How to get more volume?
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Lt. Bob
Spread the Daf!
I've read some seriously positive reviews on those.
The goal of your mix should be to make is sound good between -10 and -6db. Loudness is can be dealt with later. As long as it sounds good to you, then you should either learn some mastering or send it to someone. If you want me to have a go at it (no $$$ involved) send me a message at Montreal Mastering
Cheers,
Erick
jimmys69
MOODerator
Sorry Mr. Norman, but you need to either reply and explain yourself to any of your last 6 months of posts or PM's or I will have no choice but to make you a site moderator.
Lol!
ashcat_lt
Well-known member
Just to be a prick...
So let's say our goal is to get the loudest peaks in our recording to be as loud as possible without any clipping. You set your brickwall limiter to -0.01dbfs, and find that it's still getting intersample peaks which distort the converters. So you turn down the ceiling to 0.3dbfs and find that sometimes mp3 compression/decompression routines still cause oversea. So you turn down the ceiling a bit further. I usually set my upper limit at -0.63dbfs for no good reason, but it has seemed to work for the last two decades... Anyhow, you settle on a level where you feel you're "safe" against clipping on playback.
But guess what! If you ever send a signal into the limited that is above the ceiling you've set (after whatever input gain is applied) then you're clipping the signal. There's no way around it. Some limiters try to be nicer about it by applying a sort of curved gain reduction as it reaches the limit, and some don't actually place a hard limit - being essentially just really high-ratio compression, but if it really is a brickwall, and it actually stops a peak from going over its ceiling, then it is adding distortion.
Loudness is, in fact, a subject psychoacoustic phenomenon that does have quite a bit to do with frequency content. (Not that I have any clue what Rod was trying to say...)
Loudness in a final product must start where everything starts - writing, arrangement, and performance. If you start from a careful and deliberately controlled dynamic range in the piece, have the performers control their dynamics in the performance, and choose sounds and treatments which support this effectively, then you're almost done before you even started.
Then you should be able to do just minor things in the mix stage - a little automation or compression on individual tracks, subgroups, and/or the master fader and leave almost nothing for the mastering engineer to do at all.
Then the mastering engineer maybe uses some small amount of compression and/or limiting to try to knock down. The truly abberant peaks which may have gotten past the mixer. If you're doing mush more than that at this stage, and you really want more apparent loudness without "squashing" or "clipping", then you need to step back. First decide if the sort of loudness you're looking for is really appropriate for the piece and then go back to mixing stage and see if you can get a bit more there without completely destroying it. Maybe you have to go back and retrack the part with a tweak to the sound and more discipline from the musician. If you find yourself then questioning the arrangement... Then you have to again assess whether you have appropriate expectations. Maybe this song just isn't that loud! Do we change it to be louder, or do we let it sit where it sits?
So let's say our goal is to get the loudest peaks in our recording to be as loud as possible without any clipping. You set your brickwall limiter to -0.01dbfs, and find that it's still getting intersample peaks which distort the converters. So you turn down the ceiling to 0.3dbfs and find that sometimes mp3 compression/decompression routines still cause oversea. So you turn down the ceiling a bit further. I usually set my upper limit at -0.63dbfs for no good reason, but it has seemed to work for the last two decades... Anyhow, you settle on a level where you feel you're "safe" against clipping on playback.
But guess what! If you ever send a signal into the limited that is above the ceiling you've set (after whatever input gain is applied) then you're clipping the signal. There's no way around it. Some limiters try to be nicer about it by applying a sort of curved gain reduction as it reaches the limit, and some don't actually place a hard limit - being essentially just really high-ratio compression, but if it really is a brickwall, and it actually stops a peak from going over its ceiling, then it is adding distortion.
Loudness is, in fact, a subject psychoacoustic phenomenon that does have quite a bit to do with frequency content. (Not that I have any clue what Rod was trying to say...)
Loudness in a final product must start where everything starts - writing, arrangement, and performance. If you start from a careful and deliberately controlled dynamic range in the piece, have the performers control their dynamics in the performance, and choose sounds and treatments which support this effectively, then you're almost done before you even started.
Then you should be able to do just minor things in the mix stage - a little automation or compression on individual tracks, subgroups, and/or the master fader and leave almost nothing for the mastering engineer to do at all.
Then the mastering engineer maybe uses some small amount of compression and/or limiting to try to knock down. The truly abberant peaks which may have gotten past the mixer. If you're doing mush more than that at this stage, and you really want more apparent loudness without "squashing" or "clipping", then you need to step back. First decide if the sort of loudness you're looking for is really appropriate for the piece and then go back to mixing stage and see if you can get a bit more there without completely destroying it. Maybe you have to go back and retrack the part with a tweak to the sound and more discipline from the musician. If you find yourself then questioning the arrangement... Then you have to again assess whether you have appropriate expectations. Maybe this song just isn't that loud! Do we change it to be louder, or do we let it sit where it sits?
MusicWater
Banned
When recording engineer A says he is recording the signal level at -10 dBFS and engineer B says he is recording at that signal level too, it kind of does not mean so much. Because if engineer A is using a +24 dBu converter and engineer B is using a +18 dBu converter, what it really means is that engineer A has his signal level at 11 volts peak-to-peak (4 volts RMS) while engineer B has his signal level at 5.5 volts peak-to-peak (2 volts RMS). In other words, the difference in their signal level is 100%. They think they are recording at the same signal level relative to each other, but they really don't. This is probably the biggest confusion in recording, the ones that get it (those at the top) kind of keep silent about it because they know this is where the money/quality is. Since this sits both on the input and the output side, the relative difference in terms of POTENTIAL product quality difference just between these two setups in this particular example caused by this particular aspect is far greater than 1000%... 
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Oh! It's conspiracy theories now then??!!
0dBFS is not "open to debate" or modification, it is where you are when all the bits are used up!
Most HR's reader's interfaces will not get close to +18dBu for 0dBFS leave alone +24 and almost none of us with bog S kit have the option of re-calibrating our converters.
Loudness (in this context) is a SUBJECTIVE phenomenon and little to do with line up levels.
Revox A77s were also way below studio line level, did not stop some people making damn fine recordings on them.
Dave.
0dBFS is not "open to debate" or modification, it is where you are when all the bits are used up!
Most HR's reader's interfaces will not get close to +18dBu for 0dBFS leave alone +24 and almost none of us with bog S kit have the option of re-calibrating our converters.
Loudness (in this context) is a SUBJECTIVE phenomenon and little to do with line up levels.
Revox A77s were also way below studio line level, did not stop some people making damn fine recordings on them.
Dave.
MusicWater
Banned
Conspiracy, maybe for some, but not for me. Unimportant, maybe for some, but not for me. If you have a small headroom and utilize it inefficiently you get something completely different than when you have a large headroom and utilize it efficiently. Then of course you can mix setup A with a bunch of newbies and mix setup B with a bunch of experts, but that's a different story, a bit unrealistic too, because experts care about their headroom size and how they utilize it, hence more often than not you see setup A matched with experts and setup B matched with newbies. Then of course in between you have pros.
I think you are confusing headroom with dynamic range.
Your "headroom" stops at 0dBFS. What you really have is "legroom" and it is a poor AI today that does not have a noise floor of -100dBFS or better. After that it is just gain. If you really want +24dBu from a USB AI just bolt on a pair of NE5532s per channel as balanced amps (or get BBrown jobbies).
dAVE.
.
Your "headroom" stops at 0dBFS. What you really have is "legroom" and it is a poor AI today that does not have a noise floor of -100dBFS or better. After that it is just gain. If you really want +24dBu from a USB AI just bolt on a pair of NE5532s per channel as balanced amps (or get BBrown jobbies).
dAVE.
.
DM60
Well-known member
OK, dumb guy here, but I am going to throw it out here. Can't be any worse than some of the other posts.
If I have an analog signal that captures the sound I want, even if it is low, convert to digital add all the gain I need to get it to the level I want, there is really no negative as the digital increase does not add any noise.
In the analog world, I understand that would not be the case as the gain amps could introduce more noise, so source signal would be important. That is not the case in the digital world.
I'm sure I missed something here, but don't know it is just yet.
If I have an analog signal that captures the sound I want, even if it is low, convert to digital add all the gain I need to get it to the level I want, there is really no negative as the digital increase does not add any noise.
In the analog world, I understand that would not be the case as the gain amps could introduce more noise, so source signal would be important. That is not the case in the digital world.
I'm sure I missed something here, but don't know it is just yet.
MusicWater
Banned
I mean headroom, that starts at the nominal level +4 dBu and in the case of a +24 dBu converter ends at +24 dBu, in the case of a +18 dBu converter it ends at +18 dBu, so I am referring to this (6 dB) difference in headroom size. (20 dB vs. 14 dB)
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miroslav
Cosmic Cowboy
When recording engineer A says he is recording the signal level at -10 dBFS and engineer B says he is recording at that signal level too, it kind of does not mean so much. Because if engineer A is using a +24 dBu converter and engineer B is using a +18 dBu converter, what it really means is that engineer A has his signal level at 11 volts peak-to-peak (4 volts RMS) while engineer B has his signal level at 5.5 volts peak-to-peak (2 volts RMS). In other words, the difference in their signal level is 100%. They think they are recording at the same signal level relative to each other, but they really don't. This is probably the biggest confusion in recording, the ones that get it (those at the top) kind of keep silent about it because they know this is where the money/quality is. Since this sits both on the input and the output side, the relative difference in terms of POTENTIAL product quality difference just between these two setups in this particular example caused by this particular aspect is far greater than 1000%...
-10 dBFS is -10 dBFS......the amount of headroom of a given converter doesn't change that.
MusicWater
Banned
I am talking about the relative difference in headroom size, in the case of a +24 dBu converter -10 dBFS corresponds to +14 dBu (having utilized 10 dB of its available 20 dB headroom), in the case of a +18 dBu converter -10 dBFS corresponds to +8 dBu (having utilized 4 dB of its available 14 dB headroom), because +24 dBu and +18 dBu corresponds to their individual full scale.
miroslav
Cosmic Cowboy
You are confused with the way you are doing your math.....-10dBFS is -10dBFs.
If you take a meter, and stick it post converters, then play back two -10dBFS signals (one through your +24dBu box and one through your +18dBU box)....they will be the same signal strength post converters.
The headroom of the converter does NOT change the signal strength....
Nope, don't think you have missed anything there.
The mastering guys here have said here several time that they would much rather have tracks at neg 25 even -30 than slamming the limit.
But none of this is new! Even 40 years ago many top professionals were urging a 4-6dB drop in tape modulation because it cleaned up things so much. Far less distortion, squash and print. This was especially promoted with the coming of Dolby A since a few dBs of the 10dB noise reduction could be traded for a sweeter overall sound.
Dave.
MusicWater
Banned
If you run a signal in/out at the full scale of a +24 dBu converter, the signal level in/out is +24 dBu or 34.7 volts p-p. If you run a signal in/out at the full scale of a +18 dBu converter, the signal level in/out is +18 dBu or 17.4 volts p-p. If you run a signal in/out -10 dB from the full scale of a +24 dBu converter, the signal level in/out is +14 dBu or 11 volts p-p. If you run a signal in/out -10 dB from the full scale of a +18 dBu converter, the signal level in/out is +8 dBu or 5.5 volts p-p. This essentially means you can input/output a signal to/from the +24 dBu converter 6dB hotter before it clips, its headroom is 6dB greater relative to the +18 dBu converter, this corresponds to a 17.3 volts p-p difference in their max capacity.
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miroslav
Cosmic Cowboy
Now you are changing what you originally said.....which only confirms that you were confused to begin with.
There is no direct math relationship between dBFS and dBu
dBu
dB
dBFS
Not all the same thing...which is what you are mixing up.
<EDIT>
Also...the original signal that you sent to the converter....will be the same when it comes out, regardless of which converter you use, or its headroom.
Sending into one converter a +24dBu signal and into another a +18dBu signal is already two different things, so you can't compare them.
When you record, if you split your signal and send the SAME signal to one converter that has +24dBu headroom, and another that has +18dBu headroom....you will still get the SAME signal from both converters into the DAW. The converter with +24dBu headroom will NOT give you a hotter signal.
There is no direct math relationship between dBFS and dBu
dBu
dB
dBFS
Not all the same thing...which is what you are mixing up.
<EDIT>
Also...the original signal that you sent to the converter....will be the same when it comes out, regardless of which converter you use, or its headroom.
Sending into one converter a +24dBu signal and into another a +18dBu signal is already two different things, so you can't compare them.
When you record, if you split your signal and send the SAME signal to one converter that has +24dBu headroom, and another that has +18dBu headroom....you will still get the SAME signal from both converters into the DAW. The converter with +24dBu headroom will NOT give you a hotter signal.
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I haven't been reading / following this thread, but you're simply talking about differences in sensitives and/or gains on the analog ends of a converter, can be set (designed to be) where ever. RME- does three selectable ranges in and/or out to accommodate different gear (or nominal' levels if that's an applicable term here?). So same converter chips' what have you, same same scales in the digi sides, but selecting the 'more head room' option would be.. attenuation (or more or less gain applied) to your signals at the inputs/outputs.
Oops, Miroslav posted while I was typing -wasn't directed to your post..
Oops, Miroslav posted while I was typing -wasn't directed to your post..
BroKen_H
Re-member
MusicWater, please read the following:
dB dBu dBFS dBV
Especially the part that says "There is no decibel to dBFS converter
Notice - Comparing dBu and dBFS: There is really no fixed
world standard like e.g. −20 dBFS = +4 dBu = 0dBVU.
The digital peak scale is not equivalent to the analog RMS scale.
You can never match dBFS and dBu. "
Hope that helps...
dB dBu dBFS dBV
Especially the part that says "There is no decibel to dBFS converter
Notice - Comparing dBu and dBFS: There is really no fixed
world standard like e.g. −20 dBFS = +4 dBu = 0dBVU.
The digital peak scale is not equivalent to the analog RMS scale.
You can never match dBFS and dBu. "
Hope that helps...
ashcat_lt
Well-known member
miro - It's exactly what he's been saying all along. The guy with the -18dbu converters needs to keep his analog levels lower in order to hit -10dbfs, and when it comes back out to analog the levels will be lower. This puts you closer to the noise floor no matter which way you look at it. I don't know that it's some "secret of the masters", but it is there.
DM60 and ecc83 - Be careful here. First of all, digital gain does amplify the analog noise that came in with your signal, which really is usually the worst part of adding gain even in analog. I suppose it's obvious, but if your levels are down at -30dbfs, it means you're already way down in the analog noise floor. Then too there really is a noise floor in digital. It's really low, but not really that far away from good analog specs. If you do some absurd things, you will hear it. No matter how deep your mix engine is, you will never have any better dynamic range than what comes through the converters (without a whole lot of fucking around). Digital is a bit more forgiving than analog, but it's no excuse for poor gain staging.
DM60 and ecc83 - Be careful here. First of all, digital gain does amplify the analog noise that came in with your signal, which really is usually the worst part of adding gain even in analog. I suppose it's obvious, but if your levels are down at -30dbfs, it means you're already way down in the analog noise floor. Then too there really is a noise floor in digital. It's really low, but not really that far away from good analog specs. If you do some absurd things, you will hear it. No matter how deep your mix engine is, you will never have any better dynamic range than what comes through the converters (without a whole lot of fucking around). Digital is a bit more forgiving than analog, but it's no excuse for poor gain staging.
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