FFTW for exponential frequency axis

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 se开发者_如何学Goems 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.

This is an age old problem with FFTs and working with audio - ideally we want a log frequency scale for audio but the DFT/FFT has a linear scale. You will need to choose an FFT size that gives sufficient resolution at the low end of your frequency range, and then accumulate bins across the frequency range of interest to give yourself a pseudo-logarithmic representation. There are more complex schemes, but essentially it all boils down to the same thing.

I've seen libsndfile used all over the place:

http://www.mega-nerd.com/libsndfile/

It's LGPL too. It can read pretty much all the open source and lossless audio format you would care about. It doesn't do MP3, however, because of licensing costs.