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Gear / Technical Help => Post-Processing, Computer / Streaming / Internet Devices & Related Activity => Topic started by: cpclark on December 28, 2003, 08:35:03 PM
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alright guys, got a couple of questions, i just got CEP2 for doing post matrix's but it seems really dificult to navigate in to move and time stretch the wav's, anyone have any pointers, tips, etc.. on how to get started, ive read the help sections and im just having some trouble getting them to line up right and moving the wav's around, thanks for the help.
-chris
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I have always had problems lining up a long wav, so what I have always done is used the aud wav as the base wav, taking the SBD wave and chopping it up in CD WAV and then lining up by tracks...is ALOT easier to do in CEP and relatively easy to navigate...mixing it down at the end...
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ill try that out tomorrow when i get a chance
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on other suggestion, when you chop up the sbd wav, make sure you do it at noises...so you can line up a "pop" in the waveform...easier that way
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duly noted
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one more: when you chop of the wav into like 4 or 5 pieces, do you just add them as like say wav 2 and then 3 and 4 and so on...
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alright guys, got a couple of questions, i just got CEP2 for doing post matrix's but it seems really dificult to navigate in to move and time stretch the wav's, anyone have any pointers, tips, etc.. on how to get started, ive read the help sections and im just having some trouble getting them to line up right and moving the wav's around, thanks for the help.
-chris
seems like this discussion pops up often. i've been matrixing SBD/AUD, ALD/AUD, AUD/AUD shows for quite some time now on CEP and have achieved nice results. here are few tips that i thought i'd mention:
0. use HEADPHONES, you need to hear the details to matrix.
1. NEVER stretch or squeeze a wav by using pitch control. in all my matrix projects, i have yet to find 2 digital (DAT) recordings that align perfectly initially, but don't stretch or squeeze a source by changing the pitch.
2. choose your "base" wav. i use the AUD source in majority of the cases and align the SBD feed to my "base" wav. LOCK the base wav. it's always easier to manipulate 1 wav (non-base wav).
3. try to align the sources as soon as the show starts. look for short transient sections like a cymbal crash or drummer's stick count. i rarely use long transient sections like a bass drum or long kick drums sounds. also, vocals are not too helpful in syncing. LOCK the non-base wav.
4. once you've achieved #3 above, then listing to the matrix until you start to hear the sources de-sync. depending on your sources, this may take as short as 5 minutes or as long as 30 minutes. once you determine that the 2 sources starts to de-sync every X minutes, you need to start cutting your non-base wav every so often on that interval.
5. try to cut (there's a "CUT Wav" icon for CEP) the wav during crowd noise or a soft non-music passage (the AUD wav will mask the cut usually). since i cut the SBD/ALD sources every time, i can usually cut w/o any noticeable abnormalities. UNLOCK the non-base wav to perform #6 below.
6. once you've made a cut, slide the remainder of the wav to achieve sync. repeat these steps until the end of the show.
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several months ago, a taper buddy sent me 8 Claypool Frog's show (DAT masters) along with their respective SBD sources from their Fall '02 shows.
i recall that one of them gave me one hell of a time to sync. i had to cut the SBD of the 2.5hr show every 5-7 minutes!!! it wasn't fun, but the result was great.
it takes practice and some careful listening for details, but once you get the hang of it, it's quite fun and gets easier very fast.
good luck,
marc
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i just need to know how to slide the wav, i cant get that one figured out and what did you use for time stretch?
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is there a reason why when im playing the wavs in sync CEP would stop playing after 10 or 20 sec.
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this may or may not help...go to the "play" icon in the left corner of the screen. right click...and select "play to end of file" or whatever it is...
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Just curious for those who have tried ... if the initial 4 tracks are all done on a laptop and synced to the same clock, will they be aligned the entire time (assuming the two sources are aligned to begin with)?
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i just need to know how to slide the wav, i cant get that one figured out and what did you use for time stretch?
Just right click on the wave and drag it
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i just need to know how to slide the wav, i cant get that one figured out and what did you use for time stretch?
i highly recommend that NOT using time stretch. i would never change the pitch of a recording. i would cut the wav and slide it + or - to achieve sync.
marc
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If you have a situation where the drift is too much for the 'cutting' method (I have a 5 ms/minute drift) than I recommend trying a stretch in wavelab (high quality, preserve pitch). I tried a few other stretch functions with poor results, but this one sounds very good to me. I also have wavelab temp files set to use 32 bit float.
You have to figure out the proper stretch
- synch up 2 points - one near the beginning and one near the end
figure out the amount of drift over that period of time
- Highlight the same area in wavelab and go to stretch - only use the ratio control - play with it using values to the 3rd decimal (xxx.xxx) until the time displayed is the duration you want to match up with on the other synch point (the slightly longer or shorter one)
- Once you find the best number, use that to stretch the whole file.
- Sometimes there's some trial and error required to zero in on the right value. It's better to be a little longer and edit out the odd millisecond or two than it is to come up short.
I'd like to hear some people's opinon of the wavelab stretch - take a sample and stretch by a factor of 99.980 or something like that. See if you can tell the difference.
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If you have a situation where the drift is too much for the 'cutting' method (I have a 5 ms/minute drift) than I recommend trying a stretch in wavelab (high quality, preserve pitch). I tried a few other stretch functions with poor results, but this one sounds very good to me. I also have wavelab temp files set to use 32 bit float.
You have to figure out the proper stretch
- synch up 2 points - one near the beginning and one near the end
figure out the amount of drift over that period of time
- Highlight the same area in wavelab and go to stretch - only use the ratio control - play with it using values to the 3rd decimal (xxx.xxx) until the time displayed is the duration you want to match up with on the other synch point (the slightly longer or shorter one)
- Once you find the best number, use that to stretch the whole file.
- Sometimes there's some trial and error required to zero in on the right value. It's better to be a little longer and edit out the odd millisecond or two than it is to come up short.
I'd like to hear some people's opinon of the wavelab stretch - take a sample and stretch by a factor of 99.980 or something like that. See if you can tell the difference.
imho, there is no reason to be off by 5ms/min. (this only applies to pure digital sources). if you're off by the amount you state, you should go back further and sync it there; therefor, no need to time stretch.
it's ALWAYS easier (and more practical) to cut a wav than to time stretch + or -.
marc
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imho, there is no reason to be off by 5ms/min. (this only applies to pure digital sources). if you're off by the amount you state, you should go back further and sync it there; therefor, no need to time stretch.
it's ALWAYS easier (and more practical) to cut a wav than to time stretch + or -.
marc
marc - not quite sure what you're saying. Perhaps I wasn't clear:
Without any correction, my 2 sources drift by 5 ms/min. That's just how it is - different A>Ds. Wherever I synch in post, one minute later, one of the sources will be out by 5 ms. My ears don't seem to like the sound when I'm off by more than about 5 ms. That means I would need to cut the wave file of one source every minute, which is a huge amount of work, requires smoothing between samples AND also means that I spend more time out of synch than in it.
On the other hand, if I do a proper, pitch preserving stretch, I can synch up at the start point and end up within a couple of milliseconds an hour later.
What makes the first approach easier and more practical?
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If you have a situation where the drift is too much for the 'cutting' method (I have a 5 ms/minute drift) than I recommend trying a stretch in wavelab (high quality, preserve pitch). I tried a few other stretch functions with poor results, but this one sounds very good to me. I also have wavelab temp files set to use 32 bit float.
You have to figure out the proper stretch
- synch up 2 points - one near the beginning and one near the end
figure out the amount of drift over that period of time
- Highlight the same area in wavelab and go to stretch - only use the ratio control - play with it using values to the 3rd decimal (xxx.xxx) until the time displayed is the duration you want to match up with on the other synch point (the slightly longer or shorter one)
- Once you find the best number, use that to stretch the whole file.
- Sometimes there's some trial and error required to zero in on the right value. It's better to be a little longer and edit out the odd millisecond or two than it is to come up short.
I'd like to hear some people's opinon of the wavelab stretch - take a sample and stretch by a factor of 99.980 or something like that. See if you can tell the difference.
I use AA2, but I concur...esp with the "99.980" statement. 32bit and high quality, preserve pitch. I also lock and sync to whatever I consider my better source (rig/deck)
Nicely written.
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Just curious for those who have tried ... if the initial 4 tracks are all done on a laptop and synced to the same clock, will they be aligned the entire time (assuming the two sources are aligned to begin with)?
If it's on the same clock then you're fine. BTW, I use one of these:
http://www.cascademedia.net/products.asp?catid=129&prodid=524
It helps me with my 32 track set up...
cm