Ola Image

🚀 Ola-7B

The Ola-7B model is a remarkable creation developed by individuals from Tencent, Tsinghua University, and Nanyang Technological University. It builds upon the Qwen2.5 language model and is trained on diverse data types including text, image, video, and audio, with an impressive context window of 32K tokens. This model can accept image/video, text, and audio as inputs and generate text outputs, offering an on - demand solution for seamless and efficient processing of visual inputs with arbitrary spatial sizes and temporal lengths.

🚀 Quick Start

Installation

1. Download the speech encoder at https://huggingface.co/THUdyh/Ola_speech_encoders.
2. Replace the path in config.json with the local path of the speech encoders.

Usage

We provide a simple generation process for using our model. For more details, please refer to our Github Repo

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"
    
    # input audio and video, do not parse audio in the video, else parse audio in the video
    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('load audio')
    elif USE_SPEECH and not audio_path:
        # parse audio in the video
        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" + "User's question in speech: " + 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

✨ Features

• The model can take image/video, text, and audio as inputs and generate text outputs.
• It offers an on - demand solution to process visual inputs with arbitrary spatial sizes and temporal lengths.

📚 Documentation

Model Architecture

• Architecture: Pre - trained [Oryx - ViT](https://huggingface.co/THUdyh/Oryx - ViT) + Qwen2.5 - 7B
• Data: A mixture of more than 5M image/video/audio data, trained for 3 stages.
• Precision: BFloat16

Hardware & Software

• Hardware: 64 * NVIDIA Tesla A100
• Orchestration: HuggingFace Trainer
• Code: Pytorch

Model Information

📄 License

The model is licensed under the apache - 2.0 license.

🔧 Technical Details

The Ola - 7B model is based on the Qwen2.5 language model. It uses a pre - trained Oryx - ViT model and is trained on a large amount of image, video, and audio data. The training process is divided into 3 stages. The model has a context window of 32K tokens, which allows it to handle long - sequence inputs. The precision of the model is BFloat16, which can balance the computational efficiency and accuracy. The model is trained on 64 NVIDIA Tesla A100 GPUs using the HuggingFace Trainer with Pytorch code.

📝 Citation

@article{liu2025ola,
title={Ola: Pushing the Frontiers of Omni - Modal Language Model with Progressive Modality Alignment},
author={Liu, Zuyan and Dong, Yuhao and Wang, Jiahui and Liu, Ziwei and Hu, Winston and Lu, Jiwen and Rao, Yongming},
journal={arXiv preprint arXiv:2502.04328},
year={2025}
}