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488
vibevoice/modular/modeling_vibevoice.py
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488
vibevoice/modular/modeling_vibevoice.py
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from dataclasses import dataclass
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from typing import Dict, List, Optional, Tuple, Union, Callable
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from tqdm import tqdm
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import torch
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import torch.nn as nn
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import torch.nn.functional as F
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import torch.distributed as dist
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from transformers.models.auto import AutoModel, AutoModelForCausalLM
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from transformers.activations import ACT2FN
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from transformers.modeling_outputs import CausalLMOutput, BaseModelOutputWithPast, ModelOutput
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from transformers.models.llama.modeling_llama import LlamaRMSNorm
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from transformers import modeling_utils
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from transformers.modeling_utils import PreTrainedModel
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from transformers.modeling_flash_attention_utils import FlashAttentionKwargs
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from transformers.utils import logging
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from .modular_vibevoice_tokenizer import VibeVoiceTokenizerStreamingCache, VibeVoiceAcousticTokenizerModel, VibeVoiceSemanticTokenizerModel
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from .modular_vibevoice_diffusion_head import VibeVoiceDiffusionHead
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from vibevoice.schedule.dpm_solver import DPMSolverMultistepScheduler
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from .configuration_vibevoice import VibeVoiceConfig
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logger = logging.get_logger(__name__)
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if not hasattr(modeling_utils, "ALL_PARALLEL_STYLES") or modeling_utils.ALL_PARALLEL_STYLES is None:
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modeling_utils.ALL_PARALLEL_STYLES = ["tp", "none", "colwise", "rowwise"]
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@dataclass
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class VibeVoiceCausalLMOutputWithPast(ModelOutput):
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loss: Optional[torch.FloatTensor] = None
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diffusion_loss: Optional[torch.FloatTensor] = None
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speech_token_num: Optional[int] = None
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logits: torch.FloatTensor = None
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past_key_values: Optional[Tuple[Tuple[torch.FloatTensor]]] = None
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hidden_states: Optional[Tuple[torch.FloatTensor, ...]] = None
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attentions: Optional[Tuple[torch.FloatTensor, ...]] = None
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@dataclass
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class VibeVoiceGenerationOutput(ModelOutput):
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"""
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Output type for VibeVoice generation.
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Args:
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sequences (`torch.LongTensor` of shape `(batch_size, sequence_length)`):
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The generated sequences.
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speech_outputs (`List[torch.FloatTensor]`, *optional*):
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List of generated speech waveforms or latents for each speech segment.
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"""
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sequences: torch.LongTensor = None
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speech_outputs: Optional[List[torch.FloatTensor]] = None
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class SpeechConnector(nn.Module):
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def __init__(self, input_dim, output_dim):
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super().__init__()
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self.fc1 = nn.Linear(input_dim, output_dim)
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self.norm = LlamaRMSNorm(output_dim, eps=1e-6)
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self.fc2 = nn.Linear(output_dim, output_dim)
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def forward(self, features, **kwargs):
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x = self.fc1(features)
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x = self.norm(x)
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x = self.fc2(x)
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return x
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# @auto_docstring
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class VibeVoicePreTrainedModel(PreTrainedModel):
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config_class = VibeVoiceConfig
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base_model_prefix = "model"
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supports_gradient_checkpointing = True
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_skip_keys_device_placement = "past_key_values"
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_supports_cache_class = True
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_supports_flash_attn_2 = True
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_supports_sdpa = True
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_supports_quantized_cache = True
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_supports_static_cache = True
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_supports_attention_backend = True
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def _init_weights(self, module):
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if isinstance(module, VibeVoiceDiffusionHead):
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module.initialize_weights()
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return
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# Use the language model's initializer_range if available
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if hasattr(self.config, 'language_model_config') and hasattr(self.config.language_model_config, 'initializer_range'):
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std = self.config.language_model_config.initializer_range
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elif hasattr(self.config, 'decoder_config') and hasattr(self.config.decoder_config, 'initializer_range'):
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std = self.config.decoder_config.initializer_range
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else:
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std = 0.02 # Default value
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if isinstance(module, nn.Linear):
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module.weight.data.normal_(mean=0.0, std=std)
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if module.bias is not None:
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module.bias.data.zero_()
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elif isinstance(module, nn.LayerNorm):
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module.weight.data.fill_(1.0)
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module.bias.data.zero_()
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# @auto_docstring
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class VibeVoiceModel(VibeVoicePreTrainedModel):
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def __init__(self, config):
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super().__init__(config)
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if hasattr(config, 'torch_dtype') and config.torch_dtype is not None:
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if isinstance(config.torch_dtype, str):
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dtype = getattr(torch, config.torch_dtype)
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else:
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dtype = config.torch_dtype
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else:
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dtype = torch.float32
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# Initialize Qwen2 model for language modeling
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lm_config = config.decoder_config
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self.language_model = AutoModel.from_config(lm_config)
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# Initialize speech components if needed
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self.acoustic_tokenizer = AutoModel.from_config(config.acoustic_tokenizer_config).to(dtype)
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self.semantic_tokenizer = AutoModel.from_config(config.semantic_tokenizer_config).to(dtype)
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self.acoustic_connector = SpeechConnector(config.acoustic_vae_dim, lm_config.hidden_size).to(dtype)
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self.semantic_connector = SpeechConnector(config.semantic_vae_dim, lm_config.hidden_size).to(dtype)
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# Register scaling factors as buffers - use 1D tensors for FSDP compatibility
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self.register_buffer('speech_scaling_factor', torch.tensor(float('nan')))
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self.register_buffer('speech_bias_factor', torch.tensor(float('nan')))
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# Initialize prediction head for speech generation
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self.prediction_head = AutoModel.from_config(config.diffusion_head_config).to(dtype)
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# Initialize noise scheduler
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self.noise_scheduler = DPMSolverMultistepScheduler(
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num_train_timesteps=config.diffusion_head_config.ddpm_num_steps,
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beta_schedule=config.diffusion_head_config.ddpm_beta_schedule,
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prediction_type=config.diffusion_head_config.prediction_type
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)
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def get_input_embeddings(self):
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if hasattr(self.language_model, 'embed_tokens'):
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# If the language model has an embed_tokens attribute, return it
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return self.language_model.embed_tokens
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for name, attr in self.language_model.fullmap.items(): # parallel by nnscaler, the name is changed
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if attr.orig_name == 'embed_tokens.weight':
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return getattr(self.language_model, name)
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assert False, 'should not arrive here'
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def set_input_embeddings(self, value):
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self.language_model.embed_tokens = value
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def set_speech_tokenizers(self, acoustic_tokenizer=None, semantic_tokenizer=None):
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"""Set the speech tokenizers used for encoding and decoding speech."""
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self.acoustic_tokenizer = acoustic_tokenizer
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self.semantic_tokenizer = semantic_tokenizer
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# Reset the encoder to evaluation mode
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if self.acoustic_tokenizer is not None:
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self.acoustic_tokenizer.eval()
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if self.semantic_tokenizer is not None:
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self.semantic_tokenizer.eval()
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def forward(
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self,
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input_ids: torch.LongTensor = None,
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attention_mask: Optional[torch.Tensor] = None,
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position_ids: Optional[torch.LongTensor] = None,
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past_key_values: Optional[Tuple[Tuple[torch.FloatTensor]]] = None,
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inputs_embeds: Optional[torch.FloatTensor] = None,
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use_cache: Optional[bool] = None,
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output_attentions: Optional[bool] = None,
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output_hidden_states: Optional[bool] = None,
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return_dict: Optional[bool] = None,
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cache_position: Optional[torch.LongTensor] = None,
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**kwargs,
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) -> Union[Tuple, BaseModelOutputWithPast]:
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return_dict = return_dict if return_dict is not None else self.config.use_return_dict
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# Forward through language model
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outputs = self.language_model(
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input_ids=input_ids,
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attention_mask=attention_mask,
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position_ids=position_ids,
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past_key_values=past_key_values,
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inputs_embeds=inputs_embeds,
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use_cache=use_cache,
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output_attentions=output_attentions,
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output_hidden_states=output_hidden_states,
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return_dict=return_dict,
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cache_position=cache_position,
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**kwargs,
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)
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if not return_dict:
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return outputs
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return BaseModelOutputWithPast(
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last_hidden_state=outputs.last_hidden_state,
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past_key_values=outputs.past_key_values,
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hidden_states=outputs.hidden_states,
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attentions=outputs.attentions,
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)
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class VibeVoiceForConditionalGeneration(VibeVoicePreTrainedModel):
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_tied_weights_keys = ["lm_head.weight"]
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_tp_plan = {"lm_head": "colwise_rep"}
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def __init__(self, config):
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super().__init__(config)
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self.model = VibeVoiceModel(config)
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self.vocab_size = config.decoder_config.vocab_size
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self.lm_head = nn.Linear(config.decoder_config.hidden_size, self.vocab_size, bias=False)
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self.post_init()
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def get_input_embeddings(self):
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return self.model.get_input_embeddings()
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def set_input_embeddings(self, value):
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self.model.set_input_embeddings(value)
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def get_output_embeddings(self):
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return self.lm_head
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def set_decoder(self, decoder):
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self.model.language_model = decoder
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def get_decoder(self):
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return self.model.language_model
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def tie_weights(self):
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"""
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Tie the weights between the input embeddings and the output embeddings.
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"""
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if getattr(self.config.decoder_config, 'tie_word_embeddings', False):
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# The standard PreTrainedModel method will handle the tying.
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# It typically does a simple parameter object assignment, which is
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# CORRECT to do BEFORE FSDP wraps the model.
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output_embeddings = self.get_output_embeddings()
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input_embeddings = self.get_input_embeddings()
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if hasattr(input_embeddings, 'weight'):
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output_embeddings.weight = input_embeddings.weight
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else:
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# maybe returned input_embeddings a tensor directly
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output_embeddings.weight = input_embeddings
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if getattr(output_embeddings, "bias", None) is not None:
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output_embeddings.bias.data = nn.functional.pad(
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output_embeddings.bias.data,
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(0, output_embeddings.weight.shape[0] - output_embeddings.bias.shape[0]),
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"constant",
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0,
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)
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print("✅ Tied input and output embeddings using standard assignment.")
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else:
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print("ℹ️ tie_word_embeddings is False, not tying weights.")
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# Also, ensure set_output_embeddings is safe, though your implementation looks okay.
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# The key is to avoid calling it after accelerator.prepare().
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def set_output_embeddings(self, new_embeddings):
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# Your current implementation using data.copy_ is good practice,
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# but the best way is to not call this after prepare().
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self.lm_head = new_embeddings
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def forward_speech_features(
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self,
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speech_tensors=None,
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speech_masks=None,
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speech_type="audio",
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return_unmask=False
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):
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if speech_tensors is None:
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# Use config to get vae_dim instead of non-existent self.args
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vae_dim = self.config.acoustic_tokenizer_config.vae_dim
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audio_features = torch.zeros(1, 1, vae_dim).to(self.get_input_embeddings().weight)
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connect_features = self.model.acoustic_connector(audio_features)
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return audio_features, connect_features
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else:
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with torch.no_grad():
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if speech_type == "audio":
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with torch.no_grad():
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frames = self.model.acoustic_tokenizer.encode(speech_tensors.unsqueeze(1))[0][0]
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audio_tokens = frames.sample(self.model.acoustic_tokenizer.std_dist_type)[0]
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elif speech_type == "vae":
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# Use config to get vae_dim instead of non-existent self.args
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vae_dim = self.config.acoustic_tokenizer_config.vae_dim
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speech_mode = speech_tensors.reshape(speech_tensors.size(0), -1, vae_dim)
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# gaussian sample from the speech_mode
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batch_size = speech_mode.size(0)
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value = self.model.acoustic_tokenizer.fix_std / 0.8
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std = torch.randn(batch_size, dtype=speech_mode.dtype, device=speech_mode.device) * value
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std = std.view(-1, *[1] * (speech_mode.dim() - 1))
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audio_tokens = speech_mode + std * torch.randn(speech_mode.shape).to(speech_mode)
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else:
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raise NotImplementedError(f"Speech type {speech_type} not implemented")
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if torch.isnan(self.model.speech_scaling_factor) or torch.isnan(self.model.speech_bias_factor):
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scaling_factor = 1. / audio_tokens[speech_masks].flatten().std()
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bias_factor = -audio_tokens[speech_masks].flatten().mean()
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# Only use distributed operations if the process group is initialized
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if dist.is_available() and dist.is_initialized():
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dist.all_reduce(scaling_factor, op=dist.ReduceOp.SUM)
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dist.all_reduce(bias_factor, op=dist.ReduceOp.SUM)
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world_size = dist.get_world_size()
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self.model.speech_scaling_factor.copy_(scaling_factor / world_size)
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self.model.speech_bias_factor.copy_(bias_factor / world_size)
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print(f"Speech scaling factor (distributed): {self.model.speech_scaling_factor}, bias factor: {self.model.speech_bias_factor}", flush=True)
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else:
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# Single process case
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self.model.speech_scaling_factor.copy_(scaling_factor)
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self.model.speech_bias_factor.copy_(bias_factor)
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print(f"Speech scaling factor (single process): {self.model.speech_scaling_factor}, bias factor: {self.model.speech_bias_factor}", flush=True)
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audio_features = (audio_tokens + self.model.speech_bias_factor) * self.model.speech_scaling_factor
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connect_features = self.model.acoustic_connector(audio_features)
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if return_unmask:
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return audio_features, connect_features
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return audio_features[speech_masks], connect_features[speech_masks]
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def forward(
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self,
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input_ids: torch.LongTensor = None,
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attention_mask: Optional[torch.Tensor] = None,
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position_ids: Optional[torch.LongTensor] = None,
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past_key_values: Optional[List[torch.FloatTensor]] = None,
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inputs_embeds: Optional[torch.FloatTensor] = None,
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labels: Optional[torch.LongTensor] = None,
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use_cache: Optional[bool] = False,
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output_attentions: Optional[bool] = None,
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output_hidden_states: Optional[bool] = None,
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return_dict: Optional[bool] = None,
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cache_position: Optional[torch.LongTensor] = None,
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# New arguments for speech processing and loss calculation
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speech_tensors: Optional[torch.FloatTensor] = None,
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speech_masks: Optional[torch.BoolTensor] = None,
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speeches_loss_input: Optional[torch.FloatTensor] = None,
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speech_semantic_tensors: Optional[torch.FloatTensor] = None,
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acoustic_input_mask: Optional[torch.BoolTensor] = None,
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acoustic_loss_mask: Optional[torch.BoolTensor] = None,
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ddpm_batch_mul: int = 1,
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**kwargs: Optional[Dict[str, Union[torch.Tensor, str]]],
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) -> Union[Tuple, VibeVoiceCausalLMOutputWithPast]:
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return_dict = return_dict if return_dict is not None else self.config.use_return_dict
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x = self.get_input_embeddings()(input_ids)
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semantic_speech_all_connect_features = self.model.semantic_connector(speech_semantic_tensors)
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if speeches_loss_input is not None:
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# only part audio need diffuse
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speech_all_features, speech_all_connect_features = self.forward_speech_features(
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speech_tensors=speech_tensors.type_as(x) if speech_tensors is not None else None,
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speech_masks=speech_masks,
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speech_type=kwargs.get("speech_type", "audio"),
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return_unmask=True
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)
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if speech_tensors is not None:
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if semantic_speech_all_connect_features is not None:
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x[acoustic_input_mask] = speech_all_connect_features[speech_masks] + semantic_speech_all_connect_features[speech_masks]
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else:
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x[acoustic_input_mask] = speech_all_connect_features[speech_masks]
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speech_features = speech_all_features[speeches_loss_input.unsqueeze(-1) & speech_masks] # only part audio need diffuse
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speech_connect_features = speech_all_connect_features[speeches_loss_input.unsqueeze(-1) & speech_masks]
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else:
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speech_features, speech_connect_features = self.forward_speech_features(
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speech_tensors=speech_tensors.type_as(x) if speech_tensors is not None else None,
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speech_masks=speech_masks,
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speech_type=kwargs.get("speech_type", "audio"),
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)
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if speech_tensors is not None:
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x[acoustic_input_mask] = speech_connect_features
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outputs = self.model(
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input_ids=None,
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attention_mask=attention_mask,
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position_ids=position_ids,
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past_key_values=past_key_values,
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inputs_embeds=x,
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use_cache=use_cache,
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output_attentions=output_attentions,
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output_hidden_states=False,
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return_dict=return_dict,
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cache_position=cache_position,
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)
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hidden_states = outputs.last_hidden_state
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logits = self.lm_head(hidden_states)
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# logits = logits.float()
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loss = None
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if labels is not None:
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# The custom CE loss with masking is calculated in the training script.
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# We leave the standard loss calculation here as None.
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pass
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# --- Diffusion Loss Calculation ---
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diffusion_loss = None
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# This block is executed only if we are in a context that involves speech.
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if speech_tensors is not None and acoustic_loss_mask.sum().item() > 0:
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condition_features = hidden_states[acoustic_loss_mask]
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speech_len, latent_size = speech_features.shape
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noise = torch.randn(
|
||||
(speech_len * ddpm_batch_mul, latent_size),
|
||||
device=hidden_states.device,
|
||||
dtype=hidden_states.dtype
|
||||
)
|
||||
|
||||
timesteps = torch.multinomial(
|
||||
torch.ones(self.config.diffusion_head_config.ddpm_num_steps),
|
||||
speech_len * ddpm_batch_mul,
|
||||
replacement=True,
|
||||
).to(hidden_states.device)
|
||||
|
||||
speech_features_repeated = speech_features.repeat_interleave(ddpm_batch_mul, dim=0)
|
||||
condition_features_repeated = condition_features.repeat_interleave(ddpm_batch_mul, dim=0)
|
||||
|
||||
noisy_speech_features = self.model.noise_scheduler.add_noise(
|
||||
speech_features_repeated, noise, timesteps
|
||||
)
|
||||
|
||||
model_output = self.model.prediction_head(
|
||||
noisy_speech_features,
|
||||
timesteps.type_as(x),
|
||||
condition_features_repeated
|
||||
)
|
||||
|
||||
prediction_type = self.config.diffusion_head_config.prediction_type
|
||||
if prediction_type == "epsilon":
|
||||
target_for_loss = noise
|
||||
elif prediction_type == "v_prediction":
|
||||
target_for_loss = self.model.noise_scheduler.get_velocity(
|
||||
speech_features_repeated, noise, timesteps
|
||||
)
|
||||
else:
|
||||
raise NotImplementedError(f"Prediction type {prediction_type} not implemented")
|
||||
|
||||
diffusion_loss = F.mse_loss(model_output.float(), target_for_loss.float(), reduction='sum')
|
||||
if latent_size > 0 and ddpm_batch_mul > 0:
|
||||
diffusion_loss = diffusion_loss / latent_size / ddpm_batch_mul
|
||||
else:
|
||||
diffusion_loss = torch.tensor(0.0, device=diffusion_loss.device)
|
||||
|
||||
else:
|
||||
# Dummy loss for DDP to work when there are no speech samples in a batch,
|
||||
# but we are in a speech context.
|
||||
diffusion_loss = sum(p.sum() for p in self.model.prediction_head.parameters()) * 0.0
|
||||
diffusion_loss += sum(p.sum() for p in self.model.acoustic_connector.parameters()) * 0.0
|
||||
diffusion_loss += sum(p.sum() for p in self.model.semantic_connector.parameters()) * 0.0
|
||||
# --- End Diffusion Loss Calculation ---
|
||||
|
||||
if not return_dict:
|
||||
output = (logits, speech_len) + outputs.to_tuple()[1:]
|
||||
return (loss, diffusion_loss) + output
|
||||
|
||||
return VibeVoiceCausalLMOutputWithPast(
|
||||
loss=loss,
|
||||
diffusion_loss=diffusion_loss,
|
||||
speech_token_num=speech_len if speech_tensors is not None else 0,
|
||||
logits=logits,
|
||||
past_key_values=outputs.past_key_values,
|
||||
hidden_states=outputs.hidden_states,
|
||||
attentions=outputs.attentions,
|
||||
)
|
||||
|
||||
AutoModel.register(VibeVoiceConfig, VibeVoiceModel)
|
||||
AutoModelForCausalLM.register(VibeVoiceConfig, VibeVoiceForConditionalGeneration)
|
||||
|
||||
__all__ = [
|
||||
"VibeVoiceModel",
|
||||
"VibeVoicePreTrainedModel",
|
||||
"VibeVoiceForConditionalGeneration",
|
||||
"VibeVoiceCausalLMOutputWithPast",
|
||||
"VibeVoiceGenerationOutput",
|
||||
]
|
||||
Reference in New Issue
Block a user