awesome-repositories.com
Blog
MCP
awesome-repositories.com

Discover the best open-source repositories with AI-powered search.

ExploreCurated searchesOpen-source alternativesSelf-hosted softwareBlogSitemap
ProjectMCP serverAboutHow we rankPress
LegalPrivacyTerms
© 2026 Bringes Technology SRL·VAT RO45896025·hello@awesome-repositories.com
OpenBMB avatar

OpenBMB/MiniCPM-o

0
View on GitHub↗
23,850 stars·1,836 forks·Python·apache-2.0·37 views

MiniCPM O

MiniCPM-o is a multimodal large language model designed to function as a real-time conversational assistant on edge devices. By mapping text, image, video, and audio inputs into a unified latent space, the system enables simultaneous cross-modal reasoning and full-duplex interaction. It is built as an edge-side inference engine, utilizing quantized model weights to maintain high-performance processing on consumer hardware.

The system distinguishes itself through its integrated speech synthesis and voice cloning capabilities, which allow for the generation of expressive, personalized vocal output from short audio samples without additional training. Users can modulate the emotional tone, speed, and emphasis of synthesized speech in real time using latent prosody control tokens. Furthermore, the model supports the adoption of specific personas and roles, facilitating immersive, situation-aware dialogue.

Beyond its core conversational features, the framework provides tools for proactive visual assistance, such as monitoring environments to trigger navigation or scheduling alerts. The architecture is configurable, allowing for adjustments to visual token compression and frame sampling rates to balance accuracy and speed. The project supports fine-tuning for specialized domains, enabling developers to adapt the model to custom tasks using standard training frameworks.

Features

  • Multimodal Large Language Models - Processes real-time audio, video, and text streams using a unified vision-language model architecture.
  • Agentic Assistants - Acts as a conversational agent that maintains situational awareness through continuous visual and auditory input.
  • Edge Inference Engines - Provides a high-performance inference engine designed for executing quantized models on resource-constrained hardware.
  • Edge and Mobile - Optimizes model performance on edge devices through weight quantization and compression.
  • On-Device Inference Engines - Executes optimized and quantized machine learning models locally on edge hardware for low-latency performance.
  • Real-Time Voice Cloning - Enables real-time voice cloning by extracting vocal identity from short audio samples without additional training.
  • Edge AI Model Deployment - Optimizes and deploys complex machine learning models for efficient execution on consumer edge hardware.
  • Full-Duplex Multimodal Interaction - Enables fluid, full-duplex interaction by processing simultaneous visual, auditory, and speech streams.
  • Voice Cloning Engines - Generates expressive, personalized vocal output from reference audio samples without requiring model retraining.
  • Voice Cloning - Captures unique vocal characteristics from audio samples to generate personalized voice output.
  • Full-Duplex Conversational Streams - Supports full-duplex conversational streams by handling continuous audio-visual input and concurrent output generation.
  • Cross-Modal Representations - Maps disparate text, image, and audio inputs into a shared vector representation for cross-modal interaction.
  • Zero-Shot Voice Cloning - Enables the replication of specific vocal identities from short audio samples without requiring model retraining.
  • Model Fine-Tuning - Supports fine-tuning of pre-trained models for specialized domains using standard training frameworks.
  • Multimodal Token Interleaving - Maps text, image, and audio inputs into a unified latent space to enable simultaneous cross-modal reasoning.
  • Proactive Visual Assistance - Monitors visual environments to provide proactive alerts and navigation support for users.
  • Stream Processing Systems - Utilizes concurrent input and output buffers to enable full-duplex, real-time conversational stream processing.
  • Multimodal Conversational Interfaces - Facilitates fluid, human-like conversations by processing live audio, video, and text streams simultaneously.
  • Agent Persona Frameworks - Simulates specific characters or professional roles by adopting unique speech patterns and personality traits.
  • Speech Synthesis - Generates natural-sounding, expressive speech with customizable emotional tone and vocal emphasis.
  • Prosody Control Tokens - Injects control tokens into the generation pipeline to modulate emotional tone and speech prosody in real time.
  • Multimodal Processing - Processes text, image, video, and audio streams simultaneously for real-time multimodal interaction.
  • Multimodal Architectures - Integrates vision and speech for omni-modal live streaming capabilities.
  • Multimodal Foundation Models - Efficient multimodal model for mobile-based understanding.
  • Dialogue Interaction Engines - Maintains persistent, situation-aware dialogue to act as an immersive companion during real-world activities.
  • Agent Persona Definitions - Supports the adoption of specific personas and roles to facilitate immersive, situation-aware dialogue.
  • Voice Conditioning Encoders - Uses lightweight encoders to condition speech synthesis on reference audio samples.
  • Latent Conditioning Mechanisms - Employs latent conditioning mechanisms to adjust emotional expression and speech delivery in real time.
  • Prosody Controls - Provides real-time control over speech delivery speed and emotional prosody during synthesis.
  • Speech Emphasis Controls - Allows users to control speech emphasis to change meaning and intent through vocal stress.
  • Audio Emotion Classifiers - Modulates the tone and delivery of spoken output to convey specific emotional states.
  • Model Parameter Configurations - Provides configurable parameters for visual token compression and system profiles to balance performance.
  • Voice Personalization - Allows users to select and apply distinct vocal timbres for a customized auditory experience.
  • Visual Token Compression - Implements adaptive visual token compression to balance inference speed and accuracy on edge devices.
  • Automated Alerting Workflows - Monitors visual environments to trigger proactive alerts and navigation notifications.

Star history

Star history chart for openbmb/minicpm-oStar history chart for openbmb/minicpm-o

How this analysis was created: This summary and feature list are AI-generated from collected project material and can contain mistakes. Stars, license and language are imported from GitHub. Inclusion does not mean that we have tested or audited this project. Check the source documentation for any feature you depend on. Learn more on our About page.

AI search

Explore more awesome repositories

Describe what you need in plain English — the AI ranks thousands of curated open-source projects by relevance.

Start searching with AI

Projects sharing features with MiniCPM O

These projects share indexed features with MiniCPM O. Shared tags can include platform or build tooling; verify the primary use case before treating a result as a replacement.
  • openbmb/minicpm-vOpenBMB avatar

    OpenBMB/MiniCPM-V

    25,653View on GitHub↗

    MiniCPM-V is a multimodal large language model and vision-language system designed for complex visual and linguistic understanding. It functions as an on-device AI model, providing the capacity to process text, images, and video as a compact neural network. The project is specifically developed as an edge AI framework, utilizing quantization and weight sharding to run on memory-constrained mobile chipsets. This allows for the deployment of multimodal intelligence directly on mobile operating systems for local inference. Its capabilities cover multimodal content analysis of high-resolution im

    Python
    View on GitHub↗25,653
  • microsoft/unilmmicrosoft avatar

    microsoft/unilm

    22,030View on GitHub↗

    This project is a comprehensive framework and toolkit for developing, optimizing, and deploying transformer-based models across multimodal, document intelligence, and natural language processing tasks. It provides a unified neural architecture that processes text, vision, audio, and document layout data through a shared set of weights, enabling researchers and developers to build foundational models that align cross-modal representations. The platform distinguishes itself through advanced training and inference strategies designed for large-scale deep learning. It incorporates specialized mec

    Pythonbeitbeit-3bitnet
    View on GitHub↗22,030
  • nari-labs/dianari-labs avatar

    nari-labs/dia

    19,324View on GitHub↗

    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

    Pythonaiopen-weighttext-to-speech
    View on GitHub↗19,324
  • zyphra/zonosZyphra avatar

    Zyphra/Zonos

    7,225View on GitHub↗

    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,

    Python
    View on GitHub↗7,225
Compare all 30 related projects→

Frequently asked questions

What does openbmb/minicpm-o do?

MiniCPM-o is a multimodal large language model designed to function as a real-time conversational assistant on edge devices. By mapping text, image, video, and audio inputs into a unified latent space, the system enables simultaneous cross-modal reasoning and full-duplex interaction. It is built as an edge-side inference engine, utilizing quantized model weights to maintain high-performance processing on consumer hardware.

What are the main features of openbmb/minicpm-o?

The main features of openbmb/minicpm-o are: Multimodal Large Language Models, Agentic Assistants, Edge Inference Engines, Edge and Mobile, On-Device Inference Engines, Real-Time Voice Cloning, Edge AI Model Deployment, Full-Duplex Multimodal Interaction.

Which projects share features with openbmb/minicpm-o?

Projects with overlapping indexed features include: openbmb/minicpm-v — MiniCPM-V is a multimodal large language model and vision-language system designed for complex visual and linguistic… microsoft/unilm — This project is a comprehensive framework and toolkit for developing, optimizing, and deploying transformer-based… nari-labs/dia — Dia is a generative AI audio tool and text-to-speech synthesis engine designed for the production-ready deployment of… zyphra/zonos — Zonos is a controllable audio synthesis engine and large language model for text-to-speech. It serves as a… neuphonic/neutts — Neutts is a neural text-to-speech engine designed for real-time streaming output on edge devices such as phones and… openbmb/voxcpm — VoxCPM is a multilingual speech synthesis system and text-to-speech inference server. It functions as an AI voice…