Jack Bauer wrote on Fri, 12 October 2007 10:47 |
Hi there,
I'm a sound engineer from Greece and recently I made the decision to start my very own blog about audio, so that I can reinforce the audio community even a little bit. The blog's subjects include professional audio, music and tech news, and more. I would be glad if some of you would pass by even for a mere glance and let me know of whatever you might like or not. All comments are always welcome. I do my best to research and post quality content every day.
http://www.audioworkbench.com
Thanks in advance for your time!
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Dude, your
"Digital Audio Explained" has a bunch of fundamental errors. First and foremost, the square wave you show (wavesinesampled.jpg) as an example of why you sample at > 2x the highest frequency component is wrong. Also, your graphic that shows the effects of 8-bit words vs 16-bit words is not correct. Put simply, you don't simply draw a line to connect the dots of each sample. Look up "reconstruction filter." Dan Lavry's web site has an excellent explanation, complete with some MathCad graphs.
You also say, "As a rule, with each bit added, we gain 6 dB of dynamic range. Keep in mind that the dB is a logarithmic value, and practically means that with 6 dB more the signal gets twice as loud." The first sentence is correct; adding a bit increases the dynamic range by 6.02 dB. But the second statement, "6 dB more [means] the signal gets twice as loud," is wrong. Adding bits improves resolution at the low end, assuming your reference stays the same.
-a
PS: is your name really Jack Bauer?