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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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    Wow.

     

    Are you expecting me to recite all the maths necessary to fully understand the sampling theorem here? That would be a few hundred pages, and I doubt I'd do as a good a job of it as the numerous people who have written books on the subject. If you're interested, get one and read it.

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    Half-truths and misrepresentations. This article is written by someone who does not understand signal theory, how digital works, and how analogue works, or someone with an agenda. A shame. It is perfectly fine to like the sound of vinyl. LP recording and playback is a triumph of electro-mechanical engineering (even though not without flaws). And that is all justification the format needs. No-one needs disinformation and lies.

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    Are you suggesting there is anything but complete consensus among scientists and mathematicians regarding the validity of the sampling theorem?

     

     

    You couldn't have missed the mark by more if you were trying (or perhaps you were trying).

     

     

     

    Nope, you were asked, after saying quite cryptically that the writer was mistaken due to the Sampling Theorem, to explain how the Sampling Theorem made what he said about timing accuracy of analog signals reconstructed from digital samples incorrect.

     

     

     

    You chose to misinterpret this as Chris questioning the Sampling Theorem itself. I tried to give you an example of how unhelpful your "You just have to trust the people who know" response was, and you chose to misinterpret that.

     

     

     

    This strikes me as a lot of work to avoid an explanation you've now mostly given in real-world terms with the graphs above, so I'm not sure exactly what all of it got you.

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    Nope, you were asked, after saying quite cryptically that the writer was mistaken due to the Sampling Theorem, to explain how the Sampling Theorem made what he said about timing accuracy of analog signals reconstructed from digital samples incorrect.

     

    I referred to what esldude said: http://www.computeraudiophile.com/f3-article-comments/article-digital-vinyl-temporal-domain-31816/#post641392

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    And you and he were quite correct. Chris then asked if you could put the reason for your conclusion on a level that would help others understand:

     

     

     

    99.999% would really like help understanding how and why. It's over our heads.

     

     

     

    All you had to do then was say, "I can't or won't boil down the way the Sampling Theorem works here, but look at these two graphs: One signal can be moved by a matter of nanoseconds in relation to the other, when both are reconstructed from a 44.1KHz-sampled signal (22 microsecond time interval). So nanosecond, or in fact picosecond, time accuracy is available from real world 44.1KHz digital sampling." After all, you wound up doing that anyway. But you made a great show of *not* explaining before you eventually did. Puzzling.

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    And you and he were quite correct. Chris then asked if you could put the reason for your conclusion on a level that would help others understand:

     

     

     

    All you had to do then was say, "I can't or won't boil down the way the Sampling Theorem works here, but look at these two graphs: One signal can be moved by a matter of nanoseconds in relation to the other, when both are reconstructed from a 44.1KHz-sampled signal (22 microsecond time interval). So nanosecond, or in fact picosecond, time accuracy is available from real world 44.1KHz digital sampling." After all, you wound up doing that anyway. But you made a great show of *not* explaining before you eventually did. Puzzling.

     

    I wasn't at my desk when I first replied. Also, theoretical explanations and experimental results are not mutually exclusive.

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    Wow! This is denigrating into a "Cables-and-Interconnects-make-a-difference" conversation. I believe analog does sound better. I have a degree in Mathematics, studied the physics of sound (In 1973 I started by using the Helmholtz book "On the sensations of tone as a physiological basis for the theory of music" !?!), am a musician, recorded music in the two inch reel-to-reel days and......BIG DEAL.....I've heard all of my favorite music on vinyl and digital and vinyl wins. That being said I no longer have the physical space to store all the vinyl, all the equipment involved I would personally want, fussing over it.....I will stay with full on digital, hope for the best in future DAC's, enjoy many vinyl rips I own. But I still know that playing back The Allman Brothers "Hot Lanta" at full tilt never sounds as magnificent as it does on vinyl playback.

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    Are you suggesting there is anything but complete consensus among scientists and mathematicians regarding the validity of the sampling theorem?

     

     

     

    Yes, it's possible to calculate the effect of limited precision and whatever other imperfections there are in a practical system. For 16/44.1 the time accuracy is on the order of picoseconds, I don't remember the exact figure.

     

    Perhaps a demonstration with a DAC and scope will convince. This is the left/right zero-crossings of an iFi Nano DAC playing the same sine wave at 44.1 kHz on both channels:

     

    [ATTACH=CONFIG]33665[/ATTACH]

     

    There's an inherent skew of about 27 ns, so this is just for reference.

     

    Now we delay the right channel slightly:

     

    [ATTACH=CONFIG]33666[/ATTACH]

     

    The inter-channel difference has increased by about 8 ns which is quite substantially less than the 22 μs sample interval.

    Your example is for a constant running sine wave. It does not address the issue of resolving where in a sample a transient starts.

    Are you suggesting there is anything but complete consensus among scientists and mathematicians regarding the validity of the sampling theorem?

     

     

     

    Yes, it's possible to calculate the effect of limited precision and whatever other imperfections there are in a practical system. For 16/44.1 the time accuracy is on the order of picoseconds, I don't remember the exact figure.

     

    Perhaps a demonstration with a DAC and scope will convince. This is the left/right zero-crossings of an iFi Nano DAC playing the same sine wave at 44.1 kHz on both channels:

     

    [ATTACH=CONFIG]33665[/ATTACH]

     

    There's an inherent skew of about 27 ns, so this is just for reference.

     

    Now we delay the right channel slightly:

     

    [ATTACH=CONFIG]33666[/ATTACH]

     

    The inter-channel difference has increased by about 8 ns which is quite substantially less than the 22 μs sample interval.

     

     

    Sent from my Nexus 7 using Computer Audiophile mobile app

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    Your example is for a constant running sine wave. It does not address the issue of resolving where in a sample a transient starts.

     

     

     

    Sent from my Nexus 7 using Computer Audiophile mobile app

    Everything is a sum of sines.

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    Everything is a sum of sines.

    Yes, but can you resolve where the transient started within one 44Khz sample to the accuracy you were able to shift the constant sine wave in your example?

     

    Sent from my Nexus 7 using Computer Audiophile mobile app

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    Yes, but can you resolve where the transient started within one 44Khz sample to the accuracy you were able to shift the constant sine wave in your example?

     

    Sent from my Nexus 7 using Computer Audiophile mobile app

     

     

    As long as the transient doesn't involve frequencies equal to or higher than 22.05KHz (for a 44.1KHz sample rate), yes.

     

     

    Edit: A little more detail - As I mentioned in my previous attempt at an explanation, once you have adequate sampling (the "third sample point" in my prior explanation), you have mathematically established exactly where the signal had to be at every point along its length.

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    Wow! This is denigrating into a "Cables-and-Interconnects-make-a-difference" conversation. I believe analog does sound better. I have a degree in Mathematics, studied the physics of sound (In 1973 I started by using the Helmholtz book "On the sensations of tone as a physiological basis for the theory of music" !?!), am a musician, recorded music in the two inch reel-to-reel days and......BIG DEAL.....I've heard all of my favorite music on vinyl and digital and vinyl wins. That being said I no longer have the physical space to store all the vinyl, all the equipment involved I would personally want, fussing over it.....I will stay with full on digital, hope for the best in future DAC's, enjoy many vinyl rips I own. But I still know that playing back The Allman Brothers "Hot Lanta" at full tilt never sounds as magnificent as it does on vinyl playback.

     

    Mastering is certainly involved, and perhaps differences in analog and digital systems in terms of design and parts quality - I don't know what your analog and digital systems look like.

     

     

     

    As an example of mastering differences in my experience, the Who's Tommy and Steely Dan's Gaucho DVD-A versions weren't a patch on my LPs, but the SHM-SACD versions of each sound great.

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    I wasn't at my desk when I first replied. Also, theoretical explanations and experimental results are not mutually exclusive.

     

    Thanks, my apologies for assuming it was simply stubbornness.

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    Jud, Are you sure? I am not asking if it can resolve the frequencies but where within the width of the sampling pulse the transient starts. MQA dont seem to think it can be done, that´'s why they have come up with different sampling and filtering methods

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    Jud, Are you sure? I am not asking if it can resolve the frequencies but where within the width of the sampling pulse the transient starts. MQA dont seem to think it can be done, that´'s why they have come up with different sampling and filtering methods

     

     

    MQA has not found an exception to the Sampling Theorem. :)

     

     

     

    What they are talking about is something different.

     

     

    Mathematically, the more precise your filtering is in terms of time, the less precise it is in terms of frequency, which leads to the real world job of creating a filter that will find the appropriate balance between "time domain" distortions (ringing) and "frequency domain" distortions (aliasing and intermodulation distortion).

     

     

    MQA has made a great show of moving its balance away from the place where the vast majority of people working in digital audio have chosen to put their emphasis. They have to some extent disdained trying to have correct response in the frequency domain. Whether this is enough to provide a better time domain response as MQA claims, we don't know, since we haven't seen any data/measurements thus far. What measurements we have seen in the frequency domain would indicate it is a good idea to be skeptical of MQA's claims.

     

     

    None of this prevents you from listening to MQA and enjoying it if you like. But it doesn't appear to be any sort of game changer in technological terms.

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    As long as the transient doesn't involve frequencies equal to or higher than 22.05KHz (for a 44.1KHz sample rate), yes.

     

     

    Edit: A little more detail - As I mentioned in my previous attempt at an explanation, once you have adequate sampling (the "third sample point" in my prior explanation), you have mathematically established exactly where the signal had to be at every point along its length.

    I see what you are saying.

     

    Sent from my Nexus 7 using Computer Audiophile mobile app

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    MQA has not found an exception to the Sampling Theorem. :)

     

     

     

    What they are talking about is something different.

     

     

    Mathematically, the more precise your filtering is in terms of time, the less precise it is in terms of frequency, which leads to the real world job of creating a filter that will find the appropriate balance between "time domain" distortions (ringing) and "frequency domain" distortions (aliasing and intermodulation distortion).

     

     

    MQA has made a great show of moving its balance away from the place where the vast majority of people working in digital audio have chosen to put their emphasis. They have to some extent disdained trying to have correct response in the frequency domain. Whether this is enough to provide a better time domain response as MQA claims, we don't know, since we haven't seen any data/measurements thus far. What measurements we have seen in the frequency domain would indicate it is a good idea to be skeptical of MQA's claims.

     

     

    None of this prevents you from listening to MQA and enjoying it if you like. But it doesn't appear to be any sort of game changer in technological terms.

    Have you read their paper thar includes the triangular sampling? That is where they claim to be get better temporal accuracy in the ADC. The filtering is another part of their process.

    I haven't heard any MQA'd music yet. I have a feeling I am going to be underwhelmed!

     

    Sent from my Nexus 7 using Computer Audiophile mobile app

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    Have you read their paper thar includes the triangular sampling? That is where they claim to be get better temporal accuracy in the ADC. The filtering is another part of their process.

    I haven't heard any MQA'd music yet. I have a feeling I am going to be underwhelmed!

     

    Sent from my Nexus 7 using Computer Audiophile mobile app

     

     

    Here's Miska talking about MQA at the ADC stage (on Roon's forum):

     

     

     

    No, they just run decimation (sample rate reduction) using their filter (kernel) at mastering stage to reduce the source to 88.2/96 kHz rate for encoding. Then they use shaped high level dither to hide distortion from the leaky filter and trying to maintain dynamic range at lower frequencies despite only few bits. They use the entire top octave for filter roll-off (and aliasing), because they think that frequencies above 20 kHz are not useful so those can be sacrificed to keep ringing of the filter to minimum.

     

     

     

    See where he talks about the filter kernel? MQA uses a "triangular" filter kernel. This is part of "decimating" (reducing) the sample rate of the source. It is not magic and it does not provide starting points of between-sample signals that others cannot. And when Miska talks about "leaky" filtering he means the poor frequency response performance of the MQA filters on the DAC side.

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    Hi esldude - Igor has used and researched many products for ripping vinyl. He has incredible passion for his project. He settled on the Pure Vinyl product as being the best, but he has absolutely zero relation to the company.

     

    I'm not sure if that's what you were getting at, but I wanted to clarify for everyone.

     

    Thanks for answering that question. I too have used Channel D software for needledrops. It is excellent software.

     

    Frankly Chris I was dismayed to see this article on your site. If this had been a principle at Channel D it would have put a big dent in my respect for their company. Most of the article is untrue misinformation. Old myths that are factually not so. One need not perpetuate bad info to have a passion for vinyl.

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    From a non-engineering standpoint, there just has to be something different about sound that is born from mechanical vibration (analog) versus sound that is born from electricity (digital)- with "born" meaning how it starts in your living room. Yes, the turntable is powered- but only to spin the record. Somehow, translating vibration into electricity via a magnet just sounds different- and often better- than recreating an electrical waveform from a digital signal. I (remember) hearing a more realistic increase in loudness and image size when I had vinyl- does that reflect the way a magnet responds to larger vibrations? Have always hoped someone would delve more into these questions than samples rates and bits.

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    Thanks for answering that question. I too have used Channel D software for needledrops. It is excellent software.

     

    Frankly Chris I was dismayed to see this article on your site. If this had been a principle at Channel D it would have put a big dent in my respect for their company. Most of the article is untrue misinformation. Old myths that are factually not so. One need not perpetuate bad info to have a passion for vinyl.

     

    This is beyond my technical skill level. I honestly don't know what's right, wrong, or partially correct etc...

     

    I saw the AES article and don't know if it discusses the same thing as discussed here.

     

    Fortunately, we allow everyone to comment here and hopefully we'll get to the right place through "peer" review.

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    Hi esldude. The article does not describe the spacing of samples, but the temporal resolution. How close it can be placed in time one sound from another sound that the human ear could be the difference.

    And if you want to be precise, 192 kHz is 5.2 microseconds,

    96 kHz is 10.4 microseconds

    48 kHz is 20.8 microseconds.

    (Note, time resolution doesn't depend on the word length, e.g., 16/44.1 and 24/44.1 have exactly the same temporal resolution - 22.7 microseconds.)

     

    Best

     

    Pure Vinyl Club

     

    Listen to short demos of the LP Records

    and share your experience and observations.

     

     

    As already said, you are incorrect about this. Trying to be respectful and courteous to you, but your information is wrong.

     

    The correct theoretical number of time resolution with 16/44 is about 56 picoseconds. With dither it is even less.

     

    Real world, jitter in clocks can be above 56 picoseconds, but very nearly all modern gear has jitter at or below 300 picoseconds which is still orders of magnitude lower than the 10 microseconds of human hearing ability. It is not difficult for instance to show time delay between one meter and two meters of cable with 16/44khz digital. A meter of time for the signal is around 3 nanoseconds or so.

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    Here's Miska talking about MQA at the ADC stage (on Roon's forum):

     

     

     

     

     

     

     

    See where he talks about the filter kernel? MQA uses a "triangular" filter kernel. This is part of "decimating" (reducing) the sample rate of the source. It is not magic and it does not provide starting points of between-sample signals that others cannot. And when Miska talks about "leaky" filtering he means the poor frequency response performance of the MQA filters on the DAC side.

     

    If you read Bob Stuarts Audio Engineering Society Convention Paper 9178 ( sorry no link ) the "triangular" filter ( actually a b spline kernel, whatever that is), they do actually claim it can resolve starting points between samples because the samples overlap.

     

    There is also some mention of this in one of their patents, unfortunately the diagrams are missing:-

     

    https://www.google.com/patents/WO2014108677A1

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    If you read Bob Stuarts Audio Engineering Society Convention Paper 9178 ( sorry no link ) the "triangular" filter ( actually a b spline kernel, whatever that is), they do actually claim it can resolve starting points between samples because the samples overlap.

     

    There is also some mention of this in one of their patents, unfortunately the diagrams are missing:-

     

    https://www.google.com/patents/WO2014108677A1

     

     

    From the patent:

     

     

     

    Preferably, the downsampler comprises a decimation filter specified at the first sample rate, wherein the asymmetric component of response of the decimation filter is characterised by an attenuation of at least 32dB at frequencies that would alias to the range 0-7 kHz on decimation. The range 0-7kHz is the range where the ear is most sensitive.

     

     

    Holy s**t. Perhaps Miska and mansr have been too polite.

     

     

     

    That being said: I have heard MQA sources where I also have the hi res files. I thought it sounded pretty good, not as good as the hi res. (mansr has mentioned that the bad frequency domain performance might not be audible.) And as I think is nearly always true, almost regardless of format, where Tidal has provided MQA masters that are superior to the masters for the RedBook-resolution files they had, I think the MQA version sounds better simply because of the mastering.

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    Wow.

    Well, from someone who has studied the math and knows as much about the subject as anyone alive (James Johnston, aka "JJ"):

     

    The time resolution of a 16 bit, 44.1khz PCM channel is not limited to the 22.7µs time difference between samples. The actual minimum time resolution is equivalent to 1/(2pi * quantization levels * sample rate). For 16/44.1, that is 1/(2pi * 65536 * 44100), which is about 55 picoseconds. To put that in perspective, light travels less than an inch in that time.

     

    Shannon and Nyquist stated that as long as you keep all components of the input signal below half the sampling frequency, you can reconstruct the original signal perfectly - not just in terms of amplitude, but in terms of temporal relationships too. They only addressed sampling, and assumed infinite resolution in amplitude. With a digital signal the precision is limited by the number of amplitude steps, leading to the above formula.

     

    Edit; This is clearly illustrated in Monty's digital primer video, from about the 20 minute mark, and people still fail to understand it.

     

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