Showing 190 open source projects for "audio streaming server"

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  • 1
    Fun Audio Chat

    Fun Audio Chat

    Large Audio Language Model built for natural interactions

    ...The system supports dynamic audio input and output, meaning it can handle different voices, tones, and conversational contexts without forcing users into typed interactions. With real-time streaming, it minimizes latency and delivers responses quickly, making it suitable for applications where responsiveness matters, such as interactive demos, accessibility tools, and conversational games.
    Downloads: 0 This Week
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  • 2
    MLX-Audio

    MLX-Audio

    A text-to-speech, speech-to-text and speech-to-speech library

    MLX-Audio is a speech library built on Apple’s MLX framework and optimized for Apple Silicon machines (M-series Macs). It focuses on text-to-speech and speech-to-speech workflows, with APIs and a command-line interface that make it easy to generate high-quality audio from text. Because it uses MLX and targets Apple Silicon, inference is fast and can take advantage of hardware acceleration and quantization for efficient on-device performance. The project provides a straightforward CLI...
    Downloads: 1 This Week
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  • 3
    Triton Inference Server

    Triton Inference Server

    The Triton Inference Server provides an optimized cloud

    ...Triton delivers optimized performance for many query types, including real-time, batched, ensembles, and audio/video streaming. Provides Backend API that allows adding custom backends and pre/post-processing operations. Model pipelines using Ensembling or Business Logic Scripting (BLS). HTTP/REST and GRPC inference protocols based on the community-developed KServe protocol. A C API and Java API allow Triton to link directly into your application for edge and other in-process use cases.
    Downloads: 2 This Week
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  • 4
    OpenAI-Compatible Edge-TTS API

    OpenAI-Compatible Edge-TTS API

    Free, high-quality text-to-speech API endpoint to replace OpenAI

    ...The server supports Server-Sent Events (SSE) for streaming audio, enabling low-latency playback in chat UIs and other interactive tools. A Docker image is provided for one-command deployment, and environment variables can be used to configure default voice, language, response format, authentication, and logging options.
    Downloads: 3 This Week
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  • 5
    PaddleSpeech

    PaddleSpeech

    Easy-to-use Speech Toolkit including Self-Supervised Learning model

    PaddleSpeech is an open-source toolkit on PaddlePaddle platform for a variety of critical tasks in speech and audio, with state-of-art and influential models. Via the easy-to-use, efficient, flexible and scalable implementation, our vision is to empower both industrial application and academic research, including training, inference & testing modules, and deployment process. Low barriers to install, CLI, Server, and Streaming Server is available to quick-start your journey. ...
    Downloads: 0 This Week
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  • 6
    Sunshine

    Sunshine

    Self-hosted game stream host for Moonlight

    Sunshine is an open-source self‑hosted cloud gaming server that implements NVIDIA’s GameStream protocol. Compatible with Moonlight clients across platforms, it supports low‑latency streaming via software or hardware encoding (AMD/Intel/NVIDIA) and offers a browser‑based control UI for pairing.
    Downloads: 1,307 This Week
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  • 7
    Swing Music

    Swing Music

    Swing Music is a beautiful, self-hosted music player

    Swing Music is a beautiful, self-hosted music player and streaming server that lets you bring your personal audio library online with a modern browser-based interface, giving you a private alternative to mainstream streaming services. Designed to be both elegant and powerful, the project scans your local music files (like MP3s or FLACs), organizes metadata, and streams them on-demand to any device with a browser or its Android client.
    Downloads: 0 This Week
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  • 8
    LiveAvatar

    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. ...
    Downloads: 0 This Week
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  • 9
    Sopro TTS

    Sopro TTS

    A lightweight text-to-speech model with zero-shot voice cloning

    Sopro TTS is an open-source text-to-speech (TTS) project that implements a lightweight model capable of producing speech from text with zero-shot voice cloning, meaning it can mimic a speaker’s voice from only a few seconds of reference audio. Built with a 169 million-parameter architecture that uses dilated convolutions and cross-attention layers instead of large Transformer stacks, it achieves relatively fast real-time performance even on CPUs (about a 0.25 real-time factor measured on an M3 base). The model is designed to work with a small set of dependencies and to be accessible for developers who want offline TTS with customizable voice style, including options for streaming or non-streaming generation modes. ...
    Downloads: 0 This Week
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  • 10
    Moshi

    Moshi

    A speech-text foundation model for real time dialogue

    Moshi is a speech-text foundation model and full-duplex spoken dialogue framework. It uses Mimi, a state-of-the-art streaming neural audio codec. Mimi processes 24 kHz audio, down to a 12.5 Hz representation with a bandwidth of 1.1 kbps, in a fully streaming manner (latency of 80ms, the frame size), yet performs better than existing, non-streaming, codecs like SpeechTokenizer (50 Hz, 4kbps), or SemantiCodec (50 Hz, 1.3kbps). Moshi models two streams of audio: one corresponds to Moshi, and the other one to the user. ...
    Downloads: 0 This Week
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  • 11
    SoulSync

    SoulSync

    Automated Music Discovery and Collection Manager

    SoulSync is an intelligent music discovery and automation platform designed to bridge streaming services with self-hosted media libraries, enabling users to automatically grow and maintain curated music collections. The system continuously monitors selected artists and detects new releases, then generates personalized playlists such as Release Radar and Discovery Weekly using its built-in recommendation logic. It can automatically download missing tracks from multiple sources including Soulseek, YouTube, and Beatport, then verify file accuracy through AcoustID fingerprinting to ensure the correct audio was obtained. ...
    Downloads: 3 This Week
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  • 12
    Dolphin

    Dolphin

    Document Image Parsing via Heterogeneous Anchor Prompting”

    Dolphin — maintained by ByteDance — is a project aimed at providing a high-performance, robust, and extensible media or multimedia framework / player infrastructure (or possibly a streaming media solution), intended to meet modern demands for efficiency, flexibility, and integration in media-heavy applications. It seeks to combine performant media playback or handling (audio/video decoding, streaming, buffering) with a modular, developer-friendly API that allows easy embedding into larger applications or services. ...
    Downloads: 0 This Week
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  • 13
    MOSS-TTS Family

    MOSS-TTS Family

    MOSS‑TTS Family open‑source speech and sound generation model

    ...The broader family also includes dialogue generation, prompt-based voice creation, streaming voice-agent support, and a unified audio tokenizer. It is especially useful for developers building dubbing, podcasts, audiobooks, voice assistants, character voices, and creative audio tools.
    Downloads: 1 This Week
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  • 14
    PyLivestream

    PyLivestream

    Pure Python FFmpeg-based live video / audio streaming to YouTube

    PyLivestream is a Python-based tool that enables real-time video streaming from various input sources to platforms such as YouTube and Twitch. It acts as a wrapper around FFmpeg, allowing users to stream video from cameras, files, or screen capture devices with minimal configuration. The tool supports cross-platform operation and integrates easily into Python workflows, making it suitable for automation and scripting. It provides options for controlling streaming parameters such as bitrate,...
    Downloads: 0 This Week
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  • 15
    Qwen2.5-Omni

    Qwen2.5-Omni

    Capable of understanding text, audio, vision, video

    Qwen2.5-Omni is an end-to-end multimodal flagship model in the Qwen series by Alibaba Cloud, designed to process multiple modalities (text, images, audio, video) and generate responses both as text and natural speech in streaming real-time. It supports “Thinker-Talker” architecture, and introduces innovations for aligning modalities over time (for example synchronizing video/audio), robust speech generation, and low-VRAM/quantized versions to make usage more accessible. It holds state-of-the-art performance in many multimodal benchmarks, particularly spoken language understanding, audio reasoning, image/video understanding, etc. ...
    Downloads: 0 This Week
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  • 16
    Text Generation Web UI

    Text Generation Web UI

    Oobabooga - The definitive Web UI for local AI, with powerful features

    ...Instruct mode compatible with Alpaca and Open Assistant formats. Nice HTML output for GPT-4chan. Markdown output for GALACTICA, including LaTeX rendering. Custom chat characters. Advanced chat features (send images, get audio responses with TTS). Very efficient text streaming. Parameter presets, 8-bit mode. Layers splitting across GPU(s), CPU, and disk. CPU mode, FlexGen, DeepSpeed ZeRO-3, API with streaming and without streaming. LLaMA model, including 4-bit GPTQ. RWKV model, LoRA (loading and training), Softprompts, and extensions.
    Downloads: 14 This Week
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  • 17
    TorchAudio

    TorchAudio

    Data manipulation and transformation for audio signal processing

    The aim of torchaudio is to apply PyTorch to the audio domain. By supporting PyTorch, torchaudio follows the same philosophy of providing strong GPU acceleration, having a focus on trainable features through the autograd system, and having consistent style (tensor names and dimension names). Therefore, it is primarily a machine learning library and not a general signal processing library. The benefits of PyTorch can be seen in torchaudio through having all the computations be through PyTorch...
    Downloads: 39 This Week
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  • 18
    ElevenLabs Python

    ElevenLabs Python

    The official Python SDK for the ElevenLabs API

    ...It includes helper utilities (like play and stream) so you can either play audio locally or integrate it into your own playback or networking pipeline.
    Downloads: 2 This Week
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  • 19
    WhatsApp MCP Server

    WhatsApp MCP Server

    WhatsApp MCP server enabling AI access to chats and messaging

    whatsapp-mcp is an open source Model Context Protocol (MCP) server that enables AI agents to interact directly with a user’s WhatsApp account through a structured interface. It acts as a bridge between WhatsApp and large language models, allowing controlled access to messages, chats, and contacts. whatsapp-mcp is composed of two main components: a Go-based bridge that connects to the WhatsApp Web API and stores data locally, and a Python-based MCP server that exposes tools for AI interaction. ...
    Downloads: 0 This Week
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  • 20
    StreamSpeech

    StreamSpeech

    StreamSpeech is a seamless model for offline speech recognition

    StreamSpeech is an “all-in-one” speech model designed to perform offline and simultaneous speech recognition, speech translation, and speech synthesis within a single unified architecture. Developed as part of an ACL 2024 paper, it targets streaming and low-latency scenarios where intermediate results and final translations or synthetic speech must be produced continuously as audio is being received. The model supports eight tasks: offline ASR, speech-to-text translation, speech-to-speech translation, and TTS, as well as their streaming or simultaneous counterparts, all handled by the same underlying system. ...
    Downloads: 0 This Week
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  • 21
    VoxCPM2

    VoxCPM2

    Tokenizer-Free TTS for Multilingual Speech Generation

    VoxCPM2 is an advanced open-source text-to-speech system that redefines speech synthesis by eliminating traditional tokenization and instead generating continuous speech representations through a diffusion-based autoregressive architecture. Built on top of the MiniCPM model family, it enables highly natural, expressive, and context-aware speech generation that adapts tone, emotion, and pacing directly from input text. The system is trained on massive multilingual datasets, enabling support...
    Downloads: 25 This Week
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  • 22
    Qwen3-Omni

    Qwen3-Omni

    Qwen3-omni is a natively end-to-end, omni-modal LLM

    ...It achieves state-of-the-art results: across 36 audio and audio-visual benchmarks, it hits open-source SOTA on 32 and overall SOTA on 22, outperforming or matching strong closed-source models such as Gemini-2.5 Pro and GPT-4o. To reduce latency, especially in audio/video streaming, Talker predicts discrete speech codecs via a multi-codebook scheme and replaces heavier diffusion approaches.
    Downloads: 1 This Week
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  • 23
    Selkies-GStreamer

    Selkies-GStreamer

    Open-Source Low-Latency Accelerated Linux WebRTC HTML5 Remote Desktop

    selkies-gstreamer is a GStreamer-based media streaming component used in the Selkies project, a cloud-native platform designed for interactive desktop and application streaming. This module acts as a high-performance media pipeline that captures video, encodes it with low latency, and streams it via WebRTC to client browsers. It is optimized for GPU-accelerated encoding and integrates with Kubernetes-based deployments to enable scalable, real-time remote desktop sessions. This component...
    Downloads: 1 This Week
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  • 24
    WhisperLive

    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. ...
    Downloads: 12 This Week
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  • 25
    Orpheus TTS

    Orpheus TTS

    Towards Human-Sounding Speech

    ...The project ships both pretrained and finetuned English models, as well as a family of multilingual models released as a research preview, and includes data-processing scripts so users can train or finetune their own variants. Inference is provided through a Python package that uses vLLM under the hood for high-throughput, low-latency generation, including streaming examples that show how to generate audio chunks in real time. The maintainers provide Colab notebooks, a standardized prompting format, and one-click deployment via Baseten for production-grade, FP8/FP16 optimized inference with ~200 ms streaming latency.
    Downloads: 0 This Week
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