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> <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.



    _______________________________________________
    dupswapdrop: music-dsp mailing list
    music-dsp@music.columbia.edu <mailto:music-dsp@music.columbia.edu>
    https://lists.columbia.edu/mailman/listinfo/music-dsp


_______________________________________________
dupswapdrop: music-dsp mailing list
music-dsp@music.columbia.edu
https://lists.columbia.edu/mailman/listinfo/music-dsp

Reply via email to