Parametric Equalization questions

I have a couple of questions related to parametric equalization in AA 3.01.
1) Within AA, is there any harm to using parametric equalization twice on the same audio clip, or will one degrade the audio by doing so. Your answer could be based on EQ, in general, or your knowledge of how Audition works internally, or something else.
I am considering a two-pass parametric eq. The first pass would be for removing problems in a clip, for example, boomy bass. The second pass
would be for modifying the sound in some way, for example, to increase or decrease the gain in different parts of the spectrum. (I am aware that if one implemented a dip in a lower frequency band to reduce bass boom, one would need to be careful not to undo that in the second pass.)
2) Related to this, but not necessarily limited to EQ, are there any known issues for combining Ozone and AA mastering effects. For example, if one found the multiband compressor easier in Ozone (not to say that it is), but felt that the Audition hard limiter was better, can one mix and match or is this approach a recipe for disaster?
Thanks,
Steve

I'll respond to your first question, cuz I've actually experimented with this and come to some conclusions.
Here's an experiment for you to run on your own rig: put an Audition parametric on a track and maybe cut some low mids out of it, say...2 dB an octave wide at 350 Hz. Okay. Now put a SECOND Audition parametric on the same track and use exactly the same settings. I think you'll find that the second instance of the parametric doesn't degrade the audio, but it COMPOUNDS the EQ on the track.
Conclusion? Sure, run two parametrics on the same track. But don't overlap the frequencies that you dial in, cuz it'll compound the impact on the sound. Of course, you might like that... :-)

Similar Messages

Equalizer question....

i dont have a mixer...just a usb interface that i plug my mic into...i was wondering if it was possible to adjust the eq somehow while the instrumental is playing. so i can adjust my vocals as its going....without adding an effect or messing with the eq, then going back into multitrack view to see how it sounds over the beat...

Audition always records dry; it's the only way it can do it - no effects or EQ added. This is good practice, as it means that you always have the clean signal available after you've recorded, and can go back and correct any EQ/effect mistakes you made when you change your mind!
For instance, you might decide that your voice sounds nice with a bit of reverb. If you add this at recording time and then change your mind, you'll never, ever get rid of it. So the norm is to monitor with the effects (which, allowing for a bit of latency, Audition will do), and do everything else in the mixdown.

AA 3 effect results

When one applies affects in AA3, are they non-destructive? I have two questions related to this...
1) In a particular editing session (all I mean is that the audio file is still open and still being edited), if I have already applied an effect and then apply it a second time, is the effect cumulative or is the second application of the effect the same as if I did not apply the effect the same time? Does the answer to this question apply to the effect or to the editing context?
2) How does the answer to #1 change if the audio file has been changed and then reopened. Suppose one were to apply, for example, parametric equalization, close the file, reopen it and redo the parametric equalization? Would one need to start from the beginning or does reapplying the equalization with different values simply ignore the equalization from the first session. And, again, is there one answer for AA, or does it depend on the particular effect?
Thanks.

>And, again, is there one answer for AA, or does it depend on the particular effect?
All effects behave the same way - as far as Edit View is concerned, an effect becomes permanent when the file is saved, as Runaway says. Until then, 'undo' works, but it relies on a temp file being created before an effect is applied (this is automatic). The number of steps you can go back depends on you - and if you are editing very large files, and apply several effects to them, you can rapidly run out of temp file space if you go too far.
You can't edit files in multitrack mode - all you are doing here is editing a series of playback events that don't alter the original files in the slightest, whatever you do; that's why it's called 'non-destructive'.
Oh, and if you double up an applied effect by applying it twice in Edit View, the results are cumulative, yes.

Vintage SB-Live on a new

Dear Friends
I?ve become the lucky owner of a brand new P4 3-GHz PC running Win-XP with SP2 and having PCI-Express capability.
Now comes the question: What to do with my good old vintage SB-Li've model CT4620 that has served me so well over the years? I bought the card way back in times of Win95/98 and now I wonder if it will still work with my new PC or if I should rather put it to rest now and buy a new up-to-date sound card.
If you think that I can use the old CT4620 in my new PC without too many problems I would like to know if I should use any of the old Win95/98 software that came with the board or if I should just download the 23MB driver file from Creative and use only that. If this is likely to work, should I install the software first and then the board after ? of maybe the other way around?
Advise will be greatly apreciated.


Hey dijdoo2,
Back before the Compaq drivers (the package that evolved into the Li've Uni-Pack) I used to run the Creative Launcher fine. The trick was to set it up by turning off auto-hide and undocking it from the edge of the desktop real quick on its first appearance.
I forget exactly which order, but if you can manage to grab a piece of it that is not a tab or link of any kind with your mouse you can move it out of the docking position. Then you can right click on the left and it'll let you enter the properties to turn off auto-hide, docking, whatever. You just grab an end and resize it smaller or larger so it still shows all the functions but won't be big enough to touch a screen edge. With the old SBLi'veXPUpdate.exe driver update, it works fine.
If you install the SBLi've Uni-Pack, most of the functions are removed and the update automatically turns off load on Windows startup. So with the Uni-Pack it's useless. It just has links in it, but no longer can control Surround settings, etc.
Either way, I would install it along with the rest of the cd stuff. If you don't, some of the EAX stuff won't work right as the EAX presets for the coresponding speaker settings are installed with the Launcher. Of course, if you're gonna install the Uni-Pack you don't need that stuff. This is for those who like the SBLi'veXPUpdate.exe driver better. (The pure audio sound is better with these older, un-unified with Audigy, drivers.)
The trick is to get it off the top of the screen immediately. If you fail after one try, you must reboot and try again. If it unfortunatly remembers the failed attempt (it's that tricky), and it's totally unreachable even after you reboot, you must reinstall the Launcher from the cd and then try to do it on the first appearence again! I usually got the hang of it. You only need to mess with it once. It works fine with the SBLi'veXPUpdate.exe driver once you have it not touching the edge of the screen.
Of course, with those drivers you don't get the AdvancedEQ parametric equalizer and you have bugs that were fixed in the Uni-Pack driver.
The choice is yours. I would just install the newer stuff over the cd install and not worry about the Launcher. If you have a full install cd of the newer stuff, you don't even need the original cd installed first unless you have a Li'veDri've. But, if you don't have a newer one, install the original cd stuff first so you'll be able to update to the newer Creative Recorder, Wave editor, Mini-Disc, etc. That stuff needs an older version installed or it won't install. If you have a Li'veDri've, you must install your original cd first. Even the version of Remote Center on the new install cd's needs a prior version installed first. The driver won't recognize your Li'veDri've if you didn't install the original cd first either.

Advice for correcting boomy bass

Any suggestions for correcting boomy bass using AA3? I have tried several approaches with compression and parametric equalization and, I can change the sound, but I don't think that I am fixing the problem. The bass problem, I think, is in the recording.

Hi Eric and SteveG - Thanks for your advice. I am going to respond to both of you and, hopefully, get some more feedback.
Eric - The boomy bass problem, in this case, is a matter of the bass frequency (including an upright bass instrument, but perhaps not limited to just that instrument). The bass instrument, while one can hear it, does not sound clean and articulate in the way that typically does in an acoustic jazz recording. In addition to boom, I feel that there is a characteristic that sounds "cloudy". Any additional thoughts or suggestions?
SteveG - I am going to try your suggestions and will get back with my results. I have read that high Q is typically reserved for working with sound effects and eliminating undesirable sounds. Furthermore, for music, one typically use a low Q setting - probably one less than 10. While I realize that you are suggesting high Q for isolation of the problem, what are your thoughts on what I have mentioned and are there any sources that you recommend for further reading about Q?
After I opened the post, I found the following post:
http://www.adobe.com/designcenter/audition/articles/aud15freqedit_02.html
Any thoughts on it, as a way of reducing bass? In a nutshell, it recommends using the spectral frequency editor and the marquee tool and then using the hard limiter on a very small section of the frequency.
I quickly tried the suggestion (in order to get the idea), but it raises a couple of questions. Isn't the use of limiters something that one should try to avoid, if possible. By using a limiter on part of the frequency, what is the potential downside? Is there a possibility of creating some kind of audio-related seam that either revealing playback equipment and/or good ears can identify?
Thanks again.

How do I remove these "boomy" noises?

Hi, I have just finished recording a guitar tune and noticed that there are some "boomy" noises that I would like to get rid of or at least reduce.
I recorded using 2 x Shure SM57 mics (I know there not ideal for acoustic recording) and recorded into Audition CS6.
Below is a link to the recording, please listen for the boomy noises that I mention and please help me find a way to reduce it.
Thanks
LINK....
http://www.sendspace.com/file/6ro6yg

Late reply here, but using some EQ might help. Try this.
Apply the Parametric Equalizer. Use these settings:
HP, 140Hz, 24dB/Oct
Band 1, 230Hz, -6dB, Q: 5.7
The HP filter attenuates the low frequency energy. A lot of the boominess is there, especially just above 100Hz and below. There are also some harmonics being generated, which are especially noticeable around 230Hz. It may be overkill to reduce that frequency for the entire track, but it helps a lot for the boomy areas - so I think it's worth cutting a bit there as well.
You may want to drop the high frequencies a bit around the 2-4khz region. They are especially noticeable with the above bass cuts.

FFT Filter, Dynamic EQ, Scientific Filters...

Hello
I don't know what types of applications these are good for? I'm scared, help me overcome my fear of the unknown.
I always go back to the graphic eq, parametric eq, and notch filter because it seems to get the job done for me, but I would like to utilize the FFT Filter (what does that stand for by the way?) as well as the others listed, I just have no idea when it would be a good time to use them or just experiment with them.
So, I would like to hear from Audition users who utilize these processes effectively in their work: what do you usually use them to accomplish?...and in what type of situations do you find yourself using them in most often?
Thanks,
Kris

Huge topic - can't really cover all that in one post at all.
FFT stands for Fast Fourier Transform - and from there on the explanation gets technical and complicated, and whilst it won't help you with real-world processing, you do need to understand that long window sizes work better on LF than short ones do. And just to make it confusing, a long window size means a smaller number, and a short one is a large number.
In Audition, some of the filters are finite impulse response types (Graphic equalizer and FFT ) and others are infinite impulse types (Parametric equalizer, Scientific filters and the notch filter) - these are the two basic models for constructing digital filters. If you want to know more about these, then unfortunately it gets technical - again! Basically, the infinte type uses a recursive feedback model (and can have steeper filter sides as a result), and the finite one doesn't. Most people think that they sound 'different' as well, but that depends rather on program content - you have to use your ears. The big problem with things like the Graphic EQ is that if you aren't careful there will be a lot of phase shift at the transition points between one band and the next one, and this can have a pretty adverse effect on the sound in general. I never use it for anything, because there's never anything that I want to accomplish that I can't do with Parametric EQ, but YMMV. The notch filter is primarily only intended for removing hums, and this needs careful use too, because it's very easy to introduce ringing in the filter skirts.

Can this audio be cleaned up?

I have a couple of minutes of footage (that I have to use and cannot reshoot or dub) that was recorded in a tiny bathroom. There is some echo, but that is the least of my problems. The big problem is the bathroom fan was running during all of the takes and the sound of that hum is brutal.
Here is a 24 second sample: http://www.offstagefilms.com/files/soundtest.aif
I tried sending the sound file to SoundTrack Pro to remove the noise, but the results weren't great.
I have spent the last 3 hours playing with the Audio Filters in FCP. I am stumbling around in the dark and occasionally a filter does seem to help. I have had some luck with the following (even though I don't know exactly what they do):
Hum Remover
Parametric Equalizer
Band Pass Filter
But I can't seem to find the best combination. My ears are numb at this point.
I realize there is no perfect solution (bad audio is bad audio). But I was hoping someone out there who is better at this than me (most of you), could take a moment to listen to the file, try a couple of things, and offer me some specific suggestions.
If someone can help me out with this, I would really, really appreciate it.
Thanks,
Chris

Hi Chris,
Check out _This Video Tutorial_ that shows you how to send your audio file from FCP to Soundtrack Pro where you can set a noise print and then reduce noise. You mentioned that you've already attempted this technique but you may want to try playing with the threshold and bass/tremble preservation in Tone Control to get better results.
hth

Connecting External EQ on Firepod and Mixing Down in LE 7

ok, i guess this is a multi-part question, prob. needs two posts but hey. I have a Presonus Firepod, Logic Express 7.1 and an External Dual Channel 30 Band Equalizer. i was hoping i could somhow ouput from logic through Outputs 1-2 into my hardware equalizer then back in. I thought i had to connect through the Insert/Return's on the Firepod, but im not getting any audio in Logic. I prob. dont have the outputs/Bus's configred correctly in Logic. Which by the way confuse the heck out of me. Can someone also explain roughly how to utilize the Out's and/or Bus's. im sure i wasnt very clear on this, and im sure a slew of questions will come. but i dont know how to explain it any better. And quite honestly im sure the output/bus question i posed will tie back to my equalizer question. thanks to anyone who can help me out.

-First seperate all files: sessions, audio files, non-DIGI plug-ins, etc. from all your Digidesign folders. (it will save you time and sanity to do it now and foever, I just learned the hard way)
-Uninstall use the CLEAN UNINSTALL, go from 7.0 and all the way down to your base disc
You must use the start up/upgrade discs or your updates to uninstall fully.
Make sure ALL DIGI Database folders and anything belonging to DIGI Digi preferences, core audio manager, etc. is gone.
Just uninstalling will not work, if there are remanants it will think Pro Tools is still there and folders will be shared upon re-installing and it will still not work at launch, let me save you the time and the headache.
-Then re-install from the Base Disc on up.
Then you should be good!
Still didn't figure why it was doing this or what "Quit unexpectedly after it was reopened" meant but the solution works for me.
peace
Power Mac G4 Mac OS X (10.4) Processor: 933 MHz PowerPC G4, 2MB L3 cache

Most appropriate audio filter?

Hi once again all,
You've been superb with advice thus far, so I thought I'd give my first port of call another try..!
I've just uploaded a sample clip to YouTube, purely to give you an idea of the raw audio which my external condenser microphone is picking up on my camcorder.
The high-pitched noise/squeal/not sure how to describe it is often worse than this, and I wondered if you had any suggestions on what filter to use in FCE4 to recduce it as far as humanly possible?! I know it's inevitable to an extent, but the clearer i can get, the better.
I'm using FCE 4.0.1 and importing from a Sanyo HD1A camcorder then converting all clips using Mpeg Streamclip to Quicktime with Apple DV-PAL compression. The audio appears as stereo pairs.
Any suggestions would be useful. Thus far have tried with a 3-band equaliser to reduce the high frequency to around 8000 with -15 gain, but that doesn't reduce it sufficiently.
Cheers, Sahir

Hi(Bonjour)!
I just compied your movie from youtube and I've applied an parametric equalizer filter with the 6900-(tried 8000 too) hz set to -20 db with Q set to 3 and the tinny wistle vanish.
The 3 band equalizer works too. Try a -20 dB for an hight band set to 8000 hz.
Michel Boissonneault

Help Overcoming Sync Sound Problems

I have some video footage that a client has handed over where the sync sound shot on location is, shall we say, less than optimal. It is interview footage of Doctors in Emergency Rooms and was shot on location in empty ER treatment rooms. It is very echo-y/hollow and has a lot of background voices from out in the hall of the ER.
Though I would like to minimize the background voices, I can live with them. The real issue I need to some advice on is how to minimize the hollow/echo sound from shooting in an empty sheetrock/linoleum room? The levels are excellent and the voices are crystal clear.
I am true novice at Soundtrack Pro (and sound in general) and would really appreciate some advice on how to make the voice track "richer".
Quad G5, Dual G5, Dual G4 Mac OS X (10.4.4)

I can't walk you though it step by step (*maybe somebody else can). But here are some things to look at:
Echoes in a particular room, especially without significan movement of the mic, fall at ver specific frequencies. They can be greatly reduced with careful use of extremely narrow parametric equalizations.
The following is non-FCP specific but you'll figure it out. The short story: you want to set up a parametric EQ with a narrow band and with the gain pumped up.
Now sweep the frequency up from the bottom until the sound gets really boomy and annoying. You just found one of those frequencies. Center the frequency where it sounds the lousiest.
Now turn the gain all the way down. The sound should be much better. Tweak the bandwidth and the gain until the most objectionable echo is taken out without affecting the desired audio too badly. This is subjective.
You're not done yet. Repeat this process with another parametric EQ, sweeping up friom the frequency of the first one. Repeat all the above. And so on until you've got a handle on it. Most of the nasty echos will be in the boomy bass area.
If you're really finicky, you may now find yourself going back and raising the gain on some of these to get back some of the audio quality. Echo isn't all bad, but there is definitely such a thing as too much.
I guess I do go step by step... darn, I don't have time for that!

Reducing low-level noise from phone conversation

I'm using GarageBand to digitize a phone interview I did. There's a quiet sort of static noise on the track that I'd like to reduce while making the dialogue stand out more. Any tips?

There really isn't any kind of "one button" solution within GB. You could try doing some web searches on "parametric EQ" or "parametric equalization" to learn how to use the parametric EQ that GB provides.
An alternative solution would be to look at an audio editor that offers noise reduction to process the audio file either Pre or Post GB.
http://thehangtime.com/gb/gbfaq2.html#audioeditors
(believe both Audacity and Amadeus offer noise reduction, and Schneb says there is an Amadeus Pro now)

Removing audio whistle in background

Hi, I want to remove a whitle/hum in the background of an interview and need some help. I've tried using the "Hum Remover" Filter in FCP and also the "Noise Print" Reduction in Soundtrack Pro (although its the first time I've used the software so I may not be great at it).
Could anyone recommend how to get rid of the unwanted sound?
You can download a 1mb example aiff here :
https://rcpt.yousendit.com/736369137/16eb13161bacb3ccafe741fb544e3d67
(The first three seconds are unfiltered, then you can hear my rubbish attempt with the "Noise Print" Filter ... the interviewee is talking about software btw).
Any help much appreciated!
Oliver

Maybe we're talking about two different things. The example you posted does not have a hum.
A hum is a low-frequency sound, like you get from bad grounding. Double-check the example
you posted to make sure the "hum" is in that file, and when you stop playing the file,
the hum stops.
I think it's time for a quick filter tutorial: Insert FCP's Parametric Equalizer. Now turn up the Q
and the Gain really high so that you can hear the effect. Now move the Frequency back and forth
so that you can hear the filter sweep. Now keep sweeping back and forth until you find where
the filter makes the whistle worse. Try to find the exact frequency. +Now change the Gain from+
+postive to negative.+ That should pull out the problem frequency.

Unwanted audio compression

Can anyone help?!!
I'm trying to output a movie to tape but it keeps compressing the audio. Louder passages keep being knocked down. I don't understand why as I have not added an audio compression filter. It's not compressed when I play it in the timeline. Any ideas anyone??
Mac Pro Mac OS X (10.4.9)

and a teensy bit more:
Your problem could have just as much to do with the "soundsticks". I just looked them up. They are small speakers intended to be used as computer speakers. They in no way represent or emulate a studio monitor that one might use to make audio decisions. Even if they sound great... in fact especially if they sound great. If they make "improvements" on your sound, then those are improvements your listeners aren't going to hear.
This should pretty much make the case: "SoundSticks II employs a new computer-optimized multi-band parametric equalization."
So the problems you're having could be originating with the prejudices of those epeakers affecting your audio processing decisions, which in turn affect what comes out the other end.
To have it come out sounding like "compression" is a little odd, but it may just have to do with your relative perception of the sound coming from these different speakers.
The best way to judge audio is in a clean soft neutral environment, with good nearfield monitors.In addition (and not simultaneously) you should be able to hear some representation of how your mix is going to sound on a little 3" speaker in an average TV set. One way to do that is to play it through your TV set.

Animation - Lip Sync sound problems

Hey guys. I used to have Macromedia Flash 8 but I've now got
Adobe Flash CS3. I am trying to make a speaking animated person in
flash and for this, I wish to change the framerate to 24FPS for
high quality, and then import the sound to timeline and for each
individual frame, change the facial and mouth expressions. For
this, I would like to, when clicking on a frame of the sound to
animate, just the sound for that one frame to play so I know what
expression to add.
I used to be able to do this in the old version fo Flash and
I hope that you can help me with this.
Thankyou,
Ryan

I can't walk you though it step by step (*maybe somebody else can). But here are some things to look at:
Echoes in a particular room, especially without significan movement of the mic, fall at ver specific frequencies. They can be greatly reduced with careful use of extremely narrow parametric equalizations.
The following is non-FCP specific but you'll figure it out. The short story: you want to set up a parametric EQ with a narrow band and with the gain pumped up.
Now sweep the frequency up from the bottom until the sound gets really boomy and annoying. You just found one of those frequencies. Center the frequency where it sounds the lousiest.
Now turn the gain all the way down. The sound should be much better. Tweak the bandwidth and the gain until the most objectionable echo is taken out without affecting the desired audio too badly. This is subjective.
You're not done yet. Repeat this process with another parametric EQ, sweeping up friom the frequency of the first one. Repeat all the above. And so on until you've got a handle on it. Most of the nasty echos will be in the boomy bass area.
If you're really finicky, you may now find yourself going back and raising the gain on some of these to get back some of the audio quality. Echo isn't all bad, but there is definitely such a thing as too much.
I guess I do go step by step... darn, I don't have time for that!

Parametric Equalization questions

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