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16 bit / 24 bit mixing

May 17, 2012 10:48 AM

(I've been here a bunch before trying to figure out how to get what i hear in the multitrack sounding like my mixdowns, and I finally got everything sounding right--CS6 did it)

 

So now that I've got my mixdowns sounding good, I was wondering if (since my Blue Yeti Pro mic supports 24 bit recording) I should be working with a 24 bit session or a 16 bit session--because the mixdowns will mostly end up on CDs.  Will what I hear on the CDs sound best if I work in 16 bit multitrack sessions or will they sound just as good if I export the 24 bit sessions as 16 bit mixdowns?

 
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  • SteveG(AudioMasters)
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    May 17, 2012 11:08 AM   in reply to wycks

    As far as bit depth is concerned, it''s generally the case that you  record and edit with the highest number of bits, and only make a reduced bit-depth copy for duplication purposes.

     

    Audition's internal engine in Multitrack view works in 32-bit Floating Point mode, and this has significant technical advances, so stick with that for production purposes. When you want to make a CD from the results, when you create a 16-bit copy, you also get the option to choose the dither you use. You haven't had anything dithered up to this point, and since you only dither a signal once, this is the point to do it. You still have your 32-bit master though, so if you need, for instance, to make MP3 files as well, you can make these also from the best quality copy.

     

    Incidentally, there's absolutely no advantage whatsoever to recording at a higher bit rate - stick to 44.1k throughout.

     
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  • SteveG(AudioMasters)
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    May 17, 2012 1:22 PM   in reply to wycks

    Don't worry - you almost certainly won't improve them at all. You may have recorded 24-bit files, and saved 24-bit files, but Audition would have processed them internally as 32-bit floating point anyway. Chances are that if you have a reasonable output level in your mixed-down 24-bit files, then you won't notice the difference. The only times you can really get caught out with integer files is when you save a session, having inadvertently set the volume level really low on a file, and upon reopening, try to increase the level back to where it was. Chances are that this file will sound awful, because you will have reduced the actual bit depth down to a very low number, and saved it like that. If you'd done that with a 32-bit Floating Point file, you could have restored it without loss upon re-opening - no chance at all of doing this with an integer file.

     

    But by and large, saving mix sessions in Audition's native format is the way to go - just do bit reduced copies for duplication purposes.

     
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  • SteveG(AudioMasters)
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    May 18, 2012 1:54 AM   in reply to wycks

    Okay, this is slightly simplified in terms of the numbers, but generally the case with integer-based files:

     

    If you had an integer file - let's say a 16-bit one - and decided to reduce the amplitude of it by 36dB in your session, the session would ask you whether you wanted to save the file. If you said yes, then you'd have a 16-bit file with a maximum possible amplitude (and therefore signal to noise ratio) of 60dB (you can have up to 96dB of dynamic range in a 16-bit file). But, that amplitude, having been reduced by Audition, will result in your signal not using up any more than 10 of the 16 bits available.

     

    The trouble comes when you re-open the session and decide that you want that file back at its original amplitude. You've now got a rough-sounding 10-bit file masquerading as a 16-bit one...

     

    Okay, that's quite an extreme example, but the basic situation is the same with all integer files. It's less of a problem with 24-bit ones, because you have a lot more bits to play with, and even if it reduced to 16 bits it would sound fine. The real issues with integer files from this point of view is what the reduced bit depth does to quiet things like ambient sounds, and natural reverberation - it screws those up a treat. The absolute joy of the Floating Point system is that all the actual signal data remains intact (although in a normalised form). It's just the scaling that is altered - so reducing the amplitude, saving, and then increasing it again results in a file that is identical to the original; no losses at all, as the bit depth has now become insignificant.

     
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    May 18, 2012 2:42 AM   in reply to SteveG(AudioMasters)

    May I steal your explanation for somebody else, Steve?  I've been trying to explain the advantages of floating point working in another forum and so far have got nowhere.  Your version is lovely and concise!

     

    Incidentally, there's absolutely no advantage whatsoever to recording at a higher bit rate - stick to 44.1k throughout.

    Being pedantic, I've always tried to stick to the terminology "bit depth" and "sample rate" just to avoid confusion.  Since you're usually very precise in your choice of words, have I had this wrong all these years?

     
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  • SteveG(AudioMasters)
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    May 18, 2012 4:28 AM   in reply to Bob Howes

    Bob Howes wrote:

     

    May I steal your explanation for somebody else, Steve?  I've been trying to explain the advantages of floating point working in another forum and so far have got nowhere.  Your version is lovely and concise!

     

    Incidentally, there's absolutely no advantage whatsoever to recording at a higher bit rate - stick to 44.1k throughout.

    Being pedantic, I've always tried to stick to the terminology "bit depth" and "sample rate" just to avoid confusion.  Since you're usually very precise in your choice of words, have I had this wrong all these years?

    You can steal it as long as you credit it...

     

    And normally I'd refer to sample rate as well, but technically it's not actually wrong to refer to a bit rate... I only used bit rate above specifically  to distinguish two aspects of the same thing - how many there are, and how fast they go. I recognised the issue at the time though - which is why I italicised 'rate'.  Admittedly though, you'd usually expect to see it expressed like that as a rather useless kb/s figure.

     
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    May 18, 2012 8:00 AM   in reply to SteveG(AudioMasters)

    Are you talking about the kbs figures used for mp3 encoding which are called 'bitrate'?  Cuz that's not useless at all - it's a pretty good predictor of playback fidelity.

     
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  • SteveG(AudioMasters)
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    May 18, 2012 12:13 PM   in reply to wycks

    wycks wrote:

     

    Thanks for the clarification.  But just to be sure, if I'm working in a 16/24 bit session and record tracks into the multitrack, but don't lower the db, when I come back to the session, the quality of the recorded tracks will still be intact?  But if I lower the db in the 16/24 bit session, save the session, and open it again, I'll have lost quality in the tracks that i've lowered the db?

     

    If you save the files with the much lower levels, yes. Much safer to do all production work in 32-bit floating point, and not worry about it!

     
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  • SteveG(AudioMasters)
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    May 18, 2012 12:27 PM   in reply to therealdobro

    therealdobro wrote:

     

    Are you talking about the kbs figures used for mp3 encoding which are called 'bitrate'?  Cuz that's not useless at all - it's a pretty good predictor of playback fidelity.

     

    It's only any use if you relate it to a constant bit depth, or a constant sample rate. If you express neither, then you get ambiguities. With apologies to Wikipedia, let me quote the start of the problem:

     

    'Compression efficiency of encoders is typically defined by the bit rate, because compression ratio depends on the bit depth and sampling rate of the input signal. Nevertheless, compression ratios are often published. They may use the Compact Disc (CD) parameters as references (44.1 KHz 2 channels at 16 bits per channel), or sometimes the Digital Audio Tape (DAT) SP parameters (48 kHz, 2 times at 16 bit). Compression ratios with this latter reference are higher, which demonstrates the problem with use of the term compression ratio for lossy encoders.'

     

    But it doesn't end there - encoder quality makes a mockery of quoting bit rate figures - more Wikipedia:

     

    'Besides the bit rate of an encoded piece of audio, the quality of MP3 files also depends on the quality of the encoder itself, and the difficulty of the signal being encoded. As the MP3 standard allows quite a bit of freedom with encoding algorithms, different encoders may feature quite different quality, even with identical bit rates. As an example, in a public listening test featuring two different MP3 encoders at about 128 kbit/s, one scored 3.66 on a 1–5 scale, while the other scored only 2.22.  Quality is dependent on the choice of encoder and encoding parameters.'

     

    So like I said, pretty useless!

     
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    May 18, 2012 3:54 PM   in reply to SteveG(AudioMasters)

    Wow.  Okay, so conditions are necessary to make the statement true.  For instance: 'Given one particular wav file being encoded by the same software, a higher bitrate is a predictor of playback fidelity.'

     

    Like you say, useless for predicting fidelity on anything other than your own stuff when you're encoding it (because you know the quality it was recorded at).

     
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