Re: [music-dsp] Resampling

2018-10-07 Thread Scott Cotton
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

Re: [music-dsp] Resampling

2018-10-07 Thread Alex Dashevski
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

Re: [music-dsp] Resampling

2018-10-06 Thread Scott Cotton
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

Re: [music-dsp] Resampling

2018-10-06 Thread Scott Cotton
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

Re: [music-dsp] Resampling

2018-10-06 Thread Alex Dashevski
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

Re: [music-dsp] Resampling

2018-10-06 Thread gm
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,

Re: [music-dsp] Resampling

2018-10-06 Thread Alex Dashevski
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.

Re: [music-dsp] Resampling

2018-10-06 Thread Daniel Varela
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

Re: [music-dsp] Resampling

2018-10-06 Thread gm
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

Re: [music-dsp] Resampling

2018-10-06 Thread Ethan Duni
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

Re: [music-dsp] Resampling

2018-10-06 Thread 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" ? ‫בתאריך שבת, 6 באוק׳ 2018 ב-20:29 מאת ‪gm‬‏ <‪g...@voxangelica.net‬‏>:‬ > > > Am 06.10.2018 um 19:07 schrieb Alex Dashevski: > > What

Re: [music-dsp] Resampling

2018-10-06 Thread gm
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

Re: [music-dsp] Resampling

2018-10-06 Thread Alex Dashevski
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

Re: [music-dsp] Resampling

2018-10-06 Thread gm
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

Re: [music-dsp] Resampling

2018-10-06 Thread Alex Dashevski
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

Re: [music-dsp] Resampling

2018-10-06 Thread gm
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

Re: [music-dsp] Resampling

2018-10-06 Thread Alex Dashevski
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

Re: [music-dsp] Resampling

2018-10-06 Thread gm
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

Re: [music-dsp] Resampling

2018-10-06 Thread Alex Dashevski
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 (<

Re: [music-dsp] Resampling

2018-10-06 Thread Ethan Fenn
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

Re: [music-dsp] Resampling

2018-10-06 Thread Alex Dashevski
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.

Re: [music-dsp] Resampling

2018-10-03 Thread Spencer Jackson
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

Re: [music-dsp] Resampling

2018-10-03 Thread Alex Dashevski
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

Re: [music-dsp] Resampling

2018-10-03 Thread Spencer Jackson
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

Re: [music-dsp] resampling

2018-07-25 Thread Tom O'Hara
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

Re: [music-dsp] resampling

2018-07-25 Thread rolfsassinger
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…)

Re: [music-dsp] resampling

2018-07-24 Thread Nigel Redmon
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

Re: [music-dsp] resampling

2018-07-24 Thread Nigel Redmon
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.

Re: [music-dsp] resampling

2018-07-24 Thread Tom O'Hara
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,

Re: [music-dsp] resampling

2018-07-24 Thread rolfsassinger
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

Re: [music-dsp] resampling

2018-07-24 Thread Alex Dashevski
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

Re: [music-dsp] resampling

2018-07-23 Thread Tom O'Hara
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

Re: [music-dsp] resampling

2018-07-23 Thread Nigel Redmon
Some articles on my website: http://www.earlevel.com/main/category/digital-audio/sample-rate-conversion/ , especially the 2010 articles, but the Amp Sim article might be a helpful overview. 48k -> 8k: Filter with a

Re: [music-dsp] resampling

2018-07-23 Thread Felix Homann
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 ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu

Re: [music-dsp] resampling

2018-07-23 Thread Kjetil Matheussen
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: >

Re: [music-dsp] resampling

2018-07-23 Thread Kjetil Matheussen
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

Re: [music-dsp] resampling

2018-07-22 Thread Sound of L.A. Music and Audio
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

Re: [music-dsp] resampling

2018-07-22 Thread Henrik G. Sundt
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

Re: [music-dsp] resampling

2018-07-22 Thread Alex Dashevski
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++;

Re: [music-dsp] resampling

2018-07-22 Thread Kjetil Matheussen
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 =

Re: [music-dsp] resampling

2018-07-22 Thread Alex Dashevski
real time On Sun, Jul 22, 2018, 22:52 jpff wrote: > Were you expecting real-time/time-critical resampling or offline? > > ___ > dupswapdrop: music-dsp mailing list > music-dsp@music.columbia.edu > https://lists.columbia.edu/mailman/listinfo/music-dsp

Re: [music-dsp] resampling

2018-07-22 Thread jpff
Were you expecting real-time/time-critical resampling or offline? ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp

Re: [music-dsp] resampling

2018-07-22 Thread Alex Dashevski
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 : >

Re: [music-dsp] resampling

2018-07-22 Thread 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:

Re: [music-dsp] resampling

2018-07-22 Thread Alex Dashevski
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: > > https://ccrma.stanford.edu/~jos/resample/ > > > Using Google, I found somebody who used the LGPL code available at Julius' >

Re: [music-dsp] resampling

2018-07-19 Thread Esteban Maestre
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