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Linear Interpolation as Resampling | Physical Audio Signal ...

    https://www.dsprelated.com/freebooks/pasp/Linear_Interpolation_Resampling.html
    Linearly interpolated fractional delay is equivalent to filtering andresampling a weighted impulse train (the input signal samples) with acontinuous-time filter having the simple triangular impulse response In discrete time processing, the operation Eq.(4.5) can beapproximated arbitrarily closely by digital upsampling by alarge integ…

DigitalAudioResamplingHomePage

    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

Audio Resampling — PyTorch Tutorials 1.10.1+cu102 ...

    https://pytorch.org/tutorials/beginner/audio_resampling_tutorial.html
    To resample an audio waveform from one freqeuncy to another, you can use transforms.Resample or functional.resample. transforms.Resample precomputes and caches the kernel used for resampling, while functional.resample computes it on the fly, so using transforms.Resample will result in a speedup when resampling multiple waveforms using the …

resampling or interpolation? - Stack Overflow

    https://stackoverflow.com/questions/17852639/resampling-or-interpolation
    In the case of audio, these are the amplitude values sampled at each time point. Resampling is used to either increase the sample rate (make the image larger) or decrease it (make the image smaller). Interpolation is the process of calculating values between sample points. So, if you resample an image you can use interpolation to do it.

Resampling - dspGuru

    https://dspguru.com/dsp/faqs/multirate/resampling/
    4.1.3 What is the “resampling factor”? The interpolation factor is simply the ratio of the output rate to the input rate. Given that the interpolation factor is L and the decimation factor is M, the resampling factor is L / M. In the above example, the resampling factor is 147 / 160 = 0.91875.

The current state of the art in audio signal interpolation

    https://dsp.stackexchange.com/questions/1277/the-current-state-of-the-art-in-audio-signal-interpolation
    Many resampling algorithms use a linear-phase low-pass filter poly-phase interpolation kernel. A minimum-phase filter interpolation kernel, as opposed to linear-phase interpolation kernel (with the same frequency response), would produce less pre-ringing during resampling to a higher sample rate, which may objectively measure as less accurate ...

c - Resampling a sound sample, what filter do I use ...

    https://stackoverflow.com/questions/4393545/resampling-a-sound-sample-what-filter-do-i-use
    Don't use linear interpolation unless both sample rates (source and destination) are well above the highest frequency in your data. It's a very poor low-pass filter. What you want is an interpolating low pass filter with a stop-band starting below half the lower of the two sample rates you are dealing with.

What is resampling? | AREFYEV Studio

    https://arefyevstudio.com/en/2019/01/11/what-is-resampling/
    What is resampling? Digital audio is always sampled, which means that any digital audio file is created at a fixed sampling rate (and resolution). The sampling frequency of the Red Book Audio CD is 44.1 kHz (16-bit resolution). Audio …

GitHub - avaneev/r8brain-free-src: High-quality pro audio ...

    https://github.com/avaneev/r8brain-free-src
    High-quality pro audio resampler / sample rate converter C++ library. Very fast, for both audio resampling and time-series interpolation. - GitHub - avaneev/r8brain-free-src: High-quality pro audio resampler / sample rate converter C++ library. Very fast, for both audio resampling and time-series interpolation.

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