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--- |
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license: mit |
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pipeline_tag: image-text-to-text |
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library_name: transformers |
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base_model: |
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- OpenGVLab/InternViT-300M-448px |
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- internlm/internlm2_5-7b-chat |
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new_version: OpenGVLab/InternVL2_5-8B |
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base_model_relation: merge |
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language: |
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- multilingual |
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tags: |
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- internvl |
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- custom_code |
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--- |
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# InternOmni |
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## Quick Start |
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We provide an example code to run `InternOmni` using `transformers`. |
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> Please use transformers>=4.37.2 to ensure the model works normally. |
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### Inference with Transformers |
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```python |
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import numpy as np |
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import torch |
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import torchvision.transforms as T |
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from PIL import Image |
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from torchvision.transforms.functional import InterpolationMode |
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from transformers import AutoModel, AutoTokenizer |
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import librosa |
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from transformers.processing_utils import ProcessorMixin |
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import torch |
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class WhisperProcessor(ProcessorMixin): |
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attributes = ["feature_extractor"] |
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feature_extractor_class = "WhisperFeatureExtractor" |
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def __init__(self, feature_extractor): |
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super().__init__(feature_extractor) |
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self.current_processor = self.feature_extractor |
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self._in_target_context_manager = False |
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def get_decoder_prompt_ids(self, task=None, language=None, no_timestamps=True): |
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return self.tokenizer.get_decoder_prompt_ids(task=task, language=language, no_timestamps=no_timestamps) |
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def get_T_after_cnn(self,L_in, dilation=1): |
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for (padding, kernel_size, stride) in eval("[(1,3,1)] + [(1,3,2)] "): |
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L_out = L_in + 2 * padding - dilation * (kernel_size - 1) - 1 |
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L_out = 1 + L_out // stride |
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L_in = L_out |
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return L_out |
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def __call__(self, *args, **kwargs): |
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if self._in_target_context_manager: |
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return self.current_processor(*args, **kwargs) |
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audio = kwargs.pop("audio", None) |
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sampling_rate = kwargs.pop("sampling_rate", 16000) |
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text = kwargs.pop("text", None) |
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if len(args) > 0: |
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audio = args[0] |
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args = args[1:] |
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if audio is None and text is None: |
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raise ValueError("You need to specify either an `audio` or `text` input to process.") |
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if audio is not None: |
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L = (audio.shape[0] if audio.shape[0] <= 480000 else 480000) # max_length < 30s |
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mel_len = L // 160 |
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audio_len_after_cnn = self.get_T_after_cnn(mel_len) |
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audio_token_num = (audio_len_after_cnn - 2) // 2 + 1 |
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inputs = self.feature_extractor(audio, *args, sampling_rate=sampling_rate, **kwargs) |
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inputs['audio_len_after_cnn'] = torch.tensor(audio_len_after_cnn, dtype=torch.long) |
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inputs['audio_token_num'] = torch.tensor(audio_token_num, dtype=torch.long) |
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if text is not None: |
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encodings = self.tokenizer(text, **kwargs) |
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if text is None: |
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return inputs |
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elif audio is None: |
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return encodings |
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else: |
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inputs["labels"] = encodings["input_ids"] |
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return inputs |
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def batch_decode(self, *args, **kwargs): |
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return self.tokenizer.batch_decode(*args, **kwargs) |
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def decode(self, *args, **kwargs): |
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return self.tokenizer.decode(*args, **kwargs) |
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def get_prompt_ids(self, text: str, return_tensors="np"): |
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return self.tokenizer.get_prompt_ids(text, return_tensors=return_tensors) |
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IMAGENET_MEAN = (0.485, 0.456, 0.406) |
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IMAGENET_STD = (0.229, 0.224, 0.225) |
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def build_transform(input_size): |
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MEAN, STD = IMAGENET_MEAN, IMAGENET_STD |
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transform = T.Compose([ |
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T.Lambda(lambda img: img.convert('RGB') if img.mode != 'RGB' else img), |
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T.Resize((input_size, input_size), interpolation=InterpolationMode.BICUBIC), |
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T.ToTensor(), |
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T.Normalize(mean=MEAN, std=STD) |
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]) |
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return transform |
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def find_closest_aspect_ratio(aspect_ratio, target_ratios, width, height, image_size): |
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best_ratio_diff = float('inf') |
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best_ratio = (1, 1) |
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area = width * height |
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for ratio in target_ratios: |
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target_aspect_ratio = ratio[0] / ratio[1] |
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ratio_diff = abs(aspect_ratio - target_aspect_ratio) |
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if ratio_diff < best_ratio_diff: |
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best_ratio_diff = ratio_diff |
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best_ratio = ratio |
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elif ratio_diff == best_ratio_diff: |
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if area > 0.5 * image_size * image_size * ratio[0] * ratio[1]: |
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best_ratio = ratio |
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return best_ratio |
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def dynamic_preprocess(image, min_num=1, max_num=12, image_size=448, use_thumbnail=False): |
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orig_width, orig_height = image.size |
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aspect_ratio = orig_width / orig_height |
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# calculate the existing image aspect ratio |
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target_ratios = set( |
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(i, j) for n in range(min_num, max_num + 1) for i in range(1, n + 1) for j in range(1, n + 1) if |
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i * j <= max_num and i * j >= min_num) |
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target_ratios = sorted(target_ratios, key=lambda x: x[0] * x[1]) |
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# find the closest aspect ratio to the target |
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target_aspect_ratio = find_closest_aspect_ratio( |
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aspect_ratio, target_ratios, orig_width, orig_height, image_size) |
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# calculate the target width and height |
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target_width = image_size * target_aspect_ratio[0] |
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target_height = image_size * target_aspect_ratio[1] |
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blocks = target_aspect_ratio[0] * target_aspect_ratio[1] |
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# resize the image |
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resized_img = image.resize((target_width, target_height)) |
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processed_images = [] |
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for i in range(blocks): |
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box = ( |
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(i % (target_width // image_size)) * image_size, |
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(i // (target_width // image_size)) * image_size, |
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((i % (target_width // image_size)) + 1) * image_size, |
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((i // (target_width // image_size)) + 1) * image_size |
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) |
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# split the image |
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split_img = resized_img.crop(box) |
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processed_images.append(split_img) |
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assert len(processed_images) == blocks |
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if use_thumbnail and len(processed_images) != 1: |
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thumbnail_img = image.resize((image_size, image_size)) |
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processed_images.append(thumbnail_img) |
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return processed_images |
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def load_image(image_file, input_size=448, max_num=12): |
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image = Image.open(image_file).convert('RGB') |
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transform = build_transform(input_size=input_size) |
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images = dynamic_preprocess(image, image_size=input_size, use_thumbnail=True, max_num=max_num) |
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pixel_values = [transform(image) for image in images] |
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pixel_values = torch.stack(pixel_values) |
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return pixel_values |
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def load_audio(audio_file, audio_processor): |
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audio_values, _ = librosa.load(audio_file, sr=16000) # sample rate should be 16000 |
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audio_process_values = audio_processor(audio_values, sampling_rate=16000, return_tensors="pt") |
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input_features = audio_process_values['input_features'] |
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audio_len_after_cnn = audio_process_values['audio_len_after_cnn'] |
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audio_token_num = audio_process_values['audio_token_num'] |
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audio_input = {'audio_values': input_features, |
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'audio_len_after_cnn': audio_len_after_cnn, |
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'audio_token_num': audio_token_num, |
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} |
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return audio_input |
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path = 'OpenGVLab/InternOmni' |
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model = AutoModel.from_pretrained( |
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path, |
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torch_dtype=torch.bfloat16, |
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low_cpu_mem_usage=True, |
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trust_remote_code=True).eval().cuda() |
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tokenizer = AutoTokenizer.from_pretrained(path, trust_remote_code=True, use_fast=False) |
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audio_processor = WhisperProcessor.from_pretrained(path) |
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# set the max number of tiles in `max_num` |
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pixel_values = load_image('./1.jpg', max_num=12).to(torch.bfloat16).cuda() |
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audio = load_audio('./1.wav', audio_processor) |
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generation_config = dict(max_new_tokens=1024, do_sample=True) |
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# question = '请将这段语音识别成文字,并以文字形式展示出来。' |
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response = model.Audio_chat(tokenizer=tokenizer, pixel_values=pixel_values,audio=audio, question=None, generation_config) |
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print(f'Assistant: {response}') |
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``` |
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## License |
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This project is released under the MIT License. This project uses the pre-trained internVL2_8b as a component, which is licensed under the Apache License 2.0. |
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## Citation |
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If you find this project useful in your research, please consider citing: |
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```BibTeX |
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@article{chen2024expanding, |
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title={Expanding Performance Boundaries of Open-Source Multimodal Models with Model, Data, and Test-Time Scaling}, |
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author={Chen, Zhe and Wang, Weiyun and Cao, Yue and Liu, Yangzhou and Gao, Zhangwei and Cui, Erfei and Zhu, Jinguo and Ye, Shenglong and Tian, Hao and Liu, Zhaoyang and others}, |
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journal={arXiv preprint arXiv:2412.05271}, |
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year={2024} |
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} |
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@article{gao2024mini, |
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title={Mini-internvl: A flexible-transfer pocket multimodal model with 5\% parameters and 90\% performance}, |
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author={Gao, Zhangwei and Chen, Zhe and Cui, Erfei and Ren, Yiming and Wang, Weiyun and Zhu, Jinguo and Tian, Hao and Ye, Shenglong and He, Junjun and Zhu, Xizhou and others}, |
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journal={arXiv preprint arXiv:2410.16261}, |
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year={2024} |
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} |
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@article{chen2024far, |
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title={How Far Are We to GPT-4V? Closing the Gap to Commercial Multimodal Models with Open-Source Suites}, |
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author={Chen, Zhe and Wang, Weiyun and Tian, Hao and Ye, Shenglong and Gao, Zhangwei and Cui, Erfei and Tong, Wenwen and Hu, Kongzhi and Luo, Jiapeng and Ma, Zheng and others}, |
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journal={arXiv preprint arXiv:2404.16821}, |
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year={2024} |
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} |
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@inproceedings{chen2024internvl, |
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title={Internvl: Scaling up vision foundation models and aligning for generic visual-linguistic tasks}, |
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author={Chen, Zhe and Wu, Jiannan and Wang, Wenhai and Su, Weijie and Chen, Guo and Xing, Sen and Zhong, Muyan and Zhang, Qinglong and Zhu, Xizhou and Lu, Lewei and others}, |
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booktitle={Proceedings of the IEEE/CVF Conference on Computer Vision and Pattern Recognition}, |
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pages={24185--24198}, |
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year={2024} |
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} |
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``` |
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