I have a 24/96khz ad/da and maintain audio at 32 bits internally for processing and mixing prior to dithering back to 16 bits for burning to CD.
Is there any advantage in recording at 88.2 or 96khz and then sample rate converting back to 44.1 at the end for CD? My guess is that any advantage will be completely lost due to both artifacts caused by the sample rate conversion process and the ultimate 22.5khz upper frequency limit and associated filtering. :confused:
Comments
First off to Ethan, Respect!! But ... I have done a/b compariso
First off to Ethan, Respect!! But ... I have done a/b comparisons using a 2" analog tape playback through a MCI 600 console and an Apogee PSX 100 at 96, 48 and 44.1 and the difference between 48 and 96 is dramatic! It doesn't take a golden ear to hear it. I concede that it may not be so noticeable in all but a direct a/b comparison. So when tracking in digital at higher rates and better high frequency response, the lower harmonics are excited by the upper ones. While you are going to down sample to a lower rate for Redbook, there is a difference in the end product. Fats
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Tannoy, Dynaudio, Blue Sky, JBL, Earthworks, Westlake, NS 10's :D , Genelec, Hafler, KRK, and PMC
Those are good. …………………….. Pick one.
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Fats, this is true. Another reason to do things in hi-bit is you
Fats, this is true. Another reason to do things in hi-bit is you may want to release a DVD-A disc.
Their are two consumer audio high resolution digital formats. Both of them are growing but their is a huge division between the users of the formats. It is like the VCR wars of the past between beta and vhs.
The formats are:
DVD-A (digital versatile disc/Audio format)
SACD DSD (Super audio compact disc/Direct stream digital)
DVD-A discs are played back through a DVD-A player. This is a different machine that the DVD machines for movies. All DVD-A compatible playback machines are also DVD players but DVD players may or may not have DVD-A compatibility. DVD-A's are multichannel (6 channel/5.1) and they must sum this downmix internally for 2 channel reproduction. They are using PCM (pulse code modulation) at 24 bits, 88.1, 96 or 192K. Most discs are 24/96 or 24/192. The authoring in mastering for them is slightly more expensive that redbook (40% more) but if you are prepared to release your material in DVD-A, upsampling an original 16/44.1 to the format is ill advised unless it is remastered professionally and authored professionally, including text, footnotes and video.
For me to do a DVD-A, it cost MUCH more than redbook. I have the goods to do it..but the format is considered "new" and not many takers yet...really no takers. I have done a few for ME and it is a wonderful format. Head and shoulders above redbook.
SACD/DSD
This format seems to be the choice for the audiophiles. Direct stream digital samples at single bit 2.82MHZ! The format can be multichannel (5.1) stereo channels (2-bus..not summed) and contains the redbook layer as well. The days of an SACD only disc are soon to pass as all of the current releases on SACD are "dual layer hybrid" and can work on redbook or SACD players. At this time, the cats that are hand selected by Sony are butchering the format IMHO. Of the SACD's I own, I have redbook CD's that slay the SACD's. The guys in charge of remastering for SACD on "average" are missing the mark. Roger Waters "in the flesh" and "TOTO IV" are sad examples of this format. For good examples, look toward TELARC. They do it better. Sony has its proverbial head up its ass with this "superior" technology. It has been out now 4 years and their are only 800 titles. The one to watch for will be released 03/03/03. It is "dark side of the moon" by Pink Floyd. Unfortunantly, Alans original Quad mix was not used and he was not invited to remaster it (Head up ass syndrome again) so until this is officially released, we will have to wait and see if it can come close to the QS4 Quad album. To master to SACD means it basically will cost you an arm and two legs. Right now this boys club is doing them at 10 grand a pop and pressup is around 7.75 per unit. IMO, the format is not worth it due to the crappy mastering I have experienced. Every SACD I own could have been done much better by your truly. This is fact..not blowing smoke here. Michael has heard it too.
Ok, sorry for the book but really, if you want to use one of these emerging formats, you should be recording at least 24/192 or 24/96..
SACD/DSD mastering stations are still basically too expensive to make money with right now and DVD-A is extreamly affordable, time consuming (multichannel) and can be burned in house without being a slave to Sony. Their is only one pressing plant for SACD's in the USA at this time. The total now is up to 4. Anyone with 500 dollars can burn a DVD-A.
Fats, > First off to Ethan, Respect!! Hey, all of this is
Fats,
> First off to Ethan, Respect!! <
Hey, all of this is with great respect for each other, and a desire to learn and understand. And also with the knowledge that what works for one person doesn't have to work for another.
> I have done a/b comparisons using a 2" analog tape playback through a MCI 600 console and an Apogee PSX 100 at 96, 48 and 44.1 and the difference between 48 and 96 is dramatic! <
Just so I'm clear, you are saying that you did two identical mixdowns, one right after the other, where nothing changed except the sample rate? And I assume you recorded into a computer DAW?
If the mixes were identical and a difference was clearly audible, I wonder what actually changed from one pass to the other. I don't believe anyone can hear much past 20 KHz., or even that our hearing is influenced by supersonic content. But some people do claim to hear a meaningful difference between 44.1/48 and higher sample rates. So my interest is finding out what is changing in the audible band when the sample rate is switched.
> I concede that it may not be so noticeable in all but a direct a/b comparison. <
If it's clearly audible then it's definitely worth understanding!
--Ethan
PS: Why is it that whenever thus stuff comes up, the other person is too far away for me to drive over so I can listen for myself? :D
Ethan, you are right that doubting anyone can hear past 20K; ALT
Ethan, you are right that doubting anyone can hear past 20K; ALTHOUGH... the point is it is not that you can hear that high, it is what the presence of the freqencies above 20K do to the shape of the waves we do hear. Take a 30KHZ frquency sine wave...pulse it from -20 to -6dB in slow (about 3 seconds between) fade ups and fade downs While you have a steady state 1KhZ @ -4dB tone running. You see how the presense of the 30khz wave changes the shape of the 1khZ tone. Do this with loudspeakers making sure A, they can get up there and B, you are point source from the tweeters as loudspeakers dispersion at 30 K is very narrow. Providing your mic pres and the microphones can do it, the exact same thing happens on playback only more so. The presence of the signal changes the shape electrically and the presence of the signal "in the air" changes the shape acoustically. When I put a 50khZ wave through the system at 85dB reference to a 1KhZ signal..my Little Dogs head will twist and turn looking at the speakers. It is there.
Now, like I say, it is not that we can hear these freqencies but it is there presence that changes the shape of the ones we do hear and that...is clearly audible.
When downconverting, the changes in the shapes of audible freqencies are retained.
Flat response to 100KHZ is the goal for high fidelity reproduction IMHO
Bill, > the point is it is not that you can hear that high, i
Bill,
> the point is it is not that you can hear that high, it is what the presence of the freqencies above 20K do to the shape of the waves we do hear. <
I don't see why that would matter. Yes, if you have material with audible content and also supersonic content, the supersonic content will be visible on a 'scope and will come out your speakers if they can reproduce that high. But since nobody can hear the supersonic portion, I don't see how it's presence affects the audible portion.
For example, if you strike a cymbal or orchestra triangle, frequencies far beyond 20 KHz. are generated. You can then capture that with a microphone that responds to super high frequencies. I have done this, and watched the result on a hardware spectrum analyzer and seen content all the way out to 50 KHz. But though the content past 20 KHz. is present and visible, it contributes nothing to the perceived sound.
Years ago a friend who's an engineer at Hewlett-Packard brought over some high end test gear, including the spectrum analyzer mentioned above. He also brought over a sweepable low-pass filter, and we played with that too using a set of car keys jingling in front of a good mike. This is the test where we saw content out to 50 KHz. We then lowered the filter cutoff frequency until we could hear a difference - the knob read 18 KHz. or something close to that.
Which brings us back to why a recording made at a high sample rate sounds different than 44.1.
--Ethan
I guess I was not clear...put it in other terms. The interfer
I guess I was not clear...put it in other terms.
The interference patterns that the presence of ultra high frequencies create add to the color of the ones we do hear. Removing them and the inteference is removed. Buy having them there in the first place, the interference that they caused will hold over on the audible freqencies. If they were never there to begin with, the intererence would not be there.
If you drop a large rock into a pond and watch the waves, dropping a small pebble within those waves "interferes" with the large waves. Had the small pebble not been dropped, it would not have happened. Now if you have the fundamental of the longer wave intact with the interference of the pebble saved, you would not need said pebble to duplicate the event. The smaller pebbles signature is already "set in stone" (pun intended).
This make any sense at all?
By going back to 44.1K you are not reshaping the lower waves, the interference is still there, you are simply removing the source of the interference.
Perhaps I need to do some screen shots and show you what I mean.
It will take time though..and really today, I have paint work to finish.
Bill, > The interference patterns that the presence of ultra
Bill,
> The interference patterns that the presence of ultra high frequencies create add to the color of the ones we do hear ... dropping a small pebble within those waves "interferes" with the large waves ... This make any sense at all? <
I did understand you the first time, but I'm afraid it does not make sense. Using your analogy, fast ripples correspond to the high frequency content and slow ripples to the audible content. But unless there is a nonlinearity in the system - distortion - one does not affect the other. And one can be filtered out completely without affecting the other in any way.
I know it may seem that this would be audible, but it's really not. Again, in order for frequencies to interact there needs to be a non-linearity somewhere. And then you'll have IM distortion which creates sum and difference products. But in modern digital systems IM and other distortion components are extremely low - likely below the -96 dB. noise floor of 16 bits. A distortion figure of 0.01% means the distortion components are 80 dB. below the signal. I think most modern A/D converters can manage even better than that. Further, anything that's -80 below the music, and also masked by the music in addition to being so soft, is inaudible as far as I'm concerned.
> Perhaps I need to do some screen shots and show you what I mean. <
No, I already know what high and low frequencies at the same time looks like. :)
--Ethan
Bill, > You will always get wave interference when you pass t
Bill,
> You will always get wave interference when you pass two soundwaves through a medium called "Air" <
Yes, but not in the way you are thinking! Again, the interference must be nonlinear in order for one frequency to "interact" with another. Without nonlinearity the two frequencies are merely combined, and so can be easily separated again later with simple filtering. One does not affect the other in any way; they merely coexist at the same time on the same piece of wire, or in the same volume of air.
Let's back up a few messages...
> The interference patterns that the presence of ultra high frequencies create add to the color of the ones we do hear. <
What sort of color do they add? What does it sound like? Most important of all, how do you know this is true, and what tests have you performed to arrive at this conclusion?
> having them there in the first place, the interference that they caused will hold over on the audible freqencies. <
Again, the same questions: What sort of interference specifically? What does that interference sound like? How do you know this is really the case?
I'm sorry if this sounds confrontational, because that's really not my intent. :) But you can't just say there's "interference" and so it must be audible.
--Ethan
I need to paint today..but I can demonstrate the shift in the sh
I need to paint today..but I can demonstrate the shift in the shape of 1KhZ when 50K is applied and this is at 24/192 then I can resample this to 16/44.1K and the 1K wave is still shifted, even with the absense of 50K.
No system is absolutely linear.
I suppose that manufactures made a huge mistake in developing higherbit recording technologies?
What test have I done,
Many. Their would be no point whatsoever for any sampling frqency above 32K if we could only hear to 16KHZ? IS that wat you are saying?
So, test one involves 3 computer towers running at 3 different sampling frqencies. Tower one was running 24 bit/192K. Tower 2 at 24/96K and tower 3 running at 16/44.1K when I recorded a classical concert in the year 2001. I did this so I could burn 3 separate 16/44.1K and put this to rest for me. Hell, I don't like rendering a 4gig file anymore than the next cat.
I burned 3 CD's of each recording directly and I had another output going to a stand alone 1" machine (16track @15ips) .
The files at 192K that I burned 3 CD's from were the best. They were alive, cleaner, clearer. Their are members of this board and even moderators of this board that have heard this CD. One moderator heard both the 24/192K transfer and the 16/44.1K transfer.
That was the test, it conclusively proved that their is a reason to go to higher sampling frequencies.
Second. Why is their any equipment that goes above 44.1K? Why are DVD-A selling? Why are SACD's Selling?
Because 44.1K is a lousy idea from the get go. Only reason it was settled on was technology was not up to par at that time and video recorders were being used to transfer and edit audio digitally in the old days.
Perhaps my loudspeakers and my room and my equipment and my ears can account for most of the differences and why I master in higher bitrates. Fats has heard the difference. Rick Hammang has heard the difference and depending on the set-up, it is there. The difference is *it sounds closer to the actual performance*
I can take the 9CD's and 100% blindfolded in even the car stereo, I tell you which is which. Ever heard of ringing? What happens to the frequency response as you approach nyquist? The filter is so sharp that when you use it in recording, use it in mixdown, use it in mastering and use it in pressing..all this adds up.
Perhaps my ears are playing huge tricks on me. You do not have to believe. But if you want to see if I know what I am doing or not, you can go over to the Audio Projects site and look at the comments of artist I have remastered their wares and see what they think of the work.
We are moderators here. To tell me I do not know what I am talking about will meet resistance, especially when I can prove it time and time again.
Bill, Let me address your last point first. > But if you w
Bill,
Let me address your last point first.
> But if you want to see if I know what I am doing or not ... To tell me I do not know what I am talking about will meet resistance <
This is what I'm trying to avoid! My goal is to stay on topic and discuss the issues only. If I implied anything otherwise I apologize as that surely was not my intent.
> I can demonstrate the shift in the shape of 1KhZ when 50K is applied and this is at 24/192 then I can resample this to 16/44.1K and the 1K wave is still shifted, even with the absense of 50K. <
Yes, this is exactly what I'm getting at. Though I'd rather see a spectral analysis than a picture of waveshapes, because waveshapes can be changed dramatically with no audible affect by, for example, phase shift.
> No system is absolutely linear. <
Agreed, and that's why I made the point that nonlinearities in modern digital systems are low and further masked by the program itself. So this is one avenue worth persuing: independent of ultrasonic content, are components that are 80 dB. or whatever below the program audible?
> Their would be no point whatsoever for any sampling frqency above 32K if we could only hear to 16KHZ? IS that wat you are saying? <
Almost. I am not saying that higher sample rates cannot sound different than standard 44.1. But what I do question is the reason why they sound different. If it turns out that the difference is due to avoiding ringing near the filter cutoff, then perhaps the ultimate solution is to make better filters, or oversample at an even high rate, to avoid the waste of resources used by high sample rates yet still retain the improved fidelity.
> Hell, I don't like rendering a 4gig file anymore than the next cat. <
Right, and this is precisely my interest in getting to the bottom of what matters, what doesn't, and why.
--Ethan
OMG this is so funny! MODERATORS in debate! Nobody's getting rud
OMG this is so funny! MODERATORS in debate! Nobody's getting rude or disrespectful, it's all good, so I aint toughing this! Keep it clean, guys, because this is great! FINALLY, two guys that can have a good debate without going below the belt!
BTW ... even though I only have a little home studio, and seldom participate in the studio threads, even I know there is no grey area on this question. One of you is right, and one of you is wrong, but I'll be damned if I'll say a word! I wanna watch this! ROTFLMFAO!
A better filter. Well they should have figured out that at least
A better filter. Well they should have figured out that at least 350K sampling frequency would allow for shallow slope filters to be used but you have to be at least 70dB under the source level at the sampling frequency. 22.5K means 70dB per octave. and the therum is more like 83dB. Filters that cut off that steep will ring. No doubt about it. It is audible to even this 44 year olds ears. The ringing starts in the 12K region (9th order harmonics) and is clearly audible. Now if you have the same situation done 5 times over (recording, mixing, editing, mastering, in the CD player) you cannot possibly say that this is inaudible. The reason for 192K is to get you at least flat to 50K before the ringing at 65K starts. Filters are filters. The steeper the slope, the more they will ring. When I designed loudspeaker crossovers commercialy, I was adimate that the tweeters be beefy enough in construction but light enough for transient response that I could do 1st order because it sounds so much better and intergrates with other components so much better. I actually came up with 3dB/octave slopes using a battery of bypass resistors to cut the sharpness of even a 6dB/octave slope down some.
Nope, filters are the culpret and the only way to overcome the ringing filter syndrome (not to mention leaving the upper harmonics intact whether you hear them or not..I DO... I know when they are there or not instantly) is to raise the sampling frquency to astronimical levels and employ less steep curves to the shelf of the filter. A multishelved filter (as philips employes in my A/D D/A) is fully adjustable and audible. I got this piece of gear from a corsortum of manufactures because of my involvement in testing, calibration and reviewing in the Higher fidelity realum.
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I can simply say this. At no time will digital ever be able to give flat freqency response that is necessary to humans to achieve the best the format can offer without limitation, if steep filters are employed in the audible spectrum. Many folks have the ability to hear the absense of upper harmonics. I for one can every day , even with a headcold. I have been employing wave shaping techniques to digital mastering now for 20 years...yes since late 1982. I know that going back to analog and repairing the losses and putting it back together does restore the build-up of all these filters and it is not so bad of it happens once, instead of multiple times.
Ok Smawg, IF I am wrong, Fire me. I got other things to do than beat this tired old horse.
I don't want to step into the middle of this and incur anyone's
I don't want to step into the middle of this and incur anyone's wrath, but…..there is actually a question at the end of a bit of rambling.
When asked about sample rates by people who know nothing about technology, I draw them an analog sine wave. Think of sample rate as taking with a camera "snapshots" of that sine wave. At 44.1k we really have 44,100 snap shots per second. If we were to zoom in, we would see a stair step effect as there was signal happening between snap shots. Simple logic (as works in my simple mind) would seem to indicate that by doubling or tripling the number of snap shots taken in that same amount of time, would result in smaller "steps", getting us closer to the pure smoothness of our original analog signal. It would seem to me that regardless of output, starting with the highest possible resolution would have to result in a smother (truer to analog, less digitized) sound.
Q: Does that not seem logical regardless of what frequencies I can hear?
Doug, > It would seem to me that regardless of output, starti
Doug,
> It would seem to me that regardless of output, starting with the highest possible resolution would have to result in a smother (truer to analog, less digitized) sound. <
Yes, it would seem that way, but that's not really how digital audio works. The filtering by the D/A converter smooths out the steps so as to completely restore the original waveform.
What results when the bit rate is low is distortion that can be directly computed based on the number of bits. Assuming, of course, a D/A converter that is high quality and thus limited by the number of bits and not limitations in its own circuitry. Even 16 bits, when recorded full scale, can yield very acceptable distortion values.
And what results when the sample rate is low is a reduction of the highest frequency that can be captured. Theoretically, 44.1 KHz. can capture at least to 20 KHz., though as you can see there is some disagreement as to the audibility of other factors, such as filter ringing and other artifacts.
--Ethan
Once again,.... I have done a/b comparisons using a 2" analog t
Once again,....
I have done a/b comparisons using a 2" analog tape playback through a MCI 600 console and an Apogee PSX 100 at 96, 48 and 44.1 and the difference between 48 and 96 is dramatic! It doesn't take a golden ear to hear it.
I know this is true. Higher sample rates in tracking sound better when downsampled. Not dithering , that's bits but sample rates. I have heard this in A/B tests in a controlled environment on a good monitor system. Believe me "the earth ain't flat!". It's not like I have an agenda regarding this. I would love to be able to say don't worry about it, it makes no difference, save your money. But I can't. it makes a difference. Fats
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Tannoy, Dynaudio, Blue Sky, JBL, Earthworks, Westlake, NS 10's :D , Genelec, Hafler, KRK, and PMC
Those are good. …………………….. Pick one.
------------------------------------------------------------------------
Originally posted by Doug Milton: I don't want to step into th
Originally posted by Doug Milton:
I don't want to step into the middle of this and incur anyone's wrath, but…..there is actually a question at the end of a bit of rambling.When asked about sample rates by people who know nothing about technology, I draw them an analog sine wave. Think of sample rate as taking with a camera "snapshots" of that sine wave. At 44.1k we really have 44,100 snap shots per second. If we were to zoom in, we would see a stair step effect as there was signal happening between snap shots. Simple logic (as works in my simple mind) would seem to indicate that by doubling or tripling the number of snap shots taken in that same amount of time, would result in smaller "steps", getting us closer to the pure smoothness of our original analog signal. It would seem to me that regardless of output, starting with the highest possible resolution would have to result in a smother (truer to analog, less digitized) sound.
Q: Does that not seem logical regardless of what frequencies I can hear?
I agree, I also don't want to spoil the vibe cause it's a good one. I like it!
The same applies with (okay, I hear them coming to get me lol) images. If I scan two pictures, one at 72 dpi and the same picture at 300 dpi, which one sounds..opps......, looks better?
In the end, I want to put that image on the internet (which is 72 dpi) right? The one at 72 dpi doesn't need to be altered right? Why... because the "internet" is already at 72 dpi, that is the dpi for monitors. (CD's Monitors...what ever) It would technically make sense to avoid all the run around to scan it at 72 dpi and be done with it right? I'm letting a trick slip lol. Now... the image I scanned at 300 dpi would be HUGE if I didn't resize it to 72 dpi and transform/ maybe dither etc. it's original size (am I making sense?) Why? because it would have many dpi added in the "recording of the image" (300 ver 72 dpi).. The 300 dpi image would have more dots on the monitor which would make it larger. In the end...here it comes........The one scanned at 300 dpi, then transformed (dithered maybe) to 72 dpi looks much better than the one that was scanned at 72 dpi that didn't need any altering. Why is that? This same technology applies to audio. Hope that made sense and I'm sticking to it.
:c:
hey, I am really amazed now. A short time ago guys told me not t
hey, I am really amazed now. A short time ago guys told me not to go into the trend of PT HD 96k. It would be better stay with Mix plus @ 44.1k/24 bits and invest on a nice mic pre like the API´s.
Bill, I understood completly your points.. so... shall we all throw away our dinossaur gear that runs under 44.1k/48k?
ACB, I was one of those guys. I still say that ….. IMO 96 is j
ACB,
I was one of those guys. I still say that ….. IMO 96 is just a bus stop. If you have tons of cash it would be the cool thing to do to get up to 96 but in a year or less everyone will be having this same discussion about 96 vs. 192. Knowing your situation, I still recommend you look at better MONITORS :D , mics and preamps This has been more of a discussion on the theory of this rather than a recommendation. If you were starting from scratch at this point I would say go 96K but since you already have the 48K gear, I say “Ride that puppy ‘till it drops!” .... Fats
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Tannoy, Dynaudio, Blue Sky, JBL, Earthworks, Westlake, NS 10's :D , Genelec, Hafler, KRK, and PMC
Those are good. …………………….. Pick one.
------------------------------------------------------------------------
:w:
My Brain hurts... In my experience, even if I get 24/44.1 and
My Brain hurts...
In my experience, even if I get 24/44.1 and i'm staying in the digital domain, I upsample the track to 96k and do all my processing. The filters on the eq's are much higher and this makes a huge difference in the "audible" range. When I then downsample this signal, the high end is much smoother than if I stuck to 44.1. Everything is less pinched sounding. I know there is just about 2 books worth of stuff here so I'm not going to add any more. There are so many things we can spend our money on and so little of it. But I agree that 24/96 is just a stepping stone and soon we'll be jumping on the 192k wagon, after I buy some new amps, oh and a new computer, and add another compressor, and make some new cables, and a new converter, and....
The most disturbing thing in this thread is seeing how rattled s
The most disturbing thing in this thread is seeing how rattled so many people get. Are folks so insecure about their mixes or equipment? You can get great results from what's available now. I'm not mixing music to please the ears of my dog. What bothers me the most is how people simply repeat what's been said before without challenging set opinions. The first time I heard about the high frequency content affecting lower frequency content was from Neve. I love his EQ's and respect his years of experience but he's not God. Just because he said it doesn't mean I'm going to close my ears and eyes and refuse to think. Even if it does affect lower freqs (and I'm inclined to believe it does to a small extent)who's to say that the result is always better. Perhaps on one piece of music it is and on another it ain't. I'm of the belief that you would hear more of a difference by moving a microphone 1/8" one way or the other. Does being anal retentive make one a great mixing engineer? Some would be inclined to say yes but I feel that those that over think things often miss the forest for the trees.
I remember Crane Song sending me their analog dither for evaluation. I put it on a mix, did a blind test and could pick it out from the undithered mix (it had fuller bass response) every single time in a quick A/B test. Of course if I let 30 seconds pass by I couldn't tell the difference. I figured a small difference is a difference nontheless but soon found out on some other mixes it made no difference whatsoever. Luckily I found this out before I turned into a raving lunatic shouting out the magic cure for all of our mixes.
It's getting to the point on this forum that folks are going to be afraid to challenge the moderators. At that point some may go elsewhere because they won't want to study in a place where everyone thinks they have the answers all figured out. Or that a moderator might quit if they disagree with them. Or even worse that they might get booted from the forum for having a different viewpoint. I personally feel that Smawg is too afraid of opposing opinions as if it might bring the forum down to the level where it might actually be interesting and informative to someone. This is not meant as an attack but as construction criticism because I care about what goes on here.
John, Right On!!! IMO the difference lies in if you are attem
John, Right On!!! IMO the difference lies in if you are attempting to reproduce a sonic event or if you are creating one. If you are doing a symphony orchestra, the reproducing a sonic event exactly is your goal. If you’re cutting a rock or pop track then you’re creating one. To reproduce a sonic event the highest fidelity is in order but if you’re doing pop, rock, R&B, country or even rap-hip hop, then IMO simply use the tools you have to the best of your ability. This is a lot like a group of painters all standing around arguing / discussing the merits of different mediums and paints. What's better? Oil or water colors? Charcoal or ink? Canvas or paper? Or even oils on glass. Great work can be done on low bit and sample rates. The first 45 I cut was done off a digital 14 bit PCM beta tape. Leo De Gar Kulka said it was the best sounding bass he had heard in years! Once again it's not what you use but how you use it and most importantly what you're recording in the first place. Talent is the primary concern. Fats
------------------------------------------------------------------------
Tannoy, Dynaudio, Blue Sky, JBL, Earthworks, Westlake, NS 10's :D , Genelec, Hafler, KRK, and PMC
Those are good. …………………….. Pick one.
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I personally feel that Smawg is too afraid of opposing opinions
I personally feel that Smawg is too afraid of opposing opinions as if it might bring the forum down to the level where it might actually be interesting and informative to someone. This is not meant as an attack but as construction criticism because I care about what goes on here.
I'd love to hear MORE constructive criticism. No one is ever too old, educated, or street smart to learn more. But if , as you say, I were afraid of opposing opinions, I'd obviously not be on this site.
Opposing opinions are indeed, as you said, what make this site interesting. However, threads sometimes get downright ridiculous when one side of the discussion is easily demonstrated to be fact, yet the other party simply cannot admit to the fact that they are wrong. This was the reason for my attempt at humor when I posted that I was biting my lip.
I'll also mention that this is definately not an isolated circumstance. It's very common on RO that people get in a discussion, one person is wrong yet the other person is right, but the person who is wrong simply will not say, "Hmmm, you're right! I learned something today". I sometimes must moderate in these circumstances, as they sometimes can get ugly. It's part of human nature to dislike admitting when you're wrong. However, I believe that we're all here both to teach and to learn, and there's an old saying that goes something like, "It's hard to learn anything when you already know it all".
(Dead Link Removed)
When I started that thread, I'd only just started to read a bit about Linux. I wanted to learn more about it, so I started a thread that I was hoping would spark some interest, and get a discussion going. I admittedly was just learning, asked for help, and the guys in the thread helped me out immensely. I now have Mandrake on a peecee and a Mac, and I'm slowly learning more about Linux. The point there is that if you are a willing "student", you're almost certain to find a willing "teacher".
On the other side of the coin, when I see a question asked, if I know the answer, or can shed light on a subject , then I try to help out. I'm not always right then either, nor is anyone else "always right". But when someone points out that I have been wrong, I thank them for showing me how I was wrong. OPUS is great that way ... an excellent teacher.
Then there's the "this versus that" topics, such as Mac versus peecee. There really IS no "correct answer", and the discussions don't ever really come to conclusion, it's just open-ended discussion where a lot of facts are brought-up that participants may never have thought-of or discovered otherwise. Since I own both, I get the benefit from both sides of the discussion!
I hope you can realize that in my position as the site administrator, I must strive to keep the peace, otherwise this site would be a free for all where teaching and learning would become impossible. Most of the real problems that I have to deal with usually involve people who are new to the site, or people who simply come here specifically to stir up trouble (usually from other audio sites). I moderate a thread if people are getting rude, insulting, vulgar, racial, or similar things.
On some occasions, I even get messages from people asking for moderation simply because the "other party" is "dead-wrong" but won't admit it. In those situations, as long as the discussion isn't becoming volatile, I stay out of it, in hopes that the person who is incorrect will eventually understand or concede. However, sometimes in those situations, the incorrect party just cannot understand that they're wrong, or simply refuses to ever admit they're wrong. In those situations the discussion just basically gets dropped.
So, I have a couple of questions for you...
Do you feel that RO is neither interesting nor informative? Do you feel that if we let NASTY feuds (as opposed to intelligent discussion) go on here that RO would be more "interesting"? I really would like to know how you feel about those questions. Perhaps I could indeed learn something.
Bill, > At no time will digital ever be able to give flat fre
Bill,
> At no time will digital ever be able to give flat freqency response ... if steep filters are employed in the audible spectrum. <
Correct me if I'm wrong, but my understanding is that all converters made in the past several years use oversampling, which allows the filters to start operating far past 20 KHz. Is this not the case?
--Ethan
John, > The most disturbing thing in this thread is seeing ho
John,
> The most disturbing thing in this thread is seeing how rattled so many people get ... What bothers me the most is how people simply repeat what's been said before without challenging set opinions. <
100% agreement for sure. I have no attitude with any of this stuff. I want to learn what matters and what doesn't as much as the next person. I still can't see an advantage to capturing frequencies beyond 20 KHz., but that is not necessarily the same as using a higher sample rate which may sound different for other reasons.
> The first time I heard about the high frequency content affecting lower frequency content was from Neve. <
Yes, and I question that too. I'm sure he and the other folks in the room heard something, but I truly doubt it was due to the ear's ability to be influenced by ultrasonic content. As I recall the story, a transformer that was "improperly terminated" was found to be the culprit. But that can cause a lot of audible artifacts, like ringing and distortion - depending on where the transformer is in the signal path and how it interacts with the rest of the circuit.
I am a firm believer in Occam's Razor. Given several possible explanations, the simpler one is usually correct. I think Mr. Neve should have considered other explanations before jumping to the conclusion that supersonic components are audible.
> I'm of the belief that you would hear more of a difference by moving a microphone 1/8" one way or the other. <
No fooling! I am constantly amazed at how people will argue about the importance of minutiae like super high sample rates, jitter artifacts that are 120 dB. below the music, etc., all the while ignoring frequency response variations of 15 dB. or more in their studio and control rooms!
--Ethan
"Do you feel that RO is neither interesting nor informative? Do
"Do you feel that RO is neither interesting nor informative? Do you feel that if we let NASTY feuds (as opposed to intelligent discussion) go on here that RO would be more "interesting"? I really would like to know how you feel about those questions. Perhaps I could indeed learn something."
Ok, honest opinion here. I feel that RO is both interesting and informative. One of my favorite sites. I have yet to see a NASTY feud here that requires moderation. I'm sure that there are some but I haven't seen 'em. I do know that some folks comments have disappeared for reasons unknown to me (such as Fletcher) so that I wasn't able to get an answer to a question I asked him. In any case I appreciate the work you've put into the site and I realise that it can be a thankless job so let me just say thank you.
Fats, > IMO the difference lies in if you are attempting to r
Fats,
> IMO the difference lies in if you are attempting to reproduce a sonic event or if you are creating one. <
Excellent point!
> it's not what you use but how you use it and most importantly what you're recording in the first place. Talent is the primary concern. <
Exactly. Thanks for keeping the important stuff at the forefront.
--Ethan
Thanks Ethan, But....in regards to this... I still can't
Thanks Ethan,
But....in regards to this...
I still can't see an advantage to capturing frequencies beyond 20 KHz., but that is not necessarily the same as using a higher sample rate which may sound different for other reasons.
It is a fact this is not the case. There have been scientific case studies that support the idea that information over 20kHz can be perceived by humans. A group of listeners were hooked up with sensors and charted on EKG while being played analog and digital versions of the same material. There was far more brain activity with the analog signal and listeners reported getting more satisfaction from listening to analog. This is a very old and substantiated study. I am surprised you haven't heard of it. It was conducted by JVC in the early 90's. I know for a fact that my wife hears at way over 20kHz. We used to call her "Dog Ears". :D 20 kHz is not the limit in human hearing. And there is evidence that while we may not hear above a certain point we still perceive it as sound pressure. This was discussed at great length here at RO several months ago. Dig up the thread that I started called "Why Digital Still Sucks" .... Toodles .... Fats
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Tannoy, Dynaudio, Blue Sky, JBL, Earthworks, Westlake, NS 10's :D , Genelec, Hafler, KRK, and PMC
Those are good. …………………….. Pick one.
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OH! Okay John, thank you for the clarifications! I have yet
OH! Okay John, thank you for the clarifications!
I have yet to see a NASTY feud here that requires moderation.
Well, then we've done our job well! Thank you!
I think perhaps a clarification of "moderation" is what's needed. 99% of the moderation on RO is in the form of posts that are made to keep a potentially bad situation from going awry. If a moderator sees a heated debate, he simply watches it to ensure that it remains productive, and to see if the situation could tend to get nasty or abusive. If he sees it start to go that way, then he may post into the thread in a way that will attempt to bring the situation back to productive, healthy discussion. Sometimes this is just in the form of a joke, in order to "lighten-up" the atmosphere. Normally, that is all that is needed. Very seldom does anything more than this need to be done on RO.
So, if you've seen these sort of posts, you have indeed seen moderation. It's usually done BEFORE it gets NASTY.
I do know that some folks comments have disappeared for reasons unknown to me (such as Fletcher) so that I wasn't able to get an answer to a question I asked him.
I assume that by "some folks" that you are referring ONLY to Fletcher's two posts. That was one of the very few rare occasions where posts have actually been removed. The posts were removed due to spamming from his site, done by him and others. His banning was not due to that particular spamming, but rather for many reasons that have culminated over a very long period of time. Many things happen other than on the forums, which are of course unknown to you, but suffice it to say that the time had come where the "parting of ways" had become a necessity. He was asked to refrain from further postings here, and he responded in a gentlemanly fashion.
Again, John, I'm sorry that the timing of these events had to interfere with that thread, and that you were inconvenienced. Be assured that we do everything we can to avoid major disruptions, but sometimes outside forces simply will not allow that to happen. I hope you understand.
Fats, Yes, I am aware of studies like that, but you just know
Fats,
Yes, I am aware of studies like that, but you just know I'll have a comeback anyway! :D
> It is a fact this is not the case. There have been scientific case studies <
One study is never accepted by science as proof of anything. They must be repeated many times, by many different groups of researchers, all with similar results, before being accepted as "fact." But let's suppose that study is correct anyway.
> A group of listeners were hooked up with sensors and charted on EKG while being played analog and digital versions of the same material. There was far more brain activity with the analog signal and listeners reported getting more satisfaction from listening to analog. <
The main problem with this test is it's so subjective. What kind of music was played? Was it music most of them liked, or did most of them not like the music? How old were the people? How much more brain activity? What type of brain activity? More important, what percentage of people reported "more satisfaction" and how much more satisfaction? That is surely too elusive a term to be given much scientific weight. But most important of all, that tests analog versus digital, not whether people can hear or be affected by ultrasonic content!
Do you happen to have a link to the full report?
> I know for a fact that my wife hears at way over 20kHz. <
Okay, 20.5 KHz. maybe. Or even 22 KHz. But surely not anywhere even close to the 50+ KHz. Rupert Neve was talking about.
> there is evidence that while we may not hear above a certain point we still perceive it as sound pressure. This was discussed at great length here at RO <
Maybe, maybe not. Again the only way to determine this with any certainty is in legitimate double-blind tests that are properly designed by folks who actually know how to design such tests. A bunch of people making claims in an audio discussion group doesn't count! Moreover, I'm sure you're aware of placebo effects? And if you are not aware, I'll point out that many audio tests use A/A comparisons, where nothing at all changes from one test to the other. Yet it is very common for people to report a difference anyway.
All of this is good stuff, and needs to be brought out into the open and discussed a lot more. As John Grimm said earlier, "people simply repeat what's been said before without challenging set opinions" which I agree with totally. I've gotten into discussions not unlike this one in other forums, where people argue points they read in magazines. When I ask them how they know it's true, they then admit they don't and were just parroting what they read. I think this is a big problem because I see many of the same "facts" repeated over and over again in the popular press, and so often that many myths are now commonly accepted as true.
Last - sorry! - and perhaps most important of all, how important really is some aspect of audio like high sample rates in the overall scheme of things? Even if it can be proven for certain that some of these things really can be discerned, that does not necessarily mean it's worth spending extra money on.
Call me a Luddite, but I think CDs sound fine. :D When I am unhappy with my recording efforts, I am absolutely certain the real culprit is my own lack of mixing chops, and not the medium I record to.
--Ethan
Originally posted by Michael Fossenkemper: My Brain hurts...
Originally posted by Michael Fossenkemper:
My Brain hurts...In my experience, even if I get 24/44.1 and i'm staying in the digital domain, I upsample the track to 96k and do all my processing. The filters on the eq's are much higher and this makes a huge difference in the "audible" range. When I then downsample this signal, the high end is much smoother than if I stuck to 44.1. Everything is less pinched sounding. I know there is just about 2 books worth of stuff here so I'm not going to add any more. There are so many things we can spend our money on and so little of it. But I agree that 24/96 is just a stepping stone and soon we'll be jumping on the 192k wagon, after I buy some new amps, oh and a new computer, and add another compressor, and make some new cables, and a new converter, and....
well said Michael, and everyone else for your point of view . I think we've covered and boiled this puppy. Lets close this and rest our brains.
Cheers!
:c:
anyone ever study Quantum Physics? more....ah Wave Functions..
anyone ever study Quantum Physics?
more....ah
Wave Functions..
the fact that higher and lower waves in the spectrum(or lack of them) effect the *whole* spectrum has been proven.
theres is really nothing to debate...
its basicly the clash of "old school physics"
with the proven new Quantum discoverys...
that just dont seem to make sence with some people
i admit quantum theory is strange but i do try to embrace it.
to me saying that a 50khz wave makes no difference to us...because we cant hear that particular freq seems rather nieve...
and ultimately very American.
Rowan, > Is there any advantage in recording at 88.2 or 96khz
Rowan,
> Is there any advantage in recording at 88.2 or 96khz and then sample rate converting back to 44.1 at the end for CD? <
No advantage that I can see. Others may disagree, but my feeling is that any perceived improvement from recording at high samples rates - whether you convert later to 44.1 or not - is either imagined or else due to other variables. Like one system has better A/D/A converters, which renders the comparison invalid.
--Ethan