Provides a toolkit and codebase for building, training, and deploying speech and multimodal models — Automatic Speech Recognition, Text-to-Speech, and speech-aware LLMs — with modular neural components and pre-trained checkpoints for PyTorch. Supports streaming/low-latency inference, multi-language models, and optional compiled kernels for acceleration.
Multilingual sequence-to-sequence speech model and toolkit for speech recognition, speech-to-text translation, and language identification. Offers several model sizes (tiny → large/turbo) for different speed/accuracy trade-offs and ships with a CLI and Python API for offline transcription workflows.
Web UI to train and run retrieval-based voice conversion models from small datasets (≈10 minutes), featuring top-1 feature retrieval to avoid timbre leakage, model fusion, real-time conversion, vocal separation, and multi-hardware support.
Multilingual automatic speech recognition and speech-translation model that transcribes and translates audio. Trained on a mix of weakly labeled and pseudo-labeled data (1M + 4M hours), uses 128 Mel bins and adds a Cantonese token, and supports timestamps and long-form chunking for offline ASR and translation.
Performs speaker diarization (who spoke when) with pyannote-audio: combines voice-activity detection, speaker-change and overlapped-speech detection to produce time-stamped speaker segments; compatible with Hugging Face Endpoints and ASR pipelines.
Local WebUI for video and audio dubbing: download YouTube, isolate vocals, transcribe with Whisper, translate into 100+ languages, and generate multilingual TTS or zero-shot voice clones. Uses Gradio, yt-dlp, Demucs, Whisper/Faster-Whisper, F5/E2/CosyVoice and Edge-TTS; Windows-focused with optional Azure integration.
Generates Netflix-quality single-line subtitles and optional dubbing for videos by automating download, ASR, word-level alignment, translation, terminology management and TTS integration. Emphasizes word-level alignment with WhisperX and cinematic translation/adaptation for cleaner, single-line subtitles and smoother dubbing.
Provides low‑latency on‑device speech-to-text, intent recognition, and text-to-speech for building real‑time voice agents and interfaces. Streaming-optimized models, incremental caching, multilingual TTS/ASR and cross-platform bindings (Python, iOS, Android, Linux, Raspberry Pi) target live voice use cases where sub-200ms responsiveness matters.
Captures, transcribes, and summarizes meetings entirely on the user's machine with real-time local transcription and speaker diarization. Privacy-first design keeps audio, transcripts, and models local; supports Ollama, Claude, Groq, OpenRouter or custom OpenAI-compatible endpoints.
Zero-shot, single‑reference voice cloning TTS with multilingual support (ZH/EN/JA/ES/AR), fine-grained emotion and duration control, and pronunciation hooks (Pinyin/CMU/Kana); ships model weights, Web UI and production deployment recipes for local or server use.
Provides real-time, local audio recording and transcription on macOS using Whisper and Parakeet engines, with global hotkeys and hold-to-record behavior. Includes model download, microphone selection, drag-and-drop file transcription, multilingual auto-detection and Asian-language autocorrect; Apple Silicon only.
Runs a self-hosted meeting bot and transcription API that joins Google Meet, Teams and Zoom and streams speaker-attributed transcripts in real time. Compiles meetings into a git-backed Markdown workspace and runs sandboxed agents on your infrastructure; Apache-2.0 and air-gap capable.