pipeline_tag: 图像文本到文本 license: apache-2.0 base_model:

• Qwen/Qwen2.5-7B-Instruct language:
• 英文
• 中文 datasets:
• HuggingFaceFV/finevideo

Ola-7B

模型概述

Ola-7B模型由腾讯、清华大学和南洋理工大学的研究人员联合开发。基于Qwen2.5语言模型，该模型在文本、图像、视频和音频数据上进行训练，上下文窗口为32K token。它能接收图像/视频、文本和音频作为输入，并输出文本。

Ola提供了一种按需解决方案，能够无缝且高效地处理任意空间尺寸和时间长度的视觉输入。

• 代码仓库: https://github.com/Ola-Omni/Ola
• 支持语言: 英语、中文
• 论文: https://huggingface.co/papers/2502.04328

使用方法

1. 从https://huggingface.co/THUdyh/Ola_speech_encoders下载语音编码器
2. 将config.json中的路径替换为语音编码器的本地路径

我们提供了一个简单的生成流程来使用我们的模型。更多细节请参考我们的GitHub仓库

import os
os.environ['LOWRES_RESIZE'] = '384x32'
os.environ['HIGHRES_BASE'] = '0x32'
os.environ['VIDEO_RESIZE'] = "0x64"
os.environ['VIDEO_MAXRES'] = "480"
os.environ['VIDEO_MINRES'] = "288"
os.environ['MAXRES'] = '1536'
os.environ['MINRES'] = '0'
os.environ['REGIONAL_POOL'] = '2x'
os.environ['FORCE_NO_DOWNSAMPLE'] = '1'
os.environ['LOAD_VISION_EARLY'] = '1'
os.environ['SKIP_LOAD_VIT'] = '1'
    

import gradio as gr
import torch
import re
from decord import VideoReader, cpu
from PIL import Image
import numpy as np
import transformers
import moviepy.editor as mp
from typing import Dict, Optional, Sequence, List
import librosa
import whisper
from ola.conversation import conv_templates, SeparatorStyle
from ola.model.builder import load_pretrained_model
from ola.utils import disable_torch_init
from ola.datasets.preprocess import tokenizer_image_token, tokenizer_speech_image_token, tokenizer_speech_question_image_token
from ola.mm_utils import get_model_name_from_path, KeywordsStoppingCriteria, process_anyres_video, process_anyres_highres_image_genli
from ola.constants import IGNORE_INDEX, DEFAULT_IMAGE_TOKEN, IMAGE_TOKEN_INDEX, DEFAULT_SPEECH_TOKEN

model_path = ""
tokenizer, model, image_processor, _ = load_pretrained_model(model_path, None)
model = model.to('cuda').eval()
model = model.bfloat16()

USE_SPEECH=False
cur_dir = os.path.dirname(os.path.abspath(__file__))


def load_audio(audio_file_name):
    speech_wav, samplerate = librosa.load(audio_file_name, sr=16000)
    if len(speech_wav.shape) > 1:
        speech_wav = speech_wav[:, 0]
    speech_wav = speech_wav.astype(np.float32)
    CHUNK_LIM = 480000
    SAMPLE_RATE = 16000
    speechs = []
    speech_wavs = []

    if len(speech_wav) <= CHUNK_LIM:
        speech = whisper.pad_or_trim(speech_wav)
        speech_wav = whisper.pad_or_trim(speech_wav)
        speechs.append(speech)
        speech_wavs.append(torch.from_numpy(speech_wav).unsqueeze(0))
    else:
        for i in range(0, len(speech_wav), CHUNK_LIM):
            chunk = speech_wav[i : i + CHUNK_LIM]
            if len(chunk) < CHUNK_LIM:
                chunk = whisper.pad_or_trim(chunk)
            speechs.append(chunk)
            speech_wavs.append(torch.from_numpy(chunk).unsqueeze(0))
    mels = []
    for chunk in speechs:
        chunk = whisper.log_mel_spectrogram(chunk, n_mels=128).permute(1, 0).unsqueeze(0)
        mels.append(chunk)

    mels = torch.cat(mels, dim=0)
    speech_wavs = torch.cat(speech_wavs, dim=0)
    if mels.shape[0] > 25:
        mels = mels[:25]
        speech_wavs = speech_wavs[:25]

    speech_length = torch.LongTensor([mels.shape[1]] * mels.shape[0])
    speech_chunks = torch.LongTensor([mels.shape[0]])
    return mels, speech_length, speech_chunks, speech_wavs

def extract_audio(videos_file_path):
    my_clip = mp.VideoFileClip(videos_file_path)
    return my_clip.audio

def ola_inference(multimodal, audio_path):
    visual, text = multimodal["files"][0], multimodal["text"]
    if visual.endswith("image2.png"):
        modality = "video"
        visual = f"{cur_dir}/case/case1.mp4"
    if visual.endswith(".mp4"):
        modality = "video"
    else:
        modality = "image"
    
    # 输入音频和视频时，不解析视频中的音频；否则解析视频中的音频
    if audio_path:
        USE_SPEECH = True
    elif modality == "video":
        USE_SPEECH = True
    else:
        USE_SPEECH = False
    
    speechs = []
    speech_lengths = []
    speech_wavs = []
    speech_chunks = []
    if modality == "video":
        vr = VideoReader(visual, ctx=cpu(0))
        total_frame_num = len(vr)
        fps = round(vr.get_avg_fps())
        uniform_sampled_frames = np.linspace(0, total_frame_num - 1, 64, dtype=int)
        frame_idx = uniform_sampled_frames.tolist()
        spare_frames = vr.get_batch(frame_idx).asnumpy()
        video = [Image.fromarray(frame) for frame in spare_frames]
    else:
        image = [Image.open(visual)]
        image_sizes = [image[0].size]

    if USE_SPEECH and audio_path:
        audio_path = audio_path
        speech, speech_length, speech_chunk, speech_wav = load_audio(audio_path)
        speechs.append(speech.bfloat16().to('cuda'))
        speech_lengths.append(speech_length.to('cuda'))
        speech_chunks.append(speech_chunk.to('cuda'))
        speech_wavs.append(speech_wav.to('cuda'))
        print('加载音频')
    elif USE_SPEECH and not audio_path:
        # 解析视频中的音频
        audio = extract_audio(visual)
        audio.write_audiofile("./video_audio.wav")
        video_audio_path = './video_audio.wav'
        speech, speech_length, speech_chunk, speech_wav = load_audio(video_audio_path)
        speechs.append(speech.bfloat16().to('cuda'))
        speech_lengths.append(speech_length.to('cuda'))
        speech_chunks.append(speech_chunk.to('cuda'))
        speech_wavs.append(speech_wav.to('cuda'))
    else:
        speechs = [torch.zeros(1, 3000, 128).bfloat16().to('cuda')]
        speech_lengths = [torch.LongTensor([3000]).to('cuda')]
        speech_wavs = [torch.zeros([1, 480000]).to('cuda')]
        speech_chunks = [torch.LongTensor([1]).to('cuda')]
    
    conv_mode = "qwen_1_5"
    if text:
        qs = text
    else:
        qs = ''
    if USE_SPEECH and audio_path:
        qs = DEFAULT_IMAGE_TOKEN + "\n" + "用户语音提问: " + DEFAULT_SPEECH_TOKEN + '\n'
    elif USE_SPEECH:
        qs = DEFAULT_SPEECH_TOKEN + DEFAULT_IMAGE_TOKEN + "\n" + qs
    else:
        qs = DEFAULT_IMAGE_TOKEN + "\n" + qs

    conv = conv_templates[conv_mode].copy()
    conv.append_message(conv.roles[0], qs)
    conv.append_message(conv.roles[1], None)
    prompt = conv.get_prompt()
    if USE_SPEECH and audio_path:
        input_ids = tokenizer_speech_question_image_token(prompt, tokenizer, IMAGE_TOKEN_INDEX, return_tensors="pt").unsqueeze(0).to('cuda')
    elif USE_SPEECH:
        input_ids = tokenizer_speech_image_token(prompt, tokenizer, IMAGE_TOKEN_INDEX, return_tensors="pt").unsqueeze(0).to('cuda')
    else:
        input_ids = tokenizer_image_token(prompt, tokenizer, IMAGE_TOKEN_INDEX, return_tensors="pt").unsqueeze(0).to('cuda')

    if modality == "video":
        video_processed = []
        for idx, frame in enumerate(video):
            image_processor.do_resize = False
            image_processor.do_center_crop = False
            frame = process_anyres_video(frame, image_processor)

            if frame_idx is not None and idx in frame_idx:
                video_processed.append(frame.unsqueeze(0))
            elif frame_idx is None:
                video_processed.append(frame.unsqueeze(0))
        
        if frame_idx is None:
            frame_idx = np.arange(0, len(video_processed), dtype=int).tolist()
        
        video_processed = torch.cat(video_processed, dim=0).bfloat16().to("cuda")
        video_processed = (video_processed, video_processed)

        video_data = (video_processed, (384, 384), "video")
    else:
        image_processor.do_resize = False
        image_processor.do_center_crop = False
        image_tensor, image_highres_tensor = [], []
        for visual in image:
            image_tensor_, image_highres_tensor_ = process_anyres_highres_image_genli(visual, image_processor)
            image_tensor.append(image_tensor_)
            image_highres_tensor.append(image_highres_tensor_)
        if all(x.shape == image_tensor[0].shape for x in image_tensor):
            image_tensor = torch.stack(image_tensor, dim=0)
        if all(x.shape == image_highres_tensor[0].shape for x in image_highres_tensor):
            image_highres_tensor = torch.stack(image_highres_tensor, dim=0)
        if type(image_tensor) is list:
            image_tensor = [_image.bfloat16().to("cuda") for _image in image_tensor]
        else:
            image_tensor = image_tensor.bfloat16().to("cuda")
        if type(image_highres_tensor) is list:
            image_highres_tensor = [_image.bfloat16().to("cuda") for _image in image_highres_tensor]
        else:
            image_highres_tensor = image_highres_tensor.bfloat16().to("cuda")

    pad_token_ids = 151643

    attention_masks = input_ids.ne(pad_token_ids).long().to('cuda')
    stop_str = conv.sep if conv.sep_style != SeparatorStyle.TWO else conv.sep2
    keywords = [stop_str]
    stopping_criteria = KeywordsStoppingCriteria(keywords, tokenizer, input_ids)

    gen_kwargs = {}

    if "max_new_tokens" not in gen_kwargs:
        gen_kwargs["max_new_tokens"] = 1024
    if "temperature" not in gen_kwargs:
        gen_kwargs["temperature"] = 0.2
    if "top_p" not in gen_kwargs:
        gen_kwargs["top_p"] = None
    if "num_beams" not in gen_kwargs:
        gen_kwargs["num_beams"] = 1

    with torch.inference_mode():
        if modality == "video":
            output_ids = model.generate(
                inputs=input_ids,
                images=video_data[0][0],
                images_highres=video_data[0][1],
                modalities=video_data[2],
                speech=speechs,
                speech_lengths=speech_lengths,
                speech_chunks=speech_chunks,
                speech_wav=speech_wavs,
                attention_mask=attention_masks,
                use_cache=True,
                stopping_criteria=[stopping_criteria],
                do_sample=True if gen_kwargs["temperature"] > 0 else False,
                temperature=gen_kwargs["temperature"],
                top_p=gen_kwargs["top_p"],
                num_beams=gen_kwargs["num_beams"],
                max_new_tokens=gen_kwargs["max_new_tokens"],
            )
        else:
            output_ids = model.generate(
                inputs=input_ids,
                images=image_tensor,
                images_highres=image_highres_tensor,
                image_sizes=image_sizes,
                modalities=['image'],
                speech=speechs,
                speech_lengths=speech_lengths,
                speech_chunks=speech_chunks,
                speech_wav=speech_wavs,
                attention_mask=attention_masks,
                use_cache=True,
                stopping_criteria=[stopping_criteria],
                do_sample=True if gen_kwargs["temperature"] > 0 else False,
                temperature=gen_kwargs["temperature"],
                top_p=gen_kwargs["top_p"],
                num_beams=gen_kwargs["num_beams"],
                max_new_tokens=gen_kwargs["max_new_tokens"],
            )

    outputs = tokenizer.batch_decode(output_ids, skip_special_tokens=True)[0]
    outputs = outputs.strip()
    if outputs.endswith(stop_str):
        outputs = outputs[:-len(stop_str)]
    outputs = outputs.strip()
    return outputs, None

模型架构

• 架构: 预训练的Oryx-ViT + Qwen2.5-7B

• 数据: 混合超过500万张图像/视频/音频数据，分三个阶段训练

• 精度: BFloat16

硬件与软件

• 硬件: 64张NVIDIA Tesla A100

• 编排: HuggingFace Trainer

• 代码: Pytorch

引用

@article{liu2025ola, title={Ola: 通过渐进式模态对齐推动全模态语言模型前沿}, author={刘祖岩 and 董宇豪 and 王家辉 and 刘子威 and 胡文森 and 陆继文 and 饶永明}, journal={arXiv预印本 arXiv:2502.04328}, year={2025} }