sorry, dropped a phrase by accident: shouldn't be too hard -- to use --.

On Sat, 6 Oct 2018 at 22:14, Scott Cotton <w...@iri-labs.com> wrote:

> The best open source one I know of is
> https://breakfastquay.com/rubberband/
>
> It is however very dense.  I wouldn't bet on coming to an understanding of
> how it does sample/window framing without significant investment.  The
> author himself said it was very hard to get sample accurate input samples
> to output samples ratios.
>
> However, to use it shouldn't be too hard.
>
> Scott
>
> On Sat, 6 Oct 2018 at 20:53, Alex Dashevski <alexd...@gmail.com> wrote:
>
>> Could you know where I can find phase vocoder implementaion in cpp thus I
>> can run it on real time ?
>>
>> ‫בתאריך שבת, 6 באוק׳ 2018 ב-21:21 מאת ‪Daniel Varela‬‏ <‪
>> danielvarela...@gmail.com‬‏>:‬
>>
>>> For real time you will need to do windowing and overlap add. But yeah,
>>> 5ms should be enough.
>>>
>>> This is a high level explanation with MATLAB
>>>
>>>
>>> https://se.mathworks.com/help/audio/examples/pitch-shifting-and-time-dilation-using-a-phase-vocoder-in-matlab.html
>>>
>>> El sáb., 6 oct. 2018 20:10, Alex Dashevski <alexd...@gmail.com>
>>> escribió:
>>>
>>>> Can you tell what minimum duration of buffer ? 5ms should be Ok ?
>>>>
>>>> ‫בתאריך שבת, 6 באוק׳ 2018 ב-21:06 מאת ‪Daniel Varela‬‏ <‪
>>>> danielvarela...@gmail.com‬‏>:‬
>>>>
>>>>> You can process buffers as small as your fft allows.
>>>>>
>>>>> El 6 oct. 2018 20:03, "Alex Dashevski" <alexd...@gmail.com> escribió:
>>>>>
>>>>> Hi,
>>>>> phase vocoder doesn't have restriction of duration ?
>>>>> Thanks,
>>>>> Alex
>>>>>
>>>>> ‫בתאריך שבת, 6 באוק׳ 2018 ב-20:55 מאת ‪Daniel Varela‬‏ <‪
>>>>> danielvarela...@gmail.com‬‏>:‬
>>>>>
>>>>>> You could try a phase vocoder instead of WSOLA for time stretching.
>>>>>> Latency would be the size of the fft block.
>>>>>>
>>>>>> El sáb., 6 oct. 2018 19:49, gm <g...@voxangelica.net> escribió:
>>>>>>
>>>>>>>
>>>>>>> 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"; moz-do-not-send="true">
>>>>>>> g...@voxangelica.netb
>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> 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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>>>>>
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>
>
>
> --
> Scott Cotton
> http://www.iri-labs.com
>
>
>

-- 
Scott Cotton
http://www.iri-labs.com
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