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iOS 7 seeing slower uptake than Apple's iOS 6 - report - Page 6

post #201 of 270
Quote:
Originally Posted by MagMan1979 View Post
 

Wow, so you're refusing to upgrade to a superior version of iOS because of the colours and the fact you gain automatic updates to ensure you always have the latest version, which I might add brings with it security enhancements and hardening?

 

And you're saying you prefer OS X 10.6 to 10.8? Talk about an old dog who can't learn new tricks. Fine, stay with iOS 6, hope it crashes and burns on you and you're forced to go with iOS 7. Security is only as strong as it's weakest link, and people like you are the weakest link in the chain.

 

I hate the colors in iOS7, too.  I also will not be upgrading and, for as long as iOS7 is the main theme, I will not be buying further iDevices.  (Have had 4 iPhones, 3 iPads, 1 iPad mini, 2 iPod touches).  

 

There is a rumor that Mac OS will take some design elements from iOS7 after Mavericks.  If that is the case, I will buy a Mac that will last four years or so in order to give it some time to get fixed.  My first Mac was a MacPlus back somewhere around 1989.  Have owned countless since.  If iOS7 takes over Mac OS, the next Mac I buy could be my last.  That's life.

 

Yes, I feel that strongly about the design of iOS7.  I think it is hideous.  Lots of people like it.  That's fine. 

 

Your = the possessive of you, as in, "Your name is Tom, right?" or "What is your name?"

 

You're = a contraction of YOU + ARE as in, "You are right" --> "You're right."

 

 

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Your = the possessive of you, as in, "Your name is Tom, right?" or "What is your name?"

 

You're = a contraction of YOU + ARE as in, "You are right" --> "You're right."

 

 

Reply
post #202 of 270
Quote:
Originally Posted by Bergermeister View Post
 

 

 

 

Have a bad weekend?  

Nope, just sick and tired of people claiming they're refusing the upgrade because of colours. People always put security on the back burner, when in this day and age, with the amount of content and personal data we put on these devices, security MUST be #1 priority.

post #203 of 270
Quote:
Originally Posted by MagMan1979 View Post
 

Nope, just sick and tired of people claiming they're refusing the upgrade because of colours. People always put security on the back burner, when in this day and age, with the amount of content and personal data we put on these devices, security MUST be #1 priority.

Especially, perhaps, when the colors are changeable!

post #204 of 270
From today's Apple event 200 million devices upgraded to iOS in five days. 64 percent of iDevices are now on iOS 7.
melior diabolus quem scies
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melior diabolus quem scies
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post #205 of 270
Quote:
Originally Posted by drblank View Post
 

The major beefs I have with iTunes are things like not supports FLAC, DSD.  It doesn't switch the Audio/MIDI settings on the fly like 3rd party players do, or have iZotrope 64 SRC and a few other features that players like Audirvana, Amarra, etc. have.

 

You don't honestly expect Apple to support any of that stuff in iTunes, do you?  I agree about switching Audio/MIDI settings on the fly, but the rest of that stuff is really beyond the realm of what I'd expect from a free consumer grade music player and store application.  FLAC is pretty much a guaranteed no-go because of Apple's agreements with record labels.

 

The question I have is what is it that programs like Audirvana, Amarra, etc. don't have that iTunes does that even keeps iTunes on your radar in the first place?  Is it mostly to do with syncing to iOS?

post #206 of 270
Quote:
Originally Posted by mutoneon View Post
 

 

You don't honestly expect Apple to support any of that stuff in iTunes, do you?  I agree about switching Audio/MIDI settings on the fly, but the rest of that stuff is really beyond the realm of what I'd expect from a free consumer grade music player and store application.  FLAC is pretty much a guaranteed no-go because of Apple's agreements with record labels.

 

The question I have is what is it that programs like Audirvana, Amarra, etc. don't have that iTunes does that even keeps iTunes on your radar in the first place?  Is it mostly to do with syncing to iOS?

 

 

Audirvana, Amarra, leverage iTunes as the catalog and some of them add the ability to add or links to FLAC, DSD, etc.  What these 3rd party products do is able to add DSD, FLAC, a better audio engine, etc.   There is a GROWING push towards adding USB DACs to computers and going from these DACs to REAL stereo systems.  The MacMini, is becoming quite popular amongst the audio crowd since it's an inexpensive media server.  There are people that will buy a MacMini just for a media server and add JRiver Media Center, or any of these 3rd party apps and they get all kinds of files from a variety of download sites and ripping from disc.  

 

I think Apple would do VERY well creating iTunes into a REALLY good media server software and iTunes is it, it just needs to add more functionality, get a better audio engine, etc.

 

I don't see any problem with Apple buying out any number of these third party players and adding more functionality to iTunes.

 

I see iTunes as a repository for content, some is ripped from disc, some bought from iTunes, etc.

 

It would certainly make life a lot easier for the users if they went in this direction so that we can put more content in iTunes and manage it a LOT easier.

 

Switching Audio/MIDI settings on the fly should be done at a bare minimum, but there needs to be more people requesting that feature.

post #207 of 270
Originally Posted by drblank View Post

Audirvana, Amarra, leverage iTunes as the catalog and some of them add the ability to add or links to FLAC, DSD, etc. 

 

FLAC will never be supported. ALAC’s better, anyway. What’s DSD?

 

EDIT: Oh, Sony? Never gonna happen either, for that reason. I would have thought Sony had learned their lesson on making useless, proprietary Schmidt that no one wants to use.

Originally Posted by asdasd

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Originally Posted by asdasd

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post #208 of 270
Quote:
Originally Posted by RemE View Post

Based on my own personal experience, I ask every iphone user I come across while I'm in the field two simple questions, "did you upgrade to IOS7" and "how do you like it".

I've asked probably 100 folks so far and overwhelming response is negative, many find the apps less intuitive and harder to see.  Very few like the flat icons, they are just not "pretty".  Some said, "were they trying to copy Microsoft?".  I honestly have not talked with anyone who said that it was better, even though it has some great features.

I bet the uptake slows even more as people talk.  If Apple doesn't listen and react to this, there will be a price to pay moving forward.

Sounds like you hang out with a lot of negative, entitled whiners.

Proud AAPL stock owner.

 

GOA

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Proud AAPL stock owner.

 

GOA

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post #209 of 270
Quote:
Originally Posted by Tallest Skil View Post
 

 

FLAC will never be supported. ALAC’s better, anyway. What’s DSD?

 

EDIT: Oh, Sony? Never gonna happen either, for that reason. I would have thought Sony had learned their lesson on making useless, proprietary Schmidt that no one wants to use.

Um, people can add FLAC and DSD through the use of these 3rd party apps that leverages iTunes.  I'm currently using Audirvana with great success.  I don't have FLAC files, but others do have them.  DSD is becoming more popular as there are more and more USB DACs that are supporting DSD and there are more titles coming out on DSD.  Obviously, DSD is meant for the audiophile crowd and it's not mainstream, but the shear number of USB DACs is increasing and DSD support is getting more popular with that crowd of people.

 

So, there are ways around it.  It would just make for a slicker app if Apple just added DSD support.  because ultimately, it's what users want and the non-DSD users won't know the difference, but the DSD users WILL.  Otherwise, we have to spend X number of dollars buying a 3rd party app, when Apple could just easily buy one of these companies and just add more features to iTunes, whether or not they start selling DSD files.

 

Apple only sells AAC music files, but they support other file formats, right?  So I don't see what the problem is. 

post #210 of 270
Quote:
Originally Posted by Tallest Skil View Post
 

 

FLAC will never be supported. ALAC’s better, anyway. What’s DSD?

 

EDIT: Oh, Sony? Never gonna happen either, for that reason. I would have thought Sony had learned their lesson on making useless, proprietary Schmidt that no one wants to use.

DSD?  It's used by Sony and Philips.  It's what SACDs use, but the downloads aren't reliant on SACDs.  They just have DSD downloads and they are kicking up the bit rate to get better sound quality.  It's a format that the higher end audiophiles are liking and it's becoming yet another standard for high res files.


Remember, Sony has a HUGE catalog of classic material and I was given the option of paying a little extra for higher resolution than I will do so.  All DSD requires is a DSD DAC, not a SACD player.  That's where Sony messed up was with the player.

 

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.  I haven't heard DSD compared, but others have said it sounds even better, plus it can be converted, but that's an extra step that some don't want to have to do.

post #211 of 270
Quote:
Originally Posted by Tallest Skil View Post
 

 

FLAC will never be supported. ALAC’s better, anyway. What’s DSD?

 

EDIT: Oh, Sony? Never gonna happen either, for that reason. I would have thought Sony had learned their lesson on making useless, proprietary Schmidt that no one wants to use.

I can add FLAC files to iTunes with Audirvana, Amarra and others. So, there are ways around it but it just would be nice if Apple made iTunes software support all file formats so that we didn't have to go through a conversion process.  It would just make people's lives a little easier that enjoy high res files and I don't see what the problem is.  They can obviously sell whatever they want, but high res files are becoming more popular as USB DACs become more popular.

 

At last count, there are a lot of USB DACs on the market that start in the $250 range all the way up to $20K or more.  Almost every week or so a new DAC comes out.

post #212 of 270
Quote:
Originally Posted by drblank View Post
 
Quote:
Originally Posted by Tallest Skil View Post
 

 

FLAC will never be supported. ALAC’s better, anyway. What’s DSD?

 

EDIT: Oh, Sony? Never gonna happen either, for that reason. I would have thought Sony had learned their lesson on making useless, proprietary Schmidt that no one wants to use.

DSD?  It's used by Sony and Philips.  It's what SACDs use, but the downloads aren't reliant on SACDs.  They just have DSD downloads and they are kicking up the bit rate to get better sound quality.  It's a format that the higher end audiophiles are liking and it's becoming yet another standard for high res files.


Remember, Sony has a HUGE catalog of classic material and I was given the option of paying a little extra for higher resolution than I will do so.  All DSD requires is a DSD DAC, not a SACD player.  That's where Sony messed up was with the player.

 

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.  I haven't heard DSD compared, but others have said it sounds even better, plus it can be converted, but that's an extra step that some don't want to have to do.

 

DSD is direct-stream-digital. It's a crappy format, equivalent to how they made music with the 1-bit buzzer of the original IBM PC, just with a much higher sampling rate and some noise shaping filters, which for all intents and purposes needs to be converted back and forth to PCM at various stages in the production and playback process when any digital processing is needed. 

It's neither as space efficient nor higher-fidelity when compared with similar bit rate PCM streams.

 

Bit-for-bit, PCM results in better quality/higher information density than DSD, and there's even a mathematical proof floating around somewhere on that subject.

 

The advantage of DSD is that it allows for a dirt cheap implementation, which means a lousy implementation won't sound much worse than a great implementation, while PCM is trickier to implement in hardware, which means less-than-stellar PCM playback devices will do bad justice to PCM content, because they don't render the information in the data properly.

 

In short: properly implemented PCM is provably better than DSD, however a lot of PCM devices are not properly implemented and thus suffer from all sorts of quality issues, particularly digital devices created by audio companies that have an "analog" mindset, which tend to solve problems that don't matter in digital audio and overlook the things that do matter.

DSD, thanks to its 1-bit nature, is easier to implement, most of the error gets noise-shaped out of the audible spectrum, and a bunch of decent filters make the whole thing bearable.

 

You can't do any audio processing with DSD without subjecting it to a PCM conversion, or doing the processing in the analog domain after DA conversion.

 

A few links for the curious:

http://www.craigmandigital.com/education/PCM_vs_DSD.aspx

http://en.wikipedia.org/wiki/Direct_Stream_Digital#DSD_vs._PCM

post #213 of 270
Quote:
Originally Posted by rcfa View Post
 

 

DSD is direct-stream-digital. It's a crappy format, equivalent to how they made music with the 1-bit buzzer of the original IBM PC, just with a much higher sampling rate and some noise shaping filters, which for all intents and purposes needs to be converted back and forth to PCM at various stages in the production and playback process when any digital processing is needed. 

It's neither as space efficient nor higher-fidelity when compared with similar bit rate PCM streams.

 

Bit-for-bit, PCM results in better quality/higher information density than DSD, and there's even a mathematical proof floating around somewhere on that subject.

 

The advantage of DSD is that it allows for a dirt cheap implementation, which means a lousy implementation won't sound much worse than a great implementation, while PCM is trickier to implement in hardware, which means less-than-stellar PCM playback devices will do bad justice to PCM content, because they don't render the information in the data properly.

 

In short: properly implemented PCM is provably better than DSD, however a lot of PCM devices are not properly implemented and thus suffer from all sorts of quality issues, particularly digital devices created by audio companies that have an "analog" mindset, which tend to solve problems that don't matter in digital audio and overlook the things that do matter.

DSD, thanks to its 1-bit nature, is easier to implement, most of the error gets noise-shaped out of the audible spectrum, and a bunch of decent filters make the whole thing bearable.

 

You can't do any audio processing with DSD without subjecting it to a PCM conversion, or doing the processing in the analog domain after DA conversion.

 

A few links for the curious:

http://www.craigmandigital.com/education/PCM_vs_DSD.aspx

http://en.wikipedia.org/wiki/Direct_Stream_Digital#DSD_vs._PCM

DSD a crappy format?  Well, better tell some of these high end recording studios that are putting out DSD vs PCM.  And you better tell these high end hardware and software companies like Pyramix, DAD, Light Harmonic, MyTek, 

 

You are relying on Wikipedia?  Wikipedia is great for certain types of information.  All I am saying is that there are a lot of companies that make recording equipment and playback equipment for DSD and it's becoming more popular. Right now, Sonic Studio (which owns Philips' DSD licenses) allows people to have DSD on their Mac and it puts a tag in iTunes so it fools the user into thinking it's there.  It works, but it's not as clean.

 

MacMinis are real popular amongst the Audiophile community for a server.  They are small, relatively cheap, easy to use and Sonic Studio's Amarra only works on OS X.

 

What's dumb is preventing people from using a product.  I look at audio formats just I look at file formats for a anything else.  Doesn't Numbers and Pages recognize the same file formats as Excel and Word?

 

All I am saying is for Apple to allow iTunes to allow the user to import whatever audio format they have.  So it should NOT matter whether Apple sell them or not.  There are a LOT of different ways to get content.  There are some files that are ONLY DSD and yeah, there is a free converter by Korg we can use, but it changes it to PCM.  I haven't used it so I can't comment on if or how much it alters the file in terms of sound quality.  But that's how they have to do it now.

 

Whether people SAY they can tell a difference is one's opinion and they are coming out with higher res DSD,  but it is a format that's out there.

 

FYI, Steve Jobs was a big SACD fan, which uses DSD, only they are pushing DSD even further than what they did with SACD, so people can download these files.

 

There are people that are buying even the portable DSD recording units and recording live concerts and they want to play them back on their system that might comprise of a computer audio server.

post #214 of 270
Quote:
Originally Posted by akqies View Post


As Ppietra notes, your numbers don't add up. The fact that you extrapolated 6 quarters worth of data from just one quarter is a huge red flag.

They add up just fine.

 

Do some actual research into sales data, before you try to talk about extrapolating data. The numbers I quoted for that quarter was accurate, and also was an average of the previous 3 quarters. So, before you just run your mouth off without looking anything up, try thinking next time.

 

Apple sales have been going up. Him extrapolating the last year, then applying it back to previous years doesn't work. If they have a 33% or 50% YoY growth, and they sold 200 million in the last year, then in the previous year they sold 133 million devices.

post #215 of 270
Quote:
Originally Posted by harharhar View Post

They add up just fine.

Do some actual research into sales data, before you try to talk about extrapolating data. The numbers I quoted for that quarter was accurate, and also was an average of the previous 3 quarters. So, before you just run your mouth off without looking anything up, try thinking next time.

Apple sales have been going up. Him extrapolating the last year, then applying it back to previous years doesn't work. If they have a 33% or 50% YoY growth, and they sold 200 million in the last year, then in the previous year they sold 133 million devices.

You said 6 quarters but you only used 1 quarter. You lied to make your point. If you used all 6 quarters the data would have been different.
post #216 of 270
Quote:
Originally Posted by drblank View Post
 
Quote:
Originally Posted by rcfa View Post
 

 

DSD is direct-stream-digital. It's a crappy format, equivalent to how they made music with the 1-bit buzzer of the original IBM PC, just with a much higher sampling rate and some noise shaping filters, which for all intents and purposes needs to be converted back and forth to PCM at various stages in the production and playback process when any digital processing is needed. 

It's neither as space efficient nor higher-fidelity when compared with similar bit rate PCM streams.

 

Bit-for-bit, PCM results in better quality/higher information density than DSD, and there's even a mathematical proof floating around somewhere on that subject.

 

The advantage of DSD is that it allows for a dirt cheap implementation, which means a lousy implementation won't sound much worse than a great implementation, while PCM is trickier to implement in hardware, which means less-than-stellar PCM playback devices will do bad justice to PCM content, because they don't render the information in the data properly.

 

In short: properly implemented PCM is provably better than DSD, however a lot of PCM devices are not properly implemented and thus suffer from all sorts of quality issues, particularly digital devices created by audio companies that have an "analog" mindset, which tend to solve problems that don't matter in digital audio and overlook the things that do matter.

DSD, thanks to its 1-bit nature, is easier to implement, most of the error gets noise-shaped out of the audible spectrum, and a bunch of decent filters make the whole thing bearable.

 

You can't do any audio processing with DSD without subjecting it to a PCM conversion, or doing the processing in the analog domain after DA conversion.

 

A few links for the curious:

http://www.craigmandigital.com/education/PCM_vs_DSD.aspx

http://en.wikipedia.org/wiki/Direct_Stream_Digital#DSD_vs._PCM

DSD a crappy format?  Well, better tell some of these high end recording studios that are putting out DSD vs PCM.  And you better tell these high end hardware and software companies like Pyramix, DAD, Light Harmonic, MyTek, 

 

You are relying on Wikipedia?  Wikipedia is great for certain types of information.  All I am saying is that there are a lot of companies that make recording equipment and playback equipment for DSD and it's becoming more popular. Right now, Sonic Studio (which owns Philips' DSD licenses) allows people to have DSD on their Mac and it puts a tag in iTunes so it fools the user into thinking it's there.  It works, but it's not as clean.

 

[...]

 

There are some files that are ONLY DSD and yeah, there is a free converter by Korg we can use, but it changes it to PCM.  I haven't used it so I can't comment on if or how much it alters the file in terms of sound quality.  But that's how they have to do it now.

 

[...]

 

FYI, Steve Jobs was a big SACD fan, which uses DSD, only they are pushing DSD even further than what they did with SACD, so people can download these files.


No, I don't need to rely on Wikipedia, mathematics speak for themselves. Wikipedia however is pretty good at bundling the references e.g. to the relevant AES papers. As I said, there's a mathematical proof that for any N-bits used PCM captures more information about the audio signal than DSD does. More information means better sound. Period. Information theory says so.

 

DSD is easier to implement, so if you're a consumer electronics company, and you're trying to sell a $20 cost item as a $999.- high-end audio product, then of course you're going to fucking love DSD. If you're trying to produce the best audio for any given amount of bits used and without using massive noise shaping which then means you have to use an analog chain that's bandwidth limited, then you won't use DSD.

 

Further, it's IMPOSSIBLE to do any EQ, level adjustment, compressing (not storage, but audio), etc. without converting the signal to PCM first. In other words, EVERY SINGLE COMMERCIAL (read: major label) DSD product (SACD, etc.) has been AT LEAST ONCE converted to PCM and then back to DSD. Only some rare audiophile purist recordings that are recorded and never processed, mastered, etc. are pure DSD. And while it's possible to convert without loss PCM to DSD, the inverse isn't really true.

 

So yes, DSD is an inferior format. The issue is, that people compare 16-bit/44.1kHz PCM to SACD DSD streams. But if you look at the storage requirement, then a SACD DSD stream requires more bits than 24-bit/192kHz PCM, and just about nobody, in double-blind testing can tell the difference between DSD and PCM when equivalent bit rates are used and the quality of playback hardware is matched and not biased towards one format or the other.

 

The point is: signal processing is pure math. It's not about snake oil or which device has a nicer anodized aluminum front plate or what freak like Levinson quotes what sort of outdated pseudo-science on badly implemented early PCM audio out of context to make DSD look superior: numbers don't lie, and the same say business people. The difference is: scientists look at information content, business people look at profit margins, and those are higher when it comes to DSD. And as long as consumers buy any shit that's hyped well enough and as long as musicians without any training in math or engineering and easily swayed by suggestive marketing, endorsement deals, etc. would pick a badly aligned analog reel to reel tape over a state of the art digital recording equipment, as long you'll find people who claim that DSD is better because it's "not digital" (even though it's if anything more digital than PCM).

 

There are a lot of pseudo-authorities who spew a lot of crap. If you want to check out companies who understand digital, you don't go to Sony or Philips, you go to Meridian, or Metric Halo, or Kurzweil, or... and none of these companies would dream of DSD, they do PCM and they do it for good reason. (They may have DSD compatibility options, but it's on a DSD-to-PCM basis).

post #217 of 270
Quote:
Originally Posted by drblank View Post
 

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?

post #218 of 270
Quote:
Originally Posted by v5v View Post
 

 

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?

Well, first off, you have to have a good enough system and have your ears get accustomed to listening to sonic differences.   But the difference that I hear in a regular 16 bit recording from a RedBook CD on my system that I ripped into my iMac to an AIFF or Apple Lossless compared to the 24/96 version I downloaded from HDTracks was I could hear things more clearly.  I could actually hear the suble sound effects the guitarist used on passages that were low in the mix.  I could hear bass notes with more clarity, the high frequency things like cymbals were nice and smooth and not harsh sounding.  There seems to be better separation between each instrument and not blended together.   I am listening to an album I bought back in the 70's on vinyl and have heard this album literally a few thousand times and know just about ever note by heart.  I used to listen to with headphones back when I first bought the album.  I have various mastering jobs ranging from the original CD mastering, to remastered versions done later to the 24/96 version.  It's just so much clearer there is NO contest.    I could pick it out blindfolded so easily, it's no contest.  Worth the money from my point of view.  I am using a decent DAC that actually up samples the 16 bit version and puts it through an Apodising filter to remove pre-ringing and it's going through decent interconnect cables to a powered set of speakers that only cost around $700.  My total investment in the cables, speakers, and DAC is less than $1800 and I would consider it a very nice low end high fi system.  I'm sure on more expensive, more detailed system it would sound even better, but this is my bedroom setup and I listen to music on it about 4 hours a day on average.

 

40kHz are harmonic frequencies, that's not where you really hear the differences between 16 bit and 24/96 bit even though that's what the curves show you.   Obviously, it's going to depend on a LOT of other factors.  What the higher res files essentially do is make it more analog sounding where the music content is more accurately closer to what it would sound like if it were analog.  Now, there are ultra high end playback systems that have compared just a regular 16bit Redbook CD on a SUPER expensive transport/DAC compared to one of the most expensive and precise turntables/cartridges/phono pre amp on a VERY nice high end system that costs well over over $100K and listeners sat in the room and compared vinyl to 16 bit digital and had a hard time telling the difference.  Digital can sound REALLY good, but it's also part of the mastering job.

 

if they are taking analog tape and converting it using 16 bit AD vs 24 AD, then that conversion is much better and then the playback is much better.

 

All you have to do is get a decent DAC/speaker playback system and try it yourself as you can listen to tracks at HDTracks before you buy them. Same thing with DSD.  There are a handful of high res download sites that have free downloads or samples that you can compare, but obviously, your system, your ears and any room problems will give you the as accurate of listening experience as it will allow.  Actually, you don't want any of the frequencies above a certain level to be too loud as it might mask the note being played.  I don't know if you understand music theory, but when a musician plays a single note on a piano, it plays the fundamental frequency the highest and then all of the harmonics at a gradually lower level (HOPEFULLY), but if you have a upper harmonic that's at the same volume as the fundamental, then you aren't hearing the fundamental clearly, thus it will sound like crap.

 

Now, people like Rupert Neve and others have argued as to what the use of having equipment reproduce frequencies beyond normal human hearing as there is recording and playback equipment is supposed to capture and reproduce beyond 20KHz.   It might make things a little clearer, but you also run the risk of having it too bright/harsh sounding, but that's a whole other discussion.

 

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.

 

Do you know how many AD and DA converters are on the market and how they WILL sound better or worse than the other and it's also has to do with not only the quality of the DAC or ADC chip, but the input circuits, output circuits, isolation of power supply, etc. etc. etc.

post #219 of 270
Quote:
Originally Posted by v5v View Post
 

 

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?

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 and they can achieve such good quality sound from even 16 bit, but they can do even higher than that. Check out what Light Harmonic or any number of these super expensive USB DACs are doing just to get used to what you are SUPPOSED to be hearing. It helps you figure out what to strive for in a playback system and it's possible to get pretty close without having to spend MEGA BUCKS on a system.   Light Harmonic is selling a 24/384 USB DAC for $300 for just headphones and I'm sure that thing sounds great.    Meridian makes a great USB DAC called the Director DAC for playing through a regular stereo, or there are MyTek DACs that will also do DSD, which there isn't much of right now. But DSD is growing, especially if Sony starts releasing their catalog in DSD. I have not heard DSD, so I can't comment on that.  But I've heard people that are involved in producing recordings that swear by it and they've been doing PCM recordings for many years and have experience with both.  Blue Coast Records is doing that and they have free files that you can download to check out and if you don't have DSD playback, you can convert to PCM using Korg's software.  But it also depends on what you want in a system and what kind of music you listen to.  If you listen to a lot of heavily distorted metal that has lots of compression, sound effects, etc. or rap music or mostly computer generated samples, modeled, etc. it may not make much difference.

 

 

Oh, one thing to do, if you really want good audio for headphones, the better sets cost money and stay away from Beats, they are too colored and have too much phony bottom end. Sennheiser makes some of the better headphones and their HD800's a VERY nice if you listen to headphones a lot, it's worth the money in the long run.  There are others, but the HD800s consistently come up as one of the headphones of choice for the audiophile crowd. Yeah, I know, they are expensive. I'd buy them if I had the money, but I don't listen through headphones much.  I prefer traditional speakers.


I also play my music at around 80 to MAYBE 95dB for the loudest passages, anything louder than that, and it's just not good for long term listening.


Edited by drblank - 10/25/13 at 6:04pm
post #220 of 270
Quote:
Originally Posted by drblank View Post
 

[...] 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!)

post #221 of 270
Quote:
Originally Posted by v5v View Post
 

 

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.

 

http://www.youtube.com/watch?v=z1lacX2DJ_U

http://www.youtube.com/watch?v=-vXSR-qgwXs

http://www.youtube.com/watch?v=T3rJjRzfN_0

http://www.youtube.com/watch?v=_LR0SObVo6c

http://www.youtube.com/watch?v=c0_pPjxCbVs

post #222 of 270
Quote:
Originally Posted by drblank View Post
 

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.

post #223 of 270
Quote:
Originally Posted by v5v View Post
 
Quote:
Originally Posted by drblank View Post
 

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...

post #224 of 270
Quote:
Originally Posted by rcfa View Post

[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?

post #225 of 270
Quote:
Originally Posted by v5v View Post
 

 

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?  

post #226 of 270
Quote:
Originally Posted by drblank View Post
 

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.

post #227 of 270
Quote:
Originally Posted by rcfa View Post
 

 

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.  

post #228 of 270
Quote:
Originally Posted by v5v View Post
 

 

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.

post #229 of 270
Quote:
Originally Posted by drblank View Post
 

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 View Post
 

 

 

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.

post #230 of 270
Quote:
Originally Posted by drblank View Post
 
Quote:
Originally Posted by rcfa View Post
 

 

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.

 

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.

post #231 of 270
Quote:
Originally Posted by rcfa View Post
 

 

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.

post #232 of 270
Quote:
Originally Posted by rcfa View Post
 

 

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?  

post #233 of 270
Quote:
Originally Posted by drblank View Post
 

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.

post #234 of 270
Quote:
Originally Posted by drblank View Post
 

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.

post #235 of 270
Quote:
Originally Posted by drblank View Post
 

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.


Edited by v5v - 10/26/13 at 1:33am
post #236 of 270
Quote:
Originally Posted by v5v View Post
 

 

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.

post #237 of 270
Quote:
Originally Posted by v5v View Post
 

 

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.

post #238 of 270
Quote:
Originally Posted by drblank View Post
 

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.

post #239 of 270
Quote:
Originally Posted by v5v View Post
 

 

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.

post #240 of 270
Quote:
Originally Posted by v5v View Post
 

 

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.

 

Based on the way you think, you probably think there is such a thing as a perfect cable. sorry, they don't exist, that's just in THEORY. THEORY is THEORY, but it's not REALITY. I've seen various types of measurements of just analog cables of how it affects an analog signal. But MOST engineers are not trained to do these types of measurements, have the types of equipment they are using and actually understand what can alter the signal which can be measured and heard. 

 

You live in the dark ages, these AD and DA converters are getting better even if they are still doing 16/44.   But they are also seeing vast improvements in taking old analog recordings, converting them with better AD converters and getting better sounding digital files that we can listen to.   Some equipment do a better job at reproducing these files and some people can hear differences.  But not everyone because the vast majority of people can't tell the difference.  have you ever watched a movie with Dolby Digital ACC vs HD Master audio tracks on a decent home theater?   If you can't hear a major difference, then you are deaf.  Dolby Digital is 16/44 and HD Master is 24/96.  Even less expensive equipment it can be a vast difference to even average people. 

 

Obviously, better processors, better amps, cables, speakers will also have an impact as well as the room acoustics, but some people will spend the money on better this or that to get a marginal improvement if they so desired.  Yeah, some make bigger and smaller improvements, but when someone is going to dump $50K or more on a nice home theater (which is becoming more and more popular), they'll spend the money to get those improvements within their budget.  And they'll focus on what yields the biggest bang for the buck.  But like ANYTHING, price vs quality is on a log scale.  Just like cars, bicycles, motorcycles, etc. etc.  to get to a certain level of quality it costs X amount, to get a marginal improvement, the costs start escalating more rapidly.  Some people know this and accept this as reality.


That's why some people WILL spend $6000 on a microphone whereas someone else can't tell the difference with a $100 microphone.  Some just can't afford it.

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