I have a group of related questions regarding FFTW and audio analysis on Linux.
- What is the easiest-to-use, most comprehensive audio library in Linux/Ubuntu that will allow me to decode any of a variety of audio formats (MP3, etc.) and acquire a buffer of raw 16-bit PCM values?
gstreamer
? - I intend on taking that raw buffer and feeding it to FFTW to acquire frequency-domain data (without complex information or phase information). I think I should use one of their "r2r" methods, probably the DHT. Is this correct?
- It seems that FFTW's output frequency axis is discretized in linear increments that are based on the buffer length. It further seems that I can't change this discretization within FFTW so I must do it after the DHT. Instead of a linear frequency axis, I need an exponential axis that follows
2^(i/12)
. I think I'll have to take the DHT output and run it through some custom anti-aliasing function. Is there a Linux library to do such anti-aliasing? If not, would a basic cosine-based anti-aliasing function work?
Thanks.