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I want to make a voice changer, found the description of some effects, but I don't exactly understand what the digital audio looks like.

How much information can I get from different audio formats, or input audio streams?

Can I get frequency, pitch and what the channels are?

Should I apply filters and stuff to original track, or to spectral representation via Fourier transform?

Where can I get that kind of information?

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  • I don't get it. Not even the first sentence. I though by the title this would be some sort of philosophical debate!
    – n00dles ♦
    Commented Feb 22, 2017 at 16:39
  • @MarcW I just want to know what information can I get from an audio file.
    – Barsik the Cat
    Commented Feb 23, 2017 at 11:47
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    Digital audio is simply digital data representing the exact voltage level of the audio signal at the sampling point. Digital audio has a sampling rate which determines how many times per second the audio signal is "sampled". I suggest you read up on digital audio sampling theory and then read up on file formats and audio interfaces such as "portaudio". This will give you a better idea of how to deal with audio in a processing context.
    – Mark
    Commented Feb 23, 2017 at 11:49
  • All you can get from an audio file is the samples. check out libsndfile documentation - this will give you a useful interface to most sound file formats and allow you to access and manipulate the audio.
    – Mark
    Commented Feb 23, 2017 at 11:50
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    That's an ambitious goal :) Good luck!
    – Linuxios
    Commented Feb 24, 2017 at 15:56

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There are many audio formats, but generally, the file headers and format chunk can quite easily give you some specific format information if that's all you need; things like the amount of channels, samples per second, bits per sample, etc. Frequency and other data information is locked up in the data, which would need to be interpreted.

A MediaInfo report of a .wav file, most of which is reported by the header and format chunks: enter image description here

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