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── 99349 repos indexed ── | filtered by tag: #neural-audio-codec
kyutai-labs/pocket-tts ★ 9342 965 APPLICATION

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.

RATING: 8.3/10 MAINT: active DRIFT: medium
▸ 4 reusable components found
open-mmlab/Amphion ★ 10307 852 DATA_RESEARCH

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.

RATING: 7.6/10 REPRO: 6.5/10 MAINT: active DRIFT: medium
▸ 5 reusable components found
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