Parallel compression, my new love & best friend

[live2496]

Voxengo Elephant is really good but no free. Try the demo!

You will need to provide more details of the setup to get some advice on the phasing. Is this using outboard equipment or parallel processing of tracks in a DAW?

[SmokinJoe]

I've been doing some multi-tracking and mixing with Reaper (software).  Reaper comes with a bunch of plugins, and in particular there is one I use called "ReaComp".  Once you open up the plugin, you can fiddle with all the attack/release settings, or you can pick a drop down preset that some intellegent person left for you.  One of those presets is "New York compression", and it works pretty slick.  I tend to leave most of the settings per the preset, and then change "100% wet" to "50% wet".

The plugin was free, and it's a VST plugin.  I'm just not sure if all VST plugins are created equal, and if you can use it in your editor.
http://www.reaper.fm/reaplugs/index.php

[beatkilla]

Hey thanks for the reaper vst link.They work fine in Sony Vegas but there is no presets in reacomp for NY compression....could you tell me all the parameters in that preset.....and in the spirit of sharing great free VST plugins here are over 30 fantastic full interface Antress VSTs .....i use Modern Channel all the time .If you don't have these already Merry Christmas....
http://gersic.com/plugins/hosted/antress/

[page]

When running matrices I was digging (compressing) my soundboard channel looking for vocals

and fiddle and acoustic...this technique works great for compressing a board feed and mixing with an uncompressed audience source.

QFT     (Bump to the top... this is too important . )

Yep, it's saved my hide a couple of times. In a similar vein, the general motions worked on a recent show. The room mix had drums being just a touch too prominant at the show. I get home and find out that the board feed has vocals, and drums, (and some guitar, but it's really burried) and the drums are about 20db louder then the vocals (which are almost non-existant on the audience pull).

Trying this out and i get phasing what am i doing wrong?

Sounds like the resulting output from the plugin shifts in the timeline just a bit. Izotope's Ozone does that for all non-EQ/Amp-only operations in my experience so I make a single sample mark at 0db near the beginning of all of my files and then after I'm done processing, I go back and find it and move the processed file back to where I made the mark on the raw tracks, then adjust back and forth to taste for stuff that's smeared or shifted (usually the adjustments after re-alignment are counted in single ms forward or backward).

[live2496]

Trying this out and i get phasing what am i doing wrong?

If doing this in the computer, your software has to have latency compensation to adjust for latency added when using plugins.

[bdasilva]

I mix in Adobe Audition (3.0) to mix my matrixs   I load the board  track into AA  twice..  Put  one of these tracks into the multitrack side straight up.  on the other I use "hard limiting process"  I boost the input 13 to 15 db  and then "limit max amplitude" to -1 and then move this on to the multitrack side.  So when I mix I have a Mic stage right track,  A Mic stage left track , the soundboard signal untouched, and then the compressed to shit soundboard track (and if I'm lucky a track with just vocals)  With it all layed out it is easy to "see" what you are working with.  Often I do a mixdown to new stereo file  and then put this back with the other tracks on the multitrack side and mix it again... there is a certain blending that occurs.  Then   I do another mixdown to stereo file and it becomes my stereo master.

[live2496]

To keep things in phase in Samplitude/Sequoia you really only need to keep the cursor at the same position or put it a the border of an object or split point. I render a file (ie. bounce) with the compression and then import that file into the project. To align it you snap the new object to the cursor position (on another track) at the start of the range of the bounce.

Where things can get out of phase is if you are going outside of the daw and then bringing the compressed audio back in on an auxillary input. Once you go outside of the DAW you have to blend the compressed output and the original track in the analog domain or capture the audio to a file and somehow align it to the original.

DAW software will buffer/delay the playback to accomodate any delays introduced by the plugins in use.

[ashevillain]

I have an AUD recording of Corey Harris solo. Corey was playing mostly quiet stuff and crowd was really quiet during the songs but not so quiet between songs. Do you think this technique would work well?

If this is not the technique to use (I'm sure this has been answered before...too lazy to search) what is a good way to boost the songs and decrease the between song crowd noise while maintaining some level of cohesiveness?

[page]

I have an AUD recording of Corey Harris solo. Corey was playing mostly quiet stuff and crowd was really quiet during the songs but not so quiet between songs. Do you think this technique would work well?

If this is not the technique to use (I'm sure this has been answered before...too lazy to search) what is a good way to boost the songs and decrease the between song crowd noise while maintaining some level of cohesiveness?

What I'd do if I were in your shoes depends on one thing: How much range do you have between the loud audience sections and the quiet music sections? If it's barely 5db, I'd use a clean limiter (I prefer Izotope's Ozone for the job, but ymmv) and just squish it. If you have like 15 or 20db difference, then I'd look at compressing it on a curve and if that's not enough, limiting the result a little.

[Gutbucket]

..what is a good way to boost the songs and decrease the between song crowd noise while maintaining some level of cohesiveness?

Parallel compression is not the tool you want for adjusting this.

What page describes can work if you set things correctly so that the limiter / compressor only works on the loud, between song portions and doesn't do much to the songs.  But for what you want to do, I'd suggest manually drawing volume changes with the envelope tool- higher when he's playing and lower when when the loud crowd comes in.  If all the loud parts are mostly between songs there shouldn't be too many events to make doing that overly difficult and its easy to make the change up and down happen as quickly or slowly as needed by adjusting the slope between points.  Alternately you could achieve the same thing using fader automation if you prefer working that way and your software does it.

Why?
If I understand correctly the dynamic range during the songs themselves is not the problem and doing it this way doesn't change it at all.   You'll preserve the existing dynamics within the music segments and and within the louder between song segments, but will make difference between them less which is what you want.  Compression or limiting will change the sound of the song portions, even if set well. Standard compression reduces dynamics by working most on the loudest parts and transients.  Parallel compression instead changes the quiet end of things- which in this case are the song portions.  Your recording may or may not benefit from that, but it's a separate dynamics issue than the song/applause level differences.

[ashevillain]

^thanks for the info.

[hi and lo]

When working with a continuous live recording that varies widely in dynamic range from acoustic to full band, how should I go about parallel compression? For example, working on a recent Trey show where first set is all acoustic minus the last two songs where it's full band and very loud. I can use a lot of compression on the parallel track to where the acoustic majority sounds great, but the last two are highly brickwalled. Is it bad to mix in any portion that might be brickwalled? Do I need to apply different compression amounts and cross fade it?

[JasonSobel]

Do I need to apply different compression amounts and cross fade it?

I would do this.  the dynamics change a lot from the acoustic portion to the full band playing for the last two songs, so it makes sense that they two portions require different post processing.

[bdasilva]

I love to run matrixes... It is what I do better than 90% of the time.  Last night I'm recording a blues man (Louisiana Red) in my favorite club. (Rhythm Room)  So channels 1 and 2 are split (8') omnis at the stage lip between the monitors and PA. I split the board signal between channels 3 and 4... channel 5 is an aux out of the board with straight vocals. Channel 3 is normal and I set the trim to get a decent level... on channel 4 I turn the limiter on and set the gain as high as it will go. I had wondered if the limiter was robust enough to handle this and it is... I just  turned down this channel into the mix and it worked like a champ .  on the fly New York Compression in Phoenix...  Right now I'm mixing the 5 channels (and the mix)   God I love this machine.
Yea yea... I know this in recording section but this subject  just needs to be at the top once in while

[Patrick]

I love to run matrixes... It is what I do better than 90% of the time.  Last night I'm recording a blues man (Louisiana Red) in my favorite club. (Rhythm Room)  So channels 1 and 2 are split (8') omnis at the stage lip between the monitors and PA. I split the board signal between channels 3 and 4... channel 5 is an aux out of the board with straight vocals. Channel 3 is normal and I set the trim to get a decent level... on channel 4 I turn the limiter on and set the gain as high as it will go. I had wondered if the limiter was robust enough to handle this and it is... I just  turned down this channel into the mix and it worked like a champ .  on the fly New York Compression in Phoenix...  Right now I'm mixing the 5 channels (and the mix)   God I love this machine.
Yea yea... I know this in recording section but this subject  just needs to be at the top once in while

What are you recording on?  Depends what type of limiter you're hitting but I can't imagine a built in limiter on a recording deck would be useful in this situation.