You can "freeze" audio with the phase vocoder "for ever" if that ist what you want to do.

You just keep the magnitude of the spectrum from one point in time and keep it

and update the phases with the phase differences of that moment.



Am 06.10.2018 um 20:02 schrieb Alex Dashevski:
Hi,
phase vocoder doesn't have restriction of duration ?
Thanks,
Alex

b> <b>
    You could try a phase vocoder instead of WSOLA for time
    stretching. Latency would be the size of the fft block.

    El sC!b., 6 oct. 2018 19:49, gm <g...@voxangelica.net
    <mailto:g...@voxangelica.net>> escribiC3:


        right

        the latency required is that you need to store the complete
        wavecycle, or two of them, to compare them

        (My method works a little bit different, so I only need one
        wavecycle.)

        So you always have this latency, regardless what sample rate
        you use.

        But maybe you dont need 20 Hz, for speech for instance I think
        that 100 or even 150 Hz is sufficient? I dont know



        Am 06.10.2018 um 19:34 schrieb Alex Dashevski:
        If I understand correctly, resampling will not help. Right ?
        No other technique that will help. Right ?
        What do you mean "but not the duration/latency required" ?

        b href="mailto:g...@voxangelica.net
        <mailto:g...@voxangelica.net>"
        moz-do-not-send="true">g...@voxangelica.net
        <mailto:g...@voxangelica.net>b



            Am 06.10.2018 um 19:07 schrieb Alex Dashevski:
            > What do you mean "replay" ? duplicate buffer ?

            I mean to just read the buffer for the output.
            So in my example you play back 10 ms audio (windowed of
            course), then
            you move your read pointer and play
            that audio back again, and so on, untill the next "slice"
            or "grain" or
            "snippet" of audio is played back.

            > I have the opposite problem. My original buffer size
            doesn't contain
            > full cycle of the pitch.

            then your pitch is too low or your buffer too small -
            there is no way
            around this, it's physics / causality.
            You can decrease the number of samples of the buffer with
            a lower sample
            rate,
            but not the duration/latency required.

            > How can I succeed to shift pitch ?

            You wrote you can have a latency of < 100ms, but 100ms
            should be
            sufficient for this.



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