They announced that two days ago, so no, it most definitely has not been saying that longer than three weeks. I'm part of the beta team, it really is nearing completion. I can't say anything other than that due to NDA.
They're great aside from their total lack of any sort of decent support... A reload shouldn't take a week and a half, a KVM setup because they fucked the reload up shouldn't take 4 days. They should be monitoring tickets much more actively, I've sent in a reboot ticket with them once that took 3 hours. Basically, they're a bunch of incompetents.
I moved all of my stuff over to http://www.softlayer.com/ which has a much better support system, much friendlier staff, and much better infrastructure in general. It's pricier, but hey, you get what you pay for.
Supporting this, I've had absolutely zero problems using iTunes and Quicktime on Vista, though I only sync my iPod using my MBP, so I don't know what the issues are there.
I have to say that iTunes is butt ugly on Vista, much like it was on XP. Hopefully they'll at least make some use of the compositing engine in the next one to make it fit in somewhat better.
Gotta agree with the AC here. They're incredibly reliable and very fast when it comes to any updates. They also give you incredible tools for managing multiple domains.
Disclaimer: I am a registered eNom reseller, but there's a reason why. They really do handle things very well.
I think you are missing something. I don't know if you actually work in the music business or not, and if you do and you use open source stuff, more power to you!
What I was saying with the Pro Tools thing was a large generalization. Almost all of the major studios out there use Pro Tools, and because of that market share, most of them will continue to use Pro Tools until something else comes out that interfaces 100% and does a better job. There are already plenty of tools out there that do a better job, but none of them interface perfectly with existing Pro Tools projects. As for quality? Hell, I haven't seen Pro Tools run for much longer than 4 or 5 hours without crashing. "Standard" and "Best" are not the same thing, as I think Microsoft has proven... =P
There are a lot of other great manufacturers, I was just naming a few of them. However, I must say RME isn't a brand that I've ever heard of until now, but looking at the specs and such they look like good hardware.
This still doesn't really affect the main argument though, which is that in the current state of things there isn't any software out for Linux that can do a great job competing with the main players. Hopefully some day there will be, but it'll take some time... Either that or one of them will get a Linux port made. The thing is, working in the music industry, if you aren't using Pro Tools, you really might as well not be in the music industry. It's sad but it's true.
Have you ever worked in a professional audio environment or even seen a studio? I have. You probably haven't. I've recorded over 10 tracks simultaneously in "Windoze" as you put it, with 1 millisecond audio latency. Absolutely no problem. I've also done the same on OSX, but that's besides the point.
There is NOTHING for Linux or BSD or whatever you want to use that comes close to the quality of the four programs I mentioned. Period. Go use a few of them, then come back to me and tell me to use [insert Linux DAW here]. You won't get any proper AU or VST support in Linux, which pretty much couns you out for using any mastering software. As for hardware support? Good luck getting any of the professional audio devices (Read: Echo, Mackie, MOTU, Digidesign) to work to their full capacity. You'll be lucky if you get 200 millisecond audio latency.
You seem to get off on simply insulting people instead of posting anything relevant. Do me a favor, go to a real studio, then come back after you've had some practical experience. Thanks.
As a side note, pretty much anyone who spells Windows "Windoze" or Microsoft "Micro$oft" or whatever the hell else have you loses all credibility right away. Look, ha ha, he tried to make a funny. Isn't that so clever?
Since when did you have to use Obj-C to write a program in Java? Did someone not get past printf? Obj-C is the standard programming language for Cocoa. I don't know where you even got Java from, he didn't mention it anywhere. Hardly anyone uses the Java Cocoa bindings for OSX programming...
There's nothing that's truly professional quality, which is sad, but that's the state of things. Cubase, Logic, Sonar, and Pro Tools are the four standards.
However, there IS one fairly good program for UNIX type systems, though it's nowhere near the quality of the four I mentioned above. Have a look at Rosegarden.
Well, whois can be faked, but the full proof is that the sun.com nameservers (ns1, ns2, ns7, and ns8.sun.com) handle DNS for sun.de, and also if you use nslookup to look it up from the sun nameservers, they show themselves as authoritative and resolve to the same IP. So as you said, yep, this is legit.
Heh, replied to the AC one just a few minutes before you did. This has been posted on slashdot and many other places hundreds of times in the past 6 or 7 years... When the man actually passes away, I'm sure it will make headlines. But for now, it hasn't and it won't because he hasn't passed away.
Haha, I like the comparison to an R2 unit convention. Yea, the digital I/O would solve one problem but it wouldn't solve the problem of recompressing it.
Speaking of R2 units, it's always good fun when I'm recording and my cell phone starts doing its thing because I forgot to turn the transmitter off.
Your hearing isn't as good as mine then. My point is that the loss of data gets compounded when you recompress it, and has nothing at all to do with the signal path. My point has everything to do with the simple fact of how MP3 compression works.
I don't care if you're even just running LAME on an MP3 file to recompress it without any signal path at all! There IS A LOSS OF QUALITY. That is how MP3 encoding works. It removes overtones that the human ear GENERALLY won't hear, and then does its best to recreate them at playtime. Each time you recompress it, more overtones are removed. That is a fact, that is how the compression works.
You're absolutely correct about the loss of quality in the sound hardware itself - there is hardly any at all. The loss of quality that I'm talking about is the loss of quality in the actual MP3 encoding algorithm, and there is no way at all to avoid that no matter how good your audio hardware is, as the audio hardware is completely irrelevant. I'm talking about computer programs, not audio hardware.
I'm not trying to nullify your point with my other reply to this, because your point is completely valid... Go load up a Linkin Park or some other pop type band's CD in an audio editor, you'll cringe - it's nothing but clipping. I'm just saying that it doesn't really have much to do with the kind of compression I'm talking about.
That's a completely different type of compression - dynamic range compression as opposed to data compression. One is to reduce and even out the volume differences, and is used quite a bit in recording - for example an acoustic guitar or an electric solo you'll want to run through a compressor. Yes, you do lose some sound quality by overcompressing, as in the case of most pop CDs, but definitely not the same kind of data loss that takes place in an MP3 compression.
But see, from exactly what you said, it removes overtones that you most likely can't hear. It doesn't remove them all though. What will happen as you recompress it is that more of the overtones/harmonics will be wiped out each time you do it. Give it a try, you'll see what I mean after 5 or 6 runs if you have really good hearing.:)
I wish that were the case, however, no, there was nothing incorrect about what I stated. What happens when you play an mp3 is that it is rendered into PCM wave files based on data that it doesn't have but can reproduce to a certain degree of accuracy. When you recompress an mp3, say burning it to a CD and then ripping it again, you have burned raw PCM data that was generated by the mp3 codec from the mp3. This in itself is already lossy, since the file is of the same quality as the mp3. When you go back in and recompress that PCM data, you're compressing already lossy data by removing even more data.
It really is about equivalent to opening a jpeg - which rasterizes it into a raw bitmap - then saving that raw bitmap with compression, then opening it again, and saving it. Programs these days are smart enough with jpegs not to even save it if no changes have been made, however if the jpeg is saved, it's by definition recompressed. Likewise with mp3, aac, whatever, unless you're doing a bit by bit copy of the original file - which is just a copy, not recompression, then you're not getting the same quality of audio. However, if you do what the original poster said and I replied to, then yes, there is a quality loss, because you're outputting to an audio stream, and capturing and recompressing that audio stream.
As for removing the DRM from an audio file, the only way to do that without losing sound quality is to decrypt the file. Playing it and recompressing it is by nature going to lose quality, as well using an internal hook and recompressing it.
Yea, very few people can really hear the difference. I happen to be one of those people, which is good in some ways but sucks in others. I can tolerate MP3 audio, but I'd much rather listen to an uncompressed CD.
I'm also not one of those people who claims to be an audiophile then goes ahead and decimates the recording by running it through a tube amp. That simply cracks me up - tubes, by their very nature, color the sound.
Rereading your crap about all internal and not using an audio loop, you still fail to grasp the concept of recompression. Do me a favor. Go open up a jpeg image, and save it. Then open it up again after closing it, and save it again. Do this about 20 times. You'll see some very clear artifacting. This SAME EXACT process applies to re-encoding mp3s/whatever.
Unless you're ripping the ORIGINAL STREAM by simply decrypting the encrypted stream, you are re-encoding and lose quality. Unless of course you want to convert your lossy DRMed files into raw PCM audio, in which case you'll have whatever sound your computer reproduced from the mp3 algorithm. See, the whole idea behind mp3 and any lossy format is to remove less important data in such a way that it can be reproduced fairly accurately without having that original data. This is why it is called LOSSY.
And for the record, I'm pickier than you. I have a full studio and the speakers I use are worth about $1800, and I have a fairly high quality audio interface as well. You may not hear the difference on whatever speakers you use, however, I would hear the difference on my setup. Hell, I can hear when a note is off by about 5 cents. Perfect pitch, ever heard of it?
Now, shut up, because honestly you have no idea what you're talking about at all.
Slashdot Mirror
They announced that two days ago, so no, it most definitely has not been saying that longer than three weeks. I'm part of the beta team, it really is nearing completion. I can't say anything other than that due to NDA.
"Within three weeks" isn't some form of timeframe for release?
Anyone else find the spokesman's name hilarious?
Kwak Bumjoon!
They're great aside from their total lack of any sort of decent support... A reload shouldn't take a week and a half, a KVM setup because they fucked the reload up shouldn't take 4 days. They should be monitoring tickets much more actively, I've sent in a reboot ticket with them once that took 3 hours. Basically, they're a bunch of incompetents. I moved all of my stuff over to http://www.softlayer.com/ which has a much better support system, much friendlier staff, and much better infrastructure in general. It's pricier, but hey, you get what you pay for.
Supporting this, I've had absolutely zero problems using iTunes and Quicktime on Vista, though I only sync my iPod using my MBP, so I don't know what the issues are there.
I have to say that iTunes is butt ugly on Vista, much like it was on XP. Hopefully they'll at least make some use of the compositing engine in the next one to make it fit in somewhat better.
Gotta agree with the AC here. They're incredibly reliable and very fast when it comes to any updates. They also give you incredible tools for managing multiple domains.
Disclaimer: I am a registered eNom reseller, but there's a reason why. They really do handle things very well.
I think you are missing something. I don't know if you actually work in the music business or not, and if you do and you use open source stuff, more power to you!
What I was saying with the Pro Tools thing was a large generalization. Almost all of the major studios out there use Pro Tools, and because of that market share, most of them will continue to use Pro Tools until something else comes out that interfaces 100% and does a better job. There are already plenty of tools out there that do a better job, but none of them interface perfectly with existing Pro Tools projects. As for quality? Hell, I haven't seen Pro Tools run for much longer than 4 or 5 hours without crashing. "Standard" and "Best" are not the same thing, as I think Microsoft has proven... =P
There are a lot of other great manufacturers, I was just naming a few of them. However, I must say RME isn't a brand that I've ever heard of until now, but looking at the specs and such they look like good hardware.
This still doesn't really affect the main argument though, which is that in the current state of things there isn't any software out for Linux that can do a great job competing with the main players. Hopefully some day there will be, but it'll take some time... Either that or one of them will get a Linux port made. The thing is, working in the music industry, if you aren't using Pro Tools, you really might as well not be in the music industry. It's sad but it's true.
Ah, good point. Overlooked that one. =P
Have you ever worked in a professional audio environment or even seen a studio? I have. You probably haven't. I've recorded over 10 tracks simultaneously in "Windoze" as you put it, with 1 millisecond audio latency. Absolutely no problem. I've also done the same on OSX, but that's besides the point.
There is NOTHING for Linux or BSD or whatever you want to use that comes close to the quality of the four programs I mentioned. Period. Go use a few of them, then come back to me and tell me to use [insert Linux DAW here]. You won't get any proper AU or VST support in Linux, which pretty much couns you out for using any mastering software. As for hardware support? Good luck getting any of the professional audio devices (Read: Echo, Mackie, MOTU, Digidesign) to work to their full capacity. You'll be lucky if you get 200 millisecond audio latency.
You seem to get off on simply insulting people instead of posting anything relevant. Do me a favor, go to a real studio, then come back after you've had some practical experience. Thanks.
As a side note, pretty much anyone who spells Windows "Windoze" or Microsoft "Micro$oft" or whatever the hell else have you loses all credibility right away. Look, ha ha, he tried to make a funny. Isn't that so clever?
There's nothing that's truly professional quality, which is sad, but that's the state of things. Cubase, Logic, Sonar, and Pro Tools are the four standards.
However, there IS one fairly good program for UNIX type systems, though it's nowhere near the quality of the four I mentioned above. Have a look at Rosegarden.
You must work for Verizon... 21000000000000 bytes is 19.09 terabytes, not gigabytes.
Well, whois can be faked, but the full proof is that the sun.com nameservers (ns1, ns2, ns7, and ns8.sun.com) handle DNS for sun.de, and also if you use nslookup to look it up from the sun nameservers, they show themselves as authoritative and resolve to the same IP. So as you said, yep, this is legit.
Heh, replied to the AC one just a few minutes before you did. This has been posted on slashdot and many other places hundreds of times in the past 6 or 7 years... When the man actually passes away, I'm sure it will make headlines. But for now, it hasn't and it won't because he hasn't passed away.
My guess would be 10.4.10. ;-)
Gotta love the AC that calls the developer of the application the article's talking about a fanboi.
Haha, I like the comparison to an R2 unit convention. Yea, the digital I/O would solve one problem but it wouldn't solve the problem of recompressing it.
Speaking of R2 units, it's always good fun when I'm recording and my cell phone starts doing its thing because I forgot to turn the transmitter off.
Bzzt bzzt. bzt. bzzzztt. "CRAP!"
Your hearing isn't as good as mine then. My point is that the loss of data gets compounded when you recompress it, and has nothing at all to do with the signal path. My point has everything to do with the simple fact of how MP3 compression works.
I don't care if you're even just running LAME on an MP3 file to recompress it without any signal path at all! There IS A LOSS OF QUALITY. That is how MP3 encoding works. It removes overtones that the human ear GENERALLY won't hear, and then does its best to recreate them at playtime. Each time you recompress it, more overtones are removed. That is a fact, that is how the compression works.
You're absolutely correct about the loss of quality in the sound hardware itself - there is hardly any at all. The loss of quality that I'm talking about is the loss of quality in the actual MP3 encoding algorithm, and there is no way at all to avoid that no matter how good your audio hardware is, as the audio hardware is completely irrelevant. I'm talking about computer programs, not audio hardware.
My apologies for trolling you as an AC earlier.
I'm not trying to nullify your point with my other reply to this, because your point is completely valid... Go load up a Linkin Park or some other pop type band's CD in an audio editor, you'll cringe - it's nothing but clipping. I'm just saying that it doesn't really have much to do with the kind of compression I'm talking about.
That's a completely different type of compression - dynamic range compression as opposed to data compression. One is to reduce and even out the volume differences, and is used quite a bit in recording - for example an acoustic guitar or an electric solo you'll want to run through a compressor. Yes, you do lose some sound quality by overcompressing, as in the case of most pop CDs, but definitely not the same kind of data loss that takes place in an MP3 compression.
But see, from exactly what you said, it removes overtones that you most likely can't hear. It doesn't remove them all though. What will happen as you recompress it is that more of the overtones/harmonics will be wiped out each time you do it. Give it a try, you'll see what I mean after 5 or 6 runs if you have really good hearing. :)
I wish that were the case, however, no, there was nothing incorrect about what I stated. What happens when you play an mp3 is that it is rendered into PCM wave files based on data that it doesn't have but can reproduce to a certain degree of accuracy. When you recompress an mp3, say burning it to a CD and then ripping it again, you have burned raw PCM data that was generated by the mp3 codec from the mp3. This in itself is already lossy, since the file is of the same quality as the mp3. When you go back in and recompress that PCM data, you're compressing already lossy data by removing even more data.
It really is about equivalent to opening a jpeg - which rasterizes it into a raw bitmap - then saving that raw bitmap with compression, then opening it again, and saving it. Programs these days are smart enough with jpegs not to even save it if no changes have been made, however if the jpeg is saved, it's by definition recompressed. Likewise with mp3, aac, whatever, unless you're doing a bit by bit copy of the original file - which is just a copy, not recompression, then you're not getting the same quality of audio. However, if you do what the original poster said and I replied to, then yes, there is a quality loss, because you're outputting to an audio stream, and capturing and recompressing that audio stream.
As for removing the DRM from an audio file, the only way to do that without losing sound quality is to decrypt the file. Playing it and recompressing it is by nature going to lose quality, as well using an internal hook and recompressing it.
Yea, very few people can really hear the difference. I happen to be one of those people, which is good in some ways but sucks in others. I can tolerate MP3 audio, but I'd much rather listen to an uncompressed CD.
I'm also not one of those people who claims to be an audiophile then goes ahead and decimates the recording by running it through a tube amp. That simply cracks me up - tubes, by their very nature, color the sound.
Rereading your crap about all internal and not using an audio loop, you still fail to grasp the concept of recompression. Do me a favor. Go open up a jpeg image, and save it. Then open it up again after closing it, and save it again. Do this about 20 times. You'll see some very clear artifacting. This SAME EXACT process applies to re-encoding mp3s/whatever. Unless you're ripping the ORIGINAL STREAM by simply decrypting the encrypted stream, you are re-encoding and lose quality. Unless of course you want to convert your lossy DRMed files into raw PCM audio, in which case you'll have whatever sound your computer reproduced from the mp3 algorithm. See, the whole idea behind mp3 and any lossy format is to remove less important data in such a way that it can be reproduced fairly accurately without having that original data. This is why it is called LOSSY. And for the record, I'm pickier than you. I have a full studio and the speakers I use are worth about $1800, and I have a fairly high quality audio interface as well. You may not hear the difference on whatever speakers you use, however, I would hear the difference on my setup. Hell, I can hear when a note is off by about 5 cents. Perfect pitch, ever heard of it? Now, shut up, because honestly you have no idea what you're talking about at all.