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bjorg: Linear Interpolation for Audio in C, C++, Java, etc.
http://blog.bjornroche.com/2010/10/linear-interpolation-for-audio-in-c-c.html
I am omitting it here and jumping straight to an example. To perform a linear interpolation of 100 samples where y [0] = 7, and y [100] = 20, our code would look something like this: double start = 7; double end = 20; for ( int i=0; i<100; ++i ) {. double ratio = i/100.0; y [i] = start * (1-ratio) + end * i; }
AUDIO INTERPOLATION
https://www.ecse.rpi.edu/~rjradke/papers/radkeaes22.pdf
Radke and Rickard Audio Interpolation 3. AUDIO INTERPOLATION We are interested in synthesizing the signal that would have been received at a microphone M α placed a fraction α of the way along the line connecting M 0 to M 1 (the “baseline”). From Figure 1 and the law of cosines, we can compute d αJ = q α(1−α)∆2 +(1−α)d2 0J +αd 2 1J (16)
Alpha-II - Programming Information: Audio Interpolation
https://alpha-ii.com/Info/AudioInt.html
AudioInterpolation. Audio interpolation is a method of making digital audio sound betterthan it really is. Thesedays, almost every digital playback device (CD players*, digital receivers, sound cards, etcetera) uses interpolation to improve your listening experience. And with computers everincreasing in speed, it's now possible to use software implementations as well without …
Audio interpolation - melonDS
http://melonds.kuribo64.net/comments.php?id=147&p=2
Anyway, audio interpolation is one of the emulation improvements that have been requested for melonDS. My general policy for emulation improvements is that they should allow for keeping the accurate code paths, and they shouldn't add too much complexity to the code. Audio interpolation is well within these bounds.
How do you do bicubic (or other non-linear) interpolation ...
https://stackoverflow.com/questions/1125666/how-do-you-do-bicubic-or-other-non-linear-interpolation-of-re-sampled-audio-da
I'm writing some code that plays back WAV files at different speeds, so that the wave is either slower and lower-pitched, or faster and higher-pitched. I'm currently using simple linear interpolation, like so: int newlength = (int)Math.Round (rawdata.Length * lengthMultiplier); float [] output = new float [newlength]; for (int i = 0; i < newlength; i++) { float realPos = i / …
Lagrange Interpolation | Physical Audio Signal Processing
https://www.dsprelated.com/freebooks/pasp/Lagrange_Interpolation.html
Matlab Code for Lagrange Interpolation. A simple matlab function for computing the coefficients of a Lagrange fractional-delay FIR filter is as follows: function h = lagrange ( N, delay ) n = 0:N; h = ones (1,N+1); for k = 0:N index = find (n ~= k); h (index) = h (index) * (delay-k)./ (n (index)-k); end.
DigitalAudioResamplingHomePage - Stanford …
https://ccrma.stanford.edu/~jos/resample/resample.pdf
Shannon’s sampling theorem says it is possible to restore an audio signal exactly from its samples, it makes sense that the best digital audio interpolators would be based on that theory. Such “ideal” interpolation is called bandlimited interpolation. A bandlimited interpolation algorithm designed along these lines is described in the theory
Multi-channel audio upsampling interpolation - Signal ...
https://dsp.stackexchange.com/questions/58032/multi-channel-audio-upsampling-interpolation
Does cubic interpolation (or any other) have any advantages over linear for the specific case of audio? You'd use neither for audio. The reason is simple: The signal models you typically assume for audio signals are very "Fourier-y", to say, they assume that sound is composed of weighted harmonic oscillations, and bandlimited in its nature.
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