALOG (that is what a best microphone can register). Vinyl will never be able to repeat this result without loss - too much interference will occur in the way of the LP creation process. And even more distortion will be when trying to play LP on your TT.

     

    In fact, this is a popular picture with a chart showing ANALOG (signal from the best microphone), as a reference standard. And the loss (or lack of loss, as they represent -))), in the case of DSD) in the temporal domain of the impulse response and energy when trying to register and then reconstruct this signal with the help of various digital standards (48, 96,192 and DSD).

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    I really do dislike these sorts of graphics (the impulse ones as presented here, and the square wave ones I've seen as well), because they are fiction for an additional reason.

     

    These impulses contain a very large frequency range that is not limited to the range of audibility (~20kHz or less), 24kHz, 48kHz, or even 96kHz. *That* is why you see the "ringing" (the additional waves) and the inability to reconstruct the full amplitude until you get to DSD frequency levels (2.8MHz).

     

    Remember the Sampling Theorem: You must sample at above double the highest "frequency of interest."

     

    All graphs like the above demonstrate is that sampling doesn't work well when the Sampling Theorem's conditions are violated. Duh. So please, folks reading marketing material that is couched in the form of technical papers: be careful.

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