
Fix `TypeError: Phi4MMProcessor.init() got an unexpected keyword argument 'chat_template'`
0cb22ab
verified
# Copyright 2024 Microsoft and the HuggingFace Inc. 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 Phi4MM | |
""" | |
import re | |
from typing import List, Optional, Tuple, Union | |
import math | |
from enum import Enum | |
import numpy as np | |
import scipy | |
import torch | |
import torchvision | |
from transformers import AutoFeatureExtractor, AutoImageProcessor | |
from transformers.feature_extraction_sequence_utils import SequenceFeatureExtractor | |
from transformers.image_processing_utils import BaseImageProcessor, BatchFeature | |
from transformers.image_utils import ( | |
ImageInput, | |
make_list_of_images, | |
valid_images, | |
) | |
from transformers.processing_utils import ProcessorMixin | |
from transformers.tokenization_utils_base import PaddingStrategy, TextInput, TruncationStrategy | |
from transformers.utils import TensorType, logging | |
from torch.nn.utils.rnn import pad_sequence | |
logger = logging.get_logger(__name__) | |
# Special tokens | |
_COMPATIBLE_IMAGE_SPECIAL_TOKEN_PATTERN = r'<\|image_\d+\|>' # For backward compatibility | |
_COMPATIBLE_AUDIO_SPECIAL_TOKEN_PATTERN = r'<\|audio_\d+\|>' # For backward compatibility | |
_IMAGE_SPECIAL_TOKEN = '<|endoftext10|>' | |
_AUDIO_SPECIAL_TOKEN = '<|endoftext11|>' | |
_IMAGE_SPECIAL_TOKEN_ID = 200010 # '<|endoftext10|>', or we can better name it (in `tokenizer_config.json`) | |
_AUDIO_SPECIAL_TOKEN_ID = 200011 # '<|endoftext11|>' | |
class InputMode(Enum): | |
LANGUAGE = 0 | |
VISION = 1 | |
SPEECH = 2 | |
VISION_SPEECH = 3 | |
class Phi4MMImageProcessor(BaseImageProcessor): | |
r""" | |
Constructs a Phi4MM image processor. | |
""" | |
model_input_names = ["input_image_embeds", "image_sizes", "image_attention_mask"] | |
def __init__( | |
self, | |
dynamic_hd, | |
**kwargs, | |
) -> None: | |
super().__init__(**kwargs) | |
self.dynamic_hd = dynamic_hd | |
def find_closest_aspect_ratio(self, aspect_ratio, target_ratios, width, height, image_size): | |
best_ratio_diff = float('inf') | |
best_ratio = (1, 1) | |
area = width * height | |
for ratio in target_ratios: | |
target_aspect_ratio = ratio[0] / ratio[1] | |
ratio_diff = abs(aspect_ratio - target_aspect_ratio) | |
if ratio_diff < best_ratio_diff: | |
best_ratio_diff = ratio_diff | |
best_ratio = ratio | |
elif ratio_diff == best_ratio_diff: | |
if area > 0.5 * image_size * image_size * ratio[0] * ratio[1]: | |
best_ratio = ratio | |
return best_ratio | |
def dynamic_preprocess(self, image, min_num=1, max_num=12, image_size=384, mask_size=27, use_thumbnail=True): | |
orig_width, orig_height = image.size | |
w_crop_num = math.ceil(orig_width/float(image_size)) | |
h_crop_num = math.ceil(orig_height/float(image_size)) | |
if w_crop_num * h_crop_num > max_num: | |
aspect_ratio = orig_width / orig_height | |
# calculate the existing image aspect ratio | |
target_ratios = set( | |
(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 | |
i * j <= max_num and i * j >= min_num) | |
target_ratios = sorted(target_ratios, key=lambda x: x[0] * x[1]) | |
# find the closest aspect ratio to the target | |
target_aspect_ratio = self.find_closest_aspect_ratio( | |
aspect_ratio, target_ratios, orig_width, orig_height, image_size) | |
# calculate the target width and height | |
target_width = image_size * target_aspect_ratio[0] | |
target_height = image_size * target_aspect_ratio[1] | |
else: | |
target_width = image_size * w_crop_num | |
target_height = image_size * h_crop_num | |
target_aspect_ratio = (w_crop_num, h_crop_num) | |
# Calculate the ratio | |
ratio_width = target_width / orig_width | |
ratio_height = target_height / orig_height | |
if ratio_width < ratio_height: | |
new_size = (target_width, int(orig_height * ratio_width)) | |
padding_width = 0 | |
padding_height = target_height - int(orig_height * ratio_width) | |
else: | |
new_size = (int(orig_width * ratio_height), target_height) | |
padding_width = target_width - int(orig_width * ratio_height) | |
padding_height = 0 | |
attention_mask = torch.ones((int(mask_size*target_aspect_ratio[1]), int(mask_size*target_aspect_ratio[0]))) | |
if padding_width >= 14: | |
attention_mask[:, -math.floor(padding_width/14):] = 0 | |
if padding_height >= 14: | |
attention_mask[-math.floor(padding_height/14):,:] = 0 | |
assert attention_mask.sum() > 0 | |
if min(new_size[1], target_height) < 10 or min(new_size[0], target_width) < 10: | |
raise ValueError(f'the aspect ratio is very extreme {new_size}') | |
image = torchvision.transforms.functional.resize(image, [new_size[1], new_size[0]],) | |
resized_img = torchvision.transforms.functional.pad(image, [0, 0, padding_width, padding_height], fill=[255,255,255]) | |
return resized_img, attention_mask | |
def pad_to_max_num_crops(self, images, max_crops=5): | |
""" | |
images: B x 3 x H x W, B<=max_crops | |
""" | |
B, _, H, W = images.shape | |
if B < max_crops: | |
pad = torch.zeros(max_crops - B, 3, H, W, dtype=images.dtype, device=images.device) | |
images = torch.cat([images, pad], dim=0) | |
return images | |
def pad_mask_to_max_num_crops(self, masks, max_crops=5): | |
B, H, W = masks.shape | |
if B < max_crops: | |
pad = torch.ones(max_crops - B, H, W, dtype=masks.dtype, device=masks.device) | |
masks = torch.cat([masks, pad], dim=0) | |
return masks | |
def preprocess( | |
self, | |
images: ImageInput, | |
return_tensors: Optional[Union[str, TensorType]] = None, | |
): | |
""" | |
Args: | |
images (`ImageInput`): | |
Image to preprocess. Expects a single or batch of images with pixel values ranging from 0 to 255. If | |
passing in images with pixel values between 0 and 1, set `do_rescale=False`. | |
return_tensors (`str` or `TensorType`, *optional*): | |
The type of tensors to return. Can be one of: | |
- Unset: Return a list of `np.ndarray`. | |
- `TensorType.TENSORFLOW` or `'tf'`: Return a batch of type `tf.Tensor`. | |
- `TensorType.PYTORCH` or `'pt'`: Return a batch of type `torch.Tensor`. | |
- `TensorType.NUMPY` or `'np'`: Return a batch of type `np.ndarray`. | |
- `TensorType.JAX` or `'jax'`: Return a batch of type `jax.numpy.ndarray`. | |
""" | |
images = make_list_of_images(images) | |
if not valid_images(images): | |
raise ValueError( | |
"Invalid image type. Must be of type PIL.Image.Image, numpy.ndarray, " | |
"torch.Tensor, tf.Tensor or jax.ndarray." | |
) | |
# Basic settings. | |
img_processor = torchvision.transforms.Compose([ | |
torchvision.transforms.ToTensor(), | |
torchvision.transforms.Normalize( | |
(0.5, 0.5, 0.5), | |
(0.5, 0.5, 0.5) | |
), | |
]) | |
dyhd_base_resolution = 448 | |
# Dynamic HD | |
base_resolution = dyhd_base_resolution | |
images = [image.convert('RGB') for image in images] | |
# cover 384 and 448 resolution | |
mask_resolution = base_resolution // 14 | |
elems, image_attention_masks = [], [] | |
for im in images: | |
elem, attention_mask = self.dynamic_preprocess(im, max_num=self.dynamic_hd, image_size=base_resolution, mask_size=mask_resolution) | |
elems.append(elem) | |
image_attention_masks.append(attention_mask) | |
hd_images = [img_processor(im) for im in elems] | |
global_image = [torch.nn.functional.interpolate(im.unsqueeze(0).float(), size=(base_resolution, base_resolution), mode='bicubic',).to(im.dtype) for im in hd_images] | |
shapes = [[im.size(1), im.size(2)] for im in hd_images] | |
mask_shapes = [[mask.size(0), mask.size(1)] for mask in image_attention_masks] | |
global_attention_mask = [torch.ones((1, mask_resolution, mask_resolution)) for _ in hd_images] | |
hd_images_reshape = [im.reshape(1, 3, | |
h//base_resolution, | |
base_resolution, | |
w//base_resolution, | |
base_resolution | |
).permute(0,2,4,1,3,5).reshape(-1, 3, base_resolution, base_resolution).contiguous() for im, (h, w) in zip(hd_images, shapes)] | |
attention_masks_reshape = [mask.reshape(1, | |
h//mask_resolution, | |
mask_resolution, | |
w//mask_resolution, | |
mask_resolution | |
).permute(0,1,3,2,4).reshape(-1, mask_resolution, mask_resolution).contiguous() for mask, (h, w) in zip(image_attention_masks, mask_shapes)] | |
downsample_attention_masks = [mask[:,0::2,0::2].reshape(1, | |
h//mask_resolution, | |
w//mask_resolution, | |
mask_resolution//2+mask_resolution%2, | |
mask_resolution//2+mask_resolution%2 | |
).permute(0,1,3,2,4) for mask, (h,w) in zip(attention_masks_reshape, mask_shapes)] | |
downsample_attention_masks = [mask.reshape(mask.size(1)*mask.size(2), mask.size(3)*mask.size(4))for mask in downsample_attention_masks] | |
num_img_tokens = [256 + 1 + int(mask.sum().item()) + int(mask[:,0].sum().item()) + 16 for mask in downsample_attention_masks] | |
hd_images_reshape = [torch.cat([_global_image] + [_im], dim=0) for _global_image, _im in zip(global_image, hd_images_reshape)] | |
hd_masks_reshape = [torch.cat([_global_mask] + [_mask], dim=0) for _global_mask, _mask in zip(global_attention_mask, attention_masks_reshape)] | |
max_crops = max([img.size(0) for img in hd_images_reshape]) | |
image_transformed = [self.pad_to_max_num_crops(im, max_crops) for im in hd_images_reshape] | |
image_transformed = torch.stack(image_transformed, dim=0) | |
mask_transformed = [self.pad_mask_to_max_num_crops(mask, max_crops) for mask in hd_masks_reshape] | |
mask_transformed = torch.stack(mask_transformed, dim=0) | |
returned_input_image_embeds = image_transformed | |
returned_image_sizes = torch.tensor(shapes, dtype=torch.long) | |
returned_image_attention_mask = mask_transformed | |
returned_num_img_tokens = num_img_tokens | |
data = { | |
"input_image_embeds": returned_input_image_embeds, | |
"image_sizes": returned_image_sizes, | |
"image_attention_mask": returned_image_attention_mask, | |
"num_img_tokens": returned_num_img_tokens, | |
} | |
return BatchFeature(data=data, tensor_type=return_tensors) | |
AudioInput = Tuple[Union[np.ndarray, torch.Tensor], int] | |
AudioInputs = List[AudioInput] | |
def speechlib_mel(sample_rate, n_fft, n_mels, fmin=None, fmax=None): | |
"""Create a Mel filter-bank the same as SpeechLib FbankFC. | |
Args: | |
sample_rate (int): Sample rate in Hz. number > 0 [scalar] | |
n_fft (int): FFT size. int > 0 [scalar] | |
n_mel (int): Mel filter size. int > 0 [scalar] | |
fmin (float): lowest frequency (in Hz). If None use 0.0. | |
float >= 0 [scalar] | |
fmax: highest frequency (in Hz). If None use sample_rate / 2. | |
float >= 0 [scalar] | |
Returns | |
out (numpy.ndarray): Mel transform matrix | |
[shape=(n_mels, 1 + n_fft/2)] | |
""" | |
bank_width = int(n_fft // 2 + 1) | |
if fmax is None: | |
fmax = sample_rate / 2 | |
if fmin is None: | |
fmin = 0 | |
assert fmin >= 0, "fmin cannot be negtive" | |
assert fmin < fmax <= sample_rate / 2, "fmax must be between (fmin, samplerate / 2]" | |
def mel(f): | |
return 1127.0 * np.log(1.0 + f / 700.0) | |
def bin2mel(fft_bin): | |
return 1127.0 * np.log(1.0 + fft_bin * sample_rate / (n_fft * 700.0)) | |
def f2bin(f): | |
return int((f * n_fft / sample_rate) + 0.5) | |
# Spec 1: FFT bin range [f2bin(fmin) + 1, f2bin(fmax) - 1] | |
klo = f2bin(fmin) + 1 | |
khi = f2bin(fmax) | |
khi = max(khi, klo) | |
# Spec 2: SpeechLib uses trianges in Mel space | |
mlo = mel(fmin) | |
mhi = mel(fmax) | |
m_centers = np.linspace(mlo, mhi, n_mels + 2) | |
ms = (mhi - mlo) / (n_mels + 1) | |
matrix = np.zeros((n_mels, bank_width), dtype=np.float32) | |
for m in range(0, n_mels): | |
left = m_centers[m] | |
center = m_centers[m + 1] | |
right = m_centers[m + 2] | |
for fft_bin in range(klo, khi): | |
mbin = bin2mel(fft_bin) | |
if left < mbin < right: | |
matrix[m, fft_bin] = 1.0 - abs(center - mbin) / ms | |
return matrix | |
class Phi4MMAudioFeatureExtractor(SequenceFeatureExtractor): | |
model_input_names = ["input_audio_embeds", "audio_embed_sizes", "audio_attention_mask"] | |
def __init__(self, audio_compression_rate, audio_downsample_rate, audio_feat_stride, **kwargs): | |
feature_size = 80 | |
sampling_rate = 16000 | |
padding_value = 0.0 | |
super().__init__(feature_size, sampling_rate, padding_value, **kwargs) | |
self.compression_rate = audio_compression_rate | |
self.qformer_compression_rate = audio_downsample_rate | |
self.feat_stride = audio_feat_stride | |
self._eightk_method = "fillzero" | |
self._mel = speechlib_mel(16000, 512, 80, fmin=None, fmax=7690).T | |
self._hamming400 = np.hamming(400) # for 16k audio | |
self._hamming200 = np.hamming(200) # for 8k audio | |
def duration_to_frames(self, duration): | |
"""duration in s, estimated frames""" | |
frame_rate = 10 | |
num_frames = duration * 1000 // frame_rate | |
return num_frames | |
def __call__( | |
self, | |
audios: List[AudioInput], | |
return_tensors: Optional[Union[str, TensorType]] = None, | |
): | |
# Ref: https://github.com/huggingface/transformers/blob/v4.47.0/src/transformers/models/audio_spectrogram_transformer/feature_extraction_audio_spectrogram_transformer.py#L161 | |
returned_input_audio_embeds = [] | |
returned_audio_embed_sizes = [] | |
audio_frames_list = [] | |
for audio_data, sample_rate in audios: | |
audio_embeds = self._extract_features(audio_data, sample_rate) | |
audio_frames = len(audio_embeds) * self.feat_stride | |
audio_embed_size = self._compute_audio_embed_size(audio_frames) | |
returned_input_audio_embeds.append(torch.tensor(audio_embeds)) | |
returned_audio_embed_sizes.append(torch.tensor(audio_embed_size).long()) | |
audio_frames_list.append(audio_frames) | |
returned_input_audio_embeds = pad_sequence( | |
returned_input_audio_embeds, batch_first=True | |
) | |
returned_audio_embed_sizes = torch.stack(returned_audio_embed_sizes, dim=0) | |
audio_frames = torch.tensor(audio_frames_list) | |
returned_audio_attention_mask = torch.arange(0, audio_frames.max()).unsqueeze(0) < audio_frames.unsqueeze(1) if len(audios) > 1 else None | |
data = { | |
"input_audio_embeds": returned_input_audio_embeds, | |
"audio_embed_sizes": returned_audio_embed_sizes, | |
} | |
if returned_audio_attention_mask is not None: | |
data["audio_attention_mask"] = returned_audio_attention_mask | |
return BatchFeature(data=data, tensor_type=return_tensors) | |
def _extract_spectrogram(self, wav, fs): | |
"""Extract spectrogram features from waveform. | |
Args: | |
wav (1D array): waveform of the input | |
fs (int): sampling rate of the waveform, 16000 or 8000. | |
If fs=8000, the waveform will be resampled to 16000Hz. | |
Output: | |
log_fbank (2D array): a TxD matrix of log Mel filterbank features. | |
D=80, and T is the number of frames. | |
""" | |
if wav.ndim > 1: | |
wav = np.squeeze(wav) | |
# by default, we extract the mean if stereo | |
if len(wav.shape) == 2: | |
wav = wav.mean(1) | |
# Resample to 16000 or 8000 if needed | |
if fs > 16000: | |
wav = scipy.signal.resample_poly(wav, 1, fs // 16000) | |
fs = 16000 | |
elif 8000 < fs < 16000: | |
wav = scipy.signal.resample_poly(wav, 1, fs // 8000) | |
fs = 8000 | |
elif fs < 8000: | |
raise RuntimeError(f"Unsupported sample rate {fs}") | |
if fs == 8000: | |
if self._eightk_method == "resample": | |
# Input audio is 8 kHz. Convert to 16 kHz before feature | |
# extraction | |
wav = scipy.signal.resample_poly(wav, 2, 1) | |
fs = 16000 | |
# Do nothing here for fillzero method | |
elif fs != 16000: | |
# Input audio is not a supported sample rate. | |
raise RuntimeError(f"Input data using an unsupported sample rate: {fs}") | |
preemphasis = 0.97 | |
if fs == 8000: | |
n_fft = 256 | |
win_length = 200 | |
hop_length = 80 | |
fft_window = self._hamming200 | |
elif fs == 16000: | |
n_fft = 512 | |
win_length = 400 | |
hop_length = 160 | |
fft_window = self._hamming400 | |
# Spec 1: SpeechLib cut remaining sample insufficient for a hop | |
n_batch = (wav.shape[0] - win_length) // hop_length + 1 | |
# Here we don't use stride_tricks since the input array may not satisfy | |
# memory layout requirement and we need writeable output | |
# Here we only use list of views before copy to desination | |
# so it is more efficient than broadcasting | |
y_frames = np.array( | |
[wav[_stride : _stride + win_length] for _stride in range(0, hop_length * n_batch, hop_length)], | |
dtype=np.float32, | |
) | |
# Spec 2: SpeechLib applies preemphasis within each batch | |
y_frames_prev = np.roll(y_frames, 1, axis=1) | |
y_frames_prev[:, 0] = y_frames_prev[:, 1] | |
y_frames = (y_frames - preemphasis * y_frames_prev) * 32768 | |
S = np.fft.rfft(fft_window * y_frames, n=n_fft, axis=1).astype(np.complex64) | |
if fs == 8000: | |
# Need to pad the output to look like 16 kHz data but with zeros in | |
# the 4 to 8 kHz bins. | |
frames, bins = S.shape | |
padarray = np.zeros((frames, bins)) | |
S = np.concatenate((S[:, 0:-1], padarray), axis=1) # Nyquist bin gets set to zero | |
spec = np.abs(S).astype(np.float32) | |
return spec | |
def _extract_features(self, wav, fs): | |
"""Extract log filterbank features from waveform. | |
Args: | |
wav (1D array): waveform of the input | |
fs (int): sampling rate of the waveform, 16000 or 8000. | |
If fs=8000, the waveform will be resampled to 16000Hz. | |
Output: | |
log_fbank (2D array): a TxD matrix of log Mel filterbank features. | |
D=80, and T is the number of frames. | |
""" | |
spec = self._extract_spectrogram(wav, fs) | |
spec_power = spec**2 | |
fbank_power = np.clip(spec_power.dot(self._mel), 1.0, None) | |
log_fbank = np.log(fbank_power).astype(np.float32) | |
return log_fbank | |
def _compute_audio_embed_size(self, audio_frames): | |
integer = audio_frames // self.compression_rate | |
remainder = audio_frames % self.compression_rate | |
result = integer if remainder == 0 else integer + 1 | |
integer = result // self.qformer_compression_rate | |
remainder = result % self.qformer_compression_rate | |
result = integer if remainder == 0 else integer + 1 # qformer compression | |
return result | |
class Phi4MMProcessor(ProcessorMixin): | |
r""" | |
Constructs a Phi4MM processor which raps an image processor, a audio processor, and a GPT tokenizer into a single processor. | |
[`Phi4MMProcessor`] offers all the functionalities of [`Phi4MMImageProcessor`] and [`GPT2Tokenizer`]. See the | |
[`~Phi4MMProcessor.__call__`] and [`~Phi4MMProcessor.decode`] for more information. | |
Args: | |
image_processor ([`Phi4MMImageProcessor`], *optional*): | |
The image processor is a required input. | |
tokenizer ([`GPT2Tokenizer`], *optional*): | |
The tokenizer is a required input. | |
""" | |
attributes = ["image_processor", "audio_processor", "tokenizer"] | |
tokenizer_class = "GPT2TokenizerFast" | |
image_processor_class = "AutoImageProcessor" # Phi4MMImageProcessor will be registered later | |
audio_processor_class = "AutoFeatureExtractor" # Phi4MMAudioFeatureExtractor will be registered later | |
def __init__(self, image_processor, audio_processor, tokenizer, **kwargs): | |
self.image_processor = image_processor | |
self.audio_processor = audio_processor | |
self.tokenizer = tokenizer | |
def __call__( | |
self, | |
text: Union[TextInput, List[TextInput]], | |
images: Optional[ImageInput] = None, | |
audios: Optional[AudioInputs] = None, | |
padding: Union[bool, str, PaddingStrategy] = False, | |
truncation: Optional[Union[bool, str, TruncationStrategy]] = None, | |
max_length=None, | |
return_tensors: Optional[Union[str, TensorType]] = TensorType.PYTORCH, | |
) -> BatchFeature: | |
""" | |
Main method to prepare for the model one or several sequences(s) and image(s). This method forards the `text` | |
and `kwargs` arguments to GPT2Tokenizer's [`~GPT2Tokenizer.__call__`] if `text` is not `None` to encode | |
the text. To prepare the image(s), this method forwards the `images` and `kwrags` arguments to | |
Phi4MMImageProcessor's [`~Phi4MMImageProcessor.__call__`] if `images` is not `None`. Please refer to the doctsring | |
of the above two methods for more information. | |
Args: | |
text (`str`, `List[str]`, `List[List[str]]`): | |
The sequence or batch of sequences to be encoded. Each sequence can be a string or a list of strings | |
(pretokenized string). If the sequences are provided as list of strings (pretokenized), you must set | |
`is_split_into_words=True` (to lift the ambiguity with a batch of sequences). | |
images (`PIL.Image.Image`, `np.ndarray`, `torch.Tensor`, `List[PIL.Image.Image]`, `List[np.ndarray]`, `List[torch.Tensor]`): | |
The image or batch of images to be prepared. Each image can be a PIL image, NumPy array or PyTorch | |
tensor. Both channels-first and channels-last formats are supported. | |
padding (`bool`, `str` or [`~utils.PaddingStrategy`], *optional*, defaults to `False`): | |
Select a strategy to pad the returned sequences (according to the model's padding side and padding | |
index) among: | |
- `True` or `'longest'`: Pad to the longest sequence in the batch (or no padding if only a single | |
sequence if provided). | |
- `'max_length'`: Pad to a maximum length specified with the argument `max_length` or to the maximum | |
acceptable input length for the model if that argument is not provided. | |
- `False` or `'do_not_pad'` (default): No padding (i.e., can output a batch with sequences of different | |
lengths). | |
max_length (`int`, *optional*): | |
Maximum length of the returned list and optionally padding length (see above). | |
truncation (`bool`, *optional*): | |
Activates truncation to cut input sequences longer than `max_length` to `max_length`. | |
return_tensors (`str` or [`~utils.TensorType`], *optional*): | |
If set, will return tensors of a particular framework. Acceptable values are: | |
- `'tf'`: Return TensorFlow `tf.constant` objects. | |
- `'pt'`: Return PyTorch `torch.Tensor` objects. | |
- `'np'`: Return NumPy `np.ndarray` objects. | |
- `'jax'`: Return JAX `jnp.ndarray` objects. | |
Returns: | |
[`BatchFeature`]: A [`BatchFeature`] with the following fields: | |
- **input_ids** -- List of token ids to be fed to a model. | |
- **input_image_embeds** -- Pixel values to be fed to a model. | |
- **image_sizes** -- List of tuples specifying the size of each image in `input_image_embeds`. | |
- **image_attention_mask** -- List of attention masks for each image in `input_image_embeds`. | |
- **input_audio_embeds** -- Audio embeddings to be fed to a model. | |
- **audio_embed_sizes** -- List of integers specifying the size of each audio in `input_audio_embeds`. | |
- **attention_mask** -- List of indices specifying which tokens should be attended to by the model. | |
""" | |
image_inputs = self.image_processor(images, return_tensors=return_tensors) if images is not None else {} | |
audio_inputs = self.audio_processor(audios, return_tensors=return_tensors) if audios is not None else {} | |
inputs = self._convert_images_audios_text_to_inputs( | |
image_inputs, | |
audio_inputs, | |
text, | |
padding=padding, | |
truncation=truncation, | |
max_length=max_length, | |
return_tensors=return_tensors, | |
) | |
# idenfity the input mode | |
if len(image_inputs) > 0 and len(audio_inputs) > 0: | |
input_mode = InputMode.VISION_SPEECH | |
elif len(image_inputs) > 0: | |
input_mode = InputMode.VISION | |
elif len(audio_inputs) > 0: | |
input_mode = InputMode.SPEECH | |
else: | |
input_mode = InputMode.LANGUAGE | |
inputs["input_mode"] = torch.tensor([input_mode.value], dtype=torch.long) | |
return inputs | |
def special_image_token_id(self): | |
return self.tokenizer.convert_tokens_to_ids(self.special_image_token) | |
def get_special_image_token_id(self): | |
return self.tokenizer.convert_tokens_to_ids(self.special_image_token) | |
def chat_template(self): | |
return self.tokenizer.chat_template | |
def _convert_images_audios_text_to_inputs( | |
self, images, audios, text, padding=False, truncation=None, max_length=None, return_tensors=None | |
): | |
# prepare image id to image input ids | |
if len(images) > 0: | |
input_image_embeds = images["input_image_embeds"] | |
image_sizes = images["image_sizes"] | |
image_attention_mask = images["image_attention_mask"] | |
num_img_tokens = images['num_img_tokens'] | |
else: | |
input_image_embeds = torch.tensor([]) | |
image_sizes = torch.tensor([]) | |
image_attention_mask = torch.tensor([]) | |
num_img_tokens = [] | |
# prepare audio id to audio input ids | |
if len(audios) > 0: | |
input_audio_embeds = audios["input_audio_embeds"] | |
audio_embed_sizes = audios["audio_embed_sizes"] | |
audio_attention_mask = audios.get("audio_attention_mask", None) | |
else: | |
input_audio_embeds = torch.tensor([]) | |
audio_embed_sizes = torch.tensor([]) | |
audio_attention_mask = None | |
# Replace certain special tokens for compatibility | |
# Ref: https://stackoverflow.com/questions/11475885/python-replace-regex | |
if isinstance(text, str): | |
text = [text] | |
assert isinstance(text, list) | |
processed_text = [re.sub(_COMPATIBLE_IMAGE_SPECIAL_TOKEN_PATTERN, _IMAGE_SPECIAL_TOKEN, t) for t in text] | |
processed_text = [re.sub(_COMPATIBLE_AUDIO_SPECIAL_TOKEN_PATTERN, _AUDIO_SPECIAL_TOKEN, t) for t in processed_text] | |
input_ids_list = [self.tokenizer(t).input_ids for t in processed_text] | |
img_cnt, audio_cnt = 0, 0 # only needed for later assertion | |
image_token_count_iter = iter(num_img_tokens) | |
audio_embed_size_iter = iter(audio_embed_sizes.tolist()) | |
new_input_ids_list = [] | |
for input_ids in input_ids_list: | |
i = 0 | |
while i < len(input_ids): | |
token_id = input_ids[i] | |
if token_id == _AUDIO_SPECIAL_TOKEN_ID: | |
token_count = next(audio_embed_size_iter) | |
audio_cnt += 1 | |
elif token_id == _IMAGE_SPECIAL_TOKEN_ID: | |
token_count = next(image_token_count_iter) | |
img_cnt += 1 | |
else: | |
i += 1 | |
continue | |
tokens = [token_id] * token_count | |
input_ids = input_ids[:i] + tokens + input_ids[i + 1:] | |
i += token_count | |
input_ids = torch.tensor(input_ids, dtype=torch.long) | |
new_input_ids_list.append(input_ids) | |
lengths = torch.tensor([len(input_ids) for input_ids in new_input_ids_list]) | |
max_len = lengths.max() | |
input_ids = input_ids.new_full((len(new_input_ids_list), max_len), self.tokenizer.pad_token_id) | |
# batched inference requires left padding | |
for i in range(len(new_input_ids_list)): | |
input_ids[i, max_len - len(new_input_ids_list[i]):] = new_input_ids_list[i] | |
# If the below assertion fails, it might be that input pure-text | |
# messages contain image/audio special tokens literally | |
# (<|endoftext10|>, <|endoftext11|>). | |
assert ( | |
img_cnt == len(num_img_tokens) | |
), ( | |
f"Number of image tokens in prompt_token_ids ({img_cnt}) " | |
f"does not match number of images ({len(num_img_tokens)})" | |
) | |
assert ( | |
audio_cnt == len(audio_embed_sizes) | |
), ( | |
f"Number of audio tokens in prompt_token_ids ({audio_cnt}) " | |
f"does not match number of audios ({len(audio_embed_sizes)})" | |
) | |
# prepare attention mask | |
seq_range = torch.arange(max_len - 1, -1, -1) | |
attention_mask = seq_range.unsqueeze(0) < lengths.unsqueeze(1) | |
# prepare batch feature | |
data = { | |
"input_ids": input_ids, | |
"input_image_embeds": input_image_embeds, | |
"image_sizes": image_sizes, | |
"image_attention_mask": image_attention_mask, | |
"input_audio_embeds": input_audio_embeds, | |
"audio_embed_sizes": audio_embed_sizes, | |
"audio_attention_mask": audio_attention_mask, | |
"attention_mask": attention_mask, | |
} | |
return BatchFeature( | |
data=data | |
) | |
# 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 GPT2Tokenizer's [`~PreTrainedTokenizer.batch_decode`]. Please | |
refer to the docstring of this method for more information. | |
""" | |
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 GPT2Tokenizer's [`~PreTrainedTokenizer.decode`]. Please refer to | |
the docstring of this method for more information. | |
""" | |
return self.tokenizer.decode(*args, **kwargs) | |
# 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 | |
audio_processor_input_names = self.audio_processor.model_input_names | |
return list(dict.fromkeys(tokenizer_input_names + image_processor_input_names + audio_processor_input_names)) | |
AutoImageProcessor.register("Phi4MMImageProcessor", Phi4MMImageProcessor) | |
AutoFeatureExtractor.register("Phi4MMAudioFeatureExtractor", Phi4MMAudioFeatureExtractor) | |