openbmb/MiniCPM-o-4_5-awq API & Inference Endpoint

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 and Ollama support for efficient CPU inference on local devices, (2) int4 and GGUF format quantized models in 16 sizes, (3) vLLM and SGLang support for high-throughput and memory-efficient inference, (4) FlagOS support for the unified multi-chip backend plugin, (5) fine-tuning on new domains and tasks with LLaMA-Factory, and (6) online web demo on server. We also rollout a high-performing llama.cpp-omni inference framework together with a WebRTC Demo, which enables the full-duplex multimodal live streaming experience on local devices such as PCs (e.g., on a MacBook).

Model Architecture.

Evaluation

Image Understanding (Instruct)

Image Understanding (Thinking)

Video Understanding

OmniDocBench

Text Capability

Omni Simplex

Vision Duplex

Audio Understanding

Speech Generation

Long Speech Generation

Emotion Control

Inference Efficiency

Examples

🎙️ End-to-End Voice Chat Examples

Simplex speech conversation with custom reference audio and character prompts.

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: 💬/🎙️ (text or speech)

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.

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: 💬/🎙️ (text or speech)

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！

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"

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

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

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)

Speech and Audio Mode

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.

python
import librosa

model = ...
model.init_tts(streaming=False)

# 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。"
  ]
}


# Round 1
user_audio, _ = librosa.load("user_question.wav", sr=16000, mono=True)
user_msg = {"role": "user", "content": [user_audio]}

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=0.7,
    output_audio_path="result_roleplay_round_1.wav",
)
print("Round 1:", res)

# Round 2 (multi-turn conversation)
msgs.append({"role": "assistant", "content": res})
user_audio_2, _ = librosa.load("user_question_2.wav", sr=16000, mono=True)
msgs.append({"role": "user", "content": [user_audio_2]})

res = model.chat(
    msgs=msgs,
    do_sample=True,
    max_new_tokens=512,
    use_tts_template=True,
    generate_audio=True,
    temperature=0.7,
    output_audio_path="result_roleplay_round_2.wav",
)
print("Round 2:", res)

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.

python
import librosa

model = ...
model.init_tts(streaming=False)

# 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."
  ]
}


# Load user's audio question
user_audio, _ = librosa.load("user_question.wav", sr=16000, mono=True)
user_question = {"role": "user", "content": [user_audio]}

# Round 1
msgs = [sys_msg, user_question]
res = model.chat(
    msgs=msgs,
    do_sample=True,
    max_new_tokens=512,
    use_tts_template=True,
    generate_audio=True,
    temperature=0.7,
    output_audio_path="result_assistant_round_1.wav",
)

# Round 2
msgs.append({"role": "assistant", "content": res})
user_audio_2, _ = librosa.load("user_question_2.wav", sr=16000, mono=True)
msgs.append({"role": "user", "content": [user_audio_2]})
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_assistant_round_2.wav",
)
print(res)

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.

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",
)

Realtime Speech Conversation

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)

# 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"
msgs = [...]
# First, prefill system turn
for msg in msgs:
    if msg["role"] == "system":
        model.streaming_prefill(
            session_id=session_id,
            msgs=[msg],
            omni_mode=False,
            is_last_chunk=False,
        )

# 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
CHUNK_SAMPLES = IN_SAMPLE_RATE
OUT_SAMPLE_RATE = 24000

if user_audio is not None and len(user_audio) > 0:
    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)
        
        user_msg = {"role": "user", "content": [chunk_audio]}
        
        # For each 1s audio chunk, perform streaming_prefill once to reduce first-package latency
        model.streaming_prefill(
            session_id=session_id,
            msgs=[user_msg],
            omni_mode=False,
            is_last_chunk=is_last_chunk,
        )

# When VAD actives, make the model generate response in a streaming manner
# Here we create an iterator without returning anything
audio_iter = model.streaming_generate(
    session_id=session_id,
    generate_audio=True,
    max_new_tokens=512,
    do_sample=True,
    length_penalty=1.1, # For realtime speech conversation mode, we suggest length_penalty=1.1 to improve response content
)

# Yield 1s-length audio response from the iterator just created in a streaming manner
audio_chunks = []
for generated_audio in audio_iter:
    if generated_audio is None:
        break
    
    if isinstance(generated_audio, tuple):
        generated_audio = generated_audio[0]
        if generated_audio is None:
            break
    
    if isinstance(generated_audio, torch.Tensor):
        wav = generated_audio.detach().cpu().numpy()
    else:
        wav = np.asarray(generated_audio)
    
    if wav.ndim == 2:
        wav = wav[0]
    
    wav = wav.astype(np.float32)
    audio_chunks.append(wav)

output_audio_path = "realtime_speech.wav"
if audio_chunks:
    full_audio = np.concatenate(audio_chunks, axis=-1)
    sf.write(output_audio_path, full_audio, OUT_SAMPLE_RATE)

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.

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.

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

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

res = model.chat(msgs=msgs, use_tts_template=False, enable_thinking=enable_thinking, stream=stream)
print(res)

Chat with Multiple Images

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

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

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

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:

Table
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

Official website: https://flagos.io.

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, 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 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 and Llama.cpp with minimal setup.
Enterprises: Achieve high-throughput, scalable performance with vLLM and SGLang.
Researchers: Leverage advanced frameworks including Transformers, LLaMA-Factory, SWIFT, and Align-anything 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.
Quantized deployment: Maximize efficiency and minimize resource consumption using GGUF and BNB.
End devices: Bring powerful AI experiences to iPhone and iPad, 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 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 | RLPR | RLHF-V | LLaVA-UHD | 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}
}

MiniCPM-o-4_5-awq

Get help setting up a custom Dedicated Endpoints.

README