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.
MockingBird is a Chinese voice cloning system implementing the SV2TTS (Speaker Verification to Text-to-Speech) framework. It provides a complete pipeline for few-shot voice cloning: a speaker encoder extracts d-vector embeddings from reference audio, a synthesizer (Tacotron with Global Style Tokens or VITS) converts text to mel-spectrograms conditioned on the speaker embedding, and a vocoder (HiFi-GAN, Fre-GAN, or WaveRNN) generates waveforms. It also includes a PPG-based voice conversion path using a Conformer-based PPG extractor and a PPG2Mel converter. The system supports multiple Chinese datasets (aidatatang_200zh, magicdata, aishell3, data_aishell) and provides web UIs (Streamlit/FastAPI), a PyQt5 toolbox, and CLI tools for training and inference. The codebase is a fork of Real-Time-Voice-Cloning adapted for Mandarin with significant additions including VITS synthesizer, multiple vocoder options, emotion extraction via wav2vec2, and a modern web API layer. While the original author has moved to a commercial product (noiz.ai), the repository shows recent commits (2026) and maintains a large community (36k stars).
Amphion is a comprehensive open-source research toolkit for audio, music, and speech generation, developed by OpenMMLab. It covers multiple generative tasks including text-to-speech (TTS), singing voice conversion (SVC), voice conversion (VC), text-to-audio (TTA), neural audio codecs, and vocoders. The codebase is organized around config-driven experiments (JSON + shell scripts) with modular base classes for trainers, datasets, and inference. It ingests a wide variety of public datasets (LJSpeech, LibriTTS, VCTK, M4Singer, Emilia, etc.) and provides extensive preprocessing pipelines for feature extraction (mel-spectrograms, F0, energy, phonetic alignments, self-supervised features from HuBERT/Whisper). The primary research focus is on high-quality, zero-shot, and controllable generation using architectures like diffusion models, flow matching, GANs, autoregressive transformers, and vector-quantized codecs. The repository also includes evaluation metrics (F0, energy, intelligibility, speaker similarity) and visualization tools (SingVisio). While primarily a research codebase, it includes inference scripts and Gradio demos for practical use. External dependencies include PyTorch, HuggingFace Transformers, WeNet, Descript Audio Codec, and various pretrained models.
This is a Python application and library for singing voice conversion and real-time voice changing, forked from so-vits-svc and rebuilt around PyTorch, PyTorch Lightning, and a Click-based CLI. The actual code provides two user-facing entry points: the `svc` CLI with commands for inference, realtime microphone conversion, dataset preprocessing, training, clustering, and cleanup, and the `svcg` PySimpleGUI desktop application for interactive and realtime inference. Inference loads a HuBERT content encoder from Hugging Face, extracts pitch using CREPE, DIO, Harvest, or Parselmouth, then runs a VITS-style SynthesizerTrn with either HiFi-GAN or MS/MB-iSTFT decoders conditioned on speaker embeddings. Despite fastapi being listed as a dependency, no HTTP API server exists in the source; the project is a desktop/CLI tool, not a web service.