[...] As far as Nyquest's theorem and how it works, go to WikiPedia and look it up. I don't have time to regurgitate that for you.
But you do know that higher higher res will recreate a signal with more precision. Don't get all caught up with just how high the frequency range is. I'm not necessarily listening to that. Plus you are supposed to get more dynamic range and less noise with high res recordings, etc. But that also depends on your equipment. A high end 16bit DAC might sound better than a cheap 24bit DAC, similarly AD systems as well. But a really good 24/96 should sound a lot better than a really good 16/44.
In WHAT WAY will 24/96 sound better than 16/44? What do you get with 8 more bits? What does doubling the sample rate do?
Until you can answer those questions based purely on what Nyquist, the fundamental rule of digital theory dictates, you can't begin to make any kind of objective assessment. You're human and easily fooled. Think optical illusion, magicians and placebo affect.
I'm an audio engineer. I know how Nyquist works. I know what can and can NOT be accomplished with longer digital words and higher sampling rates. By asking you to explain how it works I was checking if you are able to apply the "mathematical impossibility" filter that prevents experts from being fooled into believing that the Emperor's suit looks AWESOME!
Here is all anyone needs to know on the subject:
Adding bits lowers the noise floor. Nothing more. NOTHING more. It does NOT make sound in the top 80 dB of audibility "smoother" or better. It makes it possible to capture sounds below -115dB or so. If you're recording a classical concert with REALLY wide dynamic range there might be some very slight benefit to 24 bit over 16, but for a typical pop song there will be ZERO difference. None. Nada. Zip. Period. The End.
Increasing the sample rate raises the limit on the highest frequency that can be captured. Period. Nothing more. It does NOT make sounds below 20KHz "smoother" or better. Since a 44.1KHz sample rate captures in excess of the highest frequency humans can hear, there is no benefit to a higher sample rate. None. Zero. Zip. Nada. Anyone who tells you otherwise is either lying to you to get your money or ignorant of how digital audio actually works. It's usually the former.
The beauty of this is that you don't have to take my word for it! Read Nyquist's theorem, the basis upon which digital recording works, and you'll see that there's no room for mystical fairy dust. There is room for making sound better but increasing sample rates and bit depths isn't gonna do squat. (Besides, the difference between any two speakers is a MILLION times bigger than the difference between 16/44 and 24/192!)
In what way do the higher res files sound better than a 16-bit file? What is the audible difference?
Can you please also explain how Nyquist's theorem works? After you do that, you can elaborate on the benefit of any sample rate over ~40KHz?
Here's sequence of videos of what I call, It's more expensive than I can afford, but it's F-ing cool!!! I've heard their speakers in person and it's just an experience you can't emulate, but the videos will show you a little bit of what these systems are about. The guy repeats himself a little and he's a little anal retentive, but just enjoy the videos.
One thing I will tell you to do, if you EVER have the chance to go to one of these high end audiophile shows, GO TO THE MBL, Wilson, Eggleston, Magico, and a few other's rooms. Their systems are so good sounding, you though you died and went to heaven
Wanna have exactly the OPPOSITE experience? Go to a symphony concert in a really nice venue, then rush home and play a recording of the same pieces! Don't do it... ever. Years and years of ear training and controlled listening go completely to crap as you realize that this wonderful playback system that cost more than a good car and that sounds SOOO good is actually only able to reproduce that live event about as well as a Polaroid captures how nice your girlfriend's skin feels. There's no way to describe it, you have to experience it. It's profound.
It was so depressing I got rid of my high-end rig and now listen to an iPod.
Heck, I just downloaded some 24/96 and 24/192 AIFF files from HDTracks and they just kick the living crap out of 16 bit or AAC. It's not even a contest.
In what way do the higher res files sound better than a 16-bit file? What is the audible difference?
Can you please also explain how Nyquist's theorem works? After you do that, you can elaborate on the benefit of any sample rate over ~40KHz?
Glad you bring that up....
16-bit vs. 24-bit: 16-bit is perfectly adequate as a delivery system, provided *everything* was done perfectly during production and mastering, AND the source material was higher resolution and word-length reduced and dithered as the absolutely last step.
Unfortunately, that's a lot of iffs which in reality almost never are true, sticking with 24-bit removes a variety of potential error sources in the digital production work flow, so in any material that's not perfectly produced, mixed and mastered, 24-bit will be sounding better in theory.
As for the sampling rate: 44.1 and certainly 48kHz providing 24kHz signal bandwidth is indeed theoretically sufficient to cover human hearing. The reality of the DA conversion however necessitates anti-aliasing filters. The higher the sampling rate, the lower slope antialiasing filter can be used, and the less phase error is propagating back into the audible range. If there were such a thing as a zero phase error brickwall filter with no pre-ringing and post-ringing effects, etc. then indeed, using sampling frequencies above 48kHz would be an utter waste of bandwidth. A well designed 88/96kHz (2x sampling rates) system should be perfectly adequate, but again, not every audio products company has world class engineers, and many products require shortcuts to stay within budget or power consumption constraints, so using 4x sampling rates makes things a lot more dummy proof in some respects.
So yes, there are practical reasons for these HD audio streams and systems, although most of the mumbo jumbo you hear the "audiophiles" ranting about are about as descriptive of what's really going on as "intelligent design" is about the evolution of live on this planet...
...which is why it takes about the same IQ to finance a televangelists Bentley as it takes to buy pure-oxigen free silver power cables, when from the outlet to the power-plant it's all low-grade copper...
Of course everything you wrote is true, but those are discussions to be held AFTER the very basic, real-world limitations of Nyquist are well understood. Until then this stuff is the fodder semi-informed people use to disseminate nonsense.
I obviously track at 24 bit myself (though not at 96K because the anti-aliasing filters on my converters are so good at 48K that there's no reason to double throughput and storage requirements) but before I would let a protege make that call I'd insist on an explanation of the REASONS for the choice, like what you wrote.
PS. Didn't the oversampling craze of the 1980s promise relief from the issue of ringing from excessively steep anti-aliasing filters?
In WHAT WAY will 24/96 sound better than 16/44? What do you get with 8 more bits? What does doubling the sample rate do?
Until you can answer those questions based purely on what Nyquist, the fundamental rule of digital theory dictates, you can't begin to make any kind of objective assessment. You're human and easily fooled. Think optical illusion, magicians and placebo affect.
I'm an audio engineer. I know how Nyquist works. I know what can and can NOT be accomplished with longer digital words and higher sampling rates. By asking you to explain how it works I was checking if you are able to apply the "mathematical impossibility" filter that prevents experts from being fooled into believing that the Emperor's suit looks AWESOME!
Here is all anyone needs to know on the subject:
Adding bits lowers the noise floor. Nothing more. NOTHING more. It does NOT make sound in the top 80 dB of audibility "smoother" or better. It makes it possible to capture sounds below -115dB or so. If you're recording a classical concert with REALLY wide dynamic range there might be some very slight benefit to 24 bit over 16, but for a typical pop song there will be ZERO difference. None. Nada. Zip. Period. The End.
Increasing the sample rate raises the limit on the highest frequency that can be captured. Period. Nothing more. It does NOT make sounds below 20KHz "smoother" or better. Since a 44.1KHz sample rate captures in excess of the highest frequency humans can hear, there is no benefit to a higher sample rate. None. Zero. Zip. Nada. Anyone who tells you otherwise is either lying to you to get your money or ignorant of how digital audio actually works. It's usually the former.
The beauty of this is that you don't have to take my word for it! Read Nyquist's theorem, the basis upon which digital recording works, and you'll see that there's no room for mystical fairy dust. There is room for making sound better but increasing sample rates and bit depths isn't gonna do squat. (Besides, the difference between any two speakers is a MILLION times bigger than the difference between 16/44 and 24/192!)
Just listen to the different recordings on a decent system that has a good quality DAC, speakers, etc. and see if you hear the difference. IF you can't, then you can't, if you can, then you can.
You are spouting off a LOT OF BULLCRAP. Seriously. Go listen to the same album that's been mastered at different rates on a good system.
Music is complex wave files, but more resolution will yield better dynamic range, lower noise, and better resolution of the audio signals.
The difference between 16/44 to 24/192 using the same system, there is a sonic difference. I can hear it on my system and it's not a big bucks system. On more expensive systems, you just hear the entire bandwidth better. It all boils down to the equipment used, how good your ears are able to detect differences, room acoustics play a role in it. If you room has too many hard surfaces and the music gets bounced around too much, then it's not as noticeable, but THERE IS A DIFFERENCE.
Do you not hear the difference between Dolby Digital and HD Master soundtracks for movies? One is 16 bit, the other 24 bit. If you can't hear the difference, then something is wrong. Maybe your playback system sucks and that might be the reason, or your ears aren't trained to know what to listen for.
Seriously, grow up. There is a sonic difference if the only difference is the bit and sample rates and it's can be heard on a decent system.
Go try it yourself.
What equipment are you using? Have you listened to 16/44 and 24/96 or 24/192 sound files that were from analog recordings?
Just listen to the different recordings on a decent system that has a good quality DAC, speakers, etc. and see if you hear the difference. IF you can't, then you can't, if you can, then you can.
You are spouting off a LOT OF BULLCRAP. Seriously. Go listen to the same album that's been mastered at different rates on a good system.
Music is complex wave files, but more resolution will yield better dynamic range, lower noise, and better resolution of the audio signals.
The difference between 16/44 to 24/192 using the same system, there is a sonic difference. I can hear it on my system and it's not a big bucks system. On more expensive systems, you just hear the entire bandwidth better. It all boils down to the equipment used, how good your ears are able to detect differences, room acoustics play a role in it. If you room has too many hard surfaces and the music gets bounced around too much, then it's not as noticeable, but THERE IS A DIFFERENCE.
Do you not hear the difference between Dolby Digital and HD Master soundtracks for movies? One is 16 bit, the other 24 bit. If you can't hear the difference, then something is wrong. Maybe your playback system sucks and that might be the reason, or your ears aren't trained to know what to listen for.
Seriously, grow up. There is a sonic difference if the only difference is the bit and sample rates and it's can be heard on a decent system.
Go try it yourself.
What equipment are you using? Have you listened to 16/44 and 24/96 or 24/192 sound files that were from analog recordings?
Stop now dude. Seriously. One of these days you're gonna actually READ NYQUIST'S THEOREM and you're gonna realize that the claims you're making here are just plain impossible. OF COURSE some things sound better than some other things. The point is that If you hear a difference between two sources playing normal material at normal listening levels, the difference is NOT word length or sample rate. It just plain CAN'T be. It's not possible. There is ALWAYS another explanation for the difference. You'll see.
Again, you don't have to take my word for it. The math is readily available anywhere.
16-bit vs. 24-bit: 16-bit is perfectly adequate as a delivery system, provided *everything* was done perfectly during production and mastering, AND the source material was higher resolution and word-length reduced and dithered as the absolutely last step.
Unfortunately, that's a lot of iffs which in reality almost never are true, sticking with 24-bit removes a variety of potential error sources in the digital production work flow, so in any material that's not perfectly produced, mixed and mastered, 24-bit will be sounding better in theory.
As for the sampling rate: 44.1 and certainly 48kHz providing 24kHz signal bandwidth is indeed theoretically sufficient to cover human hearing. The reality of the DA conversion however necessitates anti-aliasing filters. The higher the sampling rate, the lower slope antialiasing filter can be used, and the less phase error is propagating back into the audible range. If there were such a thing as a zero phase error brickwall filter with no pre-ringing and post-ringing effects, etc. then indeed, using sampling frequencies above 48kHz would be an utter waste of bandwidth. A well designed 88/96kHz (2x sampling rates) system should be perfectly adequate, but again, not every audio products company has world class engineers, and many products require shortcuts to stay within budget or power consumption constraints, so using 4x sampling rates makes things a lot more dummy proof in some respects.
So yes, there are practical reasons for these HD audio streams and systems, although most of the mumbo jumbo you hear the "audiophiles" ranting about are about as descriptive of what's really going on as "intelligent design" is about the evolution of live on this planet...
...which is why it takes about the same IQ to finance a televangelists Bentley as it takes to buy pure-oxigen free silver power cables, when from the outlet to the power-plant it's all low-grade copper...
Boy, you are talking a bunch of BS. When the power comes into your house, they have things like line filtering, line conditioning, etc. to clean up the mess provided by the power company. Then it goes through cables of various degrees. then it goes through a series of other equipment. EVERY SINGLE PIECE OF EQUIPMENT, including cables has measurable amounts of resistance, inductance, and capacitance, even cables. There are a LOT of things going on to remove distortion whether it's analog distortion or digital distortion (jitter) through out the entire chain. Now, if you listen to music that's highly altered through the use of eq, limiters, compressors, expanders, etc., then you are capturing music somehow and altering it, so you are MESSING up the original content, so you don't really know what the original instrument voice sounds like. That's why MOST pop/commercial music sounds like crap. The audiophile recordings are done typically using two microphones of exceptionally high quality, running through high quality cables, etc. etc. and they DO NOT alter the signal at all throughout the process.
Now, if you want a little education, what do you think one of the most prestigious mastering engineers uses? Bob Ludwig uses a pair of Eggleston speakers that cost around $100K, running from Cello amps that are about another $100K plus, with Transparent speaker and interconnect cables which are another $100K or so and that's what he uses as his high end reference system for classical music and other music that dictates a high quality mastering job. Now, if he gets a call to produce a mastering job that's going to be played on the radio or some cheap speakers, he'll use basically the same level of playback system only he'll change the speaker to NS-10s to give him a reference of a crappy pair of speakers that lack detail, bass, etc. Kind of what most people have.
Now, what do you think Abbey Road Studios use? B&W 800 speakers with Classe Electronics, and their own custom made cables. What do you think Skywalker Sound uses for audio recordings and film soundtracks? B&W 800 series speakers, MIT cables and relatively high end amplifiers. They all have power filtration/conditioning to remove the crap power coming from the power company.
Now, if you want to live in TOTAL ignorance, go right ahead. ANY REALLY GOOD HIGH QUAITY RECORDING STUDIO IS GOING TO USE REALLY EXPENSIVE playback systems as their reference and they have been switching gradually to higher quality, more expensive cables, amplification, and speaker systems. HDTracks are done with Wilson Audio speakers, high end power amps, and cables. These people LIVE AND BREATH audio, they've been doing it for years, have pristine systems and rooms and their reputation depends on their finished product.
The Nyquest theorem does NOT explain everything with digital, it only explains a portion of what's going on. Remember, it's just a theorem.
Do you know what is considered the highest quality AD/DA converters? According to Abbey Road, they prefer DAD, but there are other brands used in high end recording studios and it's not just the DAC chips, but it's the power supplies, input and output stages.
So, again, if you claim there is no difference, what makes you a better expert than the studios and people I've mentioned? What have you done to prove you are correct when there are INDUSTRY experts pumping out recordings that are seen as some of the most influential people will tell you that there is a difference in cables, etc. etc. etc. etc. Please explain YOURSELF and why YOU think you know everything about digital recordings.
Of course everything you wrote is true, but those are discussions to be held AFTER the very basic, real-world limitations of Nyquist are well understood. Until then this stuff is the fodder semi-informed people use to disseminate nonsense.
I obviously track at 24 bit myself (though not at 96K because the anti-aliasing filters on my converters are so good at 48K that there's no reason to double throughput and storage requirements) but before I would let a protege make that call I'd insist on an explanation of the REASONS for the choice, like what you wrote.
PS. Didn't the oversampling craze of the 1980s promise relief from the issue of ringing from excessively steep anti-aliasing filters?
It depends on what one thinks is GOOD ENOUGH. Remember, some people have lived and breathed around high end equipment for years, know the different in the tonal quality of one concert grand piano to another, can tell the different in the tonal quality of high end microphones and have basically really well trained ears. Did you know that there were tests done a long time ago and the average person can't tell between a perfect note and as little as 5 cent difference in pitch where as trained musicians could decern as little as 2 cent difference in pitch. That's just being able to tell pitch. A concert grand pianist can tell the difference between a Boesendorfer and a Steinway, etc. whereas the average person can't.
If you aren't doing reference quality recordings in a pristine environment, then you might not be able to hear these sonic differences, but those that have lived around these higher end systems are acclimated to them and they can hear the differences.
I know there are blind fold tests that have been done, but the tests that were performed could have been set up in a manner to give that result, since they might be using switch boxes and other cables that will prevent any differences heard between one set of cables and another. Obviously, it's all comes down to the equipment, room, content, and listener. If you can hear it, then you can, if you can't then you can't. I take a I have to hear it for myself approach, but I will listen to higher end systems to listen to differences and I decide for myself what I like and don't like and price is only decided upon on what I buy and afford. But some of the best sounding systems were VERY expensive and actually not the most expensive in the world, but then again, I haven't heard the most expensive systems to compare them to, i just know what I liked.
Oversampling can be done poorly or well. Meridian, for example, has created their own unique apodising filter to remove pre-ringing, others are doing the same thing and they achieve better sound than just oversampling by itself. So, it really depends on what and how oversampling is done. There are so many ways to get a decent digital sound, so all I can say is that on my system, I can tell the difference between a 16 bit file that gets oversampled and runs through a apodising filter to remove pre-ringing compared to a 24/96 version on the exact same equipment. So, I don't know what else to tell you, but my USB DAC certainly sounds a LOT better than the internal one and there is no comparison.
It depends on what one thinks is GOOD ENOUGH. Remember, some people have lived and breathed around high end equipment for years, know the different in the tonal quality of one concert grand piano to another, can tell the different in the tonal quality of high end microphones and have basically really well trained ears. Did you know that there were tests done a long time ago and the average person can't tell between a perfect note and as little as 5 cent difference in pitch where as trained musicians could decern as little as 2 cent difference in pitch. That's just being able to tell pitch. A concert grand pianist can tell the difference between a Boesendorfer and a Steinway, etc. whereas the average person can't.
If you aren't doing reference quality recordings in a pristine environment, then you might not be able to hear these sonic differences, but those that have lived around these higher end systems are acclimated to them and they can hear the differences.
I know there are blind fold tests that have been done, but the tests that were performed could have been set up in a manner to give that result, since they might be using switch boxes and other cables that will prevent any differences heard between one set of cables and another. Obviously, it's all comes down to the equipment, room, content, and listener. If you can hear it, then you can, if you can't then you can't. I take a I have to hear it for myself approach, but I will listen to higher end systems to listen to differences and I decide for myself what I like and don't like and price is only decided upon on what I buy and afford. But some of the best sounding systems were VERY expensive and actually not the most expensive in the world, but then again, I haven't heard the most expensive systems to compare them to, i just know what I liked.
Oversampling can be done poorly or well. Meridian, for example, has created their own unique apodising filter to remove pre-ringing, others are doing the same thing and they achieve better sound than just oversampling by itself. So, it really depends on what and how oversampling is done. There are so many ways to get a decent digital sound, so all I can say is that on my system, I can tell the difference between a 16 bit file that gets oversampled and runs through a apodising filter to remove pre-ringing compared to a 24/96 version on the exact same equipment. So, I don't know what else to tell you, but my USB DAC certainly sounds a LOT better than the internal one and there is no comparison.
Quote:
Originally Posted by v5v
Stop now dude. Seriously. One of these days you're gonna actually READ NYQUIST'S THEOREM and you're gonna realize that the claims you're making here are just plain impossible. OF COURSE some things sound better than some other things. The point is that If you hear a difference between two sources playing normal material at normal listening levels, the difference is NOT word length or sample rate. It just plain CAN'T be. It's not possible. There is ALWAYS another explanation for the difference. You'll see.
Again, you don't have to take my word for it. The math is readily available anywhere.
Good luck with it. I'm out.
I've read Nyquist's THEOREM years ago, but it does NOT explain everything. It's just showing you a small aspect of what digital is all about.
Do you not think that there's a lot more to it than that? What about measurable amounts of jitter? Does Nyquist explain that? NO. It doesn't. Nyquist is a THEOREM, what's a theorem? THEORY. And it's only using math to explain one aspect of a digital signal, but they know more about digital signals since Nyquist's THEORY and that's why companies are pushing the envelope, just like they are making speakers better than they ever have, power amps, pre amps, cables, etc. Things become better by the use of better, more precise measurement tools, different types of measurements, different materials, etc. etc. Sounds like you just hide behind some BS theory to back your BS excuse because you can't hear the difference because of the lack of quality equipment and experience you have.
IF they can measure something, they can also hear it under the right circumstances, which is what the high end guys are doing and that gets filtered down to the masses at some level. But the average person doesn't know what a real instrument sounds like unless they play an instrument.
Seriously, you sound like you don't have the money for better equipment and are afraid to list what you are using. Tell me what playback equipment you are using. If you are using cheap earbuds from an mobile device, yeah, you probably can't hear much difference because the DAC is a cheap DAC and the earbuds are POS. But take that same file and put it on a extremely nice system in a good room and people can hear a difference if they know what to listen for and have trained ears that haven't polluted by loud distorted, overly processed crap.
16-bit vs. 24-bit: 16-bit is perfectly adequate as a delivery system, provided *everything* was done perfectly during production and mastering, AND the source material was higher resolution and word-length reduced and dithered as the absolutely last step.
Unfortunately, that's a lot of iffs which in reality almost never are true, sticking with 24-bit removes a variety of potential error sources in the digital production work flow, so in any material that's not perfectly produced, mixed and mastered, 24-bit will be sounding better in theory.
As for the sampling rate: 44.1 and certainly 48kHz providing 24kHz signal bandwidth is indeed theoretically sufficient to cover human hearing. The reality of the DA conversion however necessitates anti-aliasing filters. The higher the sampling rate, the lower slope antialiasing filter can be used, and the less phase error is propagating back into the audible range. If there were such a thing as a zero phase error brickwall filter with no pre-ringing and post-ringing effects, etc. then indeed, using sampling frequencies above 48kHz would be an utter waste of bandwidth. A well designed 88/96kHz (2x sampling rates) system should be perfectly adequate, but again, not every audio products company has world class engineers, and many products require shortcuts to stay within budget or power consumption constraints, so using 4x sampling rates makes things a lot more dummy proof in some respects.
So yes, there are practical reasons for these HD audio streams and systems, although most of the mumbo jumbo you hear the "audiophiles" ranting about are about as descriptive of what's really going on as "intelligent design" is about the evolution of live on this planet...
...which is why it takes about the same IQ to finance a televangelists Bentley as it takes to buy pure-oxigen free silver power cables, when from the outlet to the power-plant it's all low-grade copper...
Boy, you are talking a bunch of BS. When the power comes into your house, they have things like line filtering, line conditioning, etc. to clean up the mess provided by the power company. Then it goes through cables of various degrees. then it goes through a series of other equipment. EVERY SINGLE PIECE OF EQUIPMENT, including cables has measurable amounts of resistance, inductance, and capacitance, even cables. There are a LOT of things going on to remove distortion whether it's analog distortion or digital distortion (jitter) through out the entire chain.
[...blah...blah...]
None of which has anything to do with silver power cables, which are utterly pointless, because any decent, and even more so any high-end equipment has an internal power supply that filters, regulates, stabilizes and transforms the power from AC to DC at levels useful for the electronics inside.
Quote:
Now, if you want a little education, what do you think one of the most prestigious mastering engineers uses? Bob Ludwig uses a pair of Eggleston speakers that cost around $100K, running from Cello amps that are about another $100K plus, with Transparent speaker and interconnect cables which are another $100K or so and that's what he uses as his high end reference system for classical music and other music that dictates a high quality mastering job. Now, if he gets a call to produce a mastering job that's going to be played on the radio or some cheap speakers, he'll use basically the same level of playback system only he'll change the speaker to NS-10s to give him a reference of a crappy pair of speakers that lack detail, bass, etc. Kind of what most people have.
Now, what do you think Abbey Road Studios use? B&W 800 speakers with Classe Electronics, and their own custom made cables. What do you think Skywalker Sound uses for audio recordings and film soundtracks? B&W 800 series speakers, MIT cables and relatively high end amplifiers. They all have power filtration/conditioning to remove the crap power coming from the power company.
Now, if you want to live in TOTAL ignorance, go right ahead. ANY REALLY GOOD HIGH QUAITY RECORDING STUDIO IS GOING TO USE REALLY EXPENSIVE playback systems as their reference and they have been switching gradually to higher quality, more expensive cables, amplification, and speaker systems. HDTracks are done with Wilson Audio speakers, high end power amps, and cables. These people LIVE AND BREATH audio, they've been doing it for years, have pristine systems and rooms and their reputation depends on their finished product.
The Nyquest theorem does NOT explain everything with digital, it only explains a portion of what's going on. Remember, it's just a theorem.
Do you know what is considered the highest quality AD/DA converters? According to Abbey Road, they prefer DAD, but there are other brands used in high end recording studios and it's not just the DAC chips, but it's the power supplies, input and output stages.
So, again, if you claim there is no difference, what makes you a better expert than the studios and people I've mentioned? What have you done to prove you are correct when there are INDUSTRY experts pumping out recordings that are seen as some of the most influential people will tell you that there is a difference in cables, etc. etc. etc. etc. Please explain YOURSELF and why YOU think you know everything about digital recordings.
For one, most of that expensive equipment you quote has nothing to do with the sampling frequencies we were discussing, because like you what you don't seem to know, speakers and amps are analog equipment.
Even in equipment that is digital and where sampling rates and word lengths do matter, there are no mono-causal relationships between equipment quality and these two parameters. Well engineered ADC/DACs, like e.g. my Metric Halo ULN-8, cost more, because they are done right, and so someone might use that equipment even if they work at 48kHz sampling rates, because everything else is done better. There are also issues internal to the various DAC/ADC chips, such as internal signal paths, which sampling clock is derived from which other, such that e.g. 192kHz may have lower jitter than 48kHz on the same equipment, resulting in 192 downsampled to 48kHz sounding better than straight 48kHz recording, even though the theoretical information content is the same, simply because the clock was better, etc.
All that said: nobody has invalidated Nyquist yet, and if someone did, it be about as huge as the discovery of quantum physics or something on that scale.
Many of the "golden ears" are not engineers and they talk mumbo-jumbo that could come straight from an astrologer or shaman trying to explain which cable has "better spirits". The fact is, that just because you can tell something sounds better doesn't mean you know why it sounds better, nor do you necessarily know if that better sound is euphonic or closer to reality, all of which means that many so-called authorities talk so much crap that physicist and engineers start running out of the room screaming when these people start talking.
Further, there are endorsement deals and the whole politics of the high-end audio world, musician clients that need to be impressed so they shut up and don't make stupid requests, etc. I knew of mastering engineers that bough boat loads of broken vintage audio gear to fill up 19" racks with moving needles and blinking LEDs, because that show would shut up know-it-all-better musicians, and in fact, 98% of what they did was done in software in a computer. They even had some A/B switches to let artists decide which version they liked better, when in fact the A/B switch did nothing except give the musician the cozy feeling in the belly that they were "actively involved" in the mastering process.
Oh yeah, and last but not least, which proves that you either didn't read my post or have language understanding issues, I was actually making the case FOR higher sampling rates and word lengths, but that seemed to have escaped you.
None of which has anything to do with silver power cables, which are utterly pointless, because any decent, and even more so any high-end equipment has an internal power supply that filters, regulates, stabilizes and transforms the power from AC to DC at levels useful for the electronics inside.
For one, most of that expensive equipment you quote has nothing to do with the sampling frequencies we were discussing, because like you what you don't seem to know, speakers and amps are analog equipment.
Even in equipment that is digital and where sampling rates and word lengths do matter, there are no mono-causal relationships between equipment quality and these two parameters. Well engineered ADC/DACs, like e.g. my Metric Halo ULN-8, cost more, because they are done right, and so someone might use that equipment even if they work at 48kHz sampling rates, because everything else is done better. There are also issues internal to the various DAC/ADC chips, such as internal signal paths, which sampling clock is derived from which other, such that e.g. 192kHz may have lower jitter than 48kHz on the same equipment, resulting in 192 downsampled to 48kHz sounding better than straight 48kHz recording, even though the theoretical information content is the same, simply because the clock was better, etc.
All that said: nobody has invalidated Nyquist yet, and if someone did, it be about as huge as the discovery of quantum physics or something on that scale.
Many of the "golden ears" are not engineers and they talk mumbo-jumbo that could come straight from an astrologer or shaman trying to explain which cable has "better spirits". The fact is, that just because you can tell something sounds better doesn't mean you know why it sounds better, nor do you necessarily know if that better sound is euphonic or closer to reality, all of which means that many so-called authorities talk so much crap that physicist and engineers start running out of the room screaming when these people start talking.
Further, there are endorsement deals and the whole politics of the high-end audio world, musician clients that need to be impressed so they shut up and don't make stupid requests, etc. I knew of mastering engineers that bough boat loads of broken vintage audio gear to fill up 19" racks with moving needles and blinking LEDs, because that show would shut up know-it-all-better musicians, and in fact, 98% of what they did was done in software in a computer. They even had some A/B switches to let artists decide which version they liked better, when in fact the A/B switch did nothing except give the musician the cozy feeling in the belly that they were "actively involved" in the mastering process.
Oh yeah, and last but not least, which proves that you either didn't read my post or have language understanding issues, I was actually making the case FOR higher sampling rates and word lengths, but that seemed to have escaped you.
All they are doing is explaining what they hear, so they use terms like soundstage, depth, harshness, detail, etc. etc. That's the only way they can explain what they HEAR.
Now, by your BS, all DAC should sound identical when at 16/44, 16/48 or whatever level they are at, but unfortunately they don't all sound the same.
Um, just an FYI, Skywalker Sound did NOT get free equipment or money for their endorsement for MIT Cables, Bob Ludwig did NOT get free equipment or get paid to give his endorsement for Transparent Cables. Yeah, the companies like Yamaha, and others companies might resort to paying endorsers, giving free gear, but a lot of these high end companies don't. They can't afford to. Some of them might get prototypes to help further development and are used to evaluate and give their feedback on new cable, or equipment designs, but a lot of them do not get paid or get free equipment. How do I know? I know people that work for those companies and asked them. Now, the heavily marketed companies might resort to those tactics because they have to, but a lot of these smaller high end companies DON'T do that.
Many of these reviewers for some magazines are musicians, have an engineering background as a recording engineer, or sometimes an electronics engineering background, so it's a case by case basic as to their background, but they get involved with listening and evaluating speakers, cables, amps, preamps, etc. in nice sounding rooms and they give their feedback on what they like and don't like, but I take it all with a grain of salt and do my own listening tests, but I use their guidance to help me identify what to listen for and I see for myself if i can or can't hear a difference.
For grins, I bought a AC noise sniffer, hooked it up to an AC outlet with a clock radio connected to the other outlet and the clock radio was turned off. The noise sniffer has a speaker that I can actually hear the line noise and I actually heard a radio station coming through fairly clearly on the sniffer. It's one of those Entech noise sniffers that are available and they are used by people to measure and hear line noise in AC. There is also a digital readout on the amount of noise as well. Interesting little box and I didn't pay all that much for it, there are more expensive one out there, but I bought this as other respected companies that make line filtration and conditioning systems use the same device.
None of which has anything to do with silver power cables, which are utterly pointless, because any decent, and even more so any high-end equipment has an internal power supply that filters, regulates, stabilizes and transforms the power from AC to DC at levels useful for the electronics inside.
For one, most of that expensive equipment you quote has nothing to do with the sampling frequencies we were discussing, because like you what you don't seem to know, speakers and amps are analog equipment.
Even in equipment that is digital and where sampling rates and word lengths do matter, there are no mono-causal relationships between equipment quality and these two parameters. Well engineered ADC/DACs, like e.g. my Metric Halo ULN-8, cost more, because they are done right, and so someone might use that equipment even if they work at 48kHz sampling rates, because everything else is done better. There are also issues internal to the various DAC/ADC chips, such as internal signal paths, which sampling clock is derived from which other, such that e.g. 192kHz may have lower jitter than 48kHz on the same equipment, resulting in 192 downsampled to 48kHz sounding better than straight 48kHz recording, even though the theoretical information content is the same, simply because the clock was better, etc.
All that said: nobody has invalidated Nyquist yet, and if someone did, it be about as huge as the discovery of quantum physics or something on that scale.
Many of the "golden ears" are not engineers and they talk mumbo-jumbo that could come straight from an astrologer or shaman trying to explain which cable has "better spirits". The fact is, that just because you can tell something sounds better doesn't mean you know why it sounds better, nor do you necessarily know if that better sound is euphonic or closer to reality, all of which means that many so-called authorities talk so much crap that physicist and engineers start running out of the room screaming when these people start talking.
Further, there are endorsement deals and the whole politics of the high-end audio world, musician clients that need to be impressed so they shut up and don't make stupid requests, etc. I knew of mastering engineers that bough boat loads of broken vintage audio gear to fill up 19" racks with moving needles and blinking LEDs, because that show would shut up know-it-all-better musicians, and in fact, 98% of what they did was done in software in a computer. They even had some A/B switches to let artists decide which version they liked better, when in fact the A/B switch did nothing except give the musician the cozy feeling in the belly that they were "actively involved" in the mastering process.
Oh yeah, and last but not least, which proves that you either didn't read my post or have language understanding issues, I was actually making the case FOR higher sampling rates and word lengths, but that seemed to have escaped you.
Um, by your way of thinking, every 16/44 DAC should sound identical regardless of mfg and model. That's what YOUR theory suggests, but reality is that they don't all sound equal. They also don't go through the battery of measurement tests equally either. so, how can you back your statement up when these different DACs measure differently, even though they are all doing 16/44 or 16/58 or 24/96, etc. etc. and how come leading well respected people in the mastering and audio recording industry that record live classical or jazz, etc. music in a really good acoustic environment using high quality microphones trying to capture the sound of musicians and being able to provide the most accurate recordings that the equipment they use are capable of, and then they listen to their recordings back and are trying to compare the listening experience to the actual performance, which is hard to do, but they are doing it the best they can? You theory doesn't explain any of this, but this is what is HAPPENING. You can talk sample rates, bit rates until you are blue in the face, but you haven't mentioned jitter, you haven't mentioned any of the other test measurements that these companies do when developing a product to decide what components, and design to use. Do you know the different between the different mfg DAC chips on the market?
Seriously, grow up. You seem to have some bitterness towards people that appreciate, listen to , and can afford more expensive equipment than you can. How come the brands and model AD converters are changing in these high end studios and the quality of the recordings have changed since the introduction of digital?
For grins, I bought a AC noise sniffer, hooked it up to an AC outlet with a clock radio connected to the other outlet and the clock radio was turned off. The noise sniffer has a speaker that I can actually hear the line noise and I actually heard a radio station coming through fairly clearly on the sniffer. It's one of those Entech noise sniffers that are available and they are used by people to measure and hear line noise in AC.
You have no idea how much grin I'm getting from this.
Quiz for the blank one: What's the VERY FIRST THING that AC power hits once it goes inside the box? What are the effects of that device?
Seriously dude, I'm not trying to beat you up here, but you just keep doing it to yourself! Your example above is EXACTLY what I'm talking about -- those with little knowledge of basic physics are very easily led to believe utter bullshit.
For an explanation of why your sniffer is the most pointless device ever built, look into my question, above, or just ask your local TV repairman.
Um, by your way of thinking, every 16/44 DAC should sound identical regardless of mfg and model.
NO! Jeez, this is like trying to teach a kid to shoot when he won't take the gun out of his mouth...
Obviously different circuits sound different. If you look back through the thread, no one here has suggested otherwise. You're arguing a point no one made.
You sound like a kid with a can of NO2 telling the engineers at McLaren that the laws of physics are just a guideline and not really very well understood. They are actually VERY well understood, just not by you.
No dude, wrong. WAY wrong. So wrong you should be disqualified from practicing audio until you get that misconception fixed.
Nyquist DOES explain absolutely every single aspect of digital recording. It is more absolute than e=mc2. There is no ambiguity.
NO it doesn't. It's just a THEORY. What about clocking? what about losing LSB? What about jitter? What about s/n, distortion, etc.? Nyquist does NOT explain everything, it's a THEORY.
By your way of thinking, all DA and AD converters sound the same, but they don't.
Go talk to Abbey Road Studios and ask them why they switched from their old AD/DA converters to DADs? Same thing with EMI Studios and other studios that are buying the big bucks AD/DA converters? There are the output and input stages, clocking, etc. etc.
The Nyquist Theorem is just PART OF IT.
Go talk to converter companies that make the high end stuff and then maybe you'll get an education.
You have no idea how much grin I'm getting from this.
Quiz for the blank one: What's the VERY FIRST THING that AC power hits once it goes inside the box? What are the effects of that device?
Seriously dude, I'm not trying to beat you up here, but you just keep doing it to yourself! Your example above is EXACTLY what I'm talking about -- those with little knowledge of basic physics are very easily led to believe utter bullshit.
For an explanation of why your sniffer is the most pointless device ever built, look into my question, above, or just ask your local TV repairman.
OK, then how come I can measure the noise in the AC line and then get a certain power conditioner/filter that cleans up that noise? Why do you think high end recording studios, data centers spend LOTS of money on power conditioning and filtration systems? For the fun of it?
Grow up. We aren't getting clean properly regulated power from the power companies.
I have a lot of noise that's visible on the TV, so how do you explain that? It's not a perfect picture.
I would rather talk to engineers that specialize in designing power conditioning/filtration systems than some IDIOT on AI.
OK, then how come I can measure the noise in the AC line and then get a certain power conditioner/filter that cleans up that noise? Why do you think high end recording studios, data centers spend LOTS of money on power conditioning and filtration systems? For the fun of it?
Grow up. We aren't getting clean properly regulated power from the power companies.
I have a lot of noise that's visible on the TV, so how do you explain that? It's not a perfect picture.
I would rather talk to engineers that specialize in designing power conditioning/filtration systems than some IDIOT on AI.
A local TV repair man? Hahahahahahahaa. Most of those guys have basic knowledge and basic tools for which they are using.
NO! Jeez, this is like trying to teach a kid to shoot when he won't take the gun out of his mouth...
Obviously different circuits sound different. If you look back through the thread, no one here has suggested otherwise. You're arguing a point no one made.
You sound like a kid with a can of NO2 telling the engineers at McLaren that the laws of physics are just a guideline and not really very well understood. They are actually VERY well understood, just not by you.
You are now becoming a worthless pile of garbage. I've talked to engineers that design power conditioning systems, they will put you in your place in 2 seconds.
AD/DA converters? I've talked to engineers that design this stuff and some of us can hear a difference because our ears are better trained and our playback systems are good enough to HEAR differences from one DAC to another or one cable from another.
I can't help it if your ears aren't trained and that you have crappy equipment where you can't tell the difference between a 16/44 audio recording and a 24/96 and 24/192. That's not MY problem. That's YOUR problem.
Do you know what articulation measurements are? Have you ever heard of that? Those measurements are used by some of these companies to see how well a audio signal is articulating which better helps them make better products. Articulation measurements started being researched by in the 1940's and are used by acoustic engineers in designing concert halls, studios, etc. for room treatment.
Comments
[...] As far as Nyquest's theorem and how it works, go to WikiPedia and look it up. I don't have time to regurgitate that for you.
But you do know that higher higher res will recreate a signal with more precision. Don't get all caught up with just how high the frequency range is. I'm not necessarily listening to that. Plus you are supposed to get more dynamic range and less noise with high res recordings, etc. But that also depends on your equipment. A high end 16bit DAC might sound better than a cheap 24bit DAC, similarly AD systems as well. But a really good 24/96 should sound a lot better than a really good 16/44.
In WHAT WAY will 24/96 sound better than 16/44? What do you get with 8 more bits? What does doubling the sample rate do?
Until you can answer those questions based purely on what Nyquist, the fundamental rule of digital theory dictates, you can't begin to make any kind of objective assessment. You're human and easily fooled. Think optical illusion, magicians and placebo affect.
I'm an audio engineer. I know how Nyquist works. I know what can and can NOT be accomplished with longer digital words and higher sampling rates. By asking you to explain how it works I was checking if you are able to apply the "mathematical impossibility" filter that prevents experts from being fooled into believing that the Emperor's suit looks AWESOME!
Here is all anyone needs to know on the subject:
Adding bits lowers the noise floor. Nothing more. NOTHING more. It does NOT make sound in the top 80 dB of audibility "smoother" or better. It makes it possible to capture sounds below -115dB or so. If you're recording a classical concert with REALLY wide dynamic range there might be some very slight benefit to 24 bit over 16, but for a typical pop song there will be ZERO difference. None. Nada. Zip. Period. The End.
Increasing the sample rate raises the limit on the highest frequency that can be captured. Period. Nothing more. It does NOT make sounds below 20KHz "smoother" or better. Since a 44.1KHz sample rate captures in excess of the highest frequency humans can hear, there is no benefit to a higher sample rate. None. Zero. Zip. Nada. Anyone who tells you otherwise is either lying to you to get your money or ignorant of how digital audio actually works. It's usually the former.
The beauty of this is that you don't have to take my word for it! Read Nyquist's theorem, the basis upon which digital recording works, and you'll see that there's no room for mystical fairy dust. There is room for making sound better but increasing sample rates and bit depths isn't gonna do squat. (Besides, the difference between any two speakers is a MILLION times bigger than the difference between 16/44 and 24/192!)
In what way do the higher res files sound better than a 16-bit file? What is the audible difference?
Can you please also explain how Nyquist's theorem works? After you do that, you can elaborate on the benefit of any sample rate over ~40KHz?
Here's sequence of videos of what I call, It's more expensive than I can afford, but it's F-ing cool!!! I've heard their speakers in person and it's just an experience you can't emulate, but the videos will show you a little bit of what these systems are about. The guy repeats himself a little and he's a little anal retentive, but just enjoy the videos.
One thing I will tell you to do, if you EVER have the chance to go to one of these high end audiophile shows, GO TO THE MBL, Wilson, Eggleston, Magico, and a few other's rooms. Their systems are so good sounding, you though you died and went to heaven
Wanna have exactly the OPPOSITE experience? Go to a symphony concert in a really nice venue, then rush home and play a recording of the same pieces! Don't do it... ever. Years and years of ear training and controlled listening go completely to crap as you realize that this wonderful playback system that cost more than a good car and that sounds SOOO good is actually only able to reproduce that live event about as well as a Polaroid captures how nice your girlfriend's skin feels. There's no way to describe it, you have to experience it. It's profound.
It was so depressing I got rid of my high-end rig and now listen to an iPod.
Heck, I just downloaded some 24/96 and 24/192 AIFF files from HDTracks and they just kick the living crap out of 16 bit or AAC. It's not even a contest.
In what way do the higher res files sound better than a 16-bit file? What is the audible difference?
Can you please also explain how Nyquist's theorem works? After you do that, you can elaborate on the benefit of any sample rate over ~40KHz?
Glad you bring that up....
16-bit vs. 24-bit: 16-bit is perfectly adequate as a delivery system, provided *everything* was done perfectly during production and mastering, AND the source material was higher resolution and word-length reduced and dithered as the absolutely last step.
Unfortunately, that's a lot of iffs which in reality almost never are true, sticking with 24-bit removes a variety of potential error sources in the digital production work flow, so in any material that's not perfectly produced, mixed and mastered, 24-bit will be sounding better in theory.
As for the sampling rate: 44.1 and certainly 48kHz providing 24kHz signal bandwidth is indeed theoretically sufficient to cover human hearing. The reality of the DA conversion however necessitates anti-aliasing filters. The higher the sampling rate, the lower slope antialiasing filter can be used, and the less phase error is propagating back into the audible range. If there were such a thing as a zero phase error brickwall filter with no pre-ringing and post-ringing effects, etc. then indeed, using sampling frequencies above 48kHz would be an utter waste of bandwidth. A well designed 88/96kHz (2x sampling rates) system should be perfectly adequate, but again, not every audio products company has world class engineers, and many products require shortcuts to stay within budget or power consumption constraints, so using 4x sampling rates makes things a lot more dummy proof in some respects.
So yes, there are practical reasons for these HD audio streams and systems, although most of the mumbo jumbo you hear the "audiophiles" ranting about are about as descriptive of what's really going on as "intelligent design" is about the evolution of live on this planet...
...which is why it takes about the same IQ to finance a televangelists Bentley as it takes to buy pure-oxigen free silver power cables, when from the outlet to the power-plant it's all low-grade copper...
[Censored]
Shhh!
Of course everything you wrote is true, but those are discussions to be held AFTER the very basic, real-world limitations of Nyquist are well understood. Until then this stuff is the fodder semi-informed people use to disseminate nonsense.
I obviously track at 24 bit myself (though not at 96K because the anti-aliasing filters on my converters are so good at 48K that there's no reason to double throughput and storage requirements) but before I would let a protege make that call I'd insist on an explanation of the REASONS for the choice, like what you wrote.
PS. Didn't the oversampling craze of the 1980s promise relief from the issue of ringing from excessively steep anti-aliasing filters?
In WHAT WAY will 24/96 sound better than 16/44? What do you get with 8 more bits? What does doubling the sample rate do?
Until you can answer those questions based purely on what Nyquist, the fundamental rule of digital theory dictates, you can't begin to make any kind of objective assessment. You're human and easily fooled. Think optical illusion, magicians and placebo affect.
I'm an audio engineer. I know how Nyquist works. I know what can and can NOT be accomplished with longer digital words and higher sampling rates. By asking you to explain how it works I was checking if you are able to apply the "mathematical impossibility" filter that prevents experts from being fooled into believing that the Emperor's suit looks AWESOME!
Here is all anyone needs to know on the subject:
Adding bits lowers the noise floor. Nothing more. NOTHING more. It does NOT make sound in the top 80 dB of audibility "smoother" or better. It makes it possible to capture sounds below -115dB or so. If you're recording a classical concert with REALLY wide dynamic range there might be some very slight benefit to 24 bit over 16, but for a typical pop song there will be ZERO difference. None. Nada. Zip. Period. The End.
Increasing the sample rate raises the limit on the highest frequency that can be captured. Period. Nothing more. It does NOT make sounds below 20KHz "smoother" or better. Since a 44.1KHz sample rate captures in excess of the highest frequency humans can hear, there is no benefit to a higher sample rate. None. Zero. Zip. Nada. Anyone who tells you otherwise is either lying to you to get your money or ignorant of how digital audio actually works. It's usually the former.
The beauty of this is that you don't have to take my word for it! Read Nyquist's theorem, the basis upon which digital recording works, and you'll see that there's no room for mystical fairy dust. There is room for making sound better but increasing sample rates and bit depths isn't gonna do squat. (Besides, the difference between any two speakers is a MILLION times bigger than the difference between 16/44 and 24/192!)
Just listen to the different recordings on a decent system that has a good quality DAC, speakers, etc. and see if you hear the difference. IF you can't, then you can't, if you can, then you can.
You are spouting off a LOT OF BULLCRAP. Seriously. Go listen to the same album that's been mastered at different rates on a good system.
Music is complex wave files, but more resolution will yield better dynamic range, lower noise, and better resolution of the audio signals.
The difference between 16/44 to 24/192 using the same system, there is a sonic difference. I can hear it on my system and it's not a big bucks system. On more expensive systems, you just hear the entire bandwidth better. It all boils down to the equipment used, how good your ears are able to detect differences, room acoustics play a role in it. If you room has too many hard surfaces and the music gets bounced around too much, then it's not as noticeable, but THERE IS A DIFFERENCE.
Do you not hear the difference between Dolby Digital and HD Master soundtracks for movies? One is 16 bit, the other 24 bit. If you can't hear the difference, then something is wrong. Maybe your playback system sucks and that might be the reason, or your ears aren't trained to know what to listen for.
Seriously, grow up. There is a sonic difference if the only difference is the bit and sample rates and it's can be heard on a decent system.
Go try it yourself.
What equipment are you using? Have you listened to 16/44 and 24/96 or 24/192 sound files that were from analog recordings?
Just listen to the different recordings on a decent system that has a good quality DAC, speakers, etc. and see if you hear the difference. IF you can't, then you can't, if you can, then you can.
You are spouting off a LOT OF BULLCRAP. Seriously. Go listen to the same album that's been mastered at different rates on a good system.
Music is complex wave files, but more resolution will yield better dynamic range, lower noise, and better resolution of the audio signals.
The difference between 16/44 to 24/192 using the same system, there is a sonic difference. I can hear it on my system and it's not a big bucks system. On more expensive systems, you just hear the entire bandwidth better. It all boils down to the equipment used, how good your ears are able to detect differences, room acoustics play a role in it. If you room has too many hard surfaces and the music gets bounced around too much, then it's not as noticeable, but THERE IS A DIFFERENCE.
Do you not hear the difference between Dolby Digital and HD Master soundtracks for movies? One is 16 bit, the other 24 bit. If you can't hear the difference, then something is wrong. Maybe your playback system sucks and that might be the reason, or your ears aren't trained to know what to listen for.
Seriously, grow up. There is a sonic difference if the only difference is the bit and sample rates and it's can be heard on a decent system.
Go try it yourself.
What equipment are you using? Have you listened to 16/44 and 24/96 or 24/192 sound files that were from analog recordings?
Stop now dude. Seriously. One of these days you're gonna actually READ NYQUIST'S THEOREM and you're gonna realize that the claims you're making here are just plain impossible. OF COURSE some things sound better than some other things. The point is that If you hear a difference between two sources playing normal material at normal listening levels, the difference is NOT word length or sample rate. It just plain CAN'T be. It's not possible. There is ALWAYS another explanation for the difference. You'll see.
Again, you don't have to take my word for it. The math is readily available anywhere.
Good luck with it. I'm out.
Glad you bring that up....
16-bit vs. 24-bit: 16-bit is perfectly adequate as a delivery system, provided *everything* was done perfectly during production and mastering, AND the source material was higher resolution and word-length reduced and dithered as the absolutely last step.
Unfortunately, that's a lot of iffs which in reality almost never are true, sticking with 24-bit removes a variety of potential error sources in the digital production work flow, so in any material that's not perfectly produced, mixed and mastered, 24-bit will be sounding better in theory.
As for the sampling rate: 44.1 and certainly 48kHz providing 24kHz signal bandwidth is indeed theoretically sufficient to cover human hearing. The reality of the DA conversion however necessitates anti-aliasing filters. The higher the sampling rate, the lower slope antialiasing filter can be used, and the less phase error is propagating back into the audible range. If there were such a thing as a zero phase error brickwall filter with no pre-ringing and post-ringing effects, etc. then indeed, using sampling frequencies above 48kHz would be an utter waste of bandwidth. A well designed 88/96kHz (2x sampling rates) system should be perfectly adequate, but again, not every audio products company has world class engineers, and many products require shortcuts to stay within budget or power consumption constraints, so using 4x sampling rates makes things a lot more dummy proof in some respects.
So yes, there are practical reasons for these HD audio streams and systems, although most of the mumbo jumbo you hear the "audiophiles" ranting about are about as descriptive of what's really going on as "intelligent design" is about the evolution of live on this planet...
...which is why it takes about the same IQ to finance a televangelists Bentley as it takes to buy pure-oxigen free silver power cables, when from the outlet to the power-plant it's all low-grade copper...
Boy, you are talking a bunch of BS. When the power comes into your house, they have things like line filtering, line conditioning, etc. to clean up the mess provided by the power company. Then it goes through cables of various degrees. then it goes through a series of other equipment. EVERY SINGLE PIECE OF EQUIPMENT, including cables has measurable amounts of resistance, inductance, and capacitance, even cables. There are a LOT of things going on to remove distortion whether it's analog distortion or digital distortion (jitter) through out the entire chain. Now, if you listen to music that's highly altered through the use of eq, limiters, compressors, expanders, etc., then you are capturing music somehow and altering it, so you are MESSING up the original content, so you don't really know what the original instrument voice sounds like. That's why MOST pop/commercial music sounds like crap. The audiophile recordings are done typically using two microphones of exceptionally high quality, running through high quality cables, etc. etc. and they DO NOT alter the signal at all throughout the process.
Now, if you want a little education, what do you think one of the most prestigious mastering engineers uses? Bob Ludwig uses a pair of Eggleston speakers that cost around $100K, running from Cello amps that are about another $100K plus, with Transparent speaker and interconnect cables which are another $100K or so and that's what he uses as his high end reference system for classical music and other music that dictates a high quality mastering job. Now, if he gets a call to produce a mastering job that's going to be played on the radio or some cheap speakers, he'll use basically the same level of playback system only he'll change the speaker to NS-10s to give him a reference of a crappy pair of speakers that lack detail, bass, etc. Kind of what most people have.
Now, what do you think Abbey Road Studios use? B&W 800 speakers with Classe Electronics, and their own custom made cables. What do you think Skywalker Sound uses for audio recordings and film soundtracks? B&W 800 series speakers, MIT cables and relatively high end amplifiers. They all have power filtration/conditioning to remove the crap power coming from the power company.
Now, if you want to live in TOTAL ignorance, go right ahead. ANY REALLY GOOD HIGH QUAITY RECORDING STUDIO IS GOING TO USE REALLY EXPENSIVE playback systems as their reference and they have been switching gradually to higher quality, more expensive cables, amplification, and speaker systems. HDTracks are done with Wilson Audio speakers, high end power amps, and cables. These people LIVE AND BREATH audio, they've been doing it for years, have pristine systems and rooms and their reputation depends on their finished product.
The Nyquest theorem does NOT explain everything with digital, it only explains a portion of what's going on. Remember, it's just a theorem.
Do you know what is considered the highest quality AD/DA converters? According to Abbey Road, they prefer DAD, but there are other brands used in high end recording studios and it's not just the DAC chips, but it's the power supplies, input and output stages.
So, again, if you claim there is no difference, what makes you a better expert than the studios and people I've mentioned? What have you done to prove you are correct when there are INDUSTRY experts pumping out recordings that are seen as some of the most influential people will tell you that there is a difference in cables, etc. etc. etc. etc. Please explain YOURSELF and why YOU think you know everything about digital recordings.
Shhh!
Of course everything you wrote is true, but those are discussions to be held AFTER the very basic, real-world limitations of Nyquist are well understood. Until then this stuff is the fodder semi-informed people use to disseminate nonsense.
I obviously track at 24 bit myself (though not at 96K because the anti-aliasing filters on my converters are so good at 48K that there's no reason to double throughput and storage requirements) but before I would let a protege make that call I'd insist on an explanation of the REASONS for the choice, like what you wrote.
PS. Didn't the oversampling craze of the 1980s promise relief from the issue of ringing from excessively steep anti-aliasing filters?
It depends on what one thinks is GOOD ENOUGH. Remember, some people have lived and breathed around high end equipment for years, know the different in the tonal quality of one concert grand piano to another, can tell the different in the tonal quality of high end microphones and have basically really well trained ears. Did you know that there were tests done a long time ago and the average person can't tell between a perfect note and as little as 5 cent difference in pitch where as trained musicians could decern as little as 2 cent difference in pitch. That's just being able to tell pitch. A concert grand pianist can tell the difference between a Boesendorfer and a Steinway, etc. whereas the average person can't.
If you aren't doing reference quality recordings in a pristine environment, then you might not be able to hear these sonic differences, but those that have lived around these higher end systems are acclimated to them and they can hear the differences.
I know there are blind fold tests that have been done, but the tests that were performed could have been set up in a manner to give that result, since they might be using switch boxes and other cables that will prevent any differences heard between one set of cables and another. Obviously, it's all comes down to the equipment, room, content, and listener. If you can hear it, then you can, if you can't then you can't. I take a I have to hear it for myself approach, but I will listen to higher end systems to listen to differences and I decide for myself what I like and don't like and price is only decided upon on what I buy and afford. But some of the best sounding systems were VERY expensive and actually not the most expensive in the world, but then again, I haven't heard the most expensive systems to compare them to, i just know what I liked.
Oversampling can be done poorly or well. Meridian, for example, has created their own unique apodising filter to remove pre-ringing, others are doing the same thing and they achieve better sound than just oversampling by itself. So, it really depends on what and how oversampling is done. There are so many ways to get a decent digital sound, so all I can say is that on my system, I can tell the difference between a 16 bit file that gets oversampled and runs through a apodising filter to remove pre-ringing compared to a 24/96 version on the exact same equipment. So, I don't know what else to tell you, but my USB DAC certainly sounds a LOT better than the internal one and there is no comparison.
It depends on what one thinks is GOOD ENOUGH. Remember, some people have lived and breathed around high end equipment for years, know the different in the tonal quality of one concert grand piano to another, can tell the different in the tonal quality of high end microphones and have basically really well trained ears. Did you know that there were tests done a long time ago and the average person can't tell between a perfect note and as little as 5 cent difference in pitch where as trained musicians could decern as little as 2 cent difference in pitch. That's just being able to tell pitch. A concert grand pianist can tell the difference between a Boesendorfer and a Steinway, etc. whereas the average person can't.
If you aren't doing reference quality recordings in a pristine environment, then you might not be able to hear these sonic differences, but those that have lived around these higher end systems are acclimated to them and they can hear the differences.
I know there are blind fold tests that have been done, but the tests that were performed could have been set up in a manner to give that result, since they might be using switch boxes and other cables that will prevent any differences heard between one set of cables and another. Obviously, it's all comes down to the equipment, room, content, and listener. If you can hear it, then you can, if you can't then you can't. I take a I have to hear it for myself approach, but I will listen to higher end systems to listen to differences and I decide for myself what I like and don't like and price is only decided upon on what I buy and afford. But some of the best sounding systems were VERY expensive and actually not the most expensive in the world, but then again, I haven't heard the most expensive systems to compare them to, i just know what I liked.
Oversampling can be done poorly or well. Meridian, for example, has created their own unique apodising filter to remove pre-ringing, others are doing the same thing and they achieve better sound than just oversampling by itself. So, it really depends on what and how oversampling is done. There are so many ways to get a decent digital sound, so all I can say is that on my system, I can tell the difference between a 16 bit file that gets oversampled and runs through a apodising filter to remove pre-ringing compared to a 24/96 version on the exact same equipment. So, I don't know what else to tell you, but my USB DAC certainly sounds a LOT better than the internal one and there is no comparison.
Stop now dude. Seriously. One of these days you're gonna actually READ NYQUIST'S THEOREM and you're gonna realize that the claims you're making here are just plain impossible. OF COURSE some things sound better than some other things. The point is that If you hear a difference between two sources playing normal material at normal listening levels, the difference is NOT word length or sample rate. It just plain CAN'T be. It's not possible. There is ALWAYS another explanation for the difference. You'll see.
Again, you don't have to take my word for it. The math is readily available anywhere.
Good luck with it. I'm out.
I've read Nyquist's THEOREM years ago, but it does NOT explain everything. It's just showing you a small aspect of what digital is all about.
Do you not think that there's a lot more to it than that? What about measurable amounts of jitter? Does Nyquist explain that? NO. It doesn't. Nyquist is a THEOREM, what's a theorem? THEORY. And it's only using math to explain one aspect of a digital signal, but they know more about digital signals since Nyquist's THEORY and that's why companies are pushing the envelope, just like they are making speakers better than they ever have, power amps, pre amps, cables, etc. Things become better by the use of better, more precise measurement tools, different types of measurements, different materials, etc. etc. Sounds like you just hide behind some BS theory to back your BS excuse because you can't hear the difference because of the lack of quality equipment and experience you have.
IF they can measure something, they can also hear it under the right circumstances, which is what the high end guys are doing and that gets filtered down to the masses at some level. But the average person doesn't know what a real instrument sounds like unless they play an instrument.
Seriously, you sound like you don't have the money for better equipment and are afraid to list what you are using. Tell me what playback equipment you are using. If you are using cheap earbuds from an mobile device, yeah, you probably can't hear much difference because the DAC is a cheap DAC and the earbuds are POS. But take that same file and put it on a extremely nice system in a good room and people can hear a difference if they know what to listen for and have trained ears that haven't polluted by loud distorted, overly processed crap.
Glad you bring that up....
16-bit vs. 24-bit: 16-bit is perfectly adequate as a delivery system, provided *everything* was done perfectly during production and mastering, AND the source material was higher resolution and word-length reduced and dithered as the absolutely last step.
Unfortunately, that's a lot of iffs which in reality almost never are true, sticking with 24-bit removes a variety of potential error sources in the digital production work flow, so in any material that's not perfectly produced, mixed and mastered, 24-bit will be sounding better in theory.
As for the sampling rate: 44.1 and certainly 48kHz providing 24kHz signal bandwidth is indeed theoretically sufficient to cover human hearing. The reality of the DA conversion however necessitates anti-aliasing filters. The higher the sampling rate, the lower slope antialiasing filter can be used, and the less phase error is propagating back into the audible range. If there were such a thing as a zero phase error brickwall filter with no pre-ringing and post-ringing effects, etc. then indeed, using sampling frequencies above 48kHz would be an utter waste of bandwidth. A well designed 88/96kHz (2x sampling rates) system should be perfectly adequate, but again, not every audio products company has world class engineers, and many products require shortcuts to stay within budget or power consumption constraints, so using 4x sampling rates makes things a lot more dummy proof in some respects.
So yes, there are practical reasons for these HD audio streams and systems, although most of the mumbo jumbo you hear the "audiophiles" ranting about are about as descriptive of what's really going on as "intelligent design" is about the evolution of live on this planet...
...which is why it takes about the same IQ to finance a televangelists Bentley as it takes to buy pure-oxigen free silver power cables, when from the outlet to the power-plant it's all low-grade copper...
Boy, you are talking a bunch of BS. When the power comes into your house, they have things like line filtering, line conditioning, etc. to clean up the mess provided by the power company. Then it goes through cables of various degrees. then it goes through a series of other equipment. EVERY SINGLE PIECE OF EQUIPMENT, including cables has measurable amounts of resistance, inductance, and capacitance, even cables. There are a LOT of things going on to remove distortion whether it's analog distortion or digital distortion (jitter) through out the entire chain.
[...blah...blah...]
None of which has anything to do with silver power cables, which are utterly pointless, because any decent, and even more so any high-end equipment has an internal power supply that filters, regulates, stabilizes and transforms the power from AC to DC at levels useful for the electronics inside.
Now, if you want a little education, what do you think one of the most prestigious mastering engineers uses? Bob Ludwig uses a pair of Eggleston speakers that cost around $100K, running from Cello amps that are about another $100K plus, with Transparent speaker and interconnect cables which are another $100K or so and that's what he uses as his high end reference system for classical music and other music that dictates a high quality mastering job. Now, if he gets a call to produce a mastering job that's going to be played on the radio or some cheap speakers, he'll use basically the same level of playback system only he'll change the speaker to NS-10s to give him a reference of a crappy pair of speakers that lack detail, bass, etc. Kind of what most people have.
Now, what do you think Abbey Road Studios use? B&W 800 speakers with Classe Electronics, and their own custom made cables. What do you think Skywalker Sound uses for audio recordings and film soundtracks? B&W 800 series speakers, MIT cables and relatively high end amplifiers. They all have power filtration/conditioning to remove the crap power coming from the power company.
Now, if you want to live in TOTAL ignorance, go right ahead. ANY REALLY GOOD HIGH QUAITY RECORDING STUDIO IS GOING TO USE REALLY EXPENSIVE playback systems as their reference and they have been switching gradually to higher quality, more expensive cables, amplification, and speaker systems. HDTracks are done with Wilson Audio speakers, high end power amps, and cables. These people LIVE AND BREATH audio, they've been doing it for years, have pristine systems and rooms and their reputation depends on their finished product.
The Nyquest theorem does NOT explain everything with digital, it only explains a portion of what's going on. Remember, it's just a theorem.
Do you know what is considered the highest quality AD/DA converters? According to Abbey Road, they prefer DAD, but there are other brands used in high end recording studios and it's not just the DAC chips, but it's the power supplies, input and output stages.
So, again, if you claim there is no difference, what makes you a better expert than the studios and people I've mentioned? What have you done to prove you are correct when there are INDUSTRY experts pumping out recordings that are seen as some of the most influential people will tell you that there is a difference in cables, etc. etc. etc. etc. Please explain YOURSELF and why YOU think you know everything about digital recordings.
For one, most of that expensive equipment you quote has nothing to do with the sampling frequencies we were discussing, because like you what you don't seem to know, speakers and amps are analog equipment.
Even in equipment that is digital and where sampling rates and word lengths do matter, there are no mono-causal relationships between equipment quality and these two parameters. Well engineered ADC/DACs, like e.g. my Metric Halo ULN-8, cost more, because they are done right, and so someone might use that equipment even if they work at 48kHz sampling rates, because everything else is done better. There are also issues internal to the various DAC/ADC chips, such as internal signal paths, which sampling clock is derived from which other, such that e.g. 192kHz may have lower jitter than 48kHz on the same equipment, resulting in 192 downsampled to 48kHz sounding better than straight 48kHz recording, even though the theoretical information content is the same, simply because the clock was better, etc.
All that said: nobody has invalidated Nyquist yet, and if someone did, it be about as huge as the discovery of quantum physics or something on that scale.
Many of the "golden ears" are not engineers and they talk mumbo-jumbo that could come straight from an astrologer or shaman trying to explain which cable has "better spirits". The fact is, that just because you can tell something sounds better doesn't mean you know why it sounds better, nor do you necessarily know if that better sound is euphonic or closer to reality, all of which means that many so-called authorities talk so much crap that physicist and engineers start running out of the room screaming when these people start talking.
Further, there are endorsement deals and the whole politics of the high-end audio world, musician clients that need to be impressed so they shut up and don't make stupid requests, etc. I knew of mastering engineers that bough boat loads of broken vintage audio gear to fill up 19" racks with moving needles and blinking LEDs, because that show would shut up know-it-all-better musicians, and in fact, 98% of what they did was done in software in a computer. They even had some A/B switches to let artists decide which version they liked better, when in fact the A/B switch did nothing except give the musician the cozy feeling in the belly that they were "actively involved" in the mastering process.
Oh yeah, and last but not least, which proves that you either didn't read my post or have language understanding issues, I was actually making the case FOR higher sampling rates and word lengths, but that seemed to have escaped you.
None of which has anything to do with silver power cables, which are utterly pointless, because any decent, and even more so any high-end equipment has an internal power supply that filters, regulates, stabilizes and transforms the power from AC to DC at levels useful for the electronics inside.
For one, most of that expensive equipment you quote has nothing to do with the sampling frequencies we were discussing, because like you what you don't seem to know, speakers and amps are analog equipment.
Even in equipment that is digital and where sampling rates and word lengths do matter, there are no mono-causal relationships between equipment quality and these two parameters. Well engineered ADC/DACs, like e.g. my Metric Halo ULN-8, cost more, because they are done right, and so someone might use that equipment even if they work at 48kHz sampling rates, because everything else is done better. There are also issues internal to the various DAC/ADC chips, such as internal signal paths, which sampling clock is derived from which other, such that e.g. 192kHz may have lower jitter than 48kHz on the same equipment, resulting in 192 downsampled to 48kHz sounding better than straight 48kHz recording, even though the theoretical information content is the same, simply because the clock was better, etc.
All that said: nobody has invalidated Nyquist yet, and if someone did, it be about as huge as the discovery of quantum physics or something on that scale.
Many of the "golden ears" are not engineers and they talk mumbo-jumbo that could come straight from an astrologer or shaman trying to explain which cable has "better spirits". The fact is, that just because you can tell something sounds better doesn't mean you know why it sounds better, nor do you necessarily know if that better sound is euphonic or closer to reality, all of which means that many so-called authorities talk so much crap that physicist and engineers start running out of the room screaming when these people start talking.
Further, there are endorsement deals and the whole politics of the high-end audio world, musician clients that need to be impressed so they shut up and don't make stupid requests, etc. I knew of mastering engineers that bough boat loads of broken vintage audio gear to fill up 19" racks with moving needles and blinking LEDs, because that show would shut up know-it-all-better musicians, and in fact, 98% of what they did was done in software in a computer. They even had some A/B switches to let artists decide which version they liked better, when in fact the A/B switch did nothing except give the musician the cozy feeling in the belly that they were "actively involved" in the mastering process.
Oh yeah, and last but not least, which proves that you either didn't read my post or have language understanding issues, I was actually making the case FOR higher sampling rates and word lengths, but that seemed to have escaped you.
All they are doing is explaining what they hear, so they use terms like soundstage, depth, harshness, detail, etc. etc. That's the only way they can explain what they HEAR.
Now, by your BS, all DAC should sound identical when at 16/44, 16/48 or whatever level they are at, but unfortunately they don't all sound the same.
Um, just an FYI, Skywalker Sound did NOT get free equipment or money for their endorsement for MIT Cables, Bob Ludwig did NOT get free equipment or get paid to give his endorsement for Transparent Cables. Yeah, the companies like Yamaha, and others companies might resort to paying endorsers, giving free gear, but a lot of these high end companies don't. They can't afford to. Some of them might get prototypes to help further development and are used to evaluate and give their feedback on new cable, or equipment designs, but a lot of them do not get paid or get free equipment. How do I know? I know people that work for those companies and asked them. Now, the heavily marketed companies might resort to those tactics because they have to, but a lot of these smaller high end companies DON'T do that.
Many of these reviewers for some magazines are musicians, have an engineering background as a recording engineer, or sometimes an electronics engineering background, so it's a case by case basic as to their background, but they get involved with listening and evaluating speakers, cables, amps, preamps, etc. in nice sounding rooms and they give their feedback on what they like and don't like, but I take it all with a grain of salt and do my own listening tests, but I use their guidance to help me identify what to listen for and I see for myself if i can or can't hear a difference.
For grins, I bought a AC noise sniffer, hooked it up to an AC outlet with a clock radio connected to the other outlet and the clock radio was turned off. The noise sniffer has a speaker that I can actually hear the line noise and I actually heard a radio station coming through fairly clearly on the sniffer. It's one of those Entech noise sniffers that are available and they are used by people to measure and hear line noise in AC. There is also a digital readout on the amount of noise as well. Interesting little box and I didn't pay all that much for it, there are more expensive one out there, but I bought this as other respected companies that make line filtration and conditioning systems use the same device.
None of which has anything to do with silver power cables, which are utterly pointless, because any decent, and even more so any high-end equipment has an internal power supply that filters, regulates, stabilizes and transforms the power from AC to DC at levels useful for the electronics inside.
For one, most of that expensive equipment you quote has nothing to do with the sampling frequencies we were discussing, because like you what you don't seem to know, speakers and amps are analog equipment.
Even in equipment that is digital and where sampling rates and word lengths do matter, there are no mono-causal relationships between equipment quality and these two parameters. Well engineered ADC/DACs, like e.g. my Metric Halo ULN-8, cost more, because they are done right, and so someone might use that equipment even if they work at 48kHz sampling rates, because everything else is done better. There are also issues internal to the various DAC/ADC chips, such as internal signal paths, which sampling clock is derived from which other, such that e.g. 192kHz may have lower jitter than 48kHz on the same equipment, resulting in 192 downsampled to 48kHz sounding better than straight 48kHz recording, even though the theoretical information content is the same, simply because the clock was better, etc.
All that said: nobody has invalidated Nyquist yet, and if someone did, it be about as huge as the discovery of quantum physics or something on that scale.
Many of the "golden ears" are not engineers and they talk mumbo-jumbo that could come straight from an astrologer or shaman trying to explain which cable has "better spirits". The fact is, that just because you can tell something sounds better doesn't mean you know why it sounds better, nor do you necessarily know if that better sound is euphonic or closer to reality, all of which means that many so-called authorities talk so much crap that physicist and engineers start running out of the room screaming when these people start talking.
Further, there are endorsement deals and the whole politics of the high-end audio world, musician clients that need to be impressed so they shut up and don't make stupid requests, etc. I knew of mastering engineers that bough boat loads of broken vintage audio gear to fill up 19" racks with moving needles and blinking LEDs, because that show would shut up know-it-all-better musicians, and in fact, 98% of what they did was done in software in a computer. They even had some A/B switches to let artists decide which version they liked better, when in fact the A/B switch did nothing except give the musician the cozy feeling in the belly that they were "actively involved" in the mastering process.
Oh yeah, and last but not least, which proves that you either didn't read my post or have language understanding issues, I was actually making the case FOR higher sampling rates and word lengths, but that seemed to have escaped you.
Um, by your way of thinking, every 16/44 DAC should sound identical regardless of mfg and model. That's what YOUR theory suggests, but reality is that they don't all sound equal. They also don't go through the battery of measurement tests equally either. so, how can you back your statement up when these different DACs measure differently, even though they are all doing 16/44 or 16/58 or 24/96, etc. etc. and how come leading well respected people in the mastering and audio recording industry that record live classical or jazz, etc. music in a really good acoustic environment using high quality microphones trying to capture the sound of musicians and being able to provide the most accurate recordings that the equipment they use are capable of, and then they listen to their recordings back and are trying to compare the listening experience to the actual performance, which is hard to do, but they are doing it the best they can? You theory doesn't explain any of this, but this is what is HAPPENING. You can talk sample rates, bit rates until you are blue in the face, but you haven't mentioned jitter, you haven't mentioned any of the other test measurements that these companies do when developing a product to decide what components, and design to use. Do you know the different between the different mfg DAC chips on the market?
Seriously, grow up. You seem to have some bitterness towards people that appreciate, listen to , and can afford more expensive equipment than you can. How come the brands and model AD converters are changing in these high end studios and the quality of the recordings have changed since the introduction of digital?
The Nyquest theorem does NOT explain everything with digital, it only explains a portion of what's going on. Remember, it's just a theorem.
No dude, wrong. WAY wrong. So wrong you should be disqualified from practicing audio until you get that misconception fixed.
Nyquist DOES explain absolutely every single aspect of digital recording. It is more absolute than e=mc2. There is no ambiguity.
For grins, I bought a AC noise sniffer, hooked it up to an AC outlet with a clock radio connected to the other outlet and the clock radio was turned off. The noise sniffer has a speaker that I can actually hear the line noise and I actually heard a radio station coming through fairly clearly on the sniffer. It's one of those Entech noise sniffers that are available and they are used by people to measure and hear line noise in AC.
You have no idea how much grin I'm getting from this.
Quiz for the blank one: What's the VERY FIRST THING that AC power hits once it goes inside the box? What are the effects of that device?
Seriously dude, I'm not trying to beat you up here, but you just keep doing it to yourself! Your example above is EXACTLY what I'm talking about -- those with little knowledge of basic physics are very easily led to believe utter bullshit.
For an explanation of why your sniffer is the most pointless device ever built, look into my question, above, or just ask your local TV repairman.
Um, by your way of thinking, every 16/44 DAC should sound identical regardless of mfg and model.
NO! Jeez, this is like trying to teach a kid to shoot when he won't take the gun out of his mouth...
Obviously different circuits sound different. If you look back through the thread, no one here has suggested otherwise. You're arguing a point no one made.
You sound like a kid with a can of NO2 telling the engineers at McLaren that the laws of physics are just a guideline and not really very well understood. They are actually VERY well understood, just not by you.
No dude, wrong. WAY wrong. So wrong you should be disqualified from practicing audio until you get that misconception fixed.
Nyquist DOES explain absolutely every single aspect of digital recording. It is more absolute than e=mc2. There is no ambiguity.
NO it doesn't. It's just a THEORY. What about clocking? what about losing LSB? What about jitter? What about s/n, distortion, etc.? Nyquist does NOT explain everything, it's a THEORY.
By your way of thinking, all DA and AD converters sound the same, but they don't.
Go talk to Abbey Road Studios and ask them why they switched from their old AD/DA converters to DADs? Same thing with EMI Studios and other studios that are buying the big bucks AD/DA converters? There are the output and input stages, clocking, etc. etc.
The Nyquist Theorem is just PART OF IT.
Go talk to converter companies that make the high end stuff and then maybe you'll get an education.
You have no idea how much grin I'm getting from this.
Quiz for the blank one: What's the VERY FIRST THING that AC power hits once it goes inside the box? What are the effects of that device?
Seriously dude, I'm not trying to beat you up here, but you just keep doing it to yourself! Your example above is EXACTLY what I'm talking about -- those with little knowledge of basic physics are very easily led to believe utter bullshit.
For an explanation of why your sniffer is the most pointless device ever built, look into my question, above, or just ask your local TV repairman.
OK, then how come I can measure the noise in the AC line and then get a certain power conditioner/filter that cleans up that noise? Why do you think high end recording studios, data centers spend LOTS of money on power conditioning and filtration systems? For the fun of it?
Grow up. We aren't getting clean properly regulated power from the power companies.
I have a lot of noise that's visible on the TV, so how do you explain that? It's not a perfect picture.
I would rather talk to engineers that specialize in designing power conditioning/filtration systems than some IDIOT on AI.
OK, then how come I can measure the noise in the AC line and then get a certain power conditioner/filter that cleans up that noise? Why do you think high end recording studios, data centers spend LOTS of money on power conditioning and filtration systems? For the fun of it?
Grow up. We aren't getting clean properly regulated power from the power companies.
I have a lot of noise that's visible on the TV, so how do you explain that? It's not a perfect picture.
I would rather talk to engineers that specialize in designing power conditioning/filtration systems than some IDIOT on AI.
A local TV repair man? Hahahahahahahaa. Most of those guys have basic knowledge and basic tools for which they are using.
NO! Jeez, this is like trying to teach a kid to shoot when he won't take the gun out of his mouth...
Obviously different circuits sound different. If you look back through the thread, no one here has suggested otherwise. You're arguing a point no one made.
You sound like a kid with a can of NO2 telling the engineers at McLaren that the laws of physics are just a guideline and not really very well understood. They are actually VERY well understood, just not by you.
You are now becoming a worthless pile of garbage. I've talked to engineers that design power conditioning systems, they will put you in your place in 2 seconds.
AD/DA converters? I've talked to engineers that design this stuff and some of us can hear a difference because our ears are better trained and our playback systems are good enough to HEAR differences from one DAC to another or one cable from another.
I can't help it if your ears aren't trained and that you have crappy equipment where you can't tell the difference between a 16/44 audio recording and a 24/96 and 24/192. That's not MY problem. That's YOUR problem.
Do you know what articulation measurements are? Have you ever heard of that? Those measurements are used by some of these companies to see how well a audio signal is articulating which better helps them make better products. Articulation measurements started being researched by in the 1940's and are used by acoustic engineers in designing concert halls, studios, etc. for room treatment.