4 ms·

Is this not the type of thing that every compression algorithm does?

by rfergie 11y ago

mschuster91 11y ago
Nope. MP3, for example, cuts off all frequencies outside the audible range and thus achieves large savings.
- GFK_of_xmaspast 11y ago
  A 44.1Khz .wav file has no frequencies outside the audible range.
  - joushou 11y ago
    How well do you hear 1Hz? A 44.1KHz doesn't have a lot of frequencies outside of the audible range, but that's not really that relevant, as that's only a minor trim compared to the crazy things lossy compression algorithms do here. Their main focus is having the result sound almost like the original. The audible sound-image is modified extensively by them, with focus on simplifications that are hard to notice for a human.
pcf 11y ago
AFAIK, every other audio compression encodes each slice of sound separately. My idea looks at the song as a whole, which I believe would be something new in the audio field.
- GFK_of_xmaspast 11y ago
  Why don't you start by taking a waveform, rip it from a CD or something, and start looking at the time-domain signal to see how many of these repeats there are.
drKarl 11y ago
That's roughly what compression algorithms (zip, rar, arj, etc) do, identify repeated bits of information and store where there are ocurrences of these. For lossy audio compression algorithms like mp3 and ogg, they use psychoacustic techniques, removing information the human ear can't hear, like very high or very low frequencies (that's why they're lossy). MP3 does something like slicing the signal in small fragments, applying some mathematical functions to them, like the Modified Discrete Cosine Transform (similar to Fourier Transform or FFT), then removing some components of it. The technique you're describing would pretty much be equivalent of using samples and sequence them, which would be akin to reinventing the .mod format (.mod, .s3m, .xm etc) To use samples though, you need the individual samples. To identify and extract the samples from the song would be much more difficult, unless maybe each instrument was in its own track or channel.

Is this not the type of thing that every compression algorithm does?

by rfergie 11y ago

Is this not the type of thing that every compression algorithm does?

mschuster91 11y ago

Nope. MP3, for example, cuts off all frequencies outside the audible range and thus achieves large savings.

GFK_of_xmaspast 11y ago
A 44.1Khz .wav file has no frequencies outside the audible range.
- joushou 11y ago
  How well do you hear 1Hz? A 44.1KHz doesn't have a lot of frequencies outside of the audible range, but that's not really that relevant, as that's only a minor trim compared to the crazy things lossy compression algorithms do here. Their main focus is having the result sound almost like the original. The audible sound-image is modified extensively by them, with focus on simplifications that are hard to notice for a human.

pcf 11y ago

AFAIK, every other audio compression encodes each slice of sound separately. My idea looks at the song as a whole, which I believe would be something new in the audio field.

GFK_of_xmaspast 11y ago
Why don't you start by taking a waveform, rip it from a CD or something, and start looking at the time-domain signal to see how many of these repeats there are.

drKarl 11y ago

That's roughly what compression algorithms (zip, rar, arj, etc) do, identify repeated bits of information and store where there are ocurrences of these. For lossy audio compression algorithms like mp3 and ogg, they use psychoacustic techniques, removing information the human ear can't hear, like very high or very low frequencies (that's why they're lossy). MP3 does something like slicing the signal in small fragments, applying some mathematical functions to them, like the Modified Discrete Cosine Transform (similar to Fourier Transform or FFT), then removing some components of it. The technique you're describing would pretty much be equivalent of using samples and sequence them, which would be akin to reinventing the .mod format (.mod, .s3m, .xm etc) To use samples though, you need the individual samples. To identify and extract the samples from the song would be much more difficult, unless maybe each instrument was in its own track or channel.