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

I'm working on a game application that routinely runs around 60-80 of these pitch shifters in a real-time callback on consumer hardware. Performance is fine. To clarify, the state of the resampler is: - Four stored samples for continuity between runs - The fractional offset from the last

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

Original Message Subject: Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation From: "Evan Balster" Date: Mon, February 22, 2016 6:45 pm To: music-dsp@music.columbia.edu

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

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. If you'd like to reference a more comprehensive implementation (with anti-aliasing and rate interpolation) I can provide my current code -- but it's part of a

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

On 23/02/2016 1:24 AM, Kjetil Matheussen wrote: 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

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

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

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

Original Message Subject: Re: [music-dsp] Tip for an audio resampler library doing cubic interpolation From: "Evan Balster" Date: Mon, February 22, 2016 1:02 pm To: k.s.matheus...@notam02.no

Re: [music-dsp] Time-domain noisiness estimator

> Am 22.02.2016 um 17:01 schrieb Dario Sanfilippo : > > I'll try studying autocorrelation more and see if I can implement a new > algorithm or combine it to the one I already have. Dario, you need to be careful with polyphonic material plus noise. The

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

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 four-point hermite) and it's fairly straightforward to implement

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

I'm not sure exactly what your requirements are (e.g. does it need to be a cubic resampler?). But I've used the resampler that is part of the speex library in the past and it did a fine job. It uses the table-based windowed sinc approach. I believe it's a BSD-style license, and it's well-separated

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

Re: [music-dsp] Time-domain noisiness estimator

Thanks, Corey, Steffan and Risto. I'll try studying autocorrelation more and see if I can implement a new algorithm or combine it to the one I already have. ZCR alone doesn't seem to work unless for specific contexts. Would you agree on that? For example, a 10kHz sine and a 2kHz bandwidth noise

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

SoX Resampler Library (https://sourceforge.net/projects/soxr) also provides a quick cubic interpolation with a header (soxr-lsr.h) compatible with libsamplerate. (audacity uses soxr) ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu

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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 channel. No interleaving is

Re: [music-dsp] Time-domain noisiness estimator

On February 22, 2016 at 12:13:39 pm +01:00, Corey K wrote: > I don't have any links on the use of autocorrelation in this context, and I > don't even know if it would work. My basic thought, however, was that the > autocorrelation of white noise should be zero at all

[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] Time-domain noisiness estimator

These properties are true, if you have only noise or only signal. In case of a mixture, also the described properties mix and this “torpedoes” that approach. So, an FFT with a subsequent processing like floor estimation (connect a line thru all floors between peaks) and peak estimation (connect

Re: [music-dsp] Time-domain noisiness estimator

I don't have any links on the use of autocorrelation in this context, and I don't even know if it would work. My basic thought, however, was that the autocorrelation of white noise should be zero at all time lags other than 0. Pitched signals, on the other hand, should have peaks at multiples of

Re: [music-dsp] Time-domain noisiness estimator

Thank you so much for the explanation, Ethan. Best, Dario On 21 February 2016 at 23:47, Ethan Duni wrote: > Not a purely time-domain approach, but you can consider comparing sparsity > in the time and Fourier domains. The idea is that periodic/tonal type > signals may be