Noel Borthwick [Cakewalk]
We had a very thought provoking discussion at NAMM with Henry Juszkiewicz, Gibson CEO about how much more important the lower frequencies were rather than the HF that people obsess about. He's really clued in to this stuff and approaches it from a perspective that factors in the physiology of the human ear.

 
I think a lot of people realize this instinctively and that is why there are so many high quality preamps on the market that offer adequate power supplies that can translate the details and nuance of low frequency dynamic response.
 
 
With regards to people who speak of the notion of "subsonics", they are simply spreading misinformation. Here in Tallahassee we have a gentleman who has developed a "woofer" that can produce low notes down to 1Hz. Once you hear discrete sine waves such as 1Hz, 4hz, 10Hz etc. played back on his woofer you realize that people who describe a hearing threshold at the low frequency simply do not know what they are speaking about. 
 
 
edit grammar 

He is. And it's true, 20hz is an approximation, some people would get down a bit further under ideal conditions. However, whatever Mike thinks he's heard, he has not heard a 1hz sine wave.

Well, I keep circling back to how good all of the music from the late 60's and 70's still sounds, having had none of the level of technology that exists today, and regardless of the mathematics and research, and whatever, I am going to simply use 48 K, at 24-bit depth, and do the most I can within that - including spending quality time to create quality compositions.
 
The above is only an expression of how I choose to proceed, at this point.
 
Bob Bone
 

LOL... just imagine if you could sit Mozart, Bach, Beethoven in front of X3!
 
No wait... after the initial euphoria, they would say is "What's up with the Staff View????"
 
Edit: That figures... Noel posts as I am typing that  [duck and cover]

Noel Borthwick [Cakewalk]

Jeff Evans
I have just mastered a very high quality Jazz CD. The mixes came to me at 24 Bit 88.2 Khz and yes they sound great. But even after mastering and dithering down to 16 Bit 44.1Khz the difference between the two sounding formats is so miniscule. It is really inaudible in fact. I did a little test and just changed down the original mixes from their original format to 16 Bit 44.1KHz and did no mastering, just compared the two and believe me I would challenge anyone to hear anything between them! The issue is the music is just so good and the moment I hear either of those formats I am off listening to the playing so once that happens I think there is no hope of hearing anything much else!

I had a similar experience with the last project we recorded. It was tracked in PT at 96K, mixed at 96K in SONAR and mastered to at  96K and finally downsampled to 44.1 for the CD master. I did a test loading up both a 44.1K and 96K track in SONAR (in a 96K project). Then I phase inverted the two tracks and bounced the result down to a new track. I was surprised with the result. What was left was some noise that was completely inaudible even if I cranked the volume all the way up :)
So unless something went wrong during the capture or mastering process that I was unaware of, this probably means that there was nothing significant in the recording that took advantage of the 96K range. I can't generalize here since its definitely possible that you may get different results with other music that has more HF content - however in my case this didn't make anyy diffference for all  practical purposes at all.

 I am not surprised, Noel.  

Noel Borthwick [Cakewalk]
 
I had a similar experience with the last project we recorded. It was tracked in PT at 96K, mixed at 96K in SONAR and mastered to at  96K and finally downsampled to 44.1 for the CD master. I did a test loading up both a 44.1K and 96K track in SONAR (in a 96K project). Then I phase inverted the two tracks and bounced the result down to a new track. I was surprised with the result. What was left was some noise that was completely inaudible even if I cranked the volume all the way up :)
So unless something went wrong during the capture or mastering process that I was unaware of, this probably means that there was nothing significant in the recording that took advantage of the 96K range. I can't generalize here since its definitely possible that you may get different results with other music that has more HF content - however in my case this didn't make anyy diffference for all  practical purposes at all.

 
That hardly seems like a logical way to draw a conclusion. :-S
 
I happen to work at 44.1 and 48, so I'm have no compelling reason to justify any other choice, however I am bemused by a lack of veracity in the explanations offered as justification for conclusions being shared.
 
It seems like an actual comparison would require a protocol featuring side by side capture, mixing, 
mastering, (all at the respective specifications) a final export to the distribution specification, and perhaps a digital to analog conversion.
 
If you arrive at the same conclusion after that, your conclusion will be based on an actual comparison rather than an implication that a comparison was made.
 
 
 
 
BTW, I think one of the more entertaining scaremonger phrases in the Science article is
 
"To people who study these things, it’s become clear that doubling and quadrupling sample rates instead of improving converters is a lousy economic tradeoff for consumers, as well as for the environment and the larger economy." 
 
 
I'm curious about the reasoning that supports this scientific statement:
 
"There is also the separate problem of distortion caused by the decreased sampling accuracy of a rate that is too fast."
 
Too fast for what? I'm just guessing here; Is this some sort of implication that the current implementation of clocking in some of the high speed converters is not suitable for the task? Are we really to believe that this issue is inherently insurmountable or is it more probable that this may be the case in specific implementations?
 
Those are the types of statements presented in that article that seem like a load of hooey.
 
 
 
best regards,
mike
 
 

Hence this too from one of my much earlier posts:
 
Real World now. I have created an AB test session where a very high quality analog signal (finest turntable, vinyl,pickup, RIAA equaliser etc) was sent to one side of an AB switch. That same signal was bottle necked through 16 bit 44.1K A to D and D to A and fed to the other side of the switch. Volume differences between the two switch positions were removed totally. Even with expert engineers and very high quality monitoring (and environment) many had no idea what they were listening to. I did this based on this article:
 
http://mixonline.com/reco...emperors_new_sampling/
 
This is test is rather interesting. The pure analog turntable signal represents the finest quality signal we can generate in the analog domain. (easily anyway) The best vinyl too are the Sheffield Lab recordings from the 80's where bands played live, were fully mixed and this signal fed direct to the cutting lathe avoiding all tape based stuff in between. You have to hear this to believe it. I have got an album with Dave Grusin featuring Lee Ritenour, Ron Carter and Harvey Mason. It sounds like you are standing right in front of Harvey's kit! It is a real eye opener in how fast the whole groove stylus thing can be when it needs to be. You need the finest headshell, cartridge (Shure V15 Type III) tone arm and RIAA preamp. (Music Fidelity, $1000 for a RIAA EQ!) I have got all this setup. (Dark Side of the Moon sounds ridiculous on this setup too even to this day)
 
Another source of super analogue would be a two track reel to reel machine playing back a very high quality master. I have got that too and it works exactly the same way. (I produced a really fine acoustic folk CD and did digital hard drive mixes but also mastered at the same time direct to a high speed mastering reel to reel machine. You have to hear that too to believe it!)
 
You would need 382Khz and 24 bit probably to represent it the best. It is hard to tell what you are listening to when you do this test. Analog or 16 bit 44.1KHz digital.  It demonstrates very clearly that even 44.1K 16 bit represents things amazingly well.
 
I discovered something amazing recently. I was in a studio recording with musicians for two days straight recently and during a break one of the guys told me about the album 'Captain Fingers' from Lee Ritenour. It is the most amazing mix and it was all on vinyl. I am amazed at how modern this sounds. Sounds like it could have been done yesterday. On the finest turntable it must be incredible. I am keen to track it down. The music is amazing too. The sound of this record is so wide and extended frequency response wise and so transient it is jaw dropping. A fine example of the medium not holding anything back in any shape or form. What you hear is the vision from some amazing producer and mix engineer.

 

8 bit version (two fingers):
http://www.youtube.com/watch?v=vYA4V81KIxE

http://www.youtube.com/watch?v=NJkbVAs8rjo
 
<click> <click> just go with the music, yeah.

"Evil rock music impresario Lars Mangros puts hypnotic signals in the music done by his best-selling band Andromeda that incites young fans to riot and rebel against authority. Andromeda are going to perform their first big concert which will be broadcast on a global basis. It's up to Buck Rogers and Twiki to thwart Mangros' dastardly plot to warp the minds of the youth all over the universe during said concert."
 
Amazing.
 

mike_mccue
It seems like an actual comparison would require a protocol featuring side by side capture, mixing, 
mastering, (all at the respective specifications) a final export to the distribution specification, and perhaps a digital to analog conversion.
 
If you arrive at the same conclusion after that, your conclusion will be based on an actual comparison rather than an implication that a comparison was made.
 

  
 
I didn't have the luxury of doing a side by side recording so couldn't do that test. However I'm curious why you consider my test to have been not valid. I compared a downsampled version with the original 96K recording by phase inverting snd mixing the two wave files. If there was something  special  in the original 96K recording the phase invert and mix test should have pulled out just the differences in the two recordings, right? I'm no mastering expert so I'm happy to be corrected here.
 
BTW I agree that the some of statements in that article were a bit over the top like the one about it being bad for the environment :)

The guy writes in a semi-humorous style. I can think of worse crimes.

Noel Borthwick [Cakewalk]

mike_mccue
It seems like an actual comparison would require a protocol featuring side by side capture, mixing, 
mastering, (all at the respective specifications) a final export to the distribution specification, and perhaps a digital to analog conversion.
 
If you arrive at the same conclusion after that, your conclusion will be based on an actual comparison rather than an implication that a comparison was made.
 

  
 
I didn't have the luxury of doing a side by side recording so couldn't do that test. However I'm curious why you consider my test to have been not valid. I compared a downsampled version with the original 96K recording by phase inverting snd mixing the two wave files. If there was something  special  in the original 96K recording the phase invert and mix test should have pulled out just the differences in the two recordings, right? I'm no mastering expert so I'm happy to be corrected here.

I don't see anything to correct, unless you're testing whether something recorded, mixed, and mastered at 44.1kHz is going to sound better than something recorded, mixed, and mastered at 96kHz or recorded, mixed, and mastered at 96kHz and then downsampled to 44.kHz. Your test gives 44.1kHz the "benefit of the doubt" by feeding it the [supposedly] higher-quality 96kHz source material.
 
One area where people might not take matters into account is the output filtering. I don't know if it changes for different sample rates, but for 44.1kHz, you have to brickwall pretty heavily (e.g., 96dB/octave) to keep the clock out of the output. With 96kHz, you can use a 48dB/octave filter and obtain even a bit more rejection.
 
In a comparison test at AES several years back between a 30 ips analog master tape and PCM reproductions at various sample rates as well as DSD, to me (and others in the audience) DSD sounded more like the reference music tape than 44.1kHz PCM. I've wondered if it's the technology, the fact that DSD can use gentler output filters, or something else altogether.

The article Noel linked to in post number 1 of this thread describes what Craig has referred to as "output filtering" under the headings:
 
Improvements at 44.1: Fixing the Filters
 
and
 
When More is Better: Making The Filters Even Better