Marvel Posted August 2, 2003 Share Posted August 2, 2003 I'm going to blather off the top of my head on this, so take with a grain of salt. It would seem the Nyquist theorem is totally accurate as long as you are dealing with a single sine wave, no harmonics or overtones. As soon as you have even a single acoustic instrument, say a guitar, the waveform becomes ever so complex and difficult to reproduce accurately. Yes? Rupert Neve's older analog consoles had bandwidth out the wazoo, easily past 50k, and I was thinking over 100k, but I can't find the interview (been trying to throw away old magazines, and always regret it at some point). To some golden ears, that high bandwidth worked its way through the chain somehow to where you could tell the difference on the final recording. The problem is, most equipment being used today doesn't even come close to that. What Dave is doing is putting as little between the mic and the signal as it is recorded, and the payoff becomes outstanding. I know my 16 input mixer doesn't have the bandwidth needed, but for $1100 how could it. Can most people tell the difference? Nope, but Dave is now ruined because he CAN tell the difference. If he had only stuck with using a cheap Fostex cassette deck and crummy tape to record with we wouldn't have these discussions. I have friends who are doing their recordings at 16/44.1 because they don't see the need. I think they are losing out, but I can't convince them otherwise. Perhaps if they could come listen to Dave's setup, they might change their minds. Marvel Quote Link to comment Share on other sites More sharing options...
ben. Posted August 2, 2003 Share Posted August 2, 2003 I agree, Marvel. I've noticed a huge difference between 16/44.1 and 24/44.1 on my Akai DAW. Quote Link to comment Share on other sites More sharing options...
Khorn Posted August 2, 2003 Share Posted August 2, 2003 For what it's worth IMHO the absolute best sounding recording that I have is an SACD derived from what was originally a 24Bits/96Khz master: Joe Weed: The Vultures" TopMusic SACD-1028. I find in gerneral that a SACD re-mastered from a well recorded original adds a certain "intagible" quality (musicality) that Redbook CD doesn't possess, also with SACD there is an absence of a "Digital Grundge" almost always present on Redbook in the past. One dynamite sounding Redbook IMO is: 'Eric Clapton Unplugged' Reprise CDW 45024 I am also impressed with some of the JVC XRCD stuff. In a musical sense I still think Higher Rez reaches beyond Redbook capability at least in my experience with SACD. Quote Link to comment Share on other sites More sharing options...
Mallette Posted August 2, 2003 Share Posted August 2, 2003 I find each post since my last to be in accordance with my belief and experience. I don't study equations. I listen music. While I am certain from excellent and reliable sources that power supplies, cables, caps, certain amp designs, etc. etc. etc. came make audible, perhaps significant, differences in reproduction, I believe much of this to be on the order of those concert goers who've been to many differenct halls and have, sometimes extreme, opinions on the sonics. OTOH, a crappy orchestra will sound even crappier in the worlds finest hall. So, it is my profound belief that the road to sonic happiness and contentment begins at the microphone. After that, you either need millions of dollars worth of equipment and expertise, OR absolutely minimal intervention and the most direct path to the distribution medium possible. As the former is not an option to me, anyone who gets one of my recordings gets a master with only the extraneous deleted. Dave Quote Link to comment Share on other sites More sharing options...
Griffinator Posted August 2, 2003 Author Share Posted August 2, 2003 I would like to request all of your indulgences, as I need to reply to each post individually in this sequence. ---------------- On 8/2/2003 10:01:19 AM Mallett wrote: Griff: You still don't quite understand that I don't have a dog in this race. I have no real interest in proving anything about CD's one way or the other. My interest is in high accuracy recording, especially sharing this 24/192 X 4 recording. I've recorded CD level for years, and my CD's sound better (to me and a few others at least) than all but a handful of those on the market. However, they don't sound remotely as good as LP's, or my 24/96 and up work. I presume you prefer to record for CD at 16/44.1 to start with. When I was first recording CD's, I used the Sony RM-500 with super bit mapping and achieved very nice results. It was my theory that it was pointless to record info you were just going to dither down inaccurately anyway. As to the comparison, we will be comparing the raw 24/88.2 material as well, apples to apples. And, of course, you would not only be welcome the 16th but anytime you can make it. I can only learn when proven wrong, which requires exposing my concepts to skeptics, not choir members. I am sure you will not care for this, but I find the "what equipment are you using?" "...is it calibrated?" etc. debate in such things as absurd. Only the fine caps and cables (no slight intended...I've made my respect for these folks known before) crowd would do this. Even your average audiophile neither considers nor indulges in such things. I am frankly more interested in how things compare on various systems of various configuration. However, we will be making some effort to keep things equal here in order to satisfy all tastes. Flynn: I am not familiar with the process you mentioned. As I mentioned before, I've largely abandoned the CD format for acoustic music and don't really pay much attention to it anymore. Dave ---------------- Goodness, man. I offered you a more accurate version of your test, and you're acting like I'm some kind of ultra audio snob. What I'm explaining to you is that the test you propose does not have proper controls. Hypothesis: You propose that ultrasonic frequencies have an impact on our perception of audible information. Your test procedures have way too many variables. 1) You propose to record the source material at 24/192 and then at 24/88.2. I'm not 100% OK with this, but I suspect there will be no significant difference between the two - certainly no audible one. 2) You propose to then downsample the 24/88.2 to 24/44.1. I have a big problem with this. Ask anyone who is involved in the recording industry what SRC (sample rate conversion) does to an audio file. They'll all tell you the same thing - that no SRC is perfect, and artifacts from the conversion process will inevitably ensue. Audible artifacts. Thus, you have already defeated the purpose of your test - see hypothesis - if anything differs between the two audio files other than ultrasonic information, then your experiment is flawed. 3) You then propose to truncate the 24/44.1 file to 16/44.1. Again, I submit that there will be audible artifacts from this process. When you truncate the wordlength on any file, there will be audible artifacts above the noise floor - why? Because the noise floor on most 24 bit ADC's is around -110dB, while on 16 bit ADC's it's up around -90dB. Big difference - when you introduce digital errors (from truncation) at +20dB above the noise floor, you will hear them. 4) You then propose to compare the two files (one that has been SRC'ed and truncated) in a blind listening test. I guarantee that 90% of your subjects will report that the 24/192 file will sound better, based on your testing procedures. I merely suggested that you perform an accurate test - one that ensures that the only difference between the files is the presence or absence of ultrasonic information. That's science, not audio snobbery. Quote Link to comment Share on other sites More sharing options...
Griffinator Posted August 2, 2003 Author Share Posted August 2, 2003 ---------------- On 8/2/2003 10:08:51 AM sunnysal wrote: http://www.geocities.com/bioelectrochemistry/nyquist100.jpg"> Harry Nyquist described the minimum rate needed to model a waveform. he did this to save BW on telegraph lines. this does not mean that higher sample rates are not better in some respects, certain applications. Many of us here at the b-board tend to talk over our heads, it is our right and part of the learning process, more power to us! this is a specific technical issue. The sample rate is one factor in perfecting musical sound sampling, storage and reconstruction. do not forget we are never seeing the ideal strived for, just what is practical or cheapest. regards, tony this article is quite interesting: http://www.iar-80.com/page25.html by J. Peter Moncrieff ---------------- Whether Nyquist made the description to save bandwidth on telegraph lines, the theorem was proven conclusively by Shannon 20 years later. That fact is unavoidable. The thing about "what is practical or cheapest" is pretty much an irrelevant non sequitur. Quote Link to comment Share on other sites More sharing options...
Griffinator Posted August 2, 2003 Author Share Posted August 2, 2003 ---------------- On 8/2/2003 12:44:48 PM bclarke421 wrote: I agree, Marvel. I've noticed a huge difference between 16/44.1 and 24/44.1 on my Akai DAW. ---------------- Of course you have. 24 bits yields a much better SNR than 16 bits. We're not talking about SNR here, we're talking about the audibility of ultrasonic frequencies. Bitrate has nothing to do with that. Quote Link to comment Share on other sites More sharing options...
Griffinator Posted August 2, 2003 Author Share Posted August 2, 2003 ---------------- On 8/2/2003 10:59:31 AM Marvel wrote: I'm going to blather off the top of my head on this, so take with a grain of salt. It would seem the Nyquist theorem is totally accurate as long as you are dealing with a single sine wave, no harmonics or overtones. As soon as you have even a single acoustic instrument, say a guitar, the waveform becomes ever so complex and difficult to reproduce accurately. Yes? Rupert Neve's older analog consoles had bandwidth out the wazoo, easily past 50k, and I was thinking over 100k, but I can't find the interview (been trying to throw away old magazines, and always regret it at some point). To some golden ears, that high bandwidth worked its way through the chain somehow to where you could tell the difference on the final recording. The problem is, most equipment being used today doesn't even come close to that. What Dave is doing is putting as little between the mic and the signal as it is recorded, and the payoff becomes outstanding. I know my 16 input mixer doesn't have the bandwidth needed, but for $1100 how could it. Can most people tell the difference? Nope, but Dave is now ruined because he CAN tell the difference. If he had only stuck with using a cheap Fostex cassette deck and crummy tape to record with we wouldn't have these discussions. I have friends who are doing their recordings at 16/44.1 because they don't see the need. I think they are losing out, but I can't convince them otherwise. Perhaps if they could come listen to Dave's setup, they might change their minds. Marvel ---------------- 1) Nope, you're way off about how the Nyquist theorem works. The theorem works with complex waves as well, as long as ultrasonics (which behave differently, and therefore are not subject to the theorem) are not involved in the equation. That's the whole purpose for the LP filters in audio recording and playback. Removing the ultrasonics makes the audible spectrum work under Nyquist's theorem. Why? Because no matter what the signal frequency is, there is ultimately a sine wave that can describe it - and anything slower than the highest frequency (22.05Khz) can be accurately expressed by the 44,100 Ks/S sampling rate. I'm not offering an opinion here, I'm offering scientific fact. 2) Your statement about Neve's consoles is also based on ambiguous information. Yes, Rupert's older consoles were capable of incredible bandwidth - but they were also superior to virtually every other console on the market in a variety of ways, not just the bandwidth. There's a lot more going into a console than just bandwidth limitations - how effectively the console can transmit information is the biggest one, which is why you see so many studios with consoles whose power supplies have been replaced - more voltage in = more effective transient response (particularly in the lower end of the spectrum) - Until you can prove that the only difference between a Neve and a Studer, for example, is the ultrasonic bandwidth, I'll tell you that you don't know the real reason it was superior. Yamaha now makes an HTiB with speakers that claim to handle 50Khz. It also happens to sound like ****, regardless of the source. Why? Because the speakers can't handle a modest amount of power (including 3/4 of what the amp is capable of delivering) without clipping. 3) I'm not suggesting that 16/44.1Khz is the end-all. Fact is, even the experts on ADC (including the guy who wrote that paper) state that 24/48 recording is a must for accurate representation. Why? Because a) 24 bits gets all the digital recording errors well below the audible spectrum (there has never been a human that could hear below -144dB) and a properly designed, high quality ADC system will be able to perform the LP filtering outside of the audible spectrum at 48Khz, but even the best systems have trouble at 44.1Khz - it's just too close to the audible range. I record my client material at 24/48 whenever possible. The only reason to record at higher than 48Khz? Cheap ADC's - or if you're authoring to DVD or DVD-A, which require the higher bandwidth. Most DVD-A and DVD material, when examined with a spectrograph, do not contain ultrasonics - most have no information above 23Khz. If you doubt this assertion, by all means, rip your DVD-A or DVD to wave, take a look at it in Wavelab or Cool Edit, and then tell me I'm wrong. Not trying to dog you, man, but understanding the way digital audio works is the key to understanding what effect different samplerates and bitrates have on your results. Most of the time, higher samplerates are a vehicle for manufacturers to use lower quality (and thus cheaper) ADC and DAC systems to acheive the same result that a high quality ADC or DAC can acheive at much lower rates. It's a lot cheaper to implement a poor reconstruction filter system at an extremely high bandwidth than to implement high grade filters at low bandwidths. Quote Link to comment Share on other sites More sharing options...
Griffinator Posted August 2, 2003 Author Share Posted August 2, 2003 ---------------- On 8/2/2003 1:25:02 PM Khorn wrote: For what it's worth IMHO the absolute best sounding recording that I have is an SACD derived from what was originally a 24Bits/96Khz master: Joe Weed: The Vultures" TopMusic SACD-1028. I find in gerneral that a SACD re-mastered from a well recorded original adds a certain "intagible" quality (musicality) that Redbook CD doesn't possess, also with SACD there is an absence of a "Digital Grundge" almost always present on Redbook in the past. One dynamite sounding Redbook IMO is: 'Eric Clapton Unplugged' Reprise CDW 45024 I am also impressed with some of the JVC XRCD stuff. In a musical sense I still think Higher Rez reaches beyond Redbook capability at least in my experience with SACD. ---------------- There's two reasons for this. Neither are high resolution. DSD does not depend on the extensive filtering systems that PCM employ - that's the first reason. The second reason (and I can refer you to legendary engineers that have worked with the SACD format to verify this) is that SACD is extremely unforgiving - try to push the dB limitations of SACD as an engineer and you'll hear it immediately - it literally forces the engineer to embrace dynamic range. Quote Link to comment Share on other sites More sharing options...
Griffinator Posted August 2, 2003 Author Share Posted August 2, 2003 ---------------- On 8/2/2003 2:50:56 PM Mallett wrote: I find each post since my last to be in accordance with my belief and experience. I don't study equations. I listen music. While I am certain from excellent and reliable sources that power supplies, cables, caps, certain amp designs, etc. etc. etc. came make audible, perhaps significant, differences in reproduction, I believe much of this to be on the order of those concert goers who've been to many differenct halls and have, sometimes extreme, opinions on the sonics. OTOH, a crappy orchestra will sound even crappier in the worlds finest hall. So, it is my profound belief that the road to sonic happiness and contentment begins at the microphone. After that, you either need millions of dollars worth of equipment and expertise, OR absolutely minimal intervention and the most direct path to the distribution medium possible. As the former is not an option to me, anyone who gets one of my recordings gets a master with only the extraneous deleted. Dave ---------------- And thus your road has led you to using high-bandwidth converters. Great. Enjoy them. But don't delude yourself into thinking that the ultra-sonic information they capture has any effect on the audio quality they deliver. If you'd like to prove your point about ultrasonics, then employ the testing procedure I submitted to you. If you want to prove that high-rez converters do a better job than low-rez converters of the same quality, there's no need. I already know that. Filtering has everything to do with how PCM is implemented, and the quality of the audio it delivers. Quote Link to comment Share on other sites More sharing options...
Mallette Posted August 2, 2003 Share Posted August 2, 2003 OK, Griff. To be honest with you, I didn't read every bit of your well crafted, obviously well thought out replys. Like I said, I don't have a dog in this race. TBHWY, I don't give a hoot in hell about scientific fact anymore in my audio than I do in my religion. I can't even remember why you said I couldn't possibly be hearing whatever it is I am hearing, but it's metaphysically absurd regardless of it's scientific basis. Consider: How can you know what I hear? I've yet to question those who claim to hear the difference between a Mullard and a light bulb (or whatever). Why should I question the abilities and motivations of people I don't even know? They seem honest and well intentioned enough. Now, on the up side, I'd like to know your thinking on why 24/88.2 downsampled to 16/44.1 should deteriorate beyond the simple data loss. There is no dithering applied, simply the tossing of every other bit. As to the 24/16, this should simply be tossing some relatively useless noise below the signal. That's what I've heard from those who are supposed to know. I've wondered about it, but the logic seems sound, and I don't seem to hear any artifacts not explainable by the (OK, debateable) limitations of the medium. Finally, due to your input, I've decided (I thought I mentioned this) to use only the 24/192, 24/96, and 24/88.2 material at the Hornhead gathering. That should lay to rest at least part of your concerns. If the group hears significant difference between the 24/88.2 and above, then I think we can rest assured 16/44.1 need not apply to this group. I hope you understand I am listening, and your input is valuable. Dave Quote Link to comment Share on other sites More sharing options...
Griffinator Posted August 2, 2003 Author Share Posted August 2, 2003 ---------------- On 8/2/2003 6:34:12 PM Mallett wrote: OK, Griff. To be honest with you, I didn't read every bit of your well crafted, obviously well thought out replys. Like I said, I don't have a dog in this race. TBHWY, I don't give a hoot in hell about scientific fact anymore in my audio than I do in my religion. I can't even remember why you said I couldn't possibly be hearing whatever it is I am hearing, but it's metaphysically absurd regardless of it's scientific basis. Consider: How can you know what I hear? I've yet to question those who claim to hear the difference between a Mullard and a light bulb (or whatever). Why should I question the abilities and motivations of people I don't even know? They seem honest and well intentioned enough. Now, on the up side, I'd like to know your thinking on why 24/88.2 downsampled to 16/44.1 should deteriorate beyond the simple data loss. There is no dithering applied, simply the tossing of every other bit. As to the 24/16, this should simply be tossing some relatively useless noise below the signal. That's what I've heard from those who are supposed to know. I've wondered about it, but the logic seems sound, and I don't seem to hear any artifacts not explainable by the (OK, debateable) limitations of the medium. Finally, due to your input, I've decided (I thought I mentioned this) to use only the 24/192, 24/96, and 24/88.2 material at the Hornhead gathering. That should lay to rest at least part of your concerns. If the group hears significant difference between the 24/88.2 and above, then I think we can rest assured 16/44.1 need not apply to this group. I hope you understand I am listening, and your input is valuable. Dave ---------------- Fair enough. I still suggest that using a simple 22.05Khz LP filter on the original 192Ks/S source would be the most effective A/B comparison, but you are certainly a large step ahead of where your original experiment sat scientifically. As far as my thinking about 24/88.2 vs. 16/44.1, there are two significant problems I will mention, and quite a few others I won't go into (I'm not trying to bake your brain here, just explaining the issues you're dealing with) 1) 88.2Ks/S to 44.1Ks/S is not "throwing out every other bit". SRC's do not function in this manner. SRC's resample the entire signal, regardless of the original signal's samplerate. Are you seeing how this could be problematic? 2) 24 bits truncated to 16 bits is definitely not "throwing out every other bit". Bits are not relevant to the frequency of the information, only to the amplitude. Each bit results in approxamitely 6dB of dynamic range. 24 bits provides up to 144dB of dynamic range (depending on the implementation thereof, of course - cheap 24 bit converters can deliver 120dB or less) where 16 bits provide a maximum of 96dB. Considering that the human ear can very effectively hear down to -100dB, while precious few can hear below -120dB, it's not difficult to see how the dynamic range capabilities of the two bitrates are very important. Couple that with the fact that, when you truncate the wordlength (bit rate) of a given signal, you delete everything below the least significant bit of the destination wordlength, it's not difficult to see how simply truncating from 24 bits to 16 bits can have a dramatic effect on the resulting audio signal at its lower amplitude extremes. This is the whole reason dithering was created - to at least mask the digital errors created at the lower end of the 16 bit scale when the processing of that signal was done at a higher (24,32, and in some systems even 48 bit) wordlength. Given these two issues alone, are you understanding the issues I have with the resampling and truncation of the original signal? Obviously, you're working with a digital audio workstation of some sort. I may be assuming too much to suggest that it's a computer-based system. It's going to be a lot easier to record the source once at 24/192 and then run a 22.05Khz highpass filter on it. It will also give you far more accurate results. Quote Link to comment Share on other sites More sharing options...
Mallette Posted August 2, 2003 Share Posted August 2, 2003 It's entirely computer based. Obviously, 24 to 16 is not throwing out every other bit. Just disposing of the least significant 8. Not sure how you assumed otherwise, though probably through my quick, not tailored, response not being clear. As to resampling, I do not resample to downsample from rates divisible by 2. I do simply tell the computer to dispose of every other bit. Call it destructive compression without compensation. Dithering would be pointless and impossible to do accurately (OK, I shouldn't say impossible...but I don't have the software to do it or know were to get same). In theory, it should be identical to the data that would result from recording at 16/44.1 in the first place. Is there a problem with this logic? Dave Quote Link to comment Share on other sites More sharing options...
Griffinator Posted August 2, 2003 Author Share Posted August 2, 2003 One more thing - I'm not arguing whether you are hearing what you are hearing, but rather why you are hearing what you are hearing. There's a lot more to A-D-A than mere samplerates. If it were that simple, we wouldn't have A-D and D-A converters ranging in price from $50 up to $7000 per channel. It's not about the "audio snob" game of esoteric capacitors and all that crap - it's about how well one manufacturer implements the PCM system vs. how well another manufacturer implements the same system. Do you do it with quality components and complex circuitry to acheive high-quality algorhithm? Or do you do it with cheap circuitry and simple, but poorer quality algorhithm? Quote Link to comment Share on other sites More sharing options...
Griffinator Posted August 2, 2003 Author Share Posted August 2, 2003 ---------------- On 8/2/2003 7:21:44 PM Mallett wrote: It's entirely computer based. Obviously, 24 to 16 is not throwing out every other bit. Just disposing of the least significant 8. Not sure how you assumed otherwise, though probably through my quick, not tailored, response not being clear. As to resampling, I do not resample to downsample from rates divisible by 2. I do simply tell the computer to dispose of every other bit. Call it destructive compression without compensation. Dithering would be pointless and impossible to do accurately (OK, I shouldn't say impossible...but I don't have the software to do it or know were to get same). In theory, it should be identical to the data that would result from recording at 16/44.1 in the first place. Is there a problem with this logic? Dave ---------------- 1) rates divisible by 2 do not make for more accurate sampling. This seems logical to the algebraic mind, but it is not the reality of sample rate conversion (SRC). You're not telling the computer to dispose of every other bit in this situation, you're trying to tell it to dispose of every other sample - but it's not listening to you - it's instead reconstructing the waveform and resampling it. No SRC "removes every other sample" when downconverting from 88.2 to 44.1 (or 192 to 96, or 48 to 24, or 2 to 1, for that matter) 2) Dithering has absolutely nothing to do with your sampling frequencies - only the bitrate. When you remove 8 bits from the bottom of your dynamic range, dithering steps in and goes "if this was going to fade away to -96dB (in a 16 bit final), what are the odds that there would be sound at this given sample point at -96 to -90dB? It then fills in the blanks based on randomization algorhithms. If you wish to experiment with dithering, I'd point you immediately to Wavelab - an excellent program for this purpose. It has several high-quality dithering algorhithms, with several noise-shaping options to enhance the process. Try dithering with noise shaping a few times versus truncated wave files, you'll immediately hear the difference - especially on fades. Is this a clearer explanation of why 24/88.2 does not convert to 16/44.1 without loss? Quote Link to comment Share on other sites More sharing options...
Mallette Posted August 2, 2003 Share Posted August 2, 2003 As to 1, I am precisely telling the computer to dispose of every other bit. As to 2, are you suggesting the computer is dithering anyway when I am saying "no" to dithering? In either case, I am not hearing any artifacting or other loss in the signal that does not seem to be connected to the idea that ultrasonic information does not in some way contribute to accurate, realistic reproduction. Again, I state: The results are not predictable by equations, algebra, or any other quantifiable means. Human perception is no closer to predictabilty than something that explains gravity. I've listened to audio systems whose specs implied they were near perfection that sounded like refried crap. I've also heard systems of humble origins and dirty specs that sounded like music. I trust my ears. Nothing else. Dave Quote Link to comment Share on other sites More sharing options...
Marvel Posted August 2, 2003 Share Posted August 2, 2003 Where do the studies show that ultrasonics behave different than what we consider audio frequencies? Wave theory doesn't necessarily change as the frequency goes up. This fits with Dave saying he listens to electronic music on CD. It makes so much sense, as most electronic music (rock included) does not use any acoustic instruments. Acoustic guitars more often than not use either magnetic pickups or piezo pickups, and they certainly reproduce more of the fundamental as opposed to the fundamental + harmonic content. That's what makes an acoustic guitar sound like an acoustic guitar. Plug in a piezo and you really can't tell much difference from a $150 Epiphone and a $3000 Martin. Put them in front of a mic and the difference is obvious. I can imagine what voices would sound like with a piezo pickup. Oh, perhaps early crystal mics? Now I remember. I didn't give ambiguous information about Neve consoles. Maybe just not complete. They have been superb for more than their wide bandwidth, but it was something Neve himself commented on. The wide bandwidth makes a huge difference. The resolution given by using a higher sample rate is a plus in every case, the only drawback is file size and throughput on the playback system (and record system). It is a trade off that Sony/Philipps made when coming up with the Redbook spec, given the technology of the times. Early brickwall filters were awful and introduced artifacts of their own. Using a higher samplerate means you can make a smoother/better lp filter. We all win. It is really no different than our fellow Klipschers who say they can only hear up to 12k, but can still hear a difference in speakers whose response is far higher. Marvel Quote Link to comment Share on other sites More sharing options...
Griffinator Posted August 3, 2003 Author Share Posted August 3, 2003 ---------------- On 8/2/2003 8:48:07 PM Marvel wrote: Early brickwall filters were awful and introduced artifacts of their own. Using a higher samplerate means you can make a smoother/better lp filter. We all win. It is really no different than our fellow Klipschers who say they can only hear up to 12k, but can still hear a difference in speakers whose response is far higher. Marvel ---------------- And here is the crux of the matter. "Using a higher samplerate means you can make a smoother/better lp filter." - exactly my point. It's not the presence/absence of ultrasonics that make for a better sounding product, it's the smoother LP filters in PCM (and the extremely high rolloff points - @ 50Khz - in SACD) that generate the better sound. BTW - those who read the white paper, what was your response? Quote Link to comment Share on other sites More sharing options...
3dzapper Posted August 3, 2003 Share Posted August 3, 2003 All of this information is very nice, but CDs are and will continue to be "harsher" than vinyl for other reasons as well. One is the law of physics. The moving mass of the stylus/magnet/coil assembly takes time to accelerate when tracking the groove both in cutting and playback. This has the effect of softening the "attack" of the recorded instrument. The other is electrical. A moving magnet in a coil or moving coil in a magnet produces a softening of the attack. A vinyl recording can not play back a very good square wave, a cd can. A tube amp will ramp up the front edge of a square wave, a SS amp will more closely follow it. This is why most CDs sound like Crap on our Klipschorns. When CD producers figure out that life like music does not so much resemble what the digital tape sees as accurate reproduction as opposed to the sound of an open miked hall, we will finally get listenable CDs. 'Til then I'll listen to Chicago on vinyl. Just my .02. I don't have any fancy scientific journals to cite just my ears. Rick Quote Link to comment Share on other sites More sharing options...
MBM135 Posted August 3, 2003 Share Posted August 3, 2003 I've noticed a great disparity in the quality of CD recordings too. CD remakes of older stuff like Chicago, the Eagles, Fleetwood Mac etc sound weak. But when I play a CD made since the late 1990s they sound 100% better--louder and more clear. 1 Quote Link to comment Share on other sites More sharing options...
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