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from typing import Optional, Union |
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import numpy as np |
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import torch |
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import transformers |
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from .ultravox_config import UltravoxConfig |
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class UltravoxProcessor(transformers.ProcessorMixin): |
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""" |
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Constructs an Ultravox processor which wraps an audio processor and a tokenizer into a single processor. |
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Args: |
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audio_processor: The audio processor for the audio encoder. |
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tokenizer: The tokenizer for the language model. |
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""" |
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attributes = ["audio_processor", "tokenizer"] |
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audio_processor_class = ( |
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"Wav2Vec2Processor", |
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"SeamlessM4TFeatureExtractor", |
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"WhisperProcessor", |
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) |
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tokenizer_class = ( |
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"PreTrainedTokenizer", |
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"PreTrainedTokenizerFast", |
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) |
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tokenizer: transformers.PreTrainedTokenizerBase |
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audio_processor: transformers.ProcessorMixin |
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def __init__( |
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self, |
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audio_processor=None, |
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tokenizer=None, |
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audio_padding: str = "longest", |
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encoder_ds_factor: int = 320, |
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stack_factor: int = 8, |
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audio_placeholder: str = "<|audio|>", |
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): |
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""" |
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Args: |
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audio_processor: The audio processor for the audio encoder. |
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tokenizer: The tokenizer for the language model. |
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audio_padding: The padding strategy for the audio encoder. |
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encoder_ds_factor: The downsample factor of the audio encoder. |
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stack_factor: The factor by which the audio encoder output is stacked in the multimodal projector. |
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audio_placeholder: The placeholder for the audio in the text. |
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""" |
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self.audio_padding = audio_padding |
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self.encoder_ds_factor = encoder_ds_factor |
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self.stack_factor = stack_factor |
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self.audio_placeholder = audio_placeholder |
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self.audio_token_replacement = tokenizer.eos_token |
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assert ( |
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self.audio_token_replacement is not None |
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), "The tokenizer has no EOS token. Cannot recover." |
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if tokenizer.pad_token_id is None: |
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tokenizer.pad_token_id = tokenizer.eos_token_id |
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super().__init__(audio_processor=audio_processor, tokenizer=tokenizer) |
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@classmethod |
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def from_pretrained(cls, pretrained_model_name_or_path: str, **kwargs): |
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config: UltravoxConfig = transformers.AutoConfig.from_pretrained( |
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pretrained_model_name_or_path, **kwargs |
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) |
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audio_processor = transformers.AutoProcessor.from_pretrained( |
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config.audio_model_id |
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or config.audio_config._name_or_path |
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or "facebook/wav2vec2-base-960h" |
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) |
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tokenizer = transformers.AutoTokenizer.from_pretrained( |
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pretrained_model_name_or_path, **kwargs |
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) |
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tokenizer.padding_side = "left" |
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tokenizer.pad_token = tokenizer.eos_token |
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return cls( |
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audio_processor=audio_processor, |
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tokenizer=tokenizer, |
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stack_factor=config.stack_factor, |
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) |
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def __call__( |
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self, |
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text: Optional[str] = None, |
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audio: Optional[Union[np.ndarray, torch.Tensor]] = None, |
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sampling_rate: Optional[int] = None, |
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return_tensors: Optional[ |
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Union[str, transformers.TensorType] |
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] = transformers.TensorType.PYTORCH, |
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**kwargs, |
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) -> transformers.BatchFeature: |
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""" |
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Main method to prepare for the model one text sequence and audio. This method forwards the `text` |
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and `kwargs` arguments to PreTrainedTokenizerFast's [`~PreTrainedTokenizerFast.__call__`] if `text` is not `None` to encode |
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the text. To prepare the audio(s), this method forwards the `audio`, `sampling_rate` and `kwargs` arguments to |
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audio processor's [`~Wav2Vec2Processor.__call__`] if `audio` is not `None`. Please refer to the docstring |
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of the above two methods for more information. |
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Args: |
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text (`str`, `List[str]`): |
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The sequence to be encoded. Sequence can be a string or (pretokenized string). |
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audio (`np.ndarray`, `torch.Tensor`, `List[np.ndarray]`, `List[torch.Tensor]`): |
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The audio to be prepared. Audio can be NumPy array or PyTorch tensor. In case of a |
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NumPy array/PyTorch tensor, each audio should be of shape (C, T), where C is a number of channels, and T the |
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sample length of the audio. |
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sampling_rate (`int`, *optional*, defaults to 16000): |
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Sampling rate of the input audio. We expect 16kHz audio. Don't change this value unless you know what |
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you are doing. |
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return_tensors (`str` or [`~utils.TensorType`], *optional*): |
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If set, will return tensors of a particular framework. Acceptable values are: |
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- `'tf'`: Return TensorFlow `tf.constant` objects. |
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- `'pt'`: Return PyTorch `torch.Tensor` objects. |
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- `'np'`: Return NumPy `np.ndarray` objects. |
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- `'jax'`: Return JAX `jnp.ndarray` objects. |
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Returns: |
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[`BatchFeature`]: A [`BatchFeature`] with the following fields: |
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- **input_ids** -- List of token ids to be fed to a model. Returned when `text` is not `None`. |
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- **attention_mask** -- List of indices specifying which tokens should be attended to by the model (when |
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`return_attention_mask=True` or if *"attention_mask"* is in `self.model_input_names` and if `text` is not |
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`None`). |
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- **audio_values** -- Processed audio values to be fed to a model. Returned when `audio` is not `None`. |
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- **audio_token_len** -- Predicted number of audio frames: this value is guaranteed to be a close upper bound. |
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Returned when `audio` is not `None`. |
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- **audio_token_start_idx** -- The index in the tokenized text where the audio starts. Returned when `audio` is not `None`. |
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""" |
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data = {} |
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if audio is not None and len(audio) > 0: |
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x = self.audio_processor( |
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audio, |
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sampling_rate=sampling_rate, |
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padding="longest", |
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return_attention_mask=True, |
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**kwargs, |
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) |
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if "input_features" in x: |
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data["audio_values"] = x.input_features |
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else: |
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data["audio_values"] = x.input_values |
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data["audio_len"] = x.attention_mask.sum(-1) - 1 |
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def cnn_out_len(in_len, kernel, stride=1, padding=1, dilation=1): |
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return np.floor((in_len + (2*padding) - (dilation * (kernel - 1)) - 1)/stride + 1) |
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def stack_frame_len(T): |
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T_pad = (T + self.stack_factor - 1) // self.stack_factor * self.stack_factor |
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return int((T_pad + self.stack_factor) // self.stack_factor) |
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nb_encoder_frames = [cnn_out_len(cnn_out_len(feat_len, kernel=3), kernel=3, stride=2) for feat_len in data["audio_len"]] |
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data["audio_token_len"] = [stack_frame_len(x) for x in nb_encoder_frames] |
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if text is not None: |
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assert isinstance( |
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text, list |
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), "Text must be a list." |
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processed_text = [] |
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data["audio_token_start_idx"] = [] |
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for i, t in enumerate(text): |
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assert self.audio_placeholder in t |
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if "audio_token_len" not in data: |
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raise ValueError( |
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f"audio must be provided when using audio placeholder ({self.audio_placeholder}) in text." |
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) |
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start_idx = len( |
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self.tokenizer.encode( |
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t.split(self.audio_placeholder)[0], |
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add_special_tokens=False, |
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) |
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) |
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data["audio_token_start_idx"].append(start_idx) |
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t = t.replace( |
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self.audio_placeholder, |
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self.audio_token_replacement * data["audio_token_len"][i], |
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) |
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processed_text.append(t) |
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data.update(self.tokenizer(processed_text, add_special_tokens=False, padding='longest', **kwargs)) |
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return transformers.BatchFeature(data=data, tensor_type=return_tensors) |
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def batch_decode(self, *args, **kwargs): |
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return self.tokenizer.batch_decode(*args, **kwargs) |
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def decode(self, *args, **kwargs): |
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return self.tokenizer.decode(*args, **kwargs) |
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@property |
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def model_input_names(self): |
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tokenizer_input_names = self.tokenizer.model_input_names |
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audio_processor_input_names = self.audio_processor.model_input_names |
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return list(set(tokenizer_input_names + audio_processor_input_names)) |
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UltravoxProcessor.register_for_auto_class() |
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transformers.AutoProcessor.register(UltravoxConfig, UltravoxProcessor) |
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