[Libav-user] Calculate spectrogram from the audio channel

[J Decker]

On Mon, May 5, 2014 at 12:25 PM, Ricky Huang <rhuang.work at gmail.com> wrote:

> On May 2, 2014, at 6:27 PM, J Decker <d3ck0r at gmail.com> 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?
>
>
No; I guess there is some sort of filter available for that; but I thought
that was what you wanted to do ... that is apply your FFT algorithm to
data, and just needed the data.


> Thank again.
>
>
>
>
> On Fri, May 2, 2014 at 5:48 PM, Ricky Huang <rhuang.work at gmail.com> 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.
>>
>>
>>
>>
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>>
>>
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