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| | """ |
| | 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 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, |
| | ): |
| | def get_audio_placeholder(audio_lens, chunk_input): |
| | 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 |
| |
|
| | 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 = [] |
| |
|
| | |
| | for audios in audios_list: |
| | if audios: |
| | audio_ph_list.append([get_audio_placeholder(len(a), chunk_input) for a in audios]) |
| | else: |
| | audio_ph_list.append([]) |
| |
|
| | for idx, audios in enumerate(audios_list): |
| | if audio_parts is not None: |
| | |
| | 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 = [] |
| |
|
| | |
| | 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 |
| |
|
| | |
| | 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 |
| | |
| |
|
| | |
| | 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) |
| |
|
| | |
| | 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_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 |
| |
|
| | @property |
| | |
| | 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) |
| | input_ids_ = input_ids_.squeeze(0) |
| | input_ids_varlen.append(input_ids_) |
| |
|
| | audio_features_varlen = [] |
| | for audio in audio_list: |
| | assert audio.shape.__len__() == 1 |
| | try: |
| | mel = self.audio_processor(audio) |
| | except Exception as e: |
| | raise e |
| | audio_features_varlen.append(mel) |
| |
|
| | return { |
| | "tts_input_ids_varlen": input_ids_varlen, |
| | "tts_input_features_varlen": audio_features_varlen, |
| | } |
| |
|