Reverb Tail & ER customization...

Hi,
I'm setting up an orchestral template and am interested in programming "early reflecting" Space Designer IR's for Section auxillary channels (i.e. Strings, Brass, etc.) in order to enhance 'placement'; and a reverb 'tail' on a subsequent aux - (or master out)- to put everything in the 'same room'.
But I'm not especially experienced with customizing Sound Designer IR's - and reading the manual hasn't helped much in this case.
I'm not finding much via searches here, on the Web or on the Logic Pro Help site.
Can anyone advise & or point me to useful tutorials on the topic(s)?
Thanks, Wm

Maybe you want it to sound like a bunch of fake people in a real room?
A place to start might be like this:
For this starter kit just use a couple SpaceDesigners with the same preset.
The instruments that are real close to you would get more pre-delay. As much as you can dial in without sounding weird/slap-backy. So those would all go to aux(1) with that on it.
Next away dial the predelay a bit shorter on aux(2).
Further yet you would dial in even shorter predelay on aux(3).
The idea is that it takes longer for the near instruments to travel to the walls and back. That's what predelay is for, you just don't want to separate it so much that it sounds like a short echo instead of reverb - different attack characteristics will require different settings. Farther away (towards the back of the room) the direct sound arrives closer to the same time as the reflections - so less pre-delay.
I'm sure even more realism is possible by combining short early reflection type of settings along with rooms by using more auxs, or putting the early refs on the auxes and sending those to the rooms, but this may help you visualize how to start applying things and experimenting.

Similar Messages

Batch Processing, Reverb Tails & Video Tracks

Hi Adobe. Love Audition CS6 ! Good Stuff !
But,
Really really need to be able to add the reverb and delay effect tails on to files while batch processing!
It's pretty much the only thing that stops me dead in my tracks while adding reverb and room ambience or any sort of delay effect on to vocals. Currently If I want to have anything with a reverb tail I need to manually add sielence to each file. when you have 1000s of variations this is not ideal.
A little tick box on the delay or reverb plugin would be awesome.
Also how come there is a limit to only 1 video per multi-track session? Protools does this too (unless you pay for the production toolkit) but Reaper and Ableton Live don't. This is a bit of a pain as sometimes your working to several videos and it's great to be able to do it all in the one session for the project.
I really don't understand this limitation.
A better system for arranging editing and renaming presets and favorites would be really cool too. It's a bit of a pain not being able to make category folders for those and put what you end up making related to projects in there. Or even build up a collection of good working types such as vocals, instrument settings ect...(Like iZotope Alloy does currently in it's preset manager)
I hope you consider these as options in a future release.
Cheers !
JR

Hi,
yes, I know. But I am looking for a way how to automatize this task. I am talking about a thousands of short movies and I need to process their audiotracks. I don't want to do it manually ...
Peter

Reverb-Early Reflections & Reverb Tail...

Hi,
I'm setting up an orchestral template and (as advised) am interested in programming "early reflection" Space Designer IR's for Section auxillary channels (i.e. Strings, Brass, etc.) in order to enhance instrument 'placement'; and a reverb 'tail' on a subsequent aux - (or master out)- to put everything in the 'same room'.
But I'm not highly experienced with customizing Sound Designer IR's - and reading the manual hasn't helped much in this case.
I'm not finding much via 'searches' here, on the Web or on the Logic Pro Help site.
Can anyone advise & or point me to useful tutorials on the topic?
Thanks, Wm

Maybe you want it to sound like a bunch of fake people in a real room?
A place to start might be like this:
For this starter kit just use a couple SpaceDesigners with the same preset.
The instruments that are real close to you would get more pre-delay. As much as you can dial in without sounding weird/slap-backy. So those would all go to aux(1) with that on it.
Next away dial the predelay a bit shorter on aux(2).
Further yet you would dial in even shorter predelay on aux(3).
The idea is that it takes longer for the near instruments to travel to the walls and back. That's what predelay is for, you just don't want to separate it so much that it sounds like a short echo instead of reverb - different attack characteristics will require different settings. Farther away (towards the back of the room) the direct sound arrives closer to the same time as the reflections - so less pre-delay.
I'm sure even more realism is possible by combining short early reflection type of settings along with rooms by using more auxs, or putting the early refs on the auxes and sending those to the rooms, but this may help you visualize how to start applying things and experimenting.

8.0.2 horrendous bug still exists!!!!!!!!!!!!!

The effects and sample caching bug is horrendous in my opinion. It's been there for years. It wastes huge amounts of my time. While there are some things you can do to minimize these glitches, there are NO 100% reliable work arounds. If you work in Logic and want a professional result you absolutely must listen to the finished bounce very carefully for glitches. These glitches come in many forms and will often times only be noticed after you receive the finished product from the mastering house.
The main culprits are:
1. Reverb and delay tails. If you have reverb or delay on a track and you stop playback and move your position, the reverb or delay will begin where it left off.
2. Samples. For example I often replace kicks and snares with samples. When I go to start a bounce random samples will play at the beginning of the track.
3. Clicks from compressor and EQ plug ins. These are less obvious, but generally mastering brings them out loud and clear after it's too late.
Keep in mind that these glitches don't generally happen until a split second before a region starts. That means it is very common for them to happen in the middle of the song. Starting and stopping playback a few times does NOT completely clear the cache.
It's a total craps shoot and the most embarrassing, time wasting aspect of Logic.
I really wish I could trust Logic to give me a bounce that is exactly "as expected."
I just sent a bug report... I encourage you all to do the same. They are obviously working on stuff and I think it's our responsibility to make it a priority for them.
http://www.apple.com/feedback/logicpro.html

Hi Rohan,
I meant to begin my last post "with all due respect". I read you all the time, you are always well informed, plus you came up with the 8.02 key command fix/discovery..
Rohan Stevenson1 wrote:
hi pancentre,
no mate that's not correct re logic flushing the buffer. that is actually my point - it can't, not the way the audio engine is designed - god, but my memory is sketchy on this - i remember getting the ins and outs of this a long time ago...
Right, I understand exactly what you're saying, this is indeed the way Logic works... my point was it's an archaic system that was devised at a time when computers had 1/1000th the processing power we have now. There's no need for such great efficiency, plugins can be kept partially active, this is how 9 out of 10 DAWs flush the buffers, the plugin is always active to some extent.
one of the things that makes logic efficient is the way it handles plugs - it doesn't process them the whole time - only when there is sound to process.
Agreed, see above. I think this is related to a couple of Logic's other quirks.
Efficiency in one area is causing less efficiency in other areas, the overload messages. What happens when a song is playing and it comes upon three regions with a couple of space designers and an instance sculpture? Instead of these plugs being partially active and using a little processor, they all become active at once and often cause a spike... ie; system overload or other kind of glitch.
it is the 3rd party plugs that have to flush its own audio buffer - its not something logic can effect without changing the way it works to become much less efficient.
but if you were to call it a 'logic bug' then i would find it hard to disagree with you:
Well, I guess it's hard to say but it seems like Apple came up with a workaround for Logic's behaviour and then asked everyone else to do the same. Until Apple version 7.xx Logic's plugins used to do the reverb tails and all other manner of little tics & glitches.
There's still a couple of Logic's plugs that aren't all the way compliant.
pancenter-

9.0.0 horrendous bug still exists!!!!!!!!!!!

Just reintroducing my pet peeve with Logic in general. It's the one thing that frustrates me most about Logic and it still persists after many many years. Here's my original post from 8.0.2:
http://discussions.apple.com/thread.jspa?threadID=1531297&start=0&tstart=129
The effects and sample caching bug is horrendous in my opinion. It's been there for years. It wastes huge amounts of my time. While there are some things you can do to minimize these glitches, there are NO 100% reliable work arounds. If you work in Logic and want a professional result you absolutely must listen to the finished bounce very carefully for glitches. These glitches come in many forms and will often times only be noticed after you receive the finished product from the mastering house.
The main culprits are:
1. Reverb and delay tails. If you have reverb or delay on a track and you stop playback and move your position, the reverb or delay will begin where it left off.
2. Samples. For example I often replace kicks and snares with samples. When I go to start a bounce random samples will play at the beginning of the track.
3. Clicks from compressor and EQ plug ins. These are less obvious, but generally mastering brings them out loud and clear after it's too late.
Keep in mind that these glitches don't generally happen until a split second before a region starts. That means it is very common for them to happen in the middle of the song. Starting and stopping playback a few times does NOT completely clear the cache.
It's a total craps shoot and the most embarrassing, time wasting aspect of Logic.
I really wish I could trust Logic to give me a bounce that is exactly "as expected."
I just sent a bug report... I encourage you all to do the same. They are obviously working on stuff and I think it's our responsibility to make it a priority for them.
http://www.apple.com/feedback/logicpro.html

lwilliam wrote:
It's an inexcusable bug. It's one of the few reasons left why I can't use it in front of clients. How embarassing to have to trim the beginning of a bounced file to get rid of a reverb tail!
1. If it were truly an AU-only bug, then wouldn't Cubase also exhibit the same behavior? I've not heard of Cubase users complaining about that.
2. If were really an plugin not following the AU spec, then why do these 3rd party plugins pass the validation test?
Since I go way back with this... I can tell you this behavior started with version 4.xx when Emjk adapted Logic to use VST instruments. At that time all of Logic's plugins (reverbs...internal synths) exhibited these problems as well as 3rd party VST. it was a KNOWN issue and had to do with the manner in which Logic handles buffers and and unhooks plugins/AU from the CPU at stop. Plugins, VST & AU become inactive with data in their buffers. It was never a problem with any other DAW as the plugins are kept active long enough to empty the buffers. The onus being put on the plugins was added to the AU spec at a later date to cover Logic's behavior. (which really P.O.'d some plugin developers, so much so that some won't even fix what to them is not their problem.)
I don't know about the latest version of Garageband but interestingly enough, it doesn't exhibit this problem, or if it does, nowhere near as bad as Logic. It was something that was "fixed" in Garageband.
I dislike when companies try and rewrite history, as mentioned, this was a
-known- issue. It was discussed between users and programmers... (back in the day).
over and out on this.
pancenter-

How do I turn off part repeat in record?

Hi this is probably very simple but it's driving me nuts. Every time I record a software instrument part it repeats that recorded part in loop mode until the end. How do I turn this function off? thanks for any help.

Hey Al,
that makes sense for long or live recording (I don't think it puts too much stress on processor).
but as you notice it's not pratical.
There's a white & transparent square around the last number on your session (srcoll all the way to the right) tha's the ending point; it means that the playback and recording stops there, as you know,
it should be a couple of bars ahead of the last note to not cut the reverb tails or delays.
After determining the time you want to record, drag it back to that number, also you can drag the correspondent number under the tempo, say; I'm working on a small song around the 3 minutes, with the project tempo at 120 bpm, 100 bars it's cool for me.
Cheers,
Jorge

Can I mix down to 32 bit at a higher sample rate than 44.1 kHz ?

When I use Ableton Live, it lets me choose 16, 24, or 32 bit, and then I can choose a sample rate all the way up to 192000. Is this possible in Audition ? I have been going through all the preferences and all the tabs and I can't find this option. All I find is a convert option, or the adjust option. But that's not what I want. I want to mix down this way.
The closest thing I found is when I go to "Export Audio Mix Down", I can select 32 bit. Then there is a box for sample rate, with all the different values. But it won't allow me to change it from 44100.

JimMcMahon85 wrote:
Can someone explain this process in laymens terms:
http://www.izotope.com/products/audio/ozone/OzoneDitheringGuide.pdf ---> specifically Section: VIII "Don't believe the hype"
I don't read graphs well, can someone put in laymens terms how to do this test, step by step, and where do i get a pure sinewave to import into audition in the first place??
UNbelievable: So I have to first run visual tests using a sinewave to make sure dither is working properly, then do listening tests with different types of dither to hear which I like best on my source material, and then for different source material it's best to use different types of dither techniques???... Am I getting this right???...
Hmm... you only need to run tests and do all that crap if you are completely paranoid. Visual tests prove nothing in terms of what you want to put on a CD - unless it's test tones, of course. For the vast majority of use, any form of dither at all is so much better than no dither that it simply doesn't matter. At the extreme risk of upsetting the vast majority of users, I'd say that dither is more critical if you are reproducing wide dynamic range acoustic material than anything produced synthetically in a studio - simply because the extremely compressed nature of most commercial music means that even the reverb tails drop off into noise before you get to the dither level. And that's one of the main points really - if the noise floor of your recording is at, say, -80dB then you simply won't be hearing the effects of dither, whatever form it takes - because that noise is doing the dithering for you. So you'd only ever hear the effect of LSB dither (what MBIT+, etc. does) when you do a fade to the 16-bit absolute zero at the end of your track.
Second point: you cannot dither a 32-bit Floating Point recording, under any circumstances at all. You can only dither a recording if it's stored in an integer-based format - like the 16-bit files that go on a CD. Technically then, you can dither a 24-bit recording - although there wouldn't be any point, because the dither would be at a level which was impossible to reproduce on real-world electronics - which would promptly swamp the entire effect you were listening for with its own noise anyway. Bottom line - the only signals you need to dither are the 16-bit ones on the file you use for creating CD copies. And you should only dither once - hence the seemingly strange instructions in the Ozone guide about turning off the Audition dither when you save the converted copy. The basic idea is that you apply the dither to the 32-bit file during the truncation process - and that's dithered the file (albeit 'virtually') just the once. Now if you do the final file conversion in Audition, you need to make sure the dithering is turned off during the process, otherwise all the good work that Ozone did is undone. What you need to do is to transfer the Ozone dithering at the 16-bit level directly to a 16-bit file proper, without anything else interfering with it at all. So what you do with Ozone is to do the dithering to your master file, and save that as something else - don't leave the master file like that at all. After saving it, undo the changes to the original, in fact - otherwise it's effectively not a file you could use to generate a master with a greater-than-16-bit depth from any more, because it will all have been truncated. Small point, but easy to overlook!
just what's the easiest way to test if a simple dithering setting is
working for 32-bit down to 16-bit in Audition?... Why is there no info
about dithering from 32 bit to 16 bit (which is better then dithering
from 24-bit isn't it)?
I hope that the answers to at least some of this are clearer now, but just to reiterate: The easiest way to test if its working is to burn a CD with your material on it, and at the end of a track, turn the volume right up. If it fades away smoothly to absolute zero on a system with lower noise than the CD produces then the dither has worked. If you hear a strange sort-of 'crunchy' noise at the final point, then it hasn't. There is info about the 32 to 16-bit dithering process in the Ozone manual, but you probably didn't understand it, and the reason that there's nothing worth talking about in the Audition manual is because it's pretty useless. Earlier versions of it were better, but Adobe didn't seem to like that too much, so it's been systematically denuded of useful information over the releases. Don't ask me why; I don't know what the official answer to the manual situation is at all, except that manuals are expensive to print, and have also to be compatible with the file format for the help files - which are essentially identical to it.
Part of the answer will undoubtedly be that Audition is a 'professional' product, and that 'professionals' should know all this stuff already, therefore the manual only really has to be a list of available functions, and not how to use them. I don't like that approach very much - there's no baseline definition of what a 'professional' should know (or even how they should behave...), and it's an unrealistic view of the people that use Audition anyway. Many of them would regard themselves as professional journalists, or whatever, but they still have to use the software, despite knowing very little about it technically. For these people, and probably a lot of others, the manual sucks big time.
It's all about educating people in the end - and as you are in the process of discovering, all education causes brain damage - otherwise it hasn't worked.

Weird verb behavior...

I just discovered today on one track of a project that the reverb tail was cut before it is supposed. It really stop instantly after 1, 2 or 3 seconds randomly when it's supposed to last 17 seconds in a smooth fade. Looking for the source, I tried to modify the plugin setting (it is a PlatinumVerb), I tried other verbs (GoldVerb, SilverVerb, Space Designer), always cutting really fast. Then I found that reopening the project made the problem disapear but only the first time it was played...the problem came back immediately after. After several attempts, I could bounce it as it should sound and then put it back in my project on another track as a solution. Then I found that putting the reverb on a bus didn't reproduce the problem.
The other reverbs on other tracks acts normally.
Did anyone experienced this kind of bug??

Oh I found the solution.
When converting from GLSL to AGAL the conversor is displacing the locations of the values.
for example:
In the vertex shader: In the fragment shader:
vPosition = v0 vPosition = v1
vNormal = v1
vSomething = v2
I had to create a function that searches for matches on variable names and look if the index has been moved. For now it has fixed the problem.

Leopard take off iWeb

I was installing Leopard to my Mac Mini and iPhoto etc. was ok, but iWeb was off. Why??
My iLife (was coming also, when I by Mini) is 6. Even I doesn't need iWeb very much I'd like it to be in my Mini.

as the guys said, it's embedded with the audio.
but...... logics enveloper plug has been known to do something for this.
if the signals exposed, you got probs, if in the bg, the reverb tails can be masked. i've this and it works, sort of.
"any port in a storm".

Upgraded to 7.2 and totally lost audio output

I wonder if anyone can help.
I have upgraded from 7.1 to 7.2 logic pro. now when i try to open songs i've been working on, i get no audio output at alll, just some nasty clicks and pops.
all tracks look like they're playing and show bouncing meters, but the master output is showing no audio output at all.
this only seems to happen on songs i had started in 7.1 because if i just load up logic and start a new song there's no problem.
i thought an upgrade was supposed to iron out problems not create them!
anyone had the same problem or got any ideas i would appreciate hearing from you.
my system:
G5 dual 1.8 Gig
mac OS 10.4.6
logic pro 7.2
audio interface - Novation X-station

i've tried disconnecting the X-station and running logic on built in audio but exactly the same happens.
after more messing, i've found that there are a whole load of songs which won't play audio, these are all in one folder and are an album project.
other songs in other foldres seem to play OK.
i haven't made any changes to the folder and am totally confused as to what is happening.
when i open these songs and press play, i get a horrible click or pop but then no audio. however when i press stop i get the reverb tail of an audio track.
i thought i'd finished the album which is why i thought it was safe to install the 7.2 upgrade, however now the artist wants a few changes and i can't do anything at the moment.
andy
Do you get any sound from any sessions other than
what you've described?
Does this also happen with internal/BuiltIn Audio?
J

UPGRADED TO 7.2 & STILL WORKING -- KEEPING FINGERS CROSSED!

Just so everyone knows... I just upgraded the other day to iTunes 7.2 and bought an album to test. STILL PLAYING ON THE V3i!! SO FAR SO GOOD -- KEEPING FINGERS CROSSED! THANKS APPLE!!
Compaq Windows XP

i've tried disconnecting the X-station and running logic on built in audio but exactly the same happens.
after more messing, i've found that there are a whole load of songs which won't play audio, these are all in one folder and are an album project.
other songs in other foldres seem to play OK.
i haven't made any changes to the folder and am totally confused as to what is happening.
when i open these songs and press play, i get a horrible click or pop but then no audio. however when i press stop i get the reverb tail of an audio track.
i thought i'd finished the album which is why i thought it was safe to install the 7.2 upgrade, however now the artist wants a few changes and i can't do anything at the moment.
andy
Do you get any sound from any sessions other than
what you've described?
Does this also happen with internal/BuiltIn Audio?
J

Audio Is Randomly Cutting In And Out In Playback

I have encountered this problem on all recent Premiere projects I've worked on with a variety of audio types. My audio is cutting in and out in a seemlingly random way while the video keeps running fine. Sometimes the audio will play fine for 15 seconds then stop, sometimes it stops and starts after about 2 seconds. I have tried adjusting all of my preferences, settings, audio types, and even video types but the problem always comes back. Spoke to a friend who's new to Premiere and he's having the same problem. Considering moving back to FCP despite the weak interface.
Currently I am editing Canon 5d Mkii footage at 1080p, 29.97 fps. I am using none of the audio attached to the video clips - only editing to a separate m4a music track. My audio is at 44100 Hz. I am playing back video at 1/4 quality. I am running a fully updated Mac Powerbook with a 2.6GHz Intel Core i5 and 8gb of RAM.
Please help, almostly completely unable to edit. Thanks.

I am still having the same issue in Premiere CC 8.0.1 (21)
Exact same issue as the original poster on this thread, as well as the original poster on this other thread:
Premiere Pro CC Audio Cuts Out
So this audio issue persists from version 7 now in version 8.0.1!
For me, audio is cutting in and out randomly and never in the same place. If I have multiple audio tracks on the timeline, then one or more tracks might drop-out at the same time. Sometimes all tracks drop out at the same time. It seems completely random.
I am editing with a combination of P2 footage and DNxHD 290 footage. (I originally had some AVCHD footage, but then I transcoded all of it to DNxHD 290 -- but this did not improve the audio issue at all.)
Two of my audio tracks contain the native camera audio (stereo); 1 track has the music bed (stereo .wav files) and 1 track has sound effects (stereo .wav files)
In one of my projects, I had a very strong reverb effect applied to the track (not to the clip). When the audio dropped out, I could still hear the tail of the reverb. This leads me to conclude that it's not the entire audio track that is muting -- it's definitely the source audio clip that's dropping out. The reverb tail continued after the audio dropped out.
Playing the same part of the timeline again produces randomly different results -- audio might drop out on a different track, or at a different time, or not drop out at all.
This bug is making it nearly impossible to edit! I have filed a bug report at http://www.adobe.com/go/wish_fr
During timeline playback, my CPU utilization never goes above 15% and at least 12 GB of RAM is staying free, so I am pretty confident these drop-outs are NOT caused by lack of computer resources. Also, my hard drives seem to be fast enough to play back 4 streams of HD simultaneously, but I am getting these audio drop-outs when playing back 1 stream of HD plus 4 channels of audio. Therefore, I don't believe this can be a hard drive bandwidth issue.
System specs:
Windows 7 64 bit, SP1
RAM: 24 GB
CPU: Intel Xeon X5680 @ 3.33Ghz (12 threads)
VIDEO: NVIDIA Quadro 4000

Logic playing delay effect from instruments at start of song PLEASE READ

Hi there,
I have had this strange problem where when i bounce the song I get weird delay or reverb effects from one of the tracks/instruments that comes in at the begining of the track.
almost like the delay and reverb tail at the end of the track appears at the begining. can anyone explain why logic is doing this?

We were discussing this earlier, it has to do with Logic emptying it's buffers. A quick start stop should get rid of it...

Pre and Post for Sends..Confusion?

Dudes...am I crazy?
I have 2 tracks. One plays a vocal and one plays a guitar. Both are sent to the same Aux or Bus track with a reverb on it, via the sends on their respective channels.
If I solo the guitar track, I still get reverb tails from the vocal track on the bus channel as well as the guitar!
If I switch the vocal track send to "post", then the reverb tails go away for the vocal. By the way, I'm soloing from the mixer channels, not the arrange window.
Now I understand that if it were the vocal track that I had soloed then switching between Pre and Post on the send would dictate if I hear the reverb tails or not for the vocal track.
But that's not the track I have soloed! I have the guitar track soloed. So why would I hear reverb tails on a track that's not soloed. And why would the reverb tail go away on the vocal track if I change the send to Post, when that's not the channel I have soloed.
Am I crazy, or should logic know that if I have a track soloed, it should not play anything else but what's on that track? Regardless of switching from pre to post, if the vocal track is not the track that's soloed, why am I hearing anything from that track at all? I shouldn't have to mute the vocal track or switch it's send from pre to post, just to hear the soloed guitar track by itsself.
Thanks
Noah
G4 Dual 1.42 Mac OS X (10.4.6) 2 UAD Cards/Dynaudio/Martin/U47

You say you love Logic's "pre/post mute send solo safe function". Sounds like quite a function Does that mean it's changeable in Logic? You make it sound like it's an option.
By the way, I understand that if you solo a track, it might be cool to also hear it's effects channel without having to solo the bus track its feeding.
But why would I hear another tracks effects? This only happens when two different tracks have the same bus inserted in their sends. Just because I'm using the same send for two different tracks, doesn't mean I want to hear both sends even if only one track is soloed. That's the part that makes no sense for me. Solo should only address and isolate the track that's soloed and whatever else is related to it (ie...busses/sends/plugs etc..)
I dunno, I guess when you solo a track it keeps open the bus/send it's using unless your exercise the pre/post send option. And if there's someting else using that bus send as well, it must need to stay open for it too.
By the way, for me sometimes Logic will continue to play a certain audio track, usually a single track of it's choosing, even if another track is the only one soloed. There are no busses, no plugs, no sends on it, just a straight track and I still have to mute it after soloing some other track. Now that's not right...pre/post etc...is not in the equasion here, can someone explain that?
Thanks for the help.

Anyone out there creating impulse responses?

Those of you that use sine sweeps to create your impulse responses, when you deconvolve them do they end up being the same length as your sweep file or does space designer create the proper length reverb tail?
When I attempt to create my impulse responses space designer does not create a reverb tail. It just seems to remove the sound of the sweep form the wet file.
Other convolution reverb programs that I have used create a reverb tail automatically and I believe sound designer should do this as well.
If anyone else is having this problem or has a fix for this issue please let me know. If not then if anyone out there is successfully capturing impulse responses using a sine sweep please post detailed step-by-step instructions on how you are able to do it.
At this point I am at wits end. I have contacted apple care only to find out that they don’t support space designer, so they were no help. You guys are my only hope…
Please Help
iMac Mac OS X (10.4.7)

The process is simple, but has many steps and they must be takin in the proper order:
Play the sweep tone into the space or reverb device while recording the output of the space or reverb device. If you use a reverb device, be sure to set the mix to 100% effect.
Be certain that the space or device you are sampling has fully decayed before you stop the recording. If you were sampling a 4 second plate reverb, do not stop the recording for at least 8 to 12 seconds after the sweep tone feeding the plate has ended.
Edit the resulting recording to eliminate 'dead space' before the sweep begins, and remove any excess space at the end of the file. Be careful not to truncate the decay portion after the end of the sweep.
Place the original sweep file and your edited recording of the effected sweep somewhere easy to find, I use a new folder and save them there.
In "Space Designer" press the "Deconvolution" button in the upper right corner. Load the effected sweep recording, then the original sweep signal and then save the resulting IR in a new folder.
Load the resulting IR file using the "IR Sample" menu and save the preset.
I've created hundreds of impulse files using sweep tones and it has always yielded proper reverb tail lengths for the space or reverb I was sweeping. In other words, if the sweep was 12 seconds long and the space I was sweeping has a 3 second RT60, the resulting impulse would be 3 seconds. On some files Space designer will create an impulse file which is much much too long. In these cases I remove the unnecessary minute or so at the end of the file and resave it.

Reverb Tail & ER customization...

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