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Resampling - dspGuru
https://dspguru.com/dsp/faqs/multirate/resampling/
A practical and well-known example results from the fact that professional audio equipment uses a sampling rate of 48 kHz, but consumer audio equipment uses a rate of 44.1 kHz. Therefore, to transfer music from a professional recording to a CD, the sampling rate must be changed by a factor of: (44100 / 48000) = (441 / 480) = (147 / 160)
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
• The free Open Source Audio Library Project (OSALP)15 (LGPL) contains a C++ class based on resample. • The Speex16 speech coder/decoder (based on CELP) contains a variation of the resample algorithm in the file resample.c, is free and open-source, and is released under a BSD-style license (i.e., free for both commercial and noncommercial uses).
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;
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
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 need to resample a sampled IF/RF signal. The sampled signal contains data originally in the frequency range f0-BW/2 to f0+BW/2, shifted to 0..BW.
Resampling - Hydrogenaudio Knowledgebase
https://wiki.hydrogenaudio.org/index.php?title=Resampling
Audio on DVDs is sampled at 96kHz (24-bit resolution). 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 …
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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