DSP amp Audio Programming in NET

>> YOUR LINK HERE: ___ http://youtube.com/watch?v=GyRixqQ613A

NWaves is a .NET DSP library with a lot of audio processing functions. • NWaves is available on NuGet. • Read wiki documentation: • https://github.com/ar1st0crat/NWaves/.... • NWaves was initially intended for research, visualizing and teaching basics of DSP and sound programming. • Usually, DSP code is quite complicated and difficult to read, because it's full of optimizations (and it's actually a very good thing). NWaves project aims in particular at achieving a tradeoff between good understandable code/design and satisfactory performance. Yet, the main purpose of this lib is to offer the DSP codebase that would be: • easy to read and understand • easy to incorporate into existing projects • easy to port to other programming languages and frameworks • even possibly treated as the DSP/audio textbook. • Main features • major DSP transforms (FFT, DCT, MDCT, STFT, FWT, Hilbert, Hartley, Mellin, cepstral, Goertzel) • signal builders (sine, white/pink/red/Perlin noise, awgn, triangle, sawtooth, square, pulse, ramp, ADSR, wavetable) • basic LTI digital filters (moving average, comb, Savitzky-Golay, pre/de-emphasis, DC removal, RASTA) • FIR/IIR filtering (offline and online), zero-phase filtering • BiQuad filters (low-pass, high-pass, band-pass, notch, all-pass, peaking, shelving) • 1-pole filters (low-pass, high-pass) • IIR filters (Bessel, Butterworth, Chebyshev I II, Elliptic, Thiran) • basic operations (convolution, cross-correlation, rectification, amplification, fade / crossfade) • block convolution (overlap-add / overlap-save offline and online) • basic filter design analysis (group delay, zeros/poles, BP, BR, HP from/to LP, SOS, combining filters) • state space representation of LTI filters • FIR filter design: frequency sampling, window-sinc, equiripple (Remez / Parks-McClellan) • IIR filter design: IirNotch / IirPeak / IirCombNotch / IirCombPeak • non-linear filters (median filter, distortion effects, bit crusher) • windowing functions (Hamming, Blackman, Hann, Gaussian, Kaiser, KBD, triangular, Lanczos, flat-top, Bartlett) • periodograms (Welch / Lomb-Scargle) • psychoacoustic filter banks (Mel, Bark, Critical Bands, ERB, octaves) and VTLN warping • customizable feature extraction (time-domain, spectral, MFCC, PNCC/SPNCC, LPC, LPCC, PLP, AMS) • preconfigured MFCC extractors: HTK (MFCC-FB24), Slaney (MFCC-FB40) • LPC conversions: LPC - cepstrum, LPC - LSF • feature post-processing (mean and variance normalization, adding deltas) and CSV serialization • spectral features (centroid, spread, flatness, entropy, rolloff, contrast, crest, decrease, noiseness, MPEG7) • harmonic features (harmonic centroid and spread, inharmonicity, tristimulus, odd-to-even ratio) • time-domain characteristics (rms, energy, zero-crossing rate, entropy) • pitch tracking (autocorrelation, YIN, ZCR + Schmitt trigger, HSS/HPS, cepstrum) • chromagram (chroma feature extractor) • time scale modification (phase vocoder, PV with identity phase locking, WSOLA, PaulStretch) • simple resampling, interpolation, decimation • bandlimited resampling • wavelets: haar, db, symlet, coiflet • polyphase filters • noise reduction (spectral subtraction, sciPy-style Wiener filtering) • sound effects (echo, tremolo, wahwah, phaser, chorus, vibrato, flanger, pitch shift, morphing, robotize, whisperize) • 3D/Stereo audio (stereo panning, stereo and ping-pong delay, ITD-ILD, binaural panning) • envelope following • dynamics processing (limiter / compressor / expander / noise gate) • harmonic/percussive separation • Griffin-Lim algorithm • Karplus-Strong synthesis • PADSynth synthesis • adaptive filtering (LMS, NLMS, LMF, SignLMS, RLS) • simple modulation/demodulation (AM, ring, FM, PM) • simple audio playback and recording • • 0:00 Intro • 0:42 Transforms • 1:29 Feature extractors • 3:08 Filtering and FDA • 7:28 Operations • 7:49 Block Convolvers • 8:11 Time scale modification • 9:08 Resampling • 9:38 Audio effects • 10:31 Signal synthesis (builders) • 11:08 Outro

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