Any-to-Any
Transformers
Safetensors
multilingual
minicpmo
feature-extraction
minicpm-o
omni
vision
ocr
multi-image
video
custom_code
audio
speech
voice cloning
live Streaming
realtime speech conversation
asr
tts
Instructions to use Y0316/MiniCPM-o-2_6 with libraries, inference providers, notebooks, and local apps. Follow these links to get started.
- Libraries
- Transformers
How to use Y0316/MiniCPM-o-2_6 with Transformers:
# Load model directly from transformers import AutoModel model = AutoModel.from_pretrained("Y0316/MiniCPM-o-2_6", trust_remote_code=True, dtype="auto") - Notebooks
- Google Colab
- Kaggle
| # coding=utf-8 | |
| # Copyright 2025 The OpenBMB Team. All rights reserved. | |
| # | |
| # Licensed under the Apache License, Version 2.0 (the "License"); | |
| # you may not use this file except in compliance with the License. | |
| # You may obtain a copy of the License at | |
| # | |
| # http://www.apache.org/licenses/LICENSE-2.0 | |
| # | |
| # Unless required by applicable law or agreed to in writing, software | |
| # distributed under the License is distributed on an "AS IS" BASIS, | |
| # WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. | |
| # See the License for the specific language governing permissions and | |
| # limitations under the License. | |
| """ | |
| Processor class for MiniCPMO. | |
| """ | |
| import math | |
| import re | |
| from typing import List | |
| from typing import Literal | |
| from typing import Optional | |
| from typing import Union | |
| import numpy as np | |
| import torch | |
| import torchaudio | |
| from transformers.image_utils import ImageInput | |
| from transformers.processing_utils import ProcessorMixin | |
| from transformers.tokenization_utils_base import PreTokenizedInput | |
| from transformers.tokenization_utils_base import TextInput | |
| from transformers.utils import TensorType | |
| from .image_processing_minicpmv import MiniCPMOBatchFeature | |
| class MiniCPMOProcessor(ProcessorMixin): | |
| r""" | |
| Constructs a MiniCPMV processor which wraps a MiniCPMV image processor and a MiniCPMV tokenizer into a single processor. | |
| [`MiniCPMVProcessor`] offers all the functionalities of [`MiniCPMVImageProcessor`] and [`LlamaTokenizerWrapper`]. See the | |
| [`~MiniCPMVProcessor.__call__`] and [`~MiniCPMVProcessor.decode`] for more information. | |
| Args: | |
| image_processor ([`MiniCPMVImageProcessor`], *optional*): | |
| The image processor is a required input. | |
| tokenizer ([`LlamaTokenizerWrapper`], *optional*): | |
| The tokenizer is a required input. | |
| """ | |
| attributes = ["image_processor", "feature_extractor", "tokenizer"] | |
| feature_extractor_class = "WhisperFeatureExtractor" | |
| image_processor_class = "AutoImageProcessor" | |
| tokenizer_class = "AutoTokenizer" | |
| def __init__(self, image_processor=None, feature_extractor=None, tokenizer=None): | |
| super().__init__(image_processor, feature_extractor, tokenizer) | |
| self.version = image_processor.version | |
| def __call__( | |
| self, | |
| text: Union[TextInput, PreTokenizedInput, List[TextInput], List[PreTokenizedInput]], | |
| images: ImageInput = None, | |
| audios: Union[np.ndarray, List[np.ndarray], List[List[np.ndarray]]] = None, | |
| audio_parts: Optional[list] = None, | |
| max_length: Optional[int] = None, | |
| do_pad: Optional[bool] = True, | |
| max_slice_nums: int = None, | |
| use_image_id: bool = True, | |
| chunk_input: bool = False, | |
| return_tensors: Optional[Union[str, TensorType]] = TensorType.PYTORCH, | |
| sampling_rate: Optional[int] = 16000, | |
| **kwargs, | |
| ) -> MiniCPMOBatchFeature: | |
| if images is not None: | |
| image_inputs = self.image_processor( | |
| images, do_pad=do_pad, max_slice_nums=max_slice_nums, return_tensors=return_tensors | |
| ) | |
| else: | |
| image_inputs = None | |
| if audios is not None: | |
| audio_features, audio_feature_lens, audio_phs = self.audio_feature_extract( | |
| audios, audio_parts, chunk_input, sampling_rate | |
| ) | |
| else: | |
| audio_features, audio_feature_lens, audio_phs = [], [], [] | |
| model_inputs = self._convert_omni_to_inputs( | |
| image_inputs, | |
| audio_phs, | |
| text, | |
| max_slice_nums=max_slice_nums, | |
| use_image_id=use_image_id, | |
| max_length=max_length, | |
| **kwargs, | |
| ) | |
| model_inputs["audio_features"] = audio_features | |
| model_inputs["audio_feature_lens"] = audio_feature_lens | |
| return MiniCPMOBatchFeature(data={**model_inputs}) | |
| def get_audio_placeholder(self, audio_lens, chunk_input, chunk_length): | |
| pool_step = 2 | |
| feature_lens = math.ceil(audio_lens / self.feature_extractor.hop_length) | |
| feature_lens = (feature_lens - 1) // 2 + 1 | |
| output_lens = (feature_lens - pool_step) // pool_step + 1 | |
| if chunk_input: | |
| fbank_feat_in_chunk = int(chunk_length * 100) | |
| cnn_feat_in_chunk = (fbank_feat_in_chunk - 1) // 2 + 1 | |
| audio_embeds_in_chunk = (cnn_feat_in_chunk - pool_step) // pool_step + 1 | |
| num_audio_chunks = (output_lens + audio_embeds_in_chunk - 1) // audio_embeds_in_chunk | |
| place_holders = "" | |
| total_unk_len = 0 | |
| for _ in range(num_audio_chunks): | |
| unk_len = min(audio_embeds_in_chunk, output_lens - total_unk_len) | |
| place_holders += self.tokenizer.audio_start + "<unk>" * unk_len + self.tokenizer.audio_end | |
| total_unk_len += unk_len | |
| audio_placeholder = place_holders | |
| else: | |
| audio_placeholder = self.tokenizer.audio_start + "<unk>" * output_lens + self.tokenizer.audio_end | |
| return audio_placeholder | |
| def audio_feature_extract( | |
| self, | |
| audios: Union[np.ndarray, List[np.ndarray], List[List[np.ndarray]]], | |
| audio_parts: Optional[list] = None, | |
| chunk_input: Optional[bool] = False, | |
| sampling_rate: Optional[int] = None, | |
| chunk_length: Optional[int] = 1, | |
| **kwargs, | |
| ): | |
| if isinstance(audios, np.ndarray): | |
| audios_list = [[audios]] | |
| elif isinstance(audios[0], np.ndarray): | |
| audios_list = [audios] | |
| else: | |
| audios_list = audios | |
| if audio_parts is not None: | |
| assert len(audio_parts) == len(audios_list) | |
| for parts, audios in zip(audio_parts, audios_list): | |
| assert len(parts) == len(audios) | |
| audio_feature_lens_list = [] | |
| audio_ph_list = [] | |
| audio_features_all = [] | |
| # audio placeholder not dependent on audio_parts | |
| for audios in audios_list: | |
| if audios: | |
| audio_ph_list.append([self.get_audio_placeholder(len(a), chunk_input, chunk_length) for a in audios]) | |
| else: | |
| audio_ph_list.append([]) | |
| for idx, audios in enumerate(audios_list): | |
| if audio_parts is not None: | |
| # same audio part merge | |
| audio_part = audio_parts[idx] | |
| merge_audio = [] | |
| cur_audio = [] | |
| for aid, (part, audio) in enumerate(zip(audio_part, audios)): | |
| if aid == 0 or audio_part[aid] == audio_part[aid - 1]: | |
| cur_audio.append(audio) | |
| else: | |
| merge_audio.append(np.hstack(cur_audio)) | |
| cur_audio = [audio] | |
| if cur_audio: | |
| merge_audio.append(np.hstack(cur_audio)) | |
| else: | |
| merge_audio = audios | |
| audio_feature_lens = [] | |
| # If the audio exceeds 30 seconds, split it into chunks every 30 seconds. | |
| final_merge_audio = [] | |
| max_audio_inp_len = 30 * sampling_rate | |
| for audio in merge_audio: | |
| if len(audio) <= max_audio_inp_len: | |
| final_merge_audio.append(audio) | |
| else: | |
| for i in range(math.ceil(len(audio) / max_audio_inp_len)): | |
| final_merge_audio.append(audio[i * max_audio_inp_len : (i + 1) * max_audio_inp_len]) | |
| if audios: | |
| audio_inputs = self.feature_extractor( | |
| final_merge_audio, | |
| sampling_rate=sampling_rate, | |
| return_attention_mask=True, | |
| padding="max_length", | |
| return_tensors="pt", | |
| **kwargs, | |
| ) | |
| audio_feature = audio_inputs["input_features"] | |
| actual_lens = audio_inputs["attention_mask"].sum(dim=1) | |
| for feat, lens in zip(audio_feature, actual_lens): | |
| audio_features_all.append(feat[:, :lens]) | |
| audio_feature_lens.append(lens) | |
| audio_feature_lens = torch.hstack(audio_feature_lens) | |
| audio_feature_lens_list.append(audio_feature_lens) | |
| else: | |
| audio_feature_lens_list.append([]) | |
| if audio_features_all: | |
| audio_features = [i.permute(1, 0) for i in audio_features_all] | |
| audio_features = torch.nn.utils.rnn.pad_sequence( | |
| audio_features, batch_first=True, padding_value=0.0 | |
| ).permute(0, 2, 1) | |
| else: | |
| audio_features = [] | |
| return audio_features, audio_feature_lens_list, audio_ph_list | |
| # Copied from transformers.models.clip.processing_clip.CLIPProcessor.batch_decode with CLIP->Llama | |
| def batch_decode(self, *args, **kwargs): | |
| """ | |
| This method forwards all its arguments to LlamaTokenizerFast's [`~PreTrainedTokenizer.batch_decode`]. Please | |
| refer to the docstring of this method for more information. | |
| """ | |
| output_ids = args[0] | |
| result_text = [] | |
| for result in output_ids: | |
| result = result[result != 0] | |
| if result[0] == self.tokenizer.bos_id: | |
| result = result[1:] | |
| if result[-1] == self.tokenizer.eos_id: | |
| result = result[:-1] | |
| result_text.append(self.tokenizer.decode(result, *args[1:], **kwargs).strip()) | |
| return result_text | |
| # return self.tokenizer.batch_decode(*args, **kwargs) | |
| # Copied from transformers.models.clip.processing_clip.CLIPProcessor.decode with CLIP->Llama | |
| def decode(self, *args, **kwargs): | |
| """ | |
| This method forwards all its arguments to LlamaTokenizerFast's [`~PreTrainedTokenizer.decode`]. Please refer to | |
| the docstring of this method for more information. | |
| """ | |
| result = args[0] | |
| result = result[result != 0] | |
| if result[0] == self.tokenizer.bos_id: | |
| result = result[1:] | |
| if result[-1] == self.tokenizer.eos_id or ( | |
| hasattr(self.tokenizer, "eot_id") and result[-1] == self.tokenizer.eot_id | |
| ): | |
| result = result[:-1] | |
| return self.tokenizer.decode(result, *args[1:], **kwargs).strip() | |
| def _convert(self, input_str, max_inp_length: Optional[int] = None, **kwargs): | |
| input_ids = self.tokenizer.encode(input_str, **kwargs) | |
| if max_inp_length is not None: | |
| input_ids = input_ids[:max_inp_length] | |
| input_ids = torch.tensor(input_ids, dtype=torch.int32) | |
| ## image bound | |
| start_cond = (input_ids == self.tokenizer.im_start_id) | (input_ids == self.tokenizer.slice_start_id) | |
| end_cond = (input_ids == self.tokenizer.im_end_id) | (input_ids == self.tokenizer.slice_end_id) | |
| image_start_idx = torch.where(start_cond)[0] | |
| image_start_idx += 1 | |
| image_end_idx = torch.where(end_cond)[0] | |
| valid_image_nums = max(len(image_start_idx), len(image_end_idx)) | |
| image_bounds = torch.hstack( | |
| [ | |
| image_start_idx[:valid_image_nums].unsqueeze(-1), | |
| image_end_idx[:valid_image_nums].unsqueeze(-1), | |
| ] | |
| ) | |
| ## audio bound | |
| audio_start_idx = torch.where(input_ids == self.tokenizer.audio_start_id)[0] | |
| audio_end_idx = torch.where(input_ids == self.tokenizer.audio_end_id)[0] | |
| assert len(audio_start_idx) == len(audio_end_idx) | |
| audio_bounds = torch.hstack([(audio_start_idx + 1).unsqueeze(-1), audio_end_idx.unsqueeze(-1)]) | |
| spk_start_idx = torch.where(input_ids == self.tokenizer.spk_start_id)[0] | |
| spk_end_idx = torch.where(input_ids == self.tokenizer.spk_end_id)[0] | |
| assert len(spk_start_idx) == len(spk_end_idx) | |
| spk_bounds = torch.hstack([(spk_start_idx + 1).unsqueeze(-1), spk_end_idx.unsqueeze(-1)]) | |
| return input_ids, image_bounds, audio_bounds, spk_bounds | |
| def _convert_omni_to_inputs( | |
| self, | |
| images, | |
| audio_phs, | |
| texts: Union[str, List[str]], | |
| truncation=None, | |
| max_length=None, | |
| max_slice_nums=None, | |
| use_image_id=None, | |
| return_tensors=None, | |
| **kwargs, | |
| ): | |
| if images is None and audio_phs is None: | |
| model_inputs = self.tokenizer( | |
| texts, return_tensors=return_tensors, truncation=truncation, max_length=max_length, **kwargs | |
| ) | |
| return MiniCPMOBatchFeature(data={**model_inputs}) | |
| image_tag = "(<image>./</image>)" | |
| image_pattern = "\(<image>./</image>\)" | |
| audio_tag = "(<audio>./</audio>)" | |
| audio_pattern = "\(<audio>./</audio>\)" | |
| split_pattern = f"({image_pattern}|{audio_pattern})" | |
| if isinstance(texts, str): | |
| texts = [texts] | |
| bs = len(texts) | |
| if images is not None: | |
| images, image_sizes, tgt_sizes = images["pixel_values"], images["image_sizes"], images["tgt_sizes"] | |
| else: | |
| images, image_sizes, tgt_sizes = [[]] * bs, [[]] * bs, [[]] * bs | |
| input_ids_list = [] | |
| image_bounds_list = [] | |
| audio_bounds_list = [] | |
| spk_bounds_list = [] | |
| for index, text in enumerate(texts): | |
| text_chunks = re.split(split_pattern, text) | |
| image_tags = re.findall(image_pattern, text) | |
| audio_tags = re.findall(audio_pattern, text) | |
| if image_tags: | |
| assert images is not None | |
| assert len(image_tags) == len(image_sizes[index]) | |
| if audio_tags: | |
| assert audio_phs is not None | |
| assert len(audio_tags) == len(audio_phs[index]) | |
| image_id = 0 | |
| audio_id = 0 | |
| for i, chunk in enumerate(text_chunks): | |
| if chunk == image_tag: | |
| image_placeholder = self.image_processor.get_slice_image_placeholder( | |
| image_sizes[index][image_id], image_id, max_slice_nums, use_image_id | |
| ) | |
| image_id += 1 | |
| text_chunks[i] = image_placeholder | |
| elif chunk == audio_tag: | |
| audio_placeholder = audio_phs[index][audio_id] | |
| audio_id += 1 | |
| text_chunks[i] = audio_placeholder | |
| final_text = "".join(text_chunks) | |
| input_ids, image_bounds, audio_bounds, spk_bounds = self._convert(final_text, max_length, **kwargs) | |
| input_ids_list.append(input_ids) | |
| image_bounds_list.append(image_bounds) | |
| audio_bounds_list.append(audio_bounds) | |
| spk_bounds_list.append(spk_bounds) | |
| padded_input_ids, padding_lengths = self.pad(input_ids_list, padding_side="left") | |
| attention_mask = torch.ones_like(padded_input_ids, dtype=torch.bool) | |
| for i, length in enumerate(padding_lengths): | |
| image_bounds_list[i] = image_bounds_list[i] + length | |
| audio_bounds_list[i] = audio_bounds_list[i] + length | |
| spk_bounds_list[i] = spk_bounds_list[i] + length | |
| attention_mask[i, :length] = False | |
| data = { | |
| "input_ids": padded_input_ids, | |
| "attention_mask": attention_mask, | |
| "pixel_values": images, | |
| "image_sizes": image_sizes, | |
| "image_bound": image_bounds_list, | |
| "tgt_sizes": tgt_sizes, | |
| "audio_bounds": audio_bounds_list, | |
| "spk_bounds": spk_bounds_list, | |
| } | |
| return data | |
| # Copied from transformers.models.clip.processing_clip.CLIPProcessor.model_input_names | |
| def model_input_names(self): | |
| tokenizer_input_names = self.tokenizer.model_input_names | |
| image_processor_input_names = self.image_processor.model_input_names | |
| feature_extractor_input_names = self.feature_extractor.model_input_names | |
| return list(dict.fromkeys(tokenizer_input_names + image_processor_input_names + feature_extractor_input_names)) | |
| def pad(self, inputs, max_length=None, padding_value=0, padding_side="left"): | |
| items = [] | |
| if isinstance(inputs[0], list): | |
| assert isinstance(inputs[0][0], torch.Tensor) | |
| for it in inputs: | |
| for tr in it: | |
| items.append(tr) | |
| else: | |
| assert isinstance(inputs[0], torch.Tensor) | |
| items = inputs | |
| batch_size = len(items) | |
| shape = items[0].shape | |
| dim = len(shape) | |
| assert dim <= 2 | |
| if max_length is None: | |
| max_length = 0 | |
| max_length = max(max_length, max(item.shape[-1] for item in items)) | |
| min_length = min(item.shape[-1] for item in items) | |
| dtype = items[0].dtype | |
| if dim == 0: | |
| return torch.stack([item for item in items], dim=0), [0] | |
| elif dim == 1: | |
| if max_length == min_length: | |
| return torch.stack([item for item in items], dim=0), [0] * batch_size | |
| tensor = torch.zeros((batch_size, max_length), dtype=dtype) + padding_value | |
| else: | |
| tensor = torch.zeros((batch_size, max_length, shape[-1]), dtype=dtype) + padding_value | |
| padding_length = [] | |
| for i, item in enumerate(items): | |
| if dim == 1: | |
| if padding_side == "left": | |
| tensor[i, -len(item) :] = item.clone() | |
| else: | |
| tensor[i, : len(item)] = item.clone() | |
| elif dim == 2: | |
| if padding_side == "left": | |
| tensor[i, -len(item) :, :] = item.clone() | |
| else: | |
| tensor[i, : len(item), :] = item.clone() | |
| padding_length.append(tensor.shape[-1] - len(item)) | |
| return tensor, padding_length | |
| class MelSpectrogramFeatures(torch.nn.Module): | |
| def __init__( | |
| self, | |
| sample_rate=24000, | |
| n_fft=1024, | |
| hop_length=256, | |
| n_mels=100, | |
| padding: Literal["center", "same"] = "center", | |
| ): | |
| super().__init__() | |
| if padding not in ["center", "same"]: | |
| raise ValueError("Padding must be 'center' or 'same'.") | |
| self.padding = padding | |
| self.mel_spec = torchaudio.transforms.MelSpectrogram( | |
| sample_rate=sample_rate, | |
| n_fft=n_fft, | |
| hop_length=hop_length, | |
| n_mels=n_mels, | |
| center=padding == "center", | |
| power=1, | |
| ) | |
| def __call__(self, audio: torch.Tensor) -> torch.Tensor: | |
| """ | |
| audio: Tensor([num_channels, num_samples]) | |
| """ | |
| return super().__call__(audio) | |
| def forward(self, audio: torch.Tensor) -> torch.Tensor: | |
| """ | |
| audio: Tensor([num_channels, num_samples]) | |
| """ | |
| mel: torch.Tensor = self.mel_spec(audio) | |
| features = torch.log(torch.clip(mel, min=1e-5)) | |
| return features | |
| class ChatTTSProcessor: | |
| def __init__(self, text_tokenizer): | |
| self.audio_processor = MelSpectrogramFeatures() | |
| self.text_tokenizer = text_tokenizer | |
| def __call__(self, text_list, audio_list): | |
| assert len(text_list) == len(audio_list) | |
| input_ids_varlen = [] | |
| for text in text_list: | |
| input_ids_ = self.text_tokenizer.encode(text, return_tensors="pt", add_special_tokens=False) # [1, seq_len] | |
| input_ids_ = input_ids_.squeeze(0) # [seq_len] | |
| input_ids_varlen.append(input_ids_) | |
| audio_features_varlen = [] | |
| for audio in audio_list: | |
| assert audio.shape.__len__() == 1 # [seq_len] | |
| try: | |
| mel = self.audio_processor(audio) # [100(num_mel_bins), seq_len_mel] | |
| except Exception as e: | |
| raise e | |
| audio_features_varlen.append(mel) | |
| return { | |
| "tts_input_ids_varlen": input_ids_varlen, # return List[Tensor] | |
| "tts_input_features_varlen": audio_features_varlen, # return List[Tensor] | |
| } | |