Search Results for "audio" - Page 4
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Showing 1069 open source projects for "audio"
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WhisperLive
A nearly-live implementation of OpenAI's Whisper
WhisperLive is a “nearly live” implementation of OpenAI’s Whisper model focused on real-time transcription. It runs as a server–client system in which the server hosts a Whisper backend and clients stream audio to be transcribed with very low delay. The project supports multiple inference backends, including Faster-Whisper, NVIDIA TensorRT, and OpenVINO, allowing you to target GPUs and different CPU architectures efficiently. It can handle microphone input, pre-recorded audio files, and network streams such as RTSP and HLS, making it flexible for live events, monitoring, or accessibility workflows. ... -
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LiveAvatar
Streaming Real-time Audio-Driven Avatar Generation
LiveAvatar is an open-source research and implementation project that provides a unified framework for real-time, streaming, interactive avatar video generation driven by audio and other control signals. It implements techniques from state-of-the-art diffusion-based avatar modeling to support infinite-length continuous video generation with low latency, enabling interactive AI avatars that maintain continuity and realism over extended sessions. The project co-designs algorithms and system optimizations, such as block-wise autoregressive processing and fast sampling strategies, to deliver real-time frame rates (e.g., ~45 FPS on appropriate GPU clusters) while handling non-stop generation without quality degradation. ... -
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Open Notebook
An Open Source implementation of Notebook LM with more flexibility
...The platform supports 16+ AI providers—including OpenAI, Anthropic, Ollama, Google, and LM Studio—allowing flexible model choice and cost optimization. Open Notebook enables users to organize and analyze multi-modal content such as PDFs, videos, audio files, web pages, and Office documents. It combines full-text and vector search with context-aware AI chat to deliver insights grounded in your own research materials. With advanced features like multi-speaker podcast generation, customizable content transformations, and a comprehensive REST API, Open Notebook provides a powerful and extensible research environment. -
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Fish Speech
SOTA Open Source TTS
...The models are evaluated with Seed TTS metrics and achieve exceptionally low word and character error rates, indicating strong intelligibility and alignment between text and audio. Fish Speech emphasizes expressive and controllable voices: it supports a long list of emotion tags, tone markers, and special audio effect markers that can be embedded in the text to drive prosody and vocal style, from basic emotions to nuanced states like sarcastic, conciliative, or hysterical. The system is multilingual and cross-lingual, handling multiple languages in a single input without explicit phoneme markup, and is trained on large-scale datasets. -
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AbletonMCP
Ableton Live Model Context Protocol Integration
...The system combines a Python MCP server with an Ableton MIDI Remote Script that exchanges JSON commands through TCP sockets. It can inspect sessions, create MIDI or audio tracks, build clips, insert notes, trigger clips, and control playback. The integration can also search Ableton’s browser, load instruments and effects, change tempo, and construct sections such as intros, drops, breakdowns, and outros. It supports Claude Desktop and Cursor and requires Ableton Live 10 or newer, Python 3.8 or newer, and the uv package manager. ... -
6
Hugging Face - Speech To Speech
Open speech-to-speech models and pipelines by Hugging Face toolkit AI
This project from Hugging Face focuses on enabling direct speech-to-speech processing using modern machine learning models. It provides tools and reference implementations that allow audio input to be transformed into audio output without requiring an intermediate text representation. Hugging Face - Speech To Speech builds on recent advances in speech modeling, combining components such as speech recognition, translation, and synthesis into unified pipelines. It is designed to help researchers and developers experiment with multilingual and cross-lingual voice applications. ... -
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OmniVoice
High-Quality Voice Cloning TTS for 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, it supports voice design through configurable attributes such as gender, accent, pitch, and speaking style, giving users fine-grained control over generated speech. The system also includes advanced features like non-verbal expression tags and pronunciation overrides, enabling expressive and precise output. With support for both API-based and command-line usage, it is designed for research, production, and experimentation alike. -
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RealtimeTTS
Converts text to speech in realtime
RealtimeTTS is a low-latency text-to-speech library built for real-time applications such as voice chat with LLMs, assistants, and interactive tools. It is designed around a streaming model: you can feed it text incrementally (for example, as an LLM responds) and get audio output almost immediately, which keeps end-to-end latency very low. The library is engine-agnostic and plugs into a wide range of cloud and local TTS systems, including OpenAI, ElevenLabs, Azure, Coqui, Piper, StyleTTS2, Edge TTS, Google TTS, system TTS and others, so you can swap providers without rewriting your pipeline. It supports both internet-based engines and fully local engines, which lets you choose between privacy, cost, and quality trade-offs. ... -
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NetEase-MusicBox
NetEase cloud music command line version
The high-quality command line version of NetEase Cloud Music is simple, elegant, silky and smooth, and is written based on Python. 320kbps high-quality music. Song, artist, album search. NetEase 22 song charts. Netease new disc recommendation. NetEase Featured Playlist. NetEase Anchor Radio. Private playlist, recommended daily. DJing, local collection, add at any time. Play progress and play mode display. Now playing and desktop lyrics display. Song comment display. One-click to enter the... -
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LinuxPlay
An open-source, ultra-low-latency remote desktop for Linux hosts
LinuxPlay is a multimedia playback project designed to provide a lightweight media player for Linux environments with support for common audio and video formats. It focuses on implementing playback functionality using FFmpeg and native Linux APIs to ensure efficient decoding and rendering. The project demonstrates how to build a media player from scratch, including handling audio-video synchronization and playback controls. It supports local file playback and may include streaming capabilities depending on configuration. ... -
11
pyttsx3
Offline Text To Speech synthesis for python
...The library exposes a simple but flexible API for controlling voice selection, speaking rate, volume, and other synthesis parameters from Python code. It supports both a high-level speak convenience function and a lower-level engine object with event hooks, queuing, and saving output to audio files. The repository includes examples and documentation that show how to adjust properties dynamically, persist synthesized output, and integrate pyttsx3 into GUIs or background services. -
12
BlogWizard
Generate blog articles from video or audio
BlogWizard is a demo/utility project built on top of Groq’s LLM infrastructure that converts video or audio content into well-structured blog posts, enabling creators to repurpose multimedia content into text — useful for SEO, accessibility, or reaching audiences that prefer reading. The tool uses transcription (e.g. via Whisper) to extract text from audio/video, then runs an LLM-based generation pipeline to transform that content into coherent, readable blog-format posts — with sections, formatting, and possibly metadata. ... -
13
AsrTools
AsrTools: Smart Voice-to-Text Tool
...Output formats include SRT, TXT, and ASS, making the tool suitable for subtitles, transcripts, and later editing. Video files can be imported directly and converted to audio automatically before recognition. Windows users can run a packaged executable, while developers can also launch the Python source directly. -
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Whisper
Robust Speech Recognition via Large-Scale Weak Supervision
OpenAI Whisper is a general-purpose speech recognition model. It is trained on a large dataset of diverse audio and is also a multitasking model that can perform multilingual speech recognition, speech translation, and language identification. A Transformer sequence-to-sequence model is trained on various speech processing tasks, including multilingual speech recognition, speech translation, spoken language identification, and voice activity detection. -
15
pyTheory
Music Theory for Humans
...A score API can combine melodic parts, bass lines, chords, drums, synthesis, effects, automation, and sequencing. Compositions can be played immediately or exported as MIDI for continued production in a digital audio workstation. The command-line interface includes chord analysis, progression generation, tuning, playback, live MIDI tools, and an interactive REPL. A browser playground provides access without installation, while optional integrations support Ableton Link and external MIDI input. The project also offers Claude Code skills for AI-assisted composition and analysis. -
16
yt-dlp
A youtube-dl fork with additional features and fixes
yt-dlp is a youtube-dl fork based on the now inactive youtube-dlc. The main focus of this project is adding new features and patches while also keeping up to date with the original project -
17
edge-tts
Use Microsoft Edge's online text-to-speech service from Python
...It wraps the same cloud voices used by Edge, exposing them through a simple CLI (edge-tts, edge-playback) and a Python API, so you can script high-quality speech generation in your own applications. The tool lets you list available voices, specify locale and voice name, and generate audio files in common formats like MP3 or WAV. It also supports generating subtitle files (such as SRT or VTT) alongside the speech, which is handy for video narration, e-learning, or accessibility workflows. From the CLI you can adjust parameters such as speaking rate, volume, and pitch, giving you some control over prosody without diving into SSML. ... -
18
Pixeltable
Data Infrastructure providing an approach to multimodal AI workloads
...Developers define data transformations and AI operations using computed columns on tables, allowing pipelines to evolve incrementally as new data or models are added. The framework supports multimodal content including images, video, text, and audio, enabling applications such as retrieval-augmented generation systems, semantic search, and multimedia analytics. -
19
PersonaPlex
PersonaPlex code
...PersonaPlex also supports persona and voice control, allowing developers to define the role and speaking style of the agent using text prompts and voice conditioning, making it suitable for applications like customized voice assistants, interactive character agents, or domain-specific conversational tools. Internally, it processes continuous audio streams in a hybrid input format so that speech understanding and generation occur jointly. -
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HunyuanCustom
Multimodal-Driven Architecture for Customized Video Generation
HunyuanCustom is a multimodal video customization framework by Tencent Hunyuan, aimed at generating customized videos featuring particular subjects (people, characters) under flexible conditions, while maintaining subject/identity consistency. It supports conditioning via image, audio, video, and text, and can perform subject replacement in videos, generate avatars speaking given audio, or combine multiple subject images. The architecture builds on HunyuanVideo, with added modules for identity reinforcement and modality-specific condition injection. Text-image fusion module based on LLaVA for improved multimodal understanding. ... -
21
video-use
Edit videos with Claude Code
...Designed to work with Claude Code, it automates the entire editing process—from cutting clips to rendering the final output—without requiring manual timelines or complex software interfaces. The system intelligently analyzes audio transcripts and visual cues to make precise, context-aware editing decisions. It supports a wide range of content types, including interviews, tutorials, montages, and talking-head videos. By combining structured text representations with on-demand visual previews, it minimizes processing overhead while maintaining high-quality results. ... -
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GLM-TTS
Controllable & emotion-expressive zero-shot TTS
...It uses a two-stage architecture where a generative LLM first converts text into intermediate speech token sequences and then a Flow-based neural model converts those tokens into natural audio waveforms, enabling rich prosody and voice character even for unseen speakers. The system introduces a multi-reward reinforcement learning framework that jointly optimizes for voice similarity, emotional expressiveness, pronunciation, and intelligibility, yielding output that can rival commercial options in naturalness and expressiveness. ... -
23
Label Studio
Label Studio is a multi-type data labeling and annotation tool
...Configurable label formats let you customize the visual interface to meet your specific labeling needs. Support for multiple data types including images, audio, text, HTML, time-series, and video. -
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yami
An open-source music player with simple UI
Yami is a lightweight, open-source music player built in Python. It focuses on simplicity and ease of use, providing an intuitive user interface (UI) for users to manage and play their music. Whether you're playing local files or downloading from online sources using spotdl, Yami offers a seamless experience. This project is designed for users who want a minimalistic, cross-platform music player with the ability to integrate external sources like Spotify/YouTube Music. -
25
Whisper-WebUI
A Web UI for easy subtitle using whisper model
Whisper WebUI is an open-source browser-based interface that simplifies the use of Whisper speech recognition models by providing an intuitive graphical environment for transcription, translation, and subtitle generation. Built with Gradio, it allows users to upload audio or video files, process them locally, and generate accurate text outputs without relying on command-line tools. The platform integrates optimized implementations such as faster-whisper, significantly improving transcription speed and reducing memory usage compared to standard models. It supports multiple input sources including local files, YouTube content, and microphone input, making it versatile for different workflows. ...
