For voice profile managers 255c, the strongest matches are jasonppy/voicecraft (VoiceCraft is a neural speech generation and voice cloning), babysor/mockingbird (MockingBird is a voice cloning and neural text-to-speech system) and mozilla/tts (This repository provides a comprehensive neural text-to-speech engine with). resemble-ai/chatterbox and nari-labs/dia round out the shortlist. Each is ranked by relevance to your query, popularity and recent activity.
Compare the best open-source voice profile managers on GitHub, ranked by stars and activity, to find the right tool for your project.
VoiceCraft is a neural speech generation and manipulation system consisting of a text-to-speech system, a voice cloning tool, and an audio inpainting engine. It uses a large language model approach to synthesize high-fidelity audio from text and replicate speaker identities. The system provides zero-shot voice cloning and speech editing capabilities, allowing users to modify spoken content within existing recordings. This includes an audio inpainting engine that replaces specific sections of audio with new speech while preserving the original acoustic characteristics and speaker identity. Th
VoiceCraft is a neural speech generation and voice cloning system that provides zero-shot voice cloning, text-to-speech, and audio editing capabilities matching this search.
MockingBird is an AI voice cloning tool and text-to-speech system designed to generate synthetic speech. It functions as a voice synthesis trainer for building custom models from audio datasets, a command-line generator for producing audio files, and a text-to-speech server for remote application integration. The project specializes in real-time voice cloning, which extracts vocal characteristics from short audio samples to mimic a target speaker's unique timbre. It utilizes reference-driven audio synthesis to condition pre-trained models on specific audio samples, allowing for the generation
MockingBird is a voice cloning and neural text-to-speech system that extracts vocal characteristics from audio samples for custom speech generation, fitting the speaker embedding and voice synthesis toolkit category well though lacking a distinct speaker recognition component.
This project is a comprehensive suite for neural speech synthesis, featuring a deep learning text-to-speech engine, a neural speech synthesis trainer, and a voice cloning toolkit. It provides a system for synthesizing human-like speech from text using neural network models and high-fidelity vocoders. The suite includes a speech model conversion utility to transform deep learning models between different formats for deployment across various hardware runtimes. It also provides a self-contained HTTP server to expose pre-trained text-to-speech models as a remote audio API. Capabilities include
This repository provides a comprehensive neural text-to-speech engine with built-in voice cloning and speaker embedding capabilities, perfectly matching the requirements for managing and synthesizing voice profiles.
Chatterbox is a comprehensive machine learning platform designed for multilingual speech synthesis and real-time audio generation. It functions as an engine that converts text into natural-sounding speech, capable of replicating specific human vocal characteristics and emotional expressions from short audio samples. The platform distinguishes itself through advanced control over the synthesis process, allowing for the manipulation of emotional intensity and the injection of non-verbal vocalizations such as laughter or coughing. It is engineered for low-latency performance, utilizing an optimi
Chatterbox is a comprehensive speech synthesis platform that supports voice cloning from short audio samples, neural text-to-speech, and real-time processing, making it a strong fit for your speaker embedding and voice management needs.
Dia is a generative AI audio tool and text-to-speech synthesis engine designed for the production-ready deployment of machine learning models. It provides a framework for creating lifelike synthetic speech by conditioning generation on reference audio samples to replicate specific vocal characteristics, emotional tones, and delivery styles. The system distinguishes itself through its ability to perform custom voice cloning and precise control over audio output. Users can adjust generation parameters such as temperature and guidance scale to modify the pacing, creativity, and style of the synt
Dia is a text-to-speech and generative audio engine featuring voice cloning and reference-audio conditioning, though it lacks dedicated speaker recognition utilities.
PaddleSpeech is a comprehensive toolkit of neural models for speech recognition, synthesis, and translation built on the PaddlePaddle deep learning framework. It provides a collection of frameworks and tools for converting spoken audio into written text, synthesizing natural audio from text, and performing direct speech translation. The toolkit includes specialized capabilities for keyword spotting to detect trigger words and speaker verification systems that extract unique voiceprints to identify and distinguish between individuals. It also features end-to-end translation tools that map audi
PaddleSpeech is a comprehensive speech processing toolkit that provides speaker embeddings, voice cloning, and neural text-to-speech capabilities, directly matching your search for a voice profile and speaker management framework.
GPT-SoVITS is a text-to-speech synthesis engine and voice cloning toolkit designed for generating natural-sounding human speech. It functions as a neural audio processing pipeline that maps input text to high-fidelity audio waveforms, utilizing conditional variational autoencoders and flow-based decoders to ensure expressive output. The platform distinguishes itself through its ability to perform few-shot voice cloning and cross-lingual speech generation, allowing users to maintain a specific speaker's vocal identity and emotional delivery across multiple languages. By employing cross-modal l
GPT-SoVITS is a voice cloning and text-to-speech toolkit that provides speaker embedding management, fine-tuning pipelines, and neural speech synthesis, directly matching the core requirements of this search.
Voicebox is a local speech processing system that provides text-to-speech generation, speech-to-text transcription, and voice cloning. It utilizes local machine learning inference and GPU acceleration to process audio and text data without relying on external API calls. The project features a voice cloning toolkit for creating synthetic profiles from audio samples and a timeline-based voice editor for composing multi-character conversations. It also includes an AI voice management API that allows external applications and AI agents to programmatically manage voice profiles and generate speech
Voicebox is a local speech processing system that provides voice profile management, voice cloning, and text-to-speech generation, directly matching the required speaker embedding and audio processing capabilities.
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
Sherpa-ONNX is a comprehensive speech processing toolkit providing speaker identification, local speech recognition, and on-device voice synthesis capabilities that squarely match your voice profile and embedding needs.
This project is an expressive text-to-speech foundation model and voice cloning system designed to synthesize human-like speech with emotional nuance and high fidelity. It functions as a finetunable speech model that can generate audio mimicking a specific person using a reference voice sample. The system distinguishes itself through a high-performance inference engine that utilizes memory caching and hardware compilation to reduce latency during the audio generation process. It further allows for synthesis quality improvements by training the language model on custom datasets consisting of a
This project provides a text-to-speech foundation model and voice cloning system that lets you synthesize speech mimicking a specific person using a reference voice sample, matching the core requirement for voice profile management and cloning.
This project is a neural text-to-speech engine and voice cloning toolkit designed to generate synthetic speech that mimics the vocal characteristics of a target speaker. It functions as a real-time audio synthesizer, utilizing a deep learning pipeline to convert written text into high-fidelity speech output with minimal latency. The system employs a transfer learning framework that leverages pre-trained speaker verification models to adapt synthesis to new, unseen vocal identities. By using an encoder-based speaker embedding process, the toolkit maps variable-length audio samples into a laten
This repository provides a neural text-to-speech engine and voice cloning toolkit that handles speaker embeddings and synthesis, aligning well with your search despite missing a fully-fledged voice profile manager interface.
WhisperSpeech is a multilingual speech synthesizer and neural text-to-speech system. It functions by inverting the Whisper model architecture to convert text into high-fidelity synthetic audio. The system enables voice cloning by using reference audio files to mimic specific speakers. It supports multilingual speech production, which includes the ability to generate audio across different languages and handle language switching within a single sentence. The project covers a broad range of speech capabilities, including text-to-speech generation and speech dataset preparation. It incorporates
WhisperSpeech is a neural text-to-speech system and voice cloning engine that lets you mimic specific speakers using reference audio, aligning well with your search for voice profile and embedding toolkits despite leaning toward synthesis rather than management.
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
Tortoise-tts provides zero-shot voice cloning and speaker latent extraction for speech synthesis, fitting the category well though focused primarily on text-to-speech generation rather than a standalone profile manager.
This application is a platform for AI voice synthesis and neural voice cloning. It provides a comprehensive toolkit for converting text into natural-sounding human speech by applying custom-trained neural network models to specific audio samples. The system facilitates the entire lifecycle of voice model development, including the preparation of raw audiobooks and video transcriptions into structured training datasets. It supports the training of these models on local or remote hardware, utilizing multi-GPU distributed processing to handle large-scale data and accelerate model convergence. B
This repository provides a voice cloning and neural text-to-speech platform that manages custom voice models and synthesizes speech, though it lacks dedicated features for real-time speaker recognition.
OpenVoice is a multilingual text-to-speech framework and voice cloning AI model designed for high-fidelity voice replication and low-latency audio generation. It functions as an instant speech synthesis engine that converts text to audio while replicating a specific speaker's tone and color. The system is distinguished by its ability to perform cross-lingual cloning, allowing the vocal characteristics of a reference speaker to be applied to speech in different languages regardless of the original training data. It utilizes a decoupled representation to separate the physical identity of a voic
OpenVoice is a zero-shot voice cloning and text-to-speech framework that replicates specific speaker tones and colors, though it focuses more on speech generation and cloning rather than general speaker recognition or embedding management.
Higgs-audio is a generative text-to-speech engine that transforms text into natural conversational speech using large language model architectures. It functions as a multilingual speech synthesizer capable of generating high-fidelity audio across different languages with control over emotional tone and prosody. The system includes a voice cloning tool that creates synthetic replicas of specific speakers from short audio samples without requiring extensive model training. It also provides a streaming audio API designed to deliver generated speech incrementally to minimize playback delay. The
Higgs-audio is a generative text-to-speech engine featuring zero-shot voice cloning and streaming audio capabilities, matching the core requirements for voice synthesis and cloning.
Qwen3-TTS is a large language model text-to-speech engine designed to convert written text into natural-sounding human speech. It functions as an audio tokenizer and a generative system for speech synthesis. The project features a promptable voice designer for creating synthetic vocal personas based on natural language descriptions. It also includes a zero-shot voice cloning tool that mimics a target speaker using a short reference audio clip and a transcript. The system provides a framework for speech model fine-tuning to improve speaker likeness and quality through supervised training. Add
Qwen3-TTS is a generative text-to-speech engine featuring zero-shot voice cloning and prompt-based voice profiling tools, though it lacks dedicated speaker recognition functionality.
Pocket-tts is a text-to-speech server and neural speech synthesizer that converts written text into audible speech. It includes a CPU-optimized inference engine and a voice cloning tool capable of analyzing audio samples to reproduce specific speaker characteristics. The system differentiates itself through the use of dynamic int8 quantization to reduce memory usage and increase generation speed on processors. It supports real-time speech synthesis by streaming audio chunks incrementally and utilizes voice state caching to store processed embeddings as portable files, bypassing redundant proc
Pocket-tts is a neural text-to-speech server and voice cloning tool that extracts voice characteristics to reproduce specific speakers, though it focuses primarily on synthesis rather than a comprehensive speaker recognition management system.
Bark Voice Cloning is a text-to-speech synthesis engine designed to generate natural-sounding audio and replicate specific vocal characteristics. The system utilizes a transformer-based autoregressive model to convert written text into high-fidelity speech, supporting multilingual output and expressive delivery. The project distinguishes itself through zero-shot voice cloning, which extracts speaker identity embeddings from short audio samples to condition the generative model without requiring extensive fine-tuning. It also provides specialized workflows for voice identity conversion, allowi
This repository provides zero-shot voice cloning and text-to-speech synthesis capabilities, making it a fitting tool for managing and generating specific voice profiles despite lacking a general-purpose speaker embedding management database.
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
Index-tts is a neural text-to-speech engine featuring zero-shot voice cloning and speaker embeddings, though it focuses primarily on speech generation rather than being a comprehensive profile management toolkit.
Neutts is a neural text-to-speech engine designed for real-time streaming output on edge devices such as phones and laptops. It supports voice cloning from short audio references, enabling zero-shot reproduction of a target speaker's voice, and can be fine-tuned or retrained from scratch for custom voices and styles. The system distinguishes itself through a decoder-only architecture that halves memory and accelerates generation on constrained hardware, combined with quantized model inference for reduced memory footprint. Its streaming decoder loop interleaves synthesis with playback, deliver
Neutts is a neural text-to-speech engine supporting zero-shot voice cloning from short audio samples and real-time streaming output on edge devices, though it lacks dedicated speaker recognition and general audio feature extraction tools.
Zonos is a controllable audio synthesis engine and large language model for text-to-speech. It serves as a multilingual speech generator capable of producing audio in English, Japanese, Chinese, French, and German. The system provides zero-shot voice cloning, allowing the replication of specific human voices using short audio samples. It supports the capture of nuanced behaviors, such as whispering, and provides parametric control over speaking rate, pitch, frequency, and emotional tone. The project covers a broad range of expressive speech synthesis and custom audio generation capabilities,
Zonos is a controllable audio synthesis engine that provides zero-shot voice cloning and multilingual speech generation, fitting the voice profile management and speech synthesis intent well despite lacking dedicated speaker recognition or feature extraction tooling.
This project is a GPU-accelerated speech engine and AI voice cloning tool. It functions as a text-to-speech synthesizer and voice-to-voice converter that replicates specific human voices to generate synthetic speech. The system creates digital voice profiles by analyzing short audio samples or capturing live microphone input. These profiles enable the transformation of existing audio recordings into a target speaker's voice or the synthesis of new audio from written text. The engine supports subtitle-based speech generation for batch processing and automated dubbing workflows. A web-based au
This project is a GPU-accelerated voice cloning tool and text-to-speech engine that manages voice profiles and performs audio conversion, fitting the required category well despite missing explicit speaker recognition features.
Piper is a local neural text-to-speech engine designed to convert written text into natural human speech entirely on your own hardware. By utilizing a neural synthesis framework, it operates without the need for internet connectivity, ensuring that all audio generation remains private and secure. The system distinguishes itself through a modular architecture that allows for the dynamic loading of speaker embeddings and voice configurations. This enables users to switch between various vocal personas and styles without requiring a full reload of the core synthesis model. By processing input th
Piper is a local neural text-to-speech engine that supports speaker embeddings and voice configurations for voice personalization, though it focuses primarily on synthesis rather than a complete profile management toolkit.
Spark-TTS is a deep learning text-to-speech synthesis engine designed to convert written text into high-fidelity audio. It utilizes a transformer-based architecture and autoregressive sequence modeling to generate coherent speech, transforming linguistic input into natural-sounding waveforms through neural speech codec synthesis. The platform distinguishes itself through zero-shot voice cloning, which allows users to mimic a target speaker’s unique vocal identity using only a short reference audio sample without requiring additional model training. It also features cross-lingual phonetic mapp
Spark-TTS is a deep learning text-to-speech engine featuring zero-shot voice cloning and speaker embeddings, aligning well with speech synthesis needs though lacking a standalone profile manager.
F5-TTS is a text-to-speech system that utilizes a flow matching engine and diffusion transformers to generate fluent synthetic speech. It functions as a multilingual speech synthesizer and neural training framework, providing tools for voice cloning and high-performance inference serving. The project distinguishes itself through a voice cloning toolkit capable of mimicking specific speaker characteristics and tones from reference audio clips. It supports cross-lingual generation, allowing for the synthesis of audio across various global languages or the mixing of multiple languages within a s
F5-TTS provides neural text-to-speech capabilities and voice cloning tools using diffusion transformers, aligning well with your search for voice synthesis systems even though it is focused primarily on generation rather than a broad speaker management suite.
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
This repository provides a generative speech synthesis engine supporting voice cloning, speaker embeddings, and feature extraction, making it a strong tool for voice profile and speech processing tasks despite lacking dedicated speaker recognition.
SpeechBrain is an all-in-one deep learning toolkit designed for speech and audio processing. Built as a modular library, it provides a structured environment for developing, training, and deploying neural network models across a wide range of tasks, including automatic speech recognition, speaker identification, and audio enhancement. The framework distinguishes itself through a configuration-driven approach that separates model architecture and training hyperparameters from application logic. By utilizing externalized configuration files and standardized recipes, it enables reproducible rese
SpeechBrain is a comprehensive deep learning toolkit for speech and audio processing that includes core features for speaker recognition, voice embeddings, and audio feature extraction, making it a strong fit for managing and processing voice profiles.
This project is a deep learning text-to-speech toolkit used for training and deploying neural speech synthesis models. It provides a comprehensive framework for converting written text into spoken audio, utilizing neural vocoders to transform synthesized spectrograms into high-fidelity audio waveforms. The toolkit includes a voice cloning system that replicates specific human voices by extracting speaker embeddings from short audio samples. It also supports multi-speaker audio synthesis, allowing the generation of speech across different vocal identities using specialized model architectures.
This project provides a comprehensive neural text-to-speech framework that supports voice cloning, speaker embeddings, and multi-speaker synthesis, making it well-suited for voice profile management and speech generation tasks.
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
VoxCPM is a multilingual text-to-speech and voice cloning system that provides voice profile management and fine-tuning capabilities, though it focuses primarily on speech generation rather than a general-purpose speaker embedding toolkit.
StyleTTS2 is an adversarial text-to-speech model that uses style diffusion and large speech language models to generate natural-sounding speech from text input. It combines adversarial training with large pre-trained speech models to improve speech quality and reduce artifacts, while employing a style diffusion process that extracts prosodic and timbral features from reference audio to guide speech generation. The model supports multi-speaker voice synthesis by conditioning the diffusion process on speaker-specific embeddings derived from reference utterances, enabling voice cloning and adapt
StyleTTS2 is a text-to-speech model supporting multi-speaker synthesis and few-shot voice cloning via speaker embeddings and style diffusion, though it focuses strictly on synthesis rather than a comprehensive speaker management toolkit.
This project is a comprehensive toolkit for on-device speech recognition, synthesis, and audio processing, specifically engineered for Apple Silicon. It provides a framework for building real-time, full-duplex voice agents that operate entirely offline, leveraging native hardware acceleration to maintain performance and privacy. By utilizing optimized machine learning models, the library enables local execution of complex audio tasks without reliance on external cloud services. The library distinguishes itself through its specialized focus on local, high-performance voice interaction. It incl
This Swift toolkit provides on-device speech recognition, synthesis, and audio feature extraction optimized for Apple Silicon, though its primary focus is on local voice agents rather than dedicated profile management.
Parler-TTS is a library for generating high-quality speech from text, supporting both inference and model training. It combines a transformer-based text-to-speech generator with a mel-spectrogram decoder to convert written text into natural-sounding audio. The project distinguishes itself through text-conditioned voice control, which allows speaker attributes like gender, pitch, speaking rate, and style to be adjusted via a natural-language description. It also includes speaker embedding selection for maintaining voice identity across multiple generations, and a fine-tuning recipe system that
Parler-TTS is a text-to-speech library that includes speaker embedding selection and voice attribute conditioning, fitting the voice profile and generation scope well even though dedicated speaker recognition and cloning are not its primary focus.
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 multimodal framework provides comprehensive tools for speech recognition, text-to-speech, and audio processing, though it focuses broadly on neural model training rather than a dedicated voice profile manager.
This project is a neural text-to-speech system and voice trainer that converts written text into spoken audio across a variety of global languages and regional dialects. It functions as an ONNX-based engine capable of performing fast offline inference and uses a phoneme-based controller to manage precise pronunciation. The system distinguishes itself through a comprehensive toolkit for neural voice training, allowing for the creation of custom single-speaker or multi-speaker models. It supports the export of these models to a standardized open format and provides hardware acceleration via gra
This repository provides a neural text-to-speech engine and voice trainer with multi-speaker support and voice profile management, though it focuses more on synthesis than full speaker recognition pipelines.
ChatTTS is a conversational text-to-speech generative model designed to convert written dialogue into natural sounding audio. It functions as a multilingual speech synthesis framework capable of producing human-like audio across different languages and speaker profiles. The system is distinguished by its ability to generate interactive dialogue with realistic vocal nuances. It utilizes a speech nuance controller to insert specific tokens that trigger non-verbal elements, such as laughter, pauses, and interjections, during the synthesis process. The project includes a streaming audio generato
ChatTTS is a conversational text-to-speech generative model that handles speaker profiles and speech synthesis, though it is primarily a TTS model rather than a comprehensive profile manager.
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
ESPnet is a comprehensive speech processing toolkit that includes support for voice identity conversions and speech synthesis, matching the core requirements for managing and processing voice profiles and speech models.
This project is a TensorFlow voice conversion framework and deep learning audio toolkit designed for neural voice style transfer. It functions as a speech synthesis engine that transforms the spectral characteristics of a source speaker's voice to match the vocal identity of a target speaker. The system employs a phoneme-based approach to voice conversion, classifying audio utterances into speaker-independent phonemes and resynthesizing them using a target voice. This pipeline allows for the transformation of voice characteristics by mapping audio features between different speakers. The too
This project is a deep learning voice conversion framework that maps audio features between speakers for neural voice style transfer, fitting the voice profile and cloning category well despite lacking some explicit text-to-speech features.
Voice Clone is an AI-powered tool that lets you instantly clone any voice in just seconds. Built for creators, developers, and businesses, it delivers high-quality, natural-sounding results with a simple API and web interface. Perfect for podcasts, videos, games, and more — no recording studio required.
Voice Clone is an AI-powered voice cloning tool that provides a web interface and API for generating natural-sounding cloned voices, though it lacks the broader feature depth of a full speaker embedding toolkit.
| Repository | Stars | Language | License | Last push |
|---|---|---|---|---|
| jasonppy/voicecraft | 8.5K | Jupyter Notebook | NOASSERTION | |
| babysor/mockingbird | 36.9K | Python | NOASSERTION | |
| mozilla/tts | 10.2K | Jupyter Notebook | MPL-2.0 | |
| resemble-ai/chatterbox | 22.8K | Python | mit | |
| nari-labs/dia | 19.3K | Python | Apache-2.0 | |
| paddlepaddle/paddlespeech | 12.6K | Python | Apache-2.0 | |
| rvc-boss/gpt-sovits | 58.7K | Python | MIT | |
| jamiepine/voicebox | 30K | TypeScript | MIT | |
| k2-fsa/sherpa-onnx | 13K | C++ | Apache-2.0 | |
| metavoiceio/metavoice-src | 4.2K | Python | Apache-2.0 |