Uneven output levels using Adaptive Limiter

I am placing an adaptive limiter on the master output in Soundtrack Pro. The levels are fine when I playback the project, but as soon as I try to export, I get slightly higher levels on the left channel (about 1 or 2dB). When I check the exported file, the levels are indeed uneven. It seems to be a bug with Soundtrack Pro as I have had the same problem in all my projects. Has anyone encountered this issue, or know a workaround?

but if i've inserted a limiter with output max 0db at the end of the signal chain (no other plug ins) why should my master fader clip to levels +2 etc? the master fader is at the end of the signal chain no, so should limit it to 0db?
cheers

Similar Messages

How do I use Adaptive Limiter to make sure bounces are same volume?

Hi,
I've been making ear training exercises for myself - you can understand what here:
http://discussions.apple.com/thread.jspa?threadID=490846&tstart=0
Here's what I need to do - and don't know how:
I want all bounced exercises to play at the same volume.
Each exercise has two MIDI regions. The first region (Region #1) plays at the begining of all exercises. When I look at the Output track when Region #1 plays, it is at 0.2 DB - no clipping. However, Region #2 clips in varying degrees depending on what octave an interval is being played at. For instance, a C and E played together around C1 doesn't clip - but transposed to C5 or C6 it clips anywhere from 3.7 to 4.6. I can't just lower the output volume before it clips - because it's important the Region #1 stays the same volume-wise on all of the exercises.
Some people have said just add an adaptive limiter to the Output 1-2 track... but how do I set it?
Any suggestions? Anyone understand this?? It's hard to explain on this end...
Thanks.

you don't use adaptive limiter to make sure bounces are the same volume.
that is not the function of the adaptive limiter.
the limiter puts a ceiling on the signal and optionally raises the gain. You'd use this on certain instruments that are known to produce loud peaks, such as Bass Guitar and Vocal,
It is used as the last plugin in the chain on the master fader when mixing something down, to prevent the mix from peaking.
IMO, the logic limiter doesn't destroy anything unless the mix itself has something wrong with it.
A lot of people reach for limiter when what their ears are really looking for is to multiband compress.
Your solution is to use a COMPRESSOR (not the multiband compressor, just the simple Logic compressor).
Do NOT use the default setting of the logic compressor in your case, because it defaults with its 'auto Gain' turned on which will raise your volume (a lot).
Put a compressor as the first plugin in all of your piano channels, or into a Bus, and set all your piano tracks to output to the bus with the compessor on it.
Settings:
THRESHOLD : -10
ATTACK : 0 (play with it, pianos feel good with a low one,
could try 15ms or less)
DECAY : play with it, I'd use 50ms with a Piano track
AUTO GAIN : off
GAIN : +4
RATIO : try 12.0
This setting will attenuate anything above -10 and will raise the gain up to a healthy level to compensate.
Anything below -10 won't be touched.
Since the compressor will 'squash' your signal slightly,
some frequencies will appear louder to the ear after the piano is compressed. So just AFTER the compressor on the same track, put a Logic Channel EQ and conservatively pull down some of the freq ranges to bring them back to normal.
And that's it.
a tip: When working in logic, no matter what the project, when it's time to mixdown put a COMPRESSOR on every single channel as the first plugin. Then play with plugins, etc.
I do have many issues with Logic's workflow, but the dynamics plugins IMO are very good.

Some DISTORTION using Adaptive Limiter

Hi there,
Wierd! Master Sound like Distortion when apply Adaptive Limiter!
AL apply after Multipressor! E'thing below Overload Peak level.. Gain Level upto 9 to get the level as loud as commercial standard!
What's wrong do u suspected on my case?

You really add 9 dB using AdL? That is too much, try to keep it under 6 dB somehow... Even better (imho) is 'my' rule of thumb: if you 'need' to add more than 3 dB, something's not quite right yet in the mix...
Alternative: assign drums and bass tracks (+rythm section, add to taste) to it's own aux, compress/adlimit that with some/much force, and make the final master compression much subtler (add perhaps only 2 dB or less with adlimiter or similar loudness maximizer plugin.)
The distortion could also come from the combination - how does it sound when you bypass MP? And how does MP sound on its' own?
regards, Erik.

Limiter/adaptive limiter output, still clipped

hey ppl,
I am trying to push up the gain of a track using the limiter or adaptive limiter. despite setting the output max at 0db the track still clips in the channel strip meter. the audio itself was not recorded clipped and the limiter is the only thing in the signal chain....
the only thing I can think of is that the limiter takes a bit of time to kick in so misses the initial clip at the start of a peak. but doesn't that defeat the purpose of a limiter?
any ideas?
thanks

but if i've inserted a limiter with output max 0db at the end of the signal chain (no other plug ins) why should my master fader clip to levels +2 etc? the master fader is at the end of the signal chain no, so should limit it to 0db?
cheers

XI message status at Adapter engine level using a table (SAP table)

Hello Experts,
XI message status at Adapter engine level using a table (SAP table).
We want to write a custom report using ABAP so Pls tell why the status u2018Holdingu2019 and u2018To be deliveredu2019 are present in message monitoring of RWB but not in the status (MSGSTATE) field of SXMSPMAST table.
My need is to write a report to get the messages based on the these status from table level.
Please let me know the table name and field name for this and the table name for the desciption of the status of XI messages at Adapter level.
Thanks
Gopesh

Hi Gopesh,
the Adapter Engine Messaging System messages are on the Java schema,
i.e., see the following -
[XI/PI tables|https://www.sdn.sap.com/irj/scn/wiki?path=/display/xi/xi+tables]
Regards
Kenny

Limiting Output Levels (as in Photoshop)

Hi, I'm submitting my work to several places that require a limit on the output levels of my photographs. Essentially I need to ensure my maximum output level is 252 (instead of 255) and the minimum 3 (instead of 0). Photoshop allows this very simply and easily with the levels palette, but it would seem as though this isn't such a simple task in Aperture. Exporting Aperture to Photoshop for this tiny edit is of course slightly undesireable. Is there any way to do this task in Aperture?

Aperture is not the best tool for this (NB: afaik), but you can get something that may suffice using the "output" control in the Curves Adjustment Brick. (There is no equivalent in the Levels Adjustment Brick.)
Some version of the following:
- Enable the Curves Brick
- Set the range to "Normal"
- Click inside the histogram grid and drag to the bottom left. This selects the far left point on the displayed linear adjustment.
- Click in the output field and change it to the percentage equal to 3/256 (0.012)
- Click again inside the histogram grid and drag to the top right. This selects the far right point on the displayed linear adjustment
- Click in the output field and change it to the percentage equal to 253/256 (0.988)
After the above adjustments you should see the diagonal line in the Curves Brick rotated _very_ slightly clockwise. Your image should appear slightly grayer: your darkest dark is less dark, and your lightest light is less light.
Screenshot showing exaggerated outputs:
You can save this as an Effect Preset and apply it to any selected Images.
As I hope I communicated, there may be better way to do this, and -- even more important -- it may not suffice for your need.
HTH.
Message was edited by: Kirby Krieger -- minor clarifications

2 Adaptive Limiter in series circuit - do they work properly that way?

A few days ago, I noticed that the Adaptive Limiter plugin does crazy
things. Thats the setup I used:
A:
- audio channel with audio file with full volume level range (0dB)
- channel volume set 0dB
- Adaptive Limiter included, all controller set 0db, only out-celling set -2dB
B:
- stereo output with channel volume set 0dB
- Adaptive Limiter included, all controller set 0db, only out-celling set -0.5dB
The audio material in A is limited by the Adaptive Limiter to -2dB and routed to B.
If I play the audio file from A, by switching the Adaptive Limiter in stereo output to "bypass", the maximum stereo output volume level remains -2dB.
If I play the audio file from A while all Adaptive Limiter are activated (in A and B), the maximum stereo output volume level is -0.5dB.
Only by switching the stereo output limiter to "bypass", the stereo output volume level remains -2dB.
How could it be that the Adaptive Limiter in B raises the volume level of A (-2dB) to -0.5dB that way? do they work properly that way?
Signalfloh

If I understand it all correctly, the Out Ceiling parameter seems to work as an expander ? Even if you throw -2dB at it, the second one pulls it up to -0.5?
I wouldn't have expected that, no, I thought it was just a brickwall setting, especially with the word "ceiling" used. From what I understand of your setup it seems to work like some sort of normalisation.
Why have you created this setup? Just to test? And, more importantly, what kind of audio did you throw at it? A test tone, or a (musical) track with (wide) dynamics?
And why do you say it does "crazy" things? You can only conclude that if you somehow know how it is supposed to work, right? I'm surprised at what (you describe) happens, I think it is unexpected, but not necessarily "crazy".
To really observe this from all angles, I think you need to insert three level meters: one as the very first plugin (before any AL), one after the first and before the 2nd AL and one after the second. LM>AL>LM>AL>LM

Adaptive Limiter Clipping

Why does the Adaptive Limiter clip if the gain is under 6.1?
15" PowerBook G4 1.25GHZ; 2GB RAM; 80GB 5400rpm; OWC 250GB (2) Mac OS X (10.4.9) Logic Pro 7.2.3; Edirol FA-101; Roland GI-20; Korg Kontrol49; Mackies HR624

Here is the setting procedure outlined in the manual....
In order to use AL properly, you need to Start the process by adjusting the Input Scale,
just as you would set a mixing desk’s Gain parameter,
or a digital recorder’s recording level.
The parameter behaves much like a Gain control,
but its purpose is to adjust the input level,
which must never exceed 0 dBFS.
Adjust the Gain parameter to musically control the internal process of peak
smoothing and gain increase.
Out Ceiling reduces the output level of the process in very fine steps within a range of
only 2 dB. This is no threshold control, just a simple output gain.
The Mode menu in the Adaptive Limiter’s Controls view allows you to choose between
two different forms of peak smoothing. If you choose:
• OptFit, the signal will be limited by following a linear curve. This form of peak
smoothing allows signal peaks that exceed 0 dB.
• NoOver ensures that the signal never surpasses 0 dB, avoiding distortion artefacts.
Remove DC (Controls view) activates a highpass filter that removes direct current from
the signal. When using poorly constructed audio hardware, direct current (DC) can be
undesirably layered over the audio signal.

Bouncing in Logic, Suppressed and Quiet Volume Output Levels

So I've just finished mixing a song from a friend's band's performance and it sounds great. The Output is hot, usually staying upwards of -3.5 decibels and less than 0 thanks to the compressor/limiter plugin and some basic volume automation. I have the computer output levels set at halfway and the master logic output set at half as well, and the mix is slightly louder than I would prefer comfort-wise in my headphones. By all means, my bounce out of logic and onto my desktop should be at a similar volume level, but it is not. In comparison to my songs in my itunes collection, the output is very significantly affected and diminished. A listener would have to turn the volume up on their stereo or computer to reach an equal volume level as their other music.
I have tried bouncing with normalization turned on and off and have bounced out both aiff and m4a files. My Compressor/limiter plugin is on, but the output never reaches clipping levels as the plugin seems to be working as it should. I have played my bounces in both Itunes (preamp in eq is at half as well) and quicktime with the volumes set at halfway, so it's not a blockhead issue of not having my volume up in those programs. Something is seriously up here, since I have tried many possibilities, my output levels are not requiring a reduction by a plugin and I have not come to a satisfying conclusion.
My overall questions are: 1) Would could possibly explain what's amiss here since nothing theoretically should be. 2) What explains the difference between the output I hear playing my mix within logic and playing the bounce in other programs such as I tunes (again I did not bonehead here and leave my volume or eq level set low in my Itunes, nor did I normalize my bounce). 3) Should I just send my Mix to an auxillary "master" channel strip, record as audio, and export it as is? I would prefer it not have to come to this. 4) Is it possible that one cannot truly master at a professional level in logic? Do I need to get some expensive plugins in addition, or perhaps an external mastering-specific program?
I would sincerely appreciate any informative answers here, Thanks

, and large reason for that is the fact that it always seems to lack any sort of conclusive solution
I think it always yields a conclusive solution.
After all, if my question was "I need to be able to play the guitar to put some guitar down" most answers would be either "You should get a guitarist who has the skills already" or "Learn to play the guitar".
A forum post is not going to suddenly impart the skills necessary to get commercial sounding mixes instantly - the commercial mixers charge a fortune for their work, largely because they are good at it!
If a mix sounds very loud while I'm bouncing in realtime, what can possibly explain a significant volume reduction in the sound file that comes out the other end onto my desktop?
You are comparing two different things. When you are hearing it in Logic, there is a playback level going through your interface and to your speakers. When you play the same audio through iTunes, you are playing through an entirely different chain - iTunes, with it's own volume control, through the system with it's own volume control, and so on. Also in iTunes, often people have all kindsof stuff switched on like EQ settings, or the audio leveller function, which is further going to change stuff.
Just remember that the "volume" you hear is more dependent on how loud your are setting various volume controls than what the contents of the file are. If you're not sure on this stuff, then probably the best way to compare your mixes to commercial stuff is to burn them to CD and play them back on the same system.
Again, I have no need to normalize at all since my mix has a loud average volume throughout without spilling over into clipping territory on the peaks.
Are you using metering? You'd probably find it useful to compare average and peak levels on your mixes, compared to average and peak levels on commercial mixes. I think you'd be surprised how "loud" commercial mixes really are - many don't have more than a few dB of dynamic range - at least in the regular pop/rock/commercial categories.
Presumably, I have exercised all the skill necessary to get a satisfyingly loud mix. If it (the mix) came out sounding exactly as it does in logic it would be louder than I would need it to be.
It's difficult to know what you are doing, but as long as you are not clipping your outputs, what Logic writes into the audio file is exactly what you are hearing when bouncing. How you then play the file back, in a completely different set of applications, is not going to let you listen at the exact same level unless you set all the affecting volume controls to match everything else. Remember, Logic is talking directly to your audio interface, not playing through OSX's system output.
So, divide your situation into two areas - am I getting my mix to the relative levels I want, and how am I playing back and hearing that mix, and go from there.

Adaptive limiter is broken

New Intersample clipping feature does not allow get enough rms level, if you shut it down, Out ceiling is dead.

Ed Egned wrote:
Question is not about what you like, or how pleasant it is.
Au contraire, mon ami, the ONLY question is what it sounds like - this is not scientific equipment, it is ARTISTIC equipment.
As you can see, in left picture I can't get RMS more than ~ -12 with max settings ( intersample clipping on), in right it works as expected, but Out ceiling is dead and output is overloaded ( intersample clipping off). Of course, both inputs are same ( pink noise) . On my system it is broken, can anybody confirm that ? This is not normal brickwall limiter anymore and yes, I prefer, let it be my choice, what RMS get on final product.
1. Your choice? And what would change if I "forbade" it? What if you told me to do or not do something? O wait, you just did: you told me to "let it be your choice". Well, it's my choice to not respect other peoples' silly choices. Etcetera. Futile point.
2. Your test? With pink noise? It is an Adaptive Limiter, NOT a brickwall limiter, and pink noise at +12 dB Adaptive gain, well, I'm no audio-engineer, but obviously neither are you. That's just a test.
I don't think the Adaptive Limiter is broken.

Logic Adaptive Limiter OptFit / NoOver

Hello
Logic 6.3 /Pro
When you switch the adaptive limter plugin from editor mode to slider mode you get one more option to choose from: "OptFit" and "NoOver". What is the difference ?
And why does the adaptive limiter produce "overs" when the out ceiling is set to 0,0 ? The Master channel doesn't show "overs."
Thanks for answering. J.

What Jord said, and…
Apparently there are differences in the AdLimiter versions. 7+ has three control options - DC -Yes, Lookahead (100ms for me) and NoOvers.
If you play through your material, the input side should never go over. Adjust it as high as possible, but never over. Output I usually set to -0.3
Now use the gain to set the amount of peak control and perceived volume. Each source material will be different. Experiment - judiciously. For me, I find it introduces some nice analog-type warmth and fullness, but it can destroy the dynamics of a track if used with a heavy hand.
For me NoOvers works 100% of the time. It's a brickwall. Maybe earlier versions had a bug?

Using Adapter-specific attributes in Mail adapter

Hi,
I try to use adapter-specific attributes in my Mail adapter. I set the "Use Adapter-Specific Message Attributes" indicator, but I do not get a result in the mapping and also in the SXMB_MONI the attributes are not visible.
The documentation also refers to Variable Transport Binding indicator:<a href="http://help.sap.com/saphelp_nw04/helpdata/en/23/c093409c663228e10000000a1550b0/content.htm">http://help.sap.com/saphelp_nw04/helpdata/en/23/c093409c663228e10000000a1550b0/content.htm</a>
Could this be the problem? I cannot find this indiciator! Does anyone know where I can find this indicator?
Regards,
Torsten

Hi,
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We need to write UDF's that can extract the values from this path.
For Ex:
DynamicConfiguration conf = (DynamicConfiguration) container.getTransformationParameters() .get(StreamTransformationConstants.DYNAMIC_CONFIGURATION);
DynamicConfigurationKey key = DynamicConfigurationKey.create(
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    FileName);
String valueOld = conf.get(key);
if (valueOld != null) {
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    conf.put(key, valueNew);
If you check Dynamic configuration in sxmb_moni, you would be able to find the values for all the attributes that have been set in the adapter.
Regards,
Smitha.

Problem in displaying a field in webdynpro using Adaptive RFC model

HI,
I created a webdynpro application using adaptive RFC model and i have a problem in displaying one of the output fields.
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i deleted the model and recreated it still it doesnot work.
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points will be awarded for sure
Bala

Hi Bala,
Whenever you reimport the model server restart is necessary otherwise the changes will not get reflected. You will get an error when you deploy the application.There is no go around except restarting the server as the meta data gets cached in the server as long as the server is runnig.So, Once you restart the server the cache will be cleared.
Check once again whether you are fetching data from correct BAPI field. Still problem persist try debugging.
Regards, Suresh KB

System Sound Output Level Reset After Reboot ?

Whenever my iMac Core Duo is rebooted, system sound output level always get reset to value 0.56 (-24dB) as shown in utility "/Applications/Utilities/Audio Midi Setup". I have to increase the system sound output level to desired value of 0.75(-12.75dB) to match with built-in amplifier of Altec Lansing FX6021 speaker system. The desired setting lasts until next reboot. Is there any way I can fix the system sound output level at desired level between reboots ? Why I have to be involved in such constant struggle with the iMac for such simple audio configuration ?
Audio Hijack Pro 2.6.7 wihtout Soundflower 1.1 virtual audio device has been installed. Audio hijack function of this utility has not been used at all. Thank you for any pointer.

Thanks for the confirmation that my iMac is not an isolated case. In case you are not willing to use Audio Hijack Pro, just use Automator to create an Automator script to set system volume (In Automator's Library plane: Applications>System>Set System Volume) and save it as application. Set this application as Login Items for your account through following menu path:
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iMac Intel Core Duo 20" 2 GHz 2 GB RAM Mac OS X (10.4.7) Firewire External Drive, Belkin Wireless Router, iPod nano

Return a multidimensional array using Adaptive WS model

Hi all,
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Then, I tried using Adaptive Web Service model and tried to bind the values of the output String array in my webdynpro table. I was able to get only the first element of the 2D String array. Don't know where I am going wrong.
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Earlier i used with "wenservice model" and it was working fine. Now that with SP17, "WebService model"is depricated and we are forced to use "adaptive webservice model" and it is giving the problem. the detailed error is :
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Uneven output levels using Adaptive Limiter

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