Headphone Mixes from Logic

I am considering rationalising my studio to a mixerless set up ( see Jim Frazier's comments in a recent thread http://discussions.apple.com/message.jspa?messageID=8174674#8174674)
I bring live projects from studios and add over dubs such as vocals, horns, percussion etc. I have a simple headphone monitoring system which is perfect for up to 4 musicians to record to what is already in Logic - say a horn section and percussionist.
So far all monitoring in my set up has run from the mixing desk which is the hub of all the inputs and outputs via a loom to my MOTU 828 - but my ideas have been opened up by comments such as Jim's ( see above thread) where he describes an essentially mixerless set up involving such devices as a Mackie Big Knob or Presonus + an Apogee Ensemble + A control Surface ( Mackie or Euphonix type thing)
I have 3 questions:
i) Practically speaking what is the best way to plug mics and line inputs into the back of the audio interface. Do most 'mixerless' studios use some kind of rack patch bay.. or am I exaggerating the hassle of having to lean over a rack where the inputs for mic are facing the wrong way?
ii) Is it practical and easy to create separate software mixes ( say one for horn section and one for percussionist) in Logic and route them to headphones via outputs of the Presonus/Mackie Big Knob type studio control box? Do you use Software Monitoring and is latency an issue here?
iii) Could someone just confirm the basic bits of gear that one needs to do this: ie Mac + Audio Interface + Patch Bay (?) + Studio Control Box + Control Surface
many thanks to anyone who can contribute
m s

Hi ms,
I'd be curious what the other "mixerless" guys are doing as well, but I can tell you how I'm managing things here.
In my studio, I have dedicated mic pres, so I don't plug my mics into an audio interface. The mics get plugged from the mic preamps right to the converters, which in my case, is a Apogee Rosetta 800, as well as Lavry 4496's.
I have XRL jacks out in the studio, and these jacks have their cable run through the wall, and into the control room, where they terminate to a patch bay. I also have the ins and outs of my mic pres, and hardware compressors coming up at the patch bay, as well as the I/O's of my converters. Basically, any piece of gear in my control room with an input and output is set-up to be available at the patch bays.
So when I plug a mic into the XLR wall jack out in the studio, I simply connect patch cables at the patch bay, to designate whatever signal flow I want for that session. This is how the "big" studios have done it for decades, and once you take the time and money to set this up, you'll kick yourself for not doing it sooner.
I too, do all overdubs here at my studio, Any live tracking with a rhythm section, is always done elsewhere. But aside from me recording my own guitars, I too often work with percussionists, horn sections, string players, and vocal groups, sometimes as many as 6 to 8 singers at a time.
My solution for having separate headphone mixes without a mixer was to get a dedicated headphone system. I ended up going with the Hear Technologies Hear Back system.
http://www.heartechnologies.com/hb/hearbackintro.htm
This system can send something like ten different audio paths to the Hearback satellite mixers, where the talent out in the studio can customize their mix to their hearts content. However, I don't ever give anyone THAT much control . I keep it pretty simple... a stereo mix of the track, a "more me" knob, so they can control their voice or instrument level within the track, and I also give a reverb send, so they can add reverb to their voice or instrument, without me ever having to hear it in the control room.
This system is fed from the separate outputs of my converters, and my Logic template that all songs start from is set-up so that all my Aux tracks are feeding this headphone hub in the manner I described above. So all routing does take place from within Logic.
I do monitor with software monitoring on, but it wasn't until I got the Apogee Symphony system that I was able to do that reliably. Before the Apogee system, I used the Total mix latency free software that came with my RME FF800, much like the cue mix software that came with your MOTU piece. But monitoring in software outside of Logic always felt "left footed" to me. I MUCH prefer the DAW work like a tape machine as far as that aspect goes.
And I do have to admit, it's finally nice to be able monitor through Logic, in real time, with no perceivable latency at all. I went 6 years without a mixer, waiting for technology to allow this type of performance from a native system. The Apogee Symphony card along with the 8 core MacPro finally got me there.
And as I mentioned in that thread you linked to, I use a Presonus Central Station as talkback to the talent, as well as a master control room volume knob, and speaker selector.
All in all, it's working out really, really well.
I will say though, that having a dedicated hardware mixer is not necessarily a "bad" thing, and until you have a system where latency THROUGH Logic is basically undetectable, I would move slowly, and cautiously. It's imperative that the talent be able to hear their instrument or voice as they are used to. Any detectable latency, even a small amount, WILL deteriorate a performance, no matter what any body tries to tell you. And, it makes your studio appear "semi-pro".
Latency free software like Cue Mix or Total Mix can make the mixerless studio a reality though, when throughput latency is detectable. It worked for me for years.
If I can answer any more questions, don't hesitate to ask...

Similar Messages

How to set up a pro headphone mix in logic express 8

using logic express 8 software, avalon tube amp, 4 channel head phone mixer, yamaha O1V digital mixer and running through a line 6 toneport UX2 unit cant figure out how to get a nice individual headphone mix for vocals with effect inserts inside logic express 8 don't care how to achieve it just need to do it thanks

I've copied above post and corrected confusing language. Sorry I was typing too fast.
Cheers
Dee
Hi
To run RMX multi timbrally in stereo mode:
Go to RMX in the environment. Change the MIDI channel of RMX to "all".
Create a multi instrument (using the new menu). It's a good idea to rename the mulit to RMX Multi. Cable multi instrument icon to the RMX channel using arrow tool. (Chose "remove" if a dialogue box promots you to "Remove channels port setting?").
Beginning on channel 8 move backwards and click 8 thru 1on the multi instrument using the Midi tool.
Go back to the arrange window. You should see a track called "RMX Multi 1" (if you renamed the Multi in environment "RMX Multi").
Create a new track for the remaining of the 8 Midi channels (if your Multi requires 8 channels). Also .. create an instrument tracks and assign it to RMX - that is your master RMX track where RMX is instansiated.
You now have a master track where RMX is instansiated and one track for each of your 8 channels within RMX.
You can play/place MIDI data in to each of the MIDI tracks - or drag and drop from RMX into any of these sub channels. You can also use automation on these. For any DSP use the master track.
To run RMX as a multi:
Essentially it's the same thing. Except in the mixer section of RMX assign track 2 to output B, track 3 to output C, etc ....
Go to the environment and create a stereo aux track for each MIDI channel you need in your multi. In the input section of the aux strip assign as follows: for B, 3/4, for C 5/6, etc ... (A is 1/2 and is where you have your instance of RMX).
The only advantage of multi mode is if you need discreet DSP on individual tracks.
In Stereo multi timbral mode you can freeze tracks.
Hope this helps.
Let me know.
Cheers
Dee

Correct way bounce a master mix from Logic?

I've been trying to bounce a master mix straight from Logic using AdLimiter (and in some cases multipressor) in the master output. I must be doing something wrong, because I always end up with totally mashed, pumping and overdriven bounced mixes, even with just a 2-3 dB AdLimiter gains.
I've previously used setup on PC where I bounced a dry mix, loaded the bounced mix to Soundforge (or similar editor), normalized to 0dB, then added Waves C4 and L1 (in that order) and used maybe 5dB gain with L1. Then render.. and that way I got a nice loud "master mix".
I now don't have access to waves plugins nor an external wave editor. So I figured I could do pretty much the same in Logic, if I monitor the mix, so that the master output doesn't clip, but peaks stays at around -0.5dB, so that I don't really need to normalize. Then I add Multipressor and AdLimiter in that order and start tweaking. But I'm not doing something right here, because now I get a very quiet (and limited?) signal, because I have previously had to lower the master output volume to prevent clipping. Any gain over 1dB I do to AdLimiter makes the mix sound like crap. Plus the whole mix is now -6dB or so, because of the previous volume limiting to prevent clipping.
That was a bit of a long story, but I hope someone understands the question I tried to ask.
But in short, I'd like to bounce a nice mix which is compressed and limited to be around -5 to -7 dB louder than the original mix. What is the correct way to do this?
Thanks!

But in short, I'd like to bounce a nice mix which is
compressed and limited to be around -5 to -7 dB
louder than the original mix. What is the correct way
to do this?
You might want to check out other mastering plugins. I'm not a big fan of the AdLimiter. Universal Audio's UAD1 (http://www.uaudio.com/) with the Precision Limiter gives very nice results. There are also some nice compressors available for the UAD1. Another possibility is buying the waves L3 to do the limiting, but this plugin is relatively expensive. Both options aren't cheap, though, but they will give better results than the Adlimiter.
by the way: Audacity (http://audacity.sourceforge.net/) is a nice freeware audio editor. Nothing fancy, but it works.

Bouncing a Quadrophonic mix from Logic Studio

I have a quadrophonic mix ready in Logic Studio and am having trouble bouncing the individual files. Playback is correct so the audio is routing where it should, but when I go to 'bounce' I get a single file called 'surround' instead of four individual files to bring into Compressor.
Help?
Message was edited by: Jamie Drouin

First let's just make absolutely sure that you have assigned your outputs correctly in prefs/audio/surround.
If I were you, I would make a session in logic that had unique sounds going to each output, something like a Kick drum to the front left, then a snare to the front right, then a closed hihat to the left rear, and an open hihat to the right rear. Bounce it, and see if everything landed where it was supposed to, or if, for example the closed hihat, which was supposed to be in the left rear shows up in the left front. That would indicate a different situation than if the hihat landed in the center channel.
Also ,make sure your "bounce extensions" match your surround routing in the same prefs page, one tab over.

How to set up headphone mix logic pro x

Trying to figure out how to set up headphone mix in Logic Pro X but cannot get stereo sends.

Maybe this will help, in Logic Pro 9 there was a box under preferences/audio called universal track mode. If you unchecked this box then when settng a send or creating a aux track you had the ability to select mono or stereo busses. Logic Pro X doesn't seem to have this. So when creating a aux channel with output stereo 5-6 I can't select stereo thus I can't create sens to the left or right side of this aux channel for headphone mixes. In Logic Pro 9 I can create a stereo aux channel with bus 1-2 output 5-6 (to headphone amp) I can then use sends to send musical instrument to bus 1 (left side) and vocals to bus 2 (right side). I hope this helps I just want to know if this function is still in Logic Pro x or we have to go back to OLD SCHOOL way of doing this.
Thanks

Make A headphone mix for vocalist using saphirepro14 interface

Hi. Can someone tell me the best way to set up a headphone mix in logic x. I have a saphire pro 14 interface and was wondering what the best way to be able to increase/decrease volumes of headphone mix for vocalist while not affecting the level of the overall mix. Any ideas or techniques would be greatly appreciated...i have seen videos talking about bussing out sends, but was wondering if there is a better way or anything more specific for the saphire pro14 interface...thanks

Here's an (admittedly busy) example of how I do it:
Note that all of the MON1 sends go to the channel strip marked Mon1(9). The output of this bus goes to Pamp1 (RED - right after the stereo outs), which is actually my RME's output 3&4. I labeled in Pamp1 or Phamp1 (sorry for the inconsistency) because that physical output goes to one of my headpone amps.
Just for additional info, the output (RED) marked 'Phone B' goes directly to my RME's embedded headphone output (which happens to be output 9&10 on my RME).
Hopefully this didn't confuse things further.

Headphone mix / software monitoring question

I am making the transition from Cubase to Logic Pro X. I have set up a headphone mix in the time-honoured manner, sending a (summing stack) instrument arrangement to an aux channel that goes to outputs 3-4 and drives a separate headphone amp, in this case for the vocal talent. So far so good. But now when I enable software monitoring on the vocal track I get no input signal to the headphone mix. Presumably software monitoring in Logic folds back to outputs 1 and 2 by default, hence no signal to 3-4. Cubase has something called Control Room, whereby you can send any monitored input to any pair of outputs. How does this scenario work in Logic? If I use outputs other than 1-2 for a headphone mix, is software monitoring therefore not available? I know I can send a monitored input directly via my interface, but that gives a dry signal to the vocal talent - not ideal and the whole point of software monitoring. I don't have any external reverb so software reverb inserts are the only option.
Sorry for the long ramble and hope this makes sense. Please let me know if I am missing something 101 in Logic. Thanks!
Terence

I am making the transition from Cubase to Logic Pro X. I have set up a headphone mix in the time-honoured manner, sending a (summing stack) instrument arrangement to an aux channel that goes to outputs 3-4 and drives a separate headphone amp, in this case for the vocal talent. So far so good. But now when I enable software monitoring on the vocal track I get no input signal to the headphone mix. Presumably software monitoring in Logic folds back to outputs 1 and 2 by default, hence no signal to 3-4. Cubase has something called Control Room, whereby you can send any monitored input to any pair of outputs. How does this scenario work in Logic? If I use outputs other than 1-2 for a headphone mix, is software monitoring therefore not available? I know I can send a monitored input directly via my interface, but that gives a dry signal to the vocal talent - not ideal and the whole point of software monitoring. I don't have any external reverb so software reverb inserts are the only option.
Sorry for the long ramble and hope this makes sense. Please let me know if I am missing something 101 in Logic. Thanks!
Terence

How Do You Create A Headphone Mix

Hi Everyone,
What I actually mean to ask is this: I've been reading the Logic Pro Book (great book, by the way) and in it he gives an example of creating a headphone mix which basically involves designating a separate output (7-8 in my case), then creating a bus to route the individual tracks to (via the sends on the channel strip) and then out to the outputs assigned for the Headphone mix. My problem is this, he suggests (and rightly so) that the Headphone mix sends should be set Pre Fader so as to be able to adjust the volumes of each track individually but when I do this I don't hear the FX on any of the channels (due to the fact, I assume, that they are set Post Fader). So my question is how to route the sends so that I can hear the effects and still route them through the designated output. I knew this would be a problem before hand but I thought that I could solve it. However, as I'm still trying to figure out that if it actually was Eric Clapton that shot the Sherrif, then who did shoot the Deputy so I don't have as much time on my hands as I thought I would so any suggestions (either about the Head phone mix problem or who the actual shooter might be in the case of Eric Clapton) would be greatly appreciated. Thanks as always.

The slaying of the deputy has been a mystery for many, many years. My personal opinion is that, like the theories surrounding the JFK assassination, there was a second shooter.
As to the headphones...It's early here, so forgive me if I'm not thinking this through completely... but if all your tracks outputs were assigned to busses, and then the inputs of Aux tracks set to those busses, and then the sends on those Aux tracks were set to pre fader, and assigned to another set of busses that fed the outputs for the headphones...
Simply put, if all your audio and instrumental tracks were set to output to buss 1, and buss one was then set to No Output. Then on Aux 1, set it's input to Buss 1. Then on send 1 of that Aux track, assign send 1 to buss 2, pre-fader.
Then set buss 2's output to the outputs feeding the headphones.
I know that will work, but will it include the FX from the sends that are being used on your individual tracks?
Like I said, it's still early here, and maybe I'm not thinking that through completely. Perhaps it's the deputy mystery that's clouding my judgement...

Latency Between Outputs After Project Conversion From Logic Pro 9

hiya!
i am experiencing a bizarre kind of latency – between outputs. for example, audio assigned to stereo outs 13-14 will leave my system 120 or so ticks after audio assigned to mono outputs 3 or 4. sometimes mono outputs are out of sync with one another too. there's no apparent rhyme or reason, and the latency itself seems to shift – even with plugins turned off and removed from the channel strips. if i have an aux that's outputting to channels 7-8 and do a send to an aux that outs to 13-14, this audio should be perfectly aligned, but it just isn't.
i experience this problem in sessions converted from logic 9 to logic x, though sometimes it seems to happen within sessions started in logic x. i can't for the life of me figure out why it's happening, but do believe there's some sort of error being introduced in the conversion. i can't find anything anywhere on the internet about it.
one temp fix i found was to insert an I/O plugin into the problematic aux, turn it on with an out assigned, then turn it off. sometimes that's worked, and sometimes it hasn't. i've been able to use delay compensation on certain tracks, but it's not an exact science, and sometimes the compensations themselves need to change in order for the tracks to sound aligned. it's beyond aggravating. in some situations, i've had to re-record entire sessions. not tight!
to be clear, this is not about monitoring latency, or having to slide tracks into alignment after recording, or the waveforms on the screen not matching up with what i'm hearing. this is not about plug-in latency. the situation i'm describing happens when plugins are entirely removed: audio that is supposed to be leaving logic at the same exact time via different outputs is somehow not aligned.
my signal chain:
logic pro x.0.7
mac pro quad core (trash can) into -->
apogee symphony 16x16 -->
neve 5059 16x2x2 -->
api 2500 -->
back into the symphony
i've been assured by both apogee and neve and sweetwater that this is not their issue--it's logic pro x. i'm inclined to agree, since it seems to be happening before the audio leaves logic.
please help! many thanks for anything!

godmodeinternet wrote:
thanks for your reply!
to answer some of your answers:
-i have all my channels go through auxes, in all projects, but i did attempt to route audio straight off the channels and encountered the same situation i described above.
-plug-in latency is likely set to ALL at all times, since i often track while in the 'mixing stage'. this is how i've always done everything.
ugh. the worst!
do you think there's anything in environment i might be able to swap around that might have gotten screwed up in the conversion?
thanks again.
I don't think it's an Environment issue, however, quite a few people have reported some corruption when importing Logic 9 projects, it usually takes the form of random Inspector Settings for Delay/Velocity and Preferences/Audio - Recording Delay being set to something other than 0 <zero>. This doesn't sound like your problem. Also sounds like your Plugin Latency setting (ALL) isn't the problem.. although if you're recording while mixing it should be set to Audio & Software instruments.
Have you used your hardware system successfully on an earlier version of OSX and Logic 9? If so then the answer is apparent, especially so if you notice it on projects started in Logic-X (current system) although it my take a certain level of project complexity before it shows up.

Streaming audio from Logic 7.2

Has anyone found a good way to stream audio from Logic to another computer via the web? I'm tweaking a sound design mix for a client in another city and would love to be able to get them on the phone and have them be able to participate in real time. Where's the Rocket Network when you need it? : )Thanks.

2 solutions:
1. Old school: get a telephone coupler and send the mix down the phone line.
2. iChat: feed your mix into a small mixer, open a mic when you want to talk, and feed the mix into your "line input" on iChat (make sure it's selected in preferences) and do an audio chat.
DP800 G4/1 Ghz Ti PB 15" Mac OS X (10.4.5)

How to export multiple separate tracks as one song from Logic to iTunes?

I know how to export the separate tracks that i have recorded, but i want to be able to save all of the tracks into one song, in other words, i want to be able to take my recording from logic 8 and export the entire thing, not just separate tracks, into a file that iTunes can read. Any help/suggestions?

Select "Bounce" (upper right part of Logic arrange window) and make sure MP3 is check under "Destination". This will cause all tracks to be written to a single MP3 stereo file. However, if you have any tracks muted they will not be included in the bounce. Also, make sure no track is "Solo" else you will only get THAT track.
I am assuming that you have mixed all the tracks and done any "mastering".
Once the MP3 file is generated just drag it into iTunes, right click on the song, and then select Get Info and the "Info" tab to assign Artist/Album/etc.
It is very easy......

Can't send mono outputs from Logic 9 to my Motu Ultralite

Hi all, any help would be most appreciated. I want to send mono channels from Logic 9 to my Soundcraft mixer via my Motu Ultralite in order to try out some live dubbing. When I set a channel to mono nothing comes out of the Motu. I can make it work by panning things left and right but surely there is a better way?! I went through all the mono outputs but none seem to work.
Any ideas?
Cheers
Jamie

Work
I have a Powermac G5 at work and tried with the Mail Client and with Entourage. Both were successfull.
Home
My ISP at home is All2Easy.net, but I also recently signed up for MSN, I'm not sure if you can even set up the computer to send through MSN though. I have tried smtp.email.msn.com but can't get it to work (not sure if it is because I can't send anything, or if they are the wrong settings)
I have it set to "log-in requires authentication". The settings hadn't changed when the problems started occuring though. An interesting new development is that I can't connect through All2Easy anymore (I chose them for the price).

Creating a headphone mix in LP9

Hello folks,
How do I create a 'More me' headphone mix with a multi-output audio interface using Logic Studio.
Thank you!
Boe

You might want to search:
MacProVideo Tutorials
28 Cameo - Jonathan Perl - Creating Headphone Mixes
or
http://youtu.be/CbgNQsvUxXQ
Kind regards.

Multipressor bug (from Logic 7 to Logic 8)

Logic 8's new multipressor does not read the Gain Change levels that were set when using its ancestor in Logic 7...
While being more precise, I would have liked to keep my fine-tuned (after hours of work) settings from 7 to 8... Any work around or way to retrieve them easily?
Help appreciated
Giac

I have been so discouraged about the undocumented changes from Logic 7 to Logic 8 and the enormous amount of time that I had to spend to trace these that I did not touch my computer for 9 months...
I use the Multipressor on the Main Out and I really like it (some purists say I should not do this, I like it). I was wrong that my Multipressor settings in Logic 7 were wrongly read by Logic 8. The settings are correct BUT I identified the issues why my current song sounds much less punchy than with Logic 7 :
The Multipressor plugin has been completely rewritten it seems, especially on the "Make up" functions and the Autogain.
It seems that the new "Make up" ranges have been done in a non-linear way. What is the compensation I should apply to each of the 4 bands? I have found, by ear, that it should be:
* Band 4 : +9db
* Band 3 : +8db
* Band 2 : +12.5db
* Band 1 : +16.5db
And what is the exact effect of the Autogain?
In Logic 7, when you suppressed the bands one by one, when all were turned off you could not hear anything left. With this new Multipressor I still hear the source sound coming through. Why?
This is driving me insane as I cannot find the fantastic punchy sound I had for that song in Logic 7.
Fermusic wrote : "the malfunction are usually related with the BUSS... in LP8 the BUSS has been unified in the Auxiliary channel
So when you open a LP7 song ... the first action is: open Environments Audio Mixer window and check about AUX and BUSS!)
if you have Multioutput instruments used in LP7 project you can get some malfunction..."
Could this be an additional or related problem? What should I check about AUX and BUS?
please help me before it drives me mad
ciao
Giacomo

Master volume adjustments aren't translating when i bounce projects from Logic Express, any tips?

hi folks!
whenever i bounce a track from Logic Express it's always significantly quieter than any other aiff, mp3 or wave file i have, including those that i have bounced from garageband. Even if i crank the 'master' volume in the mixing window, it sounds loud in Logic, but the bounced file is a big 'wanh wanh' and has absolutely no change in volume.
i must be doing something stupid.
is there another place i can adjust the master volume that the bounce will read?
does anyone else have this problem?
i'm pulling out my hair.
any suggestions would be very welcome!
thank you!

Hi,
first of all: The "Master" Fader is for trimming your mix to your monitors only. The bounce will be taken from the "stereo output" channel, so look there for the correct volume.
Now for the loudness issue: Garageband has kind of a built in foolproof bouncing. Normally, you would have to adjust not only volume to be an loud as you wish, but you have to insert some plugins into the stereo output channel to ensure that the signal doesn't get any louder than 0db, because this will produce clipping and with raising volume, distort the sound. But:
You can raise the volume if you insert the LIMITER plugin as very last plugin into the stereo output channel. Make sure it is the last plugin so there are no changes to loudness after it, not even an eq!
Any change of loudness before the limiter will be limited (hence the name) to a maximum of 0db by default (you can change the value - i use -0.1 db just to be sure.) while the softer parts of the music are being raised in loudness. If you listen to it, you'll notice the whole song is louder without distorting. Of course, if you overdo it, you'll hear the squished dynamics. Experiment!
Garageband seems to have a built in preset to perform these tasks fairly well. In Logic, you have the freedom of using or not using these settings and to customize them, which is quite normal for a pro tool.
Fox

Headphone Mixes from Logic

Similar Messages

Maybe you are looking for