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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
Digital Audio Resampling Home Page - CCRMA
https://ccrma.stanford.edu/~jos/resample/
Digital Audio Resampling Home Page. Abstract: This document describes digital audio sampling-rate conversion and related concepts. Open-source software is provided, and pointers are given to related projects and papers. Detailed Contents (and Navigation) What is Bandlimited Interpolation? Free Resampling Software;
What is resampling? | AREFYEV Studio
https://arefyevstudio.com/en/2019/01/11/what-is-resampling/
Poor resampling algorithms, whether upsampling or downsampling, can introduce artifacts that are clearly audible during playback. A typical low-quality, but extremely fast algorithm will be based on linear interpolation. High-quality oversampling algorithms use more processor time because they require conversion to the frequency domain.
Resampling - Hydrogenaudio Knowledgebase
https://wiki.hydrogenaud.io/index.php?title=Resampling
Resampling or the more specific terms upsampling or downsampling usually mean sample rate conversion only. Low-quality resampling algorithms, whether upsampling or downsampling, can introduce artifacts which are clearly audible in the resampled audio file. A typical low-quality, but extremely fast resampling algorithm will just be based on ...
Which anti aliasing filter algorithm for efficient audio ...
https://dsp.stackexchange.com/questions/23416/which-anti-aliasing-filter-algorithm-for-efficient-audio-resampling
I'm building a real-time audio resampler (think pitch bend) that needs to have a several different performance vs quality configuration options. I understand that I'll need to apply a low-pass filter before resampling in order to avoid aliasing. The requirements for the first filter are this: very efficient; better than nothing
algorithm - How can I use an audio-resampler to …
https://stackoverflow.com/questions/3779608/how-can-i-use-an-audio-resampler-to-resample-if-signals
There are a lot of audio-resampling libraries available. My favorite in Secret Rabbit Code. Audio resamplers keep the correct frequency of the resampled signals, so a 4KHz sine in the source, will remain a 4KHz sine in the destination. This time, however, I …
Resampling - dspGuru
https://dspguru.com/dsp/faqs/multirate/resampling/
Resampling is usually done to interface two systems which have different sampling rates. If the ratio of two system’s rates happens to be an integer, decimation or interpolation can be used to change the sampling rate (depending on whether the rate is being decreased or increased); otherwise, interpolation and decimation must be used together ...
c - Resampling a sound sample, what filter do I use ...
https://stackoverflow.com/questions/4393545/resampling-a-sound-sample-what-filter-do-i-use
A few comments, although I'm only guessing at your actual intent: You are up-sampling at a rate 44100 times the original sample rate. For example, if your input was at 10kHz your intermediate cbuf[] would be at 441MHz which is a tad high for most audio analysis. Assuming you want cbuf[] to be at 44100Hz then you only need to create …
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