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FNet Model with a `next sentence prediction (classification)` head on top.
This model is a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) sub-class. Use
it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage and
behavior.
Parameters:
config ([`FNetConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Methods: forward | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/fnet.md | https://huggingface.co/docs/transformers/en/model_doc/fnet/#fnetfornextsentenceprediction | #fnetfornextsentenceprediction | .md | 165_10 |
FNet Model transformer with a sequence classification/regression head on top (a linear layer on top of the pooled
output) e.g. for GLUE tasks.
This model is a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) sub-class. Use
it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage and
behavior.
Parameters:
config ([`FNetConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Methods: forward | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/fnet.md | https://huggingface.co/docs/transformers/en/model_doc/fnet/#fnetforsequenceclassification | #fnetforsequenceclassification | .md | 165_11 |
FNet Model with a multiple choice classification head on top (a linear layer on top of the pooled output and a
softmax) e.g. for RocStories/SWAG tasks.
This model is a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) sub-class. Use
it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage and
behavior.
Parameters:
config ([`FNetConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Methods: forward | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/fnet.md | https://huggingface.co/docs/transformers/en/model_doc/fnet/#fnetformultiplechoice | #fnetformultiplechoice | .md | 165_12 |
FNet Model with a token classification head on top (a linear layer on top of the hidden-states output) e.g. for
Named-Entity-Recognition (NER) tasks.
This model is a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) sub-class. Use
it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage and
behavior.
Parameters:
config ([`FNetConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Methods: forward | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/fnet.md | https://huggingface.co/docs/transformers/en/model_doc/fnet/#fnetfortokenclassification | #fnetfortokenclassification | .md | 165_13 |
FNet Model with a span classification head on top for extractive question-answering tasks like SQuAD (a linear
layers on top of the hidden-states output to compute `span start logits` and `span end logits`).
This model is a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) sub-class. Use
it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage and
behavior.
Parameters:
config ([`FNetConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Methods: forward | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/fnet.md | https://huggingface.co/docs/transformers/en/model_doc/fnet/#fnetforquestionanswering | #fnetforquestionanswering | .md | 165_14 |
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--> | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/unispeech.md | https://huggingface.co/docs/transformers/en/model_doc/unispeech/ | .md | 166_0 |
|
The UniSpeech model was proposed in [UniSpeech: Unified Speech Representation Learning with Labeled and Unlabeled Data](https://arxiv.org/abs/2101.07597) by Chengyi Wang, Yu Wu, Yao Qian, Kenichi Kumatani, Shujie Liu, Furu Wei, Michael
Zeng, Xuedong Huang .
The abstract from the paper is the following:
*In this paper, we propose a unified pre-training approach called UniSpeech to learn speech representations with both
unlabeled and labeled data, in which supervised phonetic CTC learning and phonetically-aware contrastive
self-supervised learning are conducted in a multi-task learning manner. The resultant representations can capture
information more correlated with phonetic structures and improve the generalization across languages and domains. We
evaluate the effectiveness of UniSpeech for cross-lingual representation learning on public CommonVoice corpus. The
results show that UniSpeech outperforms self-supervised pretraining and supervised transfer learning for speech
recognition by a maximum of 13.4% and 17.8% relative phone error rate reductions respectively (averaged over all
testing languages). The transferability of UniSpeech is also demonstrated on a domain-shift speech recognition task,
i.e., a relative word error rate reduction of 6% against the previous approach.*
This model was contributed by [patrickvonplaten](https://huggingface.co/patrickvonplaten). The Authors' code can be
found [here](https://github.com/microsoft/UniSpeech/tree/main/UniSpeech). | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/unispeech.md | https://huggingface.co/docs/transformers/en/model_doc/unispeech/#overview | #overview | .md | 166_1 |
- UniSpeech is a speech model that accepts a float array corresponding to the raw waveform of the speech signal. Please
use [`Wav2Vec2Processor`] for the feature extraction.
- UniSpeech model can be fine-tuned using connectionist temporal classification (CTC) so the model output has to be
decoded using [`Wav2Vec2CTCTokenizer`]. | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/unispeech.md | https://huggingface.co/docs/transformers/en/model_doc/unispeech/#usage-tips | #usage-tips | .md | 166_2 |
- [Audio classification task guide](../tasks/audio_classification)
- [Automatic speech recognition task guide](../tasks/asr) | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/unispeech.md | https://huggingface.co/docs/transformers/en/model_doc/unispeech/#resources | #resources | .md | 166_3 |
This is the configuration class to store the configuration of a [`UniSpeechModel`]. It is used to instantiate an
UniSpeech model according to the specified arguments, defining the model architecture. Instantiating a
configuration with the defaults will yield a similar configuration to that of the UniSpeech
[microsoft/unispeech-large-1500h-cv](https://huggingface.co/microsoft/unispeech-large-1500h-cv) architecture.
Configuration objects inherit from [`PretrainedConfig`] and can be used to control the model outputs. Read the
documentation from [`PretrainedConfig`] for more information.
Args:
vocab_size (`int`, *optional*, defaults to 32):
Vocabulary size of the UniSpeech model. Defines the number of different tokens that can be represented by
the `inputs_ids` passed when calling [`UniSpeechModel`]. Vocabulary size of the model. Defines the
different tokens that can be represented by the *inputs_ids* passed to the forward method of
[`UniSpeechModel`].
hidden_size (`int`, *optional*, defaults to 768):
Dimensionality of the encoder layers and the pooler layer.
num_hidden_layers (`int`, *optional*, defaults to 12):
Number of hidden layers in the Transformer encoder.
num_attention_heads (`int`, *optional*, defaults to 12):
Number of attention heads for each attention layer in the Transformer encoder.
intermediate_size (`int`, *optional*, defaults to 3072):
Dimensionality of the "intermediate" (i.e., feed-forward) layer in the Transformer encoder.
hidden_act (`str` or `function`, *optional*, defaults to `"gelu"`):
The non-linear activation function (function or string) in the encoder and pooler. If string, `"gelu"`,
`"relu"`, `"selu"` and `"gelu_new"` are supported.
hidden_dropout (`float`, *optional*, defaults to 0.1):
The dropout probability for all fully connected layers in the embeddings, encoder, and pooler.
activation_dropout (`float`, *optional*, defaults to 0.1):
The dropout ratio for activations inside the fully connected layer.
attention_dropout (`float`, *optional*, defaults to 0.1):
The dropout ratio for the attention probabilities.
feat_proj_dropout (`float`, *optional*, defaults to 0.0):
The dropout probability for output of the feature encoder.
feat_quantizer_dropout (`float`, *optional*, defaults to 0.0):
The dropout probability for the output of the feature encoder that's used by the quantizer.
final_dropout (`float`, *optional*, defaults to 0.1):
The dropout probability for the final projection layer of [`UniSpeechForCTC`].
layerdrop (`float`, *optional*, defaults to 0.1):
The LayerDrop probability. See the [LayerDrop paper](see https://arxiv.org/abs/1909.11556) for more
details.
initializer_range (`float`, *optional*, defaults to 0.02):
The standard deviation of the truncated_normal_initializer for initializing all weight matrices.
layer_norm_eps (`float`, *optional*, defaults to 1e-05):
The epsilon used by the layer normalization layers.
feat_extract_norm (`str`, *optional*, defaults to `"group"`):
The norm to be applied to 1D convolutional layers in feature encoder. One of `"group"` for group
normalization of only the first 1D convolutional layer or `"layer"` for layer normalization of all 1D
convolutional layers.
feat_extract_activation (`str, *optional*, defaults to `"gelu"`):
The non-linear activation function (function or string) in the 1D convolutional layers of the feature
extractor. If string, `"gelu"`, `"relu"`, `"selu"` and `"gelu_new"` are supported.
conv_dim (`Tuple[int]` or `List[int]`, *optional*, defaults to `(512, 512, 512, 512, 512, 512, 512)`):
A tuple of integers defining the number of input and output channels of each 1D convolutional layer in the
feature encoder. The length of *conv_dim* defines the number of 1D convolutional layers.
conv_stride (`Tuple[int]` or `List[int]`, *optional*, defaults to `(5, 2, 2, 2, 2, 2, 2)`):
A tuple of integers defining the stride of each 1D convolutional layer in the feature encoder. The length
of *conv_stride* defines the number of convolutional layers and has to match the length of *conv_dim*.
conv_kernel (`Tuple[int]` or `List[int]`, *optional*, defaults to `(10, 3, 3, 3, 3, 2, 2)`):
A tuple of integers defining the kernel size of each 1D convolutional layer in the feature encoder. The
length of *conv_kernel* defines the number of convolutional layers and has to match the length of
*conv_dim*.
conv_bias (`bool`, *optional*, defaults to `False`):
Whether the 1D convolutional layers have a bias.
num_conv_pos_embeddings (`int`, *optional*, defaults to 128):
Number of convolutional positional embeddings. Defines the kernel size of 1D convolutional positional
embeddings layer.
num_conv_pos_embedding_groups (`int`, *optional*, defaults to 16):
Number of groups of 1D convolutional positional embeddings layer.
do_stable_layer_norm (`bool`, *optional*, defaults to `False`):
Whether to apply *stable* layer norm architecture of the Transformer encoder. `do_stable_layer_norm is
True` corresponds to applying layer norm before the attention layer, whereas `do_stable_layer_norm is
False` corresponds to applying layer norm after the attention layer.
apply_spec_augment (`bool`, *optional*, defaults to `True`):
Whether to apply *SpecAugment* data augmentation to the outputs of the feature encoder. For reference see
[SpecAugment: A Simple Data Augmentation Method for Automatic Speech
Recognition](https://arxiv.org/abs/1904.08779).
mask_time_prob (`float`, *optional*, defaults to 0.05):
Percentage (between 0 and 1) of all feature vectors along the time axis which will be masked. The masking
procecure generates ''mask_time_prob*len(time_axis)/mask_time_length'' independent masks over the axis. If
reasoning from the propability of each feature vector to be chosen as the start of the vector span to be
masked, *mask_time_prob* should be `prob_vector_start*mask_time_length`. Note that overlap may decrease the
actual percentage of masked vectors. This is only relevant if `apply_spec_augment is True`.
mask_time_length (`int`, *optional*, defaults to 10):
Length of vector span along the time axis.
mask_time_min_masks (`int`, *optional*, defaults to 2):
The minimum number of masks of length `mask_feature_length` generated along the time axis, each time step,
irrespectively of `mask_feature_prob`. Only relevant if ''mask_time_prob*len(time_axis)/mask_time_length <
mask_time_min_masks''
mask_feature_prob (`float`, *optional*, defaults to 0.0):
Percentage (between 0 and 1) of all feature vectors along the feature axis which will be masked. The
masking procecure generates ''mask_feature_prob*len(feature_axis)/mask_time_length'' independent masks over
the axis. If reasoning from the propability of each feature vector to be chosen as the start of the vector
span to be masked, *mask_feature_prob* should be `prob_vector_start*mask_feature_length`. Note that overlap
may decrease the actual percentage of masked vectors. This is only relevant if `apply_spec_augment is
True`.
mask_feature_length (`int`, *optional*, defaults to 10):
Length of vector span along the feature axis.
mask_feature_min_masks (`int`, *optional*, defaults to 0):
The minimum number of masks of length `mask_feature_length` generated along the feature axis, each time
step, irrespectively of `mask_feature_prob`. Only relevant if
''mask_feature_prob*len(feature_axis)/mask_feature_length < mask_feature_min_masks''
num_codevectors_per_group (`int`, *optional*, defaults to 320):
Number of entries in each quantization codebook (group).
num_codevector_groups (`int`, *optional*, defaults to 2):
Number of codevector groups for product codevector quantization.
contrastive_logits_temperature (`float`, *optional*, defaults to 0.1):
The temperature *kappa* in the contrastive loss.
num_negatives (`int`, *optional*, defaults to 100):
Number of negative samples for the contrastive loss.
codevector_dim (`int`, *optional*, defaults to 256):
Dimensionality of the quantized feature vectors.
proj_codevector_dim (`int`, *optional*, defaults to 256):
Dimensionality of the final projection of both the quantized and the transformer features.
diversity_loss_weight (`int`, *optional*, defaults to 0.1):
The weight of the codebook diversity loss component.
ctc_loss_reduction (`str`, *optional*, defaults to `"mean"`):
Specifies the reduction to apply to the output of `torch.nn.CTCLoss`. Only relevant when training an
instance of [`UniSpeechForCTC`].
ctc_zero_infinity (`bool`, *optional*, defaults to `False`):
Whether to zero infinite losses and the associated gradients of `torch.nn.CTCLoss`. Infinite losses mainly
occur when the inputs are too short to be aligned to the targets. Only relevant when training an instance
of [`UniSpeechForCTC`].
use_weighted_layer_sum (`bool`, *optional*, defaults to `False`):
Whether to use a weighted average of layer outputs with learned weights. Only relevant when using an
instance of [`UniSpeechForSequenceClassification`].
classifier_proj_size (`int`, *optional*, defaults to 256):
Dimensionality of the projection before token mean-pooling for classification.
num_ctc_classes (`int`, *optional*, defaults to 80):
Specifies the number of classes (phoneme tokens and blank token) for phoneme-level CTC loss. Only relevant
when using an instance of [`UniSpeechForPreTraining`].
pad_token_id (`int`, *optional*, defaults to 0):
The id of the padding token.
bos_token_id (`int`, *optional*, defaults to 1):
The id of the "beginning-of-sequence" token.
eos_token_id (`int`, *optional*, defaults to 2):
The id of the "end-of-sequence" token.
replace_prob (`float`, *optional*, defaults to 0.5):
Propability that transformer feature is replaced by quantized feature for pretraining.
Example:
```python
>>> from transformers import UniSpeechConfig, UniSpeechModel
>>> # Initializing a UniSpeech facebook/unispeech-base-960h style configuration
>>> configuration = UniSpeechConfig()
>>> # Initializing a model (with random weights) from the facebook/unispeech-base-960h style configuration
>>> model = UniSpeechModel(configuration)
>>> # Accessing the model configuration
>>> configuration = model.config
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/unispeech.md | https://huggingface.co/docs/transformers/en/model_doc/unispeech/#unispeechconfig | #unispeechconfig | .md | 166_4 |
models.unispeech.modeling_unispeech.UniSpeechForPreTrainingOutput
Output type of [`UniSpeechForPreTrainingOutput`], with potential hidden states and attentions.
Args:
loss (*optional*, returned when model is in train mode, `torch.FloatTensor` of shape `(1,)`):
Total loss as the sum of the contrastive loss (L_m) and the diversity loss (L_d) as stated in the [official
paper](https://arxiv.org/pdf/2006.11477.pdf) . (classification) loss.
projected_states (`torch.FloatTensor` of shape `(batch_size, sequence_length, config.proj_codevector_dim)`):
Hidden-states of the model projected to *config.proj_codevector_dim* that can be used to predict the masked
projected quantized states.
projected_quantized_states (`torch.FloatTensor` of shape `(batch_size, sequence_length, config.proj_codevector_dim)`):
Quantized extracted feature vectors projected to *config.proj_codevector_dim* representing the positive
target vectors for contrastive loss.
hidden_states (`tuple(torch.FloatTensor)`, *optional*, returned when `output_hidden_states=True` is passed or when `config.output_hidden_states=True`):
Tuple of `torch.FloatTensor` (one for the output of the embeddings + one for the output of each layer) of
shape `(batch_size, sequence_length, hidden_size)`.
Hidden-states of the model at the output of each layer plus the initial embedding outputs.
attentions (`tuple(torch.FloatTensor)`, *optional*, returned when `output_attentions=True` is passed or when `config.output_attentions=True`):
Tuple of `torch.FloatTensor` (one for each layer) of shape `(batch_size, num_heads, sequence_length,
sequence_length)`.
Attentions weights after the attention softmax, used to compute the weighted average in the self-attention
heads. | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/unispeech.md | https://huggingface.co/docs/transformers/en/model_doc/unispeech/#unispeech-specific-outputs | #unispeech-specific-outputs | .md | 166_5 |
The bare UniSpeech Model transformer outputting raw hidden-states without any specific head on top.
UniSpeech was proposed in [UniSpeech: Unified Speech Representation Learning with Labeled and Unlabeled
Data](https://arxiv.org/abs/2101.07597) by Chengyi Wang, Yu Wu, Yao Qian, Kenichi Kumatani, Shujie Liu, Furu Wei,
Michael Zeng, Xuedong Huang.
This model inherits from [`PreTrainedModel`]. Check the superclass documentation for the generic methods the
library implements for all its model (such as downloading or saving etc.).
This model is a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) sub-class. Use
it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage and
behavior.
Parameters:
config ([`UniSpeechConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Methods: forward | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/unispeech.md | https://huggingface.co/docs/transformers/en/model_doc/unispeech/#unispeechmodel | #unispeechmodel | .md | 166_6 |
UniSpeech Model with a `language modeling` head on top for Connectionist Temporal Classification (CTC).
UniSpeech was proposed in [UniSpeech: Unified Speech Representation Learning with Labeled and Unlabeled
Data](https://arxiv.org/abs/2101.07597) by Chengyi Wang, Yu Wu, Yao Qian, Kenichi Kumatani, Shujie Liu, Furu Wei,
Michael Zeng, Xuedong Huang.
This model inherits from [`PreTrainedModel`]. Check the superclass documentation for the generic methods the
library implements for all its model (such as downloading or saving etc.).
This model is a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) sub-class. Use
it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage and
behavior.
Parameters:
config ([`UniSpeechConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
target_lang (`str`, *optional*):
Language id of adapter weights. Adapter weights are stored in the format adapter.<lang>.safetensors or
adapter.<lang>.bin. Only relevant when using an instance of [`UniSpeechForCTC`] with adapters. Uses 'eng'
by default.
Methods: forward | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/unispeech.md | https://huggingface.co/docs/transformers/en/model_doc/unispeech/#unispeechforctc | #unispeechforctc | .md | 166_7 |
UniSpeech Model with a sequence classification head on top (a linear layer over the pooled output) for tasks like
SUPERB Keyword Spotting.
UniSpeech was proposed in [UniSpeech: Unified Speech Representation Learning with Labeled and Unlabeled
Data](https://arxiv.org/abs/2101.07597) by Chengyi Wang, Yu Wu, Yao Qian, Kenichi Kumatani, Shujie Liu, Furu Wei,
Michael Zeng, Xuedong Huang.
This model inherits from [`PreTrainedModel`]. Check the superclass documentation for the generic methods the
library implements for all its model (such as downloading or saving etc.).
This model is a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) sub-class. Use
it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage and
behavior.
Parameters:
config ([`UniSpeechConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Methods: forward | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/unispeech.md | https://huggingface.co/docs/transformers/en/model_doc/unispeech/#unispeechforsequenceclassification | #unispeechforsequenceclassification | .md | 166_8 |
UniSpeech Model with a vector-quantization module and ctc loss for pre-training.
UniSpeech was proposed in [UniSpeech: Unified Speech Representation Learning with Labeled and Unlabeled
Data](https://arxiv.org/abs/2101.07597) by Chengyi Wang, Yu Wu, Yao Qian, Kenichi Kumatani, Shujie Liu, Furu Wei,
Michael Zeng, Xuedong Huang.
This model inherits from [`PreTrainedModel`]. Check the superclass documentation for the generic methods the
library implements for all its model (such as downloading or saving etc.).
This model is a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) sub-class. Use
it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage and
behavior.
Parameters:
config ([`UniSpeechConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Methods: forward | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/unispeech.md | https://huggingface.co/docs/transformers/en/model_doc/unispeech/#unispeechforpretraining | #unispeechforpretraining | .md | 166_9 |
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--> | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/cpmant.md | https://huggingface.co/docs/transformers/en/model_doc/cpmant/ | .md | 167_0 |
|
CPM-Ant is an open-source Chinese pre-trained language model (PLM) with 10B parameters. It is also the first milestone of the live training process of CPM-Live. The training process is cost-effective and environment-friendly. CPM-Ant also achieves promising results with delta tuning on the CUGE benchmark. Besides the full model, we also provide various compressed versions to meet the requirements of different hardware configurations. [See more](https://github.com/OpenBMB/CPM-Live/tree/cpm-ant/cpm-live)
This model was contributed by [OpenBMB](https://huggingface.co/openbmb). The original code can be found [here](https://github.com/OpenBMB/CPM-Live/tree/cpm-ant/cpm-live). | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/cpmant.md | https://huggingface.co/docs/transformers/en/model_doc/cpmant/#overview | #overview | .md | 167_1 |
- A tutorial on [CPM-Live](https://github.com/OpenBMB/CPM-Live/tree/cpm-ant/cpm-live). | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/cpmant.md | https://huggingface.co/docs/transformers/en/model_doc/cpmant/#resources | #resources | .md | 167_2 |
This is the configuration class to store the configuration of a [`CpmAntModel`]. It is used to instantiate an
CPMAnt model according to the specified arguments, defining the model architecture. Instantiating a configuration
with the defaults will yield a similar configuration to that of the CPMAnt
[openbmb/cpm-ant-10b](https://huggingface.co/openbmb/cpm-ant-10b) architecture.
Configuration objects inherit from [`PretrainedConfig`] and can be used to control the model outputs. Read the
documentation from [`PretrainedConfig`] for more information.
Args:
vocab_size (`int`, *optional*, defaults to 30720):
Vocabulary size of the CPMAnt model. Defines the number of different tokens that can be represented by the
`input` passed when calling [`CpmAntModel`].
hidden_size (`int`, *optional*, defaults to 4096):
Dimension of the encoder layers.
num_attention_heads (`int`, *optional*, defaults to 32):
Number of attention heads in the Transformer encoder.
dim_head (`int`, *optional*, defaults to 128):
Dimension of attention heads for each attention layer in the Transformer encoder.
dim_ff (`int`, *optional*, defaults to 10240):
Dimension of the "intermediate" (i.e., feed-forward) layer in the Transformer encoder.
num_hidden_layers (`int`, *optional*, defaults to 48):
Number of layers of the Transformer encoder.
dropout_p (`float`, *optional*, defaults to 0.0):
The dropout probability for all fully connected layers in the embeddings, encoder.
position_bias_num_buckets (`int`, *optional*, defaults to 512):
The number of position_bias buckets.
position_bias_max_distance (`int`, *optional*, defaults to 2048):
The maximum sequence length that this model might ever be used with. Typically set this to something large
just in case (e.g., 512 or 1024 or 2048).
eps (`float`, *optional*, defaults to 1e-06):
The epsilon used by the layer normalization layers.
init_std (`float`, *optional*, defaults to 1.0):
Initialize parameters with std = init_std.
prompt_types (`int`, *optional*, defaults to 32):
The type of prompt.
prompt_length (`int`, *optional*, defaults to 32):
The length of prompt.
segment_types (`int`, *optional*, defaults to 32):
The type of segment.
use_cache (`bool`, *optional*, defaults to `True`):
Whether to use cache.
Example:
```python
>>> from transformers import CpmAntModel, CpmAntConfig
>>> # Initializing a CPMAnt cpm-ant-10b style configuration
>>> configuration = CpmAntConfig()
>>> # Initializing a model from the cpm-ant-10b style configuration
>>> model = CpmAntModel(configuration)
>>> # Accessing the model configuration
>>> configuration = model.config
```
Methods: all | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/cpmant.md | https://huggingface.co/docs/transformers/en/model_doc/cpmant/#cpmantconfig | #cpmantconfig | .md | 167_3 |
Construct a CPMAnt tokenizer. Based on byte-level Byte-Pair-Encoding.
Args:
vocab_file (`str`):
Path to the vocabulary file.
bod_token (`str`, *optional*, defaults to `"<d>"`):
The beginning of document token.
eod_token (`str`, *optional*, defaults to `"</d>"`):
The end of document token.
bos_token (`str`, *optional*, defaults to `"<s>"`):
The beginning of sequence token.
eos_token (`str`, *optional*, defaults to `"</s>"`):
The end of sequence token.
pad_token (`str`, *optional*, defaults to `"<pad>"`):
The token used for padding.
unk_token (`str`, *optional*, defaults to `"<unk>"`):
The unknown token.
line_token (`str`, *optional*, defaults to `"</n>"`):
The line token.
space_token (`str`, *optional*, defaults to `"</_>"`):
The space token.
Methods: all | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/cpmant.md | https://huggingface.co/docs/transformers/en/model_doc/cpmant/#cpmanttokenizer | #cpmanttokenizer | .md | 167_4 |
The bare CPMAnt Model outputting raw hidden-states without any specific head on top.
This model is a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) sub-class. Use
it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage and
behavior.
Parameters
config ([`~CpmAntConfig`]): Model configuration class with all the parameters of the
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Methods: all | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/cpmant.md | https://huggingface.co/docs/transformers/en/model_doc/cpmant/#cpmantmodel | #cpmantmodel | .md | 167_5 |
The CPMAnt Model with a language modeling head on top (linear layer with weights tied to the input embeddings).
This model is a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) sub-class. Use
it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage and
behavior.
Parameters
config ([`~CpmAntConfig`]): Model configuration class with all the parameters of the
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Methods: all | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/cpmant.md | https://huggingface.co/docs/transformers/en/model_doc/cpmant/#cpmantforcausallm | #cpmantforcausallm | .md | 167_6 |
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|
The [`SpeechEncoderDecoderModel`] can be used to initialize a speech-to-text model
with any pretrained speech autoencoding model as the encoder (*e.g.* [Wav2Vec2](wav2vec2), [Hubert](hubert)) and any pretrained autoregressive model as the decoder.
The effectiveness of initializing speech-sequence-to-text-sequence models with pretrained checkpoints for speech
recognition and speech translation has *e.g.* been shown in [Large-Scale Self- and Semi-Supervised Learning for Speech
Translation](https://arxiv.org/abs/2104.06678) by Changhan Wang, Anne Wu, Juan Pino, Alexei Baevski, Michael Auli,
Alexis Conneau.
An example of how to use a [`SpeechEncoderDecoderModel`] for inference can be seen in [Speech2Text2](speech_to_text_2). | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/speech-encoder-decoder.md | https://huggingface.co/docs/transformers/en/model_doc/speech-encoder-decoder/#speech-encoder-decoder-models | #speech-encoder-decoder-models | .md | 168_1 |
[`SpeechEncoderDecoderModel`] can be randomly initialized from an encoder and a decoder config. In the following example, we show how to do this using the default [`Wav2Vec2Model`] configuration for the encoder
and the default [`BertForCausalLM`] configuration for the decoder.
```python
>>> from transformers import BertConfig, Wav2Vec2Config, SpeechEncoderDecoderConfig, SpeechEncoderDecoderModel
>>> config_encoder = Wav2Vec2Config()
>>> config_decoder = BertConfig()
>>> config = SpeechEncoderDecoderConfig.from_encoder_decoder_configs(config_encoder, config_decoder)
>>> model = SpeechEncoderDecoderModel(config=config)
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/speech-encoder-decoder.md | https://huggingface.co/docs/transformers/en/model_doc/speech-encoder-decoder/#randomly-initializing-speechencoderdecodermodel-from-model-configurations | #randomly-initializing-speechencoderdecodermodel-from-model-configurations | .md | 168_2 |
[`SpeechEncoderDecoderModel`] can be initialized from a pretrained encoder checkpoint and a pretrained decoder checkpoint. Note that any pretrained Transformer-based speech model, *e.g.* [Wav2Vec2](wav2vec2), [Hubert](hubert) can serve as the encoder and both pretrained auto-encoding models, *e.g.* BERT, pretrained causal language models, *e.g.* GPT2, as well as the pretrained decoder part of sequence-to-sequence models, *e.g.* decoder of BART, can be used as the decoder.
Depending on which architecture you choose as the decoder, the cross-attention layers might be randomly initialized.
Initializing [`SpeechEncoderDecoderModel`] from a pretrained encoder and decoder checkpoint requires the model to be fine-tuned on a downstream task, as has been shown in [the *Warm-starting-encoder-decoder blog post*](https://huggingface.co/blog/warm-starting-encoder-decoder).
To do so, the `SpeechEncoderDecoderModel` class provides a [`SpeechEncoderDecoderModel.from_encoder_decoder_pretrained`] method.
```python
>>> from transformers import SpeechEncoderDecoderModel
>>> model = SpeechEncoderDecoderModel.from_encoder_decoder_pretrained(
... "facebook/hubert-large-ll60k", "google-bert/bert-base-uncased"
... )
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/speech-encoder-decoder.md | https://huggingface.co/docs/transformers/en/model_doc/speech-encoder-decoder/#initialising-speechencoderdecodermodel-from-a-pretrained-encoder-and-a-pretrained-decoder | #initialising-speechencoderdecodermodel-from-a-pretrained-encoder-and-a-pretrained-decoder | .md | 168_3 |
To load fine-tuned checkpoints of the `SpeechEncoderDecoderModel` class, [`SpeechEncoderDecoderModel`] provides the `from_pretrained(...)` method just like any other model architecture in Transformers.
To perform inference, one uses the [`generate`] method, which allows to autoregressively generate text. This method supports various forms of decoding, such as greedy, beam search and multinomial sampling.
```python
>>> from transformers import Wav2Vec2Processor, SpeechEncoderDecoderModel
>>> from datasets import load_dataset
>>> import torch
>>> # load a fine-tuned speech translation model and corresponding processor
>>> model = SpeechEncoderDecoderModel.from_pretrained("facebook/wav2vec2-xls-r-300m-en-to-15")
>>> processor = Wav2Vec2Processor.from_pretrained("facebook/wav2vec2-xls-r-300m-en-to-15")
>>> # let's perform inference on a piece of English speech (which we'll translate to German)
>>> ds = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
>>> input_values = processor(ds[0]["audio"]["array"], return_tensors="pt").input_values
>>> # autoregressively generate transcription (uses greedy decoding by default)
>>> generated_ids = model.generate(input_values)
>>> generated_text = processor.batch_decode(generated_ids, skip_special_tokens=True)[0]
>>> print(generated_text)
Mr. Quilter ist der Apostel der Mittelschicht und wir freuen uns, sein Evangelium willkommen heißen zu können.
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/speech-encoder-decoder.md | https://huggingface.co/docs/transformers/en/model_doc/speech-encoder-decoder/#loading-an-existing-speechencoderdecodermodel-checkpoint-and-perform-inference | #loading-an-existing-speechencoderdecodermodel-checkpoint-and-perform-inference | .md | 168_4 |
Once the model is created, it can be fine-tuned similar to BART, T5 or any other encoder-decoder model on a dataset of (speech, text) pairs.
As you can see, only 2 inputs are required for the model in order to compute a loss: `input_values` (which are the
speech inputs) and `labels` (which are the `input_ids` of the encoded target sequence).
```python
>>> from transformers import AutoTokenizer, AutoFeatureExtractor, SpeechEncoderDecoderModel
>>> from datasets import load_dataset
>>> encoder_id = "facebook/wav2vec2-base-960h" # acoustic model encoder
>>> decoder_id = "google-bert/bert-base-uncased" # text decoder
>>> feature_extractor = AutoFeatureExtractor.from_pretrained(encoder_id)
>>> tokenizer = AutoTokenizer.from_pretrained(decoder_id)
>>> # Combine pre-trained encoder and pre-trained decoder to form a Seq2Seq model
>>> model = SpeechEncoderDecoderModel.from_encoder_decoder_pretrained(encoder_id, decoder_id)
>>> model.config.decoder_start_token_id = tokenizer.cls_token_id
>>> model.config.pad_token_id = tokenizer.pad_token_id
>>> # load an audio input and pre-process (normalise mean/std to 0/1)
>>> ds = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
>>> input_values = feature_extractor(ds[0]["audio"]["array"], return_tensors="pt").input_values
>>> # load its corresponding transcription and tokenize to generate labels
>>> labels = tokenizer(ds[0]["text"], return_tensors="pt").input_ids
>>> # the forward function automatically creates the correct decoder_input_ids
>>> loss = model(input_values=input_values, labels=labels).loss
>>> loss.backward()
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/speech-encoder-decoder.md | https://huggingface.co/docs/transformers/en/model_doc/speech-encoder-decoder/#training | #training | .md | 168_5 |
[`SpeechEncoderDecoderConfig`] is the configuration class to store the configuration of a
[`SpeechEncoderDecoderModel`]. It is used to instantiate an Encoder Decoder model according to the specified
arguments, defining the encoder and decoder configs.
Configuration objects inherit from [`PretrainedConfig`] and can be used to control the model outputs. Read the
documentation from [`PretrainedConfig`] for more information.
Args:
kwargs (*optional*):
Dictionary of keyword arguments. Notably:
- **encoder** ([`PretrainedConfig`], *optional*) -- An instance of a configuration object that defines
the encoder config.
- **decoder** ([`PretrainedConfig`], *optional*) -- An instance of a configuration object that defines
the decoder config.
Examples:
```python
>>> from transformers import BertConfig, Wav2Vec2Config, SpeechEncoderDecoderConfig, SpeechEncoderDecoderModel
>>> # Initializing a Wav2Vec2 & BERT style configuration
>>> config_encoder = Wav2Vec2Config()
>>> config_decoder = BertConfig()
>>> config = SpeechEncoderDecoderConfig.from_encoder_decoder_configs(config_encoder, config_decoder)
>>> # Initializing a Wav2Vec2Bert model from a Wav2Vec2 & google-bert/bert-base-uncased style configurations
>>> model = SpeechEncoderDecoderModel(config=config)
>>> # Accessing the model configuration
>>> config_encoder = model.config.encoder
>>> config_decoder = model.config.decoder
>>> # set decoder config to causal lm
>>> config_decoder.is_decoder = True
>>> config_decoder.add_cross_attention = True
>>> # Saving the model, including its configuration
>>> model.save_pretrained("my-model")
>>> # loading model and config from pretrained folder
>>> encoder_decoder_config = SpeechEncoderDecoderConfig.from_pretrained("my-model")
>>> model = SpeechEncoderDecoderModel.from_pretrained("my-model", config=encoder_decoder_config)
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/speech-encoder-decoder.md | https://huggingface.co/docs/transformers/en/model_doc/speech-encoder-decoder/#speechencoderdecoderconfig | #speechencoderdecoderconfig | .md | 168_6 |
This class can be used to initialize a speech-sequence-to-text-sequence model with any pretrained speech
autoencoding model as the encoder and any pretrained text autoregressive model as the decoder. The encoder is
loaded via [`~AutoModel.from_pretrained`] function and the decoder is loaded via
[`~AutoModelForCausalLM.from_pretrained`] function. Cross-attention layers are automatically added to the decoder
and should be fine-tuned on a downstream generative task, like summarization.
The effectiveness of initializing sequence-to-sequence models with pretrained checkpoints for sequence generation
tasks was shown in [Leveraging Pre-trained Checkpoints for Sequence Generation
Tasks](https://arxiv.org/abs/1907.12461) by Sascha Rothe, Shashi Narayan, Aliaksei Severyn. Michael Matena, Yanqi
Zhou, Wei Li, Peter J. Liu.
Additionally, in [Large-Scale Self- and Semi-Supervised Learning for Speech
Translation](https://arxiv.org/abs/2104.06678) it is shown how leveraging large pretrained speech models for speech
translation yields a significant performance improvement.
After such an Speech-Encoder Decoder model has been trained/fine-tuned, it can be saved/loaded just like any other
models (see the examples for more information).
This model inherits from [`PreTrainedModel`]. Check the superclass documentation for the generic methods the
library implements for all its model (such as downloading or saving, resizing the input embeddings, pruning heads
etc.)
This model is also a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) subclass.
Use it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage
and behavior.
Parameters:
config ([`SpeechEncoderDecoderConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
[`SpeechEncoderDecoderModel`] is a generic model class that will be instantiated as a transformer architecture with
one of the base model classes of the library as encoder and another one as decoder when created with the
:meth*~transformers.AutoModel.from_pretrained* class method for the encoder and
:meth*~transformers.AutoModelForCausalLM.from_pretrained* class method for the decoder.
Methods: forward
- from_encoder_decoder_pretrained | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/speech-encoder-decoder.md | https://huggingface.co/docs/transformers/en/model_doc/speech-encoder-decoder/#speechencoderdecodermodel | #speechencoderdecodermodel | .md | 168_7 |
No docstring available for FlaxSpeechEncoderDecoderModel
Methods: __call__
- from_encoder_decoder_pretrained | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/speech-encoder-decoder.md | https://huggingface.co/docs/transformers/en/model_doc/speech-encoder-decoder/#flaxspeechencoderdecodermodel | #flaxspeechencoderdecodermodel | .md | 168_8 |
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|
The Mimi model was proposed in [Moshi: a speech-text foundation model for real-time dialogue](https://kyutai.org/Moshi.pdf) by Alexandre Défossez, Laurent Mazaré, Manu Orsini, Amélie Royer, Patrick Pérez, Hervé Jégou, Edouard Grave and Neil Zeghidour. Mimi is a high-fidelity audio codec model developed by the Kyutai team, that combines semantic and acoustic information into audio tokens running at 12Hz and a bitrate of 1.1kbps. In other words, it can be used to map audio waveforms into “audio tokens”, known as “codebooks”.
The abstract from the paper is the following:
*We introduce Moshi, a speech-text foundation model and full-duplex spoken dialogue framework. Current systems for spoken dialogue rely on pipelines of independent components, namely voice activity detection, speech recognition, textual dialogue and text-to-speech. Such frameworks cannot emulate the experience of real conversations. First, their complexity induces a latency of several seconds between interactions. Second, text being the intermediate modality for dialogue, non-linguistic information that modifies meaning— such as emotion or non-speech sounds— is lost in the interaction. Finally, they rely on a segmentation into speaker turns, which does not take into account overlapping speech, interruptions and interjections. Moshi solves these independent issues altogether by casting spoken dialogue as speech-to-speech generation. Starting from a text language model backbone, Moshi generates speech as tokens from the residual quantizer of a neural audio codec, while modeling separately its own speech and that of the user into parallel streams. This allows for the removal of explicit speaker turns, and the modeling of arbitrary conversational dynamics. We moreover extend the hierarchical semantic-to-acoustic token generation of previous work to first predict time-aligned text tokens as a prefix to audio tokens. Not only this “Inner Monologue” method significantly improves the linguistic quality of generated speech, but we also illustrate how it can provide streaming speech recognition and text-to-speech. Our resulting model is the first real-time full-duplex spoken large language model, with a theoretical latency of 160ms, 200ms in practice, and is available at github.com/kyutai-labs/moshi.*
Its architecture is based on [Encodec](model_doc/encodec) with several major differences:
* it uses a much lower frame-rate.
* it uses additional transformers for encoding and decoding for better latent contextualization
* it uses a different quantization scheme: one codebook is dedicated to semantic projection. | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/mimi.md | https://huggingface.co/docs/transformers/en/model_doc/mimi/#overview | #overview | .md | 169_1 |
Here is a quick example of how to encode and decode an audio using this model:
```python
>>> from datasets import load_dataset, Audio
>>> from transformers import MimiModel, AutoFeatureExtractor
>>> librispeech_dummy = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
>>> # load model and feature extractor
>>> model = MimiModel.from_pretrained("kyutai/mimi")
>>> feature_extractor = AutoFeatureExtractor.from_pretrained("kyutai/mimi")
>>> # load audio sample
>>> librispeech_dummy = librispeech_dummy.cast_column("audio", Audio(sampling_rate=feature_extractor.sampling_rate))
>>> audio_sample = librispeech_dummy[-1]["audio"]["array"]
>>> inputs = feature_extractor(raw_audio=audio_sample, sampling_rate=feature_extractor.sampling_rate, return_tensors="pt")
>>> encoder_outputs = model.encode(inputs["input_values"], inputs["padding_mask"])
>>> audio_values = model.decode(encoder_outputs.audio_codes, inputs["padding_mask"])[0]
>>> # or the equivalent with a forward pass
>>> audio_values = model(inputs["input_values"], inputs["padding_mask"]).audio_values
```
This model was contributed by [Yoach Lacombe (ylacombe)](https://huggingface.co/ylacombe).
The original code can be found [here](https://github.com/kyutai-labs/moshi). | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/mimi.md | https://huggingface.co/docs/transformers/en/model_doc/mimi/#usage-example | #usage-example | .md | 169_2 |
This is the configuration class to store the configuration of an [`MimiModel`]. It is used to instantiate a
Mimi model according to the specified arguments, defining the model architecture. Instantiating a configuration
with the defaults will yield a similar configuration to that of the
[kyutai/mimi](https://huggingface.co/kyutai/mimi) architecture.
Configuration objects inherit from [`PretrainedConfig`] and can be used to control the model outputs. Read the
documentation from [`PretrainedConfig`] for more information.
Args:
sampling_rate (`int`, *optional*, defaults to 24000):
The sampling rate at which the audio waveform should be digitalized expressed in hertz (Hz).
frame_rate (`float`, *optional*, defaults to 12.5):
Framerate of the model.
audio_channels (`int`, *optional*, defaults to 1):
Number of channels in the audio data. Either 1 for mono or 2 for stereo.
hidden_size (`int`, *optional*, defaults to 512):
Intermediate representation dimension.
num_filters (`int`, *optional*, defaults to 64):
Number of convolution kernels of first `MimiConv1d` down sampling layer.
num_residual_layers (`int`, *optional*, defaults to 1):
Number of residual layers.
upsampling_ratios (`Sequence[int]`, *optional*):
Kernel size and stride ratios. The encoder uses downsampling ratios instead of upsampling ratios, hence it
will use the ratios in the reverse order to the ones specified here that must match the decoder order.
If not specified, will defaults to `[8, 6, 5, 4]`
kernel_size (`int`, *optional*, defaults to 7):
Kernel size for the initial convolution.
last_kernel_size (`int`, *optional*, defaults to 3):
Kernel size for the last convolution layer.
residual_kernel_size (`int`, *optional*, defaults to 3):
Kernel size for the residual layers.
dilation_growth_rate (`int`, *optional*, defaults to 2):
How much to increase the dilation with each layer.
use_causal_conv (`bool`, *optional*, defaults to `True`):
Whether to use fully causal convolution.
pad_mode (`str`, *optional*, defaults to `"constant"`):
Padding mode for the convolutions.
compress (`int`, *optional*, defaults to 2):
Reduced dimensionality in residual branches.
trim_right_ratio (`float`, *optional*, defaults to 1.0):
Ratio for trimming at the right of the transposed convolution under the `use_causal_conv = True` setup. If
equal to 1.0, it means that all the trimming is done at the right.
codebook_size (`int`, *optional*, defaults to 2048):
Number of discret codes in each codebooks.
codebook_dim (`int`, *optional*, defaults to 256):
Dimension of the unquantized codebook vectors. If not defined, uses `hidden_size`.
num_quantizers (`int`, *optional*, defaults to 32):
Number of quantizer channels, or codebooks, in the quantizer.
use_conv_shortcut (`bool`, *optional*, defaults to `False`):
Whether to use a convolutional layer as the 'skip' connection in the `MimiResnetBlock` block. If False,
an identity function will be used, giving a generic residual connection.
vector_quantization_hidden_dimension (`int`, *optional*, defaults to 256):
Intermediate representation dimension in the residual vector quantization space.
num_semantic_quantizers (`int`, *optional*, defaults to 1):
Number of semantic quantizer channels, or codebooks, in the semantic quantizer. Must be lower than `num_quantizers`.
upsample_groups (`int`, *optional*, defaults to 512):
If `frame_rate!=encodec_frame_rate`, indicates the number of groups used in the upsampling operation to go from one rate to another.
num_hidden_layers (`int`, *optional*, defaults to 8):
Number of hidden layers in the Transformer models.
intermediate_size (`int`, *optional*, defaults to 2048):
Dimension of the MLP representations.
num_attention_heads (`int`, *optional*, defaults to 8):
Number of attention heads for each attention layer in the Transformer encoder.
num_key_value_heads (`int`, *optional*, defaults to 8):
This is the number of key_value heads that should be used to implement Grouped Query Attention. If
`num_key_value_heads=num_attention_heads`, the model will use Multi Head Attention (MHA), if
`num_key_value_heads=1` the model will use Multi Query Attention (MQA) otherwise GQA is used. When
converting a multi-head checkpoint to a GQA checkpoint, each group key and value head should be constructed
by meanpooling all the original heads within that group. For more details checkout [this
paper](https://arxiv.org/pdf/2305.13245.pdf). If it is not specified, will default to `8`.
head_dim (`int`, *optional*, defaults to `hidden_size // num_attention_heads`):
The attention head dimension.
hidden_act (`str` or `function`, *optional*, defaults to `"gelu"`):
The non-linear activation function (function or string) in the decoder.
max_position_embeddings (`int`, *optional*, defaults to 8000):
The maximum sequence length that this model might ever be used with. Mimi's sliding window attention
allows sequence of up to 8000 tokens.
initializer_range (`float`, *optional*, defaults to 0.02):
The standard deviation of the truncated_normal_initializer for initializing all weight matrices.
norm_eps (`float`, *optional*, defaults to 1e-05):
The epsilon used by the LayerNorm normalization layers.
use_cache (`bool`, *optional*, defaults to `False`):
Whether or not the model should return the last key/values attentions (not used by all models). Only
relevant if `config.is_decoder=True`.
rope_theta (`float`, *optional*, defaults to 10000.0):
The base period of the RoPE embeddings.
sliding_window (`int`, *optional*, defaults to 250):
Sliding window attention window size. If not specified, will default to `250`.
attention_dropout (`float`, *optional*, defaults to 0.0):
The dropout ratio for the attention probabilities.
layer_scale_initial_scale (`float`, *optional*, defaults to 0.01):
Initiale scale of the residual rescaling operation done in the Transformer models.
attention_bias (`bool`, defaults to `False`, *optional*, defaults to `False`):
Whether to use a bias in the query, key, value and output projection layers during self-attention.
Example:
```python
>>> from transformers import MimiModel, MimiConfig
>>> # Initializing a "kyutai/mimi" style configuration
>>> configuration = MimiConfig()
>>> # Initializing a model (with random weights) from the "kyutai/mimi" style configuration
>>> model = MimiModel(configuration)
>>> # Accessing the model configuration
>>> configuration = model.config
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/mimi.md | https://huggingface.co/docs/transformers/en/model_doc/mimi/#mimiconfig | #mimiconfig | .md | 169_3 |
The Mimi neural audio codec model.
This model inherits from [`PreTrainedModel`]. Check the superclass documentation for the generic methods the
library implements for all its model (such as downloading or saving, resizing the input embeddings, pruning heads
etc.)
This model is also a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) subclass.
Use it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage
and behavior.
Parameters:
config ([`MimiConfig`]):
Model configuration class with all the parameters of the model. Initializing with a config file does not
load the weights associated with the model, only the configuration. Check out the
[`~PreTrainedModel.from_pretrained`] method to load the model weights.
Methods: decode
- encode
- forward | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/mimi.md | https://huggingface.co/docs/transformers/en/model_doc/mimi/#mimimodel | #mimimodel | .md | 169_4 |
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|
The MMS model was proposed in [Scaling Speech Technology to 1,000+ Languages](https://arxiv.org/abs/2305.13516)
by Vineel Pratap, Andros Tjandra, Bowen Shi, Paden Tomasello, Arun Babu, Sayani Kundu, Ali Elkahky, Zhaoheng Ni, Apoorv Vyas, Maryam Fazel-Zarandi, Alexei Baevski, Yossi Adi, Xiaohui Zhang, Wei-Ning Hsu, Alexis Conneau, Michael Auli
The abstract from the paper is the following:
*Expanding the language coverage of speech technology has the potential to improve access to information for many more people.
However, current speech technology is restricted to about one hundred languages which is a small fraction of the over 7,000
languages spoken around the world.
The Massively Multilingual Speech (MMS) project increases the number of supported languages by 10-40x, depending on the task.
The main ingredients are a new dataset based on readings of publicly available religious texts and effectively leveraging
self-supervised learning. We built pre-trained wav2vec 2.0 models covering 1,406 languages,
a single multilingual automatic speech recognition model for 1,107 languages, speech synthesis models
for the same number of languages, as well as a language identification model for 4,017 languages.
Experiments show that our multilingual speech recognition model more than halves the word error rate of
Whisper on 54 languages of the FLEURS benchmark while being trained on a small fraction of the labeled data.*
Here are the different models open sourced in the MMS project. The models and code are originally released [here](https://github.com/facebookresearch/fairseq/tree/main/examples/mms). We have add them to the `transformers` framework, making them easier to use. | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/mms.md | https://huggingface.co/docs/transformers/en/model_doc/mms/#overview | #overview | .md | 170_1 |
The ASR model checkpoints can be found here : [mms-1b-fl102](https://huggingface.co/facebook/mms-1b-fl102), [mms-1b-l1107](https://huggingface.co/facebook/mms-1b-l1107), [mms-1b-all](https://huggingface.co/facebook/mms-1b-all). For best accuracy, use the `mms-1b-all` model.
Tips:
- All ASR models accept a float array corresponding to the raw waveform of the speech signal. The raw waveform should be pre-processed with [`Wav2Vec2FeatureExtractor`].
- The models were trained using connectionist temporal classification (CTC) so the model output has to be decoded using
[`Wav2Vec2CTCTokenizer`].
- You can load different language adapter weights for different languages via [`~Wav2Vec2PreTrainedModel.load_adapter`]. Language adapters only consists of roughly 2 million parameters
and can therefore be efficiently loaded on the fly when needed. | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/mms.md | https://huggingface.co/docs/transformers/en/model_doc/mms/#automatic-speech-recognition-asr | #automatic-speech-recognition-asr | .md | 170_2 |
By default MMS loads adapter weights for English. If you want to load adapter weights of another language
make sure to specify `target_lang=<your-chosen-target-lang>` as well as `"ignore_mismatched_sizes=True`.
The `ignore_mismatched_sizes=True` keyword has to be passed to allow the language model head to be resized according
to the vocabulary of the specified language.
Similarly, the processor should be loaded with the same target language
```py
from transformers import Wav2Vec2ForCTC, AutoProcessor
model_id = "facebook/mms-1b-all"
target_lang = "fra"
processor = AutoProcessor.from_pretrained(model_id, target_lang=target_lang)
model = Wav2Vec2ForCTC.from_pretrained(model_id, target_lang=target_lang, ignore_mismatched_sizes=True)
```
<Tip>
You can safely ignore a warning such as:
```text
Some weights of Wav2Vec2ForCTC were not initialized from the model checkpoint at facebook/mms-1b-all and are newly initialized because the shapes did not match:
- lm_head.bias: found shape torch.Size([154]) in the checkpoint and torch.Size([314]) in the model instantiated
- lm_head.weight: found shape torch.Size([154, 1280]) in the checkpoint and torch.Size([314, 1280]) in the model instantiated
You should probably TRAIN this model on a down-stream task to be able to use it for predictions and inference.
```
</Tip>
If you want to use the ASR pipeline, you can load your chosen target language as such:
```py
from transformers import pipeline
model_id = "facebook/mms-1b-all"
target_lang = "fra"
pipe = pipeline(model=model_id, model_kwargs={"target_lang": "fra", "ignore_mismatched_sizes": True})
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/mms.md | https://huggingface.co/docs/transformers/en/model_doc/mms/#loading | #loading | .md | 170_3 |
Next, let's look at how we can run MMS in inference and change adapter layers after having called [`~PretrainedModel.from_pretrained`]
First, we load audio data in different languages using the [Datasets](https://github.com/huggingface/datasets).
```py
from datasets import load_dataset, Audio
# English
stream_data = load_dataset("mozilla-foundation/common_voice_13_0", "en", split="test", streaming=True)
stream_data = stream_data.cast_column("audio", Audio(sampling_rate=16000))
en_sample = next(iter(stream_data))["audio"]["array"]
# French
stream_data = load_dataset("mozilla-foundation/common_voice_13_0", "fr", split="test", streaming=True)
stream_data = stream_data.cast_column("audio", Audio(sampling_rate=16000))
fr_sample = next(iter(stream_data))["audio"]["array"]
```
Next, we load the model and processor
```py
from transformers import Wav2Vec2ForCTC, AutoProcessor
import torch
model_id = "facebook/mms-1b-all"
processor = AutoProcessor.from_pretrained(model_id)
model = Wav2Vec2ForCTC.from_pretrained(model_id)
```
Now we process the audio data, pass the processed audio data to the model and transcribe the model output,
just like we usually do for [`Wav2Vec2ForCTC`].
```py
inputs = processor(en_sample, sampling_rate=16_000, return_tensors="pt")
with torch.no_grad():
outputs = model(**inputs).logits
ids = torch.argmax(outputs, dim=-1)[0]
transcription = processor.decode(ids)
# 'joe keton disapproved of films and buster also had reservations about the media'
```
We can now keep the same model in memory and simply switch out the language adapters by
calling the convenient [`~Wav2Vec2ForCTC.load_adapter`] function for the model and [`~Wav2Vec2CTCTokenizer.set_target_lang`] for the tokenizer.
We pass the target language as an input - `"fra"` for French.
```py
processor.tokenizer.set_target_lang("fra")
model.load_adapter("fra")
inputs = processor(fr_sample, sampling_rate=16_000, return_tensors="pt")
with torch.no_grad():
outputs = model(**inputs).logits
ids = torch.argmax(outputs, dim=-1)[0]
transcription = processor.decode(ids)
# "ce dernier est volé tout au long de l'histoire romaine"
```
In the same way the language can be switched out for all other supported languages. Please have a look at:
```py
processor.tokenizer.vocab.keys()
```
to see all supported languages.
To further improve performance from ASR models, language model decoding can be used. See the documentation [here](https://huggingface.co/facebook/mms-1b-all) for further details. | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/mms.md | https://huggingface.co/docs/transformers/en/model_doc/mms/#inference | #inference | .md | 170_4 |
MMS-TTS uses the same model architecture as VITS, which was added to 🤗 Transformers in v4.33. MMS trains a separate
model checkpoint for each of the 1100+ languages in the project. All available checkpoints can be found on the Hugging
Face Hub: [facebook/mms-tts](https://huggingface.co/models?sort=trending&search=facebook%2Fmms-tts), and the inference
documentation under [VITS](https://huggingface.co/docs/transformers/main/en/model_doc/vits). | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/mms.md | https://huggingface.co/docs/transformers/en/model_doc/mms/#speech-synthesis-tts | #speech-synthesis-tts | .md | 170_5 |
To use the MMS model, first update to the latest version of the Transformers library:
```bash
pip install --upgrade transformers accelerate
```
Since the flow-based model in VITS is non-deterministic, it is good practice to set a seed to ensure reproducibility of
the outputs.
- For languages with a Roman alphabet, such as English or French, the tokenizer can be used directly to
pre-process the text inputs. The following code example runs a forward pass using the MMS-TTS English checkpoint:
```python
import torch
from transformers import VitsTokenizer, VitsModel, set_seed
tokenizer = VitsTokenizer.from_pretrained("facebook/mms-tts-eng")
model = VitsModel.from_pretrained("facebook/mms-tts-eng")
inputs = tokenizer(text="Hello - my dog is cute", return_tensors="pt")
set_seed(555) # make deterministic
with torch.no_grad():
outputs = model(**inputs)
waveform = outputs.waveform[0]
```
The resulting waveform can be saved as a `.wav` file:
```python
import scipy
scipy.io.wavfile.write("synthesized_speech.wav", rate=model.config.sampling_rate, data=waveform)
```
Or displayed in a Jupyter Notebook / Google Colab:
```python
from IPython.display import Audio
Audio(waveform, rate=model.config.sampling_rate)
```
For certain languages with non-Roman alphabets, such as Arabic, Mandarin or Hindi, the [`uroman`](https://github.com/isi-nlp/uroman)
perl package is required to pre-process the text inputs to the Roman alphabet.
You can check whether you require the `uroman` package for your language by inspecting the `is_uroman` attribute of
the pre-trained `tokenizer`:
```python
from transformers import VitsTokenizer
tokenizer = VitsTokenizer.from_pretrained("facebook/mms-tts-eng")
print(tokenizer.is_uroman)
```
If required, you should apply the uroman package to your text inputs **prior** to passing them to the `VitsTokenizer`,
since currently the tokenizer does not support performing the pre-processing itself.
To do this, first clone the uroman repository to your local machine and set the bash variable `UROMAN` to the local path:
```bash
git clone https://github.com/isi-nlp/uroman.git
cd uroman
export UROMAN=$(pwd)
```
You can then pre-process the text input using the following code snippet. You can either rely on using the bash variable
`UROMAN` to point to the uroman repository, or you can pass the uroman directory as an argument to the `uromanize` function:
```python
import torch
from transformers import VitsTokenizer, VitsModel, set_seed
import os
import subprocess
tokenizer = VitsTokenizer.from_pretrained("facebook/mms-tts-kor")
model = VitsModel.from_pretrained("facebook/mms-tts-kor")
def uromanize(input_string, uroman_path):
"""Convert non-Roman strings to Roman using the `uroman` perl package."""
script_path = os.path.join(uroman_path, "bin", "uroman.pl")
command = ["perl", script_path]
process = subprocess.Popen(command, stdin=subprocess.PIPE, stdout=subprocess.PIPE, stderr=subprocess.PIPE)
# Execute the perl command
stdout, stderr = process.communicate(input=input_string.encode())
if process.returncode != 0:
raise ValueError(f"Error {process.returncode}: {stderr.decode()}")
# Return the output as a string and skip the new-line character at the end
return stdout.decode()[:-1]
text = "이봐 무슨 일이야"
uromanized_text = uromanize(text, uroman_path=os.environ["UROMAN"])
inputs = tokenizer(text=uromanized_text, return_tensors="pt")
set_seed(555) # make deterministic
with torch.no_grad():
outputs = model(inputs["input_ids"])
waveform = outputs.waveform[0]
```
**Tips:**
* The MMS-TTS checkpoints are trained on lower-cased, un-punctuated text. By default, the `VitsTokenizer` *normalizes* the inputs by removing any casing and punctuation, to avoid passing out-of-vocabulary characters to the model. Hence, the model is agnostic to casing and punctuation, so these should be avoided in the text prompt. You can disable normalisation by setting `normalize=False` in the call to the tokenizer, but this will lead to un-expected behaviour and is discouraged.
* The speaking rate can be varied by setting the attribute `model.speaking_rate` to a chosen value. Likewise, the randomness of the noise is controlled by `model.noise_scale`:
```python
import torch
from transformers import VitsTokenizer, VitsModel, set_seed
tokenizer = VitsTokenizer.from_pretrained("facebook/mms-tts-eng")
model = VitsModel.from_pretrained("facebook/mms-tts-eng")
inputs = tokenizer(text="Hello - my dog is cute", return_tensors="pt")
# make deterministic
set_seed(555)
# make speech faster and more noisy
model.speaking_rate = 1.5
model.noise_scale = 0.8
with torch.no_grad():
outputs = model(**inputs)
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/mms.md | https://huggingface.co/docs/transformers/en/model_doc/mms/#inference | #inference | .md | 170_6 |
Different LID models are available based on the number of languages they can recognize - [126](https://huggingface.co/facebook/mms-lid-126), [256](https://huggingface.co/facebook/mms-lid-256), [512](https://huggingface.co/facebook/mms-lid-512), [1024](https://huggingface.co/facebook/mms-lid-1024), [2048](https://huggingface.co/facebook/mms-lid-2048), [4017](https://huggingface.co/facebook/mms-lid-4017). | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/mms.md | https://huggingface.co/docs/transformers/en/model_doc/mms/#language-identification-lid | #language-identification-lid | .md | 170_7 |
First, we install transformers and some other libraries
```bash
pip install torch accelerate datasets[audio]
pip install --upgrade transformers
````
Next, we load a couple of audio samples via `datasets`. Make sure that the audio data is sampled to 16000 kHz.
```py
from datasets import load_dataset, Audio
# English
stream_data = load_dataset("mozilla-foundation/common_voice_13_0", "en", split="test", streaming=True)
stream_data = stream_data.cast_column("audio", Audio(sampling_rate=16000))
en_sample = next(iter(stream_data))["audio"]["array"]
# Arabic
stream_data = load_dataset("mozilla-foundation/common_voice_13_0", "ar", split="test", streaming=True)
stream_data = stream_data.cast_column("audio", Audio(sampling_rate=16000))
ar_sample = next(iter(stream_data))["audio"]["array"]
```
Next, we load the model and processor
```py
from transformers import Wav2Vec2ForSequenceClassification, AutoFeatureExtractor
import torch
model_id = "facebook/mms-lid-126"
processor = AutoFeatureExtractor.from_pretrained(model_id)
model = Wav2Vec2ForSequenceClassification.from_pretrained(model_id)
```
Now we process the audio data, pass the processed audio data to the model to classify it into a language, just like we usually do for Wav2Vec2 audio classification models such as [ehcalabres/wav2vec2-lg-xlsr-en-speech-emotion-recognition](https://huggingface.co/harshit345/xlsr-wav2vec-speech-emotion-recognition)
```py
# English
inputs = processor(en_sample, sampling_rate=16_000, return_tensors="pt")
with torch.no_grad():
outputs = model(**inputs).logits
lang_id = torch.argmax(outputs, dim=-1)[0].item()
detected_lang = model.config.id2label[lang_id]
# 'eng'
# Arabic
inputs = processor(ar_sample, sampling_rate=16_000, return_tensors="pt")
with torch.no_grad():
outputs = model(**inputs).logits
lang_id = torch.argmax(outputs, dim=-1)[0].item()
detected_lang = model.config.id2label[lang_id]
# 'ara'
```
To see all the supported languages of a checkpoint, you can print out the language ids as follows:
```py
processor.id2label.values()
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/mms.md | https://huggingface.co/docs/transformers/en/model_doc/mms/#inference | #inference | .md | 170_8 |
Pretrained models are available for two different sizes - [300M](https://huggingface.co/facebook/mms-300m) ,
[1Bil](https://huggingface.co/facebook/mms-1b).
<Tip>
The MMS for ASR architecture is based on the Wav2Vec2 model, refer to [Wav2Vec2's documentation page](wav2vec2) for further
details on how to finetune with models for various downstream tasks.
MMS-TTS uses the same model architecture as VITS, refer to [VITS's documentation page](vits) for API reference.
</Tip> | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/mms.md | https://huggingface.co/docs/transformers/en/model_doc/mms/#audio-pretrained-models | #audio-pretrained-models | .md | 170_9 |
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Licensed under the Apache License, Version 2.0 (the "License"); you may not use this file except in compliance with
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|
<Tip warning={true}>
This model is in maintenance mode only, we do not accept any new PRs changing its code.
If you run into any issues running this model, please reinstall the last version that supported this model: v4.30.0.
You can do so by running the following command: `pip install -U transformers==4.30.0`.
</Tip> | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/bort.md | https://huggingface.co/docs/transformers/en/model_doc/bort/#bort | #bort | .md | 171_1 |
The BORT model was proposed in [Optimal Subarchitecture Extraction for BERT](https://arxiv.org/abs/2010.10499) by
Adrian de Wynter and Daniel J. Perry. It is an optimal subset of architectural parameters for the BERT, which the
authors refer to as "Bort".
The abstract from the paper is the following:
*We extract an optimal subset of architectural parameters for the BERT architecture from Devlin et al. (2018) by
applying recent breakthroughs in algorithms for neural architecture search. This optimal subset, which we refer to as
"Bort", is demonstrably smaller, having an effective (that is, not counting the embedding layer) size of 5.5% the
original BERT-large architecture, and 16% of the net size. Bort is also able to be pretrained in 288 GPU hours, which
is 1.2% of the time required to pretrain the highest-performing BERT parametric architectural variant, RoBERTa-large
(Liu et al., 2019), and about 33% of that of the world-record, in GPU hours, required to train BERT-large on the same
hardware. It is also 7.9x faster on a CPU, as well as being better performing than other compressed variants of the
architecture, and some of the non-compressed variants: it obtains performance improvements of between 0.3% and 31%,
absolute, with respect to BERT-large, on multiple public natural language understanding (NLU) benchmarks.*
This model was contributed by [stefan-it](https://huggingface.co/stefan-it). The original code can be found [here](https://github.com/alexa/bort/). | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/bort.md | https://huggingface.co/docs/transformers/en/model_doc/bort/#overview | #overview | .md | 171_2 |
- BORT's model architecture is based on BERT, refer to [BERT's documentation page](bert) for the
model's API reference as well as usage examples.
- BORT uses the RoBERTa tokenizer instead of the BERT tokenizer, refer to [RoBERTa's documentation page](roberta) for the tokenizer's API reference as well as usage examples.
- BORT requires a specific fine-tuning algorithm, called [Agora](https://adewynter.github.io/notes/bort_algorithms_and_applications.html#fine-tuning-with-algebraic-topology) ,
that is sadly not open-sourced yet. It would be very useful for the community, if someone tries to implement the
algorithm to make BORT fine-tuning work. | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/bort.md | https://huggingface.co/docs/transformers/en/model_doc/bort/#usage-tips | #usage-tips | .md | 171_3 |
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|
The Emu3 model was proposed in [Emu3: Next-Token Prediction is All You Need](https://arxiv.org/abs/2409.18869) by Xinlong Wang, Xiaosong Zhang, Zhengxiong Luo, Quan Sun, Yufeng Cui, Jinsheng Wang, Fan Zhang, Yueze Wang, Zhen Li, Qiying Yu, Yingli Zhao, Yulong Ao, Xuebin Min, Tao Li, Boya Wu, Bo Zhao, Bowen Zhang, Liangdong Wang, Guang Liu, Zheqi He, Xi Yang, Jingjing Liu, Yonghua Lin, Tiejun Huang, Zhongyuan Wang.
Emu3 is a multimodal LLM that uses vector quantization to tokenize images into discrete tokens. Discretized image tokens are later fused with text token ids for image and text generation. The model can additionally generate images by predicting image token ids.
The abstract from the paper is the following:
*While next-token prediction is considered a promising path towards artificial general intelligence, it has struggled to excel in multimodal tasks, which are still dominated by diffusion models (e.g., Stable Diffusion) and compositional approaches (e.g., CLIP combined with LLMs). In this paper, we introduce Emu3, a new suite of state-of-the-art multimodal models trained solely with next-token prediction. By tokenizing images, text, and videos into a discrete space, we train a single transformer from scratch on a mixture of multimodal sequences. Emu3 outperforms several well-established task-specific models in both generation and perception tasks, surpassing flagship models such as SDXL and LLaVA-1.6, while eliminating the need for diffusion or compositional architectures. Emu3 is also capable of generating high-fidelity video via predicting the next token in a video sequence. We simplify complex multimodal model designs by converging on a singular focus: tokens, unlocking great potential for scaling both during training and inference. Our results demonstrate that next-token prediction is a promising path towards building general multimodal intelligence beyond language. We open-source key techniques and models to support further research in this direction.*
Tips:
- We advise users to set `processor.tokenizer.padding_side = "left"` before batched generation as it leads to more accurate results.
- Note that the model has been trained with a specific prompt format for chatting. Use `processor.apply_chat_template(my_conversation_dict)` to correctly format your prompts.
- Emu3 has two different checkpoints for image-generation and text-generation, make sure to use the correct checkpoint when loading the model. To generate an image, it is advised to use `prefix_constraints` so that the generated tokens are sampled only from possible image tokens. See more below for usage examples.
> [!TIP]
> Emu3 implementation in Transformers uses a special image token to indicate where to merge image embeddings. The special image token isn't new and uses one of the reserved tokens: `<|extra_0|>`. You have to add `<image>` to your prompt in the place where the image should be embedded for correct generation.
This model was contributed by [RaushanTurganbay](https://huggingface.co/RaushanTurganbay).
The original code can be found [here](https://github.com/baaivision/Emu3). | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/emu3.md | https://huggingface.co/docs/transformers/en/model_doc/emu3/#overview | #overview | .md | 172_1 |
Here's how to load the model and perform inference in half-precision (`torch.bfloat16`) to generate textual output from text or text and image inputs:
```python
from transformers import Emu3Processor, Emu3ForConditionalGeneration
import torch
from PIL import Image
import requests
processor = Emu3Processor.from_pretrained("BAAI/Emu3-Chat-hf")
model = Emu3ForConditionalGeneration.from_pretrained("BAAI/Emu3-Chat-hf", torch_dtype=torch.bfloat16, device_map="cuda")
# prepare image and text prompt
url = 'http://images.cocodataset.org/val2017/000000039769.jpg'
image = Image.open(requests.get(url, stream=True).raw)
prompt = "What do you see in this image?<image>"
inputs = processor(images=image, text=prompt, return_tensors="pt").to(model.device, dtype=torch.bfloat16)
# autoregressively complete prompt
output = model.generate(**inputs, max_new_tokens=50)
print(processor.decode(output[0], skip_special_tokens=True))
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/emu3.md | https://huggingface.co/docs/transformers/en/model_doc/emu3/#text-generation-inference | #text-generation-inference | .md | 172_2 |
Emu3 can also generate images from textual input. Here is how you can do it:
```python
processor = Emu3Processor.from_pretrained("BAAI/Emu3-Gen-hf")
model = Emu3ForConditionalGeneration.from_pretrained("BAAI/Emu3-Gen-hf", torch_dtype="bfloat16", device_map="auto", attn_implementation="flash_attention_2")
inputs = processor(
text=["a portrait of young girl. masterpiece, film grained, best quality.", "a dog running under the rain"],
padding=True,
return_tensors="pt",
return_for_image_generation=True,
)
inputs = inputs.to(device="cuda:0", dtype=torch.bfloat16)
neg_prompt = "lowres, bad anatomy, bad hands, text, error, missing fingers, extra digit, fewer digits, cropped, worst quality, low quality, normal quality, jpeg artifacts, signature, watermark, username, blurry."
neg_inputs = processor(text=[neg_prompt] * 2, return_tensors="pt").to(device="cuda:0")
image_sizes = inputs.pop("image_sizes")
HEIGHT, WIDTH = image_sizes[0]
VISUAL_TOKENS = model.vocabulary_mapping.image_tokens
def prefix_allowed_tokens_fn(batch_id, input_ids):
height, width = HEIGHT, WIDTH
visual_tokens = VISUAL_TOKENS
image_wrapper_token_id = torch.tensor([processor.tokenizer.image_wrapper_token_id], device=model.device)
eoi_token_id = torch.tensor([processor.tokenizer.eoi_token_id], device=model.device)
eos_token_id = torch.tensor([processor.tokenizer.eos_token_id], device=model.device)
pad_token_id = torch.tensor([processor.tokenizer.pad_token_id], device=model.device)
eof_token_id = torch.tensor([processor.tokenizer.eof_token_id], device=model.device)
eol_token_id = processor.tokenizer.encode("<|extra_200|>", return_tensors="pt")[0]
position = torch.nonzero(input_ids == image_wrapper_token_id, as_tuple=True)[0][0]
offset = input_ids.shape[0] - position
if offset % (width + 1) == 0:
return (eol_token_id, )
elif offset == (width + 1) * height + 1:
return (eof_token_id, )
elif offset == (width + 1) * height + 2:
return (eoi_token_id, )
elif offset == (width + 1) * height + 3:
return (eos_token_id, )
elif offset > (width + 1) * height + 3:
return (pad_token_id, )
else:
return visual_tokens
out = model.generate(
**inputs,
max_new_tokens=50_000, # make sure to have enough tokens for one image
prefix_allowed_tokens_fn=prefix_allowed_tokens_fn,
return_dict_in_generate=True,
negative_prompt_ids=neg_inputs.input_ids, # indicate for Classifier-Free Guidance
negative_prompt_attention_mask=neg_inputs.attention_mask,
)
image = model.decode_image_tokens(out.sequences[:, inputs.input_ids.shape[1]: ], height=HEIGHT, width=WIDTH)
images = processor.postprocess(list(image.float()), return_tensors="PIL.Image.Image") # internally we convert to np but it's not supported in bf16 precision
for i, image in enumerate(images['pixel_values']):
image.save(f"result{i}.png")
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/emu3.md | https://huggingface.co/docs/transformers/en/model_doc/emu3/#image-generation-inference | #image-generation-inference | .md | 172_3 |
This is the configuration class to store the configuration of a [`Emu3Model`]. It is used to instantiate a
emu3 model according to the specified arguments, defining the model architecture. Instantiating a
configuration with the defaults will yield a similar configuration to that of the
[Emu3-community/Emu3-Chat-hf](https://huggingface.co/Emu3-community/Emu3-Chat-hf).
Configuration objects inherit from [`PretrainedConfig`] and can be used to control the model outputs. Read the
documentation from [`PretrainedConfig`] for more information.
Args:
vq_config (`Union[Dict, Emu3VQVAEConfig]`, *optional*):
Emu3VQVAEConfig instance containing the configuration for the VQ-VAE model.
text_config (`Union[Dict, Emu3TextConfig]``, *optional*):
Emu3TextConfig instance containing the configuration for the language model.
vocabulary_map (`dict`, *optional*):
A dictionary containing the vocabulary map from the tokenizer. Used to obtain tokens from the image inputs. | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/emu3.md | https://huggingface.co/docs/transformers/en/model_doc/emu3/#emu3config | #emu3config | .md | 172_4 |
This is the configuration class to store the configuration of a [`Emu3VQVAE`]. It is used to instantiate an VQ-VAE
model according to the specified arguments, defining the model architecture. Instantiating a configuration with the
defaults will yield a configuration to the VQ model presented in Emu3 paper.
Configuration objects inherit from [`PretrainedConfig`] and can be used to control the model outputs. Read the
documentation from [`PretrainedConfig`] for more information.
Args:
codebook_size (`int`, *optional*, defaults to 32768):
Codebook size of the VQ model.
embed_dim (`int`, *optional*, defaults to 4):
Dimension of the quantized vector in codebook.
latent_channels (`int`, *optional*, defaults to 4):
Dimension of the output channel of encoder and the input channel of decoder
double_latent (`bool`, *optional*, defaults to `False`):
Whether double the output dim of the encoder.
in_channels (`int`, *optional*, defaults to 3):
Input channel of encoder.
out_channels (`int`, *optional*, defaults to 3):
Output channel of decoder.
temporal_downsample_factor (`int`, *optional*, defaults to 4):
Temporal downsample factor.
base_channels (`int`, *optional*, defaults to 256):
Basic channel number of the intermediate blocks.
channel_multiplier (`List[int]`, *optional*, defaults to `[1, 2, 2, 4]`):
Channel scaling factor of the intermediate blocks.
num_res_blocks (`int`, *optional*, defaults to 2):
Residual block number in each stage.
attn_resolutions (`List[int]`, *optional*, defaults to `[3]`):
Stage indices to apply attention.
hidden_size (`int`, *optional*, defaults to 1024):
Dimension of the hidden representations in the attention layer.
num_attention_heads (`int`, *optional*, defaults to 1):
Number of attention heads for each attention layer.
attention_dropout (`float`, *optional*, defaults to 0.0):
The dropout ratio for the attention probabilities.
```python
>>> from transformers import Emu3VQVAE, Emu3VQVAEConfig
>>> # Initializing a video VQ model of Emu3 configuration
>>> configuration = Emu3VQVAEConfig()
>>> # Initializing a model from the Emu3 VQ model style configuration
>>> model = Emu3VQVAE(configuration)
>>> # Accessing the model configuration
>>> configuration = model.config
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/emu3.md | https://huggingface.co/docs/transformers/en/model_doc/emu3/#emu3vqvaeconfig | #emu3vqvaeconfig | .md | 172_5 |
This is the configuration class to store the configuration of a [`Emu3TextModel`]. It is used to instantiate a
emu3 model according to the specified arguments, defining the model architecture. Instantiating a
configuration with the defaults will yield a similar configuration to that of the
[Emu3-community/Emu3-Chat-hf](https://huggingface.co/Emu3-community/Emu3-Chat-hf).
Configuration objects inherit from [`PretrainedConfig`] and can be used to control the model outputs. Read the
documentation from [`PretrainedConfig`] for more information.
Args:
vocab_size (`int`, *optional*, defaults to 184622):
Vocabulary size of the Emu3 model. Defines the number of different tokens that can be represented by the
`inputs_ids` passed when calling [`Emu3Model`]
hidden_size (`int`, *optional*, defaults to 4096):
Dimension of the hidden representations.
intermediate_size (`int`, *optional*, defaults to 14336):
Dimension of the MLP representations.
num_hidden_layers (`int`, *optional*, defaults to 32):
Number of hidden layers in the Transformer decoder.
num_attention_heads (`int`, *optional*, defaults to 32):
Number of attention heads for each attention layer in the Transformer decoder.
num_key_value_heads (`int`, *optional*, defaults to 8):
This is the number of key_value heads that should be used to implement Grouped Query Attention. If
`num_key_value_heads=num_attention_heads`, the model will use Multi Head Attention (MHA), if
`num_key_value_heads=1 the model will use Multi Query Attention (MQA) otherwise GQA is used. When
converting a multi-head checkpoint to a GQA checkpoint, each group key and value head should be constructed
by meanpooling all the original heads within that group. For more details checkout [this
paper](https://arxiv.org/pdf/2305.13245.pdf). If it is not specified, will default to
`num_attention_heads`.
hidden_act (`str` or `function`, *optional*, defaults to `"silu"`):
The non-linear activation function (function or string) in the decoder.
max_position_embeddings (`int`, *optional*, defaults to 9216):
The maximum sequence length that this model might ever be used with. Emu supports up to 9216 tokens,
rms_norm_eps (`float`, *optional*, defaults to 1e-05):
The epsilon used by the rms normalization layers.
use_cache (`bool`, *optional*, defaults to `True`):
Whether or not the model should return the last key/values attentions (not used by all models). Only
relevant if `config.is_decoder=True`.
pad_token_id (`int`, *optional*, defaults to 151643):
Padding token id.
bos_token_id (`int`, *optional*, defaults to 151849):
Beginning of stream token id.
eos_token_id (`int`, *optional*, defaults to 151850):
End of stream token id.
tie_word_embeddings (`bool`, *optional*, defaults to `False`):
Whether to tie weight embeddings
rope_theta (`float`, *optional*, defaults to 1000000.0):
The base period of the RoPE embeddings.
rope_scaling (`Dict`, *optional*):
Dictionary containing the scaling configuration for the RoPE embeddings. NOTE: if you apply new rope type
and you expect the model to work on longer `max_position_embeddings`, we recommend you to update this value
accordingly.
Expected contents:
`rope_type` (`str`):
The sub-variant of RoPE to use. Can be one of ['default', 'linear', 'dynamic', 'yarn', 'longrope',
'llama3'], with 'default' being the original RoPE implementation.
`factor` (`float`, *optional*):
Used with all rope types except 'default'. The scaling factor to apply to the RoPE embeddings. In
most scaling types, a `factor` of x will enable the model to handle sequences of length x *
original maximum pre-trained length.
`original_max_position_embeddings` (`int`, *optional*):
Used with 'dynamic', 'longrope' and 'llama3'. The original max position embeddings used during
pretraining.
`attention_factor` (`float`, *optional*):
Used with 'yarn' and 'longrope'. The scaling factor to be applied on the attention
computation. If unspecified, it defaults to value recommended by the implementation, using the
`factor` field to infer the suggested value.
`beta_fast` (`float`, *optional*):
Only used with 'yarn'. Parameter to set the boundary for extrapolation (only) in the linear
ramp function. If unspecified, it defaults to 32.
`beta_slow` (`float`, *optional*):
Only used with 'yarn'. Parameter to set the boundary for interpolation (only) in the linear
ramp function. If unspecified, it defaults to 1.
`short_factor` (`List[float]`, *optional*):
Only used with 'longrope'. The scaling factor to be applied to short contexts (<
`original_max_position_embeddings`). Must be a list of numbers with the same length as the hidden
size divided by the number of attention heads divided by 2
`long_factor` (`List[float]`, *optional*):
Only used with 'longrope'. The scaling factor to be applied to long contexts (<
`original_max_position_embeddings`). Must be a list of numbers with the same length as the hidden
size divided by the number of attention heads divided by 2
`low_freq_factor` (`float`, *optional*):
Only used with 'llama3'. Scaling factor applied to low frequency components of the RoPE
`high_freq_factor` (`float`, *optional*):
Only used with 'llama3'. Scaling factor applied to high frequency components of the RoPE
mlp_bias (`bool`, *optional*, defaults to `False`):
Whether to use a bias in up_proj, down_proj and gate_proj layers in the MLP layers.
attention_bias (`bool`, *optional*, defaults to `False`):
Whether to use a bias in the query, key, value and output projection layers during self-attention.
attention_dropout (`float`, *optional*, defaults to 0.1):
The dropout ratio for the attention probabilities.
initializer_range (`float`, *optional*, defaults to 0.02):
The standard deviation of the truncated_normal_initializer for initializing all weight matrices.
```python
>>> from transformers import Emu3Model, Emu3Config
>>> # Initializing a Emu3-community/Emu3-Chat-hf style configuration
>>> configuration = Emu3Config()
>>> # Initializing a model from the Emu3-community/Emu3-Chat-hf style configuration
>>> model = Emu3Model(configuration)
>>> # Accessing the model configuration
>>> configuration = model.config
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/emu3.md | https://huggingface.co/docs/transformers/en/model_doc/emu3/#emu3textconfig | #emu3textconfig | .md | 172_6 |
Constructs a Emu3 processor which wraps a Emu3 image processor and a GPT2 tokenizer into a single
processor.
[`Emu3Processor`] offers all the functionalities of [`Emu3ImageProcessor`] and [`GPT2TokenizerFast`].
See the [`~Emu3Processor.__call__`] and [`~Emu3Processor.decode`] for more information.
Args:
image_processor ([`Emu3ImageProcessor`]):
The image processor is a required input.
tokenizer ([`Emu3TokenizerFast`]):
The tokenizer is a required input.
chat_template (`str`, *optional*): A Jinja template which will be used to convert lists of messages
in a chat into a tokenizable string. | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/emu3.md | https://huggingface.co/docs/transformers/en/model_doc/emu3/#emu3processor | #emu3processor | .md | 172_7 |
Constructs a Emu3 image processor that dynamically resizes images based on the original images.
Args:
do_resize (`bool`, *optional*, defaults to `True`):
Whether to resize the image's (height, width) dimensions.
resample (`PILImageResampling`, *optional*, defaults to `Resampling.BICUBIC`):
Resampling filter to use when resizing the image.
do_rescale (`bool`, *optional*, defaults to `True`):
Whether to rescale the image by the specified scale `rescale_factor`.
rescale_factor (`int` or `float`, *optional*, defaults to `1/255`):
Scale factor to use if rescaling the image.
do_normalize (`bool`, *optional*, defaults to `True`):
Whether to normalize the image.
image_mean (`float` or `List[float]`, *optional*, defaults to `[0.48145466, 0.4578275, 0.40821073]`):
Mean to use if normalizing the image. This is a float or list of floats for each channel in the image.
image_std (`float` or `List[float]`, *optional*, defaults to `[0.26862954, 0.26130258, 0.27577711]`):
Standard deviation to use if normalizing the image. This is a float or list of floats for each channel in the image.
do_convert_rgb (`bool`, *optional*, defaults to `True`):
Whether to convert the image to RGB.
do_pad (`bool`, *optional*, defaults to `True`):
Whether to pad the image. If `True`, will pad the patch dimension of the images in the batch to the largest
number of patches in the batch. Padding will be applied to the bottom and right with zeros.
min_pixels (`int`, *optional*, defaults to `512 * 512`):
The min pixels of the image to resize the image.
max_pixels (`int`, *optional*, defaults to `1024 * 1024`):
The max pixels of the image to resize the image.
spatial_factor (`int`, *optional*, defaults to 8):
The spatial downsample factor the image will be downsampled in feature extracting phase
Methods: preprocess | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/emu3.md | https://huggingface.co/docs/transformers/en/model_doc/emu3/#emu3imageprocessor | #emu3imageprocessor | .md | 172_8 |
The VQ-VAE model used in Emu3 for encoding/decoding images into discrete tokens.
This model follows the "Make-a-scene: Scene-based text-to-image generation with human priors" paper from
[ Oran Gafni, Adam Polyak, Oron Ashual, Shelly Sheynin, Devi Parikh, and Yaniv Taigman](https://arxiv.org/abs/2203.13131).
This model inherits from [`PreTrainedModel`]. Check the superclass documentation for the generic methods the
library implements for all its model (such as downloading or saving, resizing the input embeddings, pruning heads
etc.)
This model is also a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) subclass.
Use it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage
and behavior.
Parameters:
config ([`Emu3VQVAEConfig`]):
Model configuration class with all the parameters of the model. Initializing with a config file does not
load the weights associated with the model, only the configuration. Check out the
[`~PreTrainedModel.from_pretrained`] method to load the model weights.
Methods: forward | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/emu3.md | https://huggingface.co/docs/transformers/en/model_doc/emu3/#emu3vqvae | #emu3vqvae | .md | 172_9 |
The bare Emu3Text Model outputting raw hidden-states without any specific head on top.
This model inherits from [`PreTrainedModel`]. Check the superclass documentation for the generic methods the
library implements for all its model (such as downloading or saving, resizing the input embeddings, pruning heads
etc.)
This model is also a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) subclass.
Use it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage
and behavior.
Parameters:
config ([`Emu3Config`]):
Model configuration class with all the parameters of the model. Initializing with a config file does not
load the weights associated with the model, only the configuration. Check out the
[`~PreTrainedModel.from_pretrained`] method to load the model weights.
Transformer decoder consisting of *config.num_hidden_layers* layers. Each layer is a [`Emu3TextDecoderLayer`]
Args:
config: Emu3TextConfig
Methods: forward | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/emu3.md | https://huggingface.co/docs/transformers/en/model_doc/emu3/#emu3textmodel | #emu3textmodel | .md | 172_10 |
No docstring available for Emu3ForCausalLM
Methods: forward | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/emu3.md | https://huggingface.co/docs/transformers/en/model_doc/emu3/#emu3forcausallm | #emu3forcausallm | .md | 172_11 |
No docstring available for Emu3ForConditionalGeneration
Methods: forward | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/emu3.md | https://huggingface.co/docs/transformers/en/model_doc/emu3/#emu3forconditionalgeneration | #emu3forconditionalgeneration | .md | 172_12 |
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--> | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/squeezebert.md | https://huggingface.co/docs/transformers/en/model_doc/squeezebert/ | .md | 173_0 |
|
The SqueezeBERT model was proposed in [SqueezeBERT: What can computer vision teach NLP about efficient neural networks?](https://arxiv.org/abs/2006.11316) by Forrest N. Iandola, Albert E. Shaw, Ravi Krishna, Kurt W. Keutzer. It's a
bidirectional transformer similar to the BERT model. The key difference between the BERT architecture and the
SqueezeBERT architecture is that SqueezeBERT uses [grouped convolutions](https://blog.yani.io/filter-group-tutorial)
instead of fully-connected layers for the Q, K, V and FFN layers.
The abstract from the paper is the following:
*Humans read and write hundreds of billions of messages every day. Further, due to the availability of large datasets,
large computing systems, and better neural network models, natural language processing (NLP) technology has made
significant strides in understanding, proofreading, and organizing these messages. Thus, there is a significant
opportunity to deploy NLP in myriad applications to help web users, social networks, and businesses. In particular, we
consider smartphones and other mobile devices as crucial platforms for deploying NLP models at scale. However, today's
highly-accurate NLP neural network models such as BERT and RoBERTa are extremely computationally expensive, with
BERT-base taking 1.7 seconds to classify a text snippet on a Pixel 3 smartphone. In this work, we observe that methods
such as grouped convolutions have yielded significant speedups for computer vision networks, but many of these
techniques have not been adopted by NLP neural network designers. We demonstrate how to replace several operations in
self-attention layers with grouped convolutions, and we use this technique in a novel network architecture called
SqueezeBERT, which runs 4.3x faster than BERT-base on the Pixel 3 while achieving competitive accuracy on the GLUE test
set. The SqueezeBERT code will be released.*
This model was contributed by [forresti](https://huggingface.co/forresti). | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/squeezebert.md | https://huggingface.co/docs/transformers/en/model_doc/squeezebert/#overview | #overview | .md | 173_1 |
- SqueezeBERT is a model with absolute position embeddings so it's usually advised to pad the inputs on the right
rather than the left.
- SqueezeBERT is similar to BERT and therefore relies on the masked language modeling (MLM) objective. It is therefore
efficient at predicting masked tokens and at NLU in general, but is not optimal for text generation. Models trained
with a causal language modeling (CLM) objective are better in that regard.
- For best results when finetuning on sequence classification tasks, it is recommended to start with the
*squeezebert/squeezebert-mnli-headless* checkpoint. | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/squeezebert.md | https://huggingface.co/docs/transformers/en/model_doc/squeezebert/#usage-tips | #usage-tips | .md | 173_2 |
- [Text classification task guide](../tasks/sequence_classification)
- [Token classification task guide](../tasks/token_classification)
- [Question answering task guide](../tasks/question_answering)
- [Masked language modeling task guide](../tasks/masked_language_modeling)
- [Multiple choice task guide](../tasks/multiple_choice) | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/squeezebert.md | https://huggingface.co/docs/transformers/en/model_doc/squeezebert/#resources | #resources | .md | 173_3 |
This is the configuration class to store the configuration of a [`SqueezeBertModel`]. It is used to instantiate a
SqueezeBERT model according to the specified arguments, defining the model architecture. Instantiating a
configuration with the defaults will yield a similar configuration to that of the SqueezeBERT
[squeezebert/squeezebert-uncased](https://huggingface.co/squeezebert/squeezebert-uncased) architecture.
Configuration objects inherit from [`PretrainedConfig`] and can be used to control the model outputs. Read the
documentation from [`PretrainedConfig`] for more information.
Args:
vocab_size (`int`, *optional*, defaults to 30522):
Vocabulary size of the SqueezeBERT model. Defines the number of different tokens that can be represented by
the `inputs_ids` passed when calling [`SqueezeBertModel`].
hidden_size (`int`, *optional*, defaults to 768):
Dimensionality of the encoder layers and the pooler layer.
num_hidden_layers (`int`, *optional*, defaults to 12):
Number of hidden layers in the Transformer encoder.
num_attention_heads (`int`, *optional*, defaults to 12):
Number of attention heads for each attention layer in the Transformer encoder.
intermediate_size (`int`, *optional*, defaults to 3072):
Dimensionality of the "intermediate" (often named feed-forward) layer in the Transformer encoder.
hidden_act (`str` or `Callable`, *optional*, defaults to `"gelu"`):
The non-linear activation function (function or string) in the encoder and pooler. If string, `"gelu"`,
`"relu"`, `"silu"` and `"gelu_new"` are supported.
hidden_dropout_prob (`float`, *optional*, defaults to 0.1):
The dropout probability for all fully connected layers in the embeddings, encoder, and pooler.
attention_probs_dropout_prob (`float`, *optional*, defaults to 0.1):
The dropout ratio for the attention probabilities.
max_position_embeddings (`int`, *optional*, defaults to 512):
The maximum sequence length that this model might ever be used with. Typically set this to something large
just in case (e.g., 512 or 1024 or 2048).
type_vocab_size (`int`, *optional*, defaults to 2):
The vocabulary size of the `token_type_ids` passed when calling [`BertModel`] or [`TFBertModel`].
initializer_range (`float`, *optional*, defaults to 0.02):
The standard deviation of the truncated_normal_initializer for initializing all weight matrices.
layer_norm_eps (`float`, *optional*, defaults to 1e-12):
pad_token_id (`int`, *optional*, defaults to 0):
The ID of the token in the word embedding to use as padding.
embedding_size (`int`, *optional*, defaults to 768):
The dimension of the word embedding vectors.
q_groups (`int`, *optional*, defaults to 4):
The number of groups in Q layer.
k_groups (`int`, *optional*, defaults to 4):
The number of groups in K layer.
v_groups (`int`, *optional*, defaults to 4):
The number of groups in V layer.
post_attention_groups (`int`, *optional*, defaults to 1):
The number of groups in the first feed forward network layer.
intermediate_groups (`int`, *optional*, defaults to 4):
The number of groups in the second feed forward network layer.
output_groups (`int`, *optional*, defaults to 4):
The number of groups in the third feed forward network layer.
Examples:
```python
>>> from transformers import SqueezeBertConfig, SqueezeBertModel
>>> # Initializing a SqueezeBERT configuration
>>> configuration = SqueezeBertConfig()
>>> # Initializing a model (with random weights) from the configuration above
>>> model = SqueezeBertModel(configuration)
>>> # Accessing the model configuration
>>> configuration = model.config
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/squeezebert.md | https://huggingface.co/docs/transformers/en/model_doc/squeezebert/#squeezebertconfig | #squeezebertconfig | .md | 173_4 |
Construct a SqueezeBERT tokenizer. Based on WordPiece.
This tokenizer inherits from [`PreTrainedTokenizer`] which contains most of the main methods. Users should refer to
this superclass for more information regarding those methods.
Args:
vocab_file (`str`):
File containing the vocabulary.
do_lower_case (`bool`, *optional*, defaults to `True`):
Whether or not to lowercase the input when tokenizing.
do_basic_tokenize (`bool`, *optional*, defaults to `True`):
Whether or not to do basic tokenization before WordPiece.
never_split (`Iterable`, *optional*):
Collection of tokens which will never be split during tokenization. Only has an effect when
`do_basic_tokenize=True`
unk_token (`str`, *optional*, defaults to `"[UNK]"`):
The unknown token. A token that is not in the vocabulary cannot be converted to an ID and is set to be this
token instead.
sep_token (`str`, *optional*, defaults to `"[SEP]"`):
The separator token, which is used when building a sequence from multiple sequences, e.g. two sequences for
sequence classification or for a text and a question for question answering. It is also used as the last
token of a sequence built with special tokens.
pad_token (`str`, *optional*, defaults to `"[PAD]"`):
The token used for padding, for example when batching sequences of different lengths.
cls_token (`str`, *optional*, defaults to `"[CLS]"`):
The classifier token which is used when doing sequence classification (classification of the whole sequence
instead of per-token classification). It is the first token of the sequence when built with special tokens.
mask_token (`str`, *optional*, defaults to `"[MASK]"`):
The token used for masking values. This is the token used when training this model with masked language
modeling. This is the token which the model will try to predict.
tokenize_chinese_chars (`bool`, *optional*, defaults to `True`):
Whether or not to tokenize Chinese characters.
This should likely be deactivated for Japanese (see this
[issue](https://github.com/huggingface/transformers/issues/328)).
strip_accents (`bool`, *optional*):
Whether or not to strip all accents. If this option is not specified, then it will be determined by the
value for `lowercase` (as in the original SqueezeBERT).
clean_up_tokenization_spaces (`bool`, *optional*, defaults to `True`):
Whether or not to cleanup spaces after decoding, cleanup consists in removing potential artifacts like
extra spaces.
Methods: build_inputs_with_special_tokens
- get_special_tokens_mask
- create_token_type_ids_from_sequences
- save_vocabulary | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/squeezebert.md | https://huggingface.co/docs/transformers/en/model_doc/squeezebert/#squeezeberttokenizer | #squeezeberttokenizer | .md | 173_5 |
Construct a "fast" SqueezeBERT tokenizer (backed by HuggingFace's *tokenizers* library). Based on WordPiece.
This tokenizer inherits from [`PreTrainedTokenizerFast`] which contains most of the main methods. Users should
refer to this superclass for more information regarding those methods.
Args:
vocab_file (`str`):
File containing the vocabulary.
do_lower_case (`bool`, *optional*, defaults to `True`):
Whether or not to lowercase the input when tokenizing.
unk_token (`str`, *optional*, defaults to `"[UNK]"`):
The unknown token. A token that is not in the vocabulary cannot be converted to an ID and is set to be this
token instead.
sep_token (`str`, *optional*, defaults to `"[SEP]"`):
The separator token, which is used when building a sequence from multiple sequences, e.g. two sequences for
sequence classification or for a text and a question for question answering. It is also used as the last
token of a sequence built with special tokens.
pad_token (`str`, *optional*, defaults to `"[PAD]"`):
The token used for padding, for example when batching sequences of different lengths.
cls_token (`str`, *optional*, defaults to `"[CLS]"`):
The classifier token which is used when doing sequence classification (classification of the whole sequence
instead of per-token classification). It is the first token of the sequence when built with special tokens.
mask_token (`str`, *optional*, defaults to `"[MASK]"`):
The token used for masking values. This is the token used when training this model with masked language
modeling. This is the token which the model will try to predict.
clean_text (`bool`, *optional*, defaults to `True`):
Whether or not to clean the text before tokenization by removing any control characters and replacing all
whitespaces by the classic one.
tokenize_chinese_chars (`bool`, *optional*, defaults to `True`):
Whether or not to tokenize Chinese characters. This should likely be deactivated for Japanese (see [this
issue](https://github.com/huggingface/transformers/issues/328)).
strip_accents (`bool`, *optional*):
Whether or not to strip all accents. If this option is not specified, then it will be determined by the
value for `lowercase` (as in the original SqueezeBERT).
wordpieces_prefix (`str`, *optional*, defaults to `"##"`):
The prefix for subwords. | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/squeezebert.md | https://huggingface.co/docs/transformers/en/model_doc/squeezebert/#squeezeberttokenizerfast | #squeezeberttokenizerfast | .md | 173_6 |
The bare SqueezeBERT Model transformer outputting raw hidden-states without any specific head on top.
The SqueezeBERT model was proposed in [SqueezeBERT: What can computer vision teach NLP about efficient neural
networks?](https://arxiv.org/abs/2006.11316) by Forrest N. Iandola, Albert E. Shaw, Ravi Krishna, and Kurt W.
Keutzer
This model inherits from [`PreTrainedModel`]. Check the superclass documentation for the generic methods the
library implements for all its model (such as downloading or saving, resizing the input embeddings, pruning heads
etc.)
This model is also a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) subclass.
Use it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage
and behavior.
For best results finetuning SqueezeBERT on text classification tasks, it is recommended to use the
*squeezebert/squeezebert-mnli-headless* checkpoint as a starting point.
Parameters:
config ([`SqueezeBertConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Hierarchy:
```
Internal class hierarchy:
SqueezeBertModel
SqueezeBertEncoder
SqueezeBertModule
SqueezeBertSelfAttention
ConvActivation
ConvDropoutLayerNorm
```
Data layouts:
```
Input data is in [batch, sequence_length, hidden_size] format.
Data inside the encoder is in [batch, hidden_size, sequence_length] format. But, if `output_hidden_states == True`, the data from inside the encoder is returned in [batch, sequence_length, hidden_size] format.
The final output of the encoder is in [batch, sequence_length, hidden_size] format.
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/squeezebert.md | https://huggingface.co/docs/transformers/en/model_doc/squeezebert/#squeezebertmodel | #squeezebertmodel | .md | 173_7 |
SqueezeBERT Model with a `language modeling` head on top.
The SqueezeBERT model was proposed in [SqueezeBERT: What can computer vision teach NLP about efficient neural
networks?](https://arxiv.org/abs/2006.11316) by Forrest N. Iandola, Albert E. Shaw, Ravi Krishna, and Kurt W.
Keutzer
This model inherits from [`PreTrainedModel`]. Check the superclass documentation for the generic methods the
library implements for all its model (such as downloading or saving, resizing the input embeddings, pruning heads
etc.)
This model is also a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) subclass.
Use it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage
and behavior.
For best results finetuning SqueezeBERT on text classification tasks, it is recommended to use the
*squeezebert/squeezebert-mnli-headless* checkpoint as a starting point.
Parameters:
config ([`SqueezeBertConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Hierarchy:
```
Internal class hierarchy:
SqueezeBertModel
SqueezeBertEncoder
SqueezeBertModule
SqueezeBertSelfAttention
ConvActivation
ConvDropoutLayerNorm
```
Data layouts:
```
Input data is in [batch, sequence_length, hidden_size] format.
Data inside the encoder is in [batch, hidden_size, sequence_length] format. But, if `output_hidden_states == True`, the data from inside the encoder is returned in [batch, sequence_length, hidden_size] format.
The final output of the encoder is in [batch, sequence_length, hidden_size] format.
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/squeezebert.md | https://huggingface.co/docs/transformers/en/model_doc/squeezebert/#squeezebertformaskedlm | #squeezebertformaskedlm | .md | 173_8 |
SqueezeBERT Model transformer with a sequence classification/regression head on top (a linear layer on top of the
pooled output) e.g. for GLUE tasks.
The SqueezeBERT model was proposed in [SqueezeBERT: What can computer vision teach NLP about efficient neural
networks?](https://arxiv.org/abs/2006.11316) by Forrest N. Iandola, Albert E. Shaw, Ravi Krishna, and Kurt W.
Keutzer
This model inherits from [`PreTrainedModel`]. Check the superclass documentation for the generic methods the
library implements for all its model (such as downloading or saving, resizing the input embeddings, pruning heads
etc.)
This model is also a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) subclass.
Use it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage
and behavior.
For best results finetuning SqueezeBERT on text classification tasks, it is recommended to use the
*squeezebert/squeezebert-mnli-headless* checkpoint as a starting point.
Parameters:
config ([`SqueezeBertConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Hierarchy:
```
Internal class hierarchy:
SqueezeBertModel
SqueezeBertEncoder
SqueezeBertModule
SqueezeBertSelfAttention
ConvActivation
ConvDropoutLayerNorm
```
Data layouts:
```
Input data is in [batch, sequence_length, hidden_size] format.
Data inside the encoder is in [batch, hidden_size, sequence_length] format. But, if `output_hidden_states == True`, the data from inside the encoder is returned in [batch, sequence_length, hidden_size] format.
The final output of the encoder is in [batch, sequence_length, hidden_size] format.
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/squeezebert.md | https://huggingface.co/docs/transformers/en/model_doc/squeezebert/#squeezebertforsequenceclassification | #squeezebertforsequenceclassification | .md | 173_9 |
SqueezeBERT Model with a multiple choice classification head on top (a linear layer on top of the pooled output and
a softmax) e.g. for RocStories/SWAG tasks.
The SqueezeBERT model was proposed in [SqueezeBERT: What can computer vision teach NLP about efficient neural
networks?](https://arxiv.org/abs/2006.11316) by Forrest N. Iandola, Albert E. Shaw, Ravi Krishna, and Kurt W.
Keutzer
This model inherits from [`PreTrainedModel`]. Check the superclass documentation for the generic methods the
library implements for all its model (such as downloading or saving, resizing the input embeddings, pruning heads
etc.)
This model is also a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) subclass.
Use it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage
and behavior.
For best results finetuning SqueezeBERT on text classification tasks, it is recommended to use the
*squeezebert/squeezebert-mnli-headless* checkpoint as a starting point.
Parameters:
config ([`SqueezeBertConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Hierarchy:
```
Internal class hierarchy:
SqueezeBertModel
SqueezeBertEncoder
SqueezeBertModule
SqueezeBertSelfAttention
ConvActivation
ConvDropoutLayerNorm
```
Data layouts:
```
Input data is in [batch, sequence_length, hidden_size] format.
Data inside the encoder is in [batch, hidden_size, sequence_length] format. But, if `output_hidden_states == True`, the data from inside the encoder is returned in [batch, sequence_length, hidden_size] format.
The final output of the encoder is in [batch, sequence_length, hidden_size] format.
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/squeezebert.md | https://huggingface.co/docs/transformers/en/model_doc/squeezebert/#squeezebertformultiplechoice | #squeezebertformultiplechoice | .md | 173_10 |
SqueezeBERT Model with a token classification head on top (a linear layer on top of the hidden-states output) e.g.
for Named-Entity-Recognition (NER) tasks.
The SqueezeBERT model was proposed in [SqueezeBERT: What can computer vision teach NLP about efficient neural
networks?](https://arxiv.org/abs/2006.11316) by Forrest N. Iandola, Albert E. Shaw, Ravi Krishna, and Kurt W.
Keutzer
This model inherits from [`PreTrainedModel`]. Check the superclass documentation for the generic methods the
library implements for all its model (such as downloading or saving, resizing the input embeddings, pruning heads
etc.)
This model is also a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) subclass.
Use it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage
and behavior.
For best results finetuning SqueezeBERT on text classification tasks, it is recommended to use the
*squeezebert/squeezebert-mnli-headless* checkpoint as a starting point.
Parameters:
config ([`SqueezeBertConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Hierarchy:
```
Internal class hierarchy:
SqueezeBertModel
SqueezeBertEncoder
SqueezeBertModule
SqueezeBertSelfAttention
ConvActivation
ConvDropoutLayerNorm
```
Data layouts:
```
Input data is in [batch, sequence_length, hidden_size] format.
Data inside the encoder is in [batch, hidden_size, sequence_length] format. But, if `output_hidden_states == True`, the data from inside the encoder is returned in [batch, sequence_length, hidden_size] format.
The final output of the encoder is in [batch, sequence_length, hidden_size] format.
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/squeezebert.md | https://huggingface.co/docs/transformers/en/model_doc/squeezebert/#squeezebertfortokenclassification | #squeezebertfortokenclassification | .md | 173_11 |
SqueezeBERT Model with a span classification head on top for extractive question-answering tasks like SQuAD (a
linear layers on top of the hidden-states output to compute `span start logits` and `span end logits`).
The SqueezeBERT model was proposed in [SqueezeBERT: What can computer vision teach NLP about efficient neural
networks?](https://arxiv.org/abs/2006.11316) by Forrest N. Iandola, Albert E. Shaw, Ravi Krishna, and Kurt W.
Keutzer
This model inherits from [`PreTrainedModel`]. Check the superclass documentation for the generic methods the
library implements for all its model (such as downloading or saving, resizing the input embeddings, pruning heads
etc.)
This model is also a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) subclass.
Use it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage
and behavior.
For best results finetuning SqueezeBERT on text classification tasks, it is recommended to use the
*squeezebert/squeezebert-mnli-headless* checkpoint as a starting point.
Parameters:
config ([`SqueezeBertConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Hierarchy:
```
Internal class hierarchy:
SqueezeBertModel
SqueezeBertEncoder
SqueezeBertModule
SqueezeBertSelfAttention
ConvActivation
ConvDropoutLayerNorm
```
Data layouts:
```
Input data is in [batch, sequence_length, hidden_size] format.
Data inside the encoder is in [batch, hidden_size, sequence_length] format. But, if `output_hidden_states == True`, the data from inside the encoder is returned in [batch, sequence_length, hidden_size] format.
The final output of the encoder is in [batch, sequence_length, hidden_size] format.
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/squeezebert.md | https://huggingface.co/docs/transformers/en/model_doc/squeezebert/#squeezebertforquestionanswering | #squeezebertforquestionanswering | .md | 173_12 |
<!--Copyright 2021 The HuggingFace 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
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Unless required by applicable law or agreed to in writing, software distributed under the License is distributed on
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--> | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/wav2vec2_phoneme.md | https://huggingface.co/docs/transformers/en/model_doc/wav2vec2_phoneme/ | .md | 174_0 |
|
The Wav2Vec2Phoneme model was proposed in [Simple and Effective Zero-shot Cross-lingual Phoneme Recognition (Xu et al.,
2021](https://arxiv.org/abs/2109.11680) by Qiantong Xu, Alexei Baevski, Michael Auli.
The abstract from the paper is the following:
*Recent progress in self-training, self-supervised pretraining and unsupervised learning enabled well performing speech
recognition systems without any labeled data. However, in many cases there is labeled data available for related
languages which is not utilized by these methods. This paper extends previous work on zero-shot cross-lingual transfer
learning by fine-tuning a multilingually pretrained wav2vec 2.0 model to transcribe unseen languages. This is done by
mapping phonemes of the training languages to the target language using articulatory features. Experiments show that
this simple method significantly outperforms prior work which introduced task-specific architectures and used only part
of a monolingually pretrained model.*
Relevant checkpoints can be found under https://huggingface.co/models?other=phoneme-recognition.
This model was contributed by [patrickvonplaten](https://huggingface.co/patrickvonplaten)
The original code can be found [here](https://github.com/pytorch/fairseq/tree/master/fairseq/models/wav2vec). | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/wav2vec2_phoneme.md | https://huggingface.co/docs/transformers/en/model_doc/wav2vec2_phoneme/#overview | #overview | .md | 174_1 |
- Wav2Vec2Phoneme uses the exact same architecture as Wav2Vec2
- Wav2Vec2Phoneme is a speech model that accepts a float array corresponding to the raw waveform of the speech signal.
- Wav2Vec2Phoneme model was trained using connectionist temporal classification (CTC) so the model output has to be
decoded using [`Wav2Vec2PhonemeCTCTokenizer`].
- Wav2Vec2Phoneme can be fine-tuned on multiple language at once and decode unseen languages in a single forward pass
to a sequence of phonemes
- By default, the model outputs a sequence of phonemes. In order to transform the phonemes to a sequence of words one
should make use of a dictionary and language model.
<Tip>
Wav2Vec2Phoneme's architecture is based on the Wav2Vec2 model, for API reference, check out [`Wav2Vec2`](wav2vec2)'s documentation page
except for the tokenizer.
</Tip> | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/wav2vec2_phoneme.md | https://huggingface.co/docs/transformers/en/model_doc/wav2vec2_phoneme/#usage-tips | #usage-tips | .md | 174_2 |
Constructs a Wav2Vec2PhonemeCTC tokenizer.
This tokenizer inherits from [`PreTrainedTokenizer`] which contains some of the main methods. Users should refer to
the superclass for more information regarding such methods.
Args:
vocab_file (`str`):
File containing the vocabulary.
bos_token (`str`, *optional*, defaults to `"<s>"`):
The beginning of sentence token.
eos_token (`str`, *optional*, defaults to `"</s>"`):
The end of sentence token.
unk_token (`str`, *optional*, defaults to `"<unk>"`):
The unknown token. A token that is not in the vocabulary cannot be converted to an ID and is set to be this
token instead.
pad_token (`str`, *optional*, defaults to `"<pad>"`):
The token used for padding, for example when batching sequences of different lengths.
do_phonemize (`bool`, *optional*, defaults to `True`):
Whether the tokenizer should phonetize the input or not. Only if a sequence of phonemes is passed to the
tokenizer, `do_phonemize` should be set to `False`.
phonemizer_lang (`str`, *optional*, defaults to `"en-us"`):
The language of the phoneme set to which the tokenizer should phonetize the input text to.
phonemizer_backend (`str`, *optional*. defaults to `"espeak"`):
The backend phonetization library that shall be used by the phonemizer library. Defaults to `espeak-ng`.
See the [phonemizer package](https://github.com/bootphon/phonemizer#readme). for more information.
**kwargs
Additional keyword arguments passed along to [`PreTrainedTokenizer`]
Methods: __call__
- batch_decode
- decode
- phonemize | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/wav2vec2_phoneme.md | https://huggingface.co/docs/transformers/en/model_doc/wav2vec2_phoneme/#wav2vec2phonemectctokenizer | #wav2vec2phonemectctokenizer | .md | 174_3 |
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Licensed under the Apache License, Version 2.0 (the "License"); you may not use this file except in compliance with
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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.
⚠️ Note that this file is in Markdown but contain specific syntax for our doc-builder (similar to MDX) that may not be
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--> | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/big_bird.md | https://huggingface.co/docs/transformers/en/model_doc/big_bird/ | .md | 175_0 |
|
The BigBird model was proposed in [Big Bird: Transformers for Longer Sequences](https://arxiv.org/abs/2007.14062) by
Zaheer, Manzil and Guruganesh, Guru and Dubey, Kumar Avinava and Ainslie, Joshua and Alberti, Chris and Ontanon,
Santiago and Pham, Philip and Ravula, Anirudh and Wang, Qifan and Yang, Li and others. BigBird, is a sparse-attention
based transformer which extends Transformer based models, such as BERT to much longer sequences. In addition to sparse
attention, BigBird also applies global attention as well as random attention to the input sequence. Theoretically, it
has been shown that applying sparse, global, and random attention approximates full attention, while being
computationally much more efficient for longer sequences. As a consequence of the capability to handle longer context,
BigBird has shown improved performance on various long document NLP tasks, such as question answering and
summarization, compared to BERT or RoBERTa.
The abstract from the paper is the following:
*Transformers-based models, such as BERT, have been one of the most successful deep learning models for NLP.
Unfortunately, one of their core limitations is the quadratic dependency (mainly in terms of memory) on the sequence
length due to their full attention mechanism. To remedy this, we propose, BigBird, a sparse attention mechanism that
reduces this quadratic dependency to linear. We show that BigBird is a universal approximator of sequence functions and
is Turing complete, thereby preserving these properties of the quadratic, full attention model. Along the way, our
theoretical analysis reveals some of the benefits of having O(1) global tokens (such as CLS), that attend to the entire
sequence as part of the sparse attention mechanism. The proposed sparse attention can handle sequences of length up to
8x of what was previously possible using similar hardware. As a consequence of the capability to handle longer context,
BigBird drastically improves performance on various NLP tasks such as question answering and summarization. We also
propose novel applications to genomics data.*
This model was contributed by [vasudevgupta](https://huggingface.co/vasudevgupta). The original code can be found
[here](https://github.com/google-research/bigbird). | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/big_bird.md | https://huggingface.co/docs/transformers/en/model_doc/big_bird/#overview | #overview | .md | 175_1 |
- For an in-detail explanation on how BigBird's attention works, see [this blog post](https://huggingface.co/blog/big-bird).
- BigBird comes with 2 implementations: **original_full** & **block_sparse**. For the sequence length < 1024, using
**original_full** is advised as there is no benefit in using **block_sparse** attention.
- The code currently uses window size of 3 blocks and 2 global blocks.
- Sequence length must be divisible by block size.
- Current implementation supports only **ITC**.
- Current implementation doesn't support **num_random_blocks = 0**
- BigBird is a model with absolute position embeddings so it's usually advised to pad the inputs on the right rather than
the left. | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/big_bird.md | https://huggingface.co/docs/transformers/en/model_doc/big_bird/#usage-tips | #usage-tips | .md | 175_2 |
- [Text classification task guide](../tasks/sequence_classification)
- [Token classification task guide](../tasks/token_classification)
- [Question answering task guide](../tasks/question_answering)
- [Causal language modeling task guide](../tasks/language_modeling)
- [Masked language modeling task guide](../tasks/masked_language_modeling)
- [Multiple choice task guide](../tasks/multiple_choice) | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/big_bird.md | https://huggingface.co/docs/transformers/en/model_doc/big_bird/#resources | #resources | .md | 175_3 |
This is the configuration class to store the configuration of a [`BigBirdModel`]. It is used to instantiate an
BigBird model according to the specified arguments, defining the model architecture. Instantiating a configuration
with the defaults will yield a similar configuration to that of the BigBird
[google/bigbird-roberta-base](https://huggingface.co/google/bigbird-roberta-base) architecture.
Configuration objects inherit from [`PretrainedConfig`] and can be used to control the model outputs. Read the
documentation from [`PretrainedConfig`] for more information.
Args:
vocab_size (`int`, *optional*, defaults to 50358):
Vocabulary size of the BigBird model. Defines the number of different tokens that can be represented by the
`inputs_ids` passed when calling [`BigBirdModel`].
hidden_size (`int`, *optional*, defaults to 768):
Dimension of the encoder layers and the pooler layer.
num_hidden_layers (`int`, *optional*, defaults to 12):
Number of hidden layers in the Transformer encoder.
num_attention_heads (`int`, *optional*, defaults to 12):
Number of attention heads for each attention layer in the Transformer encoder.
intermediate_size (`int`, *optional*, defaults to 3072):
Dimension of the "intermediate" (i.e., feed-forward) layer in the Transformer encoder.
hidden_act (`str` or `function`, *optional*, defaults to `"gelu_new"`):
The non-linear activation function (function or string) in the encoder and pooler. If string, `"gelu"`,
`"relu"`, `"selu"` and `"gelu_new"` are supported.
hidden_dropout_prob (`float`, *optional*, defaults to 0.1):
The dropout probability for all fully connected layers in the embeddings, encoder, and pooler.
attention_probs_dropout_prob (`float`, *optional*, defaults to 0.1):
The dropout ratio for the attention probabilities.
max_position_embeddings (`int`, *optional*, defaults to 4096):
The maximum sequence length that this model might ever be used with. Typically set this to something large
just in case (e.g., 1024 or 2048 or 4096).
type_vocab_size (`int`, *optional*, defaults to 2):
The vocabulary size of the `token_type_ids` passed when calling [`BigBirdModel`].
initializer_range (`float`, *optional*, defaults to 0.02):
The standard deviation of the truncated_normal_initializer for initializing all weight matrices.
layer_norm_eps (`float`, *optional*, defaults to 1e-12):
The epsilon used by the layer normalization layers.
is_decoder (`bool`, *optional*, defaults to `False`):
Whether the model is used as a decoder or not. If `False`, the model is used as an encoder.
use_cache (`bool`, *optional*, defaults to `True`):
Whether or not the model should return the last key/values attentions (not used by all models). Only
relevant if `config.is_decoder=True`.
attention_type (`str`, *optional*, defaults to `"block_sparse"`)
Whether to use block sparse attention (with n complexity) as introduced in paper or original attention
layer (with n^2 complexity). Possible values are `"original_full"` and `"block_sparse"`.
use_bias (`bool`, *optional*, defaults to `True`)
Whether to use bias in query, key, value.
rescale_embeddings (`bool`, *optional*, defaults to `False`)
Whether to rescale embeddings with (hidden_size ** 0.5).
block_size (`int`, *optional*, defaults to 64)
Size of each block. Useful only when `attention_type == "block_sparse"`.
num_random_blocks (`int`, *optional*, defaults to 3)
Each query is going to attend these many number of random blocks. Useful only when `attention_type ==
"block_sparse"`.
classifier_dropout (`float`, *optional*):
The dropout ratio for the classification head.
Example:
```python
>>> from transformers import BigBirdConfig, BigBirdModel
>>> # Initializing a BigBird google/bigbird-roberta-base style configuration
>>> configuration = BigBirdConfig()
>>> # Initializing a model (with random weights) from the google/bigbird-roberta-base style configuration
>>> model = BigBirdModel(configuration)
>>> # Accessing the model configuration
>>> configuration = model.config
``` | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/big_bird.md | https://huggingface.co/docs/transformers/en/model_doc/big_bird/#bigbirdconfig | #bigbirdconfig | .md | 175_4 |
Construct a BigBird tokenizer. Based on [SentencePiece](https://github.com/google/sentencepiece).
This tokenizer inherits from [`PreTrainedTokenizer`] which contains most of the main methods. Users should refer to
this superclass for more information regarding those methods.
Args:
vocab_file (`str`):
[SentencePiece](https://github.com/google/sentencepiece) file (generally has a *.spm* extension) that
contains the vocabulary necessary to instantiate a tokenizer.
unk_token (`str`, *optional*, defaults to `"<unk>"`):
The unknown token. A token that is not in the vocabulary cannot be converted to an ID and is set to be this
token instead.
bos_token (`str`, *optional*, defaults to `"<s>"`):
The begin of sequence token.
eos_token (`str`, *optional*, defaults to `"</s>"`):
The end of sequence token.
pad_token (`str`, *optional*, defaults to `"<pad>"`):
The token used for padding, for example when batching sequences of different lengths.
sep_token (`str`, *optional*, defaults to `"[SEP]"`):
The separator token, which is used when building a sequence from multiple sequences, e.g. two sequences for
sequence classification or for a text and a question for question answering. It is also used as the last
token of a sequence built with special tokens.
mask_token (`str`, *optional*, defaults to `"[MASK]"`):
The token used for masking values. This is the token used when training this model with masked language
modeling. This is the token which the model will try to predict.
cls_token (`str`, *optional*, defaults to `"[CLS]"`):
The classifier token which is used when doing sequence classification (classification of the whole sequence
instead of per-token classification). It is the first token of the sequence when built with special tokens.
sp_model_kwargs (`dict`, *optional*):
Will be passed to the `SentencePieceProcessor.__init__()` method. The [Python wrapper for
SentencePiece](https://github.com/google/sentencepiece/tree/master/python) can be used, among other things,
to set:
- `enable_sampling`: Enable subword regularization.
- `nbest_size`: Sampling parameters for unigram. Invalid for BPE-Dropout.
- `nbest_size = {0,1}`: No sampling is performed.
- `nbest_size > 1`: samples from the nbest_size results.
- `nbest_size < 0`: assuming that nbest_size is infinite and samples from the all hypothesis (lattice)
using forward-filtering-and-backward-sampling algorithm.
- `alpha`: Smoothing parameter for unigram sampling, and dropout probability of merge operations for
BPE-dropout.
Methods: build_inputs_with_special_tokens
- get_special_tokens_mask
- create_token_type_ids_from_sequences
- save_vocabulary | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/big_bird.md | https://huggingface.co/docs/transformers/en/model_doc/big_bird/#bigbirdtokenizer | #bigbirdtokenizer | .md | 175_5 |
Construct a "fast" BigBird tokenizer (backed by HuggingFace's *tokenizers* library). Based on
[Unigram](https://huggingface.co/docs/tokenizers/python/latest/components.html?highlight=unigram#models). This
tokenizer inherits from [`PreTrainedTokenizerFast`] which contains most of the main methods. Users should refer to
this superclass for more information regarding those methods
Args:
vocab_file (`str`):
[SentencePiece](https://github.com/google/sentencepiece) file (generally has a *.spm* extension) that
contains the vocabulary necessary to instantiate a tokenizer.
bos_token (`str`, *optional*, defaults to `"<s>"`):
The beginning of sequence token that was used during pretraining. Can be used a sequence classifier token.
<Tip>
When building a sequence using special tokens, this is not the token that is used for the beginning of
sequence. The token used is the `cls_token`.
</Tip>
eos_token (`str`, *optional*, defaults to `"</s>"`):
The end of sequence token. .. note:: When building a sequence using special tokens, this is not the token
that is used for the end of sequence. The token used is the `sep_token`.
unk_token (`str`, *optional*, defaults to `"<unk>"`):
The unknown token. A token that is not in the vocabulary cannot be converted to an ID and is set to be this
token instead.
sep_token (`str`, *optional*, defaults to `"[SEP]"`):
The separator token, which is used when building a sequence from multiple sequences, e.g. two sequences for
sequence classification or for a text and a question for question answering. It is also used as the last
token of a sequence built with special tokens.
pad_token (`str`, *optional*, defaults to `"<pad>"`):
The token used for padding, for example when batching sequences of different lengths.
cls_token (`str`, *optional*, defaults to `"[CLS]"`):
The classifier token which is used when doing sequence classification (classification of the whole sequence
instead of per-token classification). It is the first token of the sequence when built with special tokens.
mask_token (`str`, *optional*, defaults to `"[MASK]"`):
The token used for masking values. This is the token used when training this model with masked language
modeling. This is the token which the model will try to predict. | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/big_bird.md | https://huggingface.co/docs/transformers/en/model_doc/big_bird/#bigbirdtokenizerfast | #bigbirdtokenizerfast | .md | 175_6 |
models.big_bird.modeling_big_bird.BigBirdForPreTrainingOutput
Output type of [`BigBirdForPreTraining`].
Args:
loss (*optional*, returned when `labels` is provided, `torch.FloatTensor` of shape `(1,)`):
Total loss as the sum of the masked language modeling loss and the next sequence prediction
(classification) loss.
prediction_logits (`torch.FloatTensor` of shape `(batch_size, sequence_length, config.vocab_size)`):
Prediction scores of the language modeling head (scores for each vocabulary token before SoftMax).
seq_relationship_logits (`torch.FloatTensor` of shape `(batch_size, 2)`):
Prediction scores of the next sequence prediction (classification) head (scores of True/False continuation
before SoftMax).
hidden_states (`tuple(torch.FloatTensor)`, *optional*, returned when `output_hidden_states=True` is passed or when `config.output_hidden_states=True`):
Tuple of `torch.FloatTensor` (one for the output of the embeddings + one for the output of each layer) of
shape `(batch_size, sequence_length, hidden_size)`.
Hidden-states of the model at the output of each layer plus the initial embedding outputs.
attentions (`tuple(torch.FloatTensor)`, *optional*, returned when `output_attentions=True` is passed or when `config.output_attentions=True`):
Tuple of `torch.FloatTensor` (one for each layer) of shape `(batch_size, num_heads, sequence_length,
sequence_length)`.
Attentions weights after the attention softmax, used to compute the weighted average in the self-attention
heads.
<frameworkcontent>
<pt> | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/big_bird.md | https://huggingface.co/docs/transformers/en/model_doc/big_bird/#bigbird-specific-outputs | #bigbird-specific-outputs | .md | 175_7 |
The bare BigBird Model transformer outputting raw hidden-states without any specific head on top.
This model is a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) sub-class. Use
it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage and
behavior.
Parameters:
config ([`BigBirdConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
The model can behave as an encoder (with only self-attention) as well as a decoder, in which case a layer of
cross-attention is added between the self-attention layers, following the architecture described in [Attention is
all you need](https://arxiv.org/abs/1706.03762) by Ashish Vaswani, Noam Shazeer, Niki Parmar, Jakob Uszkoreit,
Llion Jones, Aidan N. Gomez, Lukasz Kaiser and Illia Polosukhin.
To behave as an decoder the model needs to be initialized with the `is_decoder` argument of the configuration set
to `True`. To be used in a Seq2Seq model, the model needs to initialized with both `is_decoder` argument and
`add_cross_attention` set to `True`; an `encoder_hidden_states` is then expected as an input to the forward pass.
Methods: forward | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/big_bird.md | https://huggingface.co/docs/transformers/en/model_doc/big_bird/#bigbirdmodel | #bigbirdmodel | .md | 175_8 |
No docstring available for BigBirdForPreTraining
Methods: forward | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/big_bird.md | https://huggingface.co/docs/transformers/en/model_doc/big_bird/#bigbirdforpretraining | #bigbirdforpretraining | .md | 175_9 |
BigBird Model with a `language modeling` head on top for CLM fine-tuning.
This model is a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) sub-class. Use
it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage and
behavior.
Parameters:
config ([`BigBirdConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Methods: forward | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/big_bird.md | https://huggingface.co/docs/transformers/en/model_doc/big_bird/#bigbirdforcausallm | #bigbirdforcausallm | .md | 175_10 |
BigBird Model with a `language modeling` head on top.
This model is a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) sub-class. Use
it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage and
behavior.
Parameters:
config ([`BigBirdConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Methods: forward | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/big_bird.md | https://huggingface.co/docs/transformers/en/model_doc/big_bird/#bigbirdformaskedlm | #bigbirdformaskedlm | .md | 175_11 |
BigBird Model transformer with a sequence classification/regression head on top (a linear layer on top of the
pooled output) e.g. for GLUE tasks.
This model is a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) sub-class. Use
it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage and
behavior.
Parameters:
config ([`BigBirdConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Methods: forward | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/big_bird.md | https://huggingface.co/docs/transformers/en/model_doc/big_bird/#bigbirdforsequenceclassification | #bigbirdforsequenceclassification | .md | 175_12 |
BigBird Model with a multiple choice classification head on top (a linear layer on top of the pooled output and a
softmax) e.g. for RocStories/SWAG tasks.
This model is a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) sub-class. Use
it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage and
behavior.
Parameters:
config ([`BigBirdConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Methods: forward | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/big_bird.md | https://huggingface.co/docs/transformers/en/model_doc/big_bird/#bigbirdformultiplechoice | #bigbirdformultiplechoice | .md | 175_13 |
BigBird Model with a token classification head on top (a linear layer on top of the hidden-states output) e.g. for
Named-Entity-Recognition (NER) tasks.
This model is a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) sub-class. Use
it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage and
behavior.
Parameters:
config ([`BigBirdConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Methods: forward | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/big_bird.md | https://huggingface.co/docs/transformers/en/model_doc/big_bird/#bigbirdfortokenclassification | #bigbirdfortokenclassification | .md | 175_14 |
BigBird Model with a span classification head on top for extractive question-answering tasks like SQuAD (a linear
layers on top of the hidden-states output to compute `span start logits` and `span end logits`).
This model is a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) sub-class. Use
it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage and
behavior.
Parameters:
config ([`BigBirdConfig`]): Model configuration class with all the parameters of the model.
Initializing with a config file does not load the weights associated with the model, only the
configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights.
Methods: forward
</pt>
<jax> | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/big_bird.md | https://huggingface.co/docs/transformers/en/model_doc/big_bird/#bigbirdforquestionanswering | #bigbirdforquestionanswering | .md | 175_15 |
No docstring available for FlaxBigBirdModel
Methods: __call__ | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/big_bird.md | https://huggingface.co/docs/transformers/en/model_doc/big_bird/#flaxbigbirdmodel | #flaxbigbirdmodel | .md | 175_16 |
No docstring available for FlaxBigBirdForPreTraining
Methods: __call__ | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/big_bird.md | https://huggingface.co/docs/transformers/en/model_doc/big_bird/#flaxbigbirdforpretraining | #flaxbigbirdforpretraining | .md | 175_17 |
No docstring available for FlaxBigBirdForCausalLM
Methods: __call__ | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/big_bird.md | https://huggingface.co/docs/transformers/en/model_doc/big_bird/#flaxbigbirdforcausallm | #flaxbigbirdforcausallm | .md | 175_18 |
No docstring available for FlaxBigBirdForMaskedLM
Methods: __call__ | /Users/nielsrogge/Documents/python_projecten/transformers/docs/source/en/model_doc/big_bird.md | https://huggingface.co/docs/transformers/en/model_doc/big_bird/#flaxbigbirdformaskedlm | #flaxbigbirdformaskedlm | .md | 175_19 |
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