Bill, I hate burst your bubble but your waveforms are being shown in connect the dot fashion, as if the samples were infnitely small and the bandwidth was infinitely big.

If you take the data that you graphed and convolute it against a sinx/x impulse response (i.e. a perfect brickwall reconstruction filter) you will get exactly the same resulting analog information in both cases.

For the audio band both of your samples contain EXACTLy the same data, its just stored differently.

Try oversampling both. Somewhere around 8x oversampling they will both look very similar. At 20x oversampling they will be virtually identical.

Sorry, but you were highlighting a limit of your graphing software not the sampling systems.

Oversampling is the function of the D/A converter actually. This graphic was based on the recorded wav from an acoustic standpoint.

I took a single Bruel&Kjaer microphone and recorded it into the wave editor from loudspeaker source at 2 sampling fq and word lenghts.

This is the result I got.

The actually O scope was very simalar. Perhaps I need to bring this up to the scope and do an actual digital photo of it.

This file was mono.

Stereo is much worse. The mic was 1/2 inch from the source so the room is a non issue.

I see major differences in wave shape of a solo clarinet between these capacities.

The loudspeakers have no choice than to follow the output signal. This is the tracking of the speaker actually.

You have more points of resolution..in all forms including dynamics.

I see your point...I hear the difference.

Expound?

Consider this: You shouldn't be able to tell the difference between an 8kHz sine and an 8kHz square... the lowest added harmonic would be the 3rd, which would occur at 24kHz, beyond the theoretical range of audibility.

If you could test this aurally, There would be too many other parts of the chain that can cause intermodulation and in-band differences. Even removing A/Ds and D/As from the chain, leaving only an oscillator, an amplifier and a loudspeaker.

If the loudspeaker produces any amount of IM distortion -and ALL do- then you have to consider the amount of in-band artifacts that are added... think of it as a sort of in-band aliasing.

Now... still convinced that your tests are valid?

You may hear a difference. But it is not a result of what you may think it is.

Respectfully...

Brent,

> Nobody is disputing that 16-bit sucks. <

Okay, I will. :D

Cassettes suck. RealAudio over a modem sucks. But 16 bits at 44.1 can sound very good.

All of your other points are right on the money. And Steve Devino's comment that Bill was showing the limit of his software is exactly on the money. Bill, even if you graph it properly it doesn't matter how poor the waveform looks. It's easy to create waves that sound identical but look very different due to simple phase shift.

--Ethan

It fascinates me how much emotional energy people put into arguing about things they've never even done. I've been working at 96k for almost 2 years now, and I hate working any other way. It is smoother, cleaner, and easier to work with my tracks at 96. Take something as simple and everyday essential as putting some 16k high-end presence into a vocal, for example. I won't do it at 48k. It doesn't do much except make sibilance more shrill and hard. But at 96, it will make a vocal pop right out. You won't hear it when you first use the higher sample rates, but after a while, you can't imagine working without them. Quit bitching and experiment.

I have and do record at 96kHz. I record at 192kHz as well, just to say that I did. It is a fad and a necessity with classical artists. On a scope I do see the differences, but again they were inaudible to the naked ear, so why reproduce them?

Again, if you had problems adding 16kHz in your 48kHz system, it was more than likely a crapola eq plug in and/or a crapola converters. I have a Pro Tools HD rig. I have heard the Prisms and the Genex converters. Because it was my money, I bought the Digi 192's and stuck with them. There wasn't that much difference. If you heard a difference, it was not the sample rate that made that difference. Read my other posts.

What system do you use, and do you find the internal dithering sufficient, or do you dither at the masterer? How is 96kHz easier to record with?

I suspect that we will never resolve this issue with any certainty because we all hear differently. I don’t have the greatest hearing anymore, but I absolutely hear the difference between analog and digital.. I have had the opportunity to conduct a test of this. I have told this story here before at least twice, so if this all seems like deja’ vu all over again.. 2” 24 track into a MCI 600 console, Apogee PSX 100 converter at 96k with the Apogees analog outs patched back to the console 2 track return so I could switch between the analog and digital playback. Even at 96k the difference between analog and digital was very apparent.. I could also switch the sample rate on the Apogee to 24 bit and 16 bit - 48 and 44.1.. each step was a revelation. Soundstage collapse and a sense of a ceiling being lowered. The sound just becomes more closed in. Not so much a difference in high end response just more of a loss of depth and dimension. That’s what I was able to hear.. And yes, my wife really can hear above 20k. Kurt

I agree 100% that higher res imaging is going to happen. After all, we are better representing the waveforms by sampling more often. But once it dithered down to 16-bit again, how much do you retain?

I sure wish that the recording industry could drive the format mobile for a while.

What I have heard (not with my ears but what I have been told) is that if the higher frequencies are captured, they actually generate harmonics in the lower frequencies.. therfore creating a more complex wave in the audible regions. That was simplified so that I could understand it, but that is the basics of it. Now on the other hand, I can say that recording to 24 bit at 44.1 and then brought down to 16 bit 44.1 produces a superior product. I know that, I have heard it. Do you disagree with that? Kurt

Steve has a Pro Tools HD system in his studio with the HD192 converters - so I'm guessing that comparing high and low F(s) is something that he's had a chance to do. I haven't personally been able to hear a difference. I did hear a Masterlink at 96kHz and it sounded great - but so did the CD that came out of it, as well.

I _CAN_ tell the difference between 24-bit and 16-bit, but not between F(s). I just went the the audiologist, as well - no hearing loss, just tinnitus. Can still hear up to 17,500, depending on the amount of gain applied :p . I've got some high end rolloff starting around 8kHz, but not anything ridiculous. So, I'm not working with broken equipment here :p .

Dan Roth
Otitis Media

Damn, where you been oldtimer?

WOW it is good to see one of the first posters and members of RO here.

Welcome again pal!

(feeble voice) I remember when Chris (audiokid) started this site...back then we were all talking about PT 5.1.0 ....

Heh - been trying to make some CASH! Haven't found it yet...

Dan

In my own tests I have done between all the available sample rates I have came to one conclusion. I don't like even hearing that high of a frequency it's not musical in any way.
Sooo... who cares? I think companies just keep
pushing the envelope on these things to try and be on the top of the technology chain really thats all Protools does I mean it's just not musical. So big deal if you can hear that high
do you like what you hear?That is the true question.

Pro Tools itself is as musical as what you put into it. I like the linearity of digital compared to the somewhat unpredictible results of analog tape.

Dan Roth
Otitis Media

Ok a couple of more comments.

1. Bill the fact that you can still see the difference on the Oscope is a function of the 20MHz bandwidth of the scope.
Put a perfect brickwall filter between the scope and the source and all your signals will look identical (as long as you comensate for sinx/x loss. The oversampling I was talking about would be applied in the digital domain by doing an inverse FFT with 8 to 20x the bandwidth of the original signal.

2. My main point is that the difference you hear is due to something other than the sample rate. Like the analog input path or the analog reconstruction, or the fIR filters used in the decimation at the output of the converters etc.

3. Kurt, everyone can hear the difference between analog and digital, because analog (meaning tape in your case) is a much lower "quality " signal. Many people happen to prefer this lower quality signal, but that is a matter of personal taste.

4. I have owned a great 192ksps system for almost a year now. I will be very happy to charge someone 2x my normal rate to record at 192ksps. I personaly cannot hear the difference between 44.1 and 192 k on the 192i/o. An I have a very good monitoring environment. (Dan has been here he knows). Rather my 192i/o sounds better than my older stuff at all sample rates because of improvements in the clock and analog input.

Steve

Steve, you have just written what I wanted to read
LOL

...The textronix is not 20MHZ, it is 2.2GHZ.

Calibration instrument. I have a lab, not a home environment.

Loudspeakers and acoustics as well.

Kurt, everyone can hear the difference between analog and digital, because analog (meaning tape in your case) is a much lower "quality " signal. Many people happen to prefer this lower quality signal, but that is a matter of personal taste.

Steve,
That doesn't explain why my analog source sounded worse when it was put through digital.??? And the lower the sample rate went the worse it sounded. I wasn't the only person that heard this. There were three other people in the room that heard the same thing. Kurt

Music is organic. Analog is organic. Digital is toxic waste.

:p:

Is anybody gonna stay home to watch the fireworks?

Kurt, what converters and clocking are you using?

Lowdbrent- I am talking about adding eq in the analog realm. I don't mix dig. And I'm talking about AD-8000 SE and Mytek conversion.

If you are doing classical music, I assume you aren't doing that many tracks, and that you don't manipulate the sound much. So you probably don't bump up against it as much. Also, in my experience, the Digi 192 is still not a great converter and doesn't show the differences as well the Mytek does.

Again, I am saying that you have to live with it extensively and you have to be manipulating the sound to really understand the kind artifacting/damage occurring at 48 that is absent at 96. It is not just about how high up you can hear. It is about the inherent limitations of working at 48, where filtering and aliasing are happening right in our touchy-feely, feel it in the air, is something happening, upper hearing range.

>Analog is organic

Uh - sorta. So's BS. Analog works by induction - there's inherent non-linearities and accuracy limitations to the format. Sounds F'in great sometimes, depending on the equipment.
Digital is much more linear and accurate than analog. The difference that most people hear between digital and analog systems have to do more with the quality of the clock signal sent to the digital system, and the analog path in front of both the A/D and D/A.

You wouldn't record on an A827 without first biasing it, right? Well, why would you try to get away with a digital system without clocking it with a high quality clock? We all want to have a selection of the finest preamps, be they Neve, Telefunken, Great River, etc. - why would you use a cut-rate A/D/D/A with cheap analog components and poorly implemented decimation and reconstruction filtering?

This is silly - if you know what the f*ck you're doing, the format doesn't matter as much as you'd think. If the band is good and I have good pre's and mics, I'm sure I could make a 4-track cassette that had a good sound and vibe. I could probably do the same to a Nomad jukebox. Sheesh.

Dan Roth
Otitis Media

Dan Roth makes an excellent point. My favorite CD I ever recorded is one by a Blues cat named Jackie Payne. Jackie used to sing for Johnny Otis. They cut some tracks for Arhoolie Records. The CD I am speaking of however was recorded for JSP Records. The performances were impecable. Some great players played on this project, Kenny "Blue" Ray, Rob Sutteth (Huey Lewis), Austin De Lone. I listen to this CD with great satisfaction all the time. We recorded it on Blackface ADATS, 16 bit. What you record is so much more important than how you record it. Amen, Dan.... Kurt

Can I get a hallelujah? The thing is, even if you can tell a difference between the 16/44 and 24/192, if you did a blind test I bet you can't tell which is which...

BTW - Analog doesn't sound better than digital. Some analog components sound better than some digital components, and some digital components sound better than some analog components. I hate when people make that assumption because it is entirely dependant upon the equipment that's in place.

/rant

Originally posted by Kurt Foster / Fats:

Kurt, everyone can hear the difference between analog and digital, because analog (meaning tape in your case) is a much lower "quality " signal. Many people happen to prefer this lower quality signal, but that is a matter of personal taste.

Steve,
That doesn't explain why my analog source sounded worse when it was put through digital.??? And the lower the sample rate went the worse it sounded. I wasn't the only person that heard this. There were three other people in the room that heard the same thing. Kurt

It only sounds worse because you and many others don't like the sound. The fidelity of the signal is much better in digital than analog, that is a fact.

It is also a fact that many people prefer the noisier, harmonically enhanced (i.e. lots of extra harmonic distortion), sound of analog. It is your choice and your taste. I for one think I can get digital to sound better than analog, but either I have no taste or I just have it figured out.

Originally posted by Bill Roberts:
...The textronix is not 20MHZ, it is 2.2GHZ.

Calibration instrument. I have a lab, not a home environment.

Loudspeakers and acoustics as well.

My point was your input bandwidth to the scope is much greater than 20 kHz. Most of us audio designer types use the 20 MHz BW limiter on the inputs to the old Tek 465 scopes because we did not care about bandwidth beyond 20 MHz.

If you put your waveforms through a perfect brickwall filter they will all look identical, even on your 2 GHz scope. This is fact and is part of the entire nyquist sampling system.

The difference between sound between various systems is the quality of the clocks and the reconstruction filters. At 192KSPS it is easier and cheaper to make a low pass filter that is phase linear out to 20 kHz. That is why it might sound better to some people.

Steve

Guys, this is audiophile fanatic dogma and we all know it---the listening public--the ones who buy all the records-- cannot tell and dont care too--ever.

If they are perfectly satisfied with mp3's do you really think they care about 96k?

I believe it is truly the insecurity and deficencies in individuals skills that cause them to never stop talking about better gear ect.

Some love to say---(I can hear a huge difference)--just to kinda say(I have great dog ears)--its a boast.

Heart put out a bunch of all DDD albums when the technology was nothing like it is right now and had umpteem hits with them---it just doesnt matter. If your good--your good--so if you think 48 more hz's will put you over the top--dream on and dream until your dreams come true:)

Smooth you ROCK!!!!!!!!!!
I love your attitude finally someone who feels the way I do!

Heheh my first post here!

I picked up this quote off the digido website and it brings up a point not yet adressed in this thread.

"Recently, another advance in the audio art was introduced, a digital equalizer which employs double-sampling technology. This digital equalizer accepts up to a 24-bit word at 44.1 kHz (or 48K), upsamples it to to 88.2 (96), performs longword EQ calculations, and before output, resamples back to 44.1/24-bits. I was very skeptical, thinking that these heavy calculations would deteriorate sound, but this equalizer won me over. Its sound is open in the midrange, because of demonstrably low distortion products. The improvement is measurable and quite audible, more...well... analog, than any other digital equalizer I've ever heard. This confirms the hypothesis of Dr. James A. (Andy) Moorer of Sonic Solutions, "[in general], keeping the sound at a high sampling rate, from recording to the final stage will...produce a better product, since the effect of the quantization will be less at each stage". In other words, errors are spread over a much wider bandwidth, therefore we notice less distortion in the 20-20K band. Sources of such distortion include cumulative coefficient inaccuracies in filter (eq), and level calculations."

It seems like he's saying that a wider bandwidth will spread digital errors/distortion out into the inaudible range instead of having them concentrated in 20-20 so that is a reason why it will sound better. Perhaps someone more knowledgeable than me could confirm or refute this?

96k sounds better to me (Hedd and Digi 192 for A/D, monitoring through a DAC-1), so I work at 96k, but if I had to do everything at 44.1 I wouldn't be tearing my hair out or anything.

Man that was a really great first reply dude
I have to say though that it still doesn't matter
to the "untrained ear" society accepts whatever
the industry gives them and that is the simple truth if society didn't large format studios would still be doing amazing in the business
and not worrying about the next project rolling through the doors I am only speaking these thing through expierience also. As long as there are dipshits in the world that think ogg and mp3 sound as good as the original it doesn't really matter at all because that is what people make
standard hell they even make mp3 players and that's including Apple the company that pretty much owns the recording industry right now
let's face it Protools and DP3 are probably the best in the digital world right? and you can truly only do anything on macs with either one
well you can on protools with both but who can afford a ripass protools rig? So even Apple is capitalizing off the MP3 movement. So the lesson for everyone today is DON'T LET THE INDUSTRY ALLOW NEW AUDIO STANDARDS THAT DEGRADE A WHOLE OTHER STANDARD!

PEACE AND LOVE IN THE MIDDLE EAST
AND EVERYONE PRAY FOR MY SOON TO BE WIFE WHO IS OVER THERE NOW HELPING OTHER PEOPLE HAVE THE SAME FREEDOMS WE DO LIKE SITTING HERE ON OUR BUTTS
BICKERING ABOUT SAMPLERATES

THANKS GOD

chris, i hope your soon to be wife is ok. im sure she is man.