Dave Bradley wrote:

>> OK...so am I understanding correctly that you are transferring 
>> through your converter into your computer at 16bit, but then 
>> importing it into ProTools for 24bit?
>>
>> If this is the case, you're not "enhancing" the sound quality, but 
>> just using some kind of algorithm(s) in order to get a more complex 
>> file - information that was added, by the way, and has nothing to do 
>> with the original.
>
>
> That's not entirely accurate. In fact, it's quite INaccurate.
>
> Any adjustments made to the sound file will result in a mathematical 
> calculation which is never done as simple addition or subtraction. The 
> simple process of even just boosting the level by 3 db results in a 
> multiplication process that when fed 16 bits of data will result in 
> more than 16 bits of final calculation in the form of a decimal.
>
> Some people would simply say truncate that result back to 16 bits and 
> you're accurate enough for government work.
>
> The problem with that is that any additional work done on the file 
> will also result in a similar calculation.  Let's say this time it's a 
> bit of hiss reduction, and maybe some EQ boost to the 1KHz to 3KHz 
> band.  If each of those is working with the 16 bit PLUS decimal value, 
> then each calculation will result in a more accurate representation of 
> the final value. Only when all processing is done should you drop back 
> to 16 bits through whatever process, truncation, dithering, noise 
> shaping, etc. that you feel is appropriate.
>
> If, on the other hand, you leave the file as 16-bit, each calculation 
> done for each adjustment you make will leave an error which the next 
> calculation will amplify until you start to get noticeable problems.
>
> A totally made up example which gets the idea across could be as follows.
>
> Initial value 179.
> Your first calculation is your initial value * 0.5749.
> Your second calculation is the result of your first value * 4.7768.
> Your third calculation is the result of your second value * 2.48.
> Your fourth calculation is the result of your third value * 1.17.
> Your final value is the result of the fourth calculation brought back 
> to 16-bit world.
>
> If you start at 16 bit and stay at 16 bit through this entire process, 
> then here's what you get.
> First calculation (179 * 0.5749) results in 102.90710000 which is 
> immediately truncated to 102 so it will fit entirely in the 16-bits 
> which has no room for the decimal value. No, it won't round it up to 103.
> Second calculation (102 * 4.7768) results in 487.23360000 which is 
> immediately truncated to 487 so it will fit entirely in the 16-bits 
> which has no room for the decimal value.
> Third calculation (487 * 2.48) results in 1207.76000000 which is 
> immediately truncated to 1207 so it will fit entirely in the 16-bits 
> which has no room for the decimal value. No, it still won't round it 
> up to 1208.
> Fourth calculation (1207 * 1.17) results in 1412.19000000 which is 
> immediately truncated to 1412 so it will fit entirely in the 16-bits 
> which has no room for the decimal value.
> Your final value is 1412 after 4 simple adjustments to the audio data.
>
> Now, if you take that 16-bit file and import it as 24-bit, do your 
> calculations and then truncate, dither, or otherwise produce a 16-bit 
> final file, you'll notice the value changes drastically.
> First calculation (179 * 0.5749) results in 102.90710000 (the extra 8 
> bits holding the value to the right of the decimal place).
> Second calculation (102.90710000 * 4.7768) results in 491.56663528.
> Third calculation (491.56663528 * 2.48) results in 1219.08525549.
> Fourth calculation (1219.08525549 * 1.17) results in 1426.32974893.
> Let's assume you do a simple truncation at this point to go back to 
> 16-bit. That value is now 1426, whereas when you worked strictly in 
> 16-bit it turned out to be 1412.
>
> Hmm, which one was more accurate?
>
> You'll notice that in the "stay in 16-bit" example above, the 
> truncated results each time were small enough that they certainly 
> weren't using the full 16 bit capabilities of the digital sample.  
> (102, 487, 1207, 1412) The first number, 102, could be represented in 
> 6 bits, the second number, 487, could be represented in 9 bits, the 
> third number, 1207, could be represented in 11 bits, and the fourth 
> number, 1412, could be represented in 11 bits. So it's obvious that 
> we're not dealing with audio that is pressing the limit of 16-bit 
> sampling. The argument that because cassette tape is not up to CD 
> quality so any processing can remain in the 16-bit realm has just now 
> been invalidated.
>
> You MAY be able to capture all of the existing audio from a cassette 
> in 16-bit sampling, but NEVER do any processing in 16-bit. Do it in 24 
> or even 32 bit and then drop your final result to 16-bit through 
> whatever method you wish and you'll end up with a far more accurate 
> digital audio file than if you allowed all those computational errors 
> to creep in.
>
> (And in case you didn't notice, the difference between 1412 and 1426 
> is NOT just a small rounding up or down error.  It's 14 values 
> different.)
>
>
>
> -----------------
> Diamond Productions
> Preserving the past for the future.
> Dave Bradley   President
>
>

I'm sorry to say that I do not agree with you at all.  Your calculations 
look very impressive, but as you said, the examples cited are totally 
made up.  And there seems to be numerous factors not included in the 
equations (huge increase in digital noise and relative distortion from 
the original 16 bit file being upsampled to 24 bits; and one major 
oversight, that is, dithering [or truncating] down a 24-bit file to 
16-bits creates aliasing and quantizational errors [ie dithering 
noise]).  And this is regardless of what software or noise-shaping 
algorithms used.  Why do all this to audio from a cassette that most 
likely already suffers from noise/hiss problems?  Oh, and last but not 
least, my ears can pick up the other problems that this scenario creates 
-- audible digital artifacts.  The only way to avoid having to deal with 
this mess is to do the job right in the first place and make the 
original transfers in 24 bit.  It might be best for the original poster 
to invest a few dollars in obtaining an Alesis Masterlink.

One man's opinion.

Bob Conrad
Fort Lee, NJ