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Re: Measure Frequency

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  • jotape1960
    ... looking ... the ... which ... have ... Hi Richard!!! I m not a programmer but I love all about the computers and, specially the sound issues, because I m a
    Message 1 of 9 , Jul 31, 2007
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      --- In libertybasic@yahoogroups.com, "Richard Russell" <yahoo@...>
      wrote:
      >
      > --- In libertybasic@yahoogroups.com, "rod_bird001" wrote:
      > > Keep track of the directional changes and hold an array to record
      > > the "frequency" ie time to change direction, and "amplitude" ie
      > > how far did it move up or down. Keep track of a second or third
      > > level minimum and maximum to record the underlying tones over and
      > > above the harmonics.
      >
      > In practice it's very hard to determine anything other than the
      > fundamental frequency that way. Instead, one conventionally
      > performs a Fourier Transform to convert the audio waveform (time
      > domain) to its spectrum (frequency domain). Having done that,
      > discovering what frequencies are present is simply a case of
      looking
      > for peaks in the spectrum.
      >
      > At first glance performing a Fourier Transform sounds very
      > difficult, and it would be if you were to try to code it in BASIC,
      > but fortunately there are DLLs available to do the job. One of
      the
      > best known is the Fastest Fourier Transform in the West (FFTW)
      which
      > you can get as a free pre-compiled DLL for Windows:
      >
      > http://www.fftw.org/install/windows.html
      >
      > Although I've not personally called FFTW from Liberty Basic, I
      have
      > from another dialect of BASIC and it was quite straightforward.
      >
      > Richard.
      >

      Hi Richard!!!

      I'm not a programmer but I love all about the computers and,
      specially the sound issues, because I'm a music teacher.

      I understand your points about the audio representation. In fact, I
      worked on it but not with Liberty Basic. So I have a great doubt
      about it: all the samples you're talking about are 8 bits per
      sample, one audio channel *.wav files, but everybody knows the
      standard audio files are 16 bits per sample, two or more channels
      (stereo or some surround system). Now, supposing all the people in
      the forum knows about how to work with the 16 bits/more than one
      audio channel files, my doubt is: IS LIBERTYBASIC ENOUGH FASTER TO
      PROCESS THE AUDIO DATA STREAM TO REPRESENT THE GRAPHIC AUDIO WAVE IN
      REAL TIME?

      Thanks for your time!!!

      GOD BLESS YOU ALL!!!!!!!


      Juan J. Paredes G.
      From Curicó, Chile, South America, with love
    • rod_bird001
      ... I accept that FFT will provide precision and the .dll link looks useful. However you can still have a lot of fun trying to analyse it yourself.
      Message 2 of 9 , Aug 1, 2007
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        > > In practice it's very hard to determine anything other than the
        > > fundamental frequency that way. Instead, one conventionally
        > > performs a Fourier Transform to convert the audio waveform (time
        > > domain) to its spectrum (frequency domain).

        I accept that FFT will provide precision and the .dll link looks
        useful. However you can still have a lot of fun trying to analyse it
        yourself. http://www.lbdownloads.com/files/viewtopic.php?t=84 This
        simply tracks the changes of direction and gives you a spectrum
        spread of frequency and number of occurrences of that frequency. Full
        of bugs and and limited to 8bit samples but it kinda works.

        > the samples you're talking about are 8 bits per
        > sample, one audio channel *.wav files, but everybody knows the
        > standard audio files are 16 bits per sample, two or more channels
        > (stereo or some surround system).

        It is relatively straight forward to move from 8bit you just need to
        deal with bigger numbers in the format that .wav files store stereo
        and surround sound. usually each channel is listed in sequence. 16bit
        just uses two bytes instead of one so you need to uprate the analysis
        code.

        Real time? well that would require a real time audio stream. What we
        are talking about so far is analysis of recorded .wav files.

        Anyways the point I make is it isn't impossible its just numbers and
        Liberty BASIC is vastly adept at handling numbers.
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