Re: [music-dsp] Correctable signal processing (to arrive at "wire connection")
>So in the digital sense, or in combination with the analog domain, >is it reasonable to think about "correctable" operations, which as >it were can be inverted, so that applying a digital signal transformation >*and* it's converse, we end up with the same signal or something similar. I think you mean to write "inverse" rather than "converse" there, but the basic answer is yes. >Of course that first of all leads to the necessary condition that or signal >transformations are a bijection, which is hard considering most of these >operations will be filters, and in the digital domain, except for a few pathetic >cases, these will have bit depth issues. The bijection condition for LTI filters basically means that none of the roots are on the unit circle. Which you'd want anyway, since otherwise the inverse filter is going to be unstable. That said, you *can* actually invert filters with roots on the unit circle (or anywhere else) by using z-transform techniques (i.e., evaluate all of your integrals on a contour other than the unit circle which doesn't contain any of the roots), although there are various practical limitations associated with that as well. Moreover, indeed, bit-depth issues are the main thing here. No digital system can ever be truly linear, due to finite word-length. Typically we assume that the bit depth is big enough that we can ignore that, but you can certainly construct cases where it shows up. For example, suppose we hit a signal with a very aggressive (but invertible) low pass filter, which drives the high frequency content below the noise floor implied by the bit depth. Then if we run the output through the inverse of the filter, we won't get back the original high frequency content - instead we'll get end up amplifying the noise floor back up so it becomes audible. Running the filters in the opposite order might well work, though (supposing the first filter doesn't run out of headroom to amplify the band in question). This effect can come up even with relatively mild filters, depending on the content of the signals being processed - if the high frequency content is only a few bits above the noise floor to begin with, then even a gentle low-pass response is going to submerge that content below the noise floor and render it unrecoverable. >So if we have N digital poles, can we create N digital zeros at the same >frequencies, convolve those two filters and arrive at a digital wire ? Convolving an all-pole filter's response with its inverse is always going to result in the identity system (theoretically). And not a delayed identity system, but the canonical one with zero delay. But that doesn't necessarily imply that we can run a signal through an all-pole filter, and then run the output through the inverse filter, and end up with a digital wire (even allowing for a few LSBs error to account for round-off effects). For this to work, we need a further assumption that finite word length effects are not having a significant impact in either of the two filters. If they are, then the system is no longer approximately linear, and so its output is not explained by the convolution integral, and so the convolution of the two filter responses then fails to describe the operation of the chain of the two filters. And that assumption depends on both the response of the filter in question, and the content of the signals being processed. In particular, signals and/or filters with a large spectral dynamic range require special care. In practice, this all generally works fine if the bit-depth used for intermediate processing is significantly larger than the one used for final rendering, and we avoid filters with aggressive frequency responses. I.e., a gentle EQ that compensates for deviations in a microphone frequency response is unlikely to cause any problems. But attempting to invert a system with (for example) a strong low-pass response is asking for trouble. E On Thu, May 8, 2014 at 7:23 AM, Theo Verelst wrote: > Hi all, > > In the analog domain, where most interesting DSP originates since, well, > the time of radio and early telephone, it is a commonality to search for > "signal neutrality" in certain reasonableness. So a phone line would be > specified to transfer certain frequencies with a certain amplitude and > phase reliability, and a specified absence of noise and linear distortion, > echo damping, etc. > > In the digital age, of course in principle a connection or file with > sufficient number of bits and well qualified sampling frequency is > considered pretty neutral as is. That's not all a correct hypothesis, for > instance think about the sampling issues like the intended reconstruction > filtering, but enough about that. Also, there are people seemingly more > concerned with adding and dealing with dithers than actually passing a > signal from A to B, but that aside too (even though that is an interesting > subject for other professional reasons). > > So in the digital sense, or
Re: [music-dsp] a weird but salient, LTI-relevant question
The 2-second area is my recollection as well, from when I played with noise sequence length, probably 20 years ago. Under 2, you don’t really have to pay attention to hear a repeat—your latches onto it easily—and as you get longer, you have to listen more carefully, and you get to the point quickly where you’re questioning yourself whether you’re hearing it repeat. The bottom line, I think, is that yes white noise is random, but the low frequency components are, well, low frequency. And it’s pretty each to pick out a repeating bump there. Not necessarily the bass end, more in the kids, and is probably a tradeoff between frequency and ear sensitivity. I just did a q&d app, which generates 10 seconds of white noise, then adjusts the loop length as a percentage of the mouse vertical in the window, and start point based on horizontal position. Anyway, like I said, I’ve been through this before so there were no surprises. Sampo, were you looking for a particular revelation? I’m not sure if I’m listening for what you were getting at. On May 8, 2014, at 8:43 AM, STEFFAN DIEDRICHSEN wrote: > I bounced some 100 secs of noise taken from the test oscillator in Logic Pro. > Loaded this in the IRU and did some cycling. > My finding: There are portions in the noise, that allows me to go down to 2 > seconds and it still sounded like straight (un-looped) noise. Other noise > portions had “features”, that sounded like persons talking in the background > or a squeak, or so. That’s so prominent, that it was easy to identify the > cycle. > Than, I did some experimentation, the IRU allows you to playback a selection > in a cycle and the cycle can be dragged around without interrupting the > playback. Doing that over a length of about 4 seconds with a 2 second long > selection sounded to me like straight noise. > > Steffan > > > On 08 May 2014, at 12:35, Sampo Syreeni wrote: > >> Interestingly, nobody's taken the test as of yet. Even if it ain't in the >> least bit a contest, and I already said to begin with that the result might >> be rather interesting for any and all. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] a weird but salient, LTI-relevant question
Having quickly browsed over this subject, think about it that it normally isn't needed to do more than be accurate enough when dealing with audio and the human hearing, unless you want to explicitly deal with Loudness Curve sensitivity, or exotic subjects like creating stereo images for cinema! Mathematically Linear and Time Invariance is usually needed for certain (important) classes of Differential Equations, and of course usually a desirable property for a lot of types of equipment dealing with signals. About the noise, I'll say it only one time: IS YOUR NOISE SOURCE BANDWIDTH LIMITED, and it so (important undergrad EE question, seriously): how ? I mean going to things like Gaussian Distributions goes deeper than most here will want to (or can) go, and I don't see the point of it much, unless it is made more specific what that is for. T. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] a weird but salient, LTI-relevant question
On 5/8/14 1:25 AM, Sampo Syreeni wrote: On 2014-05-08, robert bristow-johnson wrote: there was a way that you could do "subtractive dither" in that the dither that you added before quantizing to a short word could be subtracted (to regain 4.77 dB) [...] I have some code for just that, where the RNG for the dither is derived from the LSBs of the last N quantized words? is that how you're synchronizing the dither between the quantizer and the later expansion of the word? even, and even better ideas. i'm all ears. when you loop the noise, is it a "butt-splice"? (i.e. no crossfade.) Yes. Otherwise the splice might introduce an interpolation artifact which would invalidate the experiment from the start. i think that if you cannot hear a different with a butt splice, you won't hear it with a cross fade. it's news to me that human hearing is LTI. Yes, well, it ain't. But even conventional psychophysical theory treats it as such. if that were the case, Fletcher-Munson curves (or Robinson-Dadson, or pick your researcher) would have equal spacing for all frequencies. the fact that they get squished at the very low and very high frequencies is ostensibly not linear behavior. -- r b-j r...@audioimagination.com "Imagination is more important than knowledge." -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] a weird but salient, LTI-relevant question
>the human hearing system is kind of an LTI... only at very low level >processing. The consistency of measured signal (= perceiving the same >signal the same way at all time as somebody wrote here) is present in >the ear canal up to brainstem -> inferior colliculus. My understanding is that there are measurable nonlinear effects even in the cochlea. Apparently when a loud frequency is present that excites one region of the membrane, the surrounding fibers react to dampen nearby sound and reinforce the purity of the dominant frequency. The pithy way of phrasing this is "the frequency response of the ear of a dead cat is different from that of a live cat." Not sure if anybody actually did that exact experiment to verify that... Of course, that doesn't invalidate the "same sound sounds the same later" property, but if you ask me that's a much much broader thing than LTI. For example, any static nonlinearity - no matter how extreme and nonlinear - will always produce the same output given the same input. That doesn't mean it's linear, it just means it's time-invariant. E E On Wed, May 7, 2014 at 10:59 PM, Enr G wrote: > My two cents as a person in the field: > > the human hearing system is kind of an LTI... only at very low level > processing. The consistency of measured signal (= perceiving the same > signal the same way at all time as somebody wrote here) is present in > the ear canal up to brainstem -> inferior colliculus. But once we go > to higher neuronal processing of auditory signals things get > complicated and the same signal can be perceived in many different > ways (e.g. google for top-down mechanism of auditory attention). The > (non linear) fourier analysis and interpreting sounds as sinusoid are > valid at ear canal level, and there are models with filterbanks to > simulate that. But once we go to conscious perception (=cerebral > cortex) evidence from animal research seems to point to a more complex > analysis performed by the neurons: the so called spectro-temporal > modulation (basically a 2D fourier transform). I.e. envelopes and > phases are treated in different ways to identify "sound objects". For > those interested, this is a nice starting point (open access): > > http://www.ploscompbiol.org/article/info%3Adoi%2F10.1371%2Fjournal.pcbi.1003412#pcbi-1003412-g007 > > e. > > On Thu, May 8, 2014 at 8:28 AM, eric wrote: > > It would appear to me that the human hearing system is an LTI system. > It doesn't react in a linear fashion to frequency or loudness, but it > perceives the same signal the same way at all times, disregarding aging, > hearing loss, etc. > > > > On 5/8/2014 1:25:28 AM, Sampo Syreeni wrote: > > On 2014-05-08, robert bristow-johnson wrote: > > > >> there was a way that you could do "subtractive dither" in that the > >> dither that you added before quantizing to a short word could be > >> subtracted (to regain 4.77 dB) [...] > > > > I have some code for just that, even, and even better ideas. Maybe I > > even mentioned them somewhere a while back? If not, will fully share > > given interest. (The code is rather shitty, and even the ideas would > > benefit from development. But still better than you see implemented > > anywhere.) > > > > Yet why-oh-why doesn't anybody just pop up their Audacity and a few > > megabytes of randomness, the way I originally asked? Because the stuff > > I'm talking about really is kind of interesting and unexpected, once you > > try it out on your own ears... > > > >> when you loop the noise, is it a "butt-splice"? (i.e. no crossfade.) > > > > Yes. Otherwise the splice might introduce an interpolation artifact > > which would invalidate the experiment from the start. > > > >> it's news to me that human hearing is LTI. > > > > Yes, well, it ain't. But even conventional psychophysical theory treats > > it as such. For example, why would we hear frequencies unless the ear > > was LTI? Fourier analysis, that is sinusoids as something special, > > doesn't make much sense unless you assume... Well, you know, at least > > something having to do with linearity and shift-variance... ;) > > -- > > Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front > > +358-40-3255353, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2 > > -- > > dupswapdrop -- the music-dsp mailing list and website: > > subscription info, FAQ, source code archive, list archive, book reviews, > dsp links > > http://music.columbia.edu/cmc/music-dsp > > http://music.columbia.edu/mailman/listinfo/music-dsp > > > > > > --- > > This email is free from viruses and malware because avast! Antivirus > protection is active. > > http://www.avast.com > > -- > > dupswapdrop -- the music-dsp mailing list and website: > > subscription info, FAQ, source code archive, list archive, book reviews, > dsp links > > http://music.columbia.edu/cmc/music-dsp > > http://music.columbia.edu/mailman/listinfo/music-dsp > -- > dupswapdrop -- the music-dsp mailing list and website: > subscription inf
Re: [music-dsp] a weird but salient, LTI-relevant question
>It would appear to me that the human hearing system is an LTI system. > It doesn't react in a linear fashion to frequency or loudness, but it perceives >the same signal the same way at all times, disregarding aging, hearing loss, etc. One of the easiest ways to see that hearing must be nonlinear is to think about masking effects. In isolation, a signal will sound like one thing. But if you add another signal to it, then the sum will not sound like the sum of the two signals (generally). One of the signals may even disappear from your perception entirely! So, this can't be the product of a linear system. Also, you don't necessarily perceive the same signal the same way at all times (even ignoring aging, etc.). This is relevant to Sampo's loop experiment, actually. Take a loop of sound (any sound), say 1-2 seconds in length, and listen to it for a long time (say 1 minute or more). You'll find that after a while weird things happen to your perception of the sound, with some components seeming to move in and out of phase with others and so on. Which doesn't sound like a time-invariant system to me! E On Wed, May 7, 2014 at 10:28 PM, eric wrote: > It would appear to me that the human hearing system is an LTI system. It > doesn't react in a linear fashion to frequency or loudness, but it > perceives the same signal the same way at all times, disregarding aging, > hearing loss, etc. > > On 5/8/2014 1:25:28 AM, Sampo Syreeni wrote: > On 2014-05-08, robert bristow-johnson wrote: > > > there was a way that you could do "subtractive dither" in that the > > dither that you added before quantizing to a short word could be > > subtracted (to regain 4.77 dB) [...] > > I have some code for just that, even, and even better ideas. Maybe I > even mentioned them somewhere a while back? If not, will fully share > given interest. (The code is rather shitty, and even the ideas would > benefit from development. But still better than you see implemented > anywhere.) > > Yet why-oh-why doesn't anybody just pop up their Audacity and a few > megabytes of randomness, the way I originally asked? Because the stuff > I'm talking about really is kind of interesting and unexpected, once you > try it out on your own ears... > > > when you loop the noise, is it a "butt-splice"? (i.e. no crossfade.) > > Yes. Otherwise the splice might introduce an interpolation artifact > which would invalidate the experiment from the start. > > > it's news to me that human hearing is LTI. > > Yes, well, it ain't. But even conventional psychophysical theory treats > it as such. For example, why would we hear frequencies unless the ear > was LTI? Fourier analysis, that is sinusoids as something special, > doesn't make much sense unless you assume... Well, you know, at least > something having to do with linearity and shift-variance... ;) > -- > Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front > +358-40-3255353, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2 > -- > dupswapdrop -- the music-dsp mailing list and website: > subscription info, FAQ, source code archive, list archive, book reviews, > dsp links > http://music.columbia.edu/cmc/music-dsp > http://music.columbia.edu/mailman/listinfo/music-dsp > > > --- > This email is free from viruses and malware because avast! Antivirus > protection is active. > http://www.avast.com > -- > dupswapdrop -- the music-dsp mailing list and website: > subscription info, FAQ, source code archive, list archive, book reviews, > dsp links > http://music.columbia.edu/cmc/music-dsp > http://music.columbia.edu/mailman/listinfo/music-dsp > -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] a weird but salient, LTI-relevant question
On Thu, May 8, 2014 at 12:59 AM, Enr G wrote: > My two cents as a person in the field: > > the human hearing system is kind of an LTI... LTI is a very specific thing. It's not sort of, kind of, LTI--it's just either LTI or not. > only at very low level > processing. The consistency of measured signal (= perceiving the same > signal the same way at all time as somebody wrote here) is present in > the ear canal up to brainstem -> inferior colliculus. No, it's not. LTI means always stationary. Two easy ones that originate from the named region: 1. Stapedius response 2. Tinnitus I agree with the sentiment: There are multiple concurrent representations of sound, and at some level of auditory processing, sounds are frequently represented the same way. but it's not LTI--you have to ignore a lot of things to treat your auditory system as approximately LTI > But once we go > to higher neuronal processing of auditory signals things get > complicated and the same signal can be perceived in many different > ways (e.g. google for top-down mechanism of auditory attention). The > (non linear) fourier analysis and interpreting sounds as sinusoid are > valid at ear canal level, and there are models with filterbanks to > simulate that. But once we go to conscious perception (=cerebral > cortex) evidence from animal research seems to point to a more complex > analysis performed by the neurons: the so called spectro-temporal > modulation (basically a 2D fourier transform). I.e. envelopes and > phases are treated in different ways to identify "sound objects". For > those interested, this is a nice starting point (open access): > http://www.ploscompbiol.org/article/info%3Adoi%2F10.1371%2Fjournal.pcbi.1003412#pcbi-1003412-g007 Looks good--I read some very good articles from almost a decade ago (sigh) about the planum temporale (posterior temporal gyrus, right?). The Robert Zatorre articles on this topic were my favorite ones. > > e. > > On Thu, May 8, 2014 at 8:28 AM, eric wrote: >> It would appear to me that the human hearing system is an LTI system. It >> doesn't react in a linear fashion to frequency or loudness, but it perceives >> the same signal the same way at all times, disregarding aging, hearing loss, >> etc. >> >> On 5/8/2014 1:25:28 AM, Sampo Syreeni wrote: >> On 2014-05-08, robert bristow-johnson wrote: >> >>> there was a way that you could do "subtractive dither" in that the >>> dither that you added before quantizing to a short word could be >>> subtracted (to regain 4.77 dB) [...] >> >> I have some code for just that, even, and even better ideas. Maybe I >> even mentioned them somewhere a while back? If not, will fully share >> given interest. (The code is rather shitty, and even the ideas would >> benefit from development. But still better than you see implemented >> anywhere.) >> >> Yet why-oh-why doesn't anybody just pop up their Audacity and a few >> megabytes of randomness, the way I originally asked? Because the stuff >> I'm talking about really is kind of interesting and unexpected, once you >> try it out on your own ears... >> >>> when you loop the noise, is it a "butt-splice"? (i.e. no crossfade.) >> >> Yes. Otherwise the splice might introduce an interpolation artifact >> which would invalidate the experiment from the start. >> >>> it's news to me that human hearing is LTI. >> >> Yes, well, it ain't. But even conventional psychophysical theory treats >> it as such. For example, why would we hear frequencies unless the ear >> was LTI? Fourier analysis, that is sinusoids as something special, >> doesn't make much sense unless you assume... Well, you know, at least >> something having to do with linearity and shift-variance... ;) >> -- >> Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front >> +358-40-3255353, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2 >> -- >> dupswapdrop -- the music-dsp mailing list and website: >> subscription info, FAQ, source code archive, list archive, book reviews, dsp >> links >> http://music.columbia.edu/cmc/music-dsp >> http://music.columbia.edu/mailman/listinfo/music-dsp >> >> >> --- >> This email is free from viruses and malware because avast! Antivirus >> protection is active. >> http://www.avast.com >> -- >> dupswapdrop -- the music-dsp mailing list and website: >> subscription info, FAQ, source code archive, list archive, book reviews, dsp >> links >> http://music.columbia.edu/cmc/music-dsp >> http://music.columbia.edu/mailman/listinfo/music-dsp > -- > dupswapdrop -- the music-dsp mailing list and website: > subscription info, FAQ, source code archive, list archive, book reviews, dsp > links > http://music.columbia.edu/cmc/music-dsp > http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] a weird but salient, LTI-relevant question
Does learning count as a non-linearity? Agus, T.R., & Pressnitzer, D. (2013). The detection of repetitions in noise before and after perceptual learning. Journal of the Acoustical Society of America, 134(1), 464-473. http://lpp.psycho.univ-paris5.fr/abstract.php?id=3564 ...Diemo On 08.05.14 07:59, Enr G wrote: My two cents as a person in the field: the human hearing system is kind of an LTI... only at very low level processing. The consistency of measured signal (= perceiving the same signal the same way at all time as somebody wrote here) is present in the ear canal up to brainstem -> inferior colliculus. But once we go to higher neuronal processing of auditory signals things get complicated and the same signal can be perceived in many different ways (e.g. google for top-down mechanism of auditory attention). The (non linear) fourier analysis and interpreting sounds as sinusoid are valid at ear canal level, and there are models with filterbanks to simulate that. But once we go to conscious perception (=cerebral cortex) evidence from animal research seems to point to a more complex analysis performed by the neurons: the so called spectro-temporal modulation (basically a 2D fourier transform). I.e. envelopes and phases are treated in different ways to identify "sound objects". For those interested, this is a nice starting point (open access): http://www.ploscompbiol.org/article/info%3Adoi%2F10.1371%2Fjournal.pcbi.1003412#pcbi-1003412-g007 e. On Thu, May 8, 2014 at 8:28 AM, eric wrote: It would appear to me that the human hearing system is an LTI system. It doesn't react in a linear fashion to frequency or loudness, but it perceives the same signal the same way at all times, disregarding aging, hearing loss, etc. On 5/8/2014 1:25:28 AM, Sampo Syreeni wrote: On 2014-05-08, robert bristow-johnson wrote: there was a way that you could do "subtractive dither" in that the dither that you added before quantizing to a short word could be subtracted (to regain 4.77 dB) [...] I have some code for just that, even, and even better ideas. Maybe I even mentioned them somewhere a while back? If not, will fully share given interest. (The code is rather shitty, and even the ideas would benefit from development. But still better than you see implemented anywhere.) Yet why-oh-why doesn't anybody just pop up their Audacity and a few megabytes of randomness, the way I originally asked? Because the stuff I'm talking about really is kind of interesting and unexpected, once you try it out on your own ears... when you loop the noise, is it a "butt-splice"? (i.e. no crossfade.) Yes. Otherwise the splice might introduce an interpolation artifact which would invalidate the experiment from the start. it's news to me that human hearing is LTI. Yes, well, it ain't. But even conventional psychophysical theory treats it as such. For example, why would we hear frequencies unless the ear was LTI? Fourier analysis, that is sinusoids as something special, doesn't make much sense unless you assume... Well, you know, at least something having to do with linearity and shift-variance... ;) -- Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front +358-40-3255353, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2 -- Diemo Schwarz, PhD -- http://diemo.concatenative.net Sound–Music–Movement Interaction Team -- http://ismm.ircam.fr IRCAM - Centre Pompidou -- 1, place Igor-Stravinsky, 75004 Paris, France Phone +33-1-4478-4879 -- Fax +33-1-4478-1540 -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] a weird but salient, LTI-relevant question
I bounced some 100 secs of noise taken from the test oscillator in Logic Pro. Loaded this in the IRU and did some cycling. My finding: There are portions in the noise, that allows me to go down to 2 seconds and it still sounded like straight (un-looped) noise. Other noise portions had “features”, that sounded like persons talking in the background or a squeak, or so. That’s so prominent, that it was easy to identify the cycle. Than, I did some experimentation, the IRU allows you to playback a selection in a cycle and the cycle can be dragged around without interrupting the playback. Doing that over a length of about 4 seconds with a 2 second long selection sounded to me like straight noise. Steffan On 08 May 2014, at 12:35, Sampo Syreeni wrote: > Interestingly, nobody's taken the test as of yet. Even if it ain't in the > least bit a contest, and I already said to begin with that the result might > be rather interesting for any and all. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] a weird but salient, LTI-relevant question
Small correction: the correct name is MM5837, which is a 16 bit shiftregister device. It’s bad but can be replaced by the MM5437, a 23 bit device which can be clocked externally and has a much longer period. Steffan On 08 May 2014, at 07:35, STEFFAN DIEDRICHSEN wrote: > The MN5837 is a pretty good noise source, if clocked externally. The internal > clock is way too high and leads to audible periods. I used it in my thesis > with good results. > > Steffan > > Von meinem iPhone gesendet > >> Am 08.05.2014 um 06:51 schrieb Nigel Redmon : >> >> Reminds me…a few decades ago at Oberheim…Tom O. lamented to me bout a >> seemingly minor decision he’d made and later regretted…replacing an analog >> noise source with a digital noise generator (OBX—same in the Prophet 5). He >> took a bunch of grief from a guy who liked to meditate to noise, bought the >> OBX and was disappointed.You could hear the cycle pretty easily. > -- > dupswapdrop -- the music-dsp mailing list and website: > subscription info, FAQ, source code archive, list archive, book reviews, dsp > links > http://music.columbia.edu/cmc/music-dsp > http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] Correctable signal processing (to arrive at "wire connection")
On 2014-05-08, Theo Verelst wrote: So give or take a few LSB errors, are digital filters like filters in the analog domain? Yes. So if we have N digital poles, can we create N digital zeros at the same frequencies, convolve those two filters and arrive at a digital wire ? Of course there may be some delay here... Yes. Practical ? Yes. Already done, as your EE eminence well knows. Well, this week I was playing with my Lexicon AD convertors and a good microphone setup, driving my large monitoring system with my latest high quality ground-seperated 384 kHz DA convertor in a real-time situation, and wanted to compensate the small (few dBs here and there) the frequency sensitivity unevenness of the microphone I used, and applied some jack/jack-rack/ladspa Linux filters for that. Worked great. But did it actually constitute a digital wire? -- Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front +358-40-3255353, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2 -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
[music-dsp] Correctable signal processing (to arrive at "wire connection")
Hi all, In the analog domain, where most interesting DSP originates since, well, the time of radio and early telephone, it is a commonality to search for "signal neutrality" in certain reasonableness. So a phone line would be specified to transfer certain frequencies with a certain amplitude and phase reliability, and a specified absence of noise and linear distortion, echo damping, etc. In the digital age, of course in principle a connection or file with sufficient number of bits and well qualified sampling frequency is considered pretty neutral as is. That's not all a correct hypothesis, for instance think about the sampling issues like the intended reconstruction filtering, but enough about that. Also, there are people seemingly more concerned with adding and dealing with dithers than actually passing a signal from A to B, but that aside too (even though that is an interesting subject for other professional reasons). So in the digital sense, or in combination with the analog domain, is it reasonable to think about "correctable" operations, which as it were can be inverted, so that applying a digital signal transformation *and* it's converse, we end up with the same signal or something similar. Of course that first of all leads to the necessary condition that or signal transformations are a bijection, which is hard considering most of these operations will be filters, and in the digital domain, except for a few pathetic cases, these will have bit depth issues. So give or take a few LSB errors, are digital filters like filters in the analog domain? So if we have N digital poles, can we create N digital zeros at the same frequencies, convolve those two filters and arrive at a digital wire ? Of course there may be some delay here... Practical ? Well, this week I was playing with my Lexicon AD convertors and a good microphone setup, driving my large monitoring system with my latest high quality ground-seperated 384 kHz DA convertor in a real-time situation, and wanted to compensate the small (few dBs here and there) the frequency sensitivity unevenness of the microphone I used, and applied some jack/jack-rack/ladspa Linux filters for that. Worked great. T.V. -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] a weird but salient, LTI-relevant question
It may be fine to think of the ear as doing a Fourier transform as a first, crude approximation. For a more accurate description however, some nonlinear effects would have to be considered. And some of this already happens in the ear. As for hearing being LTI, think about forward and backward masking. Although it happens at a rather short time scale, it implies that time invariance is not always the case. Harmonic distortion and intermodulation of two sinusoids played loudly enough also speaks against linearity. I haven't tried the experiment, but I recall a composer collegue once complaining about repetition in the noise generator of Csound. I think they used a random generator with period 2^16 in those days, but it's been improved now. Risto Holopainen -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] a weird but salient, LTI-relevant question
On 2014-05-08, Olli Niemitalo wrote: Sampo's test should be carried out multiple times to gather statistics, and because repetition will aid in reinforcement of the memory, also the number of repetitions should be controlled or recorded. How about "tap to the rhythm of it"? Or, more to the point, you should always repeat the test using a different noise stream. You shouldn't be able to learn any statistical deviation from one test to another. The only learning and pattern recognition in play should take place from cycle to cycle, and possibly even so that you're limited from hearing more than two cycles of sequence. (Though it's pretty much impossible to implement that without the cutoff giving you a hint of what the repetition length was.) Interestingly, nobody's taken the test as of yet. Even if it ain't in the least bit a contest, and I already said to begin with that the result might be rather interesting for any and all. Feature-stripped noise should work better in some applications than truly random noise. Perhaps multi-band compression could be used to level it out. If you do anything of the sort, you by definition introduce structure into the signal. After that it ain't noise anymore. -- Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front +358-40-3255353, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2 -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] a weird but salient, LTI-relevant question
If there, by chance, happens to be a feature in the noise that "catches the ear" and creates a sort of (possibly first subconscious) memory, then the choo-choo effect will be more audible as that feature can be more easily recognized again, reinforcing the memory. I generated 10 seconds of Gaussian white noise and can consistently recognize a certain short rhythmic feature from it. And, minutes after stopping playback, I can still recall that memory in my mind. It's even more easy to recognize the periodicity if you train your ears to recognize a shorter piece before playing back the whole (10 second or so) loop. So I think it boils down to two things: features and learning. Learning can also turn "non-features" into "features". Sampo's test should be carried out multiple times to gather statistics, and because repetition will aid in reinforcement of the memory, also the number of repetitions should be controlled or recorded. How about "tap to the rhythm of it"? Feature-stripped noise should work better in some applications than truly random noise. Perhaps multi-band compression could be used to level it out. -olli On Thu, May 8, 2014 at 9:56 AM, Stefan Stenzel wrote: > As someone already pointed out, spend an evening to hack a website for this. > Otherwise I just don’t feel like it’s worth the hassle, this is why-oh-why I > don’t. > > Stefan > > On 08 May 2014, at 7:25 , Sampo Syreeni wrote: > >> Yet why-oh-why doesn't anybody just pop up their Audacity and a few >> megabytes of randomness, the way I originally asked? Because the stuff I'm >> talking > > > -- > dupswapdrop -- the music-dsp mailing list and website: > subscription info, FAQ, source code archive, list archive, book reviews, dsp > links > http://music.columbia.edu/cmc/music-dsp > http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
Re: [music-dsp] a weird but salient, LTI-relevant question
I'd recommend Intelligence by Jeff Hawkins for some thought-provoking insights into high-level perceptual processing in the brain. Richard > On 8 May 2014, at 06:59, Enr G wrote: > > My two cents as a person in the field: > > the human hearing system is kind of an LTI... only at very low level > processing. The consistency of measured signal (= perceiving the same > signal the same way at all time as somebody wrote here) is present in > the ear canal up to brainstem -> inferior colliculus. But once we go > to higher neuronal processing of auditory signals things get > complicated and the same signal can be perceived in many different > ways (e.g. google for top-down mechanism of auditory attention). The > (non linear) fourier analysis and interpreting sounds as sinusoid are > valid at ear canal level, and there are models with filterbanks to > simulate that. But once we go to conscious perception (=cerebral > cortex) evidence from animal research seems to point to a more complex > analysis performed by the neurons: the so called spectro-temporal > modulation (basically a 2D fourier transform). I.e. envelopes and > phases are treated in different ways to identify "sound objects". For > those interested, this is a nice starting point (open access): > http://www.ploscompbiol.org/article/info%3Adoi%2F10.1371%2Fjournal.pcbi.1003412#pcbi-1003412-g007 > > e. > >> On Thu, May 8, 2014 at 8:28 AM, eric wrote: >> It would appear to me that the human hearing system is an LTI system. It >> doesn't react in a linear fashion to frequency or loudness, but it perceives >> the same signal the same way at all times, disregarding aging, hearing loss, >> etc. >> >> On 5/8/2014 1:25:28 AM, Sampo Syreeni wrote: >> On 2014-05-08, robert bristow-johnson wrote: >> >>> there was a way that you could do "subtractive dither" in that the >>> dither that you added before quantizing to a short word could be >>> subtracted (to regain 4.77 dB) [...] >> >> I have some code for just that, even, and even better ideas. Maybe I >> even mentioned them somewhere a while back? If not, will fully share >> given interest. (The code is rather shitty, and even the ideas would >> benefit from development. But still better than you see implemented >> anywhere.) >> >> Yet why-oh-why doesn't anybody just pop up their Audacity and a few >> megabytes of randomness, the way I originally asked? Because the stuff >> I'm talking about really is kind of interesting and unexpected, once you >> try it out on your own ears... >> >>> when you loop the noise, is it a "butt-splice"? (i.e. no crossfade.) >> >> Yes. Otherwise the splice might introduce an interpolation artifact >> which would invalidate the experiment from the start. >> >>> it's news to me that human hearing is LTI. >> >> Yes, well, it ain't. But even conventional psychophysical theory treats >> it as such. For example, why would we hear frequencies unless the ear >> was LTI? Fourier analysis, that is sinusoids as something special, >> doesn't make much sense unless you assume... Well, you know, at least >> something having to do with linearity and shift-variance... ;) >> -- >> Sampo Syreeni, aka decoy - de...@iki.fi, http://decoy.iki.fi/front >> +358-40-3255353, 025E D175 ABE5 027C 9494 EEB0 E090 8BA9 0509 85C2 >> -- >> dupswapdrop -- the music-dsp mailing list and website: >> subscription info, FAQ, source code archive, list archive, book reviews, dsp >> links >> http://music.columbia.edu/cmc/music-dsp >> http://music.columbia.edu/mailman/listinfo/music-dsp >> >> >> --- >> This email is free from viruses and malware because avast! Antivirus >> protection is active. >> http://www.avast.com >> -- >> dupswapdrop -- the music-dsp mailing list and website: >> subscription info, FAQ, source code archive, list archive, book reviews, dsp >> links >> http://music.columbia.edu/cmc/music-dsp >> http://music.columbia.edu/mailman/listinfo/music-dsp > -- > dupswapdrop -- the music-dsp mailing list and website: > subscription info, FAQ, source code archive, list archive, book reviews, dsp > links > http://music.columbia.edu/cmc/music-dsp > http://music.columbia.edu/mailman/listinfo/music-dsp -- dupswapdrop -- the music-dsp mailing list and website: subscription info, FAQ, source code archive, list archive, book reviews, dsp links http://music.columbia.edu/cmc/music-dsp http://music.columbia.edu/mailman/listinfo/music-dsp
