[music-dsp] Antialias question
Dear List, Long time lurker here, learned a lot from the random posts, so thanks for those. Maybe somebody can help me out what is the best practice for realtime applications to minimize aliasing when scanning a waveform by changing speed or constantly modulating the delay time on a delay (it's like the resampling rate is changing by every step)? I guess the smaller the change, the smaller the aliasing effect is, but what if the change can be fast enough to make it audible? If you can suggest a paper or site or keywords that I should look for, I'd appreciate the help! Kevin ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Antialias question (Kevin Chi)
Thanks Frank, I use cubic interpolation to interpolate between samples and it seems a bit better than linear for me. I am more worried about the downsampling, when the playhead is going faster than the original samplerate and that's when the high frequencies of the original material start to fold back at Nyquist to cause aliasing. My concern is if it would be a constant pitching up (downsample) rate then it would be "easy" to apply an antialias filter based on the resampling rate. But as the resampling rate can change by every step (as the speed of the playhead is modulating), you should apply different antialias filters for every step. Or at least the filter needs to use different coefficients every time it's applied... That's what I am unsure of what is the best solution to solve. -- Kevin /Hi Kevin. />//>/I'm the least-expert guy here I'm sure, but as a fellow newbie I might />/have some newbie-level ideas for you. />//>/Just to mention something simple: linear interpolating between samples is />/a huge improvement in reducing aliasing over not interpolating. For />/instance if your playback math says to get sample 171.25 out of the buffer, />/use .75 of sample 171 and .25 of sample 172. I don't know the math but my />/listening tests of tricky test waveforms (eg, a waveform with a fundamental />/at 100Hz and a 60th harmonic at 6kHz pumped up to 10, 20,30x the power, />/showed aliasing reduced by about the same amount that quadrupling the />/buffer sample rate did. />//>/I looked into other interpolations besides linear, and they didn't seem />/effective enough to bother programming. Just to give a feeling for it, if />/linear eliminated 90% of aliasing, then then more complex interpolation />/might eliminate 95%. So they might reduce it by half compared to linear, />/but only a tiny bit better compared to none at all. (The percentages are />/just meant to be illustrative and actually everything totally depends on />/your input data.) />//>/Another thing is to make sure your input data has been filtered out such />/that there's no frequencies over the Nyquist frequency. But if you're />/dealing with a PC or smart phone I'd imagine the computer hardware handles />/that for you. Once the data is in memory you cannot filter it out, as it />/will already have morphed into an alias in your input buffer and be "baked />/in" no matter how you try to play the data back. />//>/Finally, listen with good headphones; you'll hear things you probably />/won't in a room even if you have a good amp and speakers. />//>/Frank />/http://moselle-synth.com// ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Antialias question
Thanks for your ideas, I'll look into those! It's actually just a digital delay effect or a sample playback system, where I have a playhead that have to read samples from a buffer, but the playhead position can be modulated, so the output will be pitching up/down depending on the actual direction. It's realtime resampling of the original material where if the playhead is moving faster than the original sample rate, then the higher frequencies will be folding back at Nyquist. So before sampling I should apply an antialias filter to prevent it, but as the rate of the playback is always modulated, there is not an exact frequency where I should apply the lowpass filter, it's changing constantly. This is what I meant by comparing to resampling. -- Kevin Hello Kevin I am not convinced that your application totally compares to a continously changed sampling rate, but anyway: The maths stays the same, so you will have to respect Nyquist and take the artifacts of your AA filter as well as your signal processing into account. This means you might use a sampling rate significantly higher than the highest frequency to be represented correctly and this is the edge frequency of the stop band of your AA-filter. For a wave form generator in an industrial device, having similar demands, we are using something like DSD internally and perform a continous downsampling / filtering. According to the fully digital representation no further aliasing occurs. There is only the alias from the primary sampling process, held low because of the high input rate. What you can / must do is an internal upsampling, since I expect to operate with normal 192kHz/24Bit input (?) Regarding your concerns: It is a difference if you playback the stream with a multiple of the sampling frequency, especially with the same frequency, performing modulation mathematically or if you perform a slight variation of the output frequency, such as with an analog PLL with modulation taking the values from a FIFO. In the first case, there is a convolution with the filter behaviour of you processing, in the second case, there is also a spreading, according to the individual ratio to the new sampling frequency. From the view of a musical application, case 2 is preferred, because any harmonics included in the stream , such as the wave table, can be preprocess, easier controlled and are a "musical" harmonic. In one of my synths I operate this way, that all primary frequencies come from a PLL buffered 2 stage DDS accesssing the wave table with 100% each so there are no gaps and jumps in the wave table as with classical DDS. j ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Antialiased OSC
Thank you for the quick ideas, the code looks nice. Currently I am not designing a wavetable OSC, I am just trying to do some basic VA waveforms (saw, tri, square) so naively I thought I just set up the 15-20 tables/waveform from fourier series whenever I start the app/plugin and that could do the job. Then interpolating between the samples of the closest wavetables. But maybe I am too naive? :) With wavetables I was also wondering how one can act when there is a pitch modulation of a running note that will go up/down an octave... as my understanding this would mean if I don't switch wavetables at a certain pitch mod, then it will introduce more and more aliasing. But checking this at every sample sounds like some overhead maybe shouldn't be there. -- Kevin @ Fine Cut Bodies Mailto: ke...@finecutbodies.com Web: http://finecutbodies.com LinkedIn: http://lnkd.in/XZiSjF On 8/3/18 3:11 PM, music-dsp-requ...@music.columbia.edu wrote: lemme know if this doesn't work:?https://www.dropbox.com/s/cybcs7tgzgplnwc/wavetable_oscillator.c remember that this is*synthesis* code.? it does not extract wavetables from a sampled sound (i wrote a paper a quarter century ago how to do that, but it's not code).? nor does it define bandlimited square, saw, PWM, hard-sync, whatever.? that's a sorta difficult problem, but one that someone has for sure solved and we can discuss here how to do that (perhaps in MATLAB).? extracting wavetables from sampled notes requires pitch detection/tracking and interpolation. ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
[music-dsp] Antialiased OSC
Hi, Is there such a thing as today's standard for softSynth antialiased oscillators? I was looking up PolyBLEP oscillators, and was wondering how it would relate to a 1-2 waveTables per octave based oscillator or maybe to some other algos. thanks for any ideas and recommendations in advance, Kevin ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Antialiased OSC
I just want to thank you guys for the amount of experience and knowledge you are sharing here! This list is a gem! I started to replace my polyBLEP oscillators with waveTables to see how it compares! Although while experimenting with PolyBLEP I just run into something I don't get and probably you will know the answer for this. I read at a couple of places if you use a leaky integrator on a Square then you can get a Triangle. But as a leaky integrator is a first order lowpass filter, you won't get a Triangle waveform, but this: https://www.dropbox.com/s/1xq321xqcb7ir3a/PolyBLEPTri.png?dl=0 Is it me doing something wrong misunderstanding the concept, or what is the best way to make a triangle with PolyBLEPs? thanks again for the great discussion on wavetables! -- Kevin @ Fine Cut Bodies ___ dupswapdrop: music-dsp mailing list music-dsp@music.columbia.edu https://lists.columbia.edu/mailman/listinfo/music-dsp
