HOW TO GET THE SOUND QUALITY SPOT ON
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HOW TO GET THE SOUND QUALITY SPOT ON
Need 2 know how to get my sound quality spot on for my tracks i EQ and compress every sample and i use master compresor and EQ for the whole tracK keeping it to just on 6 on the db metre my tunes sound good on my monitors but had tune on radio and sounded abit feeble kinda weak any tips people Cheers
www.myspace.com/smug1music
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SSSSSSSS
OOOOOOOO
NNNNNNNNN
EEEEEEEEE
RRRRRRRRRR
OOOOOOOO
NNNNNNNNN
EEEEEEEEE
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hi, i've been lurking here for a while and decided that i should finally post.
one thing about the quality of your track perhaps sounding feeble next to others is professional mastering. if you really want your tunes to stand up against others you should really consider this. just compressing and EQing the main track does not make a properly mastered tune i'm afraid.
also i don't think it's necessary to compress and EQ every track in the mix. compression and EQ should be used where needed imo, not just put on everything because that's what you think you're supposed to do.
it would be best if you read a book on mixing such as "the mixing engineers handbook" by Bobby Owinski. although a lot of it is geared towards rock music the theory is still applicable.
you should leave mastering to an ME if you don't really know what you're doing.
one thing about the quality of your track perhaps sounding feeble next to others is professional mastering. if you really want your tunes to stand up against others you should really consider this. just compressing and EQing the main track does not make a properly mastered tune i'm afraid.
also i don't think it's necessary to compress and EQ every track in the mix. compression and EQ should be used where needed imo, not just put on everything because that's what you think you're supposed to do.
it would be best if you read a book on mixing such as "the mixing engineers handbook" by Bobby Owinski. although a lot of it is geared towards rock music the theory is still applicable.
you should leave mastering to an ME if you don't really know what you're doing.
- kindofblue272
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Re: HOW TO GET THE SOUND QUALITY SPOT ON
don't keep it 6 on the meter. 0 is the ceiling with digital audio, you're just clipping your track.Sonar One wrote:Need 2 know how to get my sound quality spot on for my tracks i EQ and compress every sample and i use master compresor and EQ for the whole tracK keeping it to just on 6 on the db metre my tunes sound good on my monitors but had tune on radio and sounded abit feeble kinda weak any tips people Cheers
www.myspace.com/smug1music
i'm not a hippie,
but yo, b, i'm hip
but yo, b, i'm hip
- samkablaam
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The only point of keeping a 24bit track at -6 dB is so that a mastering engineer has some headroom to work with. If you are "mastering" the tunes yourself then you should bump it up to around 0 dB (nothing over).
Also if I was getting my stuff played on the radio then my "mastering" would involve a little more that adding an EQ and compressor. Why not send your tunes off to a proper engineer or studio?
Also if I was getting my stuff played on the radio then my "mastering" would involve a little more that adding an EQ and compressor. Why not send your tunes off to a proper engineer or studio?
yep.Junglist wrote:
Also if I was getting my stuff played on the radio then my "mastering" would involve a little more that adding an EQ and compressor. Why not send your tunes off to a proper engineer or studio?
like i said all mastering is best left to an ME tbh. at least, whenever that's possible or to your benefit, like when tuners get radio play.
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macc
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450kbit?Sonar One wrote:i keep it on -6 and the bitrate for sampling 450kbit
The correct way to express it would be in terms of a bit depth (16, 24, 32 bit, corresponding to y-axis or amplitude resolution) and a sample rate (44.1, 48, 88.2, 96 kHz, corresponding to the x-axis or time resolution).
So for example, I make my stuff at 24bit/44.1kHz.
Bit rate is not a correct term here, referring to data transfer in a computer system rather than anything to do with audio, though mp3's cause confusion here.
Oh, and when you say 'I keep it on -6'... you keep what on -6dB?
You mean the master fader is pulled down to -6? Or is the master fader at unity/0dB and the mix is peaking at -6dB on the master level meter? Or something else?
It all sounds nothing like anything to do with sample rates and all that - to me it sounds like a gain staging issue combined with a lack of understanding of signal flow. It's a question of geting your fundamentals right
We'll sort you out man!
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When you finish your mix down, go through your mixer and turn everything down by 3 db
Sounds mad, but it will give you back the headroom that you have inevitably battered in an attempt to get it sounding big and buff.
If you are working inside the box in any decent DAW, you dont need things sitting all at 0, quite the opposite in fact (headroom for the mastering engineer / you to add some post render tweaking too)
With the lower volume, things will not be clipping, and you will be able to really hear which things are out of place, instead of struggling to work out where the individual sounds are coming from through the distortion.

Sounds mad, but it will give you back the headroom that you have inevitably battered in an attempt to get it sounding big and buff.
If you are working inside the box in any decent DAW, you dont need things sitting all at 0, quite the opposite in fact (headroom for the mastering engineer / you to add some post render tweaking too)
With the lower volume, things will not be clipping, and you will be able to really hear which things are out of place, instead of struggling to work out where the individual sounds are coming from through the distortion.
Question for Bob / Jason - is this actually relevant? I can see why it'd maybe be true if I was sending the tune over in an analogue format and the whole signal chain was going to be analogue (hence meaning that S/N becomes an issue), but if I've sent you a 24 bit wav that peaks around 0db would you really think "arse, he hasn't left any headroom, now there's not much I can do with it because despite my massive rack of hardware and stuff I don't actually have the ability to turn down a digital audio file."vadarfone wrote:If you are working inside the box in any decent DAW, you dont need things sitting all at 0, quite the opposite in fact (headroom for the mastering engineer / you to add some post render tweaking too)
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macc
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@ vaderfone and all the stuff above:
Assuming you're working in virtually any modern sequencer, going through and turning everything down by 3dB is a waste of time - you may as well just turn down the master fader to regain headroom. Much faster and *exactly* the same result.
Exceptions are if you're using plugins (waves for example) that don't operate in floating point and clip. If you're driving these over 0 and then turn down the channels/groups feeding them, they won't clip (or won't clip so much) and the sound will change as a result. generally this isn't a problem, but generally be careful to keep levels sensible all the way through the signal path. Get it right as early as possible so that you aren;'t giving with one hand and taking with the other. It's inefficient
Generally the best approach is to leave the master fader at 0 (ALWAYS ALWAYS ALWAYS!) keep levels down in the box, and turn your monitors up. It's better not trick yourself by moving the goalposts all the time. Good gain structuring is a fundamental part of a good mix!
Assuming you're working in virtually any modern sequencer, going through and turning everything down by 3dB is a waste of time - you may as well just turn down the master fader to regain headroom. Much faster and *exactly* the same result.
Exceptions are if you're using plugins (waves for example) that don't operate in floating point and clip. If you're driving these over 0 and then turn down the channels/groups feeding them, they won't clip (or won't clip so much) and the sound will change as a result. generally this isn't a problem, but generally be careful to keep levels sensible all the way through the signal path. Get it right as early as possible so that you aren;'t giving with one hand and taking with the other. It's inefficient
Generally the best approach is to leave the master fader at 0 (ALWAYS ALWAYS ALWAYS!) keep levels down in the box, and turn your monitors up. It's better not trick yourself by moving the goalposts all the time. Good gain structuring is a fundamental part of a good mix!
Last edited by macc on Tue Nov 11, 2008 2:50 pm, edited 1 time in total.
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macc
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Slothrop wrote: Question for Bob / Jason - is this actually relevant? I can see why it'd maybe be true if I was sending the tune over in an analogue format and the whole signal chain was going to be analogue (hence meaning that S/N becomes an issue), but if I've sent you a 24 bit wav that peaks around 0db would you really think "arse, he hasn't left any headroom, now there's not much I can do with it because despite my massive rack of hardware and stuff I don't actually have the ability to turn down a digital audio file."
I agree, yes, though there are a few reasons for leaving headroom;
1) better to mix it down at the right level and not have to do ANY changes in the box at my end. All calculations have rounding errors. While it's being *highly* anal about it, I'd rather not do ANYTHING to a file if I didn't HAVE to.
2) assuming you are going out to analogue, then intersample peaks in the reconstructed waveform can be an issue. It *could* clip the analogue output stage of the DA, and/or the in of the first piece in the chain. Most analogue stuff will take significantly higher levels than that, but pushin it at all will make *some* difference even if pretty much inaudible. Better to mix down 3dB lower and not have it be an issue. And yes, I can turn it down, but see 1)
3) again assuming you are going out to analogue most convertors operate at their best (apparently) at around -7dB or thereabouts. I haven't done any tests to prove this but if Lavry/<ytek and all those other geeks say it, I'll take their word for it.
So; Not a big issue, but it is a small issues. and small issues are what geeks like us are here to spot and avoid
Last edited by macc on Tue Nov 11, 2008 2:53 pm, edited 1 time in total.
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Sorry to sound dumb but what do you mean exactly when you say 0? below is screenshots of both applications I use often, Reason and Soundforge.Macc wrote:
Generally the best approach is to leave the master fader at 0 (ALWAYS ALWAYS ALWAYS!) keep levels down in the box, and turn your monitors up. It's better not trick yourself by moving the goalposts all the time. Good gain structuring is a fundamental part of a good mix!


Don’t worry about people stealing an idea. If it’s original, you will have to ram it down their throats.
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macc
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^ Good advice there. Start the right way, and use others' experience!
To the Reason/SF picture bloke (Serox, sorry
) - I fuckin hate Reason's mixer etc - all numbers that mean plenty in terms of midi controllers (0-127) but FUCK ALL in the real world of mixing. PITA.
Anyway, the soundforge example looks to be peaking at around -6, so that's cool ( though it *looks* fairly compressed), and so long as the Reason red 'audio out clipping' light never goes on then you're on the right road.
I'm more than happy to help with any of this if people dunno wtf is going on btw (assuming I have time to post of course
). I don't want to patronise people though 
To the Reason/SF picture bloke (Serox, sorry
Anyway, the soundforge example looks to be peaking at around -6, so that's cool ( though it *looks* fairly compressed), and so long as the Reason red 'audio out clipping' light never goes on then you're on the right road.
I'm more than happy to help with any of this if people dunno wtf is going on btw (assuming I have time to post of course
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macc
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Wahey! Inter-thread running gag/digs. Well done youjames fox wrote:too much maths for me i'm afraid - i prefer to trust my ears....
Luckily most of this has already been proved to be audible or we wouldn't be in thie thread in the first place
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