title: Can non-standard sampling rates be used with AAC encoding?
body: <p>I am trying to up-sampling an M4a file from 41000 to another non-standard sampling rate let's say 5000.</p>
  
  <pre><code>ffmpeg -i audio.wav -ar 5000 audio_.wav
  </code></pre>
  
  <p>This worked fine with wav files whoever it didn't work with m4a. Any ideas why?</p>
  
  <p>If non-standard sampling rates don't work with AAC I need any documentation or reference for that.</p>