Old 04-03-2009, 10:28 PM   #481
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Sampling at 44.1khz doesn't mean recording 44.1khz. It is taking a sample of(or look at)a waveform 44.1 thousand times per second.
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Old 04-04-2009, 07:06 AM   #482
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Default Invisible sounds and the quest for harmonic highs

PAPT said "On my own mixes, if I try boosting at 12k or higher I notice that there is actually not much of anything there but hiss."

Yep said "Content below around 45Hz or above about 12kHz falls under the "special effect" category... just drag a copy into REAPER and filter it for those frequencies to see what they sound like... (my guess is very little if not zero on most material, and it may even improve). My point was not that the ultra-high end is bad, just that it's not ipso-facto GOOD, especially if it mostly or entirely consists of hiss and electrical noise, which is likely if there is not any significant musical content up there."

----

This point needs discussion. PAPT has not stated his signal chain. Poor quality equipment results in failure to reproduce sound with accuracy, as compared to the real life source. This shows up most evident in the "extremes". For example content above 12khz, and below 60hz can become non existent, but most often somewhat present but nasty, harsh and very unclear on bad equipment rather than absent. Bass is much more prone to disappearances with bad equipment.

While we should make the most of what we have, good quality reproduction is vital. Otherwise we spend time practicing to shoot at a target with our eyes half shut. No matter how skillful we become working with this self imposed disability, once we open our eyes we will immediately surpass our best score, and with much less effort and time.

If you reference a quality commercial recording and you can't hear any change after making a 12k lowpass cut, or if the sound improves(!) - your monitoring, DAC or hearing have a serious issue. This is going to brick wall any attempt to produce quality material of your own beyond pure good fortune. Your efforts might sound "just as good as a CD" on your hampered setup, but when presented on good monitoring/conversion it will fall flat on its face.

Above 12k contains vital frequencies that contribute to the "sheen and polish" of a track. Every commercial track has strong clear highs and upper harmonic content. PAPT can even see this on his visual analysis, this vital clue must not be ignored.

Tracking is even more complex a chain. Lets say when listening to high quality commercial recordings you do notice a distinct loss of important sound when cutting above 12k. From this we can conclude your monitoring/dac is of at least a certain standard. But moving onto your own tracks, (excluding instruments such as bass), when you isolate the 12k+, all you find is noise and hiss, in contrast to the rich airy content you hear on commercial recordings. Visual analysis confirms the absence of extreme highs in your recordings compared to commercial offerings. So you boost these frequencies. The visual analysis starts to look more like the commercial recording, but to your dismay all you have done is increase the noise, harshness and haze, as Yep describes "the clutter". Your mix is worse with the 12k boost than without.

In this scenario your issue is one or combination of: source issues, mic technique, microphone quality, mic preamp quality, ADC quality. Just one link in this chain will ruin the rest. If you have no "good" high frequency content, you are not capturing the harmonic highs that are being captured by the commercial offering. This is not a factor engineering skill can overcome. Poor equipment can never reproduce what good equipment can reproduce, and with such limitations Yeps advice is your only choice - cut out the noise and do the best with the frequencies you can capture.

I appreciate amongst the recording community there is a strong reluctance to place any blame on quality of the tools. Many declare good equipment is simply placebo, they argue that anything will do (but ironically they exclude all equipment of lower quality than *they* currently own). The fantasy is that a great engineer can cheat reality and reproduce a real sound with equipment that cannot do so.

Note that I say "good" equipment, not "expensive" - as the two don't necessarily correlate. If you form a basic understanding of the components gear is built from, you can work out which equipment competes admirably with the 10x expensive items, but which are also only 1.5-2x more expensive than bottom of the barrel.

In conclusion, extreme highs do count. A global rule of hacking them off is a disasterous conclusion to draw, but necessary action if your equipment causes them to sound bad, or simply does not reproduce them.

Keep up the great work on this thread Yep.
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Old 04-04-2009, 08:43 AM   #483
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Quote:
Originally Posted by DerMetzgermeister View Post
Hi yep.
I have a question about sample rate. Like everyone else I have read tons of threads and articles about the subject, yet it's not 100% clear to me.

I'm quite sure that my recording gear is not capable of capturing sounds above the range of a 44.1 KHz recording.
So, there is no valid reason then to record at a higher sample rate?
And, if there is an advantage to recording at 88.2 KHz or higher, what it is?

Thanks in advance.

Edit: Sorry if the theme have been addressed previously in this thread.
This is a big, sprawling, and somewhat controversial topic that might be better-suited for another thread, but here's a short version:

First of all, there is not necessarily any need to "understand" all the technical aspects of digital audio in order to make good recordings. In fact, a little bit of knowledge can be a dangerous thing, an awful lot of people would be making better-sounding recordings if they stopped to trying to "understand" or "figure out" the technical stuff and simply trusted the equipment designers to do their job.

Most serious people generally agree that human beings cannot hear above 20kHz or so. There are some who think that people may be able to subtly "perceive" higher frequencies, but it's certainly not critical for good-sounding recordings.

And a 44.1k sample rate WILL accurately reproduce content up to 20k, so it seems like that should be the end of the story. But it's not, necessarily. Because the sounds you are capturing actually have an infinite harmonic sequence that goes above 20k, which needs to be cut off at 22.5k before conversion to digital. The frequency at exactly half the the sample rate is the highest frequency that can be perfectly reproduced by the system, and is called the "Nyquist frequency."

And that filter that cuts off the super-highs, by definition, has to be an analog filter, and it has to be a fairly steep hard cutoff filter. Which is the most artifact-inducing kind of filter, and causes "ripple" effects both above and below the cutoff frequency. EQ is basically just a delay or a series of delays, and very high-frequency eq is basically adding excruciatingly short delays to cancel out certain frequencies. So a poorly-designed or poorly-made filter (or possibly even a good one) will cause artifacts in the audible frequency range, especially if the filter's cutoff frequency is close to to the range of normal human hearing.

How big a deal are these artifacts, potentially? I'll get to that in a sec. For now, we'll just say that a "perfect" AD converter with "perfect" filters WOULD BE perfectly adequate to capture audio for human consumption at 44.1, setting aside exotic theories of supernatural hearing. (and "setting aside" exotic theories can be a more dangerous business than it sounds, when it comes to subjective sensory experiences).

HOWEVER, the "capture" is not usually the only thing that happens to digital audio. It is also likely that you'll be using plugins and processing within the digital realm, and this brings up another set of potential concerns. Processors such as compressors and distortion and analog-emulating tubeifiers and tape sims and so on are all likely to add harmonic distortion, either subtle or pronounced. And harmonic distortion extends the frequency content. What happens when the compressor creates harmonics higher than the Nyquist frequency? Aliasing distortion, that's what.

And aliasing is a pretty ugly and unpleasantly "digital" distortion. When you get frequencies embedded in a digital system that are above the Nyquist frequency, they come out through the playback converters as sort of randomized, fluctuating "subharmonics" of the too-high frequencies, modulating the audible frequency range in unnatural ways. These kinds of digital nasties are responsible for a lot of the "digital synths/guitar effects sound like crap" opinion out there.

Now, clever programmers can and should and usually do come up with ways to handle these problems, which are not necessarily terribly difficult or exotic (internal anti-aliasing filters or oversampling are a pretty good start). But those in turn add another layer of complexity to the audio processing, and a lot of obsessive purist types (such as mastering engineers) prefer to stick with the simplest processors and just start out with high-sample-rate recordings so that any aliasing artifacts are all pushed up into the inaudible range (there is no need to worry about giving the mastering engineer a low-sample-rate recording, they'll upsample it if they want). To each her own.

You, the home recordist/musician, should probably NOT try to "figure out" this stuff, nor to "think through" what kind of approach to aliasing is best. YOU should probably just leave that to plugin designers, and then use the effects that sound good, and don't use the ones that don't. Or, if you have the space and processing power, you could just record at higher sample rates. But high-sample rate mixing eats up a LOT of processing power. Every plugin uses 4x as much CPU on 192k audio as it does on 48k audio, so the tradeoff is not insignificant. And get ready for your head to explode, because higher sample rates can actually sound WORSE than lower sample rates in some cases.

The single biggest problem with AD conversion is jitter. "Jitter" is what happens when the samples are taken at non-perfect intervals. The playback converter is and must be counting on perfectly-spaced samples, and it will reconstruct a waveform based on the expectation of same. If you sample a pure sine wave at non-perfect intervals, the playback converter is going to "re-space" those samples perfectly (or as perfectly as it can), and the sine wave will come out all crooked and stretched and squashed in weird ways, which sounds like lots of ugly and random harmonic distortion. "Digititis" in short.

The thing of it is, the higher the sample rate you record at, and the faster those samples have to be taken, the more potential instability there is. So especially with cheaper converters, higher sample rates can actually come out with worse jitter than lower sample-rate recordings. There is no free lunch. And forget what you may have heard, you CANNOT EVER improve an AD converter with a fancy external word clock. Always record with your audio interface set to internal clock. If you don't know what this means, ignore it, it almost certainly doesn't apply to you.

Remember all the way back in the beginning of this thread when I said that CONFIDENCE in you gear was more important than having GREAT gear? And that test, of recording a great-sounding CD or record through your soundcard's inputs to see if it still sounded great? And how that tells you the results that you can and should expect of yourself and your rig, sound-quality-wise?

Last but not least is the consideration of target medium. Sample rate conversion is very, very easy to do perfectly at exact multiples of the target sample rate. If you want to convert 88.2kHz to 44.1k, all you need to do is to throw away every other sample. But if you want to convert 96k to 44.1, then the SRC has to sort of interpolate or "figure out" where the sample points would have been on the original analog recording. And this creates the potential for aliasing (in fact it actually causes at least some aliasing in all real-world sample-rate converters, AFAIK, but the best ones are extremely good at it.) So to keep things simple for best audio integrity, if you're recording for CD, you should probably stick to 44.1, 88.2, or 176.4.

I recommend against trying to "think through" this stuff. There is way too much to know and way too much to keep track of and way too much room for "paralysis by analysis," as my father would put it. More importantly, it is way too easy to lose focus on LISTENING when you start THINKING. There is not and doesn't HAVE to be a "perfect" sample rate, and even if there was, you wouldn't have to use it to get great-sounding recordings. Record at a sample rate that makes sense practically and sonically for what you're doing. Try the "record a CD" test at different sample rates if you like, and see if you can hear a difference.

PS-- None of the above has anything at all to do with bit depth, which is much simpler: always record at 24 bit. The CPU hit is negligible on a modern DAW and the benefits are much more clear-cut than with sample rate.
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Old 04-04-2009, 09:24 AM   #484
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...Above 12k contains vital frequencies that contribute to the "sheen and polish" of a track. Every commercial track has strong clear highs and upper harmonic content....
As I was reading this, the radio happened to be playing "Don't Fear the Reaper" by Blue Oyster Cult. Say what you will about the song, but in my opinion it's a very good recording, with rich, airy highs and a great sense of spaciousness, as well as gobs of "sheen and polish."

So I went and dug up the CD, and ripped it as a WAV file, and loaded it into Voxengo SPAN for a frequency analysis. Let's take a look:



What do you know? I see a sharp cutoff starting at about... 12k! And essentially zero content above 15k! (incidentally there is also a pretty pronounced low-end rolloff starting at about 100 cycles, which is pretty high.)

Now, maybe there are some people on this message board who have higher standards and would not deign to release a recording that sounds like "Don't Fear the Reaper," but my personal opinion is that this an excellent example of a great studio track.

Again, not to say that extreme highs are BAD, nor even that they're not GOOD, just that they are not mandatory to make a recording sound clear, airy, polished, professional, and hi-fi. I think this song is all of the above.
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Old 04-04-2009, 10:32 AM   #485
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Quote:
Originally Posted by PAPT View Post
Sampling at 44.1khz doesn't mean recording 44.1khz. It is taking a sample of(or look at)a waveform 44.1 thousand times per second.
I assumed that was understood.

Quote:
Originally Posted by yep View Post
I recommend against trying to "think through" this stuff. There is way too much to know and way too much to keep track of and way too much room for "paralysis by analysis," as my father would put it. More importantly, it is way too easy to lose focus on LISTENING when you start THINKING. There is not and doesn't HAVE to be a "perfect" sample rate, and even if there was, you wouldn't have to use it to get great-sounding recordings. Record at a sample rate that makes sense practically and sonically for what you're doing. Try the "record a CD" test at different sample rates if you like, and see if you can hear a difference.
Thank you very much, yep.
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Old 04-04-2009, 01:02 PM   #486
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Point neatly made.

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Originally Posted by yep View Post
What do you know? I see a sharp cutoff starting at about... 12k! And essentially zero content above 15k! (incidentally there is also a pretty pronounced low-end rolloff starting at about 100 cycles, which is pretty high.)
I just wanted to add here that the pitch distance between 12kHz and 15kHz is about a major third--we are not talking about a giant amount of the pitch spectrum no matter where you draw the line above 12kHz and below the limit of human hearing.

So pulling a different recording out and analyzing it and finding the rolloff at 15k or 18k or whatever would not really undermine the basic idea being presented here.
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Old 04-04-2009, 01:34 PM   #487
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Your position looks extremely persuasive visually but I encourage everyone to try the following to hear the reality of 12khz+.

1) Make sure your monitoring at a decent volume, 75db+.

2) Load up a commercial track in Reaper, run ReaFIR and set the EQ point to 12k as in picture "fir0.jpg" attached (therefore blocking off everything below 12k+). You will hear an indistinct, high pitch noise, that is very quiet (as Yep point out, it is 60db or less than the rest of the content and sloping down).

Now we think - "Surely losing this cant make any difference to the sound? It sounds and *looks* like nothing!"

3) Now flick the ReaFir shelf the other way (as in attached diagram fir1.jpg) to block off everything past 12khz. Now you are only hearing from 0-12khz. Listen for about 1 minute.

You are probably thinking "Sounds like nothing is missing, if anything it sounds even smoother than before?"

4) Now go to the mixer window and flick the FX to bypass so your thrown back listening to the original track with no interruption. The high end will seem almost unbearable.

See how much you lost by shaving off 12khz+? You have to measure with your ears, not visually. But we can also see some evidence of this visually too. To get a better view, change the ReaFIR analysis floor to say -100db, you will see content and activity all the way up to 20khz. Hearing that sound in isolation does not give any representation of its effect on the sound as a whole. Neither does looking at it on a linear chart which is not to the same scale as your hearing. In the context of a the rest of the frequencies it is a vital component. If you cant hear a difference please re-read my first post to see reasons why this might be the case.

The Blue Oyster Cult example provided by Yep is a nice smooth sounding old fashioned recording. If you examine the highs at -100db ReaFIR analysis compared with a modern track, modern songs have significantly more 12khz+ frequency content. If you want to compete with modern recordings you are going to need 12khz+ even more so than if you want to reproduce the good old days.

Personally I dont believe the extended highs in modern records are "true to real life", but they are the standard, the trend and the demand. If your aim is modern sound, excellent converters and preamps are a must in order to capture those frequencies accurately. If you try to reproduce the sheen and cut of a modern track with poor equipment by boosting 12khz, you simply raise the noise of an unclear mess.

Like I said in my last post, if you equipment is lacking you should follow Yeps advice and cut back that indistinct/noisy high end and be happy with a smooth old school sound. Modern engineers regularly boost above 12khz and 15khz. But they are in a unique position, they have the means to capture above 12khz in pure accuracy with the best equipment, it is not noise they are boosting but goodness.

If your gear is not great, follow Yeps advice and go lofi! The result will be better than trying to compete frequency wise with modern commercial tracks and ending up with a nasty harsh high end. The song is the most important thing anyway.

---

Dear Yep, I hate to argue as your thread is one of the best I have ever read. Your writing and points here are better than most commercial books available on mixing/recording. Look forward to your next posts.

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Old 04-04-2009, 01:35 PM   #488
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Old 04-04-2009, 02:11 PM   #489
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GrantsV and Yep, I don't really see an argument here, fwiw. Both points are exactly correct from their relative perspectives...Post On!!!!
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Old 04-04-2009, 03:24 PM   #490
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Personally I dont believe the extended highs in modern records are "true to real life", but they are the standard, the trend and the demand.
Sounds like a similar situation to the loudness wars.
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Old 04-04-2009, 03:40 PM   #491
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I don't want to argue the point, and I certainly don't want to "bully" the thread, so this will be my last post on the topic unless a new question comes up.

The fact is that "Don't Fear the Reaper" DOES have a pretty pronounced high-end rolloff at 12k. And that's not because Columbia Records did not have access to high-quality gear, and it's certainly not because they were going for a "lo-fi" sound. That there is some tape hiss at -80 or so might or might not be significant, and if anyone thinks that the production on the track sounds "lo fi" or "old fashioned," well... to each his own. I picked the track because it struck me as a production with lots of polish and sizzle and rich airy highs. Maybe some prefer something else...

But you'll find the same results with, for instance, Miles Davis' "Kind of Blue," the Beatles' entire catalog, and basically anything released before the mid-90s, and a great deal of the stuff released since. The old RIAA mechanical rule stipulates a 30-degree shelf cut from 12k up for vinyl release (and a similar cut at 47k), and that was just applied as a matter of course before the record even went to duplication. Which is why I knew I could pick any song released pre-CD to illustrate the point (yes, I already had the answers to the test and therefore cheated).

A great many big-name producers and engineers still do the same as a matter of course, and there are still records released on vinyl, although that might sound too "lo fi" for the iPod age... to each their own. Don't take my word for it, try it and see if it sounds better.

Moreover, and with specificity to GrantV's points, is it even possible to make a rock record with a noise floor below -85? I mean, is a Les Paul plugged into an overdriven Marshall amp even capable of such a low noise floor? (mine certainly isn't, even with a noise gate pedal, but maybe that's just me) How about a Hammond B3 plugged into a real Leslie speaker cabinet? Unless I'm missing something, these are not instruments that can HELP but produce hiss and fizz above 15k, regardless of how you record them.

And even if you have excruciatingly quiet instruments, you're certainly not going to get such a low noise floor using gear such as, for example, an AKG C12, or a U47 plugged into something like a Neve or Telefunken preamp... are those "low quality"?

And if you start resorting to gates and expanders to clean up the recording, then what's going to be left above 12k on those kinds of instruments anyway? Just pumping hiss, basically. Moreover, if we assume listeners are playing back at an average 83dB SPL, then what are they going to hear at -85? If they play back louder, then is their short-term hearing really going to be sensitive enough to pick up that stuff, even if they are listening on ADAM S7s?

I know that there are a lot of modern, all-digital records that have significant content above 12k. Some of that is because the engineers don't know what they're doing, and some of it is because of the genuinely extended frequency range available with digital. But there are also a lot of big, modern-sounding, very airy commercial releases that still follow the old RIAA curve. Even in spite of Great River preamps and Lavry converters and access to noise-free virtual instruments and all the rest of it.

I do absolutely encourage anyone and everyone to think and especially to LISTEN critically, and second-guess everything, including myself. I am very happy that GrantV brought up what a lot of people think about the ultra-highs, and I don't actually even disagree with anything he's saying, only with some of the implicit conclusions that people tend to draw, e.g. that if you have good enough gear, the extreme highs always contain important content, or conversely, that if rolling off the extreme highs improves the sound, then something is "wrong" with your equipment.

There is a lot of internet feuding and such about what "should be" important, or what "ought to" sound better in a theoretical or academic sense, and some of it has some merit. But in the studio, trying to make a killer recording that has clarity, impact, drama, and sonic excitement, a lot of what "should" be the "correct" approach falls by the wayside. Especially when we are dealing with overloaded amateur arrangements such as three guitars all playing in the bottom octave behind a baritone singer.

Some of the great consoles feature high eq that is famous for the ability to crank the knob and create screaming hype and punch in the highs (the old Neve modules with the 12k high shelf spring to mind). And this was commonly done during tracking, especially to boost the highs before they got buried under tape hiss, and then it was just as commonly backed way off at mixdown, like a form of DIY noise reduction, and then rolled off at 30 degrees come mastering. But the "character" was still there, and a lot of modern engineers with only a halfway understanding of audio read and hear about those legendary studio practices and want the noisy, hissy console modules so they too can get those "magic highs," not understanding that the process was inseparably related to the sound of 15ips tape, and that just because legendary producer X did one thing at tracking, it doesn't mean she didn't do an opposite thing later at mixdown.

In fact, from the sound of "Don't Fear the Reaper," I will bet long odds that at one stage, it was a very treble-heavy project, either from high boost or low cut. But there has obviously been a steep, across-the-board high-cut applied at some stage.

Last but not least, here are the two of the hypest, most modern-sounding album tracks I could think of off the top of my head (among records I own-- something by Christina Aguilera might have been a better pick). The first is Mr Machoman by the Lords of Acid, and the second is New Kicks by Le Tigre. Both feature extremely hot, hype production, with a big mix of samples, real, and electronic instruments. I swear they are the first two songs that came to mind. Either one would soncially drop right into a Hollywood Sci-fi/Action flick. I defy anyone to call either of them "old-fashioned" sounding. And they BOTH show the distinctive, pronounced 30-degree shelf rolloff at about 12k and 50 cycles (although neither drops off as sharply as Don't Fear the Reaper). Here they are:




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Old 04-04-2009, 04:06 PM   #492
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PS-- the real point: notice how every one of these graphs is the exact opposite of the instinctive "smiley" EQ curve that people tend to reach for as a first resort. If you've been cranking the lows and highs, and producing weak, mushy, fizzy, quiet-sounding records compared to these, then it's time to re-think.

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Old 04-04-2009, 06:19 PM   #493
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Hi Yep


Thank you for continuing to contribute awesomeness and field questions.
(It just so happens that i have one, hehe )


When mixing the simple scenarios I am showing below, what can I do
to make them sound a little "larger than life" or "sonically bigger" and more
impressive without adding additional instrumentation.


Scenario 1:

[2 Tracks]

Track 1= Acoustic guitar ( rhythm )
Track 2= Lead Vox



Scenario 2:

[3 Tracks]

Track 1= Acoustic guitar ( rhythm )
Track 2= Acoustic guitar ( Lead )
Track 3= Lead Vox


I guess what I am looking for is a way to maximize the impact of a very simple recording.

If you ( or anyone else ) has any insight into this I would
surely appreciate it.

Thanks

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Old 04-04-2009, 07:52 PM   #494
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Quote:
Originally Posted by yep View Post
PS-- the real point: notice how every one of these graphs is the exact opposite of the instinctive "smiley" EQ curve that people tend to reach for as a first resort. If you've been cranking the lows and highs, and producing weak, mushy, fizzy, quiet-sounding records compared to these, then it's time to re-think.
What I find interesting about these two graphs is the pronounced cut at 300-350hz, presumably to tame the low-mids. You see the same but with a narrower Q in Don't Fear the Reaper. A trend?
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Old 04-04-2009, 08:25 PM   #495
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What I find interesting about these two graphs is the pronounced cut at 300-350hz, presumably to tame the low-mids. You see the same but with a narrower Q in Don't Fear the Reaper. A trend?
YES.

See earlier, RE: cutting the fundamental.

Mud range.
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Old 04-04-2009, 08:31 PM   #496
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Originally Posted by mamm7215 View Post
What I find interesting about these two graphs is the pronounced cut at 300-350hz, presumably to tame the low-mids. You see the same but with a narrower Q in Don't Fear the Reaper. A trend?
Yes. That struck me as well, and sent me actually look at some things through Voxengo. It's like a minor smiley in the middle of the broader inverted smiley.

I would think this is one way they're able to get the apparent loudness and fullness up, by skewing the frequencies to the right away from the mid fundamentals, with all of it supported by a carefully shaped low end.

FWIW, some samples.

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Old 04-04-2009, 09:16 PM   #497
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Quote:
Originally Posted by TedR View Post
Hi Yep


Thank you for continuing to contribute awesomeness and field questions.
(It just so happens that i have one, hehe )


When mixing the simple scenarios I am showing below, what can I do
to make them sound a little "larger than life" or "sonically bigger" and more
impressive without adding additional instrumentation.


Scenario 1:

[2 Tracks]

Track 1= Acoustic guitar ( rhythm )
Track 2= Lead Vox



Scenario 2:

[3 Tracks]

Track 1= Acoustic guitar ( rhythm )
Track 2= Acoustic guitar ( Lead )
Track 3= Lead Vox


I guess what I am looking for is a way to maximize the impact of a very simple recording.

If you ( or anyone else ) has any insight into this I would
surely appreciate it.

Thanks
Ah, well, that leads directly to something I haven't much talked about yet, which is reverb, delay, and general "thickening" effects.

Most people who get into recording start from some degree of exposure to either synthesizers, guitar sounds, or both. And these are both areas in which presets and recipes are extremely useful. So beginning recordists are often frustrated by compression and EQ in a mixing sense, where presets are close to useless.

But reverb is a an oasis of useful presets, as long as you're using good reverbs, and as long as you know how to use them. For the near term, I'm going to be using the word "reverb" as a catch-all shorthand to indicate all sorts of ambiance and delay effects, unless I'm specifically contrasting reverb per se with something else, such as short delays, chorus, etc.

"reverb" is not something we ever consciously "hear" in the real world, unless you're in a parking garage or something, but it is all around us. As I touched on earlier in this thread, if someone were to lead you blindfolded through your house (or probably even through a stranger's house), you'd be able to tell whether you were in the living room, or the kitchen, or the bathroom, or a bedroom, just from the quality of the silence.

In studio recordings, reverb might be used to simulate this kind of real-world sense of subliminal "space," or it might be used as a dramatic "effect" to increase the size and scale of a thing. Neither approach is right or wrong, but it's a significant distinction. And making everything sound like giant 80's "drums of God" is not necessarily an improvement.

If you can get the reverb to decay along with the tempo and natural decay of the instrument, and if you can get the reverb tonality and frequency shape to be less present and forward than the instrument, you can often dial in some massive reverb and still have it fall sort of "behind" the dry track, so that the dry track doesn't get washed out or lose its immediacy, but simply has an increased sense of size and spaciousness.

OTOH, if the reverb decay extends past the natural decay of the instrument, or if the reverb has its own frequency profile that is distinctly audible as a separate sound from the instrument, then you'll end up with that 80s drum sound where the it's like the audience is listening to a drum kit and also to a room or a reverb box.

Neither approach is right or wrong.

If you ever play video games, you might be familiar with the phenomenon where, just before the last level, after you've had to fight hordes of bad guys with under-powered weapons and limited ammo, you get the "big gun" that relieves you of such concerns. I recommend viewing reverb the same way. It is something to use AFTER you have got everything else knocked into place, never a substitute or easy "wash" to drown out the problems in your mix (this is an all-too-easy temptation).

So the first stages are to get the best balance of thump and body and air and presence and clarity and so on that you can get, using mic placement, mic selection, signal path, eq and dynamics, and to get the cleanest, most dramatic and flattering representation you can, dry, and THEN to bring in the big gun of reverb.

Like makeup, reverb can smooth over a lot of flaws and cover up a lot of imperfections, but it's no substitute for a beautiful face. And in the world of music, where everyone can date the prettiest girl for the same price as every other, being heavily made-up is not going to win any dates. Everyone can buy the best CD ever made (whatever that may be), so being halfway there and covering up with makeup still lands you with nobody asking you to the prom.

So... before we go any further, get right with the concepts and practices of contrast based on frequency and dynamics.
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Old 04-04-2009, 09:20 PM   #498
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PS-- as an aside, nicholas' Reamix book deals specifically with a lot of acoustic-only, bassless, drumless mixing. It's also a great primer on using some of Reaper's advanced features. Some of the approaches are a little different than what I've been talking about in this thread, but more points of view are a good thing.
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Old 04-04-2009, 09:21 PM   #499
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...it's like a minor smiley in the middle of the broader inverted smiley.

I would think this is one way they're able to get the apparent loudness and fullness up, by skewing the frequencies to the right away from the mid fundamentals, with all of it supported by a carefully shaped low end...
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Old 04-04-2009, 10:02 PM   #500
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Back to reverb...

"Reverb" specifically is the dense accumulation of reflections that occurs in a space, densely enough to cancel out much of the frequency content of the actual instrument in favor of the frequency of the space. Similar to the way that different guitars or different pianos have their own "sound" that is often more pronounced than the sound of the pure string(s).

But there are a lot of other ways to create "reverb" as a sense of sonic size and space. Short delays are a very obvious example. Delays that are not quite dense enough to disappear into "reverb" per se can still achieve a very similar psychoacoustical effect, often with less loss of clarity and without the specific "localizing" effect of true reverb. We get a similar "embiggening" effect, but without placing the instrument in a specific "place," and without sublimating the instrument "sound" into a wash of room sound. This is especially well-suited to modern-sounding "artificial" records that tend to favor a dryer sound that adapts to the playback environment, as distinguished from more naturalistic recordings that try to contain their own sense of space.

Tied directly to this approach of "delay as reverb" is the increasingly common practice of double-tracking (or triple- or quadruple- or quintuple-tracking and so on), as is the practice of "layering," for example tracking a synth part to mimic the vocal or guitar, or vice-versa. Tracking a thing more than once thickens up the tonal qualities and creates depth and texture similar to reverb. A similar effect can be achieved just by slightly modulating or detuning delays (as with a chorus effect). You could also experiment with panning different short delays and eq'ing them differently.

Of you could just load up a reverb box and flip through presets, and then tweak the best ones to taste. Which brings me to a very important point...
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Old 04-04-2009, 10:18 PM   #501
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A lot of reverb plugins suck.

No, it's not just you. Reverb is the redheaded stepchild of plugins. It gets no love and gets treated like an afterthought, or thought of as something that can be killed by just cramming enough reflections and CPU horsepower in there. They sound splashy and trashy and completely unlike any kind of real-world space that you would ever find yourself in. They might have infinite controls and massive feature lists, but a lot of them still sound like a metal room.

Creating a good reverb takes love and dedication, of the sort that is usually reserved for vintage compressor emulators.

The best across-the-board "free" solution AFAIK is to to use an impulse reverb like SIR or REAverb and then dig through bajillions of impulses looking for a good one. I have some reverbs that I like but I have not found anything close to a short list of "best" reverb plugins. My own personal go-to favorite is the Sonitus reverb, but even that is like maybe 30% of the time, and it's not free. Suggestions welcome. More later.
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Old 04-05-2009, 02:14 PM   #502
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Thanks Yep


I appreciate the response, I hope I haven't caused you to digress from your agenda too much.

I've been experimenting for some time with verb and delay ( mostly freebies ). I seem to have a hard time achieving what I am looking for with the free stuff, but I keep trying. One day I hope to try one of the nicer ( and more expensive ) verbs, like Altiverb.

Your tips and advice, as always, are very helpful.


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Old 04-05-2009, 02:24 PM   #503
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The best across-the-board "free" solution AFAIK is to to use an impulse reverb like SIR or REAverb and then dig through bajillions of impulses looking for a good one. I have some reverbs that I like but I have not found anything close to a short list of "best" reverb plugins. My own personal go-to favorite is the Sonitus reverb, but even that is like maybe 30% of the time, and it's not free. Suggestions welcome. More later.
Pipelineaudio's impulses from the resources page sound good to me and there is a great variety of them:

https://stash.reaper.fm/tag/Reverb-Impulses

(I should add that I don't have any expensive hardware or software verbs to compare with. Oh except where I hear them on CDs I guess )
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Old 04-05-2009, 02:47 PM   #504
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We get a similar "embiggening" effect, but without placing the instrument in a specific "place," and without sublimating the instrument "sound" into a wash of room sound. This is especially well-suited to modern-sounding "artificial" records that tend to favor a dryer sound that adapts to the playback environment, as distinguished from more naturalistic recordings that try to contain their own sense of space.
Hi yep. I'm hoping you might expand on that, esp the part in bold.

I'm not sure just what I'm looking for you to add to that... but there's something there... maybe the mechanism or process of the adaption to the pb environment, if that makes any sense... that I wouldn't mind having a better sense of.

Thanks!
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Old 04-05-2009, 03:13 PM   #505
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Bootsy's epicVerb is pretty good in my experience.
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Old 04-05-2009, 05:17 PM   #506
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My favorite Free ones are....

GlaceVerb (Mix-Final Master, Vocal Settings on Acoustic Instruments)
http://www.dasample.com/index.php?show=glaceverb


Antress Modern Spacer (Overall Massive Drum Hall)
http://antress.webng.com/product.html


Room Machine 844 (Vocals, Back Ground Vocals, Snare, Acoustic)
http://www.silverspike.com/?Products:RoomMachine_844


Anwida DX Reverb Lite (Vocals, Overall Mix, Acoustic & Electric Guitars, Percussion)
http://www.anwida.com/product.asp?pid=7


Kjaerhus Classic Reverb (A little bit of everything)
http://www.kjaerhusaudio.com/classic-series.php

These all sounded ok, but never "felt" right....

Even when using SIR or REAverb with all the recommended Impulses.....tho they were well ahead of the first ones mentioned, they still didn't "feel" right a lot of the time.

Then I got the Wizoo plugin, and found out that, with reverb you do get what you pay for. This plug in surrounds the vocals of mix, and even with my tin ear that I have MAJOR problems with when it comes to reverb's, I could tell when the Wizzo one was missing, even when you could not hear it working.....it is hard to explain but a joy to hear!

I still use the free ones because I like the sound of them on certain things, but the important tracks in a song, like the Vocals, gets Wizoo'd!
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Old 04-05-2009, 06:42 PM   #507
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Pipelineaudio's impulses from the resources page sound good to me and there is a great variety of them:

https://stash.reaper.fm/tag/Reverb-Impulses

(I should add that I don't have any expensive hardware or software verbs to compare with. Oh except where I hear them on CDs I guess )
There are LOTS of great impulses for impulse reverbs, and my point was not AT ALL to say that you can't get good reverb "in the box." Only that there are a lot fo bad reverb plugins out there, much more so than bad EQs for instance.
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Old 04-05-2009, 07:14 PM   #508
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Hi yep. I'm hoping you might expand on that, esp the part in bold.

I'm not sure just what I'm looking for you to add to that... but there's something there... maybe the mechanism or process of the adaption to the pb environment, if that makes any sense... that I wouldn't mind having a better sense of.

Thanks!
Here's the thing... the listening space where you listen to music has its own acoustical properties, even if you listen on headphones.

If you're in a bar, or in your living room, or in a shopping mall, or in a car, then a theoretically perfect "dry" reproduction of the band, based on close-miked sources mixed and panned, would play back through the speakers and it would literally sound like the band was playing IN THAT SPACE. There is not necessarily any need to embed ambient information in the tracks, because the listener is going to hear the space that they're in. One could make an argument that this is actually a more authentic and ideally "pure" approach to take, to simply capture and reproduce the content and leave the environment up to the listener.

Another argument might be made that the best, purest, and most authentic way to create a recording is to either capture or create (through mixing and processing) the ideal "third row center" listener experience, and then count on the listener to make sure that they are listening in a good space and on a good system. This is something like the approach taken in movie mixing, where the presumption is that the listener will be hearing calibrated, tuned speaker systems playing back at a reference level of 83dB SPL in a room with high ceilings and soft, dark walls, and so on. In this case, we want to make sure that *everything* the listener is supposed to hear is embedded in the track. And we expect them NOT to hear anything else.

You could make an argument either way. That said, I've never been much of a fan of arguments, personally, and I don't really care too much about the "right" way of doing things, assuming there is such a thing.

It's safe to say that prior to the rise of the cassette tape, records were mostly made to sound "best" in ideal circumstances. I.e. it was the engineer's job to capture that perfect "third row center" experience, and the listener's job to have a good playback system. In more recent times it has become increasingly common to make records that are made with more concessions towards real-world listening, if not making records made for the outright lowest-common-denominator. See loudness race, etc.

Part of this has been a trend towards hyper, hotter, and dryer-sounding records. With this has come an increasing tendency to use delays instead of reverb, and to use ALL ambient effects (delays, reverb, or whatever) less as a way to create a naturalistic sense of spaciousness, and more as a sweetening, embiggening effect. More gated verbs, more ambiance that is tuned to fit "behind" the dry track (e.g. decays timed to end with the note), less sustain, longer predelays, more artificially hard-panned stereo effects, and so on.

At the same time, there is also a sort of resurgence of interest in old school, naturalistic, vintage, and even outright "lo-fi" recording sounds.

And then there are also approaches that specifically try to emulate artifacts from days gone by: hard-gated 80's-style drum reverbs, vintage slapback echo and tape delays, and so on. And there are entirely new possibilities opened up by modern technology.

What is good and what is bad is entirely a subjective call, and is totally up to you.
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Old 04-05-2009, 07:37 PM   #509
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Got this in a PM and I figured I'd answer here, since for every person who asks, there are usually a dozen others wondering:

Quote:
Dear Yep,

Thanks for producing such an amazing thread over at Reaper forums. An oasis in the desert of internet information. I particularly enjoy the logical and objective structures.

I didn't want to post this question on "the thread" as it might mess up your order structure. I have limited experience OTB, instead I use Lavry conversion good monitoring etc. and all DAW. I was wondering your views on EQ and mix bus.

In your experience have you seen any benefit to mixing on an actual mixing desk such as Soundcraft, Allen Heath etc. in terms of sound quality over using DAW summing, faders and EQ. My curiosity arose since DAW reverbs as so terrible compared to Impulses, I am wondering if EQ plugins suffer similar quality issues compared to real analog mixers?
Lynn Fuston did a big and systematic project on summing busses a while back, to try and figure out whether analog vs digital summing really made a difference. You can google "awesome DAWsum" to check his tests for yourself. The short answer was no. People could generally not tell the difference between different digital summing busses, and while some people could pick out the difference between analog and digital summing, there was no clear consensus that one or the other was "better," and differences were generally pretty small, subtle, and hard to detect, even among a room full of audio engineers listening on forensic reproduction equipment.

That said, "summing" is only one tiny part of record-making, and blind listening tests might not be 100% indicative of the way that real people work in the real world of music-making. Does having the real-world tactile control of genuine zero-latency, true analog knobs and faders help a recordist to make better decisions, compared with mouse-based or stepped digital controls and computer latency? I don't know how anyone could possibly test such a thing in a scientifically conclusive manner.

And as for whether "analog" eq sounds better than digital eq... each example is its own thing. I don't know what the best-sounding eq in the world is, and I certainly don't know whether, for example, the top third are mostly digital or mostly analog. I do know that there are an awful lot of really bad analog EQs out there on the cheap end of the spectrum, and an awful lot of really good digital ones on the free end of the spectrum. On the expensive end of the spectrum, one should expect great sound as a matter of course. And most of the name processors deliver, whether analog or digital.

Don't know if that helps, but that's what I got.
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Old 04-05-2009, 07:47 PM   #510
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PS-- anybody should feel free to post anything they like. This is not is not my forum and was never meant to be a "yep tells people how to record in sequential order" thread!
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Old 04-06-2009, 03:44 AM   #511
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and was never meant to be a "yep tells people how to record in sequential order" thread!


Lol, I got a kick outa that
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Old 04-06-2009, 01:53 PM   #512
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New Yep file is up!

http://www.filesavr.com/01yepthreads...-6-09thread510

Enjoy!
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Old 04-06-2009, 11:31 PM   #513
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An oasis in the desert of internet information?

There's more information for free on the internet than you could poke a forest full of sticks at.
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Old 04-07-2009, 01:10 AM   #514
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For my understanding I will summarize what I learned for bass and treble for the poor man's one room studio (and please correct me if I'm wrong):

Roll of bass and treble on most tracks to get rid of rumble and hiss as much as possible for that particular track. Cut more than the unexperienced would think. That process is called cleaning and should be done first in mixing.

Keep the bass on just the necessary tracks, typically BassDrum and Bass. Shape these tracks differently with EQ to seperate them spatially and achieve a clean, not boomy fundament.

Keep the air (above 10 or 12k) on tracks only which can contribute hiss-free and non harsh treble. That could be hi-quality samples of Cymbals or HiHat, Triangle, Chimes and so on. If there is NO such airy-track it is better to accept that fact. Excellent mixes do exist without too much going on beyond 12 kHz.


----------------

Could this be a way to go - or would it sound amateurish?

After cleaning try to "generate" some air - if necessary - from hiss-free and non-harsh treble (7 to 10 kHz) by means of an exciter (the kind which actually adds freqs one octave above).
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Old 04-07-2009, 09:05 AM   #515
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don't forget the magic cut at 300-350hz. I have a feeling that this does more for overall clarity than most think.
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Old 04-07-2009, 09:31 AM   #516
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Yeah, 300-500ish is where all of the "rar rar!" mids hang out. They sound good soloed on a track, but they drown absolutely everything out when they start piling up.
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Old 04-07-2009, 10:59 AM   #517
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Yeah, 300-500ish is where all of the "rar rar!" mids hang out. They sound good soloed on a track, but they drown absolutely everything out when they start piling up.
In the master EQ? For certain tracks?
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Old 04-07-2009, 02:40 PM   #518
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Always record/print at 24bit. No reason against, all the reasons for...
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Old 04-07-2009, 03:12 PM   #519
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In the master EQ? For certain tracks?
Depends on the project, really. A 3db cut at 400 or so on the master will calm things down, but if you have any tracks with way too much in that range (I'm looking at you here, metal guitars) you'll want to try a deeper cut on just those as well.
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Old 04-07-2009, 03:17 PM   #520
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Really enjoying this thread.
I get a kick looking at those Spectrum pixs of different bands.
Did you see the big boost at about 125hz of the Beatles tune?
Bass boost for Paul maybe?

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