692 lines
No EOL
29 KiB
Python
692 lines
No EOL
29 KiB
Python
import math
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import warnings
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from typing import List, Optional, Union, Dict, Any, Tuple
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import os
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import re
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import numpy as np
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import torch
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from transformers.tokenization_utils_base import BatchEncoding, PaddingStrategy, PreTokenizedInput, TextInput, TruncationStrategy
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from transformers.utils import TensorType, logging
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from .vibevoice_tokenizer_processor import AudioNormalizer
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logger = logging.get_logger(__name__)
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class VibeVoiceProcessor:
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r"""
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Constructs a VibeVoice processor which wraps a VibeVoice tokenizer and audio processor into a single processor.
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[`VibeVoiceProcessor`] offers all the functionalities of [`VibeVoiceTokenizer`] and [`VibeVoiceTokenizerProcessor`].
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See the [`~VibeVoiceProcessor.__call__`] and [`~VibeVoiceProcessor.decode`] for more information.
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Args:
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tokenizer (`VibeVoiceTextTokenizer` or `VibeVoiceTextTokenizerFast`):
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The tokenizer for text processing.
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audio_processor (`VibeVoiceTokenizerProcessor`):
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The audio processor for speech processing.
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speech_tok_compress_ratio (`int`, *optional*, defaults to 3200):
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The compression ratio for speech tokenization.
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db_normalize (`bool`, *optional*, defaults to True):
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Whether to apply decibel normalization to audio inputs.
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"""
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def __init__(self, tokenizer=None, audio_processor=None, speech_tok_compress_ratio=3200, db_normalize=True, **kwargs):
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self.tokenizer = tokenizer
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self.audio_processor = audio_processor
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self.speech_tok_compress_ratio = speech_tok_compress_ratio
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self.db_normalize = db_normalize
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self.audio_normalizer = AudioNormalizer() if db_normalize else None
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self.system_prompt = " Transform the text provided by various speakers into speech output, utilizing the distinct voice of each respective speaker.\n"
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@classmethod
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def from_pretrained(cls, pretrained_model_name_or_path, **kwargs):
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"""
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Instantiate a VibeVoiceProcessor from a pretrained VibeVoice processor.
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Args:
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pretrained_model_name_or_path (`str` or `os.PathLike`):
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This can be either:
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- a string, the *model id* of a pretrained model
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- a path to a *directory* containing processor config
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Returns:
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[`VibeVoiceProcessor`]: The processor object instantiated from pretrained model.
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"""
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import os
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import json
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from transformers.utils import cached_file
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from .vibevoice_tokenizer_processor import VibeVoiceTokenizerProcessor
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from vibevoice.modular.modular_vibevoice_text_tokenizer import (
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VibeVoiceTextTokenizer,
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VibeVoiceTextTokenizerFast
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)
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# Try to load from local path first, then from HF hub
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config_path = os.path.join(pretrained_model_name_or_path, "preprocessor_config.json")
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config = None
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if os.path.exists(config_path):
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# Local path exists
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with open(config_path, 'r') as f:
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config = json.load(f)
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else:
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# Try to load from HF hub
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try:
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config_file = cached_file(
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pretrained_model_name_or_path,
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"preprocessor_config.json",
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**kwargs
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)
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with open(config_file, 'r') as f:
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config = json.load(f)
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except Exception as e:
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logger.warning(f"Could not load preprocessor_config.json from {pretrained_model_name_or_path}: {e}")
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logger.warning("Using default configuration")
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config = {
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"speech_tok_compress_ratio": 3200,
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"db_normalize": True,
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}
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# Extract main processor parameters
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speech_tok_compress_ratio = config.get("speech_tok_compress_ratio", 3200)
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db_normalize = config.get("db_normalize", True)
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# Load tokenizer - try from model path first, then fallback to Qwen
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language_model_pretrained_name = config.get("language_model_pretrained_name", None) or kwargs.pop("language_model_pretrained_name", "Qwen/Qwen2.5-1.5B")
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logger.info(f"Loading tokenizer from {language_model_pretrained_name}")
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if 'qwen' in language_model_pretrained_name.lower():
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tokenizer = VibeVoiceTextTokenizerFast.from_pretrained(
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language_model_pretrained_name,
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**kwargs
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)
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else:
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raise ValueError(f"Unsupported tokenizer type for {language_model_pretrained_name}. Supported types: Qwen, Llama, Gemma.")
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# Load audio processor
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if "audio_processor" in config:
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# Create audio processor from config
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audio_config = config["audio_processor"]
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audio_processor = VibeVoiceTokenizerProcessor(
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sampling_rate=audio_config.get("sampling_rate", 24000),
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normalize_audio=audio_config.get("normalize_audio", True),
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target_dB_FS=audio_config.get("target_dB_FS", -25),
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eps=audio_config.get("eps", 1e-6),
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)
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else:
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# Create default audio processor
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audio_processor = VibeVoiceTokenizerProcessor()
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# Create and return the processor
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return cls(
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tokenizer=tokenizer,
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audio_processor=audio_processor,
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speech_tok_compress_ratio=speech_tok_compress_ratio,
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db_normalize=db_normalize,
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)
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def save_pretrained(self, save_directory: Union[str, os.PathLike], **kwargs):
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"""
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Save a processor to a directory, so that it can be re-loaded using the
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[`~VibeVoiceProcessor.from_pretrained`] class method.
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Args:
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save_directory (`str` or `os.PathLike`):
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Directory where the processor will be saved.
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"""
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import os
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import json
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os.makedirs(save_directory, exist_ok=True)
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# Save processor configuration
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processor_config = {
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"processor_class": "VibeVoiceProcessor",
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"speech_tok_compress_ratio": self.speech_tok_compress_ratio,
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"db_normalize": self.db_normalize,
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"audio_processor": {
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"feature_extractor_type": "VibeVoiceTokenizerProcessor",
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"sampling_rate": getattr(self.audio_processor, 'sampling_rate', 24000),
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"normalize_audio": getattr(self.audio_processor, 'normalize_audio', True),
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"target_dB_FS": getattr(self.audio_processor, 'target_dB_FS', -25),
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"eps": getattr(self.audio_processor, 'eps', 1e-6),
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}
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}
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config_path = os.path.join(save_directory, "preprocessor_config.json")
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with open(config_path, 'w') as f:
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json.dump(processor_config, f, indent=2)
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logger.info(f"Processor configuration saved in {config_path}")
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def __call__(
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self,
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text: Optional[Union[str, List[str], TextInput, PreTokenizedInput, List[TextInput], List[PreTokenizedInput]]] = None,
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voice_samples: Optional[Union[List[Union[str, np.ndarray]], List[List[Union[str, np.ndarray]]]]] = None,
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padding: Union[bool, str, PaddingStrategy] = True,
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truncation: Union[bool, str, TruncationStrategy] = False,
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max_length: Optional[int] = None,
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return_tensors: Optional[Union[str, TensorType]] = None,
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return_attention_mask: bool = True,
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**kwargs,
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) -> BatchEncoding:
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"""
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Main method to process one or more podcast scripts with optional voice samples.
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Args:
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text (`str`, `List[str]`):
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The input text(s) to process. Can be:
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- A single script string
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- A list of script strings for batch processing
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- A path to a .json or .txt file
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- A list of paths
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voice_samples (`List[Union[str, np.ndarray]]`, `List[List[Union[str, np.ndarray]]]`, *optional*):
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Voice samples for each script. Can be:
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- A list of samples for a single script
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- A list of lists for batch processing
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padding (`bool`, `str` or `PaddingStrategy`, defaults to `True`):
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Whether to pad sequences to the same length
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truncation (`bool`, `str` or `TruncationStrategy`, defaults to `False`):
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Whether to truncate sequences
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max_length (`int`, *optional*):
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Maximum length of the returned sequences
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return_tensors (`str` or `TensorType`, *optional*):
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If set, will return tensors of a particular framework
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return_attention_mask (`bool`, defaults to `True`):
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Whether to return the attention mask
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Returns:
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`BatchEncoding`: A BatchEncoding with the following fields:
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- **input_ids** -- List of token id sequences or tensor
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- **attention_mask** -- List of attention masks or tensor
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- **speech_tensors** -- Padded speech inputs (if voice_samples provided)
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- **speech_masks** -- Speech masks (if voice_samples provided)
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- **speech_input_mask** -- Boolean masks indicating speech token positions
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"""
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# Handle single vs batch input
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if isinstance(text, str) or (isinstance(text, list) and len(text) > 0 and not isinstance(text[0], str)):
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# Single input
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texts = [text]
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is_batched = False
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else:
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# Batch input
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texts = text
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is_batched = True
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# Handle voice samples
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if voice_samples is not None:
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if not is_batched or (isinstance(voice_samples[0], (str, np.ndarray))):
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# Single set of voice samples
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voice_samples_list = [voice_samples]
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else:
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# Batch of voice samples
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voice_samples_list = voice_samples
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else:
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voice_samples_list = [None] * len(texts)
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# Process each input
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all_encodings = []
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for text_input, voice_input in zip(texts, voice_samples_list):
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encoding = self._process_single(text_input, voice_input)
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all_encodings.append(encoding)
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# Combine batch
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batch_encoding = self._batch_encode(
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all_encodings,
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padding=padding,
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truncation=truncation,
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max_length=max_length,
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return_tensors=return_tensors,
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return_attention_mask=return_attention_mask,
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)
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return batch_encoding
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def _process_single(
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self,
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text: Union[str, TextInput],
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voice_samples: Optional[List[Union[str, np.ndarray]]] = None,
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) -> Dict[str, Any]:
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"""Process a single podcast script."""
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# Determine if text is a file path or direct script
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script = None
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if isinstance(text, str):
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# Check if it's a file path
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if text.endswith('.json') and os.path.exists(text):
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script = self._convert_json_to_script(text)
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elif text.endswith('.txt') and os.path.exists(text):
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script = self._convert_text_to_script(text)
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else:
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# Assume it's the script content directly
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script = text
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if script is None:
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raise ValueError(f"Could not process input text: {text}")
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# Parse the script
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parsed_lines = self._parse_script(script)
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all_speakers = list(set(speaker_id for speaker_id, _ in parsed_lines))
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# Create system prompt
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# system_tokens = self.tokenizer.encode(self.system_prompt, add_special_tokens=False)
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system_tokens = self.tokenizer.encode(self.system_prompt)
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# Process voice samples if provided
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if voice_samples:
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voice_tokens, voice_speech_inputs, voice_speech_masks = self._create_voice_prompt(voice_samples[:len(all_speakers)])
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else:
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voice_tokens, voice_speech_inputs, voice_speech_masks = [], [], []
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# Build full token sequence
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full_tokens = system_tokens + voice_tokens
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speech_input_mask = [False] * len(system_tokens) + voice_speech_masks
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# Add text input section
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full_tokens += self.tokenizer.encode(' Text input:\n', add_special_tokens=False)
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speech_input_mask += [False] * len(self.tokenizer.encode(' Text input:\n', add_special_tokens=False))
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for speaker_id, speaker_text in parsed_lines:
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speaker_text_tokens = self.tokenizer.encode(f" Speaker {speaker_id}:{speaker_text}\n", add_special_tokens=False)
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full_tokens += speaker_text_tokens
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speech_input_mask += [False] * len(speaker_text_tokens)
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# Add speech output section
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full_tokens += self.tokenizer.encode(' Speech output:\n', add_special_tokens=False) + [self.tokenizer.speech_start_id]
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speech_input_mask += [False] * (len(self.tokenizer.encode(' Speech output:\n', add_special_tokens=False)) + 1)
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return {
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"input_ids": full_tokens,
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"speech_inputs": voice_speech_inputs if voice_speech_inputs else None,
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"speech_input_mask": speech_input_mask,
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"parsed_script": parsed_lines,
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"all_speakers": all_speakers,
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}
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def _batch_encode(
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self,
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encodings: List[Dict[str, Any]],
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padding: Union[bool, str, PaddingStrategy] = True,
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truncation: Union[bool, str, TruncationStrategy] = False,
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max_length: Optional[int] = None,
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return_tensors: Optional[Union[str, TensorType]] = None,
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return_attention_mask: bool = True,
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) -> BatchEncoding:
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"""Combine multiple encodings into a batch with padding."""
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# Extract input_ids and create attention_mask
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input_ids_list = [enc["input_ids"] for enc in encodings]
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speech_input_masks_list = [enc["speech_input_mask"] for enc in encodings]
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# Determine padding strategy
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if isinstance(padding, bool):
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padding_strategy = PaddingStrategy.LONGEST if padding else PaddingStrategy.DO_NOT_PAD
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elif isinstance(padding, str):
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padding_strategy = PaddingStrategy(padding)
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else:
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padding_strategy = padding
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# Apply padding to input_ids
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if padding_strategy != PaddingStrategy.DO_NOT_PAD:
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if padding_strategy == PaddingStrategy.LONGEST:
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max_len = max(len(ids) for ids in input_ids_list)
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elif padding_strategy == PaddingStrategy.MAX_LENGTH and max_length is not None:
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max_len = max_length
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else:
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max_len = max(len(ids) for ids in input_ids_list)
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# Pad sequences
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padded_input_ids = []
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attention_masks = []
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padded_speech_input_masks = []
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for input_ids, speech_mask in zip(input_ids_list, speech_input_masks_list):
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# Truncate if needed
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if truncation and len(input_ids) > max_len:
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input_ids = input_ids[:max_len]
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speech_mask = speech_mask[:max_len]
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# Pad
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padding_length = max_len - len(input_ids)
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# padded_ids = [self.tokenizer.pad_token_id] * padding_length + input_ids
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padded_ids = [self.tokenizer.pad_id] * padding_length + input_ids
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attention_mask = [0] * padding_length + [1] * len(input_ids)
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padded_speech_mask = [False] * padding_length + speech_mask
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padded_input_ids.append(padded_ids)
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attention_masks.append(attention_mask)
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padded_speech_input_masks.append(padded_speech_mask)
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input_ids_list = padded_input_ids
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speech_input_masks_list = padded_speech_input_masks
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else:
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# No padding, just create attention masks
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attention_masks = [[1] * len(ids) for ids in input_ids_list] if return_attention_mask else None
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# Process speech inputs
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all_speech_inputs = []
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has_speech = False
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for enc in encodings:
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if enc["speech_inputs"] is not None:
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all_speech_inputs.extend(enc["speech_inputs"])
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has_speech = True
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# Prepare batch encoding
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batch_encoding = BatchEncoding()
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# Handle tensor conversion
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if return_tensors is not None:
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batch_encoding["input_ids"] = torch.tensor(input_ids_list, dtype=torch.long)
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if return_attention_mask and attention_masks is not None:
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batch_encoding["attention_mask"] = torch.tensor(attention_masks, dtype=torch.long)
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batch_encoding["speech_input_mask"] = torch.tensor(speech_input_masks_list, dtype=torch.bool)
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else:
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batch_encoding["input_ids"] = input_ids_list
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if return_attention_mask and attention_masks is not None:
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batch_encoding["attention_mask"] = attention_masks
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batch_encoding["speech_input_mask"] = speech_input_masks_list
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# Process speech tensors if present
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if has_speech:
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speech_dict = self.prepare_speech_inputs(
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all_speech_inputs,
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return_tensors=return_tensors,
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)
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batch_encoding["speech_tensors"] = speech_dict["padded_speeches"]
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batch_encoding["speech_masks"] = speech_dict["speech_masks"]
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else:
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batch_encoding["speech_tensors"] = None
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batch_encoding["speech_masks"] = None
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# Add metadata
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batch_encoding["parsed_scripts"] = [enc["parsed_script"] for enc in encodings]
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batch_encoding["all_speakers_list"] = [enc["all_speakers"] for enc in encodings]
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return batch_encoding
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def _create_voice_prompt(
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self,
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speaker_samples: List[Union[str, np.ndarray]]
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) -> Tuple[List[int], List[np.ndarray], List[bool]]:
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"""
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Create voice prompt tokens and process audio samples.
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Returns:
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tuple: (voice_tokens, voice_speech_inputs, voice_speech_masks)
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"""
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vae_token_id = self.tokenizer.speech_diffusion_id
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voice_full_tokens = self.tokenizer.encode(' Voice input:\n', add_special_tokens=False)
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voice_speech_inputs = []
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voice_speech_masks = [False] * len(voice_full_tokens)
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for speaker_id, speaker_audio in enumerate(speaker_samples):
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prefix_tokens = self.tokenizer.encode(f" Speaker {speaker_id}:", add_special_tokens=False)
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# Process audio
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if isinstance(speaker_audio, str):
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# Load audio from file
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wav = self.audio_processor._load_audio_from_path(speaker_audio)
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else:
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wav = np.array(speaker_audio, dtype=np.float32)
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# Apply normalization if needed
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if self.db_normalize and self.audio_normalizer:
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wav = self.audio_normalizer(wav)
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# Calculate token length based on compression ratio
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# if speaker_audio.endswith('.pt') or speaker_audio.endswith('.npy'):
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# vae_tok_len = wav.shape[0]
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# else:
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vae_tok_len = math.ceil(wav.shape[0] / self.speech_tok_compress_ratio)
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# Build tokens and masks
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speaker_tokens = (prefix_tokens +
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[self.tokenizer.speech_start_id] +
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[vae_token_id] * vae_tok_len +
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[self.tokenizer.speech_end_id] +
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self.tokenizer.encode('\n', add_special_tokens=False))
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vae_input_mask = ([False] * len(prefix_tokens) +
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[False] +
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[True] * vae_tok_len +
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[False] +
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[False])
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voice_full_tokens.extend(speaker_tokens)
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voice_speech_masks.extend(vae_input_mask)
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voice_speech_inputs.append(wav)
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return voice_full_tokens, voice_speech_inputs, voice_speech_masks
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def prepare_speech_inputs(
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self,
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speech_inputs: List[np.ndarray],
|
|
return_tensors: Optional[Union[str, TensorType]] = None,
|
|
device: Optional[Union[str, torch.device]] = None,
|
|
dtype: Optional[torch.dtype] = None,
|
|
) -> Dict[str, Any]:
|
|
"""
|
|
Prepare speech inputs for model consumption.
|
|
|
|
Args:
|
|
speech_inputs: List of speech arrays
|
|
return_tensors: Output tensor type
|
|
device: Device to place tensors on
|
|
dtype: Data type for tensors
|
|
|
|
Returns:
|
|
Dictionary with padded_speeches and speech_masks
|
|
"""
|
|
if not speech_inputs:
|
|
return {"padded_speeches": None, "speech_masks": None}
|
|
|
|
# Calculate sequence lengths
|
|
vae_tok_seqlens = [math.ceil(s.shape[0] / self.speech_tok_compress_ratio) for s in speech_inputs]
|
|
# vae_tok_seqlens = [math.ceil(s.shape[0] / self.speech_tok_compress_ratio) if s.ndim == 1 else s.shape[0] for s in speech_inputs]
|
|
max_speech_length = max(s.shape[0] for s in speech_inputs)
|
|
|
|
# Pad speeches
|
|
if speech_inputs[0].ndim == 1:
|
|
padded_speeches = np.full((len(speech_inputs), max_speech_length), fill_value=0, dtype=np.float32)
|
|
else:
|
|
padded_speeches = np.full((len(speech_inputs), max_speech_length, speech_inputs[0].shape[-1]), fill_value=0, dtype=np.float32)
|
|
speech_masks = np.zeros((len(speech_inputs), max(vae_tok_seqlens)), dtype=np.bool_)
|
|
|
|
for i, (speech, vae_tok_length) in enumerate(zip(speech_inputs, vae_tok_seqlens)):
|
|
padded_speeches[i, :len(speech)] = speech
|
|
speech_masks[i, :vae_tok_length] = True
|
|
|
|
result = {
|
|
"padded_speeches": padded_speeches,
|
|
"speech_masks": speech_masks,
|
|
}
|
|
|
|
# Convert to tensors if requested
|
|
if return_tensors == "pt":
|
|
result["padded_speeches"] = torch.tensor(padded_speeches, device=device, dtype=dtype or torch.float32)
|
|
result["speech_masks"] = torch.tensor(speech_masks, device=device, dtype=torch.bool)
|
|
|
|
return result
|
|
|
|
def _convert_json_to_script(self, json_file: str) -> str:
|
|
"""
|
|
Convert JSON format to script format.
|
|
Expected JSON format:
|
|
[
|
|
{"speaker": "1", "text": "Hello everyone..."},
|
|
{"speaker": "2", "text": "Great to be here..."}
|
|
]
|
|
"""
|
|
import json
|
|
|
|
with open(json_file, 'r', encoding='utf-8') as f:
|
|
data = json.load(f)
|
|
|
|
if not isinstance(data, list):
|
|
raise ValueError("JSON file must contain a list of speaker entries")
|
|
|
|
script_lines = []
|
|
for item in data:
|
|
if not isinstance(item, dict):
|
|
logger.warning(f"Skipping non-dict entry: {item}")
|
|
continue
|
|
|
|
speaker = item.get('speaker')
|
|
text = item.get('text')
|
|
|
|
if speaker is None or text is None:
|
|
logger.warning(f"Skipping entry missing speaker or text: {item}")
|
|
continue
|
|
|
|
# Ensure speaker ID is valid
|
|
try:
|
|
speaker_id = int(speaker)
|
|
except (ValueError, TypeError):
|
|
logger.warning(f"Invalid speaker ID: {speaker}, skipping entry")
|
|
continue
|
|
|
|
# Clean up text
|
|
text = text.strip()
|
|
if text:
|
|
script_lines.append(f"Speaker {speaker_id}: {text}")
|
|
|
|
if not script_lines:
|
|
raise ValueError("No valid entries found in JSON file")
|
|
|
|
return "\n".join(script_lines)
|
|
|
|
def _convert_text_to_script(self, text_file: str) -> str:
|
|
"""
|
|
Convert text file to script format.
|
|
Handles multiple formats:
|
|
1. Already formatted as "Speaker X: text"
|
|
2. Plain text (assigns to Speaker 1)
|
|
|
|
Handles edge cases like multiple colons in a line.
|
|
"""
|
|
with open(text_file, 'r', encoding='utf-8') as f:
|
|
lines = f.readlines()
|
|
|
|
script_lines = []
|
|
current_speaker = 1
|
|
|
|
for line in lines:
|
|
line = line.strip()
|
|
if not line:
|
|
continue
|
|
|
|
# Try to parse as "Speaker X: text" format
|
|
# Use regex to be more robust
|
|
speaker_match = re.match(r'^Speaker\s+(\d+)\s*:\s*(.*)$', line, re.IGNORECASE)
|
|
|
|
if speaker_match:
|
|
speaker_id = int(speaker_match.group(1))
|
|
text = speaker_match.group(2).strip()
|
|
if text:
|
|
script_lines.append(f"Speaker {speaker_id}: {text}")
|
|
else:
|
|
# Treat as plain text - assign to current speaker
|
|
script_lines.append(f"Speaker {current_speaker}: {line}")
|
|
|
|
if not script_lines:
|
|
raise ValueError("No valid content found in text file")
|
|
|
|
return "\n".join(script_lines)
|
|
|
|
def _parse_script(self, script: str) -> List[Tuple[int, str]]:
|
|
"""Parse script into list of (speaker_id, text) tuples."""
|
|
lines = script.strip().split("\n")
|
|
parsed_lines = []
|
|
speaker_ids = []
|
|
|
|
# First pass: parse all lines and collect speaker IDs
|
|
for line in lines:
|
|
if not line.strip():
|
|
continue
|
|
|
|
# Use regex to handle edge cases like multiple colons
|
|
match = re.match(r'^Speaker\s+(\d+)\s*:\s*(.*)$', line.strip(), re.IGNORECASE)
|
|
|
|
if match:
|
|
speaker_id = int(match.group(1))
|
|
text = ' ' + match.group(2).strip()
|
|
parsed_lines.append((speaker_id, text))
|
|
speaker_ids.append(speaker_id)
|
|
else:
|
|
logger.warning(f"Could not parse line: '{line}'")
|
|
|
|
if not parsed_lines:
|
|
raise ValueError("No valid speaker lines found in script")
|
|
|
|
# Check if we need to normalize speaker IDs (only if all are > 0)
|
|
min_speaker_id = min(speaker_ids)
|
|
if min_speaker_id > 0:
|
|
# Normalize to start from 0
|
|
normalized_lines = []
|
|
for speaker_id, text in parsed_lines:
|
|
normalized_lines.append((speaker_id - 1, text))
|
|
return normalized_lines
|
|
else:
|
|
# Keep original IDs
|
|
return parsed_lines
|
|
|
|
def _merge_inputs(self, text_inputs: BatchEncoding, audio_inputs: Dict) -> BatchEncoding:
|
|
"""Merge text and audio inputs into a single BatchEncoding."""
|
|
# Start with text inputs
|
|
merged = BatchEncoding(text_inputs)
|
|
|
|
# Add audio-specific fields
|
|
if "audio" in audio_inputs:
|
|
merged["speech_inputs"] = audio_inputs["audio"]
|
|
if "streaming" in audio_inputs:
|
|
merged["streaming"] = audio_inputs["streaming"]
|
|
|
|
return merged
|
|
|
|
def batch_decode(self, *args, **kwargs):
|
|
"""
|
|
This method forwards all its arguments to VibeVoiceTextTokenizer's [`~PreTrainedTokenizer.batch_decode`].
|
|
Please refer to the docstring of this method for more information.
|
|
"""
|
|
return self.tokenizer.batch_decode(*args, **kwargs)
|
|
|
|
def decode(self, *args, **kwargs):
|
|
"""
|
|
This method forwards all its arguments to VibeVoiceTextTokenizer's [`~PreTrainedTokenizer.decode`].
|
|
Please refer to the docstring of this method for more information.
|
|
"""
|
|
return self.tokenizer.decode(*args, **kwargs)
|
|
|
|
@property
|
|
def model_input_names(self):
|
|
"""
|
|
Return the list of inputs accepted by the model.
|
|
"""
|
|
tokenizer_input_names = self.tokenizer.model_input_names
|
|
audio_processor_input_names = self.audio_processor.model_input_names
|
|
return list(dict.fromkeys(tokenizer_input_names + audio_processor_input_names + ["speech_inputs", "speech_input_mask"]))
|
|
|
|
def save_audio(self,
|
|
audio: Union[torch.Tensor, np.ndarray, List[Union[torch.Tensor, np.ndarray]]],
|
|
output_path: str = "output.wav",
|
|
sampling_rate: Optional[int] = None,
|
|
normalize: bool = False,
|
|
batch_prefix: str = "audio_",
|
|
) -> str:
|
|
"""
|
|
Save audio data to a file.
|
|
Args:
|
|
audio (Union[torch.Tensor, np.ndarray, List[Union[torch.Tensor, np.ndarray]]]):
|
|
The audio data to save. Can be a single tensor/array or a list of them.
|
|
output_path (str, optional): Path to save the audio file. Defaults to "output.wav".
|
|
sampling_rate (int, optional): Sampling rate for the audio. If None, uses the processor's default.
|
|
normalize (bool, optional): Whether to normalize the audio before saving. Defaults to False.
|
|
batch_prefix (str, optional): Prefix for batch audio files. Defaults to "audio_".
|
|
Returns:
|
|
str: The path to the saved audio file.
|
|
"""
|
|
return self.audio_processor.save_audio(audio, output_path=output_path, sampling_rate=sampling_rate, normalize=normalize, batch_prefix=batch_prefix)
|
|
|
|
__all__ = [
|
|
"VibeVoiceProcessor",
|
|
] |