Does rendering degrade WAV's?

[mjau]

whenever i render a track i save it as 32 bit float, because, well, why not? and they totally null so i've stopped worrying about it.

Spoken by someone with no understanding of what digital is. One wonders why MoreSpaceEcho bothers posting here.

Also, MoreSpaceEcho has a small dick.

[farview]

You'll need to explain this more. A wordclock would have no effect on conversion.

The wordclock only effects conversion and syncronization. An ITB render doesn't need to be sync'd with anything and time is irrelevent because it is just rendering one word after another as fast as it can (unless your DAW is set to render in real time). The clock only comes into play during recording and playback, when the samples need to be taken or played at a steady rate.

[farview]

At one point I tried the polarity trick with some consolidated tracks. They nulled out. On the flip side, I am always paranoid that all of the inaudible content that's getting f'ed with in such a process (rendering) adds up to a perceptible difference. Having said that, I try not to let my paranoia influence my decision making too much as it usually squashes the creative process somewhere along the line moreso than the original fear.

-J

When you render, you are just making a file of what you are listening to. The original track has been edited, crossfaded, etc... and when you listen to that edited track, the computer is performing that edit, crossfade, etc... in real time while you listen to it. When you render, you are just making a file with that already done, so the computer doesn't have to do it on the fly anymore. It's the same thing. The only parts that will be any different from the original file will be the ones that you changed, ie. the fades, edits, etc...

There is no degradation when copying digital tracks, and that is what you are doing with a render. (at least to the parts you didn't edit)

[RefD]

whenever i render a track i save it as 32 bit float, because, well, why not? and they totally null so i've stopped worrying about it.

Spoken by someone with no understanding of what digital is. One wonders why MoreSpaceEcho bothers posting here.

Also, MoreSpaceEcho has a small dick.

and a lisp!

[inverseroom]

whenever i render a track i save it as 32 bit float, because, well, why not? and they totally null so i've stopped worrying about it.

Spoken by someone with no understanding of what digital is. One wonders why MoreSpaceEcho bothers posting here.

Yeah geez!!!!! I was thinking the same thing!!!! I mean please!!!! Who does he think he is, anyway, somebody people actually like?!!!!?

[ubertar]

A post like that should get Jeff banned from TapeOp. But then, who would we have to make fun of?

[Girl Toes]

Here is a scenario, imagine Audiosuiting a track in Pro Tools. The file will be processed with the desired plugin regardless of the clock source for the session. Essentially, this would be like rendering a file to .wav. When the file is played back a given device or program, it needs a wordclock or a clocked source to set the sample rate.

Man, I hate it when people prefer the sound quality of today to the sound quality of 45 years ago.

[MoreSpaceEcho]

A post like that should get Jeff banned from TapeOp. But then, who would we have to make fun of?

precisely.

jeff apparently is BOOKED SO FUCKING SOLID at his studio that he has time to follow me around on here and attack me for every stupid post i make.

so, again jeff, please tell everyone here what is wrong with me saving my rendered files as 32 bit float.

[dubold]

Spoken by someone with no understanding of what digital is. One wonders why MoreSpaceEcho bothers posting here.

Jeff, I'd like you to explain what is wrong with MoreSpaceEcho's statement, or retract what you said and apologize. Are you claiming that you have seen him flip polarity on files he's working on, and they do not, in fact, null?

Your statement seems unnecessarily cranky, and I would speculate that it's because you guys had a difference of opinion in the thread about syncing two PT LE systems together.

In the spirit of professionalism, why don't we try to just answer John's question without resorting to personal attacks?

[wayne kerr]

Here is a scenario, imagine Audiosuiting a track in Pro Tools. The file will be processed with the desired plugin regardless of the clock source for the session. Essentially, this would be like rendering a file to .wav. When the file is played back a given device or program, it needs a wordclock or a clocked source to set the sample rate.

Audiosuite is not a verb, dicknose.

[wayne kerr]

Spoken by someone with no understanding of what digital is. One wonders why MoreSpaceEcho bothers posting here.

Jeff, I'd like you to explain what is wrong with MoreSpaceEcho's statement, or retract what you said and apologize. Are you claiming that you have seen him flip polarity on files he's working on, and they do not, in fact, null?

Your statement seems unnecessarily cranky, and I would speculate that it's because you guys had a difference of opinion in the thread about syncing two PT LE systems together.

In the spirit of professionalism, why don't we try to just answer John's question without resorting to personal attacks?

dude! wtf! no fun at all in that.

*pops top on coors light... adjusts la-z-boy*

[RefD]

*anticipates own far less inflammatory posts being deleted*

[Ryan Silva]

whenever i render a track i save it as 32 bit float, because, well, why not? and they totally null so i've stopped worrying about it.

Spoken by someone with no understanding of what digital is. One wonders why MoreSpaceEcho bothers posting here.

The following should bring some clarity:

The most important thing to importing or exporting audio is going to be first, the sample rate and bit rate it was recorded at and second, the bit rate and sample rate it will be played back at.

Taking tracks from one system to another will mean that the wordclock will be all-important. Ken Pohlmann cites in his 'Principles of digital audio' that the most crucial stage for any audio in the digital domain is having the proper sampling rate at the time of conversion from analog to digital.

Rendering tracks for someone to work on brings with it the possibility of having a different clock or one that is better or worse than the one you started with. Maintaining the same clock source throughout a project should not just desirable, but mandatory.

Remember too, the samples before and after will be same, but they may be played at a different or slightly different rate.

Here is a scenario, imagine Audiosuiting a track in Pro Tools. The file will be processed with the desired plugin regardless of the clock source for the session. Essentially, this would be like rendering a file to .wav. When the file is played back a given device or program, it needs a wordclock or a clocked source to set the sample rate.

Alright, what the hell is going on with these guys? That's 3 posts I?ve read today in which @?,*???&? bags on everything More Space Echo says, with no apparent reason. I noticed that they joined the very same day; I smell an alter-ego here. Maybe a Jekyll and Hyde.

Anyway, sorry to get off topic just thought it was interesting.

[thom yorke]

whenever i render a track i save it as 32 bit float, because, well, why not? and they totally null so i've stopped worrying about it.

Spoken by someone with no understanding of what digital is. One wonders why MoreSpaceEcho bothers posting here.

I'm so high right now.

[trodden]

whenever i render a track i save it as 32 bit float, because, well, why not? and they totally null so i've stopped worrying about it.

Spoken by someone with no understanding of what digital is. One wonders why MoreSpaceEcho bothers posting here.

The following should bring some clarity:

The most important thing to importing or exporting audio is going to be first, the sample rate and bit rate it was recorded at and second, the bit rate and sample rate it will be played back at.

Taking tracks from one system to another will mean that the wordclock will be all-important. Ken Pohlmann cites in his 'Principles of digital audio' that the most crucial stage for any audio in the digital domain is having the proper sampling rate at the time of conversion from analog to digital.

Rendering tracks for someone to work on brings with it the possibility of having a different clock or one that is better or worse than the one you started with. Maintaining the same clock source throughout a project should not just desirable, but mandatory.

Remember too, the samples before and after will be same, but they may be played at a different or slightly different rate.

Here is a scenario, imagine Audiosuiting a track in Pro Tools. The file will be processed with the desired plugin regardless of the clock source for the session. Essentially, this would be like rendering a file to .wav. When the file is played back a given device or program, it needs a wordclock or a clocked source to set the sample rate.

Alright, what the hell is going on with these guys? That's 3 posts I?ve read today in which @?,*???&? bags on everything More Space Echo says, with no apparent reason. I noticed that they joined the very same day; I smell an alter-ego here. Maybe a Jekyll and Hyde.

Anyway, sorry to get off topic just thought it was interesting.

Please don't insult my friend morespaceecho.