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Howdy!
I've been assigned a project (for the course I'm attending) to create a program that compresses wav files to mp3. I have no background in digital signal processing (or the inner workings of audio formats for that matter), and don't have much of an idea where to start. I've looked up the very basics of the wav format but any attempt to find more about audio compression hits me with terms like 'psychoacoustics' and 'modified discrete cosine transform'. Could someone suggest how to go about this? Will it require learning much about Digital Sig. Processing? If so, could you suggest a source that describes what I need to know without going into too much detail? I'll be using C/C++. Thanks. |
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MP3 Format Information (scroll down a bit)
WAV Format Information (near bottom) There's C source code for decoding MP3 files in the first link, near the middle. |
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Look up LAME. It's open-source and encodes to MP3.
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LAME is open-source, so he could look at how the encoding was performed.
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| Tags |
| audio compression, audio files, compression algorithm, mp3, wav |
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