Ryuk17/SpeechAlgorithms
You can find the speech algorithms you want here
Speech Algorithms – Overview
What it is – A collection of reference implementations for classic and modern speech‑processing algorithms. The repo is organised into self‑contained folders, each holding a small demo or library that you can download (via DownGit) and run.
Why it matters – The code covers the whole pipeline that engineers encounter when building voice‑enabled products: raw‑signal handling, noise‑reduction, echo cancellation, voice‑activity detection, feature extraction, and even a few neural‑network‑based recognisers. Having a single place with ready‑to‑run examples speeds up learning and prototyping.
Main functional areas
| Area | Typical use‑case | Example implementation |
|---|---|---|
| Signal preprocessing | Resampling, alignment, pitch‑shift, watermarking | Resample, AudioAlignment, VoiceChange |
| Sound design | Generate synthetic ambience for games or VR | DesignSound/rain, DesignSound/wind |
| Acoustic echo cancellation (AEC) | Remove far‑end echo in teleconferencing | LMS‑based AEC, Kalman‑filter AEC, WebRTC AEC |
| Automatic gain control (AGC) | Keep microphone level stable | WebRTC AGC implementation |
| Noise reduction | Suppress background hiss or transient clicks | Spectral subtraction, transient‑interference suppression, WebRTC ANR |
| Speech enhancement | Create training data or improve noisy speech | Speech augmentation, DNN mask‑based enhancement, TF‑mask separation |
| Voice‑activity detection (VAD) | Detect when a speaker is talking | WebRTC VAD, LSTM‑based VAD, AMR‑coder label generator |
| Microphone‑array processing | Beamforming and source localisation for multi‑mic devices | CGMM‑MVDR beamformer, TDOA & SRP‑PHAT localisation |
| Pattern recognition | Keyword spotting, command recognition, speaker gender, environmental sounds | CNN command recogniser, XGBoost environmental classifier, deep‑learning keyword spotter |
| Speech codecs | Low‑bitrate transport of speech | Lyra (AI codec) and classic G.711 implementations |
| Quality metrics | Objective evaluation of processed speech | Speech‑quality and intelligibility measures |
How to use the repo
- Clone the repository.
- Navigate to the folder of interest (e.g.,
AudioProcess/Goertzel). - Follow any local
READMEor script instructions; most folders contain a small Python/Matlab/C++ demo that can be run directly. - For large datasets (e.g., augmentation data) the README points to a separate repo (
noise‑xorcist).
Who will benefit?
- Students & hobbyists learning DSP concepts for speech.
- Engineers prototyping voice‑assistants, teleconferencing, or AR/VR audio pipelines who need quick reference implementations.
- Researchers looking for baseline code to compare against newer models.
Limitations
- The implementations are primarily educational; they are not production‑hardened or heavily optimised.
- Some modules rely on external libraries (e.g., WebRTC, TensorFlow/PyTorch) that must be installed manually.
- No unified build system – each sub‑folder is independent.
Bottom line – Speech Algorithms is a well‑structured toolbox of classic and modern speech‑processing code, useful for learning and rapid prototyping across the full speech‑signal chain.
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