Search Results for "speaker"
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25 projects for "speaker" with 2 filters applied:
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FireRedTTS-2
Long-form streaming TTS system for multi-speaker dialogue generation
...FireRedTTS2 supports multilingual output and speaker flexibility, enabling scenarios that involve language switching, cross-lingual voice cloning, and expressive dialogue generation that maintains consistency over longer utterances. -
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OuteTTS
Interface for OuteTTS models
...It also includes a notion of speaker profiles: you can create a speaker from a short audio sample, save it as JSON, and reuse it for consistent voice identity across generations and sessions. For best quality, the model is designed to work with a reference speaker clip and will inherit emotion, style, and accent from that reference. -
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VoxCPM
TTS for Context-Aware Speech Generation and True-to-Life Voice Cloning
...Trained on a large 1.8-million-hour bilingual corpus, VoxCPM can infer appropriate speaking style from context, dynamically adjusting intonation, rhythm, and emotional tone. It supports zero-shot voice cloning from a short reference audio clip, capturing timbre, accent, and pacing to closely mimic a target speaker without per-speaker fine-tuning. -
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VibeVoice
Open-source multi-speaker long-form text-to-speech model
VibeVoice-1.5B is Microsoft’s frontier open-source text-to-speech (TTS) model designed for generating expressive, long-form, multi-speaker conversational audio such as podcasts. Unlike traditional TTS systems, it excels in scalability, speaker consistency, and natural turn-taking for up to 90 minutes of continuous speech with as many as four distinct speakers. A key innovation is its use of continuous acoustic and semantic speech tokenizers operating at an ultra-low frame rate of 7.5 Hz, enabling high audio fidelity with efficient processing of long sequences. ... -
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Scriberr
Self-hosted AI audio transcription
...Unlike cloud-based transcription services, Scriberr runs entirely on the user’s machine, ensuring that sensitive recordings are never sent to third-party servers and remain fully under user control. It leverages modern speech recognition models such as Whisper and other advanced architectures to deliver precise transcripts with word-level timing and speaker identification. The application includes a polished user interface that simplifies the management of recordings, transcripts, and annotations, making it suitable for both casual users and professionals handling large volumes of audio. Beyond transcription, Scriberr also integrates features such as summarization, tagging, and interaction with language models, allowing users to extract insights from conversations or meetings efficiently. -
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Whisper-WebUI
A Web UI for easy subtitle using whisper model
...It supports multiple input sources including local files, YouTube content, and microphone input, making it versatile for different workflows. Whisper WebUI also includes advanced preprocessing and postprocessing features such as voice activity detection, background music separation, and speaker diarization, enabling more accurate and structured outputs. -
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DramaBox
super expressive prompting model based on ltx2.3
...It also supports additional LoRA training on top of DramaBox, making it possible to adapt the model for a specific speaker, language flavor, or performance style. DramaBox is aimed at developers, researchers, and audio creators who need highly expressive English TTS for character dialogue, narrative audio, prototyping, or voice experimentation. -
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MOSS-TTS Family
MOSS‑TTS Family open‑source speech and sound generation model
MOSS-TTS is an open-source speech and sound generation model family built for high-fidelity, expressive, and production-oriented audio workflows. It covers long-form speech, voice cloning, multi-speaker dialogue, voice design, environmental sound effects, and real-time streaming TTS. The project is designed for complex real-world use cases where a single speech model may not be enough. Its flagship model focuses on stable long speech generation, multilingual and code-switched synthesis, pronunciation control, and zero-shot voice cloning. ... -
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AutoSubs
Instantly generate AI-powered subtitles on your device
AutoSubs is an open-source, AI-powered subtitle generation tool that enables users to automatically transcribe audio and video content into accurate, editable subtitles directly on their device. It supports both standalone usage and integration with professional video editing software such as DaVinci Resolve, allowing creators to generate and edit subtitles within their existing workflows. The tool leverages speech-to-text models, including OpenAI Whisper, to produce high-quality... -
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OmniVoice
High-Quality Voice Cloning TTS for 600+ Languages
The OmniVoice project is a cutting-edge multilingual text-to-speech system designed to generate high-quality speech across more than 600 languages. Built on a diffusion language model-style architecture, it combines scalability with strong performance, enabling both natural-sounding voice synthesis and efficient inference speeds. One of its most notable capabilities is zero-shot voice cloning, allowing users to replicate a speaker’s voice using only a short reference audio clip. In addition,... -
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ChatTTS_colab
One-click deployment (including offline integration package)
...It has first-class support for long-form audio generation, making it suitable for audiobooks, podcasts, or long narration tasks. The project also implements multi-speaker or role-based reading, letting users assign different voices to different characters in a script and even use a large language model to generate that script in one step. -
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PowerPoint-ist
Web presentation editor replicating many PowerPoint features online
PPTist is a web-based presentation editing application designed to replicate many of the commonly used features found in traditional slide presentation software. It allows users to create, edit, and present slide decks directly within a web browser while maintaining a desktop-like editing experience. PPTist is built with Vue 3 and TypeScript and focuses on providing a highly interactive slide editing environment with extensive customization and extension potential. PPTist supports a wide... -
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ESPnet
End-to-end speech processing toolkit
ESPnet is a comprehensive end-to-end speech processing toolkit covering a wide spectrum of tasks, including automatic speech recognition (ASR), text-to-speech (TTS), speech translation (ST), speech enhancement, speaker diarization, and spoken language understanding. It uses PyTorch as its deep learning engine and adopts a Kaldi-style data processing pipeline for features, data formats, and experimental recipes. This combination allows researchers to leverage modern neural architectures while still benefiting from the robust data preparation practices developed in the speech community. ... -
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MARS5
MARS5 speech model (TTS) from CAMB.AI
...The model is built to handle prosodically challenging content such as sports commentary, anime dialogue, and other high-energy or highly varied speech patterns with realistic rhythm and intonation. To control speaker identity, MARS5 uses a short reference audio clip, typically between 2 and 12 seconds, from which it learns the voice characteristics. It supports two main inference modes: shallow clone, which is faster and only needs the reference audio, and deep clone, which additionally uses the transcript of the reference audio to increase similarity and naturalness at the cost of more computation. -
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MetaVoice-1B
Foundational model for human-like, expressive TTS
MetaVoice — in the form of its source repository “metavoice-src” — is a large-scale text-to-speech (TTS) model. Specifically, the base model (MetaVoice-1B) uses around 1.2 billion parameters and has been trained on a massive dataset — reportedly around 100,000 hours of speech data. The goal is to provide human-like, expressive, and flexible TTS: able to generate natural-sounding speech that can handle diverse inputs and likely generalize over voice styles, intonation, prosody, and perhaps... -
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SoniTranslate
Synchronized Translation for Videos
SoniTranslate is a video translation and dubbing system that produces synchronized target-language audio tracks for existing video content. It provides a web UI built with Gradio, allowing users to upload a video, choose source and target languages, and then run a pipeline that handles transcription, translation and re-synthesis of speech. Under the hood, it uses advanced speech and diarization models to separate speakers, align audio with timecodes and respect subtitle timing, which lets... -
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StyleTTS 2
Towards Human-Level Text-to-Speech through Style Diffusion
...The architecture uses a two-stage training process and leverages an auxiliary speech language model to guide generation toward more natural and coherent utterances. StyleTTS2 supports both single-speaker and multi-speaker configurations, with the ability to sample or transfer styles from reference audio, making it powerful for expressive TTS and character voices. The repository includes training scripts, configuration files, and pre-trained auxiliary modules such as a text aligner, pitch extractor, and PL-BERT-based linguistic encoder. -
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VITS
Conditional Variational Autoencoder with Adversarial Learning
...This architecture enables parallel generation (fast inference) while achieving speech quality that rivals or surpasses many two-stage systems. The repository provides training and inference pipelines for common datasets such as LJ Speech (single-speaker) and VCTK (multi-speaker), including filelists, configs, and preprocessing scripts. It also includes monotonic alignment search code and g2p preprocessing, which are crucial components for aligning text and speech in an end-to-end setup. -
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Resemblyzer
A python package to analyze and compare voices with deep learning
...Its main value is making speaker representation accessible through a simple Python workflow. -
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DC-TTS
TensorFlow Implementation of DC-TTS: yet another text-to-speech model
...Training scripts, data loaders, and hyperparameter configurations are provided to reproduce results on several datasets, including LJ Speech for English, a Korean single-speaker dataset, and audiobook data from Nick Offerman and Kate Winslet. -
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Audivolv
Artificial intelligence evolves musical instruments played with mouse
Artificial Intelligence (AI) evolves musical instruments you play with the mouse from small blocks of Java code that operate on numbers, each block designed to keep all numbers in range -1 to 1 if they started in that range, and combinations of these evolved codes running 44100 times per second for high quality 44.1 khz audio. You can also write the Java code into its window. Uses Javassist internal compiler on the new Java code assembled by renaming variables to overlap eachother and... -
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Arabisc is speaker independent large vocabulary continuous speech recognizer for Arabic language released under GNU license.It is also a collection of open source tools that allows researchers and developers to build speech recognition systems for Arab
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'Text to Voice' or 'Text to Speech' is 1 of the coolest Firefox add-ons. It gives ur brwsr the pwr of speech. Select txt, clck the bttn on the bttm rite & this add-on spks the selectd txt 4 u. Isn't it brllant? Moreovr odio file cn b dnloaded as
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XTTS-v2
Multilingual voice cloning model with 6-second voice samples
XTTS-v2 is an open-source multilingual text-to-speech and voice cloning model developed by Coqui. It enables zero-shot voice cloning using as little as six seconds of reference audio, allowing users to generate speech that closely matches a target speaker without additional training. The model supports 17 languages and can perform cross-language voice cloning, meaning a voice recorded in one language can be used to synthesize speech in another while preserving speaker identity. XTTS-v2 improves on the original XTTS architecture with better speaker conditioning, support for multiple reference clips, improved prosody, enhanced audio quality, and greater inference stability. ... -
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Dia-1.6B
Dia-1.6B generates lifelike English dialogue and vocal expressions
...Designed for realistic vocal performance, Dia supports expressive features like emotion, tone control, and non-verbal cues such as laughter, coughing, or sighs. The model accepts speaker conditioning through audio prompts, allowing limited voice cloning and speaker consistency across generations. It is optimized for English and built for real-time performance on enterprise GPUs, though CPU and quantized versions are planned. The format supports [S1]/[S2] tags to differentiate speakers and integrates easily into Python workflows. ...
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