Apple Logic Pro /LogicExpress Discussion

To answer the second question first, yes.  You can import any audio 
file in any supported format into the project.  You can have Logic leave 
it as-is or convert it.  The relevant section of the manual has this to say:

"Audio files imported into a Logic Pro project can be at any supported 
bit depth and sample rate. Logic Pro supports bit depths of 16, 20, and 
24 bits, and sample rates of 44.1, 48, 88.2, 96, 176.4, and 192 kHz. 
Logic Pro can use the file’s sample rate, or can perform a real time 
sample rate conversion (see “Setting the Project Tempo” on page 145)."

The answer to the first question is both simpler and more complicated.  
:)  It's simpler in that there really is no over-arching bit depth 
setting for a project, so there's nothing to set.  It will handle files 
in whatever format you gave it to them.  The way in that it's more 
complicated is that Logic Pro uses 32-bit floating point processing 
internally (I have heard some tell that it uses 64-bit "where 
appropriate", but I haven't looked into it in much depth so I can't 
verify that or say what exactly "where appropriate" is).  The "floating 
point" part is the bit that actually wasn't covered in my explanation at 
all, but can be more or less translated into "You don't need to worry 
about it."  You CAN set the bit depth for your audio interface, but that 
just affects what the signal you hear is sent to the interface it -- it 
doesn't affect Logic's processing at all.  And when you bounce to export 
audio, you can select a bit depth each time, and you could select a 
different one every time if you wanted to, but usually if that's a final 
export (i.e. you're not moving it into another app afterward for 
mastering), you'll be exporting to 44.1khz, 16-bit anyway.  If you *are* 
going to process it after in a separate mastering tool, or you're going 
to send it to a service to be processed, you could export at 24-bit 
(although you'll want to verify with the service what formats they accept).

On 10-09-17 2:07 AM, Andy Brook wrote:
> Thank you for spending so much time giving such a detailed answer
> Ifron,  that was very easy to follow and I understand what you are
> saying now. I just have two follow up questions, if I may: can you
> switch bit rates in the middle of a project, and can you import audio
> recorded in one bit rate into a project in a different bit rate. I
> think the answer to the second is going to be yes you can, but not
> sure about the first!
>
>
> thanks again
>
> Andy Brook
>
> On 17 Sep 2010, at 05:50, Irfon-Kim Ahmad wrote:
>
>> Okay, so here is the deal with bits. A bit can be a 0 or a 1. If you
>> had a 1 bit audio interface, it could either be on or off at any
>> given moment. Now, you could still do something with that. I mean,
>> we all know such an interface as the square wave oscillator, right?
>> It's off or on. The other factor, the sampling rate, gives you the
>> ability to turn it on or off really fast, thus creating various
>> frequencies and so on. But you can't make a lot of sounds with a
>> square wave.
>>
>> So if you add a second bit, you get a 0 or a 1 followed by another 0
>> or 1. You can make the numbers 00, 01, 10 and 11. That's four
>> settings. With that you can capture a little more complex of a
>> sound, you have a few more subtleties. Your interface can be either
>> full on, pretty high, pretty low, or off, and it can vary those at
>> the sampling rate. So instead of just being able to make square and
>> pulse waves you can now make a whole bunch of different waves,
>> including extremely crude sines, triangles, etc., as well as sampled
>> or recorded sounds with some more veracity. But try to imagine
>> drawing a sine wave with only four levels -- it still won't be that
>> accurate a reproduction.
>>
>> The number of possible values for each bit depth is easy to figure
>> out -- it doubles each time:
>>
>> 1 bit: 2 values
>> 2 bit: 4 values
>> 3 bit: 8 values
>> 4 bit: 16 values
>> 5 bit: 32 values
>> 6 bit: 64 values
>> 7 bit: 128 values
>> 8 bit: 256 values
>> 9 bit: 512 values
>> 10 bit: 1024 values
>> 11 bit: 2048 values
>> 12 bit: 4096 values
>> 13 bit: 8192 values
>> 12 bit: 16384 values
>> 13 bit: 32768 values
>> 14 bit: 65536 values
>> 15 bit: 131072 values
>> 16 bit: 262144 values
>>
>> So once you get to 16 bits, you are effectively drawing your sound
>> wave with 262,144 possible values for its height at any given point.
>> You can draw curves with extreme subtleties to them with that kind
>> of accuracy, and at that point we can more or less not hear the
>> difference anymore between the real thing and our stair-stepped
>> approximation, for most people.
>>
>> So why ever use more than 16 bits, especially if you master to CD,
>> which is a 16 bit medium? The answer is that you sometimes process
>> sounds in ways that make the teeny tiny bumps more apparent.
>>
>> Suppose you've recorded a wave in 3 bits, so it has 8 possible
>> values. And let's suppose that wave is a straight diagonal line.
>> Your values look something like this:
>>
>> 1,2,3,4,5,6,7,8
>>
>> That makes an okay line, right? Right. (Well, assume it is for the
>> purposes of our discussion.) I mean, you have a "stair height" of at
>> most 1 unit, and we'll suppose that looks pretty good.
>>
>> However, suppose the line you were recording were half as loud. Your
>> values would look like this:
>>
>> 1,1,2,2,3,3,4,4
>>
>> Still the stair stepping is about the same, one unit at a time.
>>
>> Now you normalize it. Do you wind up with our original line above?
>> No. The computer says that the maximum value was half as loud as the
>> maximum possible loudness, so it will double every value. Now your
>> line looks like this:
>>
>> 2,2,4,4,6,6,8,8
>>
>> The stair steps are twice as apparent now. The volume jumps by 2
>> each time it changes. Also, we're only using 4 of our 8 possible
>> values. It's as if our recording was made with 2 bits instead of 3.
>>
>> Now think how many times you cleanse, fold and manipulate your audio
>> in a given project. If you're like me, you run it through the
>> wringer and back. Every manipulation stretches and mutates the
>> resolution of your recording. The more possible values, the more
>> resolution your data had, at the outset of the process, the more
>> data you're likely to be left with at the end.
>>
>> This is exactly why many good instruments will have 16 bit outputs
>> but perform their internal processing at 24 or 48 bits. And that's
>> why, if you do a lot of processing, you can benefit from recording
>> or processing at a higher rate and then dithering down to the target
>> format at the final mastering stage.
>>
>> -- 
>> Irfon-Kim Ahmad
>> http://www.ramp-music.net
>>
>> On 2010-09-16, at 11:48 PM, brianmc7@... wrote:
>>
>>>  From what I've been told more bits equals more dynamic
>> range???????????
>>>
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