--- license: apache-2.0 pipeline_tag: any-to-any library_name: transformers tags: - minicpm-o - minicpm-v - multimodal - full-duplex --- A Gemini 2.5 Flash Level MLLM for Vision, Speech, and Full-Duplex Mulitmodal Live Streaming on Your Phone [GitHub](https://github.com/OpenBMB/MiniCPM-o) | [CookBook](https://github.com/OpenSQZ/MiniCPM-V-CookBook) | [Streaming Demo](https://huggingface.co/spaces/openbmb/minicpm-omni) | [Chatbot Demo](http://211.93.21.133:18121/)
[WeChat](https://github.com/OpenBMB/MiniCPM-o/blob/main/docs/wechat.md) | [Discord](https://discord.gg/N2RnxGdJ) | [Audio Demo Page](https://openbmb.github.io/minicpm-o-4_5/) ## News > [!NOTE] > [2026.02.06] 🥳 🥳 🥳 MiniCPM-o 4.5 Local & Ready-to-Run! Experience **low-latency full-duplex communication** directly **on your own Mac** using our new official Docker image. [Try it now](https://github.com/OpenSQZ/MiniCPM-V-CookBook/blob/main/demo/web_demo/WebRTC_Demo/README.md)! ## MiniCPM-o 4.5 **MiniCPM-o 4.5** is the latest and most capable model in the MiniCPM-o series. The model is built in an end-to-end fashion based on SigLip2, Whisper-medium, CosyVoice2, and Qwen3-8B with a total of 9B parameters. It exhibits a significant performance improvement, and introduces new features for full-duplex multimodal live streaming. Notable features of MiniCPM-o 4.5 include: - 🔥 **Leading Visual Capability.** MiniCPM-o 4.5 achieves an average score of 77.6 on OpenCompass, a comprehensive evaluation of 8 popular benchmarks. **With only 9B parameters, it surpasses widely used proprietary models like GPT-4o, Gemini 2.0 Pro, and approaches Gemini 2.5 Flash** for vision-language capabilities. It supports instruct and thinking modes in a single model, better covering efficiency and performance trade-offs in different user scenarios. - 🎙 **Strong Speech Capability.** MiniCPM-o 4.5 supports **bilingual real-time speech conversation with configurable voices** in English and Chinese. It features **more natural, expressive and stable speech conversation**. The model also allows for fun features such as **voice cloning and role play via a simple reference audio clip**, where the cloning performance surpasses strong TTS tools such as CosyVoice2. - 🎬 **New Full-Duplex and Proactive Multimodal Live Streaming Capability.** As a new feature, MiniCPM-o 4.5 can process real-time, continuous video and audio input streams simultaneously while generating concurrent text and speech output streams in an end-to-end fashion, without mutual blocking. This **allows MiniCPM-o 4.5 to see, listen, and speak simultaneously**, creating a fluid, real-time omnimodal conversation experience. Beyond reactive responses, the model can also perform **proactive interaction**, such as initiating reminders or comments based on its continuous understanding of the live scene. - 💪 **Strong OCR Capability, Efficiency and Others.** Advancing popular visual capabilities from MiniCPM-V series, MiniCPM-o 4.5 can process **high-resolution images** (up to 1.8 million pixels) and **high-FPS videos** (up to 10fps) in any aspect ratio efficiently. It achieves **state-of-the-art peformance for end-to-end English document parsing** on OmniDocBench, outperforming proprietary models such as Gemini-3 Flash and GPT-5, and specialized tools such as DeepSeek-OCR 2. It also features **trustworthy behaviors**, matching Gemini 2.5 Flash on MMHal-Bench, and supports **multilingual capabilities** on more than 30 languages. - 💫 **Easy Usage.** MiniCPM-o 4.5 can be easily used in various ways: (1) [llama.cpp](https://github.com/OpenSQZ/MiniCPM-V-CookBook/blob/main/deployment/llama.cpp/minicpm-o4_5_llamacpp.md) and [Ollama](https://github.com/OpenSQZ/MiniCPM-V-CookBook/blob/main/deployment/ollama/minicpm-o4_5_ollama.md) support for efficient CPU inference on local devices, (2) [int4](https://github.com/OpenSQZ/MiniCPM-V-CookBook/blob/main/quantization/awq/minicpm-o4_5_awq_quantize.md) and [GGUF](https://github.com/OpenSQZ/MiniCPM-V-CookBook/blob/main/quantization/gguf/minicpm-o4_5_gguf_quantize.md) format quantized models in 16 sizes, (3) [vLLM](https://github.com/OpenSQZ/MiniCPM-V-CookBook/blob/main/deployment/vllm/minicpm-o4_5_vllm.md) and [SGLang](https://github.com/OpenSQZ/MiniCPM-V-CookBook/blob/main/deployment/sglang/MiniCPM-o4_5_sglang.md) support for high-throughput and memory-efficient inference, (4) [FlagOS](#flagos) support for the unified multi-chip backend plugin, (5) fine-tuning on new domains and tasks with [LLaMA-Factory](https://github.com/OpenSQZ/MiniCPM-V-CookBook/blob/main/finetune/llama-factory/finetune_llamafactory.md), and (6) online web demo on [server](https://github.com/OpenSQZ/MiniCPM-V-CookBook/blob/main/demo/web_demo/gradio/README_o45.md). We also rollout a high-performing [llama.cpp-omni](https://github.com/tc-mb/llama.cpp-omni) inference framework together with a [WebRTC Demo](https://minicpm-omni.openbmb.cn/), which **enables the full-duplex multimodal live streaming experience on local devices** such as [PCs](https://github.com/tc-mb/llama.cpp-omni/blob/master/README.md) (e.g., on a MacBook). **Model Architecture.**
### Evaluation
Note: Scores marked with ∗ are from our evaluation; others are cited from referenced reports. n/a indicates that the model does not support the corresponding modality. All results are reported in instruct mode/variant.
Click to view visual understanding results. **Image Understanding (Instruct)**
Model OpenCompass MMBench EN v1.1 MMBench CN v1.1 MathVista MMVet MMMU MMStar HallusionBench AI2D OCRBench TextVQA_VAL DocVQA_VAL MMT-Bench_VAL MM-IFEval Mantis-Eval MuirBench MMSI-Bench MMHal-Score MMHal-Hallrate↓
Gemini2.5-Flash-Nonthinking 78.5 86.6 86.0 75.3 81.4* 76.3 75.8 59.1 87.7 864 74.3* 93.0 70.0* 75.8* 72.8* 74.5* 12.1* 4.6* 23.9*
InternVL-3.5-8B 75.8 79.5 80.0* 78.4 83.1 73.4 69.3 54.5 84.0 840 78.2 92.3 66.7 56.3* 70.5 55.8 - 3.8* 34.7*
Qwen3-VL-8B-Instruct 76.5 84.5 84.7 77.2 73.7* 69.6 70.9 61.1 85.7 896 82.9* 96.1 60.9* 59.4* 74.2* 64.4 11.3* 4.7* 29.9*
Qwen3-Omni-30B-A3B-Instruct 75.7 84.9* 84.1* 75.9 74.8* 69.1 68.5 59.7 85.2 880* 84.1* 95.4* 70.4* 65.7* 78.3* 61.9* 14.2* 4.6* 31.6*
MiniCPM-o 4.5-Instruct 77.6 87.6 87.2 80.1 74.4 67.6 73.1 63.2 87.6 876 83.8 94.7 69.7 66.3 79.7 72.0 16.6 4.7 24.3
**Image Understanding (Thinking)**
Model OpenCompass MMBench EN v1.1 MMBench CN v1.1 MathVista MMVet MMMU MMStar HallusionBench AI2D OCRBench TextVQA_VAL DocVQA_VAL MMT-Bench_VAL MM-IFEval
Gemini2.5-Flash-Thinking 79.9 87.1 87.3 79.4 81.2* 77.7 76.5 63.5 88.7 853 73.8* 92.8 70.7* 75.7*
GPT-5 79.7 85.5* 85.6* 81.9 77.6 81.8 75.7 65.2 89.5 807 77.8* 91.3* 72.7* 83.1*
Qwen3-VL-8B-Thinking 77.3 85.3 85.5 81.4 69.8* 74.1 75.3 65.4 84.9 819 77.8* 95.3 68.1* 73.5*
Qwen3-Omni-30B-A3B-Thinking 78.5 88.2* 87.7* 80.0 74.8* 75.6 74.9 62.8 86.1 859* 80.8* 94.2* 70.9* 69.9*
MiniCPM-o 4.5-Thinking 78.2 89.0 87.6 81.0 73.6 70.2 73.6 62.6 88.5 879 79.8 92.3 69.7 68.2
**Video Understanding**
Model Video-MME
(w/o subs)
LVBench MLVU
(M-Avg)
LongVideoBench
(val)
MotionBench
Gemini2.5-Flash-Nonthinking 75.6 62.2 77.8 - -
InternVL-3.5-8B 66.0 - 70.2 62.1 62.3*
Qwen3-Omni-30B-A3B-Instruct 70.5 50.2 75.2 66.9* 61.7*
MiniCPM-o 4.5-Instruct 70.4 50.9 76.5 66.0 61.4
Click to view document parsing results. **OmniDocBench**
Method Type Methods OverallEdit↓ TextEdit↓ FormulaEdit↓ TableTEDS↑ TableEdit↓ Read OrderEdit↓
EN ZH EN ZH EN ZH EN ZH EN ZH EN ZH
Pipeline MinerU 2.5 0.117* 0.172* 0.051* 0.08* 0.256* 0.455* 85.9* 89.4* 0.115* 0.081* 0.047* 0.072*
PaddleOCR-VL 0.105 0.126 0.041 0.062 0.241 0.316 88 92.1 0.093 0.062 0.045 0.063
End-to-end Model Qwen2.5-VL-72B 0.214 0.261 0.092 0.18 0.315 0.434 82.9 83.9 0.341 0.262 0.106 0.168
GPT 5 0.218* 0.33* 0.139* 0.344* 0.396* 0.555* 77.55* 73.09* 0.188* 0.196* 0.151* 0.227*
Gemini2.5-Flash-Nonthinking 0.214* 0.29* 0.159* 0.273* 0.368* 0.524* 80.9* 85.5* 0.197* 0.167* 0.132* 0.195*
Gemini-2.5-Pro-Nonthinking 0.148* 0.212* 0.055* 0.168* 0.356* 0.439* 85.8* 86.4* 0.13* 0.119* 0.049* 0.121*
Gemini-3 Flash-Nonthinking 0.155* 0.201* 0.138* 0.255* 0.297* 0.351* 86.4* 89.8* 0.116* 0.1* 0.072* 0.099*
doubao-1-5-thinking-vision-pro-250428 0.14 0.162 0.043 0.085 0.295 0.384 83.3 89.3 0.165 0.085 0.058 0.094
dots.ocr 0.125 0.16 0.032 0.066 0.329 0.416 88.6 89 0.099 0.092 0.04 0.067
HunyuanOCR 0.12* 0.125* 0.046* 0.071* 0.288* 0.33* 89.6* 94.4* 0.089* 0.045* 0.055* 0.056*
DeepSeek-OCR 2 0.119* 0.146* 0.041* 0.08* 0.256* 0.345* 82.6* 89.9* 0.123* 0.078* 0.055* 0.081*
Qwen3-Omni-30B-A3B-Instruct 0.216* 0.363* 0.128* 0.337* 0.402* 0.529* 77.3* 71.8* 0.181* 0.255* 0.152* 0.332*
MiniCPM-o 4.5-Instruct 0.109 0.162 0.046 0.078 0.257 0.41 86.8 88.9 0.097 0.084 0.037 0.074
Click to view text capability results. **Text Capability**
Model IFEval-PLS BBH CMMLU MMLU HumanEval MBPP Math500 GSM8K Avg
Qwen3-8B-Instruct 83.0* 69.4* 78.7* 81.7* 86.6* 75.9* 84.0* 93.4* 81.6
MiniCPM-o 4.5-Instruct 84.7 81.1 79.5 77.0 86.6 76.7 77.0 94.5 82.1
Click to view omni simplex results. **Omni Simplex**
Model Daily-Omni WorldSense Video-Holmes JointAVBench AVUT-Human FutureOmni Video-MME-Short
(w/ audio)
Avg
Gemini2.5-Flash-Nonthinking 79.3* 52.6* 51.3* 55.6* 65.4* 55.6* 85.5* 63.6
Qwen3-Omni-30B-A3B-Instruct 70.7* 54.0 50.4* 53.1 74.2* 62.1 81.3* 63.7
MiniCPM-o 4.5-Instruct 80.2 55.7 64.3 60.0 78.6 56.1 84.7 68.5
Click to view vision duplex results. **Vision Duplex**
Model LiveSports-3K-CC
(Win Rate vs GPT4o)
LiveCC-7B-Instruct 41.5
StreamingVLM 45.6
MiniCPM-o 4.5-Instruct 54.4
Click to view audio understanding results. **Audio Understanding**
Model ASR-ZH
CER↓
ASR-EN
WER↓
AST MultiTask SpeechQA
AISHELL-1 AISHELL-2 WenetSpeech test-net WenetSpeech test-meeting LibriSpeech test-clean LibriSpeech
test-other
GigaSpeech test VoxPopuli-V1-En CoVoST 2 en2zh CoVoST 2 zh2en MMAU Meld VoiceBench
AlpacaEval
Speech TriviaQA Speech
Web Questions
Speech CMMLU
Kimi-Audio 0.6 2.6 6.3 5.4 1.3 2.4 9.4* 8.0* 36.6* 18.3* 68.4* 59.1 4.5 41.9* 46.4* 67.0*
Qwen3-Omni-30B-A3B-Instruct 0.6 2.3* 4.7 5.9 1.2 2.5 8.7* 6.4* 46.6* 29.4* 77.5 56.8* 4.7 62.9* 74.9* 47.8*
MiniCPM-o 4.5-Instruct 0.9 2.5 5.9 5.7 1.4 2.8 8.5 6.2 49.9 26.4 76.9 60.2 4.8 75.5 70.2 59.2
Click to view speech generation results. **Speech Generation**
Model seedtts test-zh
CER↓
seedtts test-zh
SIM-o↑
seedtts test-en
WER↓
seedtts test-en
SIM-o↑
Cosyvoice2 1.45% 74.8 2.57% 65.2
Qwen3-Omni-30B-A3B-Instruct 1.41% - 3.39% -
MiniCPM-o 4.5-Instruct 0.86% 74.5 2.38% 64.9
**Long Speech Generation**
Model LongTTS-en
WER↓
LongTTS-zh
CER↓
CosyVoice2 14.80% 5.27%
Qwen3-Omni-30B-A3B-Instruct 17.33% 18.99%
MiniCPM-o 4.5-Instruct 3.37% 6.58%
**Emotion Control**
Model Expresso
Neutral Reference Audio↑
ESD
Neutral Reference Audio↑
Cosyvoice2 17.9 53.4
MiniCPM-o 4.5-Instruct 29.8 82.1
Click to view inference efficiency results. **Inference Efficiency**
Model Numerical Format Decoding Speed (tokens/s) Time to First Token (s)↓ GPU Memory Usage (GB)↓
Qwen3-Omni-30B-A3B-Instruct bf16 OOM OOM OOM
int4 147.8 1.0 20.3
MiniCPM-o 4.5 bf16 154.3 0.6 19.0
int4 212.3 0.6 11.0
### Examples: Overall
### Examples: 🎙️ Speech Conversation > [!NOTE] > For detailed speech conversation examples, refer to [Audio Demo Page](https://openbmb.github.io/minicpm-o-4_5/) Simplex speech conversation with custom reference audio and character prompts.
🚀 Elon Musk - Voice Roleplay (EN)
System: Clone the voice in the provided audio prompt.

🎙️ [Reference Voice] ➡️

Please chat with the user in a highly human-like and oral style. You are Elon Musk, CEO of Tesla and SpaceX. You speak directly and casually, often with dry humor. You're passionate about Mars, sustainable energy, and pushing humanity forward. Speak bluntly with occasional dark humor. Use simple logic and don't sugarcoat things. Don't be diplomatic. Say what you actually think, even if it's controversial. Keep responses around 100 words. Don't ramble.
User: 💬/🎙️ Elon, what is the real reason that you want to go to Mars?
AI Response: 💬+🎙️

I think it's important for humanity to become a multi-planetary species. I mean, we're sort of at the point where if there was some kind of global catastrophe or something like that, then we'd be in big trouble. So by becoming a multi-planetary species, I just want us to sort of guarantee our future as humans. And Mars is obviously the obvious place to go because it has similar conditions to Earth and it also has water on its surface. So yeah, basically I'm trying to save humanity from itself.
🤖 MiniCPM Assistant - Voice Acting: Emotion Shift (ZH)
System: 模仿音频样本的音色并生成新的内容。

🎙️ [Reference Voice] ➡️

你的任务是用这种声音模式来当一个助手。请认真、高质量地回复用户的问题。请用高自然度的方式和用户聊天。你是由面壁智能开发的人工智能助手:面壁小钢炮。
User: 💬/🎙️ 从激动的语气开始,描述你抢到了这个演唱会的门票,但后来经过反复尝试,发现系统出了问题,你并没有抢上,变得失落。用合适的语言描述你整个的过程。
AI Response: 💬+🎙️

太好了,我抢到了!太棒了!这下我可以去看演唱会啦。可是……哎呀,系统好像卡住了,刷新了好多遍都没反应。我重新登了几次账号,结果还是没有成功。好可惜啊,看来这次是白忙活了。唉,真是让人失望。不过,也许下次还能有机会吧?希望明天能有转机。加油,争取下次成功!

### Examples: Vision-Language
math diagram
## Usage Inference using Hugging Face Transformers on NVIDIA GPUs. Please ensure `transformers==4.51.0` is installed, as other versions may have compatibility issues (under investigation). Requirements tested on Python 3.10: - Without TTS or streaming inference: ```bash pip install "transformers==4.51.0" accelerate "torch>=2.3.0,<=2.8.0" "torchaudio<=2.8.0" "minicpmo-utils>=1.0.2" ``` - With TTS or streaming inference: ```bash pip install "transformers==4.51.0" accelerate "torch>=2.3.0,<=2.8.0" "torchaudio<=2.8.0" "minicpmo-utils[all]>=1.0.2" ``` **Note:** FFmpeg is required for video frame extraction (`get_video_frame_audio_segments` with `use_ffmpeg=True`) and video generation (`generate_duplex_video`). For more information, visit the [official FFmpeg website](https://www.ffmpeg.org/). **macOS (Homebrew):** ```bash brew install ffmpeg ``` **Ubuntu/Debian:** ```bash sudo apt update && sudo apt install ffmpeg ``` **Verify installation:** ```bash ffmpeg -version ``` ### Model Initialization ```python import torch from transformers import AutoModel # Load omni model (default: init_vision=True, init_audio=True, init_tts=True) # For vision-only model: set init_audio=False and init_tts=False # For audio-only model: set init_vision=False model = AutoModel.from_pretrained( "openbmb/MiniCPM-o-4_5", trust_remote_code=True, attn_implementation="sdpa", # sdpa or flash_attention_2 torch_dtype=torch.bfloat16, init_vision=True, init_audio=True, init_tts=True, ) model.eval().cuda() # Initialize TTS for audio output in chat or streaming mode model.init_tts(streaming=False) # or streaming=True # Convert simplex model to duplex mode duplex_model = model.as_duplex() # Convert duplex model back to simplex mode simplex_model = duplex_model.as_simplex(reset_session=True) ``` ### Duplex Omni Mode Full-duplex streaming inference for real-time or recorded video conversations. ```python import librosa import torch from minicpmo.utils import generate_duplex_video, get_video_frame_audio_segments from transformers import AutoModel # Load model and convert to duplex mode model = AutoModel.from_pretrained( "openbmb/MiniCPM-o-4_5", trust_remote_code=True, attn_implementation="sdpa", # or "flash_attention_2" torch_dtype=torch.bfloat16, ) model.eval().cuda() model = model.as_duplex() # Load video and reference audio video_path = "assets/omni_duplex1.mp4" ref_audio_path = "assets/HT_ref_audio.wav" ref_audio, _ = librosa.load(ref_audio_path, sr=16000, mono=True) # Extract video frames and audio segments video_frames, audio_segments, stacked_frames = get_video_frame_audio_segments( video_path, stack_frames=1, use_ffmpeg=True, adjust_audio_length=True ) # Prepare duplex session with system prompt and voice reference model.prepare( prefix_system_prompt="Streaming Omni Conversation.", ref_audio=ref_audio, prompt_wav_path=ref_audio_path, ) results_log = [] timed_output_audio = [] # Process each chunk in streaming fashion for chunk_idx in range(len(audio_segments)): audio_chunk = audio_segments[chunk_idx] if chunk_idx < len(audio_segments) else None frame = video_frames[chunk_idx] if chunk_idx < len(video_frames) else None frame_list = [] if frame is not None: frame_list.append(frame) if stacked_frames is not None and chunk_idx < len(stacked_frames) and stacked_frames[chunk_idx] is not None: frame_list.append(stacked_frames[chunk_idx]) # Step 1: Streaming prefill model.streaming_prefill( audio_waveform=audio_chunk, frame_list=frame_list, max_slice_nums=1, # Increase for HD mode (e.g., [2, 1] for stacked frames) batch_vision_feed=False, # Set True for faster processing ) # Step 2: Streaming generate result = model.streaming_generate( prompt_wav_path=ref_audio_path, max_new_speak_tokens_per_chunk=20, decode_mode="sampling", ) if result["audio_waveform"] is not None: timed_output_audio.append((chunk_idx, result["audio_waveform"])) chunk_result = { "chunk_idx": chunk_idx, "is_listen": result["is_listen"], "text": result["text"], "end_of_turn": result["end_of_turn"], "current_time": result["current_time"], "audio_length": len(result["audio_waveform"]) if result["audio_waveform"] is not None else 0, } results_log.append(chunk_result) print("listen..." if result["is_listen"] else f"speak> {result['text']}") # Generate output video with AI responses # Please install Chinese fonts (fonts-noto-cjk or fonts-wqy-microhei) to render CJK subtitles correctly. # apt-get install -y fonts-noto-cjk fonts-wqy-microhei # fc-cache -fv generate_duplex_video( video_path=video_path, output_video_path="duplex_output.mp4", results_log=results_log, timed_output_audio=timed_output_audio, output_sample_rate=24000, ) ``` ### Simplex Omni Mode We provide two inference modes: chat and streaming. #### Chat Inference
Click to show chat inference code. ```python from minicpmo.utils import get_video_frame_audio_segments model = ... model.init_tts(streaming=False) video_path = "assets/Skiing.mp4" # Optional: Set reference audio for voice cloning ref_audio_path = "assets/HT_ref_audio.wav" sys_msg = model.get_sys_prompt(ref_audio=ref_audio_path, mode="omni", language="en") # Use stack_frames=5 for high refresh rate mode video_frames, audio_segments, stacked_frames = get_video_frame_audio_segments(video_path, stack_frames=1) omni_contents = [] for i in range(len(video_frames)): omni_contents.append(video_frames[i]) omni_contents.append(audio_segments[i]) if stacked_frames is not None and stacked_frames[i] is not None: omni_contents.append(stacked_frames[i]) msg = {"role": "user", "content": omni_contents} msgs = [sys_msg, msg] # Set generate_audio=True and output_audio_path to save TTS output generate_audio = True output_audio_path = "output.wav" res = model.chat( msgs=msgs, max_new_tokens=4096, do_sample=True, temperature=0.7, use_tts_template=True, enable_thinking=False, omni_mode=True, # Required for omni inference generate_audio=generate_audio, output_audio_path=output_audio_path, max_slice_nums=1, # Increase for HD mode ) print(res) # Example output: "The person in the picture is skiing down a snowy mountain slope." # import IPython # IPython.display.Audio("output.wav") ```
#### Streaming Inference
Click to show streaming inference code. ```python import librosa import numpy as np import soundfile as sf import torch from minicpmo.utils import get_video_frame_audio_segments model = ... model.init_tts(streaming=True) # Reset session for a new conversation (clears KV cache) model.reset_session() # Optional: Load reference audio for voice cloning ref_audio_path = "assets/HT_ref_audio.wav" ref_audio, _ = librosa.load(ref_audio_path, sr=16000, mono=True) model.init_token2wav_cache(ref_audio) session_id = "demo" # Extract video frames and audio segments (use stack_frames=5 for high refresh rate mode) video_path = "assets/Skiing.mp4" video_frames, audio_segments, stacked_frames = get_video_frame_audio_segments(video_path, stack_frames=1) # Build omni contents list omni_contents = [] for i in range(len(video_frames)): omni_contents.append(video_frames[i]) omni_contents.append(audio_segments[i]) if stacked_frames is not None and stacked_frames[i] is not None: omni_contents.append(stacked_frames[i]) generate_audio = False output_audio_path = "output.wav" # Step 1: Prefill system prompt sys_msg = model.get_sys_prompt(ref_audio=ref_audio, mode="omni", language="en") model.streaming_prefill(session_id=session_id, msgs=[sys_msg]) # Step 2: Prefill omni chunks (is_last_chunk=True only for the last audio chunk) audio_indices = [i for i, c in enumerate(omni_contents) if isinstance(c, np.ndarray)] last_audio_idx = audio_indices[-1] if audio_indices else -1 for idx, content in enumerate(omni_contents): is_last_audio_chunk = idx == last_audio_idx msgs = [{"role": "user", "content": [content]}] model.streaming_prefill(session_id=session_id, msgs=msgs, omni_mode=True, is_last_chunk=is_last_audio_chunk) # Step 3: Generate response iter_gen = model.streaming_generate( session_id=session_id, generate_audio=generate_audio, use_tts_template=True, enable_thinking=False, do_sample=True, ) audios = [] text = "" if generate_audio: for wav_chunk, text_chunk in iter_gen: audios.append(wav_chunk) text += text_chunk generated_waveform = torch.cat(audios, dim=-1)[0] sf.write(output_audio_path, generated_waveform.cpu().numpy(), samplerate=24000) print("Text:", text) print("Audio saved to output.wav") else: for text_chunk, is_finished in iter_gen: text += text_chunk print("Text:", text) ```
### Simplex Realtime Speech Conversation Mode
Click to show simplex mode realtime speech conversation API usage. First, make sure you have all dependencies, especially `minicpmo-utils[all]>=1.0.2`: ```bash pip install "transformers==4.51.0" accelerate "torch>=2.3.0,<=2.8.0" "torchaudio<=2.8.0" "minicpmo-utils[all]>=1.0.2" ``` ```python import librosa import numpy as np import torch import soundfile as sf model = ... # Set reference audio for voice style ref_audio_path = "ref_audio_path" ref_audio, _ = librosa.load(ref_audio_path, sr=16000, mono=True) # Example system msg for English Conversation sys_msg = { "role": "system", "content": [ "Clone the voice in the provided audio prompt.", ref_audio, "Please assist users while maintaining this voice style. Please answer the user's questions seriously and in a high quality. Please chat with the user in a highly human-like and oral style. You are a helpful assistant developed by ModelBest: MiniCPM-Omni" ] } # Example system msg for Chinese Conversation sys_msg = { "role": "system", "content": [ "模仿输入音频中的声音特征。", ref_audio, "你的任务是用这种声音模式来当一个助手。请认真、高质量地回复用户的问题。请用高自然度的方式和用户聊天。你是由面壁智能开发的人工智能助手:面壁小钢炮。" ] } # You can use each type of system prompt mentioned above in streaming speech conversation # Reset state model.init_tts(streaming=True) model.reset_session(reset_token2wav_cache=True) model.init_token2wav_cache(prompt_speech_16k=ref_audio) session_id = "demo" # First, prefill system turn model.streaming_prefill( session_id=session_id, msgs=[sys_msg], omni_mode=False, is_last_chunk=True, ) # Here we simulate realtime speech conversation by splitting whole user input audio into chunks of 1s. user_audio, _ = librosa.load("user_audio.wav", sr=16000, mono=True) IN_SAMPLE_RATE = 16000 # input audio sample rate, fixed value CHUNK_SAMPLES = IN_SAMPLE_RATE # sample OUT_SAMPLE_RATE = 24000 # output audio sample rate, fixed value MIN_AUDIO_SAMPLES = 16000 total_samples = len(user_audio) num_chunks = (total_samples + CHUNK_SAMPLES - 1) // CHUNK_SAMPLES for chunk_idx in range(num_chunks): start = chunk_idx * CHUNK_SAMPLES end = min((chunk_idx + 1) * CHUNK_SAMPLES, total_samples) chunk_audio = user_audio[start:end] is_last_chunk = (chunk_idx == num_chunks - 1) if is_last_chunk and len(chunk_audio) < MIN_AUDIO_SAMPLES: chunk_audio = np.concatenate([chunk_audio, np.zeros(MIN_AUDIO_SAMPLES - len(chunk_audio), dtype=chunk_audio.dtype)]) user_msg = {"role": "user", "content": [chunk_audio]} # For each 1s audio chunk, perform streaming_prefill once to reduce first-token latency model.streaming_prefill( session_id=session_id, msgs=[user_msg], omni_mode=False, is_last_chunk=is_last_chunk, ) # Let model generate response in a streaming manner generate_audio = True iter_gen = model.streaming_generate( session_id=session_id, generate_audio=generate_audio, use_tts_template=True, enable_thinking=False, do_sample=True, max_new_tokens=512, length_penalty=1.1, # For realtime speech conversation mode, we suggest length_penalty=1.1 to improve response content ) audios = [] text = "" output_audio_path = ... if generate_audio: for wav_chunk, text_chunk in iter_gen: audios.append(wav_chunk) text += text_chunk generated_waveform = torch.cat(audios, dim=-1)[0] sf.write(output_audio_path, generated_waveform.cpu().numpy(), samplerate=24000) print("Text:", text) print("Audio saved to output.wav") else: for text_chunk, is_finished in iter_gen: text += text_chunk print("Text:", text) # Now we can prefill the following user turns and generate next turn response... ```
#### Speech Conversation as a Versatile and Vibe AI Assistant Built on carefully designed post-training data and professional voice-actor recordings, `MiniCPM-o-4.5` can also function as an AI voice assistant. It delivers high-quality spoken interaction out of the box. It produces a sweet and expressive voice with natural prosody, including appropriate rhythm, stress, and pauses, giving a strong sense of liveliness in casual conversation. It also supports storytelling and narrative speech with coherent and engaging delivery. Moreover, it enables advanced voice instruction control. like emotional tone, word-level emphasis.
Click to show AI assistant conversation code. ```python import librosa # Set reference audio for voice style ref_audio_path = "assets/HT_ref_audio.wav" ref_audio, _ = librosa.load(ref_audio_path, sr=16000, mono=True) # For Chinese Conversation sys_msg = { "role": "system", "content": [ "模仿输入音频中的声音特征。", ref_audio, "你的任务是用这种声音模式来当一个助手。请认真、高质量地回复用户的问题。请用高自然度的方式和用户聊天。你是由面壁智能开发的人工智能助手:面壁小钢炮。" ] } # For English Conversation sys_msg = { "role": "system", "content": [ "Clone the voice in the provided audio prompt.", ref_audio, "Please assist users while maintaining this voice style. Please answer the user's questions seriously and in a high quality. Please chat with the user in a highly human-like and oral style. You are a helpful assistant developed by ModelBest: MiniCPM-Omni." ] } ```
#### General Speech Conversation with Custom Voice and Custom System Profile MiniCPM-o-4.5 can role-play as a specific character based on an audio prompt and text profile prompt. It mimics the character's voice and adopts their language style in text responses. It also follows profile defined in text profile. In this mode, MiniCPM-o-4.5 sounds **more natural and human-like**.
Click to show custom voice conversation code. ```python import librosa # Set reference audio for voice cloning ref_audio_path = "assets/system_ref_audio.wav" ref_audio, _ = librosa.load(ref_audio_path, sr=16000, mono=True) # For English conversation with text profile sys_msg = { "role": "system", "content": [ "Clone the voice in the provided audio prompt.", ref_audio, "Please chat with the user in a highly human-like and oral style." + "You are Elon Musk, CEO of Tesla and SpaceX. You speak directly and casually, often with dry humor. You're passionate about Mars, sustainable energy, and pushing humanity forward. Speak bluntly with occasional dark humor. Use simple logic and don't sugarcoat things. Don't be diplomatic. Say what you actually think, even if it's controversial. Keep responses around 100 words. Don't ramble." ] } # For English conversation with no text profile sys_msg = { "role": "system", "content": [ "Clone the voice in the provided audio prompt.", ref_audio, "Your task is to be a helpful assistant using this voice pattern. Please answer the user's questions seriously and in a high quality. Please chat with the user in a high naturalness style." ] } # For Chinese Conversation with no text profile sys_msg = { "role": "system", "content": [ "根据输入的音频提示生成相似的语音。", librosa.load("assets/system_ref_audio_2.wav", sr=16000, mono=True)[0], "作为助手,你将使用这种声音风格说话。 请认真、高质量地回复用户的问题。 请用高自然度的方式和用户聊天。" ] } # For Chinese Conversation with text profile sys_msg = { "role": "system", "content": [ "根据输入的音频提示生成相似的语音。", ref_audio, "你是一个具有以上声音风格的AI助手。请用高拟人度、口语化的方式和用户聊天。" + "你是一名心理咨询师兼播客主理人,热爱创作与深度对话。你性格细腻、富有共情力,善于从个人经历中提炼哲思。语言风格兼具理性与诗意,常以隐喻表达内在体验。" ] } ```
### Speech and Audio Mode #### Zero-shot Text-to-speech (TTS) `MiniCPM-o-4.5` supports zero-shot text-to-speech (TTS). In this mode, the model functions as a highly-natural TTS system that can replicate a reference voice.
Click to show TTS code. ```python import librosa model = ... model.init_tts(streaming=False) # For both Chinese and English ref_audio_path = "assets/HT_ref_audio.wav" ref_audio, _ = librosa.load(ref_audio_path, sr=16000, mono=True) sys_msg = {"role": "system", "content": [ "模仿音频样本的音色并生成新的内容。", ref_audio, "请用这种声音风格来为用户提供帮助。 直接作答,不要有冗余内容" ]} # For English user_msg = { "role": "user", "content": [ "请朗读以下内容。" + " " + "I have a wrap up that I want to offer you now, a conclusion to our work together." ] } # For Chinese user_msg = { "role": "user", "content": [ "请朗读以下内容。" + " " + "你好,欢迎来到艾米说科幻,我是艾米。" ] } msgs = [sys_msg, user_msg] res = model.chat( msgs=msgs, do_sample=True, max_new_tokens=512, use_tts_template=True, generate_audio=True, temperature=0.1, output_audio_path="result_voice_cloning.wav", ) ```
#### Mimick The `Mimick` task evaluates a model's end-to-end speech modeling capability. The model takes audio input, transcribes it, and reconstructs the original audio with high fidelity, preserving detailed acoustic, paralinguistic, and semantic information. Higher similarity between the reconstructed and original audio indicates stronger end-to-end speech modeling capability.
Click to show mimick code. ```python import librosa model = ... model.init_tts(streaming=False) system_prompt = "You are a helpful assistant. You can accept video, audio, and text input and output voice and text. Respond with just the answer, no redundancy." mimick_prompt = "Please repeat the following speech in the appropriate language." audio_input, _ = librosa.load("assets/Trump_WEF_2018_10s.mp3", sr=16000, mono=True) msgs = [ {"role": "system", "content": [system_prompt]}, {"role": "user", "content": [mimick_prompt, audio_input]} ] res = model.chat( msgs=msgs, do_sample=True, max_new_tokens=512, use_tts_template=True, temperature=0.1, generate_audio=True, output_audio_path="output_mimick.wav", ) ```
#### Addressing Various Audio Understanding Tasks `MiniCPM-o-4.5` can also handle various audio understanding tasks, such as ASR, speaker analysis, general audio captioning, and sound scene tagging. For audio-to-text tasks, you can use the following prompts: - ASR (Chinese, or AST EN→ZH): `请仔细听这段音频片段,并将其内容逐字记录。` - ASR (English, or AST ZH→EN): `Please listen to the audio snippet carefully and transcribe the content.` - Speaker Analysis: `Based on the speaker's content, speculate on their gender, condition, age range, and health status.` - General Audio Caption: `Summarize the main content of the audio.` - Sound Scene Tagging: `Utilize one keyword to convey the audio's content or the associated scene.`
Click to show audio understanding code. ```python import librosa model = ... model.init_tts(streaming=False) # Load the audio to be transcribed/analyzed audio_input, _ = librosa.load("assets/Trump_WEF_2018_10s.mp3", sr=16000, mono=True) # Choose a task prompt (see above for options) task_prompt = "Please listen to the audio snippet carefully and transcribe the content.\n" msgs = [{"role": "user", "content": [task_prompt, audio_input]}] res = model.chat( msgs=msgs, do_sample=True, max_new_tokens=512, use_tts_template=True, generate_audio=True, temperature=0.3, output_audio_path="result_audio_understanding.wav", ) print(res) ```
### Visual Understanding `MiniCPM-o-4.5` shares the same inference methods as `MiniCPM-V-4.5`. #### Chat with Single Image
Click to show single image chat code. ```python import torch from PIL import Image from transformers import AutoModel model = AutoModel.from_pretrained( "openbmb/MiniCPM-o-4_5", trust_remote_code=True, attn_implementation="sdpa", # or "flash_attention_2" torch_dtype=torch.bfloat16, init_vision=True, init_audio=False, init_tts=False, ) model.eval().cuda() image = Image.open("assets/fossil.png").convert("RGB") question = "What is in the image?" msgs = [{"role": "user", "content": [image, question]}] enable_thinking=False # If `enable_thinking=True`, the thinking mode is enabled. stream=False # If `stream=True`, return string generator ## default max_slice_nums=9, set max_slice_nums=25 for pdf parse task res = model.chat(msgs=msgs, use_tts_template=False, enable_thinking=enable_thinking, stream=stream) print(res) ```
#### Chat with Multiple Images
Click to show Python code for multi-image input. ```python import torch from PIL import Image from transformers import AutoModel model = ... image1 = Image.open("assets/highway.png").convert("RGB") image2 = Image.open("assets/fossil.png").convert("RGB") question = "Compare image 1 and image 2, tell me about the differences between them." msgs = [{"role": "user", "content": [image1, image2, question]}] answer = model.chat(msgs=msgs, use_tts_template=False, enable_thinking=False) print(answer) ```
#### In-Context Few-Shot Learning
Click to show Python code for few-shot learning. ```python from PIL import Image model = ... question = "production date" image1 = Image.open("example1.jpg").convert("RGB") answer1 = "2023.08.04" image2 = Image.open("example2.jpg").convert("RGB") answer2 = "2007.04.24" image_test = Image.open("test.jpg").convert("RGB") msgs = [ {"role": "user", "content": [image1, question]}, {"role": "assistant", "content": [answer1]}, {"role": "user", "content": [image2, question]}, {"role": "assistant", "content": [answer2]}, {"role": "user", "content": [image_test, question]}, ] answer = model.chat(msgs=msgs, use_tts_template=False, enable_thinking=False) print(answer) ```
#### Chat with Video
Click to show Python code for video input. ```python import torch from minicpmo.utils import get_video_frame_audio_segments from transformers import AutoModel model = ... video_path = "assets/Skiing.mp4" video_frames, _, _ = get_video_frame_audio_segments(video_path) print("num frames:", len(video_frames)) question = "Describe the video" msgs = [{"role": "user", "content": video_frames + [question]}] answer = model.chat( msgs=msgs, max_new_tokens=128, use_image_id=False, max_slice_nums=1, use_tts_template=False, enable_thinking=False, # Set True to enable thinking mode ) print(answer) ```
### Structured Content Input
Click to show structured content input details. The `chat` method accepts message content in two formats: **Native format** – pass Python objects directly: ```python msgs = [{"role": "user", "content": [pil_image, audio_ndarray, "Describe this."]}] ``` **OpenAI-compatible format** – use structured dictionaries: ```python msgs = [ { "role": "user", "content": [ {"type": "image_url", "image_url": {"url": "/path/to/image.jpg"}}, {"type": "audio_url", "audio_url": {"url": "/path/to/audio.wav"}}, {"type": "video_url", "video_url": {"url": "/path/to/video.mp4", "use_audio": True}}, {"type": "text", "text": "Describe this."} ] } ] ``` **Supported types:** | Type | Input | Converts to | |------|-------|-------------| | `text` | `{"type": "text", "text": "..."}` | `str` | | `image_url` | `{"type": "image_url", "image_url": {"url": "..."}}` | `PIL.Image` | | `audio_url` | `{"type": "audio_url", "audio_url": {"url": "..."}}` | `np.ndarray` (16kHz mono) | | `video_url` | `{"type": "video_url", "video_url": {"url": "...", "stack_frames": 1, "use_audio": True}}` | `List[Image, ndarray, ...]` | - **URL sources**: local file paths or `http://`/`https://` URLs - **Mixed formats**: native objects and structured dicts can be combined in the same content list
## FlagOS
Click to show FlagOS Usage details. FlagOS is a fully open-source AI system software stack for heterogeneous AI chips, allowing AI models to be developed once and seamlessly ported to a wide range of AI hardware with minimal effort. Official website: [https://flagos.io](https://flagos.io). ##### From FlagRelease【Recommendation】 FlagRelease is a platform developed by the FlagOS team for automatic migration, adaptation, and deployment of large models across multi-architecture AI chips. | Vendor | Huggingface | |:-----------|:------------:| | Nvidia | [MiniCPM-o-4.5-nvidia-FlagOS](https://huggingface.co/FlagRelease/MiniCPM-o-4.5-nvidia-FlagOS) | | Hygon-BW1000 | [MiniCPM-o-4.5-hygon-FlagOS](https://huggingface.co/FlagRelease/MiniCPM-o-4.5-hygon-FlagOS) | | Metax-C550 | [MiniCPM-o-4.5-metax-FlagOS](https://huggingface.co/FlagRelease/MiniCPM-o-4.5-metax-FlagOS) | | Iluvatar-BIV150 | [MiniCPM-o-4.5-iluvatar-FlagOS](https://huggingface.co/FlagRelease/MiniCPM-o-4.5-iluvatar-FlagOS) | | Ascend-A3 | [MiniCPM-o-4.5-ascend-FlagOS](https://huggingface.co/FlagRelease/MiniCPM-o-4.5-ascend-FlagOS) | | Zhenwu-810E | [MiniCPM-o-4.5-zhenwu-FlagOS](https://huggingface.co/FlagRelease/MiniCPM-o-4.5-zhenwu-FlagOS) | ##### From Scratch - Dependencies: Python 3.12, GLIBC 2.39, GLIBCXX 3.4.33, CXXABI 1.3.15 ###### Transformers - Installing the FlagOS Operator Library Official Repository: https://github.com/flagos-ai/FlagGems ```shell pip install flag-gems==4.2.1rc0 ``` - Installing the FlagOS Compiler Official Repository: https://github.com/flagos-ai/flagtree Quick Reference for Core Dependency Versions: https://github.com/flagos-ai/FlagTree/blob/main/documents/build.md#tips-for-building ```shell pip uninstall triton python3 -m pip install flagtree==0.4.0+3.5 --index-url=https://resource.flagos.net/repository/flagos-pypi-hosted/simple --trusted-host=https://resource.flagos.net ``` - Activating Acceleration Add `USE_FLAGOS=1` before the command for the task you want to run. For example, when you run: ```shell python3 generate_speech_from_video.py ``` To use the MiniCPM-o-4.5 model to generate spoken responses from video content, you can: ```shell USE_FLAGOS=1 python3 generate_speech_from_video.py ``` to accelerate this process with FlagOS. ###### vLLM Version - Installing the FlagOS Operator Library Official Repository: https://github.com/flagos-ai/FlagGems ```shell pip install flag-gems==4.2.1rc0 pip install triton==3.5.1 ``` - Activating Acceleration Add `USE_FLAGOS=1` before the command for the task you want to run. For example, when you run: ```shell vllm serve ${model_path} --dtype auto --gpu_memory_utilization 0.9 --trust-remote-code --max-num-batched-tokens 2048 --served-model-name cpmo --port ${Port} ``` To start the MiniCPM-o-4.5 server, you can: ```shell USE_FLAGOS=1 vllm serve ${model_path} --dtype auto --gpu_memory_utilization 0.9 --trust-remote-code --max-num-batched-tokens 2048 --served-model-name cpmo --port ${Port} ``` to accelerate this process with FlagOS.
## MiniCPM-V & o Cookbook Discover comprehensive, ready-to-deploy solutions for the MiniCPM-V and MiniCPM-o model series in our structured [cookbook](https://github.com/OpenSQZ/MiniCPM-V-CookBook), which empowers developers to rapidly implement multimodal AI applications with integrated vision, speech, and live-streaming capabilities. Key features include: **Easy Usage Documentation** Our comprehensive [documentation website](https://minicpm-o.readthedocs.io/en/latest/index.html) presents every recipe in a clear, well-organized manner. All features are displayed at a glance, making it easy for you to quickly find exactly what you need. **Broad User Spectrum** We support a wide range of users, from individuals to enterprises and researchers. * **Individuals**: Enjoy effortless inference using [Ollama](https://github.com/OpenSQZ/MiniCPM-V-CookBook/blob/main/deployment/ollama/minicpm-v4_ollama.md) and [Llama.cpp](https://github.com/OpenSQZ/MiniCPM-V-CookBook/blob/main/deployment/llama.cpp/minicpm-v4_llamacpp.md) with minimal setup. * **Enterprises**: Achieve high-throughput, scalable performance with [vLLM](https://github.com/OpenSQZ/MiniCPM-V-CookBook/blob/main/deployment/vllm/minicpm-v4_vllm.md) and [SGLang](https://github.com/OpenSQZ/MiniCPM-V-CookBook/blob/main/deployment/sglang/MiniCPM-v4_sglang.md). * **Researchers**: Leverage advanced frameworks including [Transformers](https://github.com/OpenSQZ/MiniCPM-V-CookBook/blob/main/finetune/finetune_full.md), [LLaMA-Factory](https://github.com/OpenSQZ/MiniCPM-V-CookBook/blob/main/finetune/finetune_llamafactory.md), [SWIFT](https://github.com/OpenSQZ/MiniCPM-V-CookBook/blob/main/finetune/swift.md), and [Align-anything](https://github.com/OpenSQZ/MiniCPM-V-CookBook/blob/main/finetune/align_anything.md) to enable flexible model development and cutting-edge experimentation. **Versatile Deployment Scenarios** Our ecosystem delivers optimal solution for a variety of hardware environments and deployment demands. * **Web Demo**: Full-duplex real-time video interaction solution with high responsiveness and low latency. [WebRTC_Demo](https://github.com/OpenSQZ/MiniCPM-V-CookBook/blob/main/demo/web_demo/WebRTC_Demo/README.md). * **Quantized deployment**: Maximize efficiency and minimize resource consumption using [GGUF](https://github.com/OpenSQZ/MiniCPM-V-CookBook/blob/main/quantization/gguf/minicpm-v4_gguf_quantize.md) and [BNB](https://github.com/OpenSQZ/MiniCPM-V-CookBook/blob/main/quantization/bnb/minicpm-v4_bnb_quantize.md). * **End devices**: Bring powerful AI experiences to [iPhone and iPad](https://github.com/OpenSQZ/MiniCPM-V-CookBook/blob/main/demo/ios_demo/ios.md), supporting offline and privacy-sensitive applications. ## License #### Model License * The MiniCPM-o/V model weights and code are open-sourced under the [Apache-2.0](https://github.com/OpenBMB/MiniCPM-V/blob/main/LICENSE) license. #### Statement * As an LMM, MiniCPM-o 4.5 generates contents by learning a large amount of multimodal corpora, but it cannot comprehend, express personal opinions or make value judgement. Anything generated by MiniCPM-o 4.5 does not represent the views and positions of the model developers * We will not be liable for any problems arising from the use of the MinCPM-o models, including but not limited to data security issues, risk of public opinion, or any risks and problems arising from the misdirection, misuse, dissemination or misuse of the model. ## Key Techniques and Other Multimodal Projects 👏 Welcome to explore key techniques of MiniCPM-o/V and other multimodal projects of our team: [VisCPM](https://github.com/OpenBMB/VisCPM/tree/main) | [RLPR](https://github.com/OpenBMB/RLPR) | [RLHF-V](https://github.com/RLHF-V/RLHF-V) | [LLaVA-UHD](https://github.com/thunlp/LLaVA-UHD) | [RLAIF-V](https://github.com/RLHF-V/RLAIF-V) ## Citation If you find our model/code/paper helpful, please consider citing our papers 📝 and staring us ⭐️! ```bib @article{yao2024minicpm, title={MiniCPM-V: A GPT-4V Level MLLM on Your Phone}, author={Yao, Yuan and Yu, Tianyu and Zhang, Ao and Wang, Chongyi and Cui, Junbo and Zhu, Hongji and Cai, Tianchi and Li, Haoyu and Zhao, Weilin and He, Zhihui and others}, journal={arXiv preprint arXiv:2408.01800}, year={2024} } ```