Re: Audio Questions
- From: Frank <frank@xxxxxxxxxxxxxxxxxxxxxxxxxx>
- Date: Wed, 05 Jul 2006 03:33:11 GMT
On Tue, 4 Jul 2006 18:44:29 -0700, in 'rec.video.desktop',
in article <Re: Audio Questions>,
"Richard Crowley" <rcrowley@xxxxxxxxx> wrote:
"PTravel" wrote ...
Exactly as I said. It's PCM, just like a WAV file,
in another format.
"WAV" *is* "PCM"
Not necessarily, and anyone who believes this is clearly uneducated in
this area, no offense.
"AIFF" is PCM.
I'm not a Mac person, but I believe this to be true. If the data were
compressed, then the proper extension would be .aifc and not .aiff, at
least as I understand it.
All uncompressed audio files are "PCM".
Untrue. Have you ever heard of Sony's DSD (Direct Stream Digital)
format? It's a one-bit format sampled at 2.8224 or 5.6448 MHz. And the
term PCM is a bit misleading since it comes in two basic flavors: LPCM
(Linear Pulse Code Modulation) and non-linear pulse code modulation,
the latter of which isn't being discussed here, but it still exists.
In the world of DVD-Video discs, we purposely use the term "LPCM"
rather than simply "PCM" so that it's clear that we're referring to
linear PCM as opposed to non-linear PCM.
"PCM" is an archaic term for any uncompressed digital
audio encoding.
Untrue. PCM, whether linear or non-linear, refers to a specific method
of sampling analog audio data. There's nothing archaic about it. Aside
from the small amount of DSD work being done, virtually all sampling
of analog audio is done using either linear PCM or non-linear PCM,
mostly the former.
The term "PCM" generates this kind of
confusion every time it is used.
If you mean because of the two flavors, linear and non-linear, then I
agree, but I don't think that's what you mean.
May I use DAT (Digital Audio Tape) as an example? Standard DAT is
16/48 LPCM, but most DAT machines also offer 16/44.1 LPCM and 12/32
non-linear PCM. (I offer this simply as an example of the use of
non-linear PCM, that's all.)
"PCM" means "pulse-code modulation" and was coined
back in the days before "digital" became a popular term.
Even the Wikipedia entry admits that the term is
"somewhat confusing".
The data on a CD is a long string of 16-bit words, first
the left sample, then the right sample, Left/Right, Left/
Right, etc. etc, repeated 44100 times per second.
It's actually a long string of blocks of data, each block containing
1/75th of a second's worth of 2-channel 16/44.1 LPCM digital audio.
Each block is 2352 bytes in length. (I'm just trying to stress that
"block" is the correct term to use.)
For those who care, here's the simple arithmetic:
16-bits of data per sample per channel times 44100 samples per second
per channel equals 705,600 bits per second per channel times two
channels equals 1,411,200 bits per second divided by 8 bits per byte
equals 176,400 bytes per second divided by 75 equals 2352 bytes per
block.
Exactly the same as the content of a WAV file.
Here we go again.
It's not even the same as a typical LPCM-encoded .wav file in that a
typical LPCM-encoded .wav file doesn't have it's data written in
1/75th of a second blocks. Also, in point of fact, a .wav file uses
chunks, not blocks. Again, no offense, but I find that using the
correct terminology, chunks, in this case, versus blocks, helps to
avoid confusion. I think that there's enough of that in this world
already.
Only the
"housekeeping" is different, but then the "housekeeping"
of ANY digital medium is different in the same way.
--
Frank, Independent Consultant, New York, NY
[Please remove 'nojunkmail.' from address to reply via e-mail.]
Read Frank's thoughts on HDV at http://www.humanvalues.net/hdv/
.
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