Hey Sticky, I think you'll find that the 80 hertz filter on the microphone is at the capsule prior to the impedance converter/electronics? The impedance converter/electronics will not get ravaged by the low frequencies from the proximity effect.

Doing it in the software meant that the microphone had to contend with those extended low frequencies in its electronics prior to the microphone preamplifier.

One of the same reasons why I like to use compression/limiting while I am tracking as opposed to after. People are so afraid of doing something that cannot be undone which only comes from inexperience. One must take risks in order to grow. If people are worried about being safe call the time while recording, why do people drive cars? They're so dangerous and you cannot undo death. Best not to drive anywhere. NOT!

Still alive after all these years
Ms. Remy Ann David

the roll off is on the mic pre

Remy got on AIM!

I usually roll off 80 on the recording also, I usually end up taking off up to 125 hz on the acoustic. Then once you do that the acoustic usualy brightens right up without having to boost a lot of high end. I like this term

Cut to make things sound better
Boost to make them sound different.

Jason

RemyRAD said

use compression/limiting while I am tracking

Would you be willing to share some knowledge on how you determine what sounds need how much compression?

Digital EQ takes almost twice the amount of cut for the human ear to hear it, due to the fact that there is no phase shift.

I would like to address QuickDiscs, you are incorrect with your knowledge regarding phase shift in digital equalization. There are 2 popular kinds of computer equalization, some are like what you believe them to be and others, most, are still like their analog counterparts. One is known as FIR equalization and the other is known as IIR equalization. One affects corresponding harmonics and the other doesn't. They both have their applications and one cannot assume what you have assumed.

Now on to your question from cfaalm. Many of us who have been recording for quite some time generally know what we want to begin with. For example, I most always add some light compression, from 2 to 10 DB's at between 2:1 to 6:1 and/or limiting, just a couple of DB's at 8:1 to 20:1, to most contemporary popular vocalists. It really makes a difference in the sound when you do that going in as opposed to doing it when it's coming out. One can also, frequently do both, i.e. record the vocal on one track with processing and on another track clean, without processing if one is hesitant to commit during tracking? I don't usually do any of this with operatic talent, unless I'm trying to obtain a certain sound or style in the recording.

I will frequently add a few DB's of limiting to electric basses, when taken direct, whether I'm using an active direct box or an output from the bass player's preamplifier, depending on the musicians technique used. I will also frequently use some type of limiting and gating on bass drum and snare drums. I like how it tightens things up. I also frequently invert phase on the bass drum for the change in texture it affords the drum mix (it creates a more solid thud then a flabby one). I also like adding a microphone to the bottom of the snare drum, also inverting the phase of that while mixing it in with the top microphone on the snare drum, usually on a pair of tracks, separately. It definitely adds more snap and snare sound and I frequently use a SDC like an SM 81 or AKG 451or just another SM57.

If an electronic keyboard player utilizes too wide a dynamic range (which I have found frequently) I'll add some stereo limiting to the keyboards.

Again, I cannot stress enough that when tracking with dynamics processing, LESS IS MORE. Depending on the situation and circumstances, I have also been known to track clean without adding anything until mixdown. I follow no particular formula except for emotional impact, convenience and spontaneity, unless a producer specifies otherwise. I usually also do not mix down with a stereo limiter across my stereo mix bus. I'd rather do that after I mix but for numerous live broadcasts of musical events, I have sometimes been asked to do just that (for numerous different reasons).

Now don't misinterpret what I'm saying here as there are many different brands and type of limiters/compressors. Some have peak detectors, others have averaging or RMS detectors, some rely on optical ballistics. I have and use all of the above types by numerous different manufacturers. If I don't like what I'm hearing from one unit, I'll try the others until it tickles my years, ears and rear (thank God that last one is singular)! TMI!

Nobody has ever really thrown me any curves but when they do, I roll with them, adapting along the way and never getting upset over what I may believe to be a peculiar request. It's always a learning experience and a fun challenge while strutting your stuff. The more obstacles thrown at me, the more exciting I find the job (of course there is always substantial stress involved but that's part of the job). Some people are very set in their ways and they don't do well with those kinds of situations. I excel at them.

Everything old is new again
Ms. Remy Ann David

Wow, Remy, thank you for your elaborate answer. I enjoyed reading that. So I gather that you use (analog?) compression and limiting in a conservative way. That makes sense.

I have read about reverse phasing the bottom snare mic and kick drum mic before. Somebody talked me out of reverse phasing the kick drum mic last session and now I think it sounds like crap. Fancy that. :D

We still have another session ahead of us so I'll be cooking a little with your recipe. I hope we'll perform well enough to post some material.

That's pretty cool and always remember, you can still invert phase of the bass drum in mix down. Now, I don't always invert phase on the bass drum. I usually only do it when I want a harder hitting, tighter bass drum sound instead of that fatter, flabbier in phase bass drum sound which I still like depending on the music and the project involved.

Looking forward to hearing your completed results.
Ms. Remy Ann David

RemyRAD, post: 192608, member: 49130 wrote: One of the same reasons why I like to use compression/limiting while I am tracking as opposed to after. People are so afraid of doing something that cannot be undone which only comes from inexperience.

I agree.

RemyRAD, post: 192608, member: 49130 wrote: Hey Sticky, I think you'll find that the 80 hertz filter on the microphone is at the capsule prior to the impedance converter/electronics? The impedance converter/electronics will not get ravaged by the low frequencies from the proximity effect.

Doing it in the software meant that the microphone had to contend with those extended low frequencies in its electronics prior to the microphone preamplifier.

This is very a interesting comparison between compressors/limiters and microphone filters. I much prefer tracking vocals with an LA2A to a plug-in comp on the vocal channel, ITB. "Vocals" always sound fuller and silkier through quality analog tracking. But I never thought about the HPF as a deal breaker in the same way. I can think of quite a few mics without hpf. Many engineers prefer mics without the filter.
Boswell others....
What do think about hpf on the microphone? Do you use them and prefer a mic with that option?
Is this subjective to the quality of the mic/ preamp and/or headroom ?
Do you find a capture sounds better when the mic hpf is employed during tracking or ITB during mixing?

audiokid, post: 445532, member: 1 wrote: What do think about hpf on the microphone? Do you use them and prefer a mic with that option?
Is this subjective to the quality of the mic/ preamp and/or headroom ?
Do you find a capture sounds better when the mic hpf is employed during tracking or ITB during mixing?

I haven't thought about it that far.. but I have a customer who does voice over and his voice has a lot of low end. I found that it messes the compressor behavior on my LA-610 so, with him I always use a mic with HPF engage.
So not only the mic electronics could be overloaded with so much bass but it is certainly screwing up the compression, unless I use sidechain but since I'm gonna remove those frequencies anyway.. Better do it at the mic..
I have to keep that in mind with other sources !

I have always rolled off the bottom end around 80-100....but I have always done it post-recording. I actually never even thought about doing it pre-recording.

I know this thread is older, (and the OP and I are from the same city, small world) but if what he is saying is correct, what would be causing that to happen?

The only thing I can think of is the phase offset that will come with a 6, 12, or 18 db slope (anything but 24 or 48) that will be inserted directly into the recording if you have it in the chain during recording. Or is it possibly just a harmonic anomaly that may disappear if that crossover point was changed slightly? In other words, it's causing harmonic anomalies that are now in the low to mid 100's hz areas, which will bloat the bottom end (if anomalies are the cause).

And this part....

pcrecord, post: 445554, member: 46460 wrote: unless I use sidechain

I don't quite understand how that would help? Isn't the sidechaining just causing the tracks it's applied to, to duck the vocal (or whichever slave/master you choose)? But if the vocal still has some bottom bloat...how does SC take care of that?

I'm very curious bc I work with a girl who has, one of the most beautiful voices I've ever heard. And enormous range. But her tone is deep for a woman. Think Toni Braxton, but lower. And I have a hard time controlling the bottom end sometimes. The presence just becomes too large...

Commonly, compressors will react more to bass content compared to higher frequencies.
To avoid this, you can send the track to an eq, remove some lows and take the EQ output as the sidechain input signal of your compressor.
What it will do is avoid some pumping when the signal peaks in the lows.

Of course this is a hard work to do OTB. Nowaday, many compressor plugins have that fonction imbedded.
Fabfilter C2 has this. It processes the signal and when sidechain is activated you can either use an external signal or
internal signal. With internal, it takes the input signal and apply an ajustable EQ to use it as sidechain.

Check this thread for a different viewpoint on use of HP filters at the microphone input.

Boswell, post: 445969, member: 29034 wrote: Check this thread for a different viewpoint on use of HP filters at the microphone input.

Yeah it's a very good point, if the extra low end is reduced within the mic, it's own electronic won't be saturated by it, the same with the preamp and converters. In short it can only sound better, right ? ;)

pcrecord, post: 445967, member: 46460 wrote: To avoid this, you can send the track to an eq, remove some lows and take the EQ output as the sidechain input signal of your compressor.
What it will do is avoid some pumping when the signal peaks in the lows.

Of course this is a hard work to do OTB. Nowaday, many compressor plugins have that fonction imbedded.
Fabfilter C2 has this. It processes the signal and when sidechain is activated you can either use an external signal or
internal signal. With internal, it takes the input signal and apply an ajustable EQ to use it as sidechain.

You da man! I never thought of this.

Brother Junk, post: 445979, member: 49944 wrote: You da man! I never thought of this.

You know you can even do a de-esser with this technique. Little tricks go a long way ! ;)

pcrecord, post: 445980, member: 46460 wrote: You know you can even do a deesser with this technique. Little tricks go a long way ! ;)

I can't wait to try it.

Anyone know whatever happened to "Remy Rad?" Just didn't play well with others? She seems to know some stuff...

Some related history > http://recording.org/threads/hpf-on-microphone-preamp-or-when-mixing.59231/

Tracking filters vs ITB filters. How important is this?
For 3 years Sequoia 11, 12, 13 has had a problem with the stock channel EQ (plugin).

When the HPF is engaged, depending on where you set the hpf roll-off curve (80 hz and below), silent to very obvious hiss begin to increase as you move the curve lower down in sub freq roll-off.

You don't notice it happening unless the track has those freq present during the timeline. Meaning, the filter sounds fine until it see's those subs in the track.
How important is this?
Q: Does white noise in a mix cancel or effect the phase of transients? Will white dynamic noise effect good reverberation and all that comes with ambience, sweetness etc?

This code error, while the hpf is reducing sub freqs, it is also dynamically increasing upper white noise freq throughout the spectrum in a song.

which in turn effects the threshold sensitivity on the 2-bus compression (side-chain de-essing and that bad swirly phase sound are a few areas that come to mind).

Whats more interesting> no one noticed this until I pointed it out on the support Samplitude forums. My point is not to blow my horn here... but rather wonder, holy crap batman... .... how could we miss such an important code error, which hasn't been fixed!
Thank goodness, they added an optional EQ to replace the older stock one.

If they hadn't, this would have forced me to switch to another DAW. But then... I'm betting Sequoia is the lesser of evils.
I am convinced, DAW engines are not all the same.

Plugins do add anomalies and weird artifacts to a mix.
Once I discovered this, I started researching, listening to other peoples mixes and I have indeed noticed hiss problems from (just a guess) converter AD op-amps, reverb plugins, and HPFs reacting from certain freqs.

What I noticed about myself:

After I upgraded to good (conversion, speakers, headphones > improved / subjective hearing techniques) I've learned to recognize how easy it was to evolve in a mix and miss subtle, yet very important accumulative aliasing artifacts/distortions the creep into a song/mix, that end up becoming part of the fabric of a mix. I now take extra time to listen to the full bandwidth of a track the moment I engage or disengage a process.

It also makes me ask the question, how many people are actually rolling off the subs?

Finally, I am a firm believer to get the best sound you can during tracking. Digital audio is awesome but its not magic. Some plugins do more harm than good.

audiokid, post: 445984, member: 1 wrote: I am convinced, DAW engines are not all the same.

+1

audiokid, post: 445984, member: 1 wrote:
Some related history > http://recording.org/threads/hpf-on-microphone-preamp-or-when-mixing.59231/

Tracking filters vs ITB filters. How important is this?
For 3 years Sequoia 11, 12, 13 has had a problem with the stock channel EQ (plugin).

When the HPF is engaged, depending on where you set the hpf roll-off curve (80 hz and below), silent to very obvious hiss begin to increase as you move the curve lower down in sub freq roll-off.

You don't notice it happening unless the track has those freq present during the timeline. Meaning, the filter sounds fine until it see's those subs in the track.
How important is this?
Q: Does white noise in a mix cancel or effect the phase of transients? Will white dynamic noise effect good reverberation and all that comes with ambience, sweetness etc?

This code error, while the hpf is reducing sub freqs, it is also dynamically increasing upper white noise freq throughout the spectrum in a song.

which in turn effects the threshold sensitivity on the 2-bus compression (side-chain de-essing and that bad swirly phase sound are a few areas that come to mind).

Whats more interesting> no one noticed this until I pointed it out on the support Samplitude forums. My point is not to blow my horn here... but rather wonder, holy crap batman... .... how could we miss such an important code error, which hasn't been fixed!
Thank goodness, they added an optional EQ to replace the older stock one.

If they hadn't, this would have forced me to switch to another DAW. But then... I'm betting Sequoia is the lesser of evils.
I am convinced, DAW engines are not all the same.

Plugins do add anomalies and weird artifacts to a mix.
Once I discovered this, I started researching, listening to other peoples mixes and I have indeed noticed hiss problems from (just a guess) converter AD op-amps, reverb plugins, and HPFs reacting from certain freqs.

What I noticed about myself:

After I upgraded to good (conversion, speakers, headphones > improved / subjective hearing techniques) I've learned to recognize how easy it was to evolve in a mix and miss subtle, yet very important accumulative aliasing artifacts/distortions the creep into a song/mix, that end up becoming part of the fabric of a mix. I now take extra time to listen to the full bandwidth of a track the moment I engage or disengage a process.

It also makes me ask the question, how many people are actually rolling off the subs?

Finally, I am a firm believer to get the best sound you can during tracking. Digital audio is awesome but its not magic. Some plugins do more harm than good.

This is too much for me to bear on my mind. Can you explain a little bit more, please? After a re-read i'm still pretty foggy.

Kaan, post: 445987, member: 50244 wrote: This is too much for me to bear on my mind. Can you explain a little bit more, please? After a re-read i'm still pretty foggy.

No problem, I'm not the best at wording my audio experiences or process through text.

I'll put it simple : Even though the hpf "plugin is reducing sub freqs, it is also dynamically increasing upper white noise freq throughout the spectrum in a song, via bad software code.

audiokid, post: 445984, member: 1 wrote: I am convinced, DAW engines are not all the same.

I have said the same in the past and gotten hammered for it. But I tend to agree. Also the plugins are not as good imo in some daws. I have, and use, Ableton, FL, Cubase, Logic X and PT 10/11. For "extra" plugins I have a lot of Waves stuff, my beloved Izotope pi's, Komplete 10, Vienna, and I'm probably forgetting quite a few.

I have only done it twice, (made a little song with imported loops), with PT vs Logic. I chose those two bc that's where I notice the difference. And I genuinely wish it wasn't with Logic. I would rather throw Ableton or Cubase under the bus....bc I find Logic to be very convenient. And Logic 10 is pretty good. But after a while, when a mix was done, I would notice it was a little lifeless, compared to when it started. And that didn't happen to me in PT. And even with the same presets in both, same material etc...something funny just happens in Logic.

PT seems to have noticeably more noise-free gain, with anything. Imported loops, or recorded material. Same loops, same mic, same processor etc. And once you engage plugins, they work as expected. But with Logic, things start to get strange. I don't know why, and I don't want to argue with anyone about it. But LX has a linear phase EQ, as does PT. But once you engage eq in LX, and add something else to the pi chain, stuff starts to get a little weird. I can even use the same 3rd party pi's, with the same saved presets, and it sounds different in LX vs PT. It's hard to even put your finger on it, but something is different. And the end effect is a noticeable volume/dynamics difference, and also a sort of unwanted "grainy-ness" in the sound? I don't have the tracks, it was two laptops ago. But it was almost like the LX track was somehow degraded with the addition of a few pi's. But the PT track, sounded loud and clear.

I don't notice it in Ableton, Cubase or FL. I always get clean sound, no matter how much I fuss with it. But the Logic lovers in the room came down hard on me when I mentioned this. Some people said I'm just not used to Mac's (all I personally use is Mac) or there is a setting somewhere...a box that's not ticked (if any of you know what that is, please tell me. I have yet to see the box that asks, "Do you want your mix to sound grey?") And I am by no means an expert in any of this, so perhaps it is user error, but I don't understand what box I could have unticked that would cause this. This was back in version 10.1.1 that I noticed it. I'm now on 10.2.2, and it appears to be the same. I have tried 10.2.4 and it seemed the same so I went back to 10.2.2. I have been through all the menu's, looked on line etc. So I just use Logic for composition now and export to PT. Logic also does this weird thing where if you change the session SRC up and then back down (this forum helped me figure this one out) there is a fluttering sound in the tracks, like a tremolo effect. I'm pretty sure I still have some bounces of that effect if people want to hear it. I can change PT up, down, up, down etc, and while I'm sure it's not good for the project, it doesn't induce any noticeable sound change. I had to re-write at least 3 projects, some of them twice, bc this damn fluttering sound just appeared. That artist, was not happy lol. And bc I didn't understanding the cause, I re-wrote the first couple in Logic, and the same thing happened all over again. It was extremely frustrating to say the least.

So, I agree with you audiokid , I don't have the knowledge to persuade people that not all DAW engines are equal, but imo, there is obviously something different about how at least two daws interact with pi's. And if two are different, they are probably all different, even if only slightly.

So following on your posting Chris , Im thinking .... ok so why don't I use LPF and some EQ on instruments on the way in .
I think I never use LPF on anything ( maybe some vocals ) , but it makes a lot of sense when considering white noise in the
same way as low end rumble .
I will try it on the next song . Im thinking about how much to trim instruments . Obviously I don't want to kill the vibe and
overtones etc . So how much filtering would be a good starting point ?

Smashh, post: 446281, member: 45856 wrote: So following on your posting Chris , Im thinking .... ok so why don't I use LPF and some EQ on instruments on the way in .
I think I never use LPF on anything ( maybe some vocals ) , but it makes a lot of sense when considering white noise in the
same way as low end rumble .
I will try it on the next song . Im thinking about how much to trim instruments . Obviously I don't want to kill the vibe and
overtones etc . So how much filtering would be a good starting point ?

It would all depend on the track. For female vocals, I cut it around 150 and 16-18khz. For males, it depends on the voice, but usually 100 is cut at 48db per octave, and I mess around with the LPF again between 16-18khz. But it's useful on any track. Do you have an eq that does real time frequency analyzation? (Or any plugin that does it)? It's helpful for seeing where you chop off the spectrum ends, without affecting (mostly) the track, bc you can see where the meat is, and where you can easily trim.

Forgive me if you already know this....but when you create an HPF and LPF (essentially it's a bandpass now) the points you use matter. Don't just cut it at 120...bc someone said to. Often times, you slide it down to 110, and it sounds much better. Or 125 hz, will all of a sudden create a better sound than 120, because of the overtones. Sometimes you can quite easily hear them disappear, and come back, just by moving the cutoff point slightly (relative to it's scale I mean). So you can kill what you don't want, and keep what you want.

Also the crossover slopes will impact the sound greatly. A 6db crossover slope puts the phase around that x-point 90* out of phase, 12db puts it 180* So, unless you are using a linear phase eq of some sort, it's good to know what is happening. 24db, should put you back into same relative phase. Any multiple of 24. So, sometimes, the selected cutoff is a good point, and altering the slope will give you back some overtones.

But sometimes, people make it 24db per octave just "because." But sometimes, a 12db slope will sound better. Sometimes the 18 db slope sounds better, or the 6db. Both the cutoff points, and the slope...even small adjustments often make a big difference because you will be, as you said, nulling or canceling some overtones. Often just changing the cutoff by 10hz, and all the overtones that you were missing, come back.

A frequency analyzing pi would be good to find your starting point. Or, you can just do it by ear. Slide it up or down until you hear it...and then make finer adjustments.

Most DAWs implement their stock EQ using FIR linear-phase filters. Once the half-length filter time delay is corrected, there is no phase shift in the filtered output relative to the input.

There are EQ plug-ins available that use IIR techniques to emulate analogue filters, and part of the emulation is to generate an approximation to the frequency-dependent phase characteristics that an analogue filter would have. I was disappointed to find that some of the IIR emulations I have used still make a time-delay correction, which rather defeats the point.

Smashh, post: 446281, member: 45856 wrote: Im thinking .... ok so why don't I use LPF and some EQ on instruments on the way in .

This is according to my modo of minimal mixing :
Everything we do on the way in lowers the chances of screwing things up ITB
This discussion about EQs reinforce the importance of mic choice and placement doesn't it ... ;)

Boswell, post: 446284, member: 29034 wrote: Most DAWs implement their stock EQ using FIR linear-phase filters. Once the half-length filter time delay is corrected, there is no phase shift in the filtered output relative to the input.

I mentioned that the phase shift would only happen with a non-linear phase eq.

But it sounds like you are saying that most stock DAW eq's already ARE adjusted for phase? Is that the norm?

Is there a stand out EQ out there where I can download the demo and say AAH thats so much better ? ( phase aligned ? )

I looked at a few threads about this topic ,but come away thinking , maybe I don't need to know so much about this
and I will leave it at it sounds real/good , or somethings gone skewed here .

Smashh, post: 446440, member: 45856 wrote: Is there a stand out EQ out there where I can download the demo and say AAH thats so much better ? ( phase aligned ? )

I looked at a few threads about this topic ,but come away thinking , maybe I don't need to know so much about this
and I will leave it at it sounds real/good , or somethings gone skewed here .

If you want phase aligned, PT's stock EQ is fine. It just takes practice. The EQ I use that I don't think is phase aligned is a 30 band from Waves.

About the actual topic, you are just cutting off the top or bottom (or both) end of whatever is in the track so that there is less noise.

If you don't have any noise in the first place, it's not necessary.

Just play with it....you'll get it (famous last words lol).

I would love to have no noise in the first place , but It aint gonna happen in the near future .

I get a hiss squeal around 7.6K and Im wondering if it is worth LPF just below there .
At a guess Id say not, but has anyone here cut out so much in electric strat type guitar ?
Probably a dumb question but doing that vs a high Q subtractive EQ ( 16 - 20 db ) at 7.6K

Smashh, post: 446442, member: 45856 wrote: I get a hiss squeal around 7.6K

Do you know what is causing it? Bad mic, bad cable, bad connected etc? That's fairly close to 6300 which is a sensitive area for our ears. "Noise" in that range seems abnormal to me, but maybe I'll get a lesson here.

These guys are the pro's, not me, but I would think that would be a very low cutoff point. There is still a fair amount of info left.