Topic: is a Matrix w/ 2 JB3's possible?

[T.J.]

hey everyone,

i have a question regarding a matrix using 2 JB3 sources: is it possible using wavelab 5.0?  i have recorded one one DAUD source 483>V3>JB3#1 and another SBD>RCA>JB3#2. Taking these two sources and dumping them in wavelab using the montage option, is it possible to sync the two sources even though they don't run off the same word clock? i have already tracked one source out using cd wave. the other i left untouched. it seems impossible get them lined up. has anyone ever successfully done this? any tips?

thanks in advance...

[halleyscomet8]

this should do it. i've use this for every matrix i have done.

http://www.sloppyart.com/wavelab_matrix/

[Chanher]

This method can sound just as good as any other matrix, it is just more difficult and time consuming. I'm done with this method so I bought a busman R4.

Search or check the archives, there's a guide to doing this in Wavlab.

[Church-Audio]

hey everyone,

i have a question regarding a matrix using 2 JB3 sources: is it possible using wavelab 5.0?  i have recorded one one DAUD source 483>V3>JB3#1 and another SBD>RCA>JB3#2. Taking these two sources and dumping them in wavelab using the montage option, is it possible to sync the two sources even though they don't run off the same word clock? i have already tracked one source out using cd wave. the other i left untouched. it seems impossible get them lined up. has anyone ever successfully done this? any tips?

thanks in advance...

The only issue I could see is sample rates not being dead on between the two units. One of the methods I use for two recorders is using a 1k Burst about 1 second, and feeding it into both machines at the same time then I use the wave form viewer in wavlab to sync them up works like a charm if the sample rates are not the same then you have time issues but they should be solved if you sync the recording with a burst tone. In order to do this you must disconnect the inputs and hook up your burst tone and leave them in record and connect the inputs again so that you do not stop the record on the machines. This will give you a time frame. Because you can not possibly hit record at the same time on two machines its the only way to sync them up and be 100% accurate.

Chris Church


Chris Church

[T.J.]

this should do it. i've use this for every matrix i have done.

http://www.sloppyart.com/wavelab_matrix/

thanks guys. that is the link i used to initially educate myself on how to do a matrix and it is a huge help. however, i can't seem to sync the two sources correctly. i zoom in to line up the wav files and repeatidly(sp?) listen to hear them lined up, but can't get to sound perfect.

i am really leaning towards getting an R4, i just don't matrix that often and have to justify the need for four channels.

[Church-Audio]

this should do it. i've use this for every matrix i have done.

http://www.sloppyart.com/wavelab_matrix/

thanks guys. that is the link i used to initially educate myself on how to do a matrix and it is a huge help. however, i can't seem to sync the two sources correctly. i zoom in to line up the wav files and repeatidly(sp?) listen to hear them lined up, but can't get to sound perfect.

i am really leaning towards getting an R4, i just don't matrix that often and have to justify the need for four channels.

Read what I said above there is a way to do it. If you dont have a tone burst unit very cheap. find a tone burst file on the internet and use a simple cd player with a Y cable going into both recorders at the same time then you can sync them up.

Chris Church

[T.J.]

this should do it. i've use this for every matrix i have done.

http://www.sloppyart.com/wavelab_matrix/

thanks guys. that is the link i used to initially educate myself on how to do a matrix and it is a huge help. however, i can't seem to sync the two sources correctly. i zoom in to line up the wav files and repeatidly(sp?) listen to hear them lined up, but can't get to sound perfect.

i am really leaning towards getting an R4, i just don't matrix that often and have to justify the need for four channels.

Read what I said above there is a way to do it. If you dont have a tone burst unit very cheap. find a tone burst file on the internet and use a simple cd player with a Y cable going into both recorders at the same time then you can sync them up.

Chris Church

i hear what you're saying chris, but just fail to see how this can help me with the project i have already started. +T for the help

[easy jim]

The problem you are having is drift.  Drift occurs between the two sources as a result of the difference between the two clocks.  While it should be very close coming from two JB3s, it would not be unusual to have as much as 50-100 milisecs of drift over a long show.  To correct drift, you need to time-scale one of the sources to the other, which will serve as your master clock.  (Time-scaling can also be referred to as changing 'pitch'.)

I find chopping up one source and pasting it in aligned to the beginning of each track on the other to be unduly cumbersome.  And, it is not an accurate way to make a matrix because you will still have drift between the sources which may be noticeable by the end of each track, especially if it is a long one.  So, you may end up with things perfectly aligned at the beginning of a track, but it will sound echo-ey or chorus-ey by the end.  It is much better to put the two sources next to each other in your edit window and then time-scale one to the other.

To do so, find a loud transient sound (snare hit, mic pop, etc.) as close to the beginning as possible and line them up.  Then, go to the end and find a similar loud transient as close to the end as possible and see how far off the two sources are on your time ruler.  Calculate the amount of drift and use whatever time-scaling function is available in Wavelab to either stretch the shorter source or shrink the longer source to get them to line up over the entire show.  In some software, you grab then end of one source and drag it using the time ruler in the edit window to time-scale by the proper amount (this is how AudioDesk & Digital Performer do it); in other software, you calculate the percentage of the difference between the source in original form and the properly time-scaled outcome you desire:  i.e., shrinking a source by 99.999975 % or something like that to make the longer source match with the shorter one.  Generally, your editing software will create a new file/track for the time-scaled version and it will take some processing time for it to complete before you may then get your mix together to go from 4 -> 2 tracks.

I do not know Wavelab and whether or not it has a time-scale function, but I would be surprised if it did not.  Once you mix two sources from different clocks a few times, correcting drift gets a lot easier and should become something you may consider a relatively easy step to make a matrix from two separate sources.  Now, I tend to agonize more over setting the levels of the two sources and whether or not to spot mute any artifacts out of the mix if they are bad and only present in one source.

Maybe someone who knows about time-scaling in Wavelab will chime in.  Feel free to PM me if you get stuck.

By the way, you should use the V3 source as the 'master clock' assuming you patched in the JB3 digitally.  The V3's clock will be far superior and more accurate than the internal clock in the JB3.

[Church-Audio]

The problem you are having is drift.  Drift occurs between the two sources as a result of the difference between the two clocks.  While it should be very close coming from two JB3s, it would not be unusual to have as much as 50-100 milisecs of drift over a long show.  To correct drift, you need to time-scale one of the sources to the other, which will serve as your master clock.  (Time-scaling can also be referred to as changing 'pitch'.)

I find chopping up one source and pasting it in aligned to the beginning of each track on the other to be unduly cumbersome.  And, it is not an accurate way to make a matrix because you will still have drift between the sources which may be noticeable by the end of each track, especially if it is a long one.  So, you may end up with things perfectly aligned at the beginning of a track, but it will sound echo-ey or chorus-ey by the end.  It is much better to put the two sources next to each other in your edit window and then time-scale one to the other.

To do so, find a loud transient sound (snare hit, mic pop, etc.) as close to the beginning as possible and line them up.  Then, go to the end and find a similar loud transient as close to the end as possible and see how far off the two sources are on your time ruler.  Calculate the amount of drift and use whatever time-scaling function is available in Wavelab to either stretch the shorter source or shrink the longer source to get them to line up over the entire show.  In some software, you grab then end of one source and drag it using the time ruler in the edit window to time-scale by the proper amount (this is how AudioDesk & Digital Performer do it); in other software, you calculate the percentage of the difference between the source in original form and the properly time-scaled outcome you desire:  i.e., shrinking a source by 99.999975 % or something like that to make the longer source match with the shorter one.  Generally, your editing software will create a new file/track for the time-scaled version and it will take some processing time for it to complete before you may then get your mix together to go from 4 -> 2 tracks.

I do not know Wavelab and whether or not it has a time-scale function, but I would be surprised if it did not.  Once you mix two sources from different clocks a few times, correcting drift gets a lot easier and should become something you may consider a relatively easy step to make a matrix from two separate sources.  Now, I tend to agonize more over setting the levels of the two sources and whether or not to spot mute any artifacts out of the mix if they are bad and only present in one source.

Maybe someone who knows about time-scaling in Wavelab will chime in.  Feel free to PM me if you get stuck.

By the way, you should use the V3 source as the 'master clock' assuming you patched in the JB3 digitally.  The V3's clock will be far superior and more accurate than the internal clock in the JB3.

The best way to deal with already recorded tracks Is to use a program like Nuendo or Qbase. And put two sets of stereo tracks into two stereo interleave tracks and then use your mouse to grab the wave and line it up with the other stereo track. and open your view until you get a really good lineup. Then once it looks good it will be close. Then hit play and listen, then slide one of the waves left then listen to the improvement if its worse go to your right and then you can figure out if its late or early thats all you need to do. Then send me $50.00 US in an envelope for a "thank you gift"

Chris Church

[poorlyconditioned]

The problem you are having is drift.  Drift occurs between the two sources as a result of the difference between the two clocks.  While it should be very close coming from two JB3s, it would not be unusual to have as much as 50-100 milisecs of drift over a long show.  To correct drift, you need to time-scale one of the sources to the other, which will serve as your master clock.  (Time-scaling can also be referred to as changing 'pitch'.)

I find chopping up one source and pasting it in aligned to the beginning of each track on the other to be unduly cumbersome.  And, it is not an accurate way to make a matrix because you will still have drift between the sources which may be noticeable by the end of each track, especially if it is a long one.  So, you may end up with things perfectly aligned at the beginning of a track, but it will sound echo-ey or chorus-ey by the end.  It is much better to put the two sources next to each other in your edit window and then time-scale one to the other.

To do so, find a loud transient sound (snare hit, mic pop, etc.) as close to the beginning as possible and line them up.  Then, go to the end and find a similar loud transient as close to the end as possible and see how far off the two sources are on your time ruler.  Calculate the amount of drift and use whatever time-scaling function is available in Wavelab to either stretch the shorter source or shrink the longer source to get them to line up over the entire show.  In some software, you grab then end of one source and drag it using the time ruler in the edit window to time-scale by the proper amount (this is how AudioDesk & Digital Performer do it); in other software, you calculate the percentage of the difference between the source in original form and the properly time-scaled outcome you desire:  i.e., shrinking a source by 99.999975 % or something like that to make the longer source match with the shorter one.  Generally, your editing software will create a new file/track for the time-scaled version and it will take some processing time for it to complete before you may then get your mix together to go from 4 -> 2 tracks.

I do not know Wavelab and whether or not it has a time-scale function, but I would be surprised if it did not.  Once you mix two sources from different clocks a few times, correcting drift gets a lot easier and should become something you may consider a relatively easy step to make a matrix from two separate sources.  Now, I tend to agonize more over setting the levels of the two sources and whether or not to spot mute any artifacts out of the mix if they are bad and only present in one source.

Maybe someone who knows about time-scaling in Wavelab will chime in.  Feel free to PM me if you get stuck.

By the way, you should use the V3 source as the 'master clock' assuming you patched in the JB3 digitally.  The V3's clock will be far superior and more accurate than the internal clock in the JB3.

The best way to deal with already recorded tracks Is to use a program like Nuendo or Qbase. And put two sets of stereo tracks into two stereo interleave tracks and then use your mouse to grab the wave and line it up with the other stereo track. and open your view until you get a really good lineup. Then once it looks good it will be close. Then hit play and listen, then slide one of the waves left then listen to the improvement if its worse go to your right and then you can figure out if its late or early thats all you need to do. Then send me $50.00 US in an envelope for a "thank you gift"

Chris Church

(+T is all I can afford right now.)  I've tried this (Cubase).  The problem is the clocks move a little bit with time.  The human ear is very good at hearing *changes* in alignment.  You can be out 10 or 15ms as long as the clocks are matched.  But if there are two different clocks, it can be difficult to get great sound throughout.  Hence the need to split a show into several pieces and move each one.

  Richard

[Church-Audio]

The problem you are having is drift.  Drift occurs between the two sources as a result of the difference between the two clocks.  While it should be very close coming from two JB3s, it would not be unusual to have as much as 50-100 milisecs of drift over a long show.  To correct drift, you need to time-scale one of the sources to the other, which will serve as your master clock.  (Time-scaling can also be referred to as changing 'pitch'.)

I find chopping up one source and pasting it in aligned to the beginning of each track on the other to be unduly cumbersome.  And, it is not an accurate way to make a matrix because you will still have drift between the sources which may be noticeable by the end of each track, especially if it is a long one.  So, you may end up with things perfectly aligned at the beginning of a track, but it will sound echo-ey or chorus-ey by the end.  It is much better to put the two sources next to each other in your edit window and then time-scale one to the other.

To do so, find a loud transient sound (snare hit, mic pop, etc.) as close to the beginning as possible and line them up.  Then, go to the end and find a similar loud transient as close to the end as possible and see how far off the two sources are on your time ruler.  Calculate the amount of drift and use whatever time-scaling function is available in Wavelab to either stretch the shorter source or shrink the longer source to get them to line up over the entire show.  In some software, you grab then end of one source and drag it using the time ruler in the edit window to time-scale by the proper amount (this is how AudioDesk & Digital Performer do it); in other software, you calculate the percentage of the difference between the source in original form and the properly time-scaled outcome you desire:  i.e., shrinking a source by 99.999975 % or something like that to make the longer source match with the shorter one.  Generally, your editing software will create a new file/track for the time-scaled version and it will take some processing time for it to complete before you may then get your mix together to go from 4 -> 2 tracks.

I do not know Wavelab and whether or not it has a time-scale function, but I would be surprised if it did not.  Once you mix two sources from different clocks a few times, correcting drift gets a lot easier and should become something you may consider a relatively easy step to make a matrix from two separate sources.  Now, I tend to agonize more over setting the levels of the two sources and whether or not to spot mute any artifacts out of the mix if they are bad and only present in one source.

Maybe someone who knows about time-scaling in Wavelab will chime in.  Feel free to PM me if you get stuck.

By the way, you should use the V3 source as the 'master clock' assuming you patched in the JB3 digitally.  The V3's clock will be far superior and more accurate than the internal clock in the JB3.

The best way to deal with already recorded tracks Is to use a program like Nuendo or Qbase. And put two sets of stereo tracks into two stereo interleave tracks and then use your mouse to grab the wave and line it up with the other stereo track. and open your view until you get a really good lineup. Then once it looks good it will be close. Then hit play and listen, then slide one of the waves left then listen to the improvement if its worse go to your right and then you can figure out if its late or early thats all you need to do. Then send me $50.00 US in an envelope for a "thank you gift"

Chris Church

(+T is all I can afford right now.)  I've tried this (Cubase).  The problem is the clocks move a little bit with time.  The human ear is very good at hearing *changes* in alignment.  You can be out 10 or 15ms as long as the clocks are matched.  But if there are two different clocks, it can be difficult to get great sound throughout.  Hence the need to split a show into several pieces and move each one.

  Richard

Richard the clocks do not jump around its sold for each recorder. If they are jumping around the pitch would be changing since its just flanging between tracks its the difference between two recorders clocks that is the problem one could be 44.100 the other could be 44.155 this difference creates a problem called flanging between the two recorders trust me your clock is not moving on the the recorders if it was there would be wow and flutter something digital has virtually eliminated. The clocks must be solid on a digital recorder! if not your pitch will change and the recorder would be a peace of shit.

Chris Church

[T.J.]

good advise by all. a round of T's is all i can afford right now too 
i'm going to play around with with the pitch in wavelab, amongst other things. there is no "deadline" in which this needs to be done.

By the way, you should use the V3 source as the 'master clock' assuming you patched in the JB3 digitally.  The V3's clock will be far superior and more accurate than the internal clock in the JB3.

yes i did use the opti out on the V3 to record. i'm assuming when you refer to the "master" clock you mean i should use the V3 source as the "base" source. i should try to strech the other source to match the V3 source, correct?

[Gordon]

I have done a number of matrix's like this.

mics > v3 > jb3

& sbd > rca > jb3

break the sbd into tracks and leave the aud as one long wave file.  then open the aud in the monatge.  add sbd track one, do it track by track.  I have never had any problems with drift.  it takes some time and practice to line them up but it's not that hard.

[T.J.]

I have done a number of matrix's like this.

mics > v3 > jb3

& sbd > rca > jb3

break the sbd into tracks and leave the aud as one long wave file.  then open the aud in the monatge.  add sbd track one, do it track by track.  I have never had any problems with drift.  it takes some time and practice to line them up but it's not that hard.

exactly what i have and how i tried. to me, it's a pain in the a$$. i must not have the patience for lining the sources up in post.

[easy jim]

yes i did use the opti out on the V3 to record. i'm assuming when you refer to the "master" clock you mean i should use the V3 source as the "base" source. i should try to strech the other source to match the V3 source, correct?

Yep, that is how I would do it.

break the sbd into tracks and leave the aud as one long wave file.  then open the aud in the monatge.  add sbd track one, do it track by track.  I have never had any problems with drift.  it takes some time and practice to line them up but it's not that hard.

exactly what i have and how i tried. to me, it's a pain in the a$$. i must not have the patience for lining the sources up in post.

I have found that drift is still a problem when you chop up one source and align every track.  I think you will find, as well, that with this approach you will have brief gaps of a few milisecs +/- between each segment of the chopped up source when it is lined up with the continuous source.  I can often hear the added delay from drift by the end of a 10 -15 min track if the clocks differ enough.  It is a major pain to line them up so many times taking the chop-up-one-source approach, so I try and only do it once at the very beginning and then time-scale one source to the other.  If the time-scaling is done accurately, your two sources would then be as if recorded both slaved to the same clock.

The problem is the clocks move a little bit with time.  The human ear is very good at hearing *changes* in alignment.  You can be out 10 or 15ms as long as the clocks are matched.  But if there are two different clocks, it can be difficult to get great sound throughout.  Hence the need to split a show into several pieces and move each one.

Richard the clocks do not jump around its sold for each recorder. If they are jumping around the pitch would be changing since its just flanging between tracks its the difference between two recorders clocks that is the problem one could be 44.100 the other could be 44.155 this difference creates a problem called flanging between the two recorders trust me your clock is not moving on the the recorders if it was there would be wow and flutter something digital has virtually eliminated. The clocks must be solid on a digital recorder! if not your pitch will change and the recorder would be a peace of shit.

I think we're referring to a few different things here.  As for the clocks, which I've learned a lot more about as I have been multitracking, and also often grabbing additional sources from different recorders:  digital clocks will be very internally consistent in their measurement of time in general whether they are really high end (apogee, for instance) or lower end.  However, two different clocks, even two of the exact same clocks by the exact same manufacturer, will not maintain a consistency in reference to each other as they register the passage of time.  This is why drift occurs, and it is why master clocking devices like the Apogee Big Ben, etc. are necessary when using multiple digital front ends for a multitrack.  Master clocks like the Big Ben allow you to sync/slave all the digital sources to whichever device is assigned as the 'master.'  Otherwise, you get jitter (clocking errors) due to the slight inconsistency between each devices' internal clocks' measurement of the passge of time.  The longer the recording, the more apparent the drift between two sources from different clocks.