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Gear / Technical Help => Ask The Tapers => Topic started by: jdhriver on October 19, 2006, 02:47:43 PM
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I have generally taken the road of leaving my recordings alone except for the tracking. Last night in messing around with Soundforge I ran the volume up on a recent recording...What are the pros and cons of doing this and what should I watch out for. I generally run my recordings fairly low cause I suck with the meters on the JB3 and need a SVU and would rather have a quiet than clipped out recording......cause there is always the volume knob or volume in post......
Thanks for the help. PEACE
JD
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back when I ran dat, for most all my transfers on the standalone, I bumped up the levels, because with dat, I always ran the levels low in analog recording to avoid clipping.
With the v3, I run the levels hot and usually dont need to boost levels in SF, but sometimes I do bc, Id rather change levels to the whol;e set/show rather than doing it during the show.
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back when I ran dat, for most all my transfers on the standalone, I bumped up the levels, because with dat, I always ran the levels low in analog recording to avoid clipping.
With the v3, I run the levels hot and usually dont need to boost levels in SF, but sometimes I do bc, Id rather change levels to the whol;e set/show rather than doing it during the show.
funny, i was just thinking/doing this the other night. i ran the V3 but and went to track out the show cdwave. the levels seemed a little low. so, in wavlab i added 2db of gain. then when i pulled up the file in cdwave it looked about right. i had a total of 6 clipped sections according to cdwave, which i feel is okay.
i have heard it is best to change gain as opposed to normalize. i'm not sure if this is 100% true but do it b/c it doesn't seem to change much about the recording except boost lower levels. YMMV
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Off the top of my head, I'd say the big disadvantage of boosting volume post is that you boost the entire spectrum (though, I'm sure you could tweak this with some programs). By boosting the entire spectrum, you are increasing your floor noise:
You rig has a floor noise of -60dB, and you record with a max. of -16dB. Later you boost the -16dB to 0.00dB, that also brings your floor noise up to -44dB... Not a big deal on the loud sections, but on the quiet sections, you may hear a little more hiss than usual...
Of course, the main adavantage is that its louder... But louder doesn't mean better sound...
Terry
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AS SURE AS YOU SAVE THE MASTERS...GO NUTS >:D
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AS SURE AS YOU SAVE THE MASTERS...GO NUTS >:D
A friend on this board gave me a good suggestion regarding the volume. Instructions are for SoundForge
First do a scan in the normalize oiption for the peak level. I'll usually get somewhere between -6 and -4. I do a seperate scan on each channel. Write down the values and then go into the volume option and increase the negative values by the exact positive amount to get both channels peak level to 0.00 dB So if I have a -4 on the right channel I will add +4 to it to bring it to 0
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I recorded an acoustic tweener set at -18 db this summer. I tried normalizing but scrapped it after a few seconds of listening. The ambient information seemed completely out of whack. I didn't like it at all.
Chris
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Pros:
Makes it louder.
Cons:
Doesn't increase dynamic range.
Adds requantization errors. Requantization noise can be reduced via another round of dithering but it is still a problem.
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Thanks.
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AS SURE AS YOU SAVE THE MASTERS...GO NUTS >:D
A friend on this board gave me a good suggestion regarding the volume. Instructions are for SoundForge
First do a scan in the normalize oiption for the peak level. I'll usually get somewhere between -6 and -4. I do a seperate scan on each channel. Write down the values and then go into the volume option and increase the negative values by the exact positive amount to get both channels peak level to 0.00 dB So if I have a -4 on the right channel I will add +4 to it to bring it to 0
isn't this exactly what normalizing does? you can normalize each channel individually, which is nice if you didn't run your levels even, but i have found a clap or other noise in one channel will end up screwing up the balance of the levels when you normalize each one individually.
i think the main concern is what terry addressed with bumping up the noise floor, which is why you want to run as hot as possible without clipping when doing 16bit. i normalize all my 24bit recordings prior to dither/resample since I run my levels much more conservatively, I have yet to discover any cons of doing so.
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Proper gain changes are non-destructive and can be reversed.
I rarely mess with the channel balance unless I was setup non-dfc.. And even then, I produce a number of different balance tweaked versions and evaluate them before messing with it. I generally evaluate the RMS and peak levels of various sections to arrive at the amount of correction.
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this is a great VST plugin that Teddy pointed out before. really great for getting an idea of how well the 2 channels are balanced
http://www.rogernicholsdigital.com/inspector.htm
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you are doing a good thing by running conservatively...keep up the good work.
FYI..check outthis post (http://taperssection.com/index.php?topic=69144.msg929333#msg929333) for some insight on level setting..
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so here's something I've been thinking about. I've been out of the loop for a while now, so if this has been covered, I apologize.
One of the hardest situations I encountered in the 16 bit days was quiet acoustic shows with very loud, enthusiastic crowds (not that that's a bad thing). You'd have the music peaking at -18 or -20 on the DAT deck, and the crowd would still clip wretchedly right at the ends (and sometimes beginings) of songs. it's kind of unpleasant to listen to tapes like that.
so with 24 bit, obviously you get the added headroom, and can get the full range of the music and <most> of the crowd. but difference in levels is still quite unpleasant, and I've been thinking about what the best method of evening them out would be. I assume a normal compression pluggin would have an advers effect on the music as well. something like a "soft limit" that would only effect those top 7 or 8 samples, though I imagine that may effect the music as well. ideas? how do people engage this these days?
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difference in level, or dynamic range,,,is a good thing..but if you want it so you dont have to use your volume knob :P use something like voxengo elephant or the waves L2 to even things out. Personally, I leave the dynamics intact..
so here's something I've been thinking about. I've been out of the loop for a while now, so if this has been covered, I apologize.
One of the hardest situations I encountered in the 16 bit days was quiet acoustic shows with very loud, enthusiastic crowds (not that that's a bad thing). You'd have the music peaking at -18 or -20 on the DAT deck, and the crowd would still clip wretchedly right at the ends (and sometimes beginings) of songs. it's kind of unpleasant to listen to tapes like that.
so with 24 bit, obviously you get the added headroom, and can get the full range of the music and <most> of the crowd. but difference in levels is still quite unpleasant, and I've been thinking about what the best method of evening them out would be. I assume a normal compression pluggin would have an advers effect on the music as well. something like a "soft limit" that would only effect those top 7 or 8 samples, though I imagine that may effect the music as well. ideas? how do people engage this these days?
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I realize that dynamic range is a good thing, but I have one or two shows where if you set the volume for an audible level on music, you'll wake the neighborhood dogs once the crows starts in. I'm looking for something that'll leave the dynamic range of the music intact, but squish down the crowd.
and my preamp (linn) doesn't have a volume knob, and keeping one hand on the "up" and "down" buttons is a little more active participation than I'm looking for (they don't respond fast enough anyway)
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I do a seperate scan on each channel. Write down the values and then go into the volume option and increase the negative values by the exact positive amount to get both channels peak level to 0.00 dB So if I have a -4 on the right channel I will add +4 to it to bring it to 0
I think it would be better to apply the same gain to each track unless you are trying to correct for an imbalance in the recording. If you have a well balanced recording and one channel peaks higher than the other (maybe you have a clapper wide in one channel) then the resulting image will be shifted if you apply different gain to each channel.
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but difference in levels is still quite unpleasant, and I've been thinking about what the best method of evening them out would be. I assume a normal compression pluggin would have an advers effect on the music as well. something like a "soft limit" that would only effect those top 7 or 8 samples, though I imagine that may effect the music as well. ideas? how do people engage this these days?
I've used compression on a handful of recordings I've made in situations similar to the one you mention. Personally, I don't especially care about capturing the dynamic range of a rabid crowd. Two ways to do it: complete file at once, or individual applause/crowd sections. For complete file at once, you won't impact the music if you set your threshold / compression ratio appropriately. That typically means you'll have a fairly high threshold with an aggressive ratio. I've had better success isolating each crowd outburst - I'm able to set a lower threshold with a softer ratio and still not impact the music because the music will typically drop down below whatever threshold I use before the uproarious crowd chimes in, and vice versa before the next song. The latter takes a bit more effort, but I find the effort worth it on the rare occasions I need to do so.
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agree with Bri, who cares if the clapping is clipped, set your levels based on the music.
I also edit my shows where the clapping is loud by applying soft limit compression (sound forge: effects>dynamics) to the individual sections of crowd noise (I typically squash audience peaks to ~3-5 dB or so below where the music peaks. You can play around with this to taste), then boosting the whole file using waves L2. Works pretty good for jazz and acoustic shows. Then your ears are not blown out by the crowd on playback when you had the volume cranked to hear some quiet tune. :)
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I usually normalize, and often normalize on a track-to-track basis (with max gain limits set for very quiet parts).
I do this all non-destructively (Wavelab "montage" file) and keep the original (16 bit) wave file.
I give out a lot of CDs and I've had listeners tell me it is too quiet...
I have not tried compression, but this sounds like a great idea to remove loud crowd noise, particularly when you are right in the crowd (ie., stealth recordings).
Richard
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agree with Bri, who cares if the clapping is clipped, set your levels based on the music.
I should clarify. While I don't care about reducing dynamics by compressing an uproarious audience between songs, I do care about clipping the audience. Two reasons:
<1> While clapping itself is often transient enough that it doesn't clip in nasty and crunchy fashion, audience hoots and hollers definitely have the potential to clip in nasty and crunchy fashion. Once the audience hoots and hollers clip, there's no going back - and IME there's typically no good way to recover. Sure, some clip restoration tools help, but in cases like these I find they don't remove the nasty crunchiness and only minimize the audible effects. Even during audience sections, I find crunchy clipping unpleasant.
<2> IME, even fast transients that clip may pose problems during editing. For example, compressing a clipped audience section sometimes produces a distorted result. I haven't found this happens every time, but often enough that I try not to clip the audience at all.
Of course, I ran into those two issues after running my share of 16-bit recordings with levels set for the music, damn the audience clipping. I eventually found the results too inconsistent, given the above, so I started setting levels for the audience and adjusted in post. While I didn't capture the music at as high resolution as I might have, had I set levels for the music (and clipped the audience), I at least didn't run the risk of having nasty artifacts scattered throughout my recordings (if only during the audience sections).
Recording music in situations where the audience produces higher SPLs than the music certainly poses its challenges at 16-bit. Fortunately, it's not really an issue at 24-bit.
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Yeah, just set your threshold just a tiny bit above (or maybe below) the music peaks, and then use a high ratio on your compressor, and apply make-up gain to get the limited peaks back up to near 0 db. This should keep the compressor from touching most of the music but it will kick in bigtime (like a limiter) when the audience levels go above that mark. The audience will still be loud as balls, but at least it won't be quite as bad as before and at least you've left the music's dynamics mostly untouched and normalized at the same time (with the make-up gain).
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Off the top of my head, I'd say the big disadvantage of boosting volume post is that you boost the entire spectrum (though, I'm sure you could tweak this with some programs). By boosting the entire spectrum, you are increasing your floor noise:
You rig has a floor noise of -60dB, and you record with a max. of -16dB. Later you boost the -16dB to 0.00dB, that also brings your floor noise up to -44dB... Not a big deal on the loud sections, but on the quiet sections, you may hear a little more hiss than usual...
Of course, the main adavantage is that its louder... But louder doesn't mean better sound...
Terry
i agree but most of my recordings have crowd chatter and such which should hide most of it, if my logic is right.
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That would be noise floor, Terry. ;D ;D and you are dead on with Louder is not always better (http://taperssection.com/index.php?topic=69144.msg929333#msg929333)
Off the top of my head, I'd say the big disadvantage of boosting volume post is that you boost the entire spectrum (though, I'm sure you could tweak this with some programs). By boosting the entire spectrum, you are increasing your floor noise:
You rig has a floor noise of -60dB, and you record with a max. of -16dB. Later you boost the -16dB to 0.00dB, that also brings your floor noise up to -44dB... Not a big deal on the loud sections, but on the quiet sections, you may hear a little more hiss than usual...
Of course, the main adavantage is that its louder... But louder doesn't mean better sound...
Terry
i agree but most of my recordings have crowd chatter and such which should hide most of it, if my logic is right.
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you are doing a good thing by running conservatively...keep up the good work.
FYI..check outthis post (http://taperssection.com/index.php?topic=69144.msg929333#msg929333) for some insight on level setting..
I missed that post first time round. +t, very interesting.
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I usually hard limlit the crowd noise to below -3dB relative to the music. Never had distortion due to doing that. If the audience is clipped slightly I just apply a "clip restoration" plugin before the limiting. And I always set levels for the music.
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waves L2 is my friend. i tend to run a bit low then raise it withL@ later if needed
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waves L2 is my friend. i tend to run a bit low then raise it withL@ later if needed
Only with Waves L2, you're doing more than raising levels - you're also compressing dynamics. A big deal to some people, others not. As long as you're happy, that's what counts! And yeah, L2 sounds pretty darn good. Like the included IDR dither, too - my 2nd favorite to MBIT+.
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waves L2 is my friend. i tend to run a bit low then raise it withL@ later if needed
Only with Waves L2, you're doing more than raising levels - you're also compressing dynamics. A big deal to some people, others not. As long as you're happy, that's what counts! And yeah, L2 sounds pretty darn good. Like the included IDR dither, too - my 2nd favorite to MBIT+.
Yeah, waves L2 is slaughtering a lot of commercial recordings these days..the waveforms look like a 2 x 4.
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waves L2 is my friend. i tend to run a bit low then raise it withL@ later if needed
Only with Waves L2, you're doing more than raising levels - you're also compressing dynamics. A big deal to some people, others not. As long as you're happy, that's what counts! And yeah, L2 sounds pretty darn good. Like the included IDR dither, too - my 2nd favorite to MBIT+.
i did not realize it was compressing dynamics. when i have used it the adjustments are so small i have not heard major changes. i may opt for a different way to boost from now on. i don't want to compress any dynamics in my recordings. this place rocks, thanks for the good info guys.
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Yeah, waves L2 is slaughtering a lot of commercial recordings these days..the waveforms look like a 2 x 4.
True, but L2 is very good if you use it with care.
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I use the L3 and love it.
As with many things. Moderation is the key.
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I'm holding out for L11.
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I'm holding out for L11.
I use "volume" and "gain controls" . :P :P :P
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I'm holding out for L11.
I use "volume" and "gain controls" . :P :P :P
Ah, c'mon Teddy. The first step to recovery is to admit you're compressing the hell out of those classical recordings of yours... LOL!