when you slow the resynth signal down (time stretch)
you will eventually run out of buffer memory.
In this case I reduce the lag of the signal (the time stretch) to 0
within a certain time, which is determined by the bpm of the host and
a given time measured in beats, for instance, 1 quarter or one 1/16.
This gives an inverse time stretch (time compression) which is similar
effekt to a tape speeding up, but with a fixed pitch.
This is an implementation detail which is not relevant to your basic
question,
and it's probabaly not the best solution either.
Your question was, do you need resampling, and my answer to that is
you do *not* need *proper* resampling with filtering etc, you just play
your grains at different speeds like in a sampler, so you need interpolation
between samples. HTH
I found a very efficient implementation for WSOLA, that is,
for the similarity problem, which works without correlation in my case,
and is very well suited for real time implemantation,
but unfortunately I can not discuss it in detail at the moment.
Am 27.09.2018 um 15:58 schrieb alex dashevski:
Hi,
I don't understand what do you mean. Could you explain ?
2018-09-27 16:17 GMT+03:00 gm <g...@voxangelica.net
<mailto:g...@voxangelica.net>>:
I had different solution, where the lag is reset to zero during a
musical period.
Kind of a tape speed-up effekt without the pitch change.
Not always useful though.
Am 26.09.2018 um 23:25 schrieb Jacob Penn:
Ahh yeah I gotcha,B
Yes, in the case of slow down, there Is a finite amount youb
to slow down based on the size of the circular buffer of input
data in use.
In my personal applications I offer users the ability to
restart the stretch from the writehead at a musical value.
Conveniently the slowest rate for this control will stop the
overflow ; )
Can sound quite nice!
Best,B
insignia <http://jakemumu.github.io/>
JACOB PENN.MUMUKSHU
612.388.5992
On September 26, 2018 at 2:21:29 PM, robert bristow-johnson
(r...@audioimagination.com <mailto:r...@audioimagination.com>)
wrote:
---------------------------- Original Message
----------------------------
Subject: Re: [music-dsp] WSOLA on RealTime
From: "Jacob Penn" <penn.ja...@gmail.com
<mailto:penn.ja...@gmail.com>>
Date: Wed, September 26, 2018 5:00 pm
To: r...@audioimagination.com <mailto:r...@audioimagination.com>
music-dsp@music.columbia.edu
<mailto:music-dsp@music.columbia.edu>
--------------------------------------------------------------------------
> You can indeed do it on real time audio but the tricks
is like the previous
> email, youb pitching things up, as youb > be lacking the
necessary information to move faster across the buffer from
> the write head position.
>
> Youb signal, and not speed it
> up.
no, even if you slow it down, any finite-sized buffer will
eventually overflow.B i presume you mean time-scaling (not
pitch shifting) using WSOLA.
by "real-time", i mean live samples going in and (assuming
no sample rate conversion) the same number of samples
going out in a given period of time.B with an upper bound
of delay (and the lower bound is imposed by causality) and
the process can run indefinitely.B so if you're slowing
down audio in real-time and you're running this process
for a day.B or for a year.
--
r b-jB B B B B B B B B B B B B r...@audioimagination.com
<mailto:r...@audioimagination.com>
"Imagination is more important than knowledge."
B
B
B
B
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