On May 2, 2014, at 6:27 PM, J Decker <[email protected]> wrote: > wouldn't have to be a custom filter, just decompress the media file the > normal way..... could actually probably just use the ffmpeg command line tool > to strip the audio and save it as raw samples, then just read the audio file > directly…
Thank you for the reply. Can you clarify it a bit for me: does saving the audio as raw samples using ffmpeg perform the FFT necessary to convert audio to frequency-along-time output? Thank again. > > > On Fri, May 2, 2014 at 5:48 PM, Ricky Huang <[email protected]> wrote: > Hello all, > > I am trying to reproduce the Shazam algorithm as outlined in Avery Wang's > paper "An Industrial-Strength Audio Search Algorithm" > (http://www.ee.columbia.edu/~dpwe/papers/Wang03-shazam.pdf). One of the step > in this is to convert the audio to spectrogram and identify the spectrogram > peaks. I am wondering if building a custom audio-filter for ffmpeg would be > the correct way to go? If so, does anyone have any pointers on converting > the audio data to spectrogram for me? (algorithm to use, things to note, > etc?) > > > Any help would be appreciated. Thanks. > > > > > _______________________________________________ > Libav-user mailing list > [email protected] > http://ffmpeg.org/mailman/listinfo/libav-user > > > _______________________________________________ > Libav-user mailing list > [email protected] > http://ffmpeg.org/mailman/listinfo/libav-user
_______________________________________________ Libav-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/libav-user
