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AUDIO INTERPOLATION - Rensselaer Polytechnic Institute
https://www.ecse.rpi.edu/~rjradke/papers/radkeaes22.pdf#:~:text=Using%20only%20the%20audio%20signals%20from%20two%20real,speech%20mixtures.%20Experimental%20results%20are%20presented%2C%20and%20sample
AUDIO INTERPOLATION - Rensselaer Polytechnic Institute
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)
(PDF) Audio Interpolation
https://www.researchgate.net/publication/2394632_Audio_Interpolation
Our audio interpolation algorithm decouples into two parts. First, a method based on the DUET algorithm is used to blindly Scott Rickard is also with the …
AES E-Library » Long Interpolation of Audio Signals Using ...
https://www.aes.org/e-lib/online/browse.cfm?elib=13390
Within the context of sinusoidal modeling, a new method for the interpolation of sinusoidal components is proposed. It is shown that autoregressive modeling of the amplitude and frequency parameters of these components allows us to interpolate missing audio data realistically, especially in the case of musical modulations such as vibrato or tremolo.
Multi-channel audio upsampling interpolation - Signal ...
https://dsp.stackexchange.com/questions/58032/multi-channel-audio-upsampling-interpolation
In your application, linear and even piece-wise cubic (Fig. 2) interpolation will give audible fractional-delay-dependent attenuation of high frequencies, if those are present, which may also hinder their cancellation in beamforming. Piece-wise cubic polynomial interpolation Your upsampling factor is 240 kHz / (48 kHz) = 5.
Linear Interpolation | Physical Audio Signal Processing
https://www.dsprelated.com/freebooks/pasp/Linear_Interpolation.html
Linear interpolation sounds best when the signal is oversampled. Since natural audio spectra tend to be relatively concentrated at low frequencies, linear interpolation tends to sound very good at high sampling rates .
DigitalAudioResamplingHomePage - Stanford …
https://ccrma.stanford.edu/~jos/resample/resample.pdf
we must be able to interpolate the signal between samples. Since the original signal is always assumed to be bandlimited to half the sampling rate, (otherwise aliasing distortion would occur upon sampling), Shannon’s sampling theoremtellsusthe signal canbe exactlyand uniquelyreconstructed for all time from its samples by bandlimited interpolation.
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