On Sun, 7 Oct 2018 at 08:20, Alex Dashevski wrote:
> is it phase vocoder ?
>
I am not the author, so take with a grain of salt: Yes, but it treats the
input to output sample ratio
differently than say a standard pv in matlab.
> I can't understand how it work.
>
welcome to the world of
is it phase vocoder ?
I can't understand how it work.
בתאריך שבת, 6 באוק׳ 2018 ב-23:15 מאת Scott Cotton <w...@iri-labs.com
>:
> sorry, dropped a phrase by accident: shouldn't be too hard -- to use --.
>
> On Sat, 6 Oct 2018 at 22:14, Scott Cotton wrote:
>
>> The best open source one I
sorry, dropped a phrase by accident: shouldn't be too hard -- to use --.
On Sat, 6 Oct 2018 at 22:14, Scott Cotton 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
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
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
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,
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.
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 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
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
Alex, it sounds like you are confusing algorithmic latency with framing
latency. At each frame, you take in 10ms (or whatever) of input, and then
provide 10ms of output. This (plus processing time to generate the output) is
the IO latency of the process. But the algorithm itself can add
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" ?
בתאריך שבת, 6 באוק׳ 2018 ב-20:29 מאת gm <g...@voxangelica.net>:
>
>
> Am 06.10.2018 um 19:07 schrieb Alex Dashevski:
> > What
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
What do you mean "replay" ? duplicate buffer ?
I have the opposite problem. My original buffer size doesn't contain full
cycle of the pitch.
How can I succeed to shift pitch ?
Thanks,
Alex
בתאריך שבת, 6 באוק׳ 2018 ב-19:55 מאת gm <g...@voxangelica.net>:
>
> no, you don't change the
no, you don't change the buffer size, you just change the playback rate
(and speed, if you want) of your grains.
For instance, lets say the pitch is 20 Hz, or 50 ms time for one cycle.
You want to change that to 100 Hz.
Then you take 50 ms of audio, and replay this 5 times every 10 ms (with
I still don't understand. You change buffer size. Right ?
But I don't want to change.
בתאריך שבת, 6 באוק׳ 2018 ב-19:11 מאת gm <g...@voxangelica.net>:
>
> In my example, the buffer is 2 times as long as the lowest possible pitch,
> for example if your lowest pitch is 20 Hz, you need 50 ms
In my example, the buffer is 2 times as long as the lowest possible pitch,
for example if your lowest pitch is 20 Hz, you need 50 ms for one wave cycle
Think of it as magnetic tape, without sample rate, the minimum requierd
latency and the buffer length in milliesconds
are independent of
Hi,
I can't understand your answer. The duration of buffer should be bigger
than duration of pitch because I use WSOLA.
The latency also depends on sample rate and buffer length.
Thanks,
Alex
בתאריך שבת, 6 באוק׳ 2018 ב-18:26 מאת gm <g...@voxangelica.net>:
> Your numbers don't make
Your numbers don't make sense to me but probably I just dont understand it.
The latency should be independent of the sample rate, right?
You search for similarity in the wave, chop it up, and replay the grains
at different speeds and/or rates.
What you need for this is a certain amount of
I have project with pitch shifting (resampling with wsola), It implements
on android NDK.
Since duration of pitch is ~20ms, I can't use system recommended
parameters for the fast path. for example, for my device: SampleRate:48Khz
and buffer size 240 samples. That means, duration time is 5ms (<
You've got it backwards -- downsample means fewer samples. If you have a
240-sample buffer at 48kHz, then resample to 8kHz, you'll have 240/6=40
samples.
-Ethan
On Sat, Oct 6, 2018 at 4:10 AM, Alex Dashevski wrote:
> Hi,
> Let's assume that my system has sample rate = 48Khz and audio buffer
Hi,
Let's assume that my system has sample rate = 48Khz and audio buffer size =
240 samples. It should be on RealTime.
Can I do that:
1. Dowsampe to 8Khz and buffer size should be 240*6
2. To do proccessing on buffer 240*6 with 8Khz sample rate.
3. Upsample to 48khz with original buffer size.
I have only used libraries for resampling myself. I haven't looked at their
source, but it's available. The two libraries I'm aware of are at
http://www.mega-nerd.com/SRC/download.html
and
https://kokkinizita.linuxaudio.org/linuxaudio/zita-resampler/resampler.html
perhaps they can give you some
I wrote on android ndk and there is fastpath concept. Thus, I think that
resampling can help me.
Can you recommend me code example ?
Can you give me an example of resampling ? for example from 48Khz to 8Khz
and 8Khz to 48Khz.
I found this:
https://dspguru.com/dsp/faqs/multirate/resampling/
but it
On Wed, Oct 3, 2018 at 3:17 AM Alex Dashevski wrote:
>
> if I do resampling before and after processing. for example, 48Khz -> 8Khz
> and then 8Khz -> 48Khz then will it help ?
>
Lowering sample rate can help achieve lower latencies by giving you fewer
samples to process in the same amount of
Hi,
I use a sample rate :48Khz and buffer size = 240 samples.
I made pitch shifting with WSOLA and resampling.
But pitch duration is ~20ms then I need decrease rate sample or increase
buffer size. As a result of it, the delay will increase.
if I do resampling before and after processing. for
On 7/26/2018 2:27 AM, rolfsassin...@web.de wrote:
Regarding Tom's remark: Using the copied samples also requires no
additional multiplcation since the value is already stored and in use (?)
No, they require multiplication and addition as, while the samples are
the same, each coefficient is
multiplcation since the value is already stored and in use (?)
Anyway thanks.
Rolf
Gesendet: Dienstag, 24. Juli 2018 um 18:36 Uhr
Von: "Nigel Redmon"
An: music-dsp@music.columbia.edu
Betreff: Re: [music-dsp] resampling
(Not sure why I didn’t receive Rolf’s email directly…)
eeding zeros needs more
>>> filter TAPs to come to the same result.
>>>
>>> Rolf
>>>
>>>
>>> Gesendet: Montag, 23. Juli 2018 um 18:25 Uhr
>>> Von: "Nigel Redmon" <mailto:earle...@earlevel.com>
>>> An: music-dsp@musi
MATLAB and found that feeding zeros needs more
>> filter TAPs to come to the same result.
>>
>> Rolf
>>
>>
>> Gesendet: Montag, 23. Juli 2018 um 18:25 Uhr
>> Von: "Nigel Redmon" <mailto:earle...@earlevel.com>
>> An: music-dsp@music.columbia.
ult.
Rolf
*Gesendet:* Montag, 23. Juli 2018 um 18:25 Uhr
*Von:* "Nigel Redmon"
*An:* music-dsp@music.columbia.edu
*Betreff:* Re: [music-dsp] resampling
Some articles on my website:
http://www.earlevel.com/main/category/digital-audio/sample-rate-conversion/,
especially the 2010 articles,
d that feeding zeros needs more filter TAPs to come to the same result.
Rolf
Gesendet: Montag, 23. Juli 2018 um 18:25 Uhr
Von: "Nigel Redmon"
An: music-dsp@music.columbia.edu
Betreff: Re: [music-dsp] resampling
Some articles on my website: http://www.earlevel.com/main/cate
Hi,
I need to do resampling on android.
Could you give me code on c/c++/Java?
On Tue, Jul 24, 2018, 08:56 Tom O'Hara wrote:
> I've done many resamplers over the decades (48<->32, 24,16,8) and always
> used FIRs for these reasons.
>
> Tom
>
> On 7/23/2018 6:25 PM, Nigel Redmon wrote:
> > Some
I've done many resamplers over the decades (48<->32, 24,16,8) and always
used FIRs for these reasons.
Tom
On 7/23/2018 6:25 PM, Nigel Redmon wrote:
Some articles on my website:
http://www.earlevel.com/main/category/digital-audio/sample-rate-conversion/,
especially the 2010 articles, but the
libsamplerate, aka Secret Rabbit Code, has been relicensed under a 2 clause
BSD license a while ago. Maybe you want to give it a try:
https://github.com/erikd/libsamplerate
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On Mon, Jul 23, 2018 at 3:08 AM, Henrik G. Sundt wrote:
> This solution, without using any low pass filters before and after the
> desimation, will generate a lot of aliasing frequencies, Kjetil!
>
> Here is another solution:
>
On Mon, Jul 23, 2018 at 3:08 AM, Henrik G. Sundt wrote:
> This solution, without using any low pass filters before and after the
> desimation, will generate a lot of aliasing frequencies, Kjetil!
>
>
No doubt. I did write "Not the best sound quality though." :-)
Alex didn't write about his
This code is also dangerous "LGPL" :-)
Seriously, I'm afraid this is also too much for him. Code is not really
good to explain solutions. I prefer the clarification and let people
code themselves.
Let's try it this way:
1. Apply an anti aliasing filter with an edge frequency of about
This solution, without using any low pass filters before and after the
desimation, will generate a lot of aliasing frequencies, Kjetil!
Here is another solution:
https://github.com/intervigilium/libresample/tree/master/jni/resample
Henrik
On 22.07.2018 22:22, Kjetil Matheussen wrote:
Maybe
where is low pass filter?
On Sun, Jul 22, 2018, 23:22 Kjetil Matheussen
wrote:
> Maybe this will give you an idea:
>
> 48khz -> 8khz:
> float get_output_sample(get_input_sample){
>static int i=0;
>static float sample;
>
> if (i % 6 == 0)
> sample = get_input_sample();
>
> i++;
Maybe this will give you an idea:
48khz -> 8khz:
float get_output_sample(get_input_sample){
static int i=0;
static float sample;
if (i % 6 == 0)
sample = get_input_sample();
i++;
return sample;
}
8khz -> 48khz:
float get_output_sample(get_input_sample){
float ret =
real time
On Sun, Jul 22, 2018, 22:52 jpff wrote:
> Were you expecting real-time/time-critical resampling or offline?
>
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Were you expecting real-time/time-critical resampling or offline?
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This is even more incomprehensible.
I'm looking for a simple example of code and explanation how to convert
signal of 48Khz freq samples to 8Kh ,do processing of signal and convert
8Khz to 48Khz freq samples.
Thanks,
Alex
2018-07-22 22:28 GMT+03:00 Vladimir Pantelic :
>
https://en.wikipedia.org/wiki/GNU_Lesser_General_Public_License
On Sun, Jul 22, 2018, 21:23 Alex Dashevski wrote:
> Hi,
> Could you explain how to use with LGPL ? I can't understand it.
> Thanks,
> Alex
>
> 2018-07-19 21:28 GMT+03:00 Esteban Maestre :
>
>> Hi Alex,
>>
>>
>> This is a good read:
Hi Alex,
This is a good read:
https://ccrma.stanford.edu/~jos/resample/
Using Google, I found somebody who used the LGPL code available at
Julius' site:
https://github.com/intervigilium/libresample
Good luck!
Esteban
On 7/19/2018 2:15 PM, Alex Dashevski wrote:
Hi,
I need to convert
Hi,
I need to convert 48Khz to 8KHz on input and convert 8Khz to 48Khz on audio
on output.
Could you explain how to do it ?
I need to implement this on android(NDK).
Thanks,
Alex
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