Topic: Pre/Post Processing?

[jimi56]

This is more of an opinon thing than technical so i went with this forum.  What are your stances on pre and post processing.  Mostly i'm talking about using an equalizer at the show and then software post-processing, ala audacity or soundforge. 

I've bounced back and forth between yes and no myself.  Before i knew anything about recording and the like, i kind of liked to find the shows with pure sources, but now i wouldn't mind a little EQ tweak here or there maybe some volume boosts after the fact. 

For arguements sake, lets disregard powering and carrying an EQ unit or any effects/processing unit.  Obvioulsy i'm not talking preamps for gain and the like.

Thoughts?

[RebelRebel]

The problem with processing is that a lot of kids dont know what they are doing, and do it on a computer with kmart speakers and in a room that isnt suitable...with no experience mastering/mixing whatsoever. if it is for your own listening it is one thing, but to tweak something and spread it not knowing how it will translate to other peoples playback or ears even is another..Volume boosts are acceptable(IMHO), but EQ is a whole different ballgame, and is really the realm of the professionals(for mass distribution) ..if you do EQ it(and youll get some that turn their noses up at you) list exactly what you did. A lot of tapers are listening to sources to get a feel for how a mic/pre combo sounds and EQ screws with that ...my 2cents...

This is more of an opinon thing than technical so i went with this forum.  What are your stances on pre and post processing.  Mostly i'm talking about using an equalizer at the show and then software post-processing, ala audacity or soundforge. 

I've bounced back and forth between yes and no myself.  Before i knew anything about recording and the like, i kind of liked to find the shows with pure sources, but now i wouldn't mind a little EQ tweak here or there maybe some volume boosts after the fact. 

For arguements sake, lets disregard powering and carrying an EQ unit or any effects/processing unit.  Obvioulsy i'm not talking preamps for gain and the like.

Thoughts?

[scoper]

I do NO post-processing on my tapes, except for clip repair if needed, and normalizing the waveform.

I agree that everyone's ears are different, and I don't want to impose my sense of what sounds good to the next person. That's what tone controls or equilizers on playback equipment is for.

I've heard my original tapes "remastered" by some "experts" withing the Springsteen fan community, and have universally hated the result. People are way too heavy-handed, especially boosting the low and high end to try to give the tapes a little "presence".

Leave it to the pros.

Scott

[Bdifr78]

IMO volume boosts are the only thing that is cool.  But like Ray said if your not sharing the recording, do what ever sounds good to you.  One time I adjusted a recording by cutting some bass and screwing slightly with the upper end, I only did that because the recording was virtually unlistenable without any adjustment.  When I listen to the tape a few weeks or months later, I really want to know how the room sounded and how the equipment I used with specific mic configs and locations sounded.  Kinda as a learning opportunity.  When you are running entry level gear I can understand the temptation to make it sound better, but in reality you probably just need new mics.

[cleantone]

The problem with processing is that a lot of kids dont know what they are doing, and do it on a computer with kmart speakers and in a room that isnt suitable...with no experience mastering/mixing whatsoever. if it is for your own listening it is one thing, but to tweak something and spread it not knowing how it will translate to other peoples playback or ears even is another..Volume boosts are acceptable(IMHO), but EQ is a whole different ballgame, and is really the realm of the professionals(for mass distribution)

I agree with this 100%

..if you do EQ it(and youll get some that turn their noses up at you) list exactly what you did. A lot of tapers are listening to sources to get a feel for how a mic/pre combo sounds and EQ screws with that ...my 2cents...

I don't personally feel the need to list the parameters on any EQ or compression I might use. I also feel that there are far too many variables from show to show and whatnot to truely be able to get a sense of mic/pre combo's with any real results that this isn't a factor for me. My goal is to make the best recording I can make and if some EQ or compression is helpful with that I'll use it. I don't have a real control room but I try not to be extreme as a real matering engineer in a real mastering suite coudl or would be. I also have been doing it for about a decade and probably only got somewhat good in that last 6 or so years. If you do not have a decent playback system you best off sticking with volume changes if any.

[TNJazz]

Agree completely with Ray and cleantone.  If you don't have a good room and a good system, you have no idea how it will translate.

I don't do anything to my 2 track recordings.  I want them to represent what it sounded like in the room.

I try to be as minimalist as possible with my multitrack recordings as well.  A little verb, some light EQ possibly, and maybe a little compression (everything I do is a separate split, so of course it's pre-everything).  That's about all though, because I'm not a professional mixing engineer.  I track it and let other folks deal with that end of things.

So how many of you secretly chuckle and roll your eyes when you see the word "remastered" listed in a DIME torrent?  I know I do...

[Brian Skalinder]

A few thoughts from a PM discussion (slightly edited for public consumption) I had with momule and mmmatt, and a comment or two from and in response to the To Normalize or Not thread:

Of course, the decision about whether or not to master depends in large part on the master recording itself.  I generally find myself recording in smallish venues, from close-to-ideal location, with pretty good sound systems and sound engineers.  As a result, I find fewer issues in most of my recordings (to my ears) than those from larger or less ideal venues / locations.

This very minimal comparative listening [we listened to a couple original and mastered samples] illustrates, to me, anyway, one of the primary reasons I don't do much mastering beyond a bit of gain/compression (and then only if necessary, for example if the HUGE dynamic range in a recording - rarely the case when recording PAs - makes playback at reasonable volumes difficult without constantly adjusting the playback volume):  it's very, very difficult to master in such a way that everyone finds it equally pleasing.

Ultimately, for me it comes down to three issues:

<1> Numbers Game
<2> Effectiveness
<3> Personal Preference and Priorities

<1> Numbers Game
In some respect, it comes down to a numbers game:  should I master in the hopes of pleasing X% of listeners at the potential expense of disappointing Y% of listeners?  Ideally, X = a high number and Y = a low number, and *I* fall into the X (pleased) category.  From a previous post:

My target audiences for my recordings, in order of priority:

<1>  me
<2>  fellow tapers and friends
<3>  the masses

So I master accordingly:  for me.  It just so happens that many of my <2> friends and fellow tapers are also interested in hearing recordings in their full dynamic glory.  I'm not especially concerned about <3> those who wish to listen to recordings on the crappiest car sound system on the planet and therefore would prefer if I compress the snot out of my recordings.  So since I have no real interest in "mastering" my recordings for the masses, I don't.

As it is these days, I generally seed 2 sources:  24-bit and 16-bit.  The additional time, effort, HD space, and bandwidth are not worth it to me to ensure I satisfy the masses.

Were I to produce a recording on behalf of a band as an official release, then my target audience would have changed.  Hence, my decision of whether and how to master may change accordingly.

<2> Effectiveness

As others mentioned in bits and pieces in the "to normalize or not" thread, and I clarified and summarized in one of my posts, it's incredibly difficult to master unless one has ALL of the following:

• knowledge
• experience
• proper tools
• great ears
• proper monitoring gear

Very few tapers / hobbyists have the knowledge and experience.  Granted, one may learn the knowledge and gain the experience, but that takes time and for many tapers, it's just a hobby and as such other life activities take priority.  Those with the knowledge and experience are probably in a position, and motivated, to acquire the proper tools.  Not much to say about great ears - some have it, most don't.  But even if all these factors vall fall into place, I know almost no tapers who have proper monitoring gear.

<3> Personal Preference and Priorities

Anyway...to summarize my perspective (again drawn from the thread spurring the discussion), in my situation, I don't - and likely won't start to - master much, if at all, because:

• I generally like the way they sound
• I don't find the benefits worth the additional time and effort (when balanced against other life priorities)
• I don't have a proper monitoring environment
• I make the recordings for my ears, and my playback, and am not set up with proper monitors to ensure it sounds "just so" on everyone's playback system
• the venues in which I record generally sound good - have good PAs, good people behind the board, not horrible (and granted not great, either) acoustics, etc., and I make the effort to secure a good location for my ambient mics
• the minor imperfections - and I don't mean something so blaringly awful that the recording's downright unlistenable - all contribute in some way to the character of the recording;  the flaws are part of the experience for me, just as I find a live band - warts and all - appealing.

[jimi56]

Just to clarify... I personally don't ever plan on running pre-EQ or pre anything.  I'd like to think i have a good ear, but still, i want to capture  as close to whats happening then, so pre processing live recordings is just a no IMO.  Studio works are different, production can be as much a part of the album as the music.

Post production stuff... I suppose if i HAVE to i'll do a normalization/volume boost.  But sometimes that will just boost ambiance noise, so i just hope that i get the levels right so that its not needed.  Plus it seems to me low end mics/cables can introduce hiss that might not come out till an overall boost is applied, so for me its all about getting it right the first time.  I guess thats why i have a hard time just setting it an walking away, always that itching feeling that i need to double check those meters. 

I'd like to do nothing to recording except track it and shave off uneeded audiance noise pre-set and pre encore.  I tend to avoid recordings that go through any kind of production when it comes to shows.  For me, its trying to capture a moment.  I'm new to running a rig at live shows but i've dabbled enough in some home studio stuff that i didn't totally freak when i had to plug it in and hit record.  I was just thinking about this whole issue this morning and thought i'd bounce it around here.  With the BT and archive.org to distribute recordings its easy for people to get thing out to the masses, and often we get a variety of sources from a single show.  I'm seeing more production going into some distributions and was just curious about the majority opinion on it, esspecially the opinion of those more experienced than myself. 

[cleantone]

Post production stuff... I suppose if i HAVE to i'll do a normalization/volume boost.  But sometimes that will just boost ambiance noise, so i just hope that i get the levels right so that its not needed.  Plus it seems to me low end mics/cables can introduce hiss that might not come out till an overall boost is applied, so for me its all about getting it right the first time.  I guess thats why i have a hard time just setting it an walking away, always that itching feeling that i need to double check those meters.

Your definitly correct about not applying EQ or anything else as your recordings. You can't undo it and your obviously not in a control room where you can monitor properly when recording a live concert. You points about adding ambiance and or noise. I think I'm right when I say that if you have a recording with 6db of headroom, then boost the track 6 db in post. This would change your noise floor. I have to assume it would be the same change as turning the volume up 6db on playback though too. At least I think it would be the same.

[jeromejello]

i will generally normalize to -0.20dB in post... this takes care of any clips that might have popped in (i am still tweaking my rig)

on the show i just recorded sunday night... the crowes opened the second set with an acoustic number that i bumped +8dB in post with a fade into the original level during the applause/tuning before the next tune... that came out ok... oh, and i did bump the left channel in the first set +0.5odB as well (again, still getting the hang of the mics)

that was the most post i've done for a show... it was all pretty non-intrusive, but it was noted in the text file if anyone didnt want that.

as far as EQ'ing... that is something that i want nothing to do with for the previously mentioned reasons.... tim's system sounds different than ray's, which sound different from mine, and lets not forget moke... can he even hear it?   .*  anything other than a normalization, or a simple volume boost is going too far for something that is going to circulate.

** edit to add:  relax ticket nazis... it was meant in jest... moke owns your soul anyhow... it was just a general stab at team aarp, which we will all join one day **

[Brian Skalinder]

I think I'm right when I say that if you have a recording with 6db of headroom, then boost the track 6 db in post. This would change your noise floor. I have to assume it would be the same change as turning the volume up 6db on playback though too. At least I think it would be the same.

As I understand it, adding the gain in post, in the digital realm, wouldn't change the noise floor so much as raise the noise floor along with the audio.  The relative difference between audio and noise floor remains unchanged.

For example, if the audio peaks at -7 dB and the noise floor's at -77 dB (a difference of 70 dB), adding +6 dB in the digital realm would result in audio of -1 dB and noise floor at -71 dB (still a difference of 70 dB).

This is why it's so important to get your levels set well while recording - it maximizes the audio relative to the noise floor.  Using the example above:  assuming the same noise floor at the time of recording, -77 dB, setting proper levels at the time of recording to say -1 dB would result in a difference between the audio and noise floor of 76 dB.  So setting your levels properly at the time of recording maximizes the audio relative to the noise floor.

[Jhurlbs81]

So for those of you who say, "no pre-EQ"..what about HPFs on the V3?  I always note which setting was used, (Slope 1 or 2).  Thoughts?

I have heard the argument, "you can always roll off the bass after the fact" 

But if you are in a bass heavy room/location wouldn't running the HPF give you more room for the other frequencies? 

[Digital Quality]

Scaling in the digital domain is also adds a new layer of errors - requanitzation noise. It's not clear to me if the tools take care of this with some added dither or if they leave it up to us to do. Either way, errors are added each time you do a digital scaling. Dithering can fix it up somewhat but post processing is destructive.

[bconnolly]

So how many of you secretly chuckle and roll your eyes when you see the word "remastered" listed in a DIME torrent?  I know I do...

I chuckle out loud.  Verbosely.

As far as post-processing, I don't like doing anything too involved or crazy with what I record though I do delight in having them boosted a little bit in Wavelab to sound "fuller" and "cleaner".  I sent my 2006-01-28 recording to Kfrinkle and he did some jazzy jazz to it in Wavelab.  I got it back from him and was amazed at how much nicer the recording sounded.

Is that so wrong?

[jimi56]

Yeah that makes sense about the noise floor.  I might just be thinking of times when i had to do some crappy transfers to a PC via mic port.  That just introduced more noise, obviously.  (don't worry, that was long ago and i had no idea what i was doing..never will happen again )   

All this level talk just makes me want to get a set of meters for the UA-5 that much sooner.... 

[Music is alive]

It's ironic this topic appears here, now,  since - as a newbie taper - this is something I've been thinking about a lot lately.

When folks do choose to do things post production, what are examples of things that are commonly done?  This is assuming the doer is a regular civilian, not a pro mixer/engineer.

Brian S - please forgive the newbie question, but this post greatly intrigues me:

This is why it's so important to get your levels set well while recording - it maximizes the audio relative to the noise floor.  Using the example above:  assuming the same noise floor at the time of recording, -77 dB, setting proper levels at the time of recording to say -1 dB would result in a difference between the audio and noise floor of 76 dB.  So setting your levels properly at the time of recording maximizes the audio relative to the noise floor.

You've put your finger right on - without breaking out in newbie detail - a Q that I've struggled with, which, succinctly, is:  what is the difference between getting your levels just right during recording and (assuming you didn't come out with a badly clipped show) boosting it in post production?  

I understand there is a difference, but if you could use a bit of patience and expand on exactly the meaning of the "audio" and "noise floor", and their relation, for those of us who don't explicitly understand such wording, you'd be helping me a good deal.

[TNJazz]

Brian's explanation is very concise and clear.  If it seems a little fuzzy to you still, just try it at home with your stereo and you'll hear exactly what the difference is with your own ears.

Put your mics up to one of the speakers and set your levels so it hits around -2.

Record a 30 second clip or so.

Then back down the gain so the levels hit at -12 or -15.

Record another 30 second clip.

Then throw your samples into an audio editor and make them exactly the same volume.

Then listen to them and see if you can hear a difference.  I'm betting you will.

[zowie]

So how many of you secretly chuckle and roll your eyes when you see the word "remastered" listed in a DIME torrent?  I know I do...

I chuckle out loud.  Verbosely.

As far as post-processing, I don't like doing anything too involved or crazy with what I record though I do delight in having them boosted a little bit in Wavelab to sound "fuller" and "cleaner".  I sent my 2006-01-28 recording to Kfrinkle and he did some jazzy jazz to it in Wavelab.  I got it back from him and was amazed at how much nicer the recording sounded.

Is that so wrong?

"If it sounds good, it is good."  - Duke Ellington.
Corrolary: As long as there remains an unprocessed master around to go back to.

[Digital Quality]

SNR goes up 6db for each added bit in a digital signal. A full 16 bit signal correlates to 96dB available SNR; 24 bits results in 144dB of available SNR. If you record at less that optimal levels, it's just like leaving off (msb) bits on your final signal. For each bit you don't take advantage of, subtract 6db SNR.

Scaling afterwards will scale the data signal and the noise signal equally but then it also adds some new noise signal. To get the highest possible SNR in your recording it's best to use up as much of your resolution as possible up front. It can not be recovered later, all you can do is make it worse..

Now, if it's so quiet that you can't listen to it or your playback is super noisy at high gain levels, by all means scale it up. If you are trying to recover the last few dB you'd be best to leave things alone. In between there your ears are the best judge, do what sounds best to you. I try to never do anything extra to my recordings because they sound best to me that way.

[zowie]

Scaling afterwards will scale the data signal and the noise signal equally but then it also adds some new noise signal.

Could you explain the basis for that statement?

[Digital Quality]

Requantization noise is added. The scaled words must be brought back into the 16 bit domain. The difference between the scaled number and it's 16 bit word is the new noise. It can be helped by adding another level of dithering but not eliminated.

[cleantone]

SNR goes up 6db for each added bit in a digital signal. A full 16 bit signal correlates to 96dB available SNR; 24 bits results in 144dB of available SNR. If you record at less that optimal levels, it's just like leaving off (msb) bits on your final signal. For each bit you don't take advantage of, subtract 6db SNR.

Scaling afterwards will scale the data signal and the noise signal equally but then it also adds some new noise signal. To get the highest possible SNR in your recording it's best to use up as much of your resolution as possible up front. It can not be recovered later, all you can do is make it worse..

True. To the best of my knowledge anyway. I just want to note or add that the noise floor can be room tone and/or inherent noise from any peice of gear in the chain. Which of course is why you typed "available" SNR and not just SNR. Thought that might be helpful info for some people.

[zowie]

Requantization noise is added. The scaled words must be brought back into the 16 bit domain. The difference between the scaled number and it's 16 bit word is the new noise. It can be helped by adding another level of dithering but not eliminated.

If we're talking about the same thing, I don't think of that as noise but as distortion.

[Digital Quality]

Same thing. Distortion is the addition of a noise signal. Noise comes from a lot of places, quantization, interferance, components (even resistors add noise), not even considering frequency response of the system. Everything in front of the mics is considered to be "signal", even room noise. 96dB is the theoretical limit just for signal to quantization noise I believe. All the other stuff is added on.

Another strange thing about quantization is that you will still have added noise even if you say, "I'll just multiply everything by 2.0, that way there are no more rounding errors." The rounding errors are eliminated that way but you get a new effect of skipping quantization levels in your new signal. This I don't totally understand yet but apparently it's also a problem with the D/A converter. The quantization skips present themselves as noise when converting back to analog.

[wbrisette]

For me, I sometimes do it, sometimes I don't. All depends on the venue and what the recording sounds like. But, I also always do multitrack, so usually when I eq a track, it's a particular microphone and once tossed into the mix, rarely makes huge differences in the sound. I will also point out that many years ago, I use to master recordings for a living at a recording studio/cassette duplication facility. I learned a long time ago not to mess around with the recordings too much. I also learned a long time ago that it is better to subtract rather than add things to recordings (need more bottom end? drop the top and mids a few dB). Anyhow, I rarely put any of this info on my recordings, but a lot of that has to do with how I have to use a multi-track editor in post to do my work.

Wayne

[Digital Quality]

On the nicer tools, do they do extra dithering for you when you process or is that all left up to user? I have a cheapo wavelab student version it's pretty basic in terms of features. If you were doing several things it would be best to do them all at once and only requantize at the end is that possible??

As for UA-5 meters, I have a suggestion. Take it to some shows you don't really care about and run it so hot it makes you want to cry. The light should be blinking fairly regularly during the loud parts of the show but not on every beat and not glowing. Try it a little hotter during the 2nd set, remember you don't care if you limit or not. Get a feel for different brightness levels that it blinks and it gets a lot easier. I guess the led just tics on at -3db or so, is that right?

Your goal should be to just clip a few samples from time to time during the show IMO. The UA-5 handles a few clips with no problem but over all the time will wreck it. Any other opinions on that?

[Brian]

i know what i'm doing(or at least like to think so at this point ) so i'm all for it

Most 2 track shows I do now I send through Waves REQ 6 > L3 in Nuendo 3.  I do not use normalization at all as I don't think it sounds as good as a mastering limiter.  I also don't smash the hell out of the recordings either.  The out ceiling is at -.03db and I set the threshold at no lower than -3db depending on the initial recording levels so that it only limits the loudest parts and transients of the recording.

I've been quite happy doing this.

[VA_TAPER]

as far as EQ'ing... that is something that i want nothing to do with for the previously mentioned reasons.... tim's system sounds different than ray's, which sound different from mine

Maybe it's just me, but I'm an EQ guy (if it needs it).  EQ'ing your system and listening at reference levels (75-85db) is important before EQ'ing your show.  I run digital room correction, so I am not only EQ'ing out speaker/amp issues in my playback system but also phase, standing wave and time alignment issues.  But if we're talking EQ seting up your system is a cinch.  

1. Get on eBay and get an EQ ($20-$???)
2. Go to ratshack and get a dB meter
3. Get a testdisc of tones and pink/white noise
4. Sit at your listening position and calibrate the frequencies as needed

Now you are set up like a true monitoring system and can adjust you recordings to ear.  

mmmatt posted earlier a portion of an email conversation we had on this topic here's a snippet of it that shows my stance on the "debate":

>      3.  Do (did) you do any post on your files other than conversion? Everything seems so well balanced vocals > music > crowd, that I'm wondering if that is straight of the pre or if you tweaked it.  Maybe Glen travels with a kick-ass engineer?  I hear very little "room" in this recording.  Do you pull out resonance frequencies, or was this room just real dead... lack of heavy bass helps, but still.

I do post work on the shows, minor stuff but it brings
out some of the qualities you mentioned; what allows
me to do the post work is the frontend gear, by
capturing a clean signal without harshness I can adjust
several parameters (minor eq, compression, etc.) with
out bringing out the issues these tools did with other
rigs I've owned.  I do eq out seriously resonant
frequencies, this is done mostly by ear now, I run an
HTPC to do mastering.  If I hear something
(frequency wise) to strong I'll cut it using Waves
LinEq I never boost or cut by more than 3db, but 3db
can make a huge difference.

I cascade compressors, instead of using 1 compressor
for 6db of squishage, I'll use 3 different compressors
each providing a mild 2db.  This allows different
attack and release profiles to thwart pumping and
noise, but it allows the gentle equalization of volume
so that the banter can be heard and it brings out the
nice reflective qualities in the hall.

I de-ess as well to compensate for the issues of
recording a compression horn tweeter.

I used to add harmonics to the mix too, for bringing
out the highs, but the tube pre has completely
eliminated the need to do that.





     Thanks for the help.  I'm sure your gear helps make the recording sound so good, but that is just the flavors you like.  I wasn't sure how you feel about sharing so I didn't post this to the thread if you didn't want to answer these questions.  If you would rather I will post and let everyone hear the answers!  Once again if you don't want to answer these questions, there are no hard feelings.
 Matt

 
well, that turned in to more than a quick answer!

feel free to ask any thing else, really I buck
convention on alot of what I do.  People look down on
mastering, but no one plays their recordings back on
18" woofers and horn tweeters at over 105db so that
just doesn't make sense to me. I tape and master for
my ear, which is a little phucked up due to 7 years of
marine corps infantry, but generally most people like
what I do, so I keep it going.  Also, incase you haven't
picked it up I'm not an accuracy type of guy, I could
care less if my recordings sound like the venue, unless the
venue sounds good.  I'm a euphonic guy, I want my
recordings to sound like you're listening to a vinyl
record of the concert played through a vintage tube
amp.  


 

[jeromejello]

i should possibly clarify, i do EQ if necessary, but just thru my playback.  my point is that i dont EQ the file that i seed b/c of the differences (and lets face it, i am barely holding on to the craft of taping, let alone a whole new skill of post production) that can be experienced across the board.  i find most people prefer to have the sound in as close to a 'as it was' form and provide thier own tweaking with their own equipment for thier own taste.

[VA_TAPER]

i should possibly clarify, i do EQ if necessary, but just thru my playback.  my point is that i dont EQ the file that i seed b/c of the differences (and lets face it, i am barely holding on to the craft of taping, let alone a whole new skill of post production) that can be experienced across the board.  i find most people prefer to have the sound in as close to a 'as it was' form and provide thier own tweaking with their own equipment for thier own taste.

I'm not trying to convince you to EQ your recordings either, everyone gets in to this hobby for their own reasons, love of music, etc.  Not everyone gets as crazy/anal with it as some of us do (i'm in the crazy/anal group), but my point is there is not 'as it was' form.  Move your mic stand 10' to the left and you've got a potentially totally different sound than where you set up at, run a different pre, run a different polar pattern, 16/24 bit, small/large diaphgram, schopes/oktava all can do WAY more to the recording than some corrective mastering. You're the taper, be empowered (if you want to); do what ever you want to the recording that you think sounds good, seed it that way.  If people don't like they can tape it themselves, or you can learn from the critizisms. 

Either way, do what makes you happy. 

Here's an mp3 of one of the shows I taped and did a little mastering work on:
http://www.archive.org/download/glen2005-08-30.dsd.flac/glen2005-08-30set2t11_vbr.mp3

check it out, let me know what you think, I always appreciate new critics, it helps me hone in on the sound.

peace, and keep spreadin' the tunes,

chris

[jeromejello]

i should possibly clarify, i do EQ if necessary, but just thru my playback.  my point is that i dont EQ the file that i seed b/c of the differences (and lets face it, i am barely holding on to the craft of taping, let alone a whole new skill of post production) that can be experienced across the board.  i find most people prefer to have the sound in as close to a 'as it was' form and provide thier own tweaking with their own equipment for thier own taste.

but my point is there is not 'as it was' form.  Move your mic stand 10' to the left and you've got a potentially totally different sound than where you set up at, run a different pre, run a different polar pattern, 16/24 bit, small/large diaphgram, schopes/oktava all can do WAY more to the recording than some corrective mastering. You're the taper, be empowered (if you want to); do what ever you want to the recording that you think sounds good, seed it that way.  If people don't like they can tape it themselves, or you can learn from the critizisms. 

Either way, do what makes you happy. 

extremely valid points... that said i still suck at post 

+t for the empowerment

[zowie]

I have a Radio Shack spl meter.  It serves several purposes for me, but I don't think it's accurate enough for adjusting eq.  Probably the cheap mic is far from flat.  I got better results eq-ing by ear with sweeps and white noise. 

That was back when I had a mid-fi playback system.  With a good hi res system, adding a $20 or $100 equalizer plus an extra pair of cables does more harm than good, IMO.


I eq recordings sometimes in post.  A dab of eq can go a long way with a tape done with cheap or badly positioned mics or a bad room.  More than a dab is usually a bad thing.

[VA_TAPER]

I have a Radio Shack spl meter.  It serves several purposes for me, but I don't think it's accurate enough for adjusting eq.  Probably the cheap mic is far from flat.  I got better results eq-ing by ear with sweeps and white noise. 

That was back when I had a mid-fi playback system.  With a good hi res system, adding a $20 or $100 equalizer plus an extra pair of cables does more harm than good, IMO.


I eq recordings sometimes in post.  A dab of eq can go a long way with a tape done with cheap or badly positioned mics or a bad room.  More than a dab is usually a bad thing.

Today's $20 EQ is yesterdays $1000 eq, why? Because most systems (most non-audiophile systems) are multichannel and having 3 EQs to set is impractical.  That doesn't mean you can't find a reference grade EQ on the cheap.  And if you think an extra pair of cables is going to harm a line level signal, god help the poor snakes that the puny mic / instrument signals were run through in the studio.

I would also argue that a dab of eq is required in most circumstances because you are not (I say this generally, it is possible you have a V-DOSC array) playing the recording back on the same type of system it was originally played on.  In the studio is different, you can eliminate the need for EQ because the is no intermediary playback system, but to ask even a pair of $6k towers to play with the response of a bandwidth limited, compression horn driven concert PA is not realistic, let alone a car stereo (wait, make that factory car stereo =)  There is also the compensation for the fletcher-munson curve and bring the EQ in line with a reasonable playback volume.  Most music is not listened to at concert levels for home playback, so the sensitivity of different frequencies changes with respect to volume.

And about the RS spl meter, it might not be perfect, but it can be used to improve system response, it uses a panasonic omni type cap that is very flat, but very noisy.  If you use a computer to feed your DAC with nice low jitter SPDIF, I would highly reccommend renting an earthworks omni, and Lunatec V3 to run a full set of tones, sweeps and impulses through WAVES convolver, you can set it up to create and inverse acoustical image of your room approching the quality of some very expensive DRCs.  Then it's mearly a few plug-ins to route all audio playback through the inverse image and suddenly things become quite clear =)

peace, chris

[zowie]

I have a Radio Shack spl meter.  It serves several purposes for me, but I don't think it's accurate enough for adjusting eq.  Probably the cheap mic is far from flat.  I got better results eq-ing by ear with sweeps and white noise. 

That was back when I had a mid-fi playback system.  With a good hi res system, adding a $20 or $100 equalizer plus an extra pair of cables does more harm than good, IMO.


I eq recordings sometimes in post.  A dab of eq can go a long way with a tape done with cheap or badly positioned mics or a bad room.  More than a dab is usually a bad thing.

Today's $20 EQ is yesterdays $1000 eq, why? Because most systems (most non-audiophile systems) are multichannel and having 3 EQs to set is impractical.  That doesn't mean you can't find a reference grade EQ on the cheap.  And if you think an extra pair of cables is going to harm a line level signal, god help the poor snakes that the puny mic / instrument signals were run through in the studio.

I would also argue that a dab of eq is required in most circumstances because you are not (I say this generally, it is possible you have a V-DOSC array) playing the recording back on the same type of system it was originally played on.  In the studio is different, you can eliminate the need for EQ because the is no intermediary playback system, but to ask even a pair of $6k towers to play with the response of a bandwidth limited, compression horn driven concert PA is not realistic, let alone a car stereo (wait, make that factory car stereo =)  There is also the compensation for the fletcher-munson curve and bring the EQ in line with a reasonable playback volume.  Most music is not listened to at concert levels for home playback, so the sensitivity of different frequencies changes with respect to volume.

And about the RS spl meter, it might not be perfect, but it can be used to improve system response, it uses a panasonic omni type cap that is very flat, but very noisy.  If you use a computer to feed your DAC with nice low jitter SPDIF, I would highly reccommend renting an earthworks omni, and Lunatec V3 to run a full set of tones, sweeps and impulses through WAVES convolver, you can set it up to create and inverse acoustical image of your room approching the quality of some very expensive DRCs.  Then it's mearly a few plug-ins to route all audio playback through the inverse image and suddenly things become quite clear =)

peace, chris

I saw a test about 5 years ago that showed audible peaks and valleys in the freq. response of the RS SPL meter.

If you can show me a $1000 eq for $20, I'm interested.  I see $300 eqs for $20 but I don't think they're adequate.

I am utterly convinced that cables change the sound of line level signal; change may or not be "harm."  Cheap cables do harm it.  But I don't want to have that hackneyed debate here.

[RebelRebel]

might want to be careful linking to discounted software..
I only know because I got reprimanded for it here.

[momule]

On the nicer tools, do they do extra dithering for you when you process or is that all left up to user? I have a cheapo wavelab student version it's pretty basic in terms of features. If you were doing several things it would be best to do them all at once and only requantize at the end is that possible??

That's one of the Big Differences In Wavelab and SF for me. Is that Wavelab A. allows to you hear your effects Real time and B. adds a group of Effects at one time .

As for UA-5 meters, I have a suggestion. Take it to some shows you don't really care about and run it so hot it makes you want to cry. The light should be blinking fairly regularly during the loud parts of the show but not on every beat and not glowing. Try it a little hotter during the 2nd set, remember you don't care if you limit or not. Get a feel for different brightness levels that it blinks and it gets a lot easier. I guess the led just tics on at -3db or so, is that right?

This has been My experience.  I found out after running the UA-5 for just one show that the clip light was not really its clipping point.

Your goal should be to just clip a few samples from time to time during the show IMO. The UA-5 handles a few clips with no problem but over all the time will wreck it. Any other opinions on that?

I would say if your not seeing the clip light blink every few sec's your prolly not running hot enough.  Im amazed at how low some folks run. 

Just for the record I agree With VA taper.  +T

I would also argue that a dab of eq is required in most circumstances because you are not (I say this generally, it is possible you have a V-DOSC array) playing the recording back on the same type of system it was originally played on.  In the studio is different, you can eliminate the need for EQ because the is no intermediary playback system, but to ask even a pair of $6k towers to play with the response of a bandwidth limited, compression horn driven concert PA is not realistic, let alone a car stereo (wait, make that factory car stereo =)  There is also the compensation for the fletcher-munson curve and bring the EQ in line with a reasonable playback volume.  Most music is not listened to at concert levels for home playback, so the sensitivity of different frequencies changes with respect to volume.

[grtphl]

I know this thread is _old_, but it's been a really great reference for me.  I thought it was worth a bump so others can read it and in-case anybody has anything else good to add.

Edit: This thread has been really helpful too.