Projects sharing features with Linly Dubbing

Voice Pro is a comprehensive speech and audio processing toolkit that combines text-to-speech synthesis, voice cloning, speech recognition, and translation capabilities into a single application. At its core, the project enables users to generate natural-sounding speech from text, clone voices from short audio samples without requiring prior training data, and perform real-time speech translation across over 100 languages. The platform distinguishes itself through its integrated multimedia workflow, allowing users to download YouTube videos, extract audio, separate voice tracks, generate word

MuseTalk is a deep learning lip synchronization system designed to align video facial movements with audio tracks for high-fidelity video dubbing. It functions as an engine that matches facial expressions to audio input in real-time, enabling the modification of a speaker's lip movements to match new audio sources across different languages. The project features a distributed GPU training pipeline and a multi-stage processing workflow for refining the visual accuracy of synthetic speech. It distinguishes itself through the use of region-specific face masking and mouth openness control, which

Whishper is a graphical user interface for transcribing audio and video files into text using the Whisper model. It serves as a speech-to-text tool and subtitle file generator that converts spoken content into editable text and timed subtitle formats. The project features an integrated transcription and translation interface, allowing users to refine automated results and convert transcribed text into different languages. It includes a visual editor for correcting speech recognition errors, adjusting segment timecodes, and performing bilingual translation reviews. The system handles the full

This project is a framework for developing multimodal AI agents that function as programmable participants in real-time communication rooms. It enables the construction of agents that can see, hear, and speak by integrating speech-to-text, large language models, and text-to-speech pipelines to facilitate low-latency, natural conversations. The system is distinguished by its advanced orchestration of real-time media and conversational flow, including support for full-duplex speech, preemptive response generation, and sophisticated interruption management. It further differentiates itself throu

Autocut is a text-based video editor and automatic speech recognition tool. It allows users to cut and merge video clips by modifying a text transcript instead of using a traditional timeline. The system operates as an FFmpeg video processor and subtitle manipulation utility. It converts spoken audio into text and compacts subtitle files into simplified formats, enabling the removal of unwanted video segments by deleting corresponding sentences from a transcription file. The project covers automated video transcription, non-linear video cutting, and subtitle file management. It supports hard

MoneyPrinterPlus is an automated video production system designed for the mass creation of short-form AI content. It functions as an end-to-end pipeline that uses large language models to generate scripts, synthesize voiceovers, and produce visual assets to assemble complete videos. The project is distinguished by its ability to batch-process high volumes of unique content through automated mixing and randomized asset pairing. It includes a social media auto-publisher that uses browser simulation to automate the upload and distribution of generated videos to platforms such as TikTok and Xiaoh

VoxCPM is a multilingual speech synthesis system and text-to-speech inference server. It functions as an AI voice cloning tool and a synthetic voice designer, capable of generating natural speech across global languages and regional dialects using a GPU-accelerated audio generator. The project features a speech model fine-tuning framework that supports both full parameter updates and low-rank adaptation for customizing voice characteristics. It enables high-fidelity voice cloning from reference audio, including cross-lingual voice transfer and acoustic environment mimicry, as well as the crea

YouDub-webui is a multilingual video translator and AI dubbing pipeline manager featuring a web interface for automating video translation, audio dubbing, and subtitle burning. It utilizes a GPU-accelerated media processor to speed up audio transcription and video rendering tasks. The system implements a stage-based pipeline that converts original speech into new languages while preserving background audio through audio track mixing. It supports multiple localization workflows, including automated translation and subtitle-driven dubbing using SRT files to bypass automatic transcription phases

Tortoise-tts is a neural text-to-speech engine and voice cloning toolkit designed for high-quality audio generation. It functions as a zero-shot synthesis system, meaning it can generate speech for unseen speakers without requiring additional training or fine-tuning for each new voice. The system specializes in replicating human vocal characteristics using small sets of reference audio clips. It allows for the extraction of voice latents to mimic specific speakers, the generation of random synthetic identities, and the blending of multiple voice profiles to create hybrid vocal identities. Th

ESPnet is a comprehensive speech processing toolkit and PyTorch-based trainer designed for building end-to-end speech recognition, synthesis, and translation models. It provides a structured framework for developing automatic speech recognition systems using transducer and encoder-decoder architectures, alongside engines for text-to-speech synthesis and speech translation pipelines. The project distinguishes itself through a recipe-based workflow execution system that ensures experimental reproducibility by running standardized sequences of scripts for data preparation and model training. It

PocketSphinx is an offline speech recognition engine that converts raw audio from files or live microphone streams into written text without requiring a network connection. It functions as a speech-to-text library, a real-time transcription engine, and a voice command processor, capable of detecting and transcribing spoken commands from continuous audio streams with configurable acoustic and language models. The engine uses weighted finite-state transducers to represent acoustic, phonetic, and language models as a single search graph for efficient decoding. It employs fixed-point acoustic mod

NeMo is a multimodal AI framework and toolkit designed for the development, training, and scaling of large language models, generative AI systems, and speech-based models. It functions as an automatic speech recognition toolkit, a text-to-speech engine, and a framework for building models that process and generate combinations of text, image, and audio data. The project serves as a conversational AI orchestrator capable of managing real-time, interruptible voice interactions. It provides specialized workflows for speech translation, converting spoken audio from one language into text or speec

This project is a generative speech synthesis engine that converts text into high-fidelity human speech. It utilizes a two-stage autoregressive transformer architecture that separates semantic token prediction from acoustic detail reconstruction to balance linguistic accuracy with audio quality. The system is designed to support multilingual output and conversational AI development, enabling the generation of context-aware speech that maintains flow across multiple dialogue turns. The platform distinguishes itself through a production-ready inference server that employs continuous batching to

Kimi-Audio is a large language model audio foundation model designed to understand audio input and generate high-fidelity speech responses in real time. It functions as a unified system encompassing a text-to-speech synthesis engine and a speech-to-text transcription tool. The project enables real-time audio conversations through a multi-modal conversation loop and chunk-wise streaming detokenization to reduce playback latency. It provides controls over speech speed, accent, and emotional tone during conversational audio generation. The system covers audio intelligence capabilities, includin

Streamer-Sales is a digital avatar video generator and large language model sales agent framework. It functions as a multimodal AI pipeline that synthesizes talking-head videos using text-to-speech and face animation to create virtual spokespeople. The system differentiates itself by combining retrieval-augmented generation with agentic web searching to ground product descriptions and sales scripts in factual data and real-time information. It further integrates a multimodal interaction workflow that coordinates automatic speech recognition and text-to-speech for real-time voice conversations

This project is an AI-driven suite of tools designed to repurpose long-form video content into short-form clips. It integrates a speech-to-text engine for automated transcription, a highlighting system that ranks engaging segments based on emotional hooks, and a video processor that converts horizontal footage into vertical formats. The system distinguishes itself through intelligent video cropping that utilizes face tracking and motion smoothing to keep subjects centered. It also employs an analysis system to extract viral highlights by scoring segments for engagement and practical value. T

Fonoster is a conversational AI framework and multi-tenant communications platform as a service. It serves as a programmable voice gateway and SIP telephony platform, enabling the creation of voice-based assistants and automated communication workflows using large language models. The project distinguishes itself through a vendor-agnostic speech integration engine that abstracts speech-to-text and text-to-speech providers. It features a multi-tenant architecture that isolates telephony resources and user identities into distinct organizational workspaces. The system covers a broad range of t

mlx-audio is an audio processing toolkit built on Apple MLX that provides speech transcription, text-to-speech synthesis, voice cloning, and audio source separation using local models. It offers an OpenAI-compatible REST API and web interface for running audio generation and transcription tasks, enabling drop-in integration with existing tools that follow that endpoint structure. The toolkit supports text-prompted audio source separation, allowing specific sounds to be isolated from mixed recordings based on natural language descriptions. It also provides voice cloning from a short reference

AudioGPT is an LLM-driven audio framework and processing suite that uses large language models to orchestrate neural audio pipelines. It functions as a multimodal audio generator and processing system, integrating a collection of pretrained models to handle speech synthesis, sound generation, and audio manipulation. The system is distinguished by its ability to generate audio from diverse inputs, including text and images, and its capacity to produce synchronized talking head videos. It also operates as a neural speech translator, converting spoken language between different tongues while pre

EmotiVoice is an emotional text-to-speech engine and bilingual speech synthesizer designed to generate synthetic audio in English and Chinese. It utilizes a deep learning architecture to produce high-fidelity speech with controllable emotional states and timbres. The project includes a voice cloning framework for replicating specific speaker identities by training custom acoustic models on personal audio datasets. It employs a jointly-trained acoustic-vocoder pipeline and style-embedding-based synthesis to manage expression and reduce audio artifacts. The system covers a broad range of speec

Index-tts is a neural audio generation engine designed to convert written text into high-fidelity human speech. By utilizing deep learning models and phoneme-based sequence modeling, the system transforms text into natural-sounding audio waveforms suitable for a variety of accessibility and media applications. The platform functions as a server-side inference pipeline that provides a programmatic interface for integrating voice generation into external applications. It distinguishes itself through asynchronous audio streaming, which buffers and delivers generated speech chunks in real time to

RedditVideoMakerBot is a social media video creator and automation bot that transforms Reddit threads into short-form videos. It functions as a text-to-speech video generator, programmatically fetching posts and comments to create narrated clips with background visuals. The system integrates a content downloader for Reddit data with a voice engine that synthesizes spoken audio from written text. It manages the assembly of these components by combining image sequences and audio tracks into completed video files. The tool includes a web-based configuration interface for managing bot settings a

NarratoAI is an automated video production pipeline that uses large language models to generate scripts, voiceovers, and edited video commentary. It functions as a combined scriptwriter, voiceover generator, and video editor to streamline the creation of movie and television commentary content. The system automates the production workflow by converting input data into structured narrative scripts, synthesizing artificial speech for narration, and programmatically assembling video clips based on script timestamps. It also converts spoken audio from video files into written text for subtitles a

Autosub is a command-line media processor and automatic subtitle generator that converts audio streams from video and audio files into timed text overlays. It functions as an AI speech-to-text converter that uses OpenAI Whisper to generate synchronized subtitles. The tool includes a language translation pipeline to convert transcribed speech into target languages, enabling multilingual video captioning. It manages the process from audio-stream extraction to the serialization of final subtitle files for local storage. The system covers audio-to-text transcription, time-stamped text mapping, a

Sherpa-ONNX is an ONNX-based speech processing toolkit that provides a local speech recognition engine, an on-device voice synthesis tool, and a speaker identification framework. It is designed as a cross-platform speech API that enables speech-to-text, text-to-speech, and speaker verification tasks to be executed locally on a device without requiring network access. The project is distinguished by its ability to perform zero-shot voice cloning and speaker diarization on-device. It supports a wide range of hardware accelerations, including GPU and various NPU architectures, and provides a Web