Unsloth is a full-stack, local-first AI platform for fine-tuning and inference of large language models (LLMs), diffusion models, and audio models. It consists of two primary components: (1) the core `unsloth` Python library, which provides highly optimized LoRA/QLoRA/RL training with custom Triton kernels, and (2) the Unsloth Studio application, a Tauri desktop app with a React frontend (TanStack Router, Zustand) and a FastAPI backend (SQLite, structlog). The platform enables users to discover models from the Hugging Face Hub, download and manage them locally, run inference with chat, image/video generation, and RAG, fine-tune with a visual recipe editor, and export in multiple formats (GGUF, NVFP4). It also integrates external LLM providers (OpenAI, Anthropic, Ollama) and supports agentic tool loops via MCP and Codex transports. The codebase is extensively tested with both unit tests and source-level contract tests, and includes cross-platform desktop installers for Windows, macOS, and Linux.
Moonshine Voice is an open-source AI toolkit for building real-time voice agents and applications with on-device speech-to-text, text-to-speech, and voice cloning capabilities. The system emphasizes privacy, low latency, and offline operation through a cross-platform C++ core with language bindings for Python, Swift, Android, and WebAssembly. It includes optimized models for various platforms (from microcontrollers to desktops), multilingual grapheme-to-phoneme conversion, speaker diarization, and embedding-based phrase matching. The architecture features modular components for audio processing, model inference, and language-specific rules, with examples demonstrating microphone transcription, streaming TTS, agent workflows, and embedded deployments on RP2350 microcontrollers.
This repository is a Home Assistant custom component that adds OpenAI-compatible text-to-speech to a HA instance. It is distributed as an integration for HACS and registered as a config-flow service integration. In production it exposes TTS entities, an `openai_tts.say` service that announces text to media players with automatic volume save/restore and optional chimes, and an `openai_tts.set_api_key` service for credential rotation. The architecture is a HA integration split into focused modules: config flow and profile subentries, a streaming TTS engine, a sentence-level audio pipelining layer, an ffmpeg-based loudness pipeline, push-driven API health tracking, cache and validation logic, repairs, diagnostics, and a large announcement orchestration module. Provider support is driven by declarative presets for OpenAI, Mistral, Groq, Lemonfox, Kokoro, Chatterbox, and custom OpenAI-compatible endpoints, and the code intentionally modern-targets HA's streaming TTS contract rather than retaining legacy fallback paths.
OpenMontage is an open-source, agentic video production platform where an AI coding assistant (Claude Code, Cursor) orchestrates the entire pipeline. It provides a library of 100+ tools (video generation, audio, graphics, enhancement, analysis, avatar, subtitles) behind a scored provider selection layer, and 10+ production pipeline manifests (animated explainer, cinematic, documentary montage, talking head, etc.) that define stage order, approval gates, and tool assignments. The core architecture is a Python framework with a BaseTool class, ToolRegistry for auto-discovery, pipeline loader, checkpoint system, and a Remotion-based video composer. It also includes a Backlot living storyboard server for live production monitoring and an Ink Theater procedural puppetry engine. While primarily an application, it contains data/research elements such as ManimGL/ManimCE educational animation examples covering transformer internals, quantum mechanics, and ML concepts, as well as a CLIP-based corpus builder for documentary montage pipelines.
Piper is a fast, local neural text-to-speech engine that uses VITS models exported to ONNX for inference. The repository provides a comprehensive application stack: a Python package (`piper-tts`) with a clean API for voice synthesis, a CLI tool, an HTTP server with a web interface (including viseme animation), and a C/C++ shared library (`libpiper`) for embedding in other applications. It also includes a full training pipeline using PyTorch Lightning, allowing users to fine-tune or train new voices from scratch, with support for multi-speaker models, custom phonemes, and VAD silence trimming. The architecture is modular: phonemization is handled by language-specific backends (espeak-ng, Chinese pinyin/g2pW, Hebrew Nakdimon, Japanese OpenJTalk, Arabic tashkeel), and the core inference engine uses ONNX Runtime for efficient, cross-platform execution. The codebase is well-structured with clear separation of concerns, extensive testing, and CI/CD pipelines that build wheels for all major platforms. Although primarily an application, the training code (VITS model implementations, dataset caching, monotonic alignment Cython) exhibits data/research characteristics, making it a hybrid project. The repository is actively maintained, with a responsive issue tracker and regular updates, though the last commit date appears anomalous.
aspeak is a text-to-speech client for Azure TTS API, written in Rust. It provides a CLI, a Rust library, and Python bindings. The core functionality synthesizes speech from plain text or SSML using either the REST or WebSocket API. The architecture is modular: the library defines two synthesizer types (`RestSynthesizer` and `WebsocketSynthesizer`) with a unified trait, and a CLI wraps these for command-line use. The Python binding exposes a `SpeechService` class. The codebase demonstrates solid Rust engineering with builder patterns, comprehensive error handling, and support for proxies, custom audio formats, and rich SSML options. It is well-tested with unit and integration tests, has CI/CD pipelines, and is actively maintained.
VibeVoice is a production-grade speech AI framework from Microsoft Research, providing both text-to-speech (TTS) and automatic speech recognition (ASR) capabilities. The repository delivers inference and deployment infrastructure for frontier speech models built on a Qwen2.5 LLM backbone with continuous speech tokenizers at 7.5 Hz and a next-token diffusion framework. It includes a vLLM plugin for high-performance serving with OpenAI-compatible APIs, FastAPI/WebSocket streaming ASR, real-time TTS demos, and Gradio-based interactive applications. The codebase is primarily an application deployment toolkit, with embedded research code for model architectures, tokenizers, and diffusion schedulers. It supports long-form multi-speaker TTS (up to 90 minutes), streaming TTS, and single-pass ASR with diarization across multiple languages. The repository also provides LoRA fine-tuning scripts for ASR adaptation and comprehensive demo scripts for both local and cloud deployment.
Chatterbox is a state-of-the-art open-source text-to-speech library by Resemble AI, providing multiple TTS models (Chatterbox, Chatterbox-Multilingual, Chatterbox-Turbo, Chatterbox-Nano) and voice conversion. The architecture is modular: a T3 transformer generates speech tokens from text conditioned on speaker embeddings and optional reference audio; an S3Gen pipeline converts tokens to mel-spectrograms via conditional flow matching (CFM) and then to waveforms via a HiFiGAN vocoder. Voice cloning is achieved through a separate voice encoder (LSTM-based speaker embedding) and S3 tokenizer for prompt conditioning. The library supports 23+ languages with sophisticated multilingual tokenization (Cangjie for Chinese, hiragana for Japanese, diacritics for Hebrew, stress for Russian, Jamo decomposition for Korean). All models include imperceptible neural watermarking via Perth. The codebase is well-structured, with clean separation of concerns, efficient inference (KV caching, CFG, distilled single-step decoder for Turbo/Nano), and pinned dependencies. It is primarily an application library but includes research-grade model implementations.
ESPnet is an end-to-end speech processing toolkit built on PyTorch, covering a wide range of tasks: automatic speech recognition (ASR), text-to-speech (TTS), speech translation (ST), speech enhancement (SE), speaker recognition, spoken language understanding (SLU), singing voice synthesis (SVS), speech language modeling (SpeechLM), and more. The repository contains both a core Python library (`espnet2/`, `espnet3/`) and a large collection of reproducible recipes (`egs2/`, `egs3/`) for dozens of public corpora and research challenges. The architecture is modular: each task defines its own model, loss, training, and inference via a `Task` class, with abstract base classes for encoders, decoders, frontends, tokenizers, and losses. Configuration is YAML-driven, and the training pipeline uses PyTorch Lightning (in ESPnet3) or a custom trainer (ESPnet2). The system provides CLI tools (`espnet` with subcommands like `transcribe`, `synthesize`, `enhance`, `translate`) and a Python API (`Speech2Text.from_pretrained()`) for inference. It also includes Docker images for deployment and Hugging Face Spaces demos. The repository is hybrid: it is primarily a software framework/library (application) with a strong research character, as evidenced by the extensive recipe collection, benchmark results, and support for state-of-the-art models (Conformer, HuBERT, Whisper, VALL-E). The recipe system follows Kaldi-style data preparation and stage-based pipelines, enabling reproducible experimentation. External dependencies include PyTorch, torchaudio, sentencepiece, and optional components like flash-attn, fairseq, and s3prl. The codebase is actively maintained with CI across multiple OS and Python versions, unit tests, and integration tests.
YouDub WebUI is a full-stack video localization platform that enables users to import videos, transcribe speech via Whisper, translate via OpenAI-compatible APIs, clone voices with VoxCPM2, separate audio with Demucs, align subtitles with Qwen3 Forced Aligner, and export dubbed videos or subtitles. The architecture is plugin-based (Cordis framework) with a Node.js/TypeScript host orchestrating Python subprocesses for ML inference, a Next.js web frontend, and native desktop apps for macOS (Swift) and Windows (C#). The system features a task engine with queuing, cancellation, retry, and partial rerun; a worker protocol using NDJSON over stdin/stdout; and a comprehensive setup service that downloads and verifies model weights. While primarily an application, it includes research-grade model consumption and detailed validation records documenting real-media runs with SHA-256 hashes and human listening feedback. The codebase is actively maintained with recent commits, CI/CD pipelines, and extensive unit tests for contracts, protocols, and permissions.
VieNeu-TTS is an advanced on-device Vietnamese text-to-speech library with instant voice cloning, supporting bilingual (Vietnamese/English) synthesis. The system provides a Python SDK (`vieneu`) with multiple backends: v3 Turbo (default, 48kHz, ONNX or PyTorch), v3 Nano (lightweight 24kHz, ONNX only), and legacy v1/v2 (deprecated). It includes a Gradio web UI (`vieneu-web`), a FastAPI streaming server, and a browser-based HTML/JS client. The core architecture is a factory pattern (`Vieneu(mode=...)`) that returns the appropriate engine, all inheriting from a shared base class. Inference can run entirely torch-free on CPU via ONNX Runtime, with optional GPU acceleration (CUDA) for batched generation and lower latency. Feature highlights include instant voice cloning from a 3β8 second reference clip using a speaker encoder (ONNX) and denoiser, inline emotion cues (`[cΖ°α»i]`, `[thα» dΓ i]`), frame-level streaming, SRT subtitle-to-speech, and LoRA fine-tuning for custom voices (requires ~6GB GPU). The system is deployed via Docker with CPU and GPU profiles, and has CI/CD pipelines for testing and PyPI publishing. The codebase is well-structured with clear separation of concerns, though it carries some legacy weight from older model versions and a few deprecated parameters. Other notable components include a sliding-window repetition penalty, a babble guard heuristic for short utterances, and a voice persistence system for user-defined presets.
WhisperPlus is a Python library that provides enhanced speech processing capabilities built around Whisper models. It offers a modular pipeline architecture for speech-to-text transcription with optimizations like quantization and flash attention, text summarization (including long-text handling), speaker diarization, text-to-speech conversion, video auto-captioning, and retrieval-augmented generation for chatting with video content. The library integrates with Hugging Face Transformers, Apple MLX for optimized inference on Apple silicon, and various external services like YouTube downloading and vector databases for RAG functionality. The architecture follows a modular design with clearly separated pipelines for each functionality: core speech-to-text (whisper.py), summarization (summarization.py, long_text_summarization.py), text-to-speech (text2speech.py), diarization (whisper_diarize.py), auto-captioning (whisper_autocaption.py), and RAG chatbots (lancedb_chatbot.py, autollm_chatbot.py). Utility modules handle YouTube media downloading (download_utils.py) and text formatting (text_utils.py). The library provides both programmatic API access and a Gradio-based web interface (app.py) for interactive use, along with a CLI entry point (cli.py). Data flows typically begin with audio/video input (from files or YouTube URLs), proceed through specialized processing pipelines (transcription, summarization, etc.), and produce text, audio, or video outputs. For RAG features, transcripts are processed into vector stores enabling question-answering interactions. The library relies on established dependencies including torch, transformers, ffmpeg, pyannote.audio for diarization, and MLX-specific libraries for Apple silicon optimization, demonstrating practical integration of state-of-the-art models into an accessible toolkit.
Irodori-TTS is a state-of-the-art Japanese text-to-speech system built on rectified flow diffusion over DACVAE audio latents. It provides a complete training and inference stack: `train.py` supports full fine-tuning, LoRA, speaker inversion, and MeanFlow distillation with distributed training and WSD scheduling; `infer.py` offers a rich CLI with CFG, Sway Sampling, duration scaling, and watermarking; `gradio_app.py` and `gradio_app_voicedesign.py` expose interactive UIs for reference-based cloning and caption-conditioned voice design. The architecture is modular (`irodori_tts` package) with attention dispatch (FA3/SDPA), a hybrid Muon+AdamW optimizer, codec abstraction, and caching-friendly `InferenceRuntime`. While highly engineered for production use (INT4/INT8 quantization, safetensors, HuggingFace Hub integration), it remains a research-oriented codebase with experimental sampling algorithms and many YAML configs for different model sizes/versions. The code is clean and well-factored, though `train.py` is a large monolith and external dependencies (DACVAE, silentcipher) are pinned to specific commits, creating ecosystem fragility.
MoneyPrinterTurbo is a production short-video generation platform. Given a topic or script, it orchestrates LLM-based script/keyword generation, multi-provider TTS (Edge, Azure, Gemini, ElevenLabs, etc.), video material sourcing (Pexels, Pixabay, Coverr, several AI video generators), subtitle generation (Whisper or TTS timestamps), background music, and final video composition via moviepy/FFmpeg. It exposes three consistent interfaces: a FastAPI REST API, a Streamlit WebUI, and a CLI. All three share the same underlying task pipeline in `app/services/task.py`, which manages state, progress, and cross-posting to YouTube/TikTok/Instagram. The architecture is a service-oriented monolith: the core pipeline is a linear sequence of stages (script β terms β audio β subtitles β materials β video β optional BGM/cross-post). External integrations are abstracted behind provider-specific modules (e.g., `voice.py`, `material.py`, `bgm.py`). A Redis-backed or in-memory state manager tracks task progress. The configuration is runtime-editable and stored in a TOML file. The WebUI is a large Streamlit single-page application with dozens of provider-specific controls, billing confirmations, and i18n support for 12+ languages. The codebase demonstrates strong engineering practices: extensive test coverage (200+ test files), structured error handling with per-provider error classes, atomic file writes, path traversal protection, and security-conscious API key handling. However, it also shows signs of rapid growth: the WebUI is a monolithic 1300+ line script, provider integration logic is occasionally duplicated, and the dependency list is large (many third-party APIs). The repository is actively maintained with recent commits and good issue triage. It is not a research project; there are no datasets, model training, or analytical queries.
ChatTTS is a generative speech model application designed specifically for dialogue scenarios, functioning as a Text-to-Speech (TTS) library with integrated serving infrastructure. The core logic resides in the ChatTTS package, where `ChatTTS/core.py` defines the main inference interface. The architecture combines a GPT-based language model (`ChatTTS/model/gpt.py`) for token generation, a DVAE (`ChatTTS/model/dvae.py`) for audio compression, and Vocos as a vocoder. The system supports multiple speakers, prosodic control (laughter, pauses), and text normalization for Chinese and English. Deployment is handled through multiple entry points: a FastAPI-based API server (`examples/api/main.py`), a Gradio WebUI (`examples/web/webui.py`), and CLI tools (`examples/cmd/run.py`). A significant engineering highlight is the integration of vLLM (`ChatTTS/model/velocity/`) to optimize inference throughput for the GPT component. The repository includes robust asset management (`ChatTTS/utils/dl.py`) for model weights with SHA256 verification and ONNX export capabilities (`examples/onnx/exporter.py`) for cross-platform deployment. While the underlying model is research-grade (trained on 100k+ hours of audio), the repository structure prioritizes application delivery over experimental research. It provides a stable library interface with clear separation between model logic, audio processing (`tools/audio/`), and serving layers. The codebase demonstrates high craftsmanship in handling stochastic generation reproducibility (`tools/seeder/ctx.py`) and speaker embedding management (`ChatTTS/model/speaker.py`), making it a production-ready TTS solution rather than a raw research notebook collection.
pocket-tts is a production-ready text-to-speech library and application optimized for CPU inference. It provides a Python API, CLI (generate, serve, export-voice), and a FastAPI web server with streaming WAV output. The core architecture uses a FlowLMModel (transformer + flow matching with LSD) and a Mimi neural audio codec, with streaming inference via KV caching. It supports multiple languages (English, French, German, Portuguese, Italian, Spanish), voice cloning from audio prompts, and quantization (int8 dynamic) for efficiency. The repository also includes a full training pipeline (data preparation, alignment, tokenizer training, distributed training, distillation) and evaluation scripts. While primarily an application, the training code adds a research dimension. The codebase is modular and well-documented, with reusable components like stateful streaming modules, quantization utilities, and rotary embeddings. Some technical debt exists (deprecated quantization backend, hardcoded thread settings, multiple config versions), but overall it is a solid, actively maintained project.
Foliate is a desktop e-book reader for GNOME/Linux, built with GTK4, Libadwaita, and GJS (GNOME JavaScript). It supports EPUB, Mobipocket, Kindle, FB2, CBZ, and PDF formats, using WebKitGTK to render book content via a custom JavaScript rendering engine (foliate-js). The application provides a full-featured reading experience with paginated and scrolled modes, customizable fonts and themes, annotations (highlights, notes, bookmarks), text-to-speech via Speech Dispatcher, OPDS catalog browsing, and dictionary/Wikipedia lookups. Architecture: The main entry point is src/main.js, which initializes the GJS application and loads the GResource-bundled source files. The core UI is defined in src/app.js (window creation, file handling) and src/book-viewer.js (reading view with sidebar, search, navbar, TTS). The library view (src/library.js) manages book collections and OPDS integration. Reading state (annotations, bookmarks, progress) is persisted per-book as JSON files in the user's data directory. The rendering pipeline lives in the WebView: src/reader/reader.js handles content display, CSS injection, and footnotes, while the foliate-js submodule parses e-book formats into a pageable model. External dependencies: GNOME platform (GTK 4.12, Libadwaita 1.8, WebKitGTK 6.0, GJS 1.82), Speech Dispatcher for TTS, Tracker (optional) for file indexing, and third-party APIs (Wiktionary, Wikipedia, Google Translate) for lookup tools. The build system uses Meson, with Flatpak and Snap packaging provided. Internationalization across 30+ languages is maintained via gettext PO files. Data flows: User opens a file β book is parsed by foliate-js into a structured document model β rendered in WebView β user interactions (scroll, search, annotate, bookmark) are communicated back to GJS via the WebView's async RPC pattern (exec/iter/provide) β state is persisted as JSON files. OPDS catalogs are fetched via HTTP and rendered in a separate WebView frontend (src/opds/main.html).
ReadAny is a full-featured, cross-platform e-book reader application with deep AI integration. It supports reading EPUB, PDF, MOBI, FB2, and other formats via the modified foliate-js engine, and provides text-to-speech, annotations, reading statistics, and cross-device sync (WebDAV, S3, LAN). The AI subsystem offers chat with RAG (semantic search over book content), context-aware tools, and execution of user-defined skills. The application follows a local-first architecture with robust offline capabilities and optional cloud sync. The codebase is structured as a pnpm monorepo with four main packages: 'core' (shared business logic, types, stores, AI, RAG, sync, stats, TTS), 'app' (Tauri desktop client with React+shadcn UI), 'app-expo' (Expo/React Native mobile client), and 'cli' (agent integration via MCP and managed shims). The application exhibits high craftsmanship in its platform abstraction layer, persistent store pattern, and hybrid search engine. However, there is significant platform-driven code duplication (desktop vs. mobile UI components) and a complex dependency graph with multiple patched packages, indicating some architectural debt.
This repository implements a fully autonomous pipeline that generates, produces, and uploads educational YouTube videos daily. It uses the Gemini 2.5 Flash API to create lesson scripts and curricula, gTTS for narration, Pexels for background imagery, and MoviePy/Pillow for video rendering. The pipeline runs on a GitHub Actions cron schedule, requiring no manual intervention. It produces both a long-form lesson video and a YouTube Short, uploads them with custom thumbnails, and updates the content plan repository. The code is well-structured for its purpose, with proper error handling, retry logic for TTS rate limits, and credential management for YouTube OAuth. While not a reusable library, it demonstrates good software engineering practices for a CI/CD-based content generation system.
Dograh is an open-source, self-hostable voice AI platform for building, deploying, and monitoring voice agents. It provides a visual workflow builder (node-based graph editor) for designing call flows, with deep integration across telephony providers (Twilio, Plivo, Vonage, Telnyx, Cloudonix, Vobiz, Asterisk ARI), STT/TTS/LLM models (OpenAI, Deepgram, Google, Azure, Cartesia, ElevenLabs, etc.), and real-time audio pipelines via the Pipecat library. The backend is a FastAPI monolith with PostgreSQL, Redis, and object storage, complemented by separate worker services (ARQ for background tasks) and a campaign orchestrator for bulk outbound calling. The system supports multi-tenancy, call recording, knowledge base/RAG, QA analysis, embeddable chat widgets, and an MCP server for AI-assisted workflow management. SDKs for Python and TypeScript enable programmatic workflow creation and outbound call triggering. While the core is a full-stack application, it includes evaluation frameworks for STT providers and post-call QA analysis, giving it a hybrid character but dominated by production application concerns.