Provides an AI-driven English learning app suite (Enjoy) that focuses on speaking practice and pronunciation evaluation. Open-source repo backing a web app, browser extensions for YouTube/Netflix, and a local-first desktop/web client design; some scoring features require the project's paid Enjoy AI service.
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.
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.
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 a 10,000-hour Sichuanese (Chuan-Yu) speech corpus with rich annotations (timestamps, speaker age/gender/emotion, SNR, DNSMOS) and unified metadata for ASR and TTS research; includes metadata.jsonl, evaluation benchmarks, and an LLM-assisted transcription pipeline.
Automatically generates complete short-form videos from a single topic: drafts script with an LLM, produces AI images/video, synthesizes multilingual TTS (including voice cloning), adds background music, and composes the final video. Supports local ComfyUI/RunningHub or direct model APIs and customizable templates.
Orchestrates low-latency, multi-stage pipelines for omni and multimodal models by running each stage with its own scheduler and using zero-copy shared memory for tensor transfer. Emphasizes per-stage bottleneck tuning and OpenAI-compatible streaming endpoints, suitable for TTS and multimodal serving.
Provides open ASR and TTS speech data for 24 Sub‑Saharan African languages to train and evaluate speech models. Includes ~1,250 hours of transcribed ASR and ~235 hours of single‑speaker TTS with train/validation/test/unlabeled splits and mixed CC-BY licenses.
Generates controllable multilingual speech from text with nine predefined timbres and custom-voice control; supports voice design, quick voice cloning and low-latency streaming (first audio packet after a single character), suitable for real-time TTS and voice-design workflows.
Turns any topic or document into an interactive, multi-agent classroom that generates slides, quizzes, interactive simulations and project-based learning activities. Includes real-time AI teachers/classmates, whiteboard drawing, TTS/ASR, PPTX/HTML export and chat-app integration via OpenClaw.