ESPnet-VC
State-of-the-art end-to-end voice conversion for research and enterprise-scale speech transformation.
Robust, lightweight forced aligner for precise word-level audio-to-text synchronization.
Gentle is a specialized speech processing tool built on the Kaldi speech recognition toolkit, designed specifically for the task of forced alignment. Unlike standard ASR (Automatic Speech Recognition) systems that attempt to transcribe unknown audio, Gentle takes an existing transcript and an audio file to generate precise, word-level time-stamps. Its technical architecture utilizes a weighted finite-state transducer (WFST) approach, creating a dynamic language model based on the provided text. This allows for extremely high precision even in noisy environments or with non-standard accents. In the 2026 market, Gentle remains a critical piece of infrastructure for developers working on automated video captioning, lip-syncing for digital humans, and searchable audio databases. It is uniquely capable of handling 'out-of-vocabulary' segments through its 'mush' model, which attempts to find phonetic matches even when the transcript and audio diverge. As an open-source project, it provides a high-performance, local-first alternative to costly cloud-based alignment APIs, making it the preferred choice for privacy-conscious enterprise workflows and high-volume media processing pipelines.
Uses the state-of-the-art Kaldi toolkit for speech recognition, providing high-fidelity acoustic modeling.
State-of-the-art end-to-end voice conversion for research and enterprise-scale speech transformation.
End-to-End Speech Processing Toolkit for State-of-the-Art ASR, TTS, and Speech Translation.
The industry-standard open-source engine for high-precision phonetic speech alignment and acoustic modeling.
A high-performance Python library for speech data representation, manipulation, and efficient deep learning pipelines.
Verified feedback from the global deployment network.
Post queries, share implementation strategies, and help other users.
Generates a finite-state transducer (FST) on-the-fly from the input transcript.
Identifies segments where the audio doesn't match the transcript and provides best-guess phonetic alignments.
The built-in server allows it to be called via HTTP POST requests, returning JSON payloads.
Supports additional languages by swapping the acoustic model files in the data directory.
Capable of breaking down words into constituent phonemes for ultra-precise timing.
Provides a containerized version to avoid complex dependency management (e.g., Kaldi/FFmpeg).
Manually timing subtitles for long-form video is labor-intensive.
Registry Updated:2/7/2026
Character mouths must move in sync with voice-over audio.
Finding specific moments in thousands of hours of audio.