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Gear / Technical Help => Post-Processing, Computer / Streaming / Internet Devices & Related Activity => Topic started by: Fatah Ruark (aka MIKE B) on June 12, 2005, 01:35:20 AM
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Well, I just mixed up the Mule show from the Fox last Feb. So I figured I would document to process (at least my process), so others could learn how to make their own matrix recordings.
This is my first attempt to do a tutorial, so if I could get some comments from both people that are familiar with making a matrix as well as those that aren't, that would be cool. I'm not the greatest teacher, so I want to make sure this is easy to understand.
Here is the webpage I made:
http://sloppyart.bravehost.com/matrix.html (http://sloppyart.bravehost.com/matrix.html)
If everything looks good and a MOD wants to put a link to the page in the archive, that would be cool.
Hopefully that helps some people. Not sure if there's anything else like this out there but I figured I'd do my own anyway.
MIKE B
EDIT: To update link.
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Looks good +T
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Looks good, when I get ready to render it, I always check"create a specific file". If you do that, the option bar should not say wav pcm 16/44.1 32 Bit Float. THis can cause problems when opening up in a programn such as cd wav The 32 bit float option should be changed to just standard 16. Interesting you mention splitting the sbd into tracks, I have never done that but will try it. Ive done 4 so far now, ,and plan to do this every time, mix the two sources seperately, instead of doing the matrix on the fly. I noticed too, on this last recording, I had to line up the SBD file just a few ms BEFORE the AUD source, so that it would line up right. Does that make sense??Even though the waveforms are lined up, maybe the word clocks in the two different devices are different, so you would have to stagger em a bit??Anyway, thanks again man. +T
Ray
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Looks good, when I get ready to render it, I always check"create a specific file". If you do that, the option bar should not say wav pcm 16/44.1 32 Bit Float. THis can cause problems when opening up in a programn such as cd wav The 32 bit float option should be changed to just standard 16. Interesting you mention splitting the sbd into tracks, I have never done that but will try it. Ive done 4 so far now, ,and plan to do this every time, mix the two sources seperately, instead of doing the matrix on the fly. I noticed too, on this last recording, I had to line up the SBD file just a few ms BEFORE the AUD source, so that it would line up right. Does that make sense??Even though the waveforms are lined up, maybe the word clocks in the two different devices are different, so you would have to stagger em a bit??Anyway, thanks again man. +T
Ray
exactly, when recording into 2 different devices, you are dealing with 2 different word clocks.
Unless you have a sbd >v3> and aud>V3 then you have the same word clock on both recordings.
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Looks good, when I get ready to render it, I always check"create a specific file". If you do that, the option bar should not say wav pcm 16/44.1 32 Bit Float. THis can cause problems when opening up in a programn such as cd wav The 32 bit float option should be changed to just standard 16. Interesting you mention splitting the sbd into tracks, I have never done that but will try it. Ive done 4 so far now, ,and plan to do this every time, mix the two sources seperately, instead of doing the matrix on the fly. I noticed too, on this last recording, I had to line up the SBD file just a few ms BEFORE the AUD source, so that it would line up right. Does that make sense??Even though the waveforms are lined up, maybe the word clocks in the two different devices are different, so you would have to stagger em a bit??Anyway, thanks again man. +T
Ray
exactly, when recording into 2 different devices, you are dealing with 2 different word clocks.
Unless you have a sbd >v3> and aud>V3 then you have the same word clock on both recordings.
So then it would be necessary not to line the wavforms up exactly, but rather to stagger them slightly , with the "tape based" recorder slightly in front of the HD based one, from time to time.
+T man.
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This is a great tutorial, but I'm interested in the 'other' process that is more precise. The time stretch function, who here has some more experience to make this tutorial more comprehensive?
Anyone?
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Awesome job Mike. + T for your effort.
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Awesome discovery! +T for your documentation.
Looking forward to the method of multichannel DVD-A creation. My attempts thus far have yeilded nothing but static :(
Jim
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Awesome discovery! +T for your documentation.
Looking forward to the method of multichannel DVD-A creation. My attempts thus far have yeilded nothing but static :(
Jim
So Jim, how is static in 5.1? ;) I've only tried it once and really didn't get anywhere either.
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Awesome discovery! +T for your documentation.
Looking forward to the method of multichannel DVD-A creation. My attempts thus far have yeilded nothing but static :(
Jim
So Jim, how is static in 5.1? ;) I've only tried it once and really didn't get anywhere either.
Static in 5.1 is the same as stereo, 'cept louder! A 4 channel 24/48 discrete montage wound up indicated as 24/96. I saved it just in case the player was doing it's thing correctly, but the receiver couldn't process it.
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"If you want to stretch one file to match the other:
match up a precise start point and end point on each recording and divide the length of one segment by the length of the other (same section of music). You'll find that one is maybe 99.98454% of the other or something like that.
Use wavelab's stretch function (it's the only one I've tried that sounds good). Select the whole file to be shortened and do a time stretch with these settings
It's probably good to set wavelab temp files to 24 or 32 bit as well.
Note that there's no point in using anything other than the ratio box to set the stretch. All the other boxes only appear to be more precise - in fact they don't produce the stretch that they indicate. Rather, they always produce the nearest '3 decimal ratio', so it's better just to set it there so you can see the length of the the file you'll *actually* get. And 3 decimals is it - any more precision ends up getting ignored. "(thanks dklein)
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"Here's another method to time stretch in wavelab that doesn't involve calculations.
Get your tracks in sync in a montage. Then go to the last few minutes of the set & split them(in the middle/left of the screen hit "edit>split all tracks @ cursor"). Then move the end portion of the track that needs to be stretched so it is again in sync. The gap you have between the track is the difference between the 2 tapes clock speeds, or drift. Move your cursor so it is exactly @ the beg. of the end piece of the track that you intend to stretch. Right click over the longer piece of the track that you intend to stretch & hit "transform>time stretch to cursor". Just make sure "Preserve pitch" is enabled. "
(thanks the yokel)
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Good stuff, I'm learning a lot. I was afraid of multitrack mode before!
+t for y'all.
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great tutorial mike B. it takes a while to actually get used to using the zoom and syncing up each song. but once you get the hang of that, it's a breeze. thanks again. many +T's
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great tutorial mike B. it takes a while to actually get used to using the zoom and syncing up each song. but once you get the hang of that, it's a breeze. thanks again. many +T's
qft, only took me 2+ hrs to do my latest matrix :)
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okay, i now need to figure something out. when i mixed my two sources the sample size changed from 16 to 32 bit ( i guess b/c i rendered two 16 bit sources) anyway, how do i change this back to 16 bit in wavelab 5.0? i'm assuming this will change the quality of the wav file
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okay, i now need to figure something out. when i mixed my two sources the sample size changed from 16 to 32 bit ( i guess b/c i rendered two 16 bit sources) anyway, how do i change this back to 16 bit in wavelab 5.0? i'm assuming this will change the quality of the wav file
it works with 32 bit float..you do a dither to get from 32 to 16(in the master section, choose your dither scheme, then render...). if you want to change that 32 bit float option..
go to options>preferences>file and youll see a drop down menu.
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okay, i now need to figure something out. when i mixed my two sources the sample size changed from 16 to 32 bit ( i guess b/c i rendered two 16 bit sources) anyway, how do i change this back to 16 bit in wavelab 5.0? i'm assuming this will change the quality of the wav file
it works with 32 bit float..you do a dither to get from 32 to 16(in the master section, choose your dither scheme, then render...). if you want to change that 32 bit float option..
go to options>preferences>file and youll see a drop down menu.
thanks for the response teddy. i tried to dither my file to 16 bit, but i haven't been able to do it. i am learning how to use wavelab, so i must be doing something wrong. i have opened my wav file, selected to dither, intern, noise type (off), noise shape (off), and output bit resolution (16). i click on render and chose the whole file. when i did an analysis using the bit meter the file is still 32 bits. what am i doing wrong?
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Very nice tutorial Mike! +T
Does anyone know to do this Wavelab 3.0? My brother n law gave this version to me and I would like to do a couple matrix's if I can . Everything seems to be very close 3.0 vs 5.0
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you have to select the file header within the sub menu as well...do "create new file" and then there is a drop down list that will show you the different options.
okay, i now need to figure something out. when i mixed my two sources the sample size changed from 16 to 32 bit ( i guess b/c i rendered two 16 bit sources) anyway, how do i change this back to 16 bit in wavelab 5.0? i'm assuming this will change the quality of the wav file
it works with 32 bit float..you do a dither to get from 32 to 16(in the master section, choose your dither scheme, then render...). if you want to change that 32 bit float option..
go to options>preferences>file and youll see a drop down menu.
thanks for the response teddy. i tried to dither my file to 16 bit, but i haven't been able to do it. i am learning how to use wavelab, so i must be doing something wrong. i have opened my wav file, selected to dither, intern, noise type (off), noise shape (off), and output bit resolution (16). i click on render and chose the whole file. when i did an analysis using the bit meter the file is still 32 bits. what am i doing wrong?