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Audio Algorithms - Adaptive Digital
https://www.adaptivedigital.com/audio-algorithms/
The AAC-LC (Advanced Audio Coding – Low Complexity) is a natural audio coding algorithm that can handle 48 channels and sample rates up to 96kHz. This coding technique uses a perceptual filter bank, a sophisticated masking model, noise …
Home [audioalgorithms.com]
https://audioalgorithms.com/
AUDIOALGORITHMS develops software for clients in the U.S., Europe and Asia. We specialize in signal processing algorithms with a focus on audio, video and real-time communication applications. We develop software for mobile, desktop, web and embedded systems.
Algorithms_in_C: audio/alaw.c File Reference
https://thealgorithms.github.io/C/df/d27/alaw_8c.html
Algorithms_in_C: audio/alaw.c File Reference. Algorithms_in_C 1.0.0. Set of algorithms implemented in C. Macros|Functions|Variables. alaw.c File Reference. A-law algorithm for encoding and decoding (16bit pcm <=> a-law). This is the implementation of G.711in C.
audio - Algorithm to mix sound - Stack Overflow
https://stackoverflow.com/questions/376036/algorithm-to-mix-sound
So, the algorithm goes: Find out the volume in which you want the output sound to be. It can be the average or maxima of one of the signals. factor = average (s1) You assume that both signals are already OK, not overflowing the 32767.0. Normalize both signals with this factor: s1 = (s1/max (s1))*factor.
5.3.8 Algorithms for Audio Companding and …
http://digitalsoundandmusic.com/5-3-8-algorithms-for-audio-companding-and-compression/
5.3.8 Algorithms for Audio Companding and Compression. 5.3.8.1 Mu-law Encoding. Mu-law encoding (also denoted μ-law) is a nonlinear companding method used in telecommunications. Companding is a method of compressing a digital signal by reducing the bit depth before it is transmitted and then expanding it when it is received. The advantage of ...
comp.dsp | Audio Compressor/AGC in C or C++
https://www.dsprelated.com/showthread/comp.dsp/47797-1.php
Audio Compressor/AGC in C or C++. ... Of the others, I have no idea how well they work. I was hoping that those on this group would have tried various algorithms. In particular, I'm wondering about fast envelope/peak detection and signal averaging. That seems to be one of the keys, esp as regards CPU cycles.
C - The Algorithms
https://the-algorithms.com/language/c
Algorithms implemented in c. Client Server; Remote Command Exec Udp Server. More
DigitalAudioResamplingHomePage
https://ccrma.stanford.edu/~jos/resample/resample.pdf
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 of operation tutorial.1 There is also free open-source software2 available in the C programming ...
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