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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.

     

     

     

    1-Pixel.png

    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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    "Can you give an example of a signal condition which has no content above 20 khz, and yet is not resolved between samples at the sample rate of 44.1 khz?"

     

     

    .... not likely.

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    "Sorry for the long post."

     

    You'd better apologise for the nonsense in that post.

    Pray go back to school.

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    This thread has cleared up my misconceptions on the accuracy achievable with normal sampling methods, and I thank Jud, esldude and others for taking the time. I am now left pondering why MQA use a different sampling method. From their wiki page:-

     

    "One more difference to standard formats is the sampling process. The audio stream is sampled and convolved with a triangle function, and interpolated later during playback. The techniques employed, including the sampling of signals with a finite rate of innovation, were developed by a number of researchers over the preceding decade, including Pier Luigi Dragotti and others."

     

    Does anyone have an explanation ( none cynical please)?

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    This thread has cleared up my misconceptions on the accuracy achievable with normal sampling methods, and I thank Jud, esldude and others for taking the time. I am now left pondering why MQA use a different sampling method. From their wiki page:-

     

    "One more difference to standard formats is the sampling process. The audio stream is sampled and convolved with a triangle function, and interpolated later during playback. The techniques employed, including the sampling of signals with a finite rate of innovation, were developed by a number of researchers over the preceding decade, including Pier Luigi Dragotti and others."

     

    Does anyone have an explanation ( none cynical please)?

     

    The only explanations I can think of are cynical. Sorry.

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    This thread has cleared up my misconceptions on the accuracy achievable with normal sampling methods, and I thank Jud, esldude and others for taking the time. I am now left pondering why MQA use a different sampling method. From their wiki page:-

     

    "One more difference to standard formats is the sampling process. The audio stream is sampled and convolved with a triangle function, and interpolated later during playback. The techniques employed, including the sampling of signals with a finite rate of innovation, were developed by a number of researchers over the preceding decade, including Pier Luigi Dragotti and others."

     

    Does anyone have an explanation ( none cynical please)?

     

    It actually is not a "different" sampling method than anyone else has used. Filters are constructed by convolving a filter "kernel" with a function, such as (classically) a sinc function. MQA uses one of the available kernel choices, a triangular kernel.

     

    Miska describes a triangular kernel in technical terms as "crappy." :)

     

     

    Sent from my iPhone using Computer Audiophile

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    The argument Pure Vinyl seems to be trying to make here is that there can be two different 10kHz waveforms, where additional "precision" is required to reconstruct the two different, but same kHz tone waves. This has always struck me as the "my bass drum has an attack shape in its waveform that exists at 50Hz that makes it sound different than other 50kHz notes. Isn't the real science that my 50Hz bass drum wave is accompanied by higher spectral content, i.e. at 10kHz and 30kHz that further informs the sound and you need to preserve all of the content in order to capture the original sound?

    Hi, sdolezalek

     

    It doesn't affect the precision at all. Or any other *individual* frequency (taken by itself) that can be reproduced with that sample rate.

     

    What is important is that the time relationships *between* different frequencies is better preserved by using a higher sample rate. But not the precision of reproducing any single frequency.

     

    And yes, by the way, I, too, I can not hear 16kHz

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    Easily confused, yes. That's why it is a common misconception. You are still making the same mistake in thinking that digital sampling resolution is limited to the time between samples.

     

    This is an unfortunate choice of analogy. It argues for accuracy / resolution in amplitude, not time, which is also an area where digital has a large superiority in resolution over vinyl.

     

    Here you repeat your misunderstanding. You say that an event that occurs between sample times can only be captured to a time matching the closest sample time. This is incorrect. It is captured accurately, to the resolution defined by the equation in my earlier post. Your continued failure to understand how this can be so shows you lack the "training and experience on the subject" you talked of above. Have you watched the video I posted the link to? Starting from about 17:20 it shows, in real life, with a cheap DAC at 16/44.1, a transient being accurately sampled in time between sample times.

     

    Here you display a lack of understanding of digital oscillosopes and how they are used. The biggest difference is that they do not depend on being able to sample at twice the rate of the highest frequency being measured. It's quite common to use such a scope in an undersampled mode. But for our purposes (audio), we stick to the classic Shannon-Nyquist model.

     

    Make up your mind. Either digital can resolve transients in between samples, or it can't.

     

    In the rest of your post, all you have done is make the case that 16/44.1 is marginal (but close enough for rock'n'roll), and that 24/96 is enough to capture everything that might possibly be significant.

     

     

    As long as there's nothing in the signal you're sampling above 48KHz.

     

     

    Wonder how much of the response from vinyl above 20KHz is truly signal as opposed to noise.

     

     

    Sent from my iPhone using Computer Audiophile

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    By the way: None of what's been said should obscure the fact that while CD is capable of greater dynamic range than vinyl, CDs are all too often actually produced with a squashed dynamic range. No hocus pocus is necessary to explain this, only production decisions.

     

     

    Sent from my iPhone using Computer Audiophile

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    It doesn't affect the precision at all. Or any other *individual* frequency (taken by itself) that can be reproduced with that sample rate.

     

    What is important is that the time relationships *between* different frequencies is better preserved by using a higher sample rate. But not the precision of reproducing any single frequency.

     

    That makes no sense at all.

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    Here is something with an impulse.

     

    And for those unconvinced by numerical calculations, let's fire up the scope again.

     

    This is a single-sample impulse at 44.1 kHz with the right channel delayed one sample:

     

    tek00000.png

     

    As expected, we get the familiar sinc response with an offset of 23 μs.

     

    Next, we keep the left channel unchanged while replacing the right channel with the sinc function evaluated at half-sample positions:

     

    tek00001.png

     

    Same response, but now with half the delay between channels.

     

    Now a quarter sample delay:

     

    tek00002.png

     

    Again, the delay is reduced as it should.

     

    Finally, a delay of 1/100th of a sample period:

     

    tek00003.png

     

    Still the expected result.

     

    Recall that an impulse contains all frequencies up to the Nyquist limit. Since the response can be shifted by a tiny fraction of a sample period, the timing (phase) of every frequency must be accurately represented. In other words, there is nothing special about pure sine tones as has been suggested by some.

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    By the way: None of what's been said should obscure the fact that while CD is capable of greater dynamic range than vinyl, CDs are all too often actually produced with a squashed dynamic range. No hocus pocus is necessary to explain this, only production decisions.

    "The first cartridge with integrated RF shielding, the Clearaudio Goldfinger Statement MC phono cartridge is designed to play on the world's finest analog systems and reproduce the music on the greatest recordings ever made with unstinting clarity, liveliness, dynamics, realism, and responsiveness. Featuring 12 perfectly matched and symmetrical magnets – an unprecedented achievement – surrounding its coils. The Product of the Year Award-winning Goldfinger Statement allows systems to reach the long-unattainable dynamic range of 100dB." https://www.musicdirect.com/store/clearaudio-goldfinger-statement-mc-cartridge

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    As long as there's nothing in the signal you're sampling above 48KHz.

     

     

    Wonder how much of the response from vinyl above 20KHz is truly signal as opposed to noise.

     

     

    Sent from my iPhone using Computer Audiophile

     

    Well that is the other part. I have taken some of the downloads offered, steep filtered everything below 20 khz, then slowed it down to 25% normal speed so I could hear ultrasonics. There is very little up there. It is mostly surface noise and widely spaced bits of cymbals at low levels.

     

    It has been mentioned there is a slow roll off past 50 khz. Well not usually. 99% of LPs prior to digital were sourced from analog tape. Such tape isn't flat to 50 khz. Such tape also has scrape and flutter to effect timing. It certainly doesn't have infinite temporal resolution. It some situations the altered phase of high frequencies as the tape heads roll off could interact with drum-shots to cause something like pre-ringing at a lower frequency. Only unlike digital filter ringing it is well in the audible band. It is one of the DSP effects of DAW plug-ins to give a tape-like sound.

     

    Then there is the IMD effects of higher frequencies interacting with the high frequency tape bias.

     

     

    The reel project – part 3

     

    You can read about a few such effects here.

     

    Print thru anyone? That is another real effect of reel tape which will get put onto the LP. Sound like superb temporal performance?

     

    Hey liking tape is all well and good. Ditto for LP. We need to get past this idea these prior mediums are superior to digital as they aren't.

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    "The first cartridge with integrated RF shielding, the Clearaudio Goldfinger Statement MC phono cartridge is designed to play on the world's finest analog systems and reproduce the music on the greatest recordings ever made with unstinting clarity, liveliness, dynamics, realism, and responsiveness. Featuring 12 perfectly matched and symmetrical magnets – an unprecedented achievement – surrounding its coils. The Product of the Year Award-winning Goldfinger Statement allows systems to reach the long-unattainable dynamic range of 100dB." https://www.musicdirect.com/store/clearaudio-goldfinger-statement-mc-cartridge

     

     

    Said like someone who doesn't understand how to read an FFT. The graphs were done with a 256 k FFT. Such a graph would have the noise floor near -150 db for a noise level across the full band of -100db. The graph shown for the cartridge in fact would be a dynamic range as such is usually determined of maybe 60 db give or take a bit.

     

    Here is a better view of that graph down this page.

     

    graph3.gif

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    Seems the aether ate one of my posts.

     

    About the cartridge with 100 db dynamic range. Well no.

     

    The graph is using a 256k FFT. At -100 db noise floor across the 20 khz bandwidth you would see a line of noise near -150 db on such an FFT. So I would guess maybe that cartridge has maybe 60-65 db dynamic range. It really would be nice to quit trying to mislead analog fans about "blowing CD out of the water". Just say you prefer the sound of LP.

     

    A better picture of the graph on this page.

     

    http://www.6moons.com/audioreviews/clearaudio3/graph3.gif

     

    6moons audio reviews: Clearaudio Goldfinger

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    By the way: None of what's been said should obscure the fact that while CD is capable of greater dynamic range than vinyl, CDs are all too often actually produced with a squashed dynamic range. No hocus pocus is necessary to explain this, only production decisions.

     

    Amen.

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    By the way: None of what's been said should obscure the fact that while CD is capable of greater dynamic range than vinyl, CDs are all too often actually produced with a squashed dynamic range. No hocus pocus is necessary to explain this, only production decisions.

    /QUOTE]

     

    Amen to that.

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    ... 24/96 is enough to capture everything that might possibly be significant.

     

    As long as there's nothing in the signal you're sampling above 48KHz. ...

     

    Did you mean "nothing significant"? I left "for audio" off the end of my sentence. :-)

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    Said like someone who doesn't understand how to read an FFT. The graphs were done with a 256 k FFT. Such a graph would have the noise floor near -150 db for a noise level across the full band of -100db. The graph shown for the cartridge in fact would be a dynamic range as such is usually determined of maybe 60 db give or take a bit.

     

    Here is a better view of that graph down this page.

     

    [ATTACH=CONFIG]33713[/ATTACH]

     

    Then there is the RIAA curve vs. wow, and surface noise to consider, though surface noise can be "listened through." (The RIAA curve refers to low frequencies being diminished and high frequencies being boosted when a record is cut, and the reverse happening in the phono preamp at playback, to allow maximum playing time for a 33 /13 LP and minimize groove damage. The low frequency boost on playback is approximately 10db at the frequency - 400 Hz - where wow's perceived pitch distortions are centered.)

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    ... Amen to that.

    You know something's wrong when a metal album ends up less compressed than a semi-acoustic soul/blues album. (Iron Maiden, "The Final Frontier" versus Tom Jones, "Praise And Blame". Both mastered in 2010 by the same engineer.)

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    [/font]

     

    Did you mean "nothing significant"? I left "for audio" off the end of my sentence. :-)

     

     

     

    Yep. Thinking of harmonics for trumpet, and some percussion. (Cymbals can have 40% of their energy above 20kHz.)

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    You know something's wrong when a metal album ends up less compressed than a semi-acoustic soul/blues album. (Iron Maiden, "The Final Frontier" versus Tom Jones, "Praise And Blame". Both mastered in 2010 by the same engineer.)

    You got it!

     

    The audiophile community IMO needs to be more focused on how the bits are MADE than squabbling how they get distributed.

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    Well that is the other part. I have taken some of the downloads offered, steep filtered everything below 20 khz, then slowed it down to 25% normal speed so I could hear ultrasonics. There is very little up there. It is mostly surface noise and widely spaced bits of cymbals at low levels.

     

    It has been mentioned there is a slow roll off past 50 khz. Well not usually. 99% of LPs prior to digital were sourced from analog tape. Such tape isn't flat to 50 khz. Such tape also has scrape and flutter to effect timing. It certainly doesn't have infinite temporal resolution. It some situations the altered phase of high frequencies as the tape heads roll off could interact with drum-shots to cause something like pre-ringing at a lower frequency. Only unlike digital filter ringing it is well in the audible band. It is one of the DSP effects of DAW plug-ins to give a tape-like sound.

     

    Then there is the IMD effects of higher frequencies interacting with the high frequency tape bias.

     

     

    The reel project – part 3

     

    You can read about a few such effects here.

     

    Print thru anyone? That is another real effect of reel tape which will get put onto the LP. Sound like superb temporal performance?

     

    Hey liking tape is all well and good. Ditto for LP. We need to get past this idea these prior mediums are superior to digital as they aren't.

    So is 16/44 enough to capture all the music in your opinion? And if so why do the majority of CD's of analog recordings sound crap compared to the analog version? Cheers!

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    So is 16/44 enough to capture all the music in your opinion? And if so why do the majority of CD's of analog recordings sound crap compared to the analog version? Cheers!

     

    16/44 or very little more than that can capture all of the music imo.

     

    As for analog recordings vs digital ones there are multitude of factors. One inescapable factor is that LP even at it finest is not of full fidelity and has coloration. Maybe you like that coloration, maybe most do. That is okay.

     

    So many factors. Are the masters the same? Did they try and squeeze more out of the CD because they could? Do the majority of CDs sound like crap vs LP (not in my experience) as some do and some don't? If you love LP have you tailored your gear for best sound on LP which works against CD sounding its best?

     

    My basic thought is your premise of the question isn't true a majority of the time. It isn't generally true that CDs of analog recordings sound like crap. For one thing I have digitally recorded both reel tape and LP with the result sounding like reel tape or LP. So the digital CD isn't a bottleneck.

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    So is 16/44 enough to capture all the music in your opinion? And if so why do the majority of CD's of analog recordings sound crap compared to the analog version? Cheers!

     

    All the music that matters, yes. And they don't have to "sound crap", but:

     

    - I spent a lot of time and money back in the 80s and 90s duplicating my LP collection in CDs.

    - The majority of the CDs sounded similar to the LPs. Actually, they did sound different, but it was always the same difference, so I could blame the differences on my gear rather than something fundamentally wrong. I found I preferred the CD sound, and over time stopped listening to the LPs.

    - I have since repurchased some of those CDs as remasters or re-releases. The majority of them do "sound crap" compared to the early versions. Even where they have been redigitised from the original tapes, in many cases the age of the tape is showing and/or the result has been compressed to modern levels.

    - It is my belief that, in the majority of cases, a well cared for LP on a good turntable can be digitised to 24/96 then downsampled to 16/44.1 and the result will be so close to the LP as to not matter. If you're really fussy, leave it at 24/96.

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    All the music that matters, yes. And they don't have to "sound crap", but:

     

    - I spent a lot of time and money back in the 80s and 90s duplicating my LP collection in CDs.

    - The majority of the CDs sounded similar to the LPs. Actually, they did sound different, but it was always the same difference, so I could blame the differences on my gear rather than something fundamentally wrong. I found I preferred the CD sound, and over time stopped listening to the LPs.

    - I have since repurchased some of those CDs as remasters or re-releases. The majority of them do "sound crap" compared to the early versions. Even where they have been redigitised from the original tapes, in many cases the age of the tape is showing and/or the result has been compressed to modern levels.

    - It is my belief that, in the majority of cases, a well cared for LP on a good turntable can be digitised to 24/96 then downsampled to 16/44.1 and the result will be so close to the LP as to not matter. If you're really fussy, leave it at 24/96.

     

    Agree 16/44 is enough, The Trinity Sessions is one example of a great 16/44 recording.Maybe the A to D converters/process is worse now than in early years?

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