# SPDX-License-Identifier: Apache-2.0 # SPDX-FileCopyrightText: Copyright contributors to the vLLM project # 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 from typing import Literal, TypeAlias import numpy as np import regex import torch import torchaudio from transformers.image_processing_utils import BatchFeature from transformers.image_utils import ImageInput from transformers.processing_utils import ProcessorMixin from transformers.tokenization_utils_base import PreTokenizedInput, TextInput from transformers.utils import TensorType MiniCPMOBatchFeature: TypeAlias = BatchFeature 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, pool_step=2, ): super().__init__(image_processor, feature_extractor, tokenizer) # Mirror the MiniCPMVProcessor guard: newer (transformers v5.7+) # MiniCPM image processors may drop the legacy `version` attribute, # so fall back to None instead of hard-crashing. `version` only # special-cases the 2.5 tokenization path; other values take the # default branch. self.version = getattr(image_processor, "version", None) self.pool_step = pool_step def _safe_get_token_id(self, attr_name, default_token_str): """Get token ID safely, with fallback to default.""" val = getattr(self.tokenizer, attr_name, None) if val is None: val = self.tokenizer.convert_tokens_to_ids(default_token_str) if val is None: return -1 return val def _safe_get_token_str(self, attr_name, default_token_str): """Get token string safely, with fallback to default.""" return getattr(self.tokenizer, attr_name, default_token_str) def __call__( self, text: TextInput | PreTokenizedInput | list[TextInput] | list[PreTokenizedInput], images: ImageInput = None, audios: np.ndarray | list[np.ndarray] | list[list[np.ndarray]] = None, audio_parts: list | None = None, max_length: int | None = None, do_pad: bool | None = True, max_slice_nums: int | None = None, use_image_id: bool = True, chunk_input: bool = False, return_tensors: str | TensorType | None = TensorType.PYTORCH, sampling_rate: int | None = 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 = self.pool_step 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 audio_start = getattr(self.tokenizer, "audio_start", "") 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 += audio_start + "" * unk_len + audio_end total_unk_len += unk_len audio_placeholder = place_holders else: audio_placeholder = audio_start + "" * output_lens + audio_end return audio_placeholder def audio_feature_extract( self, audios: np.ndarray | list[np.ndarray] | list[list[np.ndarray]], audio_parts: list | None = None, chunk_input: bool | None = False, sampling_rate: int | None = None, chunk_length: int | None = 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 or 16000) 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 [`~PythonBackend.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 len(result) > 0 and result[0] == self.tokenizer.bos_id: result = result[1:] if len(result) > 0 and result[-1] == self.tokenizer.eos_id: result = result[:-1] result_text.append( self.tokenizer.decode(result, *args[1:], **kwargs).strip() ) return result_text # 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 [`~PythonBackend.decode`]. Please refer to the docstring of this method for more information. """ result = args[0] result = result[result != 0] if len(result) > 0 and result[0] == self.tokenizer.bos_id: result = result[1:] if len(result) > 0 and ( 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: int | None = 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] assert len(image_start_idx) == len(image_end_idx), ( f"The number of image start tokens ({len(image_start_idx)}) " f"and end tokens ({len(image_end_idx)}) must match." ) image_bounds = torch.hstack( [ image_start_idx.unsqueeze(-1), image_end_idx.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: 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_pattern = r"\(./\)" audio_tag = "()" audio_pattern = r"\(\)" split_pattern = rf"({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 = regex.split(split_pattern, text) image_tags = regex.findall(image_pattern, text) audio_tags = regex.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 @property # 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", ): if not inputs: return torch.empty(0), [] 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] }