Topic: Matrix in post best practice workflow

[tim in jersey]

n00b to the matrix in post thing...

Scenario:

Deck 1 is a SBD feed, Deck 2 is my aud recording. Neither is jammed to timecode. I mix/correct for drift in post using Vegas.

I run levels on each very conservatively. About -9 to -6 dbfs, peak.

Should I normalize/amplify each file prior to bringing it in to the DAW or just mix and then amplify?

[justink]

i amplify both sources to about -5 or -4 db (for a 50/50 mix).  then align.  then mix/render (the joining will amplify some waves).

there's a great thread about this using audacity, pretty easy.

[page]

I use the following:

1) Align via preferred method (stretch in Vegas/Reaper or time-align in Audacity)
2) Determine what I like about either source (sbd has less reverberation in the 1-3khz range, aud has a warm gooey bass, etc), and EQ to bring out what I like about both, maybe a net change of 2db tops, just enough to nudge it in a direction. Don't overdo it.
3) Make sure the peaks in either are at least -6 or lower.
3.5) double check alignment.
4) Render to 2ch. (Assuming a render is required to apply editing to both sources equally)
5) EQ/process anything else.
6) Amplify the remaining db and do any limiting and dither.

I used to keep the peaks between -3 and -6 which mathematically should bring your render to just under 0, but I kept having the occasional overage and having to redo the render so I figured the additional few DB of amping in Ozone or whatever is clean enough I'm not worried about it. Make sure to double check your alignment, I prefer to check mine now over the course of a couple of listening sessions and in both loud and quiet sections throughout the recording since non-clocked sources can drift at different speeds (ever so slightly which is a pisser but usually not much of a problem). I don't think I've forgotten anything....

[justink]

I use the following:

1) Align via preferred method (stretch in Vegas/Reaper or time-align in Audacity)
2) Determine what I like about either source (sbd has less reverberation in the 1-3khz range, aud has a warm gooey bass, etc), and EQ to bring out what I like about both, maybe a net change of 2db tops, just enough to nudge it in a direction. Don't overdo it.
3) Make sure the peaks in either are at least -6 or lower.
3.5) double check alignment.
4) Render to 2ch. (Assuming a render is required to apply editing to both sources equally)
5) EQ/process anything else.
6) Amplify the remaining db and do any limiting and dither.

I used to keep the peaks between -3 and -6 which mathematically should bring your render to just under 0, but I kept having the occasional overage and having to redo the render so I figured the additional few DB of amping in Ozone or whatever is clean enough I'm not worried about it. Make sure to double check your alignment, I prefer to check mine now over the course of a couple of listening sessions and in both loud and quiet sections throughout the recording since non-clocked sources can drift at different speeds (ever so slightly which is a pisser but usually not much of a problem). I don't think I've forgotten anything....

he said it best.

[runonce]

n00b to the matrix in post thing...

Scenario:

Deck 1 is a SBD feed, Deck 2 is my aud recording. Neither is jammed to timecode. I mix/correct for drift in post using Vegas.

I run levels on each very conservatively. About -9 to -6 dbfs, peak.

Should I normalize/amplify each file prior to bringing it in to the DAW or just mix and then amplify?

If they are 16 bit sources - you might want to skip normalization until the end (as recommened by page in the last post) - You're going to need about -6db of extra headroom to handle the higher peaks that result when you combine sources.

[tim in jersey]

That answers my questions. Thank you, gentlemen.

[Walstib62]

I could be totally wrong on this, and if so, please someone let me know. But, I have always imported the tracks and normalized them first. The reason being that normalizing removes any possible dc offset. That way, all tracks are nulled to 0 offset prior to any other processing. To make things easier level-wise, I normalize to -6bB. Then, when the tracks are combined, the result is always a little below 0 without going over.

[justink]

I could be totally wrong on this, and if so, please someone let me know. But, I have always imported the tracks and normalized them first. The reason being that normalizing removes any possible dc offset. That way, all tracks are nulled to 0 offset prior to any other processing. To make things easier level-wise, I normalize to -6bB. Then, when the tracks are combined, the result is always a little below 0 without going over.

nothing wrong with that.

[page]

I could be totally wrong on this, and if so, please someone let me know. But, I have always imported the tracks and normalized them first. The reason being that normalizing removes any possible dc offset.

I just checked in Audacity and there is an option to remove the offset when you do it (news to me). I don't recall there being one in reaper, and I know in RX it's a separate function. In Ozone, they have it chained to occur at the end before dither or the output gain structure.

but you bring up a good point; do you strip it before you do anything or at the end? All of my current gear produces so little offset naturally that I haven't had much reason to do it any earlier in the chain, but from a best practices standpoint, it's a valid question.

[morst]

I could be totally wrong on this, and if so, please someone let me know. But, I have always imported the tracks and normalized them first. The reason being that normalizing removes any possible dc offset. That way, all tracks are nulled to 0 offset prior to any other processing.

I think you might be totally wrong. Normalization just brings up overall output levels, DC Offset Removal is a separate function. Unless you are using some offset-correcting normalizer plugin?

You'll get WAY better results mixing 24-bit signals if you can master that way...

[kubacheck]

I could be totally wrong on this, and if so, please someone let me know. But, I have always imported the tracks and normalized them first. The reason being that normalizing removes any possible dc offset. That way, all tracks are nulled to 0 offset prior to any other processing.

I think you might be totally wrong. Normalization just brings up overall output levels, DC Offset Removal is a separate function. Unless you are using some offset-correcting normalizer plugin?

You'll get WAY better results mixing 24-bit signals if you can master that way...

the one time I tried to do a matrix in Vegas, it wouldn't let me render the matrix of the two 24 bit sources as a resultant 24 bit output file, it automatically dithered it down to 16/44.1.... if there was a way to keep it at 24/48, I couldn't figure it out....

[morst]

it automatically dithered it down to 16/44.1.... if there was a way to keep it at 24/48, I couldn't figure it out....

If that's really true, I would advise you to find something other than Vegas to do your work. No sense in wrecking your bits like it's the only option!!?

[kubacheck]

it automatically dithered it down to 16/44.1.... if there was a way to keep it at 24/48, I couldn't figure it out....

If that's really true, I would advise you to find something other than Vegas to do your work. No sense in wrecking your bits like it's the only option!!?

its been a couple years, but as I recall, the pull down menus only went up to 16 and 44.1, I could've rendered at lower bit rates, but 16/44.1 were the highest quality values I could select..... perhaps that was specific to the version of Vegas I was using, I think it was Vegas 8.....

[beatkilla]

Thats definetely not true for Sony Vegas. You can work with and render up to 24bit 192khz.      I just checked and you must have the lite version of vegas and that one goes no higher than 16  48.

[kubacheck]

Thats definetely not true for Sony Vegas. You can work with and render up to 24bit 192khz.

well, like I said, if there was a way ( and I looked and looked), I could not figure it out..... the details are not fresh since it's been a couple years, but I recall I could not find any options anywhere that allowed me to render the output audio file at 24/48..... as I mentioned, I was using Vegas 8, if that means anything....

[F.O.Bean]

I have used WaveLab in the past w/ the Audio Montage to do my post matrix's. I had to sync up every few minutes on one of my matrix's That was a TOTAL PITA

[Walstib62]

I could be totally wrong on this, and if so, please someone let me know. But, I have always imported the tracks and normalized them first. The reason being that normalizing removes any possible dc offset. That way, all tracks are nulled to 0 offset prior to any other processing.

I think you might be totally wrong. Normalization just brings up overall output levels, DC Offset Removal is a separate function. Unless you are using some offset-correcting normalizer plugin?

You'll get WAY better results mixing 24-bit signals if you can master that way...

In audacity, you can check a box to remove dc offset during normalizing. There is another box to manually select the dB value to normalize to.

[Chimney Top]

n00b to the matrix in post thing...

Scenario:

Deck 1 is a SBD feed, Deck 2 is my aud recording. Neither is jammed to timecode. I mix/correct for drift in post using Vegas.

I run levels on each very conservatively. About -9 to -6 dbfs, peak.

Should I normalize/amplify each file prior to bringing it in to the DAW or just mix and then amplify?

If they are 16 bit sources - you might want to skip normalization until the end (as recommened by page in the last post) - You're going to need about -6db of extra headroom to handle the higher peaks that result when you combine sources.

once they are synced, i think it adds +3db.

[drivingwheel]

n00b to the matrix in post thing...

Scenario:

Deck 1 is a SBD feed, Deck 2 is my aud recording. Neither is jammed to timecode. I mix/correct for drift in post using Vegas.

I run levels on each very conservatively. About -9 to -6 dbfs, peak.

Should I normalize/amplify each file prior to bringing it in to the DAW or just mix and then amplify?

If they are 16 bit sources - you might want to skip normalization until the end (as recommened by page in the last post) - You're going to need about -6db of extra headroom to handle the higher peaks that result when you combine sources.

I keep everything digital at -6, whether it's 16-or 24-bit...and I try to do that for both the actual source recordings and any final post mixed 'master'; in the digital world, it seems good to stay at -6.  Everything I listen to music on has a volume knob...

[acidjack]

I can't believe I find myself differing (I hesitate to say "disagree", since I don't, really) with page on something related to a topic he knows 10x as much about, but I actually like to hold off on EQ until the sources are mixed down.  IME, trying to EQ each separately sometimes resulted in a less desirable result in mixdown.  Page's method is totally logical, I just found it easier to get the right balance of sources, then EQ and do other things on the combined file.  EXCEPT in some circumstances where something was really off on one of the two files - for example, an SBD that required compression or other tweaks because it was so horribly off balance.

[page]

I can't believe I find myself differing (I hesitate to say "disagree", since I don't, really) with page on something related to a topic he knows 10x as much about, but I actually like to hold off on EQ until the sources are mixed down.  IME, trying to EQ each separately sometimes resulted in a less desirable result in mixdown.  Page's method is totally logical, I just found it easier to get the right balance of sources, then EQ and do other things on the combined file.  EXCEPT in some circumstances where something was really off on one of the two files - for example, an SBD that required compression or other tweaks because it was so horribly off balance.

yeah, I did that for a while, and to a degree I still sort of do. I didn't write out every little nuance to keep it somewhat simple, but thats why I align them first. Then I can listen to the natural combination (via the wonderful fairy land of non-destructive editing/rendering) and hear what works and what doesn't. Occasional muting of one source then gives me a better sense of what each one brings to the table and I can start to tweak the individual ones pre-mix. The reason for doing the pre-mix adjustments is so you can retain stuff that is micro-detail oriented in one source that would be overblown in another and lost when you EQ or compensate for that amount coming from the overblown source. For example, lets say you have a fast attack and fast decay on the sbd bass, and a warm creamy attack-less bass on the aud courtesy of a boomy room. If you amp the bass in the sbd first, do the render, then reduce it, you can retain that initial thwack a little more and then let the warm reverb/boom constitute the decay tail and fill around the note. It's not perfect by any means, but I find it an improvement the majority of time I try it but ymmv of course. Again, you're not doing large adjustments, just a touch here and there to benefit you.

The other trade off is time, I've been tinkering with post production techniques for a good year or so now and this sort of stuff still kills time. If I was taping zman-quantity, I'd probably give up on this sort of stuff, but I'm not, so it's not bad.

[runonce]

Thats why I gave up on post matix (for the moment) - too much time in post...back to using a mixer and OTF (on the fly) matirxing.

I usually run partial SBD + Wide Stage/Pan.
Often my mics end up almost where the soundman might put his mic on the guitar cab -  but not as close, so get a little more leak -  and get the some bass/drums.
So, I can live with that little delay there...

That approach greatly increases the chances that I will have listenable recording by the end of the next day.

When I have to capture a complete board mix and add AUD - generally in that situation...I just keep the AUD mic levels low enough that the delay is barely audible...sucks for tapers as we all love to hear our mics.
But in this situation - just a touch to add some room bass, add some "slop" so the board doesn't sound so dry...kinda sounds like an old live FM show this way...which works for one country act I work with...