A signal picked up by a microphone might look like this.Īs you all know, we live in an analog world and that means audio signals are of course continuous, i.e. The peak voltage of that signal will depend on the properties of the microphone in question and the amplitude of the audio signal. These vibrations are transferred to a permanent magnet, wich will move inside of a coil and thus generate an analog signal representing the audio waves. A microphone has a membrane which will vibrate in the same frequency as the audio signal it is picking up. So, how does a microphone turn differences in air pressure into something a computer can work with? It is actually pretty easy, at least if you know the principle of electromagnetic induction. These differences form waves that are translated into electrical signals by our ear or a microphone. anything you or I can hear, is basically just differences in air pressure. The code can be used as a whole or partially as per the terms of the MIT License allows. However I still recommend that you at least read the following sections, as they explain the reasons for certain decisions. The code is well commented and any seasoned programmer should be able to work with it.

The code can be found in this GitHub gist or you can download the archive here.

The footnotes can be found at the bottom of the page, the links to them might not work, as they are not supported by Steemit.

Feel free to give me feedback and discuss things either here or my website. I initially wrote it for my blog page but I thought, why not get some more exposure by posting it here.įrom now on I will release blog posts both on my blog site and here on Steemit. In this tutorial we will put together some C code that will be able to decode audio files using ffmpeg version 3.1.ĭisclaimer: This post was not originally written for Steemit.

Ffmpeg decoder