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Originally Posted by obuckley /img/forum/go_quote.gif
I do not disagree with Xnor either about a noise floor affecting one's ability to perceive signal. Noise is unhelpful. However, it is a particularly human gift to be able to screen out parts of the noise spectrum (even when large parts of that would normally be regarded as signal) and focus the ears on e.g. what the 3rd violin to the left in the second row of an orchestra is playing and conclude that he/she is exceptionally gifted/total rubbish. I am not sure that is easily quantifiable.
Assume you boost an extremely low-level sound in a dithered 16 bit file so that you can hear the noise floor clearly. (This noise floor usually is recorded noise but let's assume the recording is perfect and the noise floor is just dither.)
Even if that sound is below the noise floor, you can still hear it. Have you ever had such a freaking extreme clean recording where you had to turn up the volume so much that you could hear the dither noise clearly in order to hear low-level details?
While I welcome recordings with great dynamic range, it does get annoying at some point. Like in movies where you have to turn up the volume to understand what the characters are saying in dialogs but that volume would blow your ears during action scenes. Also, if we take a look at (dynamically uncompressed) concert hall recordings, we rarely see >70 dB dynamic range due to the noise floor. Sure, maybe there are some details in the noise floor like someone scratching his/her nose, so what?
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The discrete values from -32k to +32k are discrete volume levels. If you want to reduce the volume a bit e.g. to remove segments of a recording which have gone into clipping by exceeding 32k, you take the numbers down for a couple of microseconds in the waveform, or you take the whole waveform down a notch, or a larger part of it. One way or another, high numbers, high volume - one to one correlation.
No, they are discrete values for individual samples. But sounds often consist of several hundreds or thousands of samples, not single samples. So the different volumes of a sound are not fixed to those discrete levels.
Let's take 8 bit quantization to make things simpler:
127 = -0.07 dBFS
126 = -0.14 dBFS
Generate a sine wave at -0.10 dBFS and quantize it to 8 bits using simple triangular dither. Take a look at the spectral analysis and compare it to that of the initial signal. They match, despite the fact that -0.10 dBFS = 126.5 (8 bit) and an 8 bit sample can only have the value 126 or 127, nothing in between (also see below).
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I don't know at a bit level what over-sampling DACs do - I don't care much - what I do know is that if you take a digital recording in 24 bit resolution and look at the numbers, they still range from -32k to +32k but there is a decimal point in there for you to fool with which is not available in 16 bit. I know this, because I do it.
I'm not speaking about the implementation of DACs either. The range of discrete values is a function of bit depth:
24 bit: -2^23 to 2^23-1 or -8388608 to 8388607
16 bit: -2^15 to 2^15-1 or -32768 to 32767
.. and so on ..
Integers do not have a decimal or fractional component. Integers consist of natural numbers (0, 1, 2, 3 ...).
If you "blow up" a 16 bit file to a 24 bit one you will have unused values. For example 32767 -> 8388352 and 32766 -> 8388096 which, surprise, is a difference of 256 (= 2^8).
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I even kind of get the impression from the little I read of audio product reviews that non-oversampling now tends to get a better press than oversampling. You have already irrevocably lost any extra precision. Oversampling always struck me as unproductive - you can no more conjure up a change of waveform by multiplying the numbers than you can retrieve what is lost in a 128kbps MP3 by re-recording it as a 320kbps. If you start with a recording where that precision is there from the start, it stays there until you eliminate it e.g. by compressing 24bit data down to 16bit. Garbage in, garbage out.
It seems you're confusing several different things here. Oversampling, as the name suggest, deals with the sample rate not bit depth. Lossy codecs throw away information based on psychoacoustic models.
Please be more specific about each of these terms if you want a more detailed response.
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Please don't think I am claiming to be able to tell the difference. I can barely hear anything above 12,000 Hz these days, let alone 25,000 and it took me quite a while, including a trip to the doctors' to get my ears syringed before I finally realised that my teenaged kids had blown the tweeters on my stereo
. To my partial credit I did notice
something was wrong.
I can't help feeling that CD specs were put together 35 years ago when digital technology was at an early stage. If we can't do better than that with the technology that's available to us now, I'd be a bit surprised - and it is not difficult to do. Most studio masters these days are done in digital form at 24/196 and it's pretty easy to copy digits.
You have to distinguish all the recording and processing from the final playback/reproduction. Let's assume there is recording both available in 44.1/16 and 44.1/24 but you cannot hear a difference between the two. You wouldn't buy the usually more expensive 44.1/24 file right?
But those people who earn money by selling you such formats also try to sell you that there are huge differences..