I agree completely. I bitched about that same thing in a thread in another article a while back, pointing out that the version of sendmail that ships with stable is older than sand and the version of postgresql is the horrific 6.5 and that Debian required quite a bit of tarballing as a result of the ancient versions of the software represented by the packages in stable, but not surprisingly enough my post was modded down as "troll". Obviously by some clueless individual who doesn't understand that the debianized sendmail segfaults under high loads due to a problem that was long since fixed in the 11.x sendmail series. It's essentially the same thing with postgresql, 6.5 is an unstable piece of crap, 7.1 is an incredible RDBMS. But me pointing out this and providing facts somehow constituted a "troll".
"Troll"? I administer scores of Debian boxes every day at work and Debian is my distribution of choice. Are the statements I made in that message untrue?
Every time I have to tarball Sendmail on a debian box because the debianized version is ancient, I feel like Debian has "gone out of business". It's not like sendmail is the only package that is at least a major version behind. I typically have to tarball Postgresql as well, since the Debian package is 6.5 (which is crap) and 7.x runs very well. Sure, the package maintainers backport security patches, but they don't backport general bugfixes. I remember when a recent proftpd security fix for the glibc globbing bug came out. The package maintainer didn't bother with a new package and a non-maintainer upload was used for the updated package. The binary distributed with the package was compiled incorrectly which essentially rendered the software nonoperable. It took a week or two before the regular package maintainer corrected the issue.
In short, Debian is a great distribution if you don't mind tarballing the major stuff.
Since we are regurgitating bugtraq posts: although Guninski's exploit did not work on FreeBSD, the same race issue that was used in his exploit was purportedly found in several places in the FreeBSD code. Due to the somewhat generic nature of the issue, I suspect this may appear in more unix-type OSs as well.
I was thinking the same thing... if this guy starts on the unix-type OSs with the same fervor that he has demonstrated on MS systems over the last year, things could get interesting (and way more secure). This guy is an exploit whirlwind.
-
My biggest complaint about CD's is the 16 bit part. Irregardless of the encoding scheme, soft sounds take a beating. Perhaps when they create/implement an audio DVD standard we can all be happy.
-
I agree with you there. I just read an article in an audio trade rag that said quite a few bands are starting to record both at DVD-A bit and sample rates and are also starting to make 5.1 mixes. The major labels are supposedly specifically requesting DVD-A versions of new albums, or at least full compliance with such. So maybe higher bit depth in the consumer audio market is actually making some headway.
The SB16 definitely had a digital mixer. I just verified this on Creative's web site.
The SB Live can use 48 KHz on its output.
The sample rate is not related to the quality achieved by internal operations (volume, effects). You are confusing bit depth with sample rate.
Where are you getting this stuff from? The sample rate has absolutely nothing to do with digital mixing. Digital mixing has been used in sound cards since the SB16. In addition, I know of no sound card that does 96 KHz internally and has only 44.1 KHz output. I really have no idea where this stuff is coming from.
Jesus, I am not talking about consumer grade stuff. The point behind the move to 88.2 and 96 KHz in pro recording is a known fact, the artifacts exist. I cannot possibly explain it any clearer, it's like arguing the answer to 1+1. The reason consumer sound cards are moving to 96 KHz is because it's next to impossible to get a modern convertor that does less than 24 bit/96 KHz.
The problem is with the analog low pass filter's ability to remove material right on the edge of the audio spectrum. It either attenuates some of the desired audio or doesn't filter enough (it's not a brick wall). Getting the entire filtering process out of the edge of the audio spectrum is a primary point behind 96 KHz.
I can't explain it much simpler. Essentially, the sampling process at 44.1 KHz results in artifacts in the audible frequency range. Although filter techniques largely eliminate this, artifacts do still remain. Sampling at 96 KHz moves the artifacts out of the audible range, thus requiring no mucking-around in the audible spectrum.
Not the sneeze, the Zippo! Christ, the sound could be picked up from outside the building with the way some people sneeze. As for the Zippo, assuming that the microphones were actually directed at the players, and located within reasonable proximity of the playing area, I question the ability of modern microphone technology (which really hasn't had any noteable advanced in 30 years) to capture in great detail a low amplitude sound that potentially originates from outside of the microphone known area of sensitivity.
24 bits is better than 16 bits because the noise floor drops with 24 bits, thus you can record at a lower level and thus have greater dynamic range.
maru
Re:I am afraid it is you who are mistaken
on
Insanely Audiophile
·
· Score: 1
I am saying that every aspect of what you are listening to is manually controlled. Frequency response, dynamics, imaging. And a set of audiophile-grade speakers are not generally used as a reference. Assuming that what the tracks sound like in the studio's control room is "correct" (since that is the sound the engineer and producer are looking for), an audiophile system generally is not playing back the "correct" sound. It has reproduced the sound different from what the people who manually control what the track sounds like have chosen.
The exception to this, and the only point behind an audiophile system, would be to play back audiophile-specific recordings. These are made by putting a stereo paired microphone or two microphones in a room with "desired" acoustics and then a dry recording is made and this is what is released.
The point behind an increase in sample rate past 44.1 KHz is because of the interaction of harmonics on the brick-wall filtering that is taking place.
Audio frequencies above half the sample rate (Nyquist frequency) are filtered, but to avoid a particular type of aliasing the frequencies above the Nyquist frequency are translated to frequencies below, typically in the 5-8 KHz range. This creates intermodulation distortion. By increasing the sample rate to 88.2 or 96 KHz, this aliased range is moved out of the frequency spectrum in which the original source's audio information will reside, which eliminates intermodulation distortion.
It's not an audiophile thing, it's real.
You are insane if you think modern recordings can capture the level of detail you describe without the signal being tweaked all to hell at some point during the mixing/mastering process. Now about that transparency?
maru
Re:super sounding gear that isn't that expensive
on
Insanely Audiophile
·
· Score: 1
Right. Sony is making half the gear that the albums you will be playing are recorded with but somehow they are relegated to mid-class on the playback side. How exactly does that work?
Yet another audiophile who claims that the more expensive systems sound "different" but mentions nothing about whether the system sounds better.
Spend your money where it counts. On improving the acoustics of your listening area. This will make a more significant difference than any of the other items you mention, unless you are just trying to change the sound of the sound for the sake of saying it sounds different, which seems to be the objective of many of the audiophiles whose messages I have read here.
Of course there is a difference between the different "audiophile" cables. That's because the same audiophiles who think an equalizer is the product of satan, and work towards the perfectly flat frequency response through only mechanical means, use speaker cables that typically have inductive networks in them.
Of course a major difference between live and recorded music is due to the physical environment. If you are playing back studio-recorded tracks and getting a live sound then your listening area is screwed.
You may hear a difference, but one is not necessarily better than the other. They are just that: different. It's the same reason that a recording studio carries a variety of microphones. They all sound different. Since music is generally mixed for a normal grade playback system, an audiophile system essentially re-tunes the music to something that the piece was not originally intended to sound like.
Slashdot Mirror
I agree completely. I bitched about that same thing in a thread in another article a while back, pointing out that the version of sendmail that ships with stable is older than sand and the version of postgresql is the horrific 6.5 and that Debian required quite a bit of tarballing as a result of the ancient versions of the software represented by the packages in stable, but not surprisingly enough my post was modded down as "troll". Obviously by some clueless individual who doesn't understand that the debianized sendmail segfaults under high loads due to a problem that was long since fixed in the 11.x sendmail series. It's essentially the same thing with postgresql, 6.5 is an unstable piece of crap, 7.1 is an incredible RDBMS. But me pointing out this and providing facts somehow constituted a "troll".
maru
==
CiscoChick: Look, Cisco is a Real Company(tm) that makes Real Money(tm). We have no use for your amateurish --
==
In light of Cisco's recent $2.5 billion excess inventory writeoff, the "makes Real Money" part of CiscoChick's line may need to be modified.
maru
"Troll"? I administer scores of Debian boxes every day at work and Debian is my distribution of choice. Are the statements I made in that message untrue?
Crap! I certainly did not intend to post at +1, what kind of dumbass would design it so you have to explicitly turn off +1 posting each time you post?
Every time I have to tarball Sendmail on a debian box because the debianized version is ancient, I feel like Debian has "gone out of business". It's not like sendmail is the only package that is at least a major version behind. I typically have to tarball Postgresql as well, since the Debian package is 6.5 (which is crap) and 7.x runs very well. Sure, the package maintainers backport security patches, but they don't backport general bugfixes. I remember when a recent proftpd security fix for the glibc globbing bug came out. The package maintainer didn't bother with a new package and a non-maintainer upload was used for the updated package. The binary distributed with the package was compiled incorrectly which essentially rendered the software nonoperable. It took a week or two before the regular package maintainer corrected the issue.
In short, Debian is a great distribution if you don't mind tarballing the major stuff.
I am sure "the real geeks" don't account for any noteworthy percentage of red hat's revenue.
=
Seems to me like some people expect perfection from the OpenBSD crew
=
Have you ever seen any of Theo's posts in bugtraq?
maru
Since we are regurgitating bugtraq posts: although Guninski's exploit did not work on FreeBSD, the same race issue that was used in his exploit was purportedly found in several places in the FreeBSD code. Due to the somewhat generic nature of the issue, I suspect this may appear in more unix-type OSs as well.
I was thinking the same thing... if this guy starts on the unix-type OSs with the same fervor that he has demonstrated on MS systems over the last year, things could get interesting (and way more secure). This guy is an exploit whirlwind.
maru
-
My biggest complaint about CD's is the 16 bit part. Irregardless of the encoding scheme, soft sounds take a beating. Perhaps when they create/implement an audio DVD standard we can all be happy.
-
I agree with you there. I just read an article in an audio trade rag that said quite a few bands are starting to record both at DVD-A bit and sample rates and are also starting to make 5.1 mixes. The major labels are supposedly specifically requesting DVD-A versions of new albums, or at least full compliance with such. So maybe higher bit depth in the consumer audio market is actually making some headway.
maru
The SB16 definitely had a digital mixer. I just verified this on Creative's web site.
The SB Live can use 48 KHz on its output.
The sample rate is not related to the quality achieved by internal operations (volume, effects). You are confusing bit depth with sample rate.
maru
Where are you getting this stuff from? The sample rate has absolutely nothing to do with digital mixing. Digital mixing has been used in sound cards since the SB16. In addition, I know of no sound card that does 96 KHz internally and has only 44.1 KHz output. I really have no idea where this stuff is coming from.
naru
Jesus, I am not talking about consumer grade stuff. The point behind the move to 88.2 and 96 KHz in pro recording is a known fact, the artifacts exist. I cannot possibly explain it any clearer, it's like arguing the answer to 1+1. The reason consumer sound cards are moving to 96 KHz is because it's next to impossible to get a modern convertor that does less than 24 bit/96 KHz.
maru
The problem is with the analog low pass filter's ability to remove material right on the edge of the audio spectrum. It either attenuates some of the desired audio or doesn't filter enough (it's not a brick wall). Getting the entire filtering process out of the edge of the audio spectrum is a primary point behind 96 KHz.
maru
I can't explain it much simpler. Essentially, the sampling process at 44.1 KHz results in artifacts in the audible frequency range. Although filter techniques largely eliminate this, artifacts do still remain. Sampling at 96 KHz moves the artifacts out of the audible range, thus requiring no mucking-around in the audible spectrum.
maru
Not the sneeze, the Zippo! Christ, the sound could be picked up from outside the building with the way some people sneeze. As for the Zippo, assuming that the microphones were actually directed at the players, and located within reasonable proximity of the playing area, I question the ability of modern microphone technology (which really hasn't had any noteable advanced in 30 years) to capture in great detail a low amplitude sound that potentially originates from outside of the microphone known area of sensitivity.
maru
24 bits is better than 16 bits because the noise floor drops with 24 bits, thus you can record at a lower level and thus have greater dynamic range.
maru
I am saying that every aspect of what you are listening to is manually controlled. Frequency response, dynamics, imaging. And a set of audiophile-grade speakers are not generally used as a reference. Assuming that what the tracks sound like in the studio's control room is "correct" (since that is the sound the engineer and producer are looking for), an audiophile system generally is not playing back the "correct" sound. It has reproduced the sound different from what the people who manually control what the track sounds like have chosen.
The exception to this, and the only point behind an audiophile system, would be to play back audiophile-specific recordings. These are made by putting a stereo paired microphone or two microphones in a room with "desired" acoustics and then a dry recording is made and this is what is released.
maru
The point behind an increase in sample rate past 44.1 KHz is because of the interaction of harmonics on the brick-wall filtering that is taking place.
Audio frequencies above half the sample rate (Nyquist frequency) are filtered, but to avoid a particular type of aliasing the frequencies above the Nyquist frequency are translated to frequencies below, typically in the 5-8 KHz range. This creates intermodulation distortion. By increasing the sample rate to 88.2 or 96 KHz, this aliased range is moved out of the frequency spectrum in which the original source's audio information will reside, which eliminates intermodulation distortion.
It's not an audiophile thing, it's real.
maru
You are insane if you think modern recordings can capture the level of detail you describe without the signal being tweaked all to hell at some point during the mixing/mastering process. Now about that transparency?
maru
Right. Sony is making half the gear that the albums you will be playing are recorded with but somehow they are relegated to mid-class on the playback side. How exactly does that work?
maru
Yet another audiophile who claims that the more expensive systems sound "different" but mentions nothing about whether the system sounds better.
Spend your money where it counts. On improving the acoustics of your listening area. This will make a more significant difference than any of the other items you mention, unless you are just trying to change the sound of the sound for the sake of saying it sounds different, which seems to be the objective of many of the audiophiles whose messages I have read here.
maru
Of course there is a difference between the different "audiophile" cables. That's because the same audiophiles who think an equalizer is the product of satan, and work towards the perfectly flat frequency response through only mechanical means, use speaker cables that typically have inductive networks in them.
maru
Of course a major difference between live and recorded music is due to the physical environment. If you are playing back studio-recorded tracks and getting a live sound then your listening area is screwed.
maru
You may hear a difference, but one is not necessarily better than the other. They are just that: different. It's the same reason that a recording studio carries a variety of microphones. They all sound different. Since music is generally mixed for a normal grade playback system, an audiophile system essentially re-tunes the music to something that the piece was not originally intended to sound like.
maru