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Gear / Technical Help => Ask The Tapers => Topic started by: jeromejello on December 30, 2008, 11:08:58 PM
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i have been trying to figure out what is the best way to get into the zone with my 24 bit jump.
when i was using the ua-5 >jb3/ihp120 i would throtle the gain and keep it peaking...
with my 722, i was running conservatively at first, and had to add a lot of gain in post (+9dB to +12dB), but lately have been trying to dance with the reds and only adding +3dB on average.
is there an advantage to running hot? is it better to have the pre's do the work or is it better to add gain in post? I have always been under the impression that although you dont want to clip, hotter was better and gain added in post was some how inferior because it was processed.
what are your thoughts?
(fwiw, with 24b i am either running mics>722 or mics>mp2>r-09... and i am using sound forge 7.0 on a xp sp3 box with AMD Athlon 3100 (2.2GHz) & 512 MB ram [although i though i was running over a gig... hmm])
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I run my R4 to max peak @ -3dB(aprx) and average peak around -6dB(aprx) I don't me exact because every show is a little different and the R4 does not have the greatest meters. I do a lot of matrix 2 x mics and 2 x SDB recordings so I try to follow this and when I render the tracks I peak at -1.3dB. I think this works good for matrix and 2 track and I am not a huge fan of normalize so I can almost always get away w/o doing so for the 16 bit dithering.
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no need to run it hot.
so adding gain in post doesnt introduce additional noise? it seems like it would. ???
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Only if you are recording something with really, really quiet parts where the noise floor of the venue is less than that of your rig without the gain turned up so much. I often set the R-09 to peak around -12db for the start and invariably eat up a bit more headroom with peaks around -6db by the encore for something loud or from enthusiastic clapping for acoustic events (where I often need more dynamic range). Even for those classical type super dynamic things with very low level passages, the venue HVAC noise is louder than my mic self-noise so boosting later is no problem.
24bit lets you back off a bit, relax and enjoy the music.
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Running mic-in to a SD722 I always run cool. I like to see the first red hitting occasionally. If I see the second red hitting a lot I dial it back a little.
I think most people that run 722s mic in will agree this seems to be the best way to run these boxes. Adding 7-9 or even more dB in post to bring my recordings up to ~-1dB final is typical.
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I agree with Teddy 100%:
http://taperssection.com/index.php/topic,69144.msg929333.html#msg929333
My recordings started sounding better & I stopped having to worry about clipping when I stopped running hot. I even found I could record 16 bit as low as -20 dB and not hear any noise when converted to 32 bit by Adobe Audition for boosting.
I doubt anyone here will agree with me as to 16 bit, but for 24 bit recording it's a no brainer.
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I have always been under the impression that although you dont want to clip, hotter was better and gain added in post was some how inferior because it was processed.
at 16 bit, hotter was better because you only had 16 bits to play with. but at 24 bit, even if you peak at -12, you're still getting better resolution than 16 bit
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I run hot. I keep 'er around -6db on the R4. My thoughts are you
shouldn't have to crank your playback to hear the recording.
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kcmule's comment idicates that he wants to be able to use his actual masters for playback. Otherwise just boost in post. I'll give up sometimes not being able to listen to my recording on the drive back from the venue for the benefits of 24 bit recording peaking at -12 to -20.
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I agree with Teddy 100%:
http://taperssection.com/index.php/topic,69144.msg929333.html#msg929333
My recordings started sounding better & I stopped having to worry about clipping when I stopped running hot. I even found I could record 16 bit as low as -20 dB and not hear any noise when converted to 32 bit by Adobe Audition for boosting.
I doubt anyone here will agree with me as to 16 bit, but for 24 bit recording it's a no brainier.
Right on.
Teddy's post is one the best readings ever in TS.
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kcmule's comment idicates that he wants to be able to use his actual masters for playback. Otherwise just boost in post.
I do like to hear masters at decent levels. I do boost in post, but not a bunch.
Perhaps I'll try a few shows at lower levels and compare for myself.
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When I was running a 702 I learned that it does not like to hit 0 at all.. unlike the V3 I ran prior to the 702.. I would just set levels to peak at -3 or 4 and all was good..
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Running mic-in to a SD722 I always run cool. I like to see the first red hitting occasionally. If I see the second red hitting a lot I dial it back a little.
I think most people that run 722s mic in will agree this seems to be the best way to run these boxes. Adding 7-9 or even more dB in post to bring my recordings up to ~-1dB final is typical.
That's how I've been running the 702...
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Recording gain is like salt in a stew: you can always add more, but it is the Devil's own job to get it out. 8)
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What hasn't really been stated fully here is WHY running cooler in 24bit is OK versus 16bit. Remember back in computer school about how additional bits gives more information? Well, I don't know the exact numbers, but in 24bit, each sample has an order of magnitude more information recorded than 16bit.
Running hot in 16bit was a little bit more important for rendering the sound information on the media accurately as it was recorded. Since there's less information, there's less lattitude with what you can do with that information in post. In that case, the general concensus was that it was worth the risk of clipping to run hot because alot of people felt that the sound tended to be better on the final product when they didn't have to bump the levels 10db or more in post.
In 24bit, there is an order of magnitude more information. Most everyone agrees that the sound information captured with levels running conservatively and then bumped in post sounds no different than if you run hot to begin with. So, the logic therefore is to run conservatively for the sake of avoiding any potential risk of clipping.
Regarding, the concept of amplifying noise, that is a concern for sure, but as someone has explained, the noise floor for most of the music that we record is such that it will not be heard, especially with modded low-noise components in the sound chain. Bumping the noise floor 10db won't be heard anyway. I suppose if all you record is chamber music, this would something that you might want to check out on your rig, but I have the feeling that the use of high quality, low noise components will keep the noise floor sufficiently low that using the '10db bump in post' strategy would still work out OK.
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Let me ask this:
What is considered "running hot" for 24 bit?
Trying to keep near zero?
Like I say, I keep 'er around -6db which should leave
plenty of room for an any additional bumps in post.
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Let me ask this:
What is considered "running hot" for 24 bit?
Trying to keep near zero?
Like I say, I keep 'er around -6db which should leave
plenty of room for an any additional bumps in post.
That's what I used to do before reading Teddy's post.
Right now, I keep my levels around -12db. Love the results.
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That's what I used to do before reading Teddy's post.
Right now, I keep my levels around -12db. Love the results.
How much do you add in post? 10 to 12db ? That seems like a bunch to me.
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That's what I used to do before reading Teddy's post.
Right now, I keep my levels around -12db. Love the results.
How much do you add in post? 10 to 12db ? That seems like a bunch to me.
Why?
I mean, why adding 10db or 12 db sounds too much in post and not too much during the recording process?
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Like RobertNC alluded to, the flavor of a given recorder/preamp changes sometimes depending on how hot you run it. Quite simply, it's not a perfect linear amplifier with pure gain and nothing else.
I generally run so the meters on the Busman T-mod R4 so they are bouncing up around -6 on the average. One time at a show I got up from my seat and accidentally bumped the left gain knob so I was running really hot, not really clipping badly, but clipping occasionally. At any rate, it ran like that for a few songs, and when I tried to balance it out in post I was really shocked at how different it sounded. When it's at -6 or less, it's sounds nice and "airy", but run really hot it sounds bland, perhaps compressed, more in-your-face. It's listenable, but not pristine... almost like I patched that channel with an inferior source for a few minutes. So that's my profound scientific assessment based on 1 data point. I run the R4 down around -6db max.
With the V3 I don't find this to be quite so much the case. I've had cases where I started off conservatively, and was peaking -12db the first song, then I brought it up the gain 10db between songs. When I boost that first song 10db in post to match the rest of the set I don't find that it changes the flavor with respect to other songs I recorded hot.
So, if Robert is telling you tribal knowledge among the SD7xx owners is "give it room to breath", then he is probably on to something. My suggestion is to play with it during an opening act you don't care about... set the gain conservatively for half the set, then crank up the gain so you are almost clipping for the second half. Level it off in post, listen to it, and make your own decisions.
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I like 24 bit, cause you can be lazy and not worry about it. I need to do more testing, but I am pretty sure I prefer my stock UA-5 at lower gain as opposed to HAWT HAWT.
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ok i may be showing my ignorance here but who cares. all things being equal(mics, pre & recorder), if taper A peaks at -12 and taper B peaks at say -4, hasn't taper B made a "dynamically" superior recording?
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He has made a recording 8dB louder. The dynamics are the same. But he also ran the chance of clipping. Clipping just plain sounds awful.
Had he recorded at the lower gain rate he could have made it up in post.
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He has made a recording 8dB louder. The dynamics are the same. But he also ran the chance of clipping. Clipping just plain sounds awful.
Had he recorded at the lower gain rate he could have made it up in post.
BINGO! I typically add around 6-10db in post, even adding upwards to +20db in post, and the noise floor is so low in 24bit, that it sounds just as good as any other set I recorded that weekend ;)
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ok i may be showing my ignorance here but who cares. all things being equal(mics, pre & recorder), if taper A peaks at -12 and taper B peaks at say -4, hasn't taper B made a "dynamically" superior recording?
Taper B is also boosting the effective noise-floor of his recording 8db higher along with all the desired signal. So the resulting recording has the same dynamic range as Taper A's recording that left extra headroom up top. The essential line in the excellent post of Teddy's that fmaderjr linked to is this-
3) As long as the noise floor in any recording system is lower than the noise floor in the signal you're recording, you will record the full dynamic range perfectly.
Now if taper B's gear was really crappy and had a noise floor higher than the noise of the room he was recording in then he would be making a "dynamically superior" recording by running hotter and pushing the noise imposed by his gear down. But note that it would be pointless for him to be recording a 24bit file in that case since a huge part of that would be nothing but noise. It's likely he wouldn't even be using all 16bits.
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One way to think about it is that the total dynamic range of the sound being recorded is almost always smaller than the total dynamic capability of our gear. That range runs between noise at the bottom of the scale and clipping at the top. You adjust where the range of sound you are recording fits into the range of capabilities of your recording system and either leave more room at the bottom or the top. So the question then becomes, "where is the comfortable middle?"
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Like RobertNC alluded to, the flavor of a given recorder/preamp changes sometimes depending on how hot you run it. Quite simply, it's not a perfect linear amplifier with pure gain and nothing else.
Exactly. There is more to the overall equation than just the concept of headroom and dynamic range and more information in 24bit realm versus 16 bit realm.
I don't understand the electronics of it all, but, what and where you are recording, the mics, the pre-amps, the amplifier stage, the A/D stage are all part of the final signal you get, and the final signal you get will be different under different overall operating conditions.
When you turn up the knob on that box you are doing a lot more than just "adding dB in real time". You are changing the operating conditions of the analog stages in the box, and that is changing how the unit performs.
It's a complicated combination of preferences and conditions.
24 versus 16 bit aside, different recorders with different analog stage designs behave differently. So may sound better "hotter" some may sound better "cooler". Experiment. Personally I think the 7xx series mic-in is a "cool" box that sounds better run at low gains.
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Thanks to everyone for a great discussion. I have always tried to run as hot as possible even though I record in 24-bit...I guess its just an old habit. I had previously thought that I was fine as long as I wasn't clipping, and I had never considered that there may be something else going on to impact sound quality. Since I'm also a bit of a photographer nerd, I had previously equated this ("running hot") to the commonly recommended practice of siding toward slight overexposure on digital sensors rather than underexposure -- this is frequently discussed in photo forums as a way to reduce noise and possibly "extend" dynamic range.
I may have to try a few gain experiments at the next show I tape. This has been a thought provoking discussion. Thanks!
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dmonkey (and RobertNC), if my preamp behaved in the manner just described:
> When you turn up the knob on that box you are doing a lot more than just "adding dB in real time". You are changing the operating conditions of the analog stages in the box, and that is changing how the unit performs. ... It's a complicated combination of preferences and conditions.
... I would consider it a bad choice of preamp, if not defective. However, there are many audio engineers who wouldn't necessarily feel that way. There's a deep difference of attitude about this kind of thing, and that's what I'd like to alert you to. If you're not aware of it, you'll probably have a hard time making sense of the conflicting advice that people offer you. Let me try to outline the two viewpoints as fairly as I can--though I'm on one side and not the other, so I may not do full justice to the side that I don't agree with.
One side believes fundamentally that our ears are more sensitive than the best audio measuring equipment, and that there's no such thing as a sonically neutral audio component. According to them, every audio component (even a microphone cable or a line-level "interconnect") has its own "sonic signature," and when you choose microphones, preamps, A/D converters, recorders and even microphone cables and "interconnects," you're like a chef who blends ingredients so as to create a specific flavor experience.
Often the people in this group say that they consider microphones, preamps, mixing consoles, etc. to be like musical instruments. They consider their own work to be a form of direct participation in an artistic event.
The other side says that it is possible to have sonically neutral audio components in at least some cases--or to come so close that certain items of equipment effectively "drop out of the equation" as variables. If you're careful, these people believe, you can find neutral-sounding preamps, cables and digital recording devices. Even if they're not perfectly neutral sounding, they can be close enough that the remaining variables (such as room acoustics, or microphones and their placement) overwhelm them by orders of magnitude.
In general, the second group of people prefers preamps and other electronic components that they consider close to this ideal. They consider a "flavored" preamp or converter to be like the sum of a neutral preamp or converter plus a "flavoring" component that ought to be optional. In general, these people want to let the musicians be the artists; they're just trying to record what the musicians are doing.
Please note that the second group of people doesn't say that "all preamps sound alike," for example; they only say that a preamp should (and can, if care is taken in its design and use) deliver a signal that is essentially just an amplified copy of whatever you feed into it. There is always some minimum amount of noise that the laws of physics require, but apart from that, the output ought to be sonically indistinguishable in character from the input.
The thought behind this viewpoint is that there's no such thing as a "universal sonic improver"--no tweak that you can do to any audio signal, that will always make it sound better no matter what that signal was like in the first place. For every such tweak that may be built into component X (say, a mild boost in the low-mid frequency region for "warmth" and a gently increasing amount of low-order harmonic distortion to simulate "vintage tube sound"), there will be some recording that already has too much of the same thing, where any further addition will only make the result sound wrong. The second group of people prefers to record as "straight" as possible, and if there's an improvement to be made by boosting this or shifting that or reducing some other thing, you make it after the live recording is safely in the can.
There are other important differences in viewpoint between these two groups, but the more I go on about this, the more I risk stereotyping people (no pun unintended). And there's already way too much of that. Each person has his own reasons for his own opinions, but a surprising amount of this difference is over which beliefs are opinions vs. which are proven facts. That leads sometimes to the type of discussion in which people talk past each other and secretly--or sometimes not so secretly--think that each other's point of view is foolish. It's not pleasant to be anywhere near that kind of situation.
Anyway, my answer to the original question in this thread would resemble some that were already posted. 16 bits gives you a huge dynamic range to begin with, and even though no real-world recording ever has 24 full bits of resolution, the available 20 or 21 bits gives you so much that you can really afford to relax about levels. Just get the peaks somewhere into the top, I dunno, maybe 10 dB below full scale, and then you can normalize and dither down to 16 bits at your leisure when you get the recording home.
As long as your other components are well chosen and properly connected, and your gain settings make sense, with 24-bit recording there's no sonic penalty for having moderate rather than "maximum possible" peak levels. It even makes some sense to aim for them on purpose. If you're not sure what the peak sound levels will be, you can afford to set everything 6 dB too low, if that's how it should turn out. Doing so should greatly reduce the number of times that something accidentally lights your "OVER" light.
--best regards
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Apologies for going off topic, but something above caught me.
The second group of people prefers to record as "straight" as possible, and if there's an improvement to be made by boosting this or shifting that or reducing some other thing, you make it after the live recording is safely in the can.
I sometimes feel like I'm standing with a foot in both camps and that line above really sums up my philosophy. For me an absolutely essential element is capturing an uncolored 'straight' recording, yet I'm not adverse to whatever I can do to make any of my recordings sound great, period. I know and respect a lot of people who feel their raw recordings are some sort of documentary and should not be touched at all after the event, but for me that's where the potential for subjective, manipulative improvement comes to play. Personally I find the 'post' stage much more difficult and subjective, with great potential to mess things up, whereas the goals of the initial capture are somehow more clearly defined with my sonically neutral hat on. I can't do much with a sow's ear and don't need to do much with the real gems, but I'm slowly getting better at transforming a decent, neutrally recorded performance into something that can really shine.
That may say more about what I'm comfortable with and where I need to invest more personal educational effort than anything. Or that I trust my recording tools more than my playback, monitoring and manipulative ones. Seems there are a lot more uncertain variables in the 'recording as instrument and color' camp wherever it comes in the production phase.
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I'm not really in one camp or the other. I'm a hobbyist not an audio engineer. It is just my personal experience that with my specific gear as I run hotter the final sound has a harsher quality to it that I don't like. If I run cooler it sounds more natural.
For all I know it is my imagination. Most people tell me I should not even be running a 722 without a Grace in front of it, but from a strictly empirical point of view, I run cool, and like the results.
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I'm not really in one camp or the other. I'm a hobbyist not an audio engineer. It is just my personal experience that with my specific gear as I run hotter the final sound has a harsher quality to it that I don't like. If I run cooler it sounds more natural.
For all I know it is my imagination. Most people tell me I should not even be running a 722 without a Grace in front of it, but from a strictly empirical point of view, I run cool, and like the results.
I know what you mean. Same thing happens to me.
It's not your imagination. There's a very solid explanation for that.
http://taperssection.com/index.php/topic,69144.msg929333.html#msg929333
Take care.
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Um, no--sorry to say, the whole three-paragraph-long core of that message, about supposed distortion above "0 dBVU" levels, is completely unfounded. Also, some of the numbers he gives seem to be just made-up numbers. The "functional 100 db of S/N" as the limit for any digital recorder might have been true in the early 1980s, but A/D converters have been available for 10 years now with an honest dynamic range 15 - 20 dB better than that.
Look, any particular piece of gear can have problems, but most digital recording equipment doesn't sound any worse--or any different--in its top 10 dB from the rest of its range. If someone has a 24-bit recorder that really and truly sounds better when you don't use the top 10 dB of its range, I'd say get it looked at by a competent technician, because it's not supposed to behave that way. Or try padding down its outputs and recording at normal levels, because you may be overloading the input stage of the equipment you were listening to it through.
--best regards
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DSatz,
Well, I'm here to learn. That's my only goal BTW. For that,I really thank you. It's just that this piece of writing has been praised for so many members that I took it for granted.
Anyway, thanks for your message. It's always a pleasure to read your posts.
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Dede2002, "0 VU" just doesn't mean what the author of the message seems to think it does; he seems to have guessed, and guessed wrong. (It's not any one voltage level, so you can't really say that signals at or above that level are difficult for circuits to handle; they could even be very small signals, just as long as they're in linear proportion to the voltage of other signals in the same system.)
And as I said earlier, the 100 dB (or thereabouts) "functional" limit on possible signal-to-noise ratios simply doesn't exist--one Prism converter gets around 122 dB as I recall, and Apogees get 120 dB or thereabouts, for example.
But there definitely are points in the message that I agree with, and that I think are very well put. Maybe I'll have a little talk off-line with the person who wrote it, since we all have the right to revise our postings here.
--best regards
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8.2 db below 0 for 16.44
12.5 db below 0 for 24.96
my law
g
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Dede2002, "0 VU" just doesn't mean what the author of the message seems to think it does; he seems to have guessed, and guessed wrong. (It's not any one voltage level, so you can't really say that signals at or above that level are difficult for circuits to handle; they could even be very small signals, just as long as they're in linear proportion to the voltage of other signals in the same system.)
And as I said earlier, the 100 dB (or thereabouts) "functional" limit on possible signal-to-noise ratios simply doesn't exist--one Prism converter gets around 122 dB as I recall, and Apogees get 120 dB or thereabouts, for example.
But there definitely are points in the message that I agree with, and that I think are very well put. Maybe I'll have a little talk off-line with the person who wrote it, since we all have the right to revise our postings here.
--best regards
That would be awesome. One more time, thanks for sharing your knowledge.
Happy 2009.
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i would like to thank everyone for the dialogue... it has been very informative.
both of my main concerns were raised thus far... if running the 722 too hot would some how have a negative effect on the sound quality and if adding gain in post would have a negative effect on the sound quality.
i guess it seems to run more conservatively in 24 bit is the way to go... i dont know if i would be -12 peak conservative, but i dont think i will ride it to 0 either.