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Take it easy!

old.hikari3.ch will be permanantly shut down soon

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Could someone give me a quick run down?

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>>12868 Proof? skeptical

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Is Hikarin still waiting for a serious answer or was Hikarin satisfied with discussing armpits?

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nyagan all these toe hoe characters look the same.

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>>13339 You're blind.

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>>13340 cope tuhufag.


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What's hikarin's favorite soda?

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>>13094 I stopped drinking cokes years ago but Barq's was always my favorite.

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Delicious.

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>>13094 im sippin on some A 'n' dubya

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>>13126 really, a scientist's fuel its my favorit 2! esspecialy the vanilla float flavour

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>>13330 actual coke floats (coke with scoops of vanilla ice cream in it) fucking suck though.


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how do i use those emoji thingies? t. retard

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Dis nyagan 4 reel?



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Jesus fucking Christ I can't focus on anything, I switch a tab in my browser or go to a different window on my computer and I go down 20 different rabbit holes

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cry could be worse cry

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>>13319 Love it when Wonderland's White Rabbit is a shab


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So I've been working on a project to create a singing voice synthesizer I'm basing it on the description in Jordi Bonada's PhD thesis "Voice Processing and Synthesis by Performance Sampling and Spectral Models". After a lot of trouble with getting TWM f0 estimation to work, I've finally gotten to implementing MFPA (Maximally Flat Phase Alignment". And amazingly, it seems to have worked first try. Compare my results: https://i.ibb.co/dsvgv0fd/Screen-Shot-2026-03-02-at-3-54-48-PM.png To the results in the study: https://i.ibb.co/C3fjdWVd/Screen-Shot-2026-03-02-at-3-55-09-PM.png

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There's been a lot of mistakes I've known about for a while but haven't bothered to publicly mention yet. I thought I'd finally do that. https://queuesevenm.wordpress.com/2026/10/18/some-overdue-corrections/ Also, another thing I said that was wrong originally is which processing techniques VOCALOID 1 and VOCALOID 2 use. The correct answer is V1 uses Spectral Modeling Synthesis, while V2 uses Spectral Peak Processing/Scaling/Phase-locked Vocoder.

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Analysis/synthesis is a paradigm of audio processing. In this paradigm, an audio signal is analyzed to produce a model of the signal. Then this model is transformed according to our wishes. Finally, the modified model is synthesized to produce the transformed signal. This paradigm can be used not only for transforming signals, but also for synthesizing them from scratch by means of selecting prerecorded samples from a database. Also, another method is omit the analysis stage and generate the model using statistics or neural networks. In analysis/synthesis, for any real (i.e. non-synthetic) signal, there will always be a portion of it which cannot be represented well by the model. The exact composition of this component depends on exactly what the model itself models, but in this post I will be focusing on what can not be modeled by any model, that is, stochastic components. Broadly, even if they are not specifically implemented this way, analysis/synthesis techniques that consider the stochastic part due so by first separating out the part that cannot be modeled by the model. Although the remaining component is stochastic, it is not strictly random. Usually, this means it is either tied to the model, or a separate model that describes a non-random aspect of the stochastic component. For example, a typical case is that the spectrum of then stochastic part is not flat as it would be in white noise, but follows an envelope. In that case, the extraction of just the random part would be done by inverse filtering the stochastic component by that envelope, a process known as spectral whitening. From there, there are basically two ways to proceed. Either the random signal is preserved, or it is discarded and replaced with white noise at synthesis. The former has the advantage that, depending how the rest of system is implemented, it may be possible to reconstruct the original signal perfectly when no transforms are applied. Additionally, in practice, the 'random' component will in practice always include some amount of signal, so discarding loses some aspect of the original signal. On the other hand, the latter has the advantage of flexibility. Often, transforming the model requires that we now have more or less samples of the random part. One option would be to interpolate the random component. However, when more samples are required than there were originally (i.e. upsampling), the new signal is smoother than the original. This can be a significant problem, for example if we are slowing down a signal by large factor. From signal theory, we know that all other interpolation methods approximate the ideal sinc-convolution interpolation. In the frequency-domain, that method corresponds to there being no new spectral components above the old Nyquist frequency when upsampling. Given that our random component is supposed to be white noise, i.e. that it should have a flat spectral envelope, we can now see where the smoothness problem comes from. Thus, from this observation, I propose a hybrid model. We preserve the original random component, with all the advantages that brings. When requiring a different amount of samples than there were originally, we resample it. However, in the case of upsampling, we replace the new spectral bins with random values instead of zeros. (continued in next post)

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(continued from previous post) Thusfar, I have discussed how the spectrum should change under this proposal, however in practice the direct principle cannot be implemented directly. Two practical reasons are that my FFT implementation has fixed maximum size, so it cannot be used on arbitrary-length signals; and that doing so also becomes inefficient with larger and larger FFT size. However, a more fundamental issue is the circular nature of the FFT which results in artifacts whenever the signal is not entirely periodic with a period corresponding to exactly the FFT window. This problem can be either mitigated through techniques such as overlap-and-add; or time-domain convolution (i.e. filtering) can be used instead. For linear transforms, the latter is almost always used because it is much more efficient, and at the same time, it produces far lower levels of artifacts. Based on that, this proposal could be implemented by lowpass filtering the interpolated signal (i.e. resampling), and adding to that highpass-filtered white noise whose bandwidth corresponds to the difference between the new and old sampling rates. In practice, real lowpass filters are not ideal, so they have a non-zero transition band. Thus, the resulting envelope is not completely flat and would have spikes due to the transition bands. This could be reduced, although not completely, by increasing the filter size, at the cost of additional computational complexity. Another option is to adjust the bandwidths slightly such that total distortion is minimal (i.e. error can in either direction, not just positive). Another interesting idea would be to see if a window could be used whose frequency response is overlapping. In the time-domain, some windows are said to be overlapping, for example the triangular window, Hanning window, and Gaussian window. What that means is that it is possible to separate the windows by a specific distance, and the resulting of adding them together is one outside of given edges. The Kaiser-Bessel window is almost-but-not-quite overlapping, which is because it is almost-but-not-quite the Gaussian window. If a window's frequency response were to be overlapping, using it as the FIR filter construction window would solve the problem from before. Special care would need to made to ensure that the filter's transition band matches the size needing for proper overlapping as closely as possible. Usually we do not care about this since the transition band being smaller than desired is usually exclusively a good thing.

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>>13286 Actually, thinking about it, the resilient border interpolation at synthesis is not unnecessary now given by proposed WBHSM+LPC framework. The reason it became unnecessary before is that the proposed RVPM modeled each voice pulse as its full impulse response, assuming a time-invariant filter for each pulse (although I suppose the EpR differential envelope may model some of the time-variance of the filter, but this breaks as transforms are applied). The new WBHSM+LPC model is a literal combination of WBHSM and LPC where WBHSM models the source signal. Since that signal is not filtered, each voice pulse is actually independent, and pulses and periods are truly equivalent. Because of that, we actually go back to modeling each individual pitch period, reintroducing the need for the synthesis border interpolation. For this, we can do the same thing we do in the analysis stage where we shifted the window such that the window starts coincides roughly with the glottal opening instant and only the right side of the window is interpolated. This becomes especially beneficial for the WBHSM+LPC synthesis since the left side is where the voice pulse actually is, while on the right side the glottis should be closed or almost closed.

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Actually there's an alternative to converting the parallel filter to cascade filter, which is just to keep it as a parallel filter. How this can be done I will describe shortly. This may be preferred as I believe not all allophones can be modeled well by cascade filters. Furthermore, perhaps higher resonances can't be at all in general. The parallel filter is just a sum of resonances, which are each linear-time invariant filters, thus it is also a linear-time invariant filter and thus also can be expressed as a z-transfer function. This can be done in practice by starting with a z-domain expression of the collective z-transfer function, in the form of a some of all the composite z-transfer functions. Then, each of the terms are refactored such that they all have an identical. Finally, all of the numerators can be summed. Much of the terms can then be combined since they are just constants times some power of z. The result is an IIR filter. We can use the decomposition property of IIR filters that lets us turn one into an all-pole IIR filter and an FIR filter that represents the zeros. We do the inverse filtering step using the former. Finally, we add an additional step in the second WBHSM pass where we divide the spectrum by the latter's frequency response.


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Any other hikarians who suffer from speech impediments? What impediments do you have? When did it start? How do you cope? How many times do you wish death upon yourself when you're forced to speak publicly? How do you think it affected your life? I'll start: >What impediments do you have? stutter / stammer + heavy blocking >When did it start? as long as i can remember >How do you cope? obv i re-arrange my sentences and use synonyms but when i have to say a specific word that ik i will stutter in i try to skip or say the first syllable in a quieter voice this trick rarely works tho because i don't have a great control over my voice >How do you think it affected your life? ofc it made me make almost 0 friends as a kid because everyone thought i am a weird retarded kid because i sound like one hence i deserve to get bullied but actually now that i think about it, the lack of friendship was actually valuable cuz it made me develop skills and hobbies that i wouldn't have had time to

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>>11963 I pronounce V's as W's

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>>11979 >Alogia Not sure of i have this or just a very low verbal iq (even though i speak 3 languages "fluently")

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personally i don't know if i have a stand alone speech disorder but autism + adhd give me a hard time finding words, pronouncing them right, not stammering and then getting the whole sentence out. more often than not i fuck up one word and then have to restart the entire sentence over and over until i get it out one time fully start to finish without fucking it up

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I have a British accent In all seriousness though, I fumble and mutter a lot in casual exchanges, especially in my mother tongue. I can roughly tell when a social cue is dropped, just can't grasp the words and cultural sentiment to respond to it properly. It is easier for me to react to such cues in English, but even then I'm not what you'd call "socially fluent". It's a bit uncanny because this isn't a case of being raised multiculturally, by migrant parents in an English-speaking country, or what have you. I just taught myself English from a very early age, and I do mean like from 7 or 8 yrs old (TV and internet, of course). In spite of the aforementioned, I can project my voice clearly during presentations no matter the language, so it only applies to casual situations. I don't think this is considered a speech impediment as such, but figured some hikarins might relate to this. The notion of finding it easier to communicate certain things in acquired languages is understood and not uncommon. I just think my case leans toward the more extreme end where my native fluency is effectively eroded.

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>>13325 I gross at 90% of standalone expressions in my native language, like what do you mean "i love you" disdain, bitch you sound like a hog in heat


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>>13243 Just ignore the roodey p**s or mock them snicker

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wat if raymoo was r baby in th game?

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why is my wife the bg now?

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>>13326 your wife? but shes with me right now

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>>13327 your wife?? that's my wife! get your own!! angry


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Do West African hikarins exist? I want to know if anyone there still gives a shit about Fela Kuti. He's corny, but he pulls it off well.

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Probably not

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No, only South Americansnicker


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Do you like my eggs? I added EXTRA butter and Oil

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>>10696 Looks like melted popcorn neco

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It probably tastes quite good but the presentation could use some work. neco

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Did you melt chees and eggs

Your fortune: Average Luck

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These look like the type of eggs to give me diarrhea for 3 days

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>>13317 Has this happened to you before?


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