Pure Vinyl Club

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About Pure Vinyl Club

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  1. Under the means for synchronization, everyone who works in professional studios usually means only clock. Not a DAC, not an ADC, but only a clock (internal in interfaces or external Master Clock). And nothing more. I hope that you will not wipe your feet about this great man (Andrea Bocelli): Andrea Bocelli Closer to Home | Antelope Audio My good friends have this in the studio: TAS 180: Esoteric G-ORb Rubidium Master Clock Generator | The Absolute Sound
  2. Yes, the Atomic reference oscillator is not a clock, but only a frequency generator (10MHz). It just gives more precision (accurancy) for the master clock (Isochrone Trinity).
  3. Antelope Audio Pure2: Burr-Brown (TI) A/D converter PCM4222 (High-Performance, Two-Channel, 24-Bit, from 8kHz to 216kHz Sampling Multi-Bit Delta-Sigma Analog-to-Digital Converter)
  4. It all depends on what kind of synchronization tools you use when registering an analog signal in digital domain and then reconstructing it (playing it back in an analog way). If you use a standard clock (quartz oscillator, possibly with the exception of the Grimm Audio CC1), then you can not completely solve the jitter problems even at a frequency of 44.1kHz. And a bigger problem will be when you use 96 or 192kHz. We use for recording in the studio Antelope Audio Audiophile 10M (10 MHz Rubidium Atomic Reference Generator, Frequency Accurancy better than 0.03 PPB (parts per BILLION)) and therefore we have no problems, even when we record on the frequency 192kHz.
  5. "In fact, this is a popular picture with a chart showing ANALOG (signal from the best microphone), as a reference standard. And the loss (or lack of loss, as they represent -))), in the case of DSD) in the temporal domain of the impulse response and energy when trying to register and then reconstruct this signal with the help of various digital standards (48, 96,192 and DSD)."
  6. Jud, I actually wrote with irony about this "popular" picture. Look again at the picture - there the peak of the DSD is even higher than the peak ANALOG ;-))
  7. Here you are slightly incorrectly pointing to vinyl, although in fact here it is said about ANALOG (that is what a best microphone can register). Vinyl will never be able to repeat this result without loss - too much interference will occur in the way of the LP creation process. And even more distortion will be when trying to play LP on your TT. In fact, this is a popular picture with a chart showing ANALOG (signal from the best microphone), as a reference standard. And the loss (or lack of loss, as they represent -))), in the case of DSD) in the temporal domain of the impulse response and energy when trying to register and then reconstruct this signal with the help of various digital standards (48, 96,192 and DSD).
  8. Here you are slightly incorrectly pointing to vinyl, although in fact here it is said about ANALOG (that is what a best microphone can register). Vinyl will never be able to repeat this result without loss - too much interference will occur in the way of the LP creation process. And even more distortion will be when trying to play LP on your TT. In fact, this is a popular picture with a chart showing ANALOG (signal from the best microphone), as a reference standard. And the loss (or lack of loss, as they represent -))), in the case of DSD) in the temporal domain of the impulse response and energy when trying to register and then reconstruct this signal with the help of various digital standards (48, 96,192 and DSD).
  9. It seems, this whole thing has also just degenerated into yet another "CD vs. vinyl" religious war. You always accuse me of opposing vinyl and digital. Show me a place in my two articles and comments, where I praise LP and disparage DIGITAL. All just the opposite. There is no TT in any of my sound systems. And I do not have any of my LPs, none at all! And never will be! TT, on which I make rips with LP is in the studio and it is integrated into professional equipment. And even in the studio, I never hear an "analog" sound from the LP, only DIGITAL. Because the phono preamp that I use for recording does not have a RIAA curve corrector. Only the "flat" XLR output and RIAA are superimposed in the digital domain. I always hear from the LP only a digital sound and I really like what I hear. This is the essence of this project - to make the most effective TT and with the help of high-class professional equipment to make the most effective LP rip. And that's all! Farewell to LP! Put it on the shelf in the old closet and forget how terrible a dream, all those inconveniences and problems with LP. Just this! It is the high quality of the DIGITAL equipment and software that has allowed to achieve very good results. Now you can write down everything that is on LP, with great reserve, absolutely all the information. And yes, I really like high fidelity when listening to music. In the studio, I listen to professional monitors. But at home I do not have any tube amplifiers. Only active pro monitors of different sizes (4", 5", 6", 8"). And my main setup is the Grimm Audio LS-1, which many consider the most high fidelity system in the world.
  10. This article is specifically written in a clear, easy form, which would be 90% of the readers on the CA, which are not interested in all this higher mathematics and who just want to know what they need for good listening to your favorite music. It was not meant for technological discussion. And only a small passage contains some technical information and is the cause of all this disagreement and criticism (due to misunderstanding or unwillingness to understand that in fact the article is written about something else). Maybe it's my fault, because I do not know good English and I use Google translator. Quotation from an interview with Art Dudley was used precisely because it is directly related to the material of this article. And it is to this passage. Or is Art Dudley not good enough to quote? To make it clear what exactly Art Dudley wanted to say, and in the context of which he said this, I provide a larger excerpt from this interview. Sorry, but the interview is only in Polish and I again have to use Google translator: "Wojciech Pacuła: What are then the greatest sins of modern audio? Art Dudley: In my opinion the two sins are the worst: 1) We wrangled down, with every movement, by those who point to this or that element of design or construction, and say, "It does not matter." I have one answer for them: "Bullshit, EVERYTHING is important!" We hear, however, again and again. Manufacturers claim that it does not matter what material performed amplifier housing. From engineers to remasteringiem who think it does not matter that the board LP was incised with digital tape (or using a digital delay). From the people, by which high resolution is not important, because the Nyquist frequency for CD 44.1 kHz is sufficient. The latter is particularly problematic when we realize that the Nyquist frequency does not apply to work and reconstruction filters decymacyjnych composite signal. Indeed, the two samples may be used to describe a single frequency, but do not provide a sufficient density of the samples to describe the speed with which the signal increases or decreases - and this is a key distinction between the music and mere sound." High Fidelity
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  12. 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
  13. Excuse me! It was another hacker attack from Russia to the United States!;-)))) I'm still poorly understood with the forum and checked the English version of his post through Google translate and mistakenly sent the Russian transcription.
  14. Hi elstude. I'm sorry for what I have for a long time did not respond to questions. I just have no experience of commenting on the CA and I wanted to wait and see what the discussion will go in the direction. The very title of this article lies the whole essence of what I wanted to discuss on the forum. Unlike "static" characteristics (spectral and dynamic domains), which are often discussed, I would like to emphasize the importance of what we hear in real time from our acoustic system - to "dynamic" characteristics (temporal and spatial domains ). And especially I like to emphasize the importance of temporal resolution for emotional perception when listening of music. And it has nothing to do with the digital aspect. I was not going to oppose an analog (vinyl, tape) and digital (CD, Hi-Res). Just what we hear from our speakers. This temporal resolution It may be worsened by your speaker or the speaker cables or phono preamp, or cartridge, ... Any component can "slow down" transients. Just what I wanted to say. Only once, I noticed that the CD format is not enough to reconstruct high time domain, also citing in this context (Art Dudley and Rob Robinson). But you are brought down to me hurricane criticism. All about what you say - it properly. It is generally accepted and does not require a large discus. But, once again I ask - how it relates to the topic of this article? Now a substantive response to your first question on this topic: It seems that temporal accuracy and temporal resolution are being confused here. Accuracy is a given, and for the sake of argument can be as high as one wishes - picoseconds, femtoseconds, etc.. When someone says a CD samples "accurately" to the picosecond level (that is, the sample clock is stable and has low jitter), that may be true, but that has nothing to do with the temporal resolution. The distinction between accuracy and precision / resolution is one of the prime topics emphasized early in college introductory physics and chemistry (or even grammar school advanced placement classes in the same subjects): the difference between accuracy and precision are critical in science and easily confused unless one has had experience or training in the subject. Take two balance scales that are, let's say, 100% accurate. One weighs to a precision (almost the same thing as resolution) of 1 gram, the other 10 grams. Put a 10 gram mass on both scales, they will both indicate 10 grams, with perfect accuracy. Place a 12 gram mass, the more *precise* of the two scales will indicate 12 grams. Remember both scales are 100 percent accurate - but you can't expect the second scale to resolve 12 grams, because it is below the measurement *precision* of the scale. The second scale will "accurately" report 10 grams for the 12 gram mass - within a resolution of +/- 5 grams. So, at best, you know that your measurement of the mass is 10 grams plus or minus 5 grams. A 16 gram mass would be measured as 16 grams on the first scale, 20 grams on the other, etc. The second scale is accurate, but only within the specified precision. On the other hand, a scale that is not "accurate" would have some discernible error in the measurement. For example, the scales report 11 and 20 grams in the first measurement - though the resolution is the same. For the optimum result, we need both good accuracy *and* precision. Now take the CD sampler accurate to 1 picosecond, given as an example. Yes, sample after sample are within one picosecond of the sample period (or even better is possible, with a super low jitter clock). But it simply cannot accurately resolve signal events with a temporal precision greater than the sample rate - approximately 23 microseconds (rounding to the nearest microsecond; 1/44100 is actually a transcendental number which can be represented to an arbitrary precision depending on the number of decimal places, which we don't care about here). So an event occurring at a given time can be resolved with certainty at best with a temporal resolution of roughly 23 microseconds. If you increase the sample rate to 192 kHz the temporal ACCURACY may be the same (at the same time coordinate, with the same picosecond or femtosecond accurate clock) but the temporal RESOLUTION is more than 4 times better - about 5 microseconds. (or to be precise, 5.2083333 microseconds with the 3 repeating on and on and on). Temporal resolution is defined by the sample rate. Taking an extreme example for illustration of a boundary condition, if the brain and auditory system only resolved sounds with TEMPORAL resolution at the second (1,000,000 microseconds) level, this would be problematic because a threat to survival such as a leopard pouncing from behind would not be detected as readily. Imagine an experiment where we played a sound of an animal springing for two human test subjects, from a loudspeaker positioned some moderate distance behind and above the subject. On the one hand the sound is sampled at 10 Hz. On the other we sample at 192 kHz. The first case has a TEMPORAL resolution of 100,000 microseconds (0.1 second), the other about 5 microseconds. Do you think that in both cases both subjects will be able to react in the same way, including detecting the position of the sound, to each stimulus? From the argument presented in one of the responses here one should be able to reconstruct the original waveform from the lower sample rate signal, ergo both subjects by the argument presented would be expected to react identically. Of course, I am saving the red herring, which is due to the fact that the 10 Hz sampled signal will be antialias filtered, removing frequencies above 5 Hz - so there would be nothing in the audio to react to! But the argument presented is that somehow one can reconstruct a signal by calculating the intersample waveform. The problem is there is no way "in the universe" to reconstruct the ORIGINAL "leopard pouncing" signal via such an operation. There is absolutely no way that a signal sampled at a given sample rate to a digital waveform has the same temporal resolution as a signal sampled at higher sample rates. If it did, then all the research labs in the world (using high speed analog to digital converters for signal sampling and recording) could just go and throw away all of our expensive high speed digital sampling hardware. Same thing with our 100 GHz sampling oscilloscopes. Why waste all that money when a 100 MHz scope has putatively the same temporal resolution? Because it does not. If a paper were submitted for publication to a peer reviewed technical journal and reported conclusions which depended on sampling a signal and the paper tried to infer that an event was determined to occur with a temporal precision greater than the sample rate, the paper would be rejected, with a suggestion to repeat with a more capable experimental measurement (that is, appropriate sample rate). Any argument that the original signal (that is, containing events faster than the sample rate) could be reconstructed perfectly from a lower sample rate signal would be ridiculed. There is absolutely no way this information could be extracted from a measurement made at a lower sample rate - which has lower TEMPORAL resolution. If the whole reaction we're trying to observe takes place in 100 nanoseconds and we can only sample at 1 microsecond, then we won't measure anything useful, let alone could we expect to somehow reconstruct the same data set that one would obtain at higher temporal resolution (sample rate). One cannot reconstruct an ORIGINAL signal containing spectral content at frequencies above half the sample rate, from one sampled at a lower sample rate, just because "there is only one (such) signal in all the universe," because the signal bandwidth of the "one signal in the universe" that fits the points has already been band limited by the antialiasing filter, and parts of the original signal discarded. The signal you get at the lower sample rate **only corresponds to band limited version of the original signal** - which is NOT the same as the original signal. There isn't any way to determine the ORIGINAL signal without sampling at a higher rate in the first place. You can certainly calculate the sample position at arbitrary inter-sample time intervals, but that will not give you the same result as sampling the ORIGINAL signal at a higher rate. Sure, one could sample the *band limited* signal at a higher rate, and in that case the result will be exactly the same - but there would be no practical reason to oversample (as in analog to digital conversion) a (severely, in the case of analog to digital conversion) band limited signal in the first place. The scenario with analog involves ORIGINAL signals which also are band limited; in the real world, all signals are band limited to some extent - it's a matter of the criteria applied; but analog is band limited to a much lesser extent than digital, and the characteristics of the band limiting are also quite different. Any signal which is lowpass filtered to prevent aliasing at lower sample rates is going to have a greater variation in group delay / phase shift near a fixed signal frequency (say, always at 10 kHz) than a signal sampled at a higher sample rate, presuming the antialias filtering is chosen appropriately for the sample rate (and it would always be, otherwise one wouldn't go to the trouble of using higher sample rates). A signal reproduced on a vinyl LP sourced from analog sources doesn't encounter the brickwall filter used in analog to digital converters / CDs. There is a high frequency roll off, but it is a gentle 6 dB per octave above the cutter head resonance of typically 50 kHz. By comparison the brickwall filter used to record CD format audio has a much steeper slope, about 100 dB of attenuation in a small fraction of an octave (antialiasing filters aren't even expressed in terms of dB per octave; just passband, stopband, ripple and attenuation). Furthermore, compared to our analog 6 dB per octave rolloff, there is a much greater phase shift variation among frequencies in the sampled signal. And just above about 20 kHz, there is NO signal left; on the other hand with analog, and vinyl, there is considerable signal energy, continuing for octaves above. And that is a key difference between analog and digital. So, in digital recording, we try to get closer to the TEMPORAL characteristics of analog by increasing the sample rate and pushing our brickwall filter higher and higher in frequency, so that the disruption of time relationships / coherence between the frequencies comprising our audible frequency range (up to 20 kHz, more or less) audio is minimized. Then, our music signal (or sound of the panther pouncing) is reproduced with the temporal relationships between different frequencies closer to (if not entirely preserved) compared to the actual sound found "in nature" and accordingly our auditory system / brain / emotional state responds more favorably by comparison. "Engaging, toe tapping, PRAT." As a bonus we can enjoy knowing about the technical elegance of capturing more of the frequency range of analog (and vinyl LPs) which extends strongly at least two octaves above the bandwidth limit of a CD (as proven by many measurements using spectrum analysis of signals from LPs, including those referenced in the post). Sorry for the long post. Good weekend! Regards Pure Vinyl Club
  15. 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.