Hi,
The current code has problem.
if I send you the current code, can you help me with implementation ? Are
you familiar with android ?

Thanks,
Alex

2018-09-27 20:30 GMT+03:00 Spencer Jackson <ssjackso...@gmail.com>:

> Alex:
>
> Any algorithm with a deterministic computation time can be applied in
> real-time if latency is sufficient enough. Are you familiar with the idea
> of copying the stream to a buffer then applying the algorithm and
> outputting the older data which the algorithm has already been applied to?
> Adding latency like this and processing in blocks is the standard method
> that allows very cpu heavy processing to be applied to a real-time stream.
> The only real requirement for real-time processes is to output the same
> number of samples as the input (at the same sample rate).
>
> Pitch shifting methods stretch or shrink the signal creating more or less
> samples and you must figure out how to remove the extra or add samples to
> be the correct number of samples in order for it to work on a real-time
> audio stream. This is the resampling part and there are many ways to
> address it.
>
> I think we might be able to help more if you can ask more specific
> questions about the implementation details you are struggling with.
>
> _Spencer
>
> On Thu, Sep 27, 2018 at 12:16 AM Alex Dashevski <alexd...@gmail.com>
> wrote:
>
>> Hi Robert,
>>
>> I don't understand your last sentence:
>> "then, in real-time, you can use the time
>> scaler  make the more or fewer samples per block to be the same as
>> normal."
>>
>> In the beginning of your answer, you said that I need to use WSOLA with
>> resampling.
>>
>> I have problem with implementation. Can you help ?
>>
>> Thanks,
>> Alex
>>
>> 2018-09-27 0:25 GMT+03:00 Jacob Penn <penn.ja...@gmail.com>:
>>
>>> Ahh yeah I gotcha,
>>>
>>> Yes, in the case of slow down, there Is a finite amount you’re able 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,
>>>
>>> [image: 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) wrote:
>>>
>>>
>>>
>>> ---------------------------- Original Message
>>> ----------------------------
>>> Subject: Re: [music-dsp] WSOLA on RealTime
>>> From: "Jacob Penn" <penn.ja...@gmail.com>
>>> Date: Wed, September 26, 2018 5:00 pm
>>> To: r...@audioimagination.com
>>> music-dsp@music.columbia.edu
>>> ------------------------------------------------------------
>>> --------------
>>>
>>> > You can indeed do it on real time audio but the tricks is like the
>>> previous
>>> > email, you’ll need to devise strategies for pitching things up, as
>>> you’ll
>>> > be lacking the necessary information to move faster across the buffer
>>> from
>>> > the write head position.
>>> >
>>> > You’ll also obviously only be able to slow down a signal, and not
>>> speed it
>>> > up.
>>>
>>> no, even if you slow it down, any finite-sized buffer will eventually
>>> overflow.  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.  with an upper bound of delay (and the lower bound is imposed by
>>> causality) and the process can run indefinitely.  so if you're slowing down
>>> audio in real-time and you're running this process for a day.  or for a
>>> year.
>>>
>>>
>>> --
>>>
>>> r b-j                         r...@audioimagination.com
>>>
>>> "Imagination is more important than knowledge."
>>>
>>>
>>>
>>>
>>>
>>>
>>>
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>>
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