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Sampling Rates
How big of a difference does sampling rate make on a digital recording?
I just got a few DVD-A's, and unfortunately I'm stuck listening to them in DD/DTS, which isn't altogether bad, but doesn't push the DVD-A format to its limit.
Is the sampling rate the biggest advantage DVD-A has over CD audio?
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Quote:
Originally Posted by kexodusc
How big of a difference does sampling rate make on a digital recording?
I just got a few DVD-A's, and unfortunately I'm stuck listening to them in DD/DTS, which isn't altogether bad, but doesn't push the DVD-A format to its limit.
Is the sampling rate the biggest advantage DVD-A has over CD audio?
Sampling rate is way over hyped for the consumer market. CD is already well above most everyones hearing range. Can you hear 20kHz sound? Is there sufficient amount of 20kHz in music to begin with at a high enough level to be audible, well over 100dB threshold minimums?
Don't worry about it. Later, when you need to upgrade the player, get one that is universal.
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Quote:
Originally Posted by mtrycraft
Sampling rate is way over hyped for the consumer market. CD is already well above most everyones hearing range. Can you hear 20kHz sound? Is there sufficient amount of 20kHz in music to begin with at a high enough level to be audible, well over 100dB threshold minimums?
Don't worry about it. Later, when you need to upgrade the player, get one that is universal.
Mtry,
I fully understand you skeptisicm with regards to sampling rates, but it is unfounded. The sample rate defines the Nyquist frequency, or upper limit of the recording or playback of the digital audio. While the upper limit of the human hearing is 20khz(most cannot hear this high) you need a sample rate that goes beyond that frequency limit. With the current redbook sample rate of 44.1khz, the Nyquist frequency would be 22,050khz. In order to enforce that upper limit(and keep noise out of the digital system) steep brickwall filters are employed. These brickwall filters have audible effects such as time smearing that can be heard well within the range of human hearing(most time they they cause strings, cymbals and upper brass to sound harsh) If you were to measure the frequency response of a crash of cymbals, or the overtones of some string and brass instruments, you would find that significant energy can be registered as high as 40khz, so 44.1khz sampling rate is not enough to capture the FULL frequency response without some sort of aliasing gong on. Aliasing raises the noise level within the audible band of frequencies so it is desireable to have the cutoff frequency as high as it can be. This facilitates using filters with a long gradual roll off.
24/96khz (as 24/192khz) raises the Nyquist frequency to 43khz which is more than a octave above CD redbook cutoff frequency(22,050khz) so no brickwall filters are necessary(and no time domain smearing which results in a more linear response of a wide banc of frequencies) and the audible problems they bring.
Using a higher sampling rate insures that EVERY instrument within a orchestra will be recorded with harmonics and timbral textures intact. That is something the current redbook standard had a problem with from day one.
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Quote:
Originally Posted by Sir Terrence the Terrible
Mtry,
I fully understand you skeptisicm with regards to sampling rates, but it is unfounded. The sample rate defines the Nyquist frequency, or upper limit of the recording or playback of the digital audio. While the upper limit of the human hearing is 20khz(most cannot hear this high) you need a sample rate that goes beyond that frequency limit. With the current redbook sample rate of 44.1khz, the Nyquist frequency would be 22,050khz. In order to enforce that upper limit(and keep noise out of the digital system) steep brickwall filters are employed. These brickwall filters have audible effects such as time smearing that can be heard well within the range of human hearing(most time they they cause strings, cymbals and upper brass to sound harsh) If you were to measure the frequency response of a crash of cymbals, or the overtones of some string and brass instruments, you would find that significant energy can be registered as high as 40khz, so 44.1khz sampling rate is not enough to capture the FULL frequency response without some sort of aliasing gong on. Aliasing raises the noise level within the audible band of frequencies so it is desireable to have the cutoff frequency as high as it can be. This facilitates using filters with a long gradual roll off.
24/96khz (as 24/192khz) raises the Nyquist frequency to 43khz which is more than a octave above CD redbook cutoff frequency(22,050khz) so no brickwall filters are necessary(and no time domain smearing which results in a more linear response of a wide banc of frequencies) and the audible problems they bring.
.
No, no no, steep brick wall filter is not used for Red Book 44.1 sampling. That is a misnomer:
http://www.mlssa.com/pdf/Upsampling-theory-rev-2.pdf
http://www.resolutionaudio.com/Up-Oversampling.pdf
http://www.simaudio.com/upsampling.htm
Using a higher sampling rate insures that EVERY instrument within a orchestra will be recorded with harmonics and timbral textures intact. That is something the current redbook standard had a problem with from day one
Don't need it, don't miss it anything that is above 22.05khz. You just cannot hear it. Absolute nonsense that you can or it matters. Anything that is audible gets recorded.
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Quote:
Originally Posted by mtrycraft
Mtry,
Hit yourself on the head, it may clear the fog. We are not talking OVERSAMPLING, that is done at the DAC stage. We are talking about recording at a high sampling rate. BIG difference bud. Secondly, there is not an engineer on this planet that would not agree with me that recording at a 96khz sampling rate sounds noticeable better than at 44.1khz.(and 192khz sounds better than 96khz)
While you cannot hear above 20khz directly, transient information in some instruments is located above 20khz. If you limit the response of a signal or sharp attack at 20khz(or even 22,050khz), the transient information will sound blurred. With a 44.1khz sampling rate, a brickwall filter MUST be used because high frequency information has to be removed at 22,050khz(which is the Nyquist frequency for 44.1khz) or aliasing will occur out of band, and within the band of human hearing. If you are recording audio from 20-20khz, that filter has got to be brickwall because the signal level must drop below between -60 and -90db at 22,050khz so as not to be heard. This filter must do this at 20khz(and be at -60 to -90 at 22,050khz). I would say that is pretty brickwall Mtry. So it is not a misnomer at all as you stated. If no brickwall filter is used, then you would have to lower the cutoff point of the audio well into the range of hearing to eliminate any signals above 22,050khz
http://psbg.emusician.com/ar/emusic_...digital_audio/
http://grassomusic.de/english/frames...sh/digital.htm
Here is a challenge to you. If you don't think the sampling rate is important record the sound of a cymbal crash(or trumpet, glock, chimes) at 96khz sampling rate, listen, then downconvert it to 44.1khz and listen to it again. Since I have done this before I know what you'll hear ;>
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Quote:
Originally Posted by Sir Terrence the Terrible
Mtry,
Hit yourself on the head, it may clear the fog. We are not talking OVERSAMPLING, that is done at the DAC stage. We are talking about recording at a high sampling rate. BIG difference bud. Secondly, there is not an engineer on this planet that would not agree with me that recording at a 96khz sampling rate sounds noticeable better than at 44.1khz.(and 192khz sounds better than 96khz)
While you cannot hear above 20khz directly, transient information in some instruments is located above 20khz. If you limit the response of a signal or sharp attack at 20khz(or even 22,050khz), the transient information will sound blurred. With a 44.1khz sampling rate, a brickwall filter MUST be used because high frequency information has to be removed at 22,050khz(which is the Nyquist frequency for 44.1khz) or aliasing will occur out of band, and within the band of human hearing. If you are recording audio from 20-20khz, that filter has got to be brickwall because the signal level must drop below between -60 and -90db at 22,050khz so as not to be heard. This filter must do this at 20khz(and be at -60 to -90 at 22,050khz). I would say that is pretty brickwall Mtry. So it is not a misnomer at all as you stated. If no brickwall filter is used, then you would have to lower the cutoff point of the audio well into the range of hearing to eliminate any signals above 22,050khz
http://psbg.emusician.com/ar/emusic_...digital_audio/
http://grassomusic.de/english/frames...sh/digital.htm
Here is a challenge to you. If you don't think the sampling rate is important record the sound of a cymbal crash(or trumpet, glock, chimes) at 96khz sampling rate, listen, then downconvert it to 44.1khz and listen to it again. Since I have done this before I know what you'll hear ;>
Thanks for articute the issue so well. The biggest misconception most people have
is that all CD players sound the same.
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On recording, increasing sampling rate above 44.1K will not improve the recording. There is no reason to oversample on recording to capture sound which cannot be heard by human beings. However, on playback, there are a couple of advantages to oversampling. First, by sampling the same signal more times, the number of samples with errors as a percentage of the total number of samples may be reduced. Secondly, by oversampling and generating a signal which is the same as the 44.1K but repeated 4 times or 8 times as though it were recorded at a much higher sampling rate, the analog filter can be pushed out to a much higher frequency where it no longer affects the audible spectrum. By 1989, the best 20 bit players had approached the theoretical ideal and with newer 1 bit oversampled players, most others have come a long way towards matching or slightly bettering them. Fiddling with the analog frequency response to change the tonal balance of cd players slightly is hailed as a major improvement by some audiophiles but in reality, it offers no tangable benefits at all which cannot be achieved with proper use of an equalizer at far lower cost.
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Quote:
Originally Posted by skeptic
On recording, increasing sampling rate above 44.1K will not improve the recording. There is no reason to oversample on recording to capture sound which cannot be heard by human beings. However, on playback, there are a couple of advantages to oversampling. First, by sampling the same signal more times, the number of samples with errors as a percentage of the total number of samples may be reduced. Secondly, by oversampling and generating a signal which is the same as the 44.1K but repeated 4 times or 8 times as though it were recorded at a much higher sampling rate, the analog filter can be pushed out to a much higher frequency where it no longer affects the audible spectrum. By 1989, the best 20 bit players had approached the theoretical ideal and with newer 1 bit oversampled players, most others have come a long way towards matching or slightly bettering them. Fiddling with the analog frequency response to change the tonal balance of cd players slightly is hailed as a major improvement by some audiophiles but in reality, it offers no tangable benefits at all which cannot be achieved with proper use of an equalizer at far lower cost.
Sir,
You are not just in left field on this one, you are in the wrong stadim, in the wrong city, and in the wrong country, and the wrong planet!!!! You don't oversample at the recording stage anyway, you sample. Oversampling is strictly a playback function. The same benefits you mention for oversampling apply to higher sampling, but higher sampling is more predictable, and reliable for best results. Higher sampling allows you to design filters that roll off gradually, as opposed to a brickwall fashion of filters currently in use for 44.1khz CD
First, using a sampling rate above 44.1khz DOES improve recording. Every engineer on this planet is in agreement with that. That is why both DVD-A, and SACD are EXTREMELY popular among audio engineers. Almost every recording done these days is both mixed, mastered, and archived at 24/96khz. Once again, it doesn't matter that you cannot hear it directly, most of the benefits of sampling at 96khz happen right in the audible range of hearing(sharper transients, more air in the mix, and more open sound)
Secondly, an EQ CANNOT do what sampling at a higher rate can do. If you try and do this, not only will you screw up the phase of the high frequencies, but you are like to burn out your tweeter in the process. Not to mention you just be introducing alot more noise to the mix.
I do not know where you got your information, but you need to give it back. It is wrong, and it will surely damage ones speakers if implemented.
DO NOT TRY THIS AT HOME!!!
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Hit yourself on the head, it may clear the fog. We are not talking OVERSAMPLING, that is done at the DAC stage. We are talking about recording at a high sampling rate. BIG difference bud. Secondly, there is not an engineer on this planet that would not agree with me that recording at a 96khz sampling rate sounds noticeable better than at 44.1khz.(and 192khz sounds better than 96khz)
Hogwash. Please site som of the DBT to support his nonsense, thanks.
While you cannot hear above 20khz directly, transient information in some instruments is located above 20khz. If you limit the response of a signal or sharp attack at 20khz(or even 22,050khz), the transient information will sound blurred. With a 44.1khz sampling rate, a brickwall filter MUST be used because high frequency information has to be removed at 22,050khz(which is the Nyquist frequency for 44.1khz) or aliasing will occur out of band, and within the band of human hearing. If you are recording audio from 20-20khz, that filter has got to be brickwall because the signal level must drop below between -60 and -90db at 22,050khz so as not to be heard. This filter must do this at 20khz(and be at -60 to -90 at 22,050khz). I would say that is pretty brickwall Mtry. So it is not a misnomer at all as you stated. If no brickwall filter is used, then you would have to lower the cutoff point of the audio well into the range of hearing to eliminate any signals above 22,050khz
More unalderated hogwash, garbage. Brick wall filtering is not used exactely because of the oversampling.
And Ultrasonic information is not neede. THAT has been demonstrated and published.
Please consult AES on that. Or, if you require, I will site it for you.
You cannot hear it!!!.
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Quote:
Originally Posted by Andy2
Thanks for articute the issue so well. The biggest misconception most people have
is that all CD players sound the same.
Not ALL sound the same, most do in fact under bias controlled listeing. That is the critical issue being biased or not when listening and comparing them.
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Quote:
Originally Posted by Sir Terrence the Terrible
Sir,
You are not just in left field on this one, you are in the wrong stadim, in the wrong city, and in the wrong country, and the wrong planet!!!! You don't oversample at the recording stage anyway, you sample. Oversampling is strictly a playback function. The same benefits you mention for oversampling apply to higher sampling, but higher sampling is more predictable, and reliable for best results. Higher sampling allows you to design filters that roll off gradually, as opposed to a brickwall fashion of filters currently in use for 44.1khz CD
First, using a sampling rate above 44.1khz DOES improve recording. Every engineer on this planet is in agreement with that. That is why both DVD-A, and SACD are EXTREMELY popular among audio engineers. Almost every recording done these days is both mixed, mastered, and archived at 24/96khz. Once again, it doesn't matter that you cannot hear it directly, most of the benefits of sampling at 96khz happen right in the audible range of hearing(sharper transients, more air in the mix, and more open sound)
Secondly, an EQ CANNOT do what sampling at a higher rate can do. If you try and do this, not only will you screw up the phase of the high frequencies, but you are like to burn out your tweeter in the process. Not to mention you just be introducing alot more noise to the mix.
I do not know where you got your information, but you need to give it back. It is wrong, and it will surely damage ones speakers if implemented.
DO NOT TRY THIS AT HOME!!!
Oh, Terrance, brick wall filters haven't been used for a long time, very long time. Oversampling is the reason.
Please, check it out.
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In one regard, you are correct, oversampling means sampling the playback at a higher rate than the recording was made at. This is a small technical point. However, increasing the recording sampling rate to extend the captured analog frequency bandwidth will not improve the regenerated analog signal in the audible spectrum in comparison to a recording made at the lower sampling rate and oversampled at the same higher rate. I don't think there is much point in discussion of frequencies beyond the audible spectrum. Some audiophiles will never concede what scientists and engineers already know and that is that their inclusion adds nothing perceptable in terms of accuracy to humans and at worst can actually detract from accurate reproduction by introducing all kinds of new distorton. It is also pointless to argue what mathematicians, engineers and scientists already know and that is that 44.1Khz sampling on recording is sufficient to accurately capture the entire 20Khz audible analog frequency spectrum that humans can hear (if their hearing isn't imparied.)
"Secondly, an EQ CANNOT do what sampling at a higher rate can do."
I never said it could. What I said was that many so called high end cd players merely fiddle with the analog frequency response after D/A and call the altered sound a "breakthrough." On direct comparison, it will obviously sound different but the difference is easily obtained far more cheaply than the hundreds and even thousands of dollars extra they normally charge for these players. Once the analog signal is perfectly regenerated which is not merely possible but now virtually universal with 1 bit oversampling players, NO FURTHER IMPROVEMENT IN THE PROCESS IS NECESSARY OR POSSIBLE. That is one reason why SACD audio is doomed unless it is just as cheap, totally compatable, and gradually replaces RBCD. There is simply no technical reason for it but electronics companies are always looking for new classes of products to trick audiophiles into thinking they need something better than the fully developed offerings already cheaply available on the market.
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Quote:
Originally Posted by mtrycraft
Hit yourself on the head, it may clear the fog. We are not talking OVERSAMPLING, that is done at the DAC stage. We are talking about recording at a high sampling rate. BIG difference bud. Secondly, there is not an engineer on this planet that would not agree with me that recording at a 96khz sampling rate sounds noticeable better than at 44.1khz.(and 192khz sounds better than 96khz)
Hogwash. Please site som of the DBT to support his nonsense, thanks.
While you cannot hear above 20khz directly, transient information in some instruments is located above 20khz. If you limit the response of a signal or sharp attack at 20khz(or even 22,050khz), the transient information will sound blurred. With a 44.1khz sampling rate, a brickwall filter MUST be used because high frequency information has to be removed at 22,050khz(which is the Nyquist frequency for 44.1khz) or aliasing will occur out of band, and within the band of human hearing. If you are recording audio from 20-20khz, that filter has got to be brickwall because the signal level must drop below between -60 and -90db at 22,050khz so as not to be heard. This filter must do this at 20khz(and be at -60 to -90 at 22,050khz). I would say that is pretty brickwall Mtry. So it is not a misnomer at all as you stated. If no brickwall filter is used, then you would have to lower the cutoff point of the audio well into the range of hearing to eliminate any signals above 22,050khz
More unalderated hogwash, garbage. Brick wall filtering is not used exactely because of the oversampling.
And Ultrasonic information is not neede. THAT has been demonstrated and published.
Please consult AES on that. Or, if you require, I will site it for you.
You cannot hear it!!!.
Mtry,
You know me better than that. I sit in, and participate in many conferences every year with people who know digital audio in and out. You sit behind a computer demanding DBT, and have absolutely no experience in audio. You dismiss all information that does not square with your "totally unbiased" opinion{sarcasm off} I have done this before with you, but have no desire to do this again. You don't know what you are talking about or else you would demand something other than a DBT(that is your usual fall back when you don't have any information to support your arguement) . If a high sampling rate for recording is useless, the oversampling is EXTREMELY useless based on your arguement. So you lack of knowledge makes you arguement inconsistant. Would you like a DBT on that too?? You may be able to intimidate people in the cable forums , and amplifier forums also, but try the DBT crap on someone else, you bore me to tears with it.
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Irregardless of you being bored, CD players have been oversampling for a very long time now and grick wall filters are ancient history, if ever used.
And the need for ultrasonics is just plain silly, regardless where you sit day in and day out. Research has shown otherwise, regardless of the content of instruments above 20kHz as it is well below the threshold of detection, for one.
You would do well to read some of the research on ultrasonics, what effect it has or doesn't in heariong and perception, Boyk not withstanding as he has zero evidence for audibility by anyone. Maybe you can contact J. Stewar of Meridian, or Homlinson closer to home. Or, the ASE paper on audibility of ultrasonics.
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Quote:
Originally Posted by mtrycraft
Irregardless of you being bored, CD players have been oversampling for a very long time now and grick wall filters are ancient history, if ever used.
And the need for ultrasonics is just plain silly, regardless where you sit day in and day out. Research has shown otherwise, regardless of the content of instruments above 20kHz as it is well below the threshold of detection, for one.
You would do well to read some of the research on ultrasonics, what effect it has or doesn't in heariong and perception, Boyk not withstanding as he has zero evidence for audibility by anyone. Maybe you can contact J. Stewar of Meridian, or Homlinson closer to home. Or, the ASE paper on audibility of ultrasonics.
As I have stated earlier, higher sampling is used so NO steep filters are needed. This improves the sound within the frequencies we DO hear. I never said that we could hear above 20khz. The benefit of the 96khz(and higher) allows for four samples for each wave in the upper limit of human hearing, and six to twelve samples for waves in the 8-6KHz frequency range, where most of the music we hear is. That is the benefit, and I stated so in my earlier post.
You are trying to argue when there is no reason. The same reasoning for oversampling applies to higher sampling. Once again(and you are guilty of this so often) you are majoring in minors.
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As I have stated earlier, higher sampling is used so NO steep filters are needed.
It is not needed today in 44.1 due to the oversampling.
This improves the sound within the frequencies we DO hear.
Yes, it would if steep filter was used but it is not used because it is oversampled.
I never said that we could hear above 20khz.
Then I misunderstood your post on this, sorry.
The benefit of the 96khz(and higher) allows for four samples for each wave in the upper limit of human hearing, and six to twelve samples for waves in the 8-6KHz frequency range, where most of the music we hear is. That is the benefit, and I stated so in my earlier post.
But it is not needed. Forier is just fine on this, 2 sample is sufficient. One only needs to see the spectrum at 20khz and below how good it is.
The same reasoning for oversampling applies to higher sampling. Once again(and you are guilty of this so often) you are majoring in minors.
Then why use the higher sample to begin with? Just an audio hype. It is fine for the studios and archiving and mixing, not needed in consumer land. Marketing.
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The proof of the pudding is in the eating. And after more than twenty years, the market has voted with its dollars. I have so many wonderful cds and so many awful vinyls, I can't imaging the world going backwards. I also have 78 RPM shellac records too and players for them but I can't even remember the last time I heard one. Awful sound. (There's even a black and white television set in my basement but I'll bet the last time it was turned on was at least 20 years ago. I wonder why I don't just throw it out.) If cds sound as bad as some audiophiles say they are, they never would have amounted to anything. Maybe it's the music they listen to. They are hearing it for what it really is for the first time and they don't like it. Or maybe it's the rest of that audiophile overhyped junk they listen to it on. Those 8" 2 way little boxes they paid $1500 for that have no bass and shrill treble or those puny class A tube amps that can only put out a few watts. There's the culprit.
I want to thank all you audiophiles out there who sell your unwanted cds to the second hand stores; keep 'em comin'. When I can buy DG, Phillips, London, Sony/Columbia, RCA cds for a few bucks that sound as good as new, it's all I can do to stop myself while there's any money left in my bank account. I'd rather buy great recordings than chase electronic rainbows. BTW, when are you trading in your new equipment. It's already obsolete.
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Skeptic is like a cold shower for me
Everytime I start to get sucked into the vinyl is better than CD mystic, he washes away that illusion.
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Quote:
Originally Posted by mtrycraft
As I have stated earlier, higher sampling is used so NO steep filters are needed.
It is not needed today in 44.1 due to the oversampling.
This improves the sound within the frequencies we DO hear.
Yes, it would if steep filter was used but it is not used because it is oversampled.
I never said that we could hear above 20khz.
Then I misunderstood your post on this, sorry.
The benefit of the 96khz(and higher) allows for four samples for each wave in the upper limit of human hearing, and six to twelve samples for waves in the 8-6KHz frequency range, where most of the music we hear is. That is the benefit, and I stated so in my earlier post.
But it is not needed. Forier is just fine on this, 2 sample is sufficient. One only needs to see the spectrum at 20khz and below how good it is.
The same reasoning for oversampling applies to higher sampling. Once again(and you are guilty of this so often) you are majoring in minors.
Then why use the higher sample to begin with? Just an audio hype. It is fine for the studios and archiving and mixing, not needed in consumer land. Marketing.
So Mtry, if we were to use photography as an example, you are saying a 1megapixal camera is good enough, and a 5 mega pixel camera is not needed? In other words 2 sampled snapshots of the analog waveform is better than 6-12? Bull****, bull****!!!
The more samples of the analog waveform you get, the closer you get to reproducing it transparently. The more samples, the better the audio. Thats an indusputable fact.
Another reason that 96khz and higher sample rate is desireable lies in the white papers presented to AES by Julian Dunn. He sites measurements on the several high end CD players shows the anti alias, and anti image filters do not achieve full rejections of signals above the nyquist frequency. He found VERY non linear behavior in the electronic and electromechanical stages following the signal path of the DAC. This causes the non rejected signals(above the passband) to interact with signal below the passband(what we can hear) and this interaction can be heard by even inexperienced listeners. This is even with oversampling applied(which one can concluded that oversampling at the output stage CAN be of limited benefit if the filters are not well designed) These results are from some VERY expensive CD players, can you imagine what happen in CD players that imploy cheap DAC to meet a price point? Oversampling works great for 44.1khz in theory, but because of limitations in the electronic parts imployed in CD players it has been proven this is not always the most effective solution.
96khz and higher sample rate isn't chosen for its extended high frequency behavior(as I have stated previously). That is just a benefit and allows low pass filters with more gentle slopes. But the higher sampling is used to get MORE samples within the audible frequencies. Yes 2 samples is a MINIMUM point, but more samples leads to higher resolution, cleaner audio, and better imaging.
Recent research suggests that the human brain can discern a difference in a sound's arrival time between the two ears of better than 15 microseconds – around the time between samples at 96 kHz sampling – and some people can even discern a 5µS difference! So while super-high sample rates are probably unnecessary for frequency response, they may be justified for stereo and surround imaging accuracy.
Why 96khz and higher from a recording perspective.
It is clear and widely recognized that most of us can ’t hear much above 18 kHz(there no argument here), but that does not mean that there isn’t anything up there that we need to record – and here's another reason for higher sampling rates. Plenty of acoustic instruments produce usable output up to around the 30 kHz mark(harmonics which make up timbre, and transient attacks) – something that would be picked up in some form by a decent 30 in/s half-inch analog recording. A string section, for example, could well produce some significant ultrasonic energy.(its been measured out to 40khz)
The ultrasonic content of all those instruments blends together to produce audible beat frequencies which contribute to the overall timbre of the sound. If you record your string section at a distance with a stereo pair, for example, all those interactions will have taken place in the air before your microphones ever capture the sound.You can record such a signal with 44.1 kHz sampling and never worry about losing anything –as long as your filters are of good quality(not always the case as previously mentioned) and you have enough bits.
If, however, you recorded a string section with a couple of 48-track digital machines(which is the most common practice), mic on each instrument feeding its own track so that you can mix it all later, your close-mic technique does not pick up any interactions.The only time they can happen is when you mix, by which time the ultrasonic stuff has all been knocked off by your 48 kHz multitrack recorders, so that will never happen. It would thus seem that high sampling rates allow the flexibility of using different mic techniques with better results.
Think of higher sample rates and longer word lengths as a kind of “headroom.” We need higher resolution in the studio than consumers so we can start with a higher level of quality in case some gets lost on the way which might well happen.
And what happens when you modify a digital signal in the digital domain, say by EQing it, or fading it out? You create more bits – more data.You ought to have spare bits so you have room to work.You can always lose resolution, but you can’t easily get it back again.
In the end to say higher sampling rates are just a marketing ploy shows an extreme case of ignorance. Theory only works well if all else is perfect. Nothing is perfect though. Oversampling only works if the low pass, and digital filters operate perfectly. It has been shown they don't. Alot of your assumption and believes are based in a perfect world scenario. We don't live that way, or in that world.
Skeptic, with a 44.1khz sample rate, a analog signal cannot be perfectly regenerated. That is basic digital audio. There are not enough samples for PERFECT regeneration. It would take a sample rate in the neightborhood of 192khz plus for that. Your assertions a just plain false, and I could find nothing to support your arguement. Not in my experience, or the many white papers I have read on digital audio. A bit of skeptisicm is healthy, too much can make you just plain ignorant.
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I want to believe that higher sampling rates produce higher resolotion and better sound.
I shudder to think that Sony, in its cheapness, would spend millions of dollars investing into R&D to deveolp a higher resolution format based on a physical property that didn't improve sound at all. It scares me even more to think that after the failure to improve sound, Sony expects its already critical market to be fooled into thinking they actually do hear a noticeable difference in sound quality.
Such is what skeptic seems to be implying so far. I'm also afraid that if skeptic's view holds true, pure audio perfection has already been accomplished, and a new format cannot improve sound quality over what we have now.
I believe that Princess Di is still alive.
I believe that JFK was killed by the CIA.
I believe that Micheal Jackson is innocent and that OJ will one day find the real killers.
I can't believe that Sony is clever enough to fool millions of people who test audio equipment long before they buy that their newest format is superior to its predecessor when in fact it is not.
And yet I can offer no evidence that I truly do hear a noticeable difference in sound quality other than I do. I attribute this to bit-rate on SACD's, and a combination of bit-rate and word length on DVD-A's...
Quote:
Originally Posted by skeptic
I want to thank all you audiophiles out there who sell your unwanted cds to the second hand stores; keep 'em comin'. When I can buy DG, Phillips, London, Sony/Columbia, RCA cds for a few bucks that sound as good as new, it's all I can do to stop myself while there's any money left in my bank account. I'd rather buy great recordings than chase electronic rainbows. BTW, when are you trading in your new equipment. It's already obsolete.
Now that's just plain funny :)
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No, it's the multichannel capability; that said ...
Quote:
Originally Posted by kexodusc
... Is the sampling rate the biggest advantage DVD-A has over CD audio?
... I'll never really understand the sampling rate issue despite many explainations.
No doubt the theory is sound that you can perfectly record a sound a sample rating rate of twice its frequency, e.g. 20KHz, (the human limit of audibility), can perfectly reproduced a rate of 40KHz.
There are couple problems in the real world:
1) If you're sampling at 40KHz and a 25KHz sound comes along, any attempt to record it will cause errors, that is, distortion, called "aliasing", below the 20KHz, hence audible. I don't understand why this is the case, but you should understand that sampling sounds above 20KHz must be avoided.
2) Since for each sample recorded, there is a given number of bits representing the amplitude of the sound, an actual sound level might be slightly higher or lower than can be represented by the available bits. This need to round causes "quantization error". For some reason I don't clearly understand, this distortion is correlated to the overall sound level, hence is not random, (like clicks & pops on vinyl), and sounds very objectionable. Stuff must be done to conceal quantization error, usually by added ramdon "noise".
Higher sampling rates and more bits per sample can help to remove these problems "at source" so to speak. But it isn't the higher frequencies that can be recorded that matter make higher rez better -- because we can't hear these frequencies.
Another problem that afflicts digital recording is "jitter", that is, the effect of no taking the sound sample, and/or not reproducing the sampled sound, at exactly the right moment in time. It seems small amounts of jitter can sound quite unpleasant. I don't know whether higher sampling rates inherently help solver the jitter problem, but maybe not!!
OK you experts, go to town.
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So Mtry, if we were to use photography as an example, you are saying a 1megapixal camera is good enough, and a 5 mega pixel camera is not needed? In other words 2 sampled snapshots of the analog waveform is better than 6-12? Bull****, bull****!!!
The more samples of the analog waveform you get, the closer you get to reproducing it transparently. The more samples, the better the audio. Thats an indusputable fact.
Photography is not audio and even in photography there is a limit at how many pixels you can differentiate. Simple.
Check out Nyquist and what can be demonstrated with two samples. Rather simple.
Another reason that 96khz and higher sample rate is desireable lies in the white papers presented to AES by Julian Dunn. He sites measurements on the several high end CD players shows the anti alias, and anti image filters do not achieve full rejections of signals above the nyquist frequency. He found VERY non linear behavior in the electronic and electromechanical stages following the signal path of the DAC. This causes the non rejected signals(above the passband) to interact with signal below the passband(what we can hear) and this interaction can be heard by even inexperienced listeners. This is even with oversampling applied(which one can concluded that oversampling at the output stage CAN be of limited benefit if the filters are not well designed) These results are from some VERY expensive CD players, can you imagine what happen in CD players that imploy cheap DAC to meet a price point? Oversampling works great for 44.1khz in theory, but because of limitations in the electronic parts imployed in CD players it has been proven this is not always the most effective solution.
I guess someone needs better recording engineers then who can do it properly.
96khz and higher sample rate isn't chosen for its extended high frequency behavior(as I have stated previously).
Glad you don't think this, my misunderstanding, but many don't think this at all.
That is just a benefit and allows low pass filters with more gentle slopes. But the higher sampling is used to get MORE samples within the audible frequencies. Yes 2 samples is a MINIMUM point, but more samples leads to higher resolution, cleaner audio, and better imaging.
Here is the debate then. I need better convincing evidence of this.
Recent research suggests that the human brain can discern a difference in a sound's arrival time between the two ears of better than 15 microseconds –around the time between samples at 96 kHz sampling – and some people can even discern a 5µS difference! So while super-high sample rates are probably unnecessary for frequency response, they may be justified for stereo and surround imaging accuracy.
OK. But you are confusing timing for localizing with the two ears and how stereo is used on a disc to create soundstage. Totally different concept and events. Sampling has nothing to do with this. Varying the levels and amounts of signal in each speaker does, not sampling rates.
It is clear and widely recognized that most of us can ’t hear much above 18 kHz(there no argument here), but that does not mean that there isn’t anything up there that we need to record – and here's another reason for higher sampling rates. Plenty of acoustic instruments produce usable output up to around the 30 kHz mark(harmonics which make up timbre, and transient attacks) – something that would be picked up in some form by a decent 30 in/s half-inch analog recording. A string section, for example, could well produce some significant ultrasonic energy.(its been measured out to 40khz)
Useable to whom, for what? You can measure until the cows come home. It is useless information just as the masked info that is discarded in perceptual coding. Actually, it is more useless as you will never know its existancce, period. And, it doesn't affect anything in the audible bands. If there are interactions that happens to be audible, then the mic will recordid it and you will hear it with the 44.1 sampling just as well as higher sampling.
That is exactely what the research has shown.
Boyk has measured harmonics to 100kHz. So what. If he had better measuring gear he may have measured it to 200kHz. Meaningless to us. Again, just because it was measured doesn't mean we hear it, it affects anything that we hear.
The ultrasonic content of all those instruments blends together to produce audible beat frequencies which contribute to the overall timbre of the sound.
Fine. IF the instrument produces ultrasonic harmonics that creates audible byproducts which we hear before recording, the audible frequency will be recorded by the recorder and it will be on the CD at 44.1 sampling just as well as it is with 96k sampling.
Now, on the otherhand, if you claim this to happen after the recording takes place, in the electronics, it can only be as an IM byproduct, that is distortion, and not part of the music which needs to be discarded as any distortion.
If you record your string section at a distance with a stereo pair,
Unfortunately, ultrasonic frequency disperses very rapidly, much more so that ones we hear. Hence, your premise is not sound as nothing will happen.
If, however, you recorded a string section with a couple of 48-track digital machines(which is the most common practice), mic on each instrument feeding its own track so that you can mix it all later, your close-mic technique does not pick up any interactions.The only time they can happen is when you mix,
When you mix, you are hoping for Inter modulation to take place? That is distortion. And why would your premis only happen at ultrasonic frequencies? It will happen at all frequencies, down to the lowest recorded frequency. Is that what you want? IM distortion? Hardly. Your premise is false. It doesn't happen. If it happens, it is IM distortion, an undesired byproduct.
Think of higher sample rates and longer word lengths as a kind of “headroom.”We need higher resolution in the studio than consumers so we can start with a higher level of quality in case some gets lost on the way which might well happen.
You need higher quality in the recording so you can master them properly in th edigital domain, all the algorythins, additions and subtractions, averaging, etc will not diminish the final quality, nothing more. The consumer has no need for that in the final product.
And what happens when you modify a digital signal in the digital domain, say by EQing it, or fading it out? You create more bits – more data.You ought to have spare bits so you have room to work.You can always lose resolution, but you can’t easily get it back again.
This is why you need it in recording, so the gear can do all the mathematical applications, rounding off, etc, so you don't end up below what is audible, not because we can detect 96kHz samoling and 20+ bit word length. We just cannot hear it. Finite hearing ability by the end user.
No qualms for using this in th emastering and mixing stage. That is where it is needed, not on playback at home. That is all marketing.
In the end to say higher sampling rates are just a marketing ploy shows an extreme case of ignorance.
Not at all. I didn't say you have no need for that in the studio for mixing, nuimber crunching as that is what happnes. You have no need in the home for playback. You cannot hear it.
Theory only works well if all else is perfect. Nothing is perfect though. Oversampling only works if the low pass, and digital filters operate perfectly. It has been shown they don't. Alot of your assumption and believes are based in a perfect world scenario. We don't live that way, or in that world.
Are you telling me that the 96k sampling is not further oversampled? That 96 is enough? If so, that is only 2X oversampling, a fraction more. CD players have been doing at least 4X and much more for a very long time. That woul de 192k and 384k.
Skeptic, with a 44.1khz sample rate, a analog signal cannot be perfectly regenerated.
Nyquist works. Recording and playback is a bit more complicated to accomlish, hence the oversampling and fanal playback at 44.1.
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Interesting thread science Vs science, maths Vs maths...
On the left Sir Terenese champion of the higher sampling rates. On the right Mtry - champsion of the CD world.
In the middle - the rest of us - basically non the wiser and more confused with each post.
Might I suggest we let our ears make the decisions? Skeptic likes the sound of CD - it works for him and he gains from the CD's others almost throw away at presumably bargain prices.
Others do not like the sound of CD and prefer vinyl. They too gain (usually) from the low price of the media - people like me, who buy classical recordings for a couple of bucks a piece and love them to death.
Others find the sound of SACD or DVDa more palatable. They pay higher prices but hell - there is an industry to support that employs a good number of people.
Basically we are in a win win situation. We have a choice and we exercise it. How we justify that choice is up to each individual - me? I go with my ears - unreliable as they might be, they seem to be consistent in their choices.
Of course there are still others that listen to MP3 and other digital formats - in itself a growing global enterprise that seems to have found away to make it pay. Clever old Apple I say and good luck to them. MP3 may be frowned upon by those who claim its audio quality is not up to acceptable levels, but when I am in the office and have only my computer speakers to listen to, they do nicely, certainly on a par with FM transmissions IMO.
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"Basically we are in a win win situation. We have a choice and we exercise it."
Actually there really isn't much choice. People who want to buy vinyl phonograph records have to hunt them down, usually on the used market and take what they can get. Everything ever released on vinyl and even shellac is finding its way onto cd. What's more, the worlds fixed supply of vinly is not only dwindling but detriorating. In 50 years, vinyl phonograph records will be more of an antique curiousity than a viable alternative to whatever recording method is in vogue. CDs on the other hand will probably always be around because they can be reproduced indefinitely with no deterioration for almost no cost at all and you can buy a player for as little as five dollars. (Twenty years ago they were a thousand to fifteen hundred and those were more expensive dollars. Too bad for vinyl lovers. That can't be much fun. Shopping for them is more like a treasure hunt than building a library of music you want.
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Terrence
Let me invite you to post your propositions about sampling rates, the need for such high rates, the need for ultra high rates to be most accurate for reproduction, and, your premises about the ultrasonics and how they interact in the air to produce harmonics, or intermodulate in the components to make a difference in what we hear.
http://groups.google.com/groups?hl=e...audio.high-end
There are a number of posteres there with a hell of a lot more knowledge in digital audio and acoustics, especially Richard Pierce, who would be most interested to enlighten a poster on these. :)
It really matters not what I say or don't say, if I am right or wrong. I have no pull, nothing in this field.
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Quote:
Originally Posted by Feanor
Everytime I start to get sucked into the vinyl is better than CD mystic, he washes away that illusion.
But, there is nothing wrong with illusions. I enjoy the likes of David Copperfiled but I know that it is an illusion only :)
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Quote:
Originally Posted by kexodusc
I want to believe that higher sampling rates produce higher resolotion and better sound.
I shudder to think that Sony, in its cheapness, would spend millions of dollars investing into R&D to deveolp a higher resolution format based on a physical property that didn't improve sound at all. It scares me even more to think that after the failure to improve sound, Sony expects its already critical market to be fooled into thinking they actually do hear a noticeable difference in sound quality.
Such is what skeptic seems to be implying so far. I'm also afraid that if skeptic's view holds true, pure audio perfection has already been accomplished, and a new format cannot improve sound quality over what we have now.
I believe that Princess Di is still alive.
I believe that JFK was killed by the CIA.
I believe that Micheal Jackson is innocent and that OJ will one day find the real killers.
I can't believe that Sony is clever enough to fool millions of people who test audio equipment long before they buy that their newest format is superior to its predecessor when in fact it is not.
And yet I can offer no evidence that I truly do hear a noticeable difference in sound quality other than I do. I attribute this to bit-rate on SACD's, and a combination of bit-rate and word length on DVD-A's...
Now that's just plain funny :)
There are many thruths in what skeptic has said.
Audiophiles have a tendency to believe, falsly, that the ears ability to hear is limitless. Unfortunately that is not the case. We do know what the limits are. It wasn't nor is it a mystery. For the vast majority of the population and consumer homes capability for reproduction, the ears limits are surpassed, hence nothing new is needed.
However, one area is very difficult to advance as it is the most important and complex. Your speakers capability and your room's acoustics. That is where the emphasis needs to be, not in wires, etc.
As to your other premis about Sony being clever to fool the audiophile is simple. The marketing industry has been conditioning the public for more than a century and human nature is very gullible, far from being skeptical, or the marketing industry woul dbe different or non existant. So, Sony doesn't need to be very clever at all.
Just in case you want to crack open some research into what we can hear, plenty out there. A few can hear 18 bits, period. That is the limit. Not many can hear 18kHz, fewer can hear 20kz and even fewer can hear 25khz but there are a couple who can. Do you build an industry based on a couple of individuals hearing accuity?
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Might I suggest we let our ears make the decisions?
That is what I want you to do, let your ears decide, but only your ears, not your eyes which component has a better name stamped on it, which component has a better marketing hype behind it, which component has 'golden ear' approval. You see, audiophiles, most of them, will never trust only their ears as you suggest, 'trust your ears.'
That is an alien concept to them.
Others do not like the sound of CD and prefer vinyl.
Ah, the key here is 'prefer.' Hard to challenge a preference and hard or impossible to test. Unfortunately many go beyond and exhalt testable parameters, not preferences.
but hell - there is an industry to support that employs a good number of people.
Yes. :)
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So Mtry, if we were to use photography as an example, you are saying a 1megapixal camera is good enough, and a 5 mega pixel camera is not needed? In other words 2 sampled snapshots of the analog waveform is better than 6-12? Bull****, bull****!!!
The more samples of the analog waveform you get, the closer you get to reproducing it transparently. The more samples, the better the audio. Thats an indusputable fact.
"Photography is not audio and even in photography there is a limit at how many pixels you can differentiate. Simple.
Check out Nyquist and what can be demonstrated with two samples. Rather simple."
Not that simple as you believe. If everything was perfect, then you may have a point. However it has already been discovered that all is not perfect. This is simple whether you have the intelligence to understand it or not. More samples means more resolution. That's not disputable. Two samples is a bare minimum, adequate, but not superior. If I were to listen to you, then RBCD is perfect, and we do not need DVD-A or SACD. Much evidence has already said that isn't true. I guess you have been living under a rock. Theory only works when all other things are perfect.
Another reason that 96khz and higher sample rate is desireable lies in the white papers presented to AES by Julian Dunn. He sites measurements on the several high end CD players shows the anti alias, and anti image filters do not achieve full rejections of signals above the nyquist frequency. He found VERY non linear behavior in the electronic and electromechanical stages following the signal path of the DAC. This causes the non rejected signals(above the passband) to interact with signal below the passband(what we can hear) and this interaction can be heard by even inexperienced listeners. This is even with oversampling applied(which one can concluded that oversampling at the output stage CAN be of limited benefit if the filters are not well designed) These results are from some VERY expensive CD players, can you imagine what happen in CD players that imploy cheap DAC to meet a price point? Oversampling works great for 44.1khz in theory, but because of limitations in the electronic parts imployed in CD players it has been proven this is not always the most effective solution.
"I guess someone needs better recording engineers then who can do it properly."
The design of filters is not a recording engineers job. Or didn't you know this?
That is just a benefit and allows low pass filters with more gentle slopes. But the higher sampling is used to get MORE samples within the audible frequencies. Yes 2 samples is a MINIMUM point, but more samples leads to higher resolution, cleaner audio, and better imaging.
"Here is the debate then. I need better convincing evidence of this."
No matter what evidence you get, you'll debate to save your pitful ego.
Recent research suggests that the human brain can discern a difference in a sound's arrival time between the two ears of better than 15 microseconds –around the time between samples at 96 kHz sampling – and some people can even discern a 5µS difference! So while super-high sample rates are probably unnecessary for frequency response, they may be justified for stereo and surround imaging accuracy.
"OK. But you are confusing timing for localizing with the two ears and how stereo is used on a disc to create soundstage. Totally different concept and events. Sampling has nothing to do with this. Varying the levels and amounts of signal in each speaker does, not sampling rates."
Sorry, but all of this works hand in hand. Its a chain mtry. You think only in snapshots, however there is a concept stream here. I think that is why my point elludes you. When you engineer your first recording, I am sure your tune will change when you try and apply all of your internet born theories.
If you get a better snapshot(and more of them) of the varing levels, the localization is more clear. Every engineer understands this, oh wait, you not an engineer. Timing is how you get imaging. If a flute's sound arrives to the microphone before a trumpet, it will be percieved to the ear as closer to us. If a flute's sound arrives to the left microphone in a Blumlein array before the right one, it will be preceived by the ear as coming from the left speaker. When recording, the distance between the performer and the microphone, and where they sit in relation to the microphone array determines the localization, not the level or amount of signal. Once again, basic recording 101
It is clear and widely recognized that most of us can ’t hear much above 18 kHz(there no argument here), but that does not mean that there isn’t anything up there that we need to record – and here's another reason for higher sampling rates. Plenty of acoustic instruments produce usable output up to around the 30 kHz mark(harmonics which make up timbre, and transient attacks) – something that would be picked up in some form by a decent 30 in/s half-inch analog recording. A string section, for example, could well produce some significant ultrasonic energy.(its been measured out to 40khz)
"Useable to whom, for what? You can measure until the cows come home. It is useless information just as the masked info that is discarded in perceptual coding. Actually, it is more useless as you will never know its existancce, period. And, it doesn't affect anything in the audible bands. If there are interactions that happens to be audible, then the mic will recordid it and you will hear it with the 44.1 sampling just as well as higher sampling.
That is exactely what the research has shown."
Are you telling me an instruments harmonics doesn't mean anything? Do you believe that when a violin plays a C note at 256hz, that only that C note at 256hz is heard. Wrong man, you hear harmonics two octaves above that, and more. If you chop off the harmonics, then you change the timbre. This can easily be emulated by listening to a cymbal crash, and then using a eq to cut the upper frequencies off. The timbre will be altered. Even cutting just the 20khz filter on the eq will alter the perceived timbre within the range of hearing. That's simple to understand(even you can understand it)
Since a sample rate of 44.1khz contains no signals above 22,050khz, then none of the harmonics in the upper frequencies of strings and cymbals will be recorded, thus altering its timbre in the audible range. If you don't believe what I say, listen to a recording with strings, cymbals, and high brass with a eq cut at 20khz and see how dull it sounds compared to the master tape. Easy.
Boyk has measured harmonics to 100kHz. So what. If he had better measuring gear he may have measured it to 200kHz. Meaningless to us. Again, just because it was measured doesn't mean we hear it, it affects anything that we hear.
More theory, but no experience and no basic understanding of recording.
The ultrasonic content of all those instruments blends together to produce audible beat frequencies which contribute to the overall timbre of the sound.
Fine. IF the instrument produces ultrasonic harmonics that creates audible byproducts which we hear before recording, the audible frequency will be recorded by the recorder and it will be on the CD at 44.1 sampling just as well as it is with 96k sampling.
Sorry bud, if you use 44.1khz, then the ultrasonic components of the signal are cut off by the low pass filters in DAC process.
"Now, on the otherhand, if you claim this to happen after the recording takes place, in the electronics, it can only be as an IM byproduct, that is distortion, and not part of the music which needs to be discarded as any distortion."
I think my point is pretty clear.
If you record your string section at a distance with a stereo pair,
"Unfortunately, ultrasonic frequency disperses very rapidly, much more so that ones we hear. Hence, your premise is not sound as nothing will happen."
If that was the case, then no measurement of high frequency harmonics is attainable(and your comments about Boyk measurements are a lie). That mean if we step away 20 feet from the instrument we will hear no high frequencies at all. We know that's not the case. Since when know they are, they don't disperse as rapid as you think. Again your lack of experience and knowledge of basic recording techniques leaves you ignorant of the process.
If, however, you recorded a string section with a couple of 48-track digital machines(which is the most common practice), mic on each instrument feeding its own track so that you can mix it all later, your close-mic technique does not pick up any interactions.The only time they can happen is when you mix,
"When you mix, you are hoping for Inter modulation to take place? That is distortion. And why would your premis only happen at ultrasonic frequencies? It will happen at all frequencies, down to the lowest recorded frequency. Is that what you want? IM distortion? Hardly. Your premise is false. It doesn't happen. If it happens, it is IM distortion, an undesired byproduct."
You missed the boat entirely on this one. You can't get IM distortion from close miking techniques. Where did you get that from?
Iteractions only take place in the higher frequencies because their wavelengths are shorter than the path to the microphones.
Think of higher sample rates and longer word lengths as a kind of “headroom.”We need higher resolution in the studio than consumers so we can start with a higher level of quality in case some gets lost on the way which might well happen.
"You need higher quality in the recording so you can master them properly in th edigital domain, all the algorythins, additions and subtractions, averaging, etc will not diminish the final quality, nothing more. The consumer has no need for that in the final product."
What???
And what happens when you modify a digital signal in the digital domain, say by EQing it, or fading it out? You create more bits – more data.You ought to have spare bits so you have room to work.You can always lose resolution, but you can’t easily get it back again.
This is why you need it in recording, so the gear can do all the mathematical applications, rounding off, etc, so you don't end up below what is audible, not because we can detect 96kHz samoling and 20+ bit word length. We just cannot hear it. Finite hearing ability by the end user.
No qualms for using this in th emastering and mixing stage. That is where it is needed, not on playback at home. That is all marketing.
We can hear 120db dynamic range, and we can detect clarity. And that's what 20/96khz would give you. . The point is to have a delivery system that exceeds the hearing, not limit what we can hear.
In the end to say higher sampling rates are just a marketing ploy shows an extreme case of ignorance.
Not at all. I didn't say you have no need for that in the studio for mixing, nuimber crunching as that is what happnes. You have no need in the home for playback. You cannot hear it.
Theory only works well if all else is perfect. Nothing is perfect though. Oversampling only works if the low pass, and digital filters operate perfectly. It has been shown they don't. Alot of your assumption and believes are based in a perfect world scenario. We don't live that way, or in that world.
"Are you telling me that the 96k sampling is not further oversampled? That 96 is enough? If so, that is only 2X oversampling, a fraction more. CD players have been doing at least 4X and much more for a very long time. That woul de 192k and 384k."
No need to oversample when 96khz captures all of the fundemental and harmonics of all instruments. 44.1khz does not capture all of the harmonics, and oversampling at the player level is not perfect, and sometimes not very effective if VERY good filtering isn't used. Julian Dunn has already proven this. Oversampling a 44.1khz sample at the DAC output doesn't reproduce the upper harmonics, that is already lost in recording at 44.1khz
Skeptic, with a 44.1khz sample rate, a analog signal cannot be perfectly regenerated.
"Nyquist works. Recording and playback is a bit more complicated to accomlish, hence the oversampling and fanal playback at 44.1."
Nyquist works, but just like anything it can be improved. If we just stop once we learn about something, then there can be no advance in technology. If we go by your thinking, then cars shouldn't have ever been invented(the horse and buggy can get you to point B right?, jets should not have ever been invented(propellors are good enough), the CD shouldn't have ever been introduce because vinyl is good enough to reproduce the music. Understanding Nyquist theory is just the start, you take that theory and improve on it.
I think that is what you have done with all the information you glean off the net. You learn the bare minimum, and never take it any further than that. This is why so many complex issues escape you.
Perhaps a little read for you may help.
http://www.smr-home-theatre.org/surr.../page_08.shtml
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Close-mic is a bane to good recording practice, regardless
Quote:
Originally Posted by mtrycraft
...
The ultrasonic content of all those instruments blends together to produce audible beat frequencies which contribute to the overall timbre of the sound.
Fine. IF the instrument produces ultrasonic harmonics that creates audible byproducts which we hear before recording, the audible frequency will be recorded by the recorder and it will be on the CD at 44.1 sampling just as well as it is with 96k sampling.
Now, on the otherhand, if you claim this to happen after the recording takes place, in the electronics, it can only be as an IM byproduct, that is distortion, and not part of the music which needs to be discarded as any distortion.
If you record your string section at a distance with a stereo pair,
Unfortunately, ultrasonic frequency disperses very rapidly, much more so that ones we hear. Hence, your premise is not sound as nothing will happen.
If, however, you recorded a string section with a couple of 48-track digital machines(which is the most common practice), mic on each instrument feeding its own track so that you can mix it all later, your close-mic technique does not pick up any interactions.The only time they can happen is when you mix,
When you mix, you are hoping for Inter modulation to take place? That is distortion. And why would your premis only happen at ultrasonic frequencies? It will happen at all frequencies, down to the lowest recorded frequency. Is that what you want? IM distortion? Hardly. Your premise is false. It doesn't happen. If it happens, it is IM distortion, an undesired byproduct.
...
I mean, whether recording is DDD, ADD, AAD, AAA, or whatever.
Why is it that Mercury Living Presence recordings sound so real? Is it because they used 35mm film? No way!! It was because they used only three, omnidirectional mics.
To capture the sound of an actual ensemble in a real auditorium, you need to capture all sounds with a natural balance of direct and reflected sound. This is lost with close-micing and no mix-down engineer can recreat it. Skill engineers can create a plausible "virtual" space, but I would say they can never equal the sound of a good, actual auditorium.
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Quote:
Originally Posted by skeptic
"Basically we are in a win win situation. We have a choice and we exercise it."
Actually there really isn't much choice. People who want to buy vinyl phonograph records have to hunt them down, usually on the used market and take what they can get. Everything ever released on vinyl and even shellac is finding its way onto cd. What's more, the worlds fixed supply of vinly is not only dwindling but detriorating. In 50 years, vinyl phonograph records will be more of an antique curiousity than a viable alternative to whatever recording method is in vogue. CDs on the other hand will probably always be around because they can be reproduced indefinitely with no deterioration for almost no cost at all and you can buy a player for as little as five dollars. (Twenty years ago they were a thousand to fifteen hundred and those were more expensive dollars. Too bad for vinyl lovers. That can't be much fun. Shopping for them is more like a treasure hunt than building a library of music you want.
The nice thing about a treasure hunt is, upon its completion, you have a treasure.
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Um...
[QUOTE=Sir Terrence the Terrible][. If you don't believe what I say, listen to a recording with strings, cymbals, and high brass with a eq cut at 20khz and see how dull it sounds compared to the master tape. Easy.
QUOTE]
Not so easy! :) Most of us don't have access to master tapes or eq that high. Can you explain to a lay person exactly what is occurring here and how cutting frequencies we cannot hear can impact what we do hear? Thanks in advance.
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Quote:
Originally Posted by Feanor
I mean, whether recording is DDD, ADD, AAD, AAA, or whatever.
Why is it that Mercury Living Presence recordings sound so real? Is it because they used 35mm film? No way!! It was because they used only three, omnidirectional mics.
To capture the sound of an actual ensemble in a real auditorium, you need to capture all sounds with a natural balance of direct and reflected sound. This is lost with close-micing and no mix-down engineer can recreat it. Skill engineers can create a plausible "virtual" space, but I would say they can never equal the sound of a good, actual auditorium.
Sometime omnidirectional mikes pic up too much reverberation. Also some engineers spotlight certain instruments that are difficult to hear in the presence of louder instruments. Such as close miking a flute solo passage in the presence of a brass chorus. Different venues, genre's of music, and instrumentation required different miking techniques. That is why there are so many different placement techniques. Different techniques produce different results. On size doesn't fit all.
You don't record a solo piano with an spaced pair of onmidirectional mikes or you will lose all of the percussive transient attacks of the instruments hammers hitting the strings.
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Quote:
Originally Posted by mtrycraft
There are a number of posteres there with a hell of a lot more knowledge in digital audio and acoustics, especially Richard Pierce, who would be most interested to enlighten a poster on these. :)
Or more interesting still, have a debate between Pierce who presumably thinks RBCD is perfect with guys like Steve Hoffman and Jack Fenner (they don't). If you're into classic rock music, then you will recognize Hoffman's name. His favorite recording media is still a 60's vacuum tube based Ampex recorder. For classical fans, you should be well aware of Fenner who has been with Telarc for over twenty five years. I met him briefly back in 1978 when he was in Atlanta to record Stravinsky's Firebird. Telarc has released many a SACD recording.
rw
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I don't doubt the it depends
Quote:
Originally Posted by Sir Terrence the Terrible
Sometime omnidirectional mikes pic up too much reverberation. ...
You don't record a solo piano with an spaced pair of onmidirectional mikes or you will lose all of the percussive transient attacks of the instruments hammers hitting the strings.
For one thing, some venues have too much reverberation for good live listening. In particular it can be difficult to distinguish instrument from instrument or voice from voice. For sure omni mic recording won't work in these places.
From what I've read, the Mercury crews took great pains with the placement of their microphones -- using only three mics didn't make their job easier but the results were worth the extra effort.
As for piano sound, I own MLP's recording of Byron Janis playing Mussorksky's Pictures at a Exhibition, (cat# 434 346-2) I terms of percussive transients this recording is as good as any I have heard. The notes don't give many details but do state that it was made with three microphones -- presumably according to their normal practice.
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Quote:
Originally Posted by E-Stat
Or more interesting still, have a debate between Pierce who presumably thinks RBCD is perfect with guys like Steve Hoffman and Jack Fenner (they don't). If you're into classic rock music, then you will recognize Hoffman's name. His favorite recording media is still a 60's vacuum tube based Ampex recorder. For classical fans, you should be well aware of Fenner who has been with Telarc for over twenty five years. I met him briefly back in 1978 when he was in Atlanta to record Stravinsky's Firebird. Telarc has released many a SACD recording.
rw
Yes, that would be a very interesting debate but embarrassing to your friends, I would predict.
Whiule I do like those old music, I think they are the classic rock by your def, I don't follow who does what or who is behing the mixing boards. Not many names impress me.
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oh yes
Quote:
Originally Posted by E-Stat
For classical fans, you should be well aware of Fenner who has been with Telarc for over twenty five years. I met him briefly back in 1978 when he was in Atlanta to record Stravinsky's Firebird. Telarc has released many a SACD recording.
rw
I know of him and happen to have that recording and a few more from Telarc. I like their CDs just fine.
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The internet is so fun. I often find it interesting that generally speaking most people will make attacks and snyde comments on computers and not face to face.
On the Internet Mrty, Skeptic and Sir Terrence all seem to be experts on digital. Unfortunately do any of you have any technical background. Pretty sure Skeptic is at least an EE and submitted some patent on something that no one else wanted.
I would say the arguements can stand on their own regardless of background but then you have audio experts like JJ on Audio Asylum who hold the same degree level as Martin Colloms who sits seemingly in opposition and is a world wide renouned expert in the field...and then people disagree wholeheartedly with him and his impressive degree.
The relative layperson is royally screwed because it looks like a bunch of people who seem to know something about science and totally dissagree on practically every point about the actual measurement and of course the testing of human subjects...well we of course know that is not the field to be conducted for an Engineer. But besides that leap out of the qualified field I'd at least expect all the EE's to agree 100% 0n all aspects of all things audio...after all is not 100% of everything measured?
It's actually rather frustrating to read because there almost seems like 3 distinct opinions just from you three and then we add the two people I mentioned who have distinct opinions from each other and also from you three. Add in the other million EE and experts and now it's starting to look like the 6 economists in a room with 6 different answers...of course any science relying on statistical evidence is in big fat trouble but that aside it's highly frustrating.
More so when references don't directly discuss EXACTLY what was being discussed to an exact tee. Hell I even read Tom Nousaine's letters to Stereophile and even he writes a bunch of weasal worded commentary on type 1 and 2 errors. It's softpeddled back and forth. Bleggh - circular statistical clap-trap.
There is no one scientific process - and it appears when my instructor said this today he was correct...if there was you three and these others would all have the exact same result.
I suppose the easiest thing for the layperson would be for you three to lay down your degrees for all of us to see. The best and most prestigious wins. After all a claim made by an expert should be able to show his/her expertise. I'm not going to hire a Doctor who does not have proof he is a doctor...so why should anyone here take advice from non audio experts. Not that we will anyway...but many companies can just throw the non-degree people's resumes in the garbage so it would be nice if I could start a similar root out process.
Most people make buying decisions on anecdotal stuff like reviews. SO if we're going to take someone who is spouting fact then that person should provide their expertise.
Damn Damn frustrating.
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Looks like RCA and I are on the same page....
and getting no joy from the "experts".
One other thing:
"CDs on the other hand will probably always be around because they can be reproduced indefinitely with no deterioration for almost no cost at all "
I think that was from Skeptic. MY question is with regard to the no deterioration bit. I have found that of the 450 or so CD's that I own (and about 200 DVD's) several are no longer readable on any of my players. I do not know why - I have tried cleaning them but without joy. A couple are readable in the main but get stuck on certain songs. There is no visible damage to the CD.
Whilst I have already rejected the "perfect sound" hype of CD I am now also leaning towards rejecting the "forever" bit too.
Not a good sign. I have records from the 1950's that play wonderfully and CD's from the 80's that dont. Time will tell I suppose.
One other thing on the treasure hunting bit with regards to vinyl. Yes, it can be a game to find a specific recording you are looking for but it is not quitte as difficult as non-vinyl lovers would have you believe. I live in Athens, Greece. Within half an hours car journey from either my house or my office I have counted 16 shops selling vinyl - 5 within walking distance of my office which is the centre of the city.
Stocks vary, but several have over 10,000 records in at any time, and you, Skeptic, would be amazed at the classical vinyl available.
One store, typically the most distant, has what we might refer to as NOS (New Old Stock) classical - 20,000 of them - all sealed from Philips and DECCA and a couple of others in a lock-up near his shop. Viewing and buying is by appointment only.
It is worth going if only to see 2 shelves, approximately 10 feet long - filled with Philips vinyl from the '70's - all matching covers.
Final item for vinyl lovers - has anyone else noticed how much more is coming out on vinyl these days - hell - even Sony are getting back into it:
http://www.sonymusicstore.com/store/...=00003&alpha=A
Not a great selection but it is a start.
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This topic was discussed on the Analog Room
Quote:
Originally Posted by RGA
The internet is so fun. I often find it interesting that generally speaking most people will make attacks and snyde comments on computers and not face to face.
On the Internet Mrty, Skeptic and Sir Terrence all seem to be experts on digital. Unfortunately do any of you have any technical background. Pretty sure Skeptic is at least an EE and submitted some patent on something that no one else wanted.
I would say the arguements can stand on their own regardless of background but then you have audio experts like JJ on Audio Asylum who hold the same degree level as Martin Colloms who sits seemingly in opposition and is a world wide renouned expert in the field...and then people disagree wholeheartedly with him and his impressive degree.
The relative layperson is royally screwed because it looks like a bunch of people who seem to know something about science and totally dissagree on practically every point about the actual measurement and of course the testing of human subjects...well we of course know that is not the field to be conducted for an Engineer. But besides that leap out of the qualified field I'd at least expect all the EE's to agree 100% 0n all aspects of all things audio...after all is not 100% of everything measured?
It's actually rather frustrating to read because there almost seems like 3 distinct opinions just from you three and then we add the two people I mentioned who have distinct opinions from each other and also from you three. Add in the other million EE and experts and now it's starting to look like the 6 economists in a room with 6 different answers...of course any science relying on statistical evidence is in big fat trouble but that aside it's highly frustrating.
More so when references don't directly discuss EXACTLY what was being discussed to an exact tee. Hell I even read Tom Nousaine's letters to Stereophile and even he writes a bunch of weasal worded commentary on type 1 and 2 errors. It's softpeddled back and forth. Bleggh - circular statistical clap-trap.
There is no one scientific process - and it appears when my instructor said this today he was correct...if there was you three and these others would all have the exact same result.
I suppose the easiest thing for the layperson would be for you three to lay down your degrees for all of us to see. The best and most prestigious wins. After all a claim made by an expert should be able to show his/her expertise. I'm not going to hire a Doctor who does not have proof he is a doctor...so why should anyone here take advice from non audio experts. Not that we will anyway...but many companies can just throw the non-degree people's resumes in the garbage so it would be nice if I could start a similar root out process.
Most people make buying decisions on anecdotal stuff like reviews. SO if we're going to take someone who is spouting fact then that person should provide their expertise.
Damn Damn frustrating.
But no so well written. RGA, as usual, you cut to the heart of the matter with surgical precision. There are dissenting opinions on top of dissenting opinions which leads me to wonder if anybody is "right"... that is, if all of this is mostly opinion based on interpretation of the facts. I think I'm just going to stick with what sounds good to my ears.
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