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algorithm - How can I use an audio-resampler to resample ...
https://stackoverflow.com/questions/3779608/how-can-i-use-an-audio-resampler-to-resample-if-signals#:~:text=If%20you%20resample%20a%20signal%2C%20you%20will%20either,first%2C%20or%20as%20part%20of%20the%20resampling%20filter%2Finterpolator.
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.
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;
Resampling - Hydrogenaudio Knowledgebase
https://wiki.hydrogenaudio.org/index.php?title=Resampling
Resampling or Sample Rate Conversion is required when one wants to convert a digital audio file (i.e. an analog audio signal that has already been digitized) from a given sample rate into a different sample rate (resolution can stay the same or change). Upsampling (aka interpolation) is the process of converting from a lower to higher sample rate (e.g. from 44.1 kHz to 48 kHz); …
Resampling - dspGuru
https://dspguru.com/dsp/faqs/multirate/resampling/
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. 4.1.4 Is there a restriction on the resampling factor I can use? Yes.
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
Same as for audio signal, the correct frequency should be kept, but now "correct" means same distance from center - not distance from 0 as for sampled-audio. For example [Edited]: If I resample a 64KSamples/Sec IF signal to 48KSamples/Sec, a tone in 16KHz will be in 12KHz after resampling, and a tone in 14KHz will be in 12KHz after resampling. The original range …
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
Fast resampling - which algorithm? - KVR Audio
https://www.kvraudio.com/forum/viewtopic.php?t=364821
Since I'm not really a DSP guy, I wrote a decent resampling algorithm, that is very fast, but not totally aliasing-safe in all cases. I used an hermite interpolation technique (you can find it at musicdsp.org), for lowering the pitch, and a series of different algorithms for raising the pitch that implements a simple kind of boxcar lowpass filter (the most basic one).
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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