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Digital Audio Basics: Audio Sample Rate and Bit Depth
https://www.izotope.com/en/learn/digital-audio-basics-sample-rate-and-bit-depth.html#:~:text=In%20theory%2C%20it%E2%80%99s%20not%20a%20bad%20idea%20to,converted%20to%20either%2044.1%20kHz%20or%2048%20kHz.
Digital Audio Resampling Home Page - Stanford University
https://ccrma.stanford.edu/~jos/resample/
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)
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
Digital Audio Resampling · Technick.net
https://technick.net/guides/theory/dar/
guides theory. Digital Audio Resampling by Julius O. Smith III. It appears that you are using AdBlocking software. The cost of running this website is covered by advertisements. If you like it please feel free to a small amount of money to secure the future of this website.
Resampling (Chapter 9) - Sampling Theory
https://www.cambridge.org/core/books/sampling-theory/resampling/7B2AE3B3AA1067E7978C8EF96C356C4C
When an audio file is played, the samples representing the recording are passed through a DAC. The analog output is then amplified and fed into a speaker. A typical DAC supports only a small number of sampling (and reconstruction) rates.
Resampling - Wikipedia
https://en.wikipedia.org/wiki/Resampling
Theory of Upsampled Digital Audio
http://www.mlssa.com/pdf/Upsampling-theory-rev-2.pdf
Theory of Upsampled Digital Audio 2 of 6 Copyright 2002 by Douglas D. Rife For perfect waveform reproduction, sampling theory requires that the digital reconstruction filter remove all frequency components above the Nyquist frequency or ½ the sampling rate of the input digital data. Because CDs encode audio data sampled at
c - Resampling a sound sample, what filter do I use ...
https://stackoverflow.com/questions/4393545/resampling-a-sound-sample-what-filter-do-i-use
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 44100/OrigSampleRate of samples in cbuf [] per sample in sampdata [].
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