Jump to content
  • Pure Vinyl Club
    Pure Vinyl Club

    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




    User Feedback

    Recommended Comments



    I'm not disputing that a digital copy can and should sound as good as the original analog recording but in reality if you listen to the 16/44 content of artist's like Frank Sinatra on Tidal, it does sound like utter crap compared to the analog original.

     

    And that only proves they did a bad job remastering for digital. If I can make a recording at home on high quality audiophile equipment - nothing unusual or special - that sounds exactly - and I mean exactly - like the source, then certainly a record company can do the same.

    Share this comment


    Link to comment
    Share on other sites

    And that only proves they did a bad job remastering for digital. If I can make a recording at home on high quality audiophile equipment - nothing unusual or special - that sounds exactly - and I mean exactly - like the source, then certainly a record company can do the same.

    Yes, you would think so wouldn't you...

    Share this comment


    Link to comment
    Share on other sites

    MOFI's SACDs of Frankie sound very good. Maybe it's a possible to claim a rebate if you swear you will listen only to the gorgeous perfect CD layer and not to the crappy bad tech misinformed misconceived SACD layer

    And that only proves they did a bad job remastering for digital. If I can make a recording at home on high quality audiophile equipment - nothing unusual or special - that sounds exactly - and I mean exactly - like the source, then certainly a record company can do the same.

    Share this comment


    Link to comment
    Share on other sites

    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.

     

     

    Resurecting an old comment on this thread, I am not a scientist or mathematician but I enjoy reading discussions such as this. I have watched Monty's video a few times. Do you have any other sources that I can do further reading as to why the time resolution of 44.1 is not limited to the time difference between samples? The formula given for the actual minimum time resolution is proving hard for me to grasp! Thanks! :)

    Share this comment


    Link to comment
    Share on other sites

    MOFI's SACDs of Frankie sound very good. Maybe it's a possible to claim a rebate if you swear you will listen only to the gorgeous perfect CD layer and not to the crappy bad tech misinformed misconceived SACD layer

     

    I thought you were talking about the group Frankie Goes to Hollywood, and was quite interested for a second. :)

    Share this comment


    Link to comment
    Share on other sites

    Resurecting an old comment on this thread, I am not a scientist or mathematician but I enjoy reading discussions such as this. I have watched Monty's video a few times. Do you have any other sources that I can do further reading as to why the time resolution of 44.1 is not limited to the time difference between samples? The formula given for the actual minimum time resolution is proving hard for me to grasp! Thanks! :)

     

    My math is rudimentary, so I can't help much. As I understand the formula, any signal that fits in less than half the sample rate is represented by more than two samples. To capture a difference in timing of that signal, it has to change at least one of the bits of at least one of the samples.

    This paper may help:

    https://web.archive.org/web/20060614125302/http://www.lavryengineering.com/documents/Sampling_Theory.pdf

    Share this comment


    Link to comment
    Share on other sites

    My math is rudimentary, so I can't help much. As I understand the formula, any signal that fits in less than half the sample rate is represented by more than two samples. To capture a difference in timing of that signal, it has to change at least one of the bits of at least one of the samples.

    This paper may help:

    https://web.archive.org/web/20060614125302/http://www.lavryengineering.com/documents/Sampling_Theory.pdf

     

    Thanks! Probably over my head but I'll give it a shot!

    Share this comment


    Link to comment
    Share on other sites

    Thanks! Probably over my head but I'll give it a shot!

    It's worth trying, it uses a minimum of math.

    Share this comment


    Link to comment
    Share on other sites

    On 15/3/2017 at 0:51 PM, Jud said:

     

    So many vinyl lovers' preferences may come to some extent from the fact that at least part of the time they're listening to better mastering. I completely agree with the proposition that mastering trumps resolution or even vinyl vs. digital.

     

     

    And then folks get sucked in by plausible-sounding hypotheses that are incorrect, but they don't examine too closely since these hypotheses are in line with their preconceptions. I'm still in agreement, in fact perhaps more so than you: that Lipshitz and Vanderkooy article is steadfastly cited by lots of people whose DACs are doing sigma-delta modulation internally (yourself as well, perhaps?), and they don't seem to mind.

    First: Like several others, I was very happy to see the comment that the author withdrew the article, and that his stated retraction was written at the very top. Good job! Most vinyl lovers, or "OCD audiophiles" (pardon my French) never admit anything (*cough* Fremer). So again, my compliments to you :-).

    Then on to Jud's comments:

    I completely agree! One of the most famous examples is Red Hot Chili Peppers' "Stadium Arcadium" which sounds horrible on CD, which is mastered by loudness lover Vlado Meller, while it sounds lovely on vinyl, which is mastered by Steve Hoffman (there's a comparison video by Ian Shepherd on Youtube if you haven't seen it). Although people have preferences then I believe there's usually consensus. I'm convinced most would prefer the vinyl edition of the RHCP album. I'm also convinced that most would choose the vinyl edition of Mike Stern's "Upside downside" from 1986 (the CD is "typical" 80s sounding: cold, thin and shrill).

    Unfortunately, I've also come to realize lately that too much music, not only today (but especially today), is poorly produced or mastered. Up until recently I just thought people were exagerrating. Anyway, something is never going to sound good on either media if the master is poor, and yes, I do think the mastering for CDs in the 80s was less than stellar.

    What I've found is that some extremely dynamically compressed, shrill and "hard" sounding CDs MAY sound subjectively "better" on vinyl precisely because they have lower fidelity. An example is Mastodon's "Once more 'round the sun", which is mastered by Ted Jensen and measures DR5 with the DR meter. The vinyl edition is actually tolerable to listen to, while the CD sounds very "hard". I assume the vinyl edition is cut from the same master, and then the vinyl material changes the sound a bit and so does my playback equipment, and perhaps the cutting engineer also used EQ to reduce the heavy amount of treble present. Other examples of this, albeit with slightly less compressed albums, could be the first 3-4 albums by Badly Drawn Boy.

    I've found that this kind of effect (lower fidelity = more pleasant) usually starts to come into play when the CD measures around DR6 or lower.

    Share this comment


    Link to comment
    Share on other sites

    3 minutes ago, board said:

    First: Like several others, I was very happy to see the comment that the author withdrew the article, and that his stated retraction was written at the very top. Good job! Most vinyl lovers, or "OCD audiophiles" (pardon my French) never admit anything (*cough* Fremer). So again, my compliments to you :-).

     

    The retraction was from Chris, not the author.

    Share this comment


    Link to comment
    Share on other sites

    11 minutes ago, board said:

    First: Like several others, I was very happy to see the comment that the author withdrew the article, and that his stated retraction was written at the very top. Good job! Most vinyl lovers, or "OCD audiophiles" (pardon my French) never admit anything (*cough* Fremer). So again, my compliments to you :-).

    Then on to Jud's comments:

    I completely agree! One of the most famous examples is Red Hot Chili Peppers' "Stadium Arcadium" which sounds horrible on CD, which is mastered by loudness lover Vlado Meller, while it sounds lovely on vinyl, which is mastered by Steve Hoffman (there's a comparison video by Ian Shepherd on Youtube if you haven't seen it). Although people have preferences then I believe there's usually consensus. I'm convinced most would prefer the vinyl edition of the RHCP album. I'm also convinced that most would choose the vinyl edition of Mike Stern's "Upside downside" from 1986 (the CD is "typical" 80s sounding: cold, thin and shrill).

    Unfortunately, I've also come to realize lately that too much music, not only today (but especially today), is poorly produced or mastered. Up until recently I just thought people were exagerrating. Anyway, something is never going to sound good on either media if the master is poor, and yes, I do think the mastering for CDs in the 80s was less than stellar.

    What I've found is that some extremely dynamically compressed, shrill and "hard" sounding CDs MAY sound subjectively "better" on vinyl precisely because they have lower fidelity. An example is Mastodon's "Once more 'round the sun", which is mastered by Ted Jensen and measures DR5 with the DR meter. The vinyl edition is actually tolerable to listen to, while the CD sounds very "hard". I assume the vinyl edition is cut from the same master, and then the vinyl material changes the sound a bit and so does my playback equipment, and perhaps the cutting engineer also used EQ to reduce the heavy amount of treble present. Other examples of this, albeit with slightly less compressed albums, could be the first 3-4 albums by Badly Drawn Boy.

    I've found that this kind of effect (lower fidelity = more pleasant) usually starts to come into play when the CD measures around DR6 or lower.

    I really wish Chris would allow someone to write about this stuff! Hint, hint, wink, wink! ;-)

    In all seriousness though, I think more and more audiophiles are recognizing the fact that the promises of high-res audio, and to a certain extent vinyl, have more to do with the mastering behind them than their sampling rate.

    I hope for everybody's sake that trend continues.

    Share this comment


    Link to comment
    Share on other sites

    2 minutes ago, AlexMetalFi said:

    I really wish Chris would allow someone to write about this stuff! Hint, hint, wink, wink! ;-)

    In all seriousness though, I think more and more audiophiles are recognizing the fact that the promises of high-res audio, and to a certain extent vinyl, have more to do with the mastering behind them than their sampling rate.

    I hope for everybody's sake that trend continues.

    Are you winking at me?

    Sorry, about my mistake about who retracted the article, but I was still happy to see that it was retracted :-).

    Share this comment


    Link to comment
    Share on other sites

    Just now, board said:

    Are you winking at me?

    Sorry, about my mistake about who retracted the article, but I was still happy to see that it was retracted :-).

    No, at Chris.

    Share this comment


    Link to comment
    Share on other sites

    3 minutes ago, AlexMetalFi said:

    I really wish Chris would allow someone to write about this stuff! Hint, hint, wink, wink! ;-)

    In all seriousness though, I think more and more audiophiles are recognizing the fact that the promises of high-res audio, and to a certain extent vinyl, have more to do with the mastering behind them than their sampling rate.

    I hope for everybody's sake that trend continues.

    Your email is still in my inbox. Been to busy with the site upgrade to respond. Please email your thoughts and ideas. I'm always open and searching for good writers. They are harder to find than most people think.

    Share this comment


    Link to comment
    Share on other sites

    On 3/16/2017 at 7:09 AM, firedog said:

     

     

    Me too. Very good digital recordings of needle drops are indistinguishable on playback from the original vinyl - played back on the same system. I've personally experienced this and demonstrated it with others.

     

    A properly done digital recording of vinyl has all those "vinyl like" qualities that vinyl lovers love. So the digital medium itself isn't the culprit or the limitation. I refer all of you to John Atkinson's review of the Ayre ADC, where he said he compared digital conversions of his own analog recordings to the original- in fact compared them "till his ears bled" - and said he couldn't tell them apart.

    Please keep in mind that those transfers were done at 24/192 and that as I recall, at that setting the Ayre does no filtering.

    It is also true that JA's conclusion is highly suspect. As he claims to be able to hear differences between DACs and has never found one to be audibly perfect, it is not possible for the digital playback to exactly match the analog source, even if the digital copy is perfect.

    Finally, I have LP transfers made with the Ayre QA-9, and while there is no doubt they are imbued with "LPness", they don't sound identical to direct playback from the analog setup used in the transfer. Many people prefer these  LP transfers to hi res transfers from master tapes because they retain a lot of the LP sound, but they are not identical. 

    Share this comment


    Link to comment
    Share on other sites

    6 minutes ago, The Computer Audiophile said:

    Your email is still in my inbox. Been to busy with the site upgrade to respond. Please email your thoughts and ideas. I'm always open and searching for good writers. They are harder to find than most people think.

    Yeah, I had no idea you were in the middle of this. My apologies Chris for insinuating you were ignoring my request.

    Chris, do me a favor and a at least respond so I have your direct address!

    Cheers!

    Share this comment


    Link to comment
    Share on other sites

    On 15/3/2017 at 0:34 PM, Fokus said:

     

    Some didn't. But the various Sony PCM16** and F1 convertors did have dither from day one.

     

    https://www.gearslutz.com/board/attachments/mastering-forum/25537d1161190991-old-sony-pcm-users-analogue-pcm-1983-.gif

    Odd! Especially since Stanley Lipshitz says that it wasn't introduced in A/D converters until the mid 80s in the following video (this is the video I talked about when I said that even he said that early converters weren't very good), but maybe he was unaware of those particular converters you mention, although I don't really understand how your picture shows that it uses dithering (is it the lack of harmonic distortion?).

     

    On a completely different note, then someone (I couldn't find the comment) mentioned something along the lines of "yeah, maybe digital is a superior technology, but CDs usually sound like shit, whereas the vinyl edition almost always sounds better, so maybe that's worth discussing instead of this stupid discussion about technology" (I'm paraphrasing a lot).

    It would be a lot easier to have a debate about that if we didn't have to post 13 pages of comments correcting the vinylphiles in their flawed assesments of technology.

    All that said, I think even the most hardcore objectivist, who would perhaps argue that he would choose the digital version in 99,9 % of the cases, simply because it has higher fidelity, would agree that we should strive for good sounding productions and masters.

    As much as I dislike Michael Framer, then I agree with him about certain things (this or that album sounds good), and I think both I and the most hardcore pro-digital objectivist would agree with Framer that the music world would be a better place if most of the music produced nowadays was better produced and mastered.

    So, what I'm trying to say is that, yes, we need a debate, preferably one where the head of Sony, Universal, mastering engineers, musicians, etc. would participate, so we can tell them that we want better sounding music - not just louder music. But I also think the hardcore subjectivists need to learn how to read and listen, rather than repeat the same old tired claims that have been disproved numerous times in the past. On the same note, some of the hardcore objectivists also need to "loosen up" a bit, instead of always saying "but it has higher fidelity, therefore I *have* to prefer this one to the one that I actually enjoy but which has lower fidelity" (CD/vinyl or speakers/amps, etc.).

    On another note: I read a mastering engineer say that when bands ask for louder, louder, LOUDER mastering he makes one like he wants it and then makes a compressed one and adjust the volume levels to match, and the bands always choose the uncompressed one :-). Based on what I read, it's usually the artists, not the mastering engineers, who want louder masterings.

    Share this comment


    Link to comment
    Share on other sites

    Yes, the lack of (anharmonic) distortion components, and the presence of a nice and flat noise floor.

    I once had schematics for an early Sony convertor, forgot which one, and it clearly had a zener-based noise source at its input. The concept of dither originated in the 40s. Old-school engineering textbooks sometimes contain circuits for dither sources.

     

    I did not watch the SL movie. Perhaps he is meaning something else, such as the use of ADCs of more than 16 bit (Decca were early with this, and when did DG come with 4D?), followed with digital-domain dither and reduction to 16 bit? Or perhaps he was thinking of early digital editors, which indeed lacked dither during fades.

     

    Share this comment


    Link to comment
    Share on other sites

    19 hours ago, Fokus said:

    Yes, the lack of (anharmonic) distortion components, and the presence of a nice and flat noise floor.

    I once had schematics for an early Sony convertor, forgot which one, and it clearly had a zener-based noise source at its input. The concept of dither originated in the 40s. Old-school engineering textbooks sometimes contain circuits for dither sources.

     

    I did not watch the SL movie. Perhaps he is meaning something else, such as the use of ADCs of more than 16 bit (Decca were early with this, and when did DG come with 4D?), followed with digital-domain dither and reduction to 16 bit? Or perhaps he was thinking of early digital editors, which indeed lacked dither during fades.

     

    I may have this wrong, and couldn't find anything definitive about those ADs.  I do believe the first ones had no dither.  I am thinking the Sony 701s which were pretty early were the first that used zener diodes as noise sources to provide random dither. 

     

    I have seen reference that PCM 1610 and 1630 as well as the PCM F1 Sony units used one converter chip and switched between right and left channels thereby offsetting the sampling by one half of a sample period between channels.  Pretty much like the first Sony CD players used one chip and the two channels were offset by one half sample period. 

    Share this comment


    Link to comment
    Share on other sites

    3 hours ago, esldude said:

    I may have this wrong, and couldn't find anything definitive about those ADs.  I do believe the first ones had no dither.

     

    For the sake of archaelogy then. Here is a page of the PCM1630 maintenance manual. It clearly discusses the dither generator:

     

    https://www.manualslib.com/manual/452275/Sony-Pcm-1630.html?page=68

     

    Other thing I found browsing sevel forums:

    -the dither level was insufficient for 14 bit use, so it only worked running at 16 bit

    -the linearity of the early convertors was suspect, so that dither, in the end, may well have been a bit ineffective

    -the 1610/1630 editor fader was undithered

     

    BTW, the time-shared nature of early Sony convertors is a well known fact.

    Share this comment


    Link to comment
    Share on other sites

    I finally saw the Lipschitz film. He never says that dither was not used on recorders before the mid 80s.

     

    What he says is that some older digital recordings suffered from audible quantisation distortion. He attributes this among others to problematic ADCs and the use of undithered fades during editing.

     

    The part about the mid 80s, as I understand it, pertains to the development of dither theory, and the derivation of optimal dither. Before this, dither could be just about any noise. After Lipschitz and the others who looked into this more thoroughly it was cast in a mathematical frame and the correct amount and probability distributions were derived.

    Share this comment


    Link to comment
    Share on other sites

    1 hour ago, Fokus said:

     

    For the sake of archaelogy then. Here is a page of the PCM1630 maintenance manual. It clearly discusses the dither generator:

     

    https://www.manualslib.com/manual/452275/Sony-Pcm-1630.html?page=68

     

    Other thing I found browsing sevel forums:

    -the dither level was insufficient for 14 bit use, so it only worked running at 16 bit

    -the linearity of the early convertors was suspect, so that dither, in the end, may well have been a bit ineffective

    -the 1610/1630 editor fader was undithered

     

    BTW, the time-shared nature of early Sony convertors is a well known fact.

    Good.  Thanks for providing the facts of the matter. 

     

    So there was a switch on an internal board to turn dither on and it was set at a level of -82 dbm.  Which in this context I suppose is equivalent to -82 dbu. 

    Share this comment


    Link to comment
    Share on other sites

    Okay, maybe I misunderstood what Stanley Lipshitz meant :-).

    Anyway, this entire discussions seems to more or less have come to an end, and I see that as a good thing :-).

    I will stop posting now, and I extend a thank you to everybody else who has contributed with helpful info and to Chris for posting his retraction of the article. I'm glad he's keeping the article up here including all the info to help others in the future :-).

    Share this comment


    Link to comment
    Share on other sites




    Create an account or sign in to comment

    You need to be a member in order to leave a comment

    Create an account

    Sign up for a new account in our community. It's easy!

    Register a new account

    Sign in

    Already have an account? Sign in here.

    Sign In Now




×
×
  • Create New...