Analog to Digital Conversion

Steve Rowe (test lead on the sound team) points to a great article from Ars Technica on D2A:

If you want digital audio in a computer, you have to get it from somewhere.  Usually that means taking analog sound out of the air and turning it into the bits that a computer can understand.  Ars Technica gives us another installment of the AudioFile. This one covers the subject of Analog to Digital Conversion.

Well worth reading.

Comments (7)

  1. Nik says:

    Totally off-topic, but I read your post on "Save the Blibbet", and I’ve noticed that the Blibbet seems to be making a comeback as a window sticker in several cafeterias (32, 112, and maybe others).  It’s about 1-2 feet in diameter. I wonder who is behind this ?

  2. Nik, I’ve been wondering the same thing.  But I have no idea.

  3. Norman Diamond says:

    "I wonder who is behind this ?"

    I thought that behind a window sticker there would be a window?  And if you’re using a sufficiently recent version of Linux or Windows then you can check for yourself by inspecting the partially obscured window?

  4. Norman – we’re talking physical windows here, not virtual ones.

  5. Igor says:

    Larry, I believe Norman was trying to be funny 🙂

  6. Norman Diamond says:

    Yup.  I submitted a comment containing just one word, "Whoosh", but it looks like that was considered spammy.

  7. Donnie Hale says:

    A question, Larry, on the D/A article. I’ve always wondered how that process works – the usual high-level articles use a single wave and show the sampling. That’s never quite made sense to me – in the analog world of "sound through the air" the sound of a song, for example, at any instant is the sum of the amplitudes for every frequency being played / reproduced / heard. Does that make sense, and is that right?

    If so, I’ve always wondered how a single wave in the high-level examples can represent all the frequencies and amplitudes in a song. After reading that article, it seems as though it’s not an actual set of frequencies / amplitudes being sampled but a voltage which, in an analog "encoding" of sound, represents that frequency / amplitude sum at any point in time.

    Do I have that right? Care to add a brief elaboration? Thanks.