Re: [music-dsp] resampling

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: > https://github.com/intervigilium/libresample/tree/master/jni/resample

Re: [music-dsp] resampling

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 knowle

Re: [music-dsp] resampling

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 = get_inpu

Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation

On Thu, Feb 25, 2016 at 5:08 PM, Ethan Fenn wrote: > When you ask if there is an implementation > of something available, you don't expect people to teach you programming > lessons and sprout about > how useful it would be for your personal development to implement things > yourself. > > Honestly

Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation

On Thu, Feb 25, 2016 at 3:44 PM, STEFFAN DIEDRICHSEN wrote: > > On 25.02.2016|KW8, at 03:43, Ross Bencina > wrote: > > I'm surprised it's apparently so uncommon to implement a > callback interface for providing samples when resampling. It's the > really the natural thing to do. > > > > In a C-en

Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation

On Thu, Feb 25, 2016 at 3:43 AM, Ross Bencina wrote: > On 23/02/2016 7:42 PM, Kjetil Matheussen wrote: > >> But that's why I ask, so I don't have to do the implementation. It >> seems like a common task that someone, probably many, have already >> done. &g

Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation

e in a non-opensource application may be a transgression of > the GPL license under which the original code is offered! (Unless you've > purchased a commercial license to JUCE) > > – Evan Balster > creator of imitone <http://imitone.com> > > On Wed, Feb 2

Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation

On Mon, Feb 22, 2016 at 1:52 PM, Kjetil Matheussen wrote: > Hi all, > > I wonder if anyone has a tip for a C or C++ of an implementation of a > Cubic interpolating resampler. > I'm not asking about the algorithm itself, that is all covered (currently > I'm using a

Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation

On Tue, Feb 23, 2016 at 9:42 AM, Kjetil Matheussen wrote: > > >> > P.P.S. How do you communicate the amount of source data needed? the >> callback has no "count" parameter. >> >> The callback just provides as many samples as it can. If the call

Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation

On Tue, Feb 23, 2016 at 12:45 AM, Evan Balster wrote: > Kjetil -- > > I suspect you'll have a very hard time finding a library with the > functions you need that doesn't include quite a lot else. > That's not necessarily a problem. And if it would be a problem, it might be simple just to rip out

Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation

On Tue, Feb 23, 2016 at 12:36 AM, Ross Bencina wrote: > On 23/02/2016 1:24 AM, Kjetil Matheussen wrote: > >> On Mon, Feb 22, 2016 at 2:59 PM, Ross Bencina >> mailto:rossb-li...@audiomulch.com>> wrote: >> >> Hi Kjetil, >> >> On 22/02/2016 11

Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation

On Mon, Feb 22, 2016 at 8:40 PM, robert bristow-johnson < r...@audioimagination.com> wrote: > > > > Pulling in a library for just this seems like massive overkill. The > > venerable pink elephant paper > > describes > > several cubic resa

Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation

On Mon, Feb 22, 2016 at 7:02 PM, Evan Balster wrote: > Hey, Kjetil -- > > Pulling in a library for just this seems like massive overkill. The > venerable pink elephant paper > describes > several cubic resampling algorithms (I prefer fo

Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation

On Mon, Feb 22, 2016 at 5:38 PM, Kjetil Matheussen wrote: > > Thank you, but I found something else. > > It seems like JUCE actually have a resampling class that seems to do the > right thing: > > https://github.com/julianstorer/JUCE/blob/master/modules/juce

Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation

Thank you, but I found something else. It seems like JUCE actually have a resampling class that seems to do the right thing: https://github.com/julianstorer/JUCE/blob/master/modules/juce_audio_basics/sources/juce_ResamplingAudioSource.cpp I don't know what type of resampler this is, but it should

Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation

On Mon, Feb 22, 2016 at 3:24 PM, Kjetil Matheussen wrote: > * Interleaved/non-interleaved input? output? always interpolate all >> channels? >> >> > Interleaved, always interpolate all channels. > > That was wrong. It's always just one ch

Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation

On Mon, Feb 22, 2016 at 2:59 PM, Ross Bencina wrote: > Hi Kjetil, > > On 22/02/2016 11:52 PM, Kjetil Matheussen wrote: > >> I wonder if anyone has a tip for a C or C++ of an implementation of a >> Cubic interpolating resampler. I'm not asking about the algorithm &

[music-dsp] Tip for an audio resampler library doing cubic interpolation

Hi all, I wonder if anyone has a tip for a C or C++ of an implementation of a Cubic interpolating resampler. I'm not asking about the algorithm itself, that is all covered (currently I'm using a Catmull-Rom spline algorithm, but that is not so important here). What I'm asking about is a framework

Re: [music-dsp] Fast exp2() approximation?

On Tue, Sep 2, 2014 at 11:01 PM, Kjetil Matheussen wrote: > > Paul Mineiro have some nice code which uses some dirty, but fully working, > tricks: > > http://fastapprox.googlecode.com/svn/trunk/fastapprox/src/fastonebigheader.h > > > > but maybe > there are some o

Re: [music-dsp] Fast exp2() approximation?

On Tue, Sep 2, 2014 at 10:10 PM, Paul Stoffregen wrote: > I'm hoping to find a fast approximation for exp2(), which I can implement > in 32 bit fixed point. So far, the best I've turned up by searching is > this from the archives. > > http://www.musicdsp.org/showone.php?id=106 > > n = in

Re: [music-dsp] [admin] Digest Latency

Hi, I saw the same behavior in digest mode as Frank does. (I turned off digest mode because of this though) On 6/12/14, Douglas Repetto wrote: > Hi Frank, > > The list is currently set to send out a digest once there's 100k worth of > messages OR once a day (assuming there's at least one messa

Re: [music-dsp] music-dsp mailing list digest settings

Douglas Repetto: > > The digest doesn't go by number of messages, it goes by size. It was > set at 10k to trigger a new digest, I just upped it to 10k. Let me > know how that feels. > Thank you! That's much better, now there's 17 mails in the latest digest. -- dupswapdrop -- the music-dsp mailing

[music-dsp] music-dsp mailing list digest settings

Hi Douglas and others, I wonder if it would be possible to increase the number of emails contained in the digest mails we receive? We (well, at least I) only get 2 or 3 emails, which is so little that my mail box would be less messy if I didn't had subscribed to the digest version, since then I c

Re: [music-dsp] Segmentation fault: 11 when running stereo to mono .c file

Pablo Frank: > When running the file below, the message Segmentation fault: 11 is displayed > and an empty > stereo file is created. > Can someone show the problem, or better, correct the code ? (I'm a beginner). > Thanks in advance Why do you ask the same question twice? j...@cs.bath.ac.uk:

Re: [music-dsp] unable to open an aiff file

Pablo Frank: > /* memory for SF_INFO structures */ > psfinfo_in = (SF_INFO *) malloc(sizeof(SF_INFO)); > psfinfo_out = (SF_INFO *) malloc(sizeof(SF_INFO)); > Maybe try to clear psfinfo_in and/or psinfo_out before using them. -- dupswapdrop -- the music-dsp mailing list and website: subscription i