mirror of
https://github.com/modelscope/DiffSynth-Studio.git
synced 2026-04-02 15:48:11 +00:00
218 lines
8.7 KiB
Python
218 lines
8.7 KiB
Python
import torch, math
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from PIL import Image
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from typing import Union
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from tqdm import tqdm
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from einops import rearrange
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import numpy as np
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from math import prod
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from transformers import AutoTokenizer
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from ..core.device.npu_compatible_device import get_device_type
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from ..diffusion import FlowMatchScheduler
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from ..core import ModelConfig, gradient_checkpoint_forward
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from ..diffusion.base_pipeline import BasePipeline, PipelineUnit, ControlNetInput
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from ..utils.lora.merge import merge_lora
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from ..core.device.npu_compatible_device import get_device_type
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from ..core import ModelConfig
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from ..diffusion.base_pipeline import BasePipeline
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from ..models.ace_step_text_encoder import AceStepTextEncoder
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from ..models.ace_step_vae import AceStepVAE
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from ..models.ace_step_dit import AceStepConditionGenerationModelWrapper
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class AceStepAudioPipeline(BasePipeline):
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def __init__(self, device=get_device_type(), torch_dtype=torch.bfloat16):
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super().__init__(device=device, torch_dtype=torch_dtype)
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self.text_encoder: AceStepTextEncoder = None
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self.dit: AceStepConditionGenerationModelWrapper = None
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self.vae: AceStepVAE = None
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self.scheduler = FlowMatchScheduler()
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self.tokenizer: AutoTokenizer = None
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self.in_iteration_models = ("dit",)
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self.units = []
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@staticmethod
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def from_pretrained(
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torch_dtype: torch.dtype = torch.bfloat16,
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device: Union[str, torch.device] = get_device_type(),
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model_configs: list[ModelConfig] = [],
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tokenizer_config: ModelConfig = ModelConfig(model_id="ACE-Step/Ace-Step1.5", origin_file_pattern="Qwen3-Embedding-0.6B"),
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vram_limit: float = None,
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):
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# Initialize pipeline
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pipe = AceStepAudioPipeline(device=device, torch_dtype=torch_dtype)
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model_pool = pipe.download_and_load_models(model_configs, vram_limit)
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# Fetch models
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pipe.text_encoder = model_pool.fetch_model("ace_step_text_encoder")
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pipe.dit = model_pool.fetch_model("ace_step_dit")
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pipe.vae = model_pool.fetch_model("ace_step_vae")
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if tokenizer_config is not None:
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tokenizer_config.download_if_necessary()
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pipe.tokenizer = AutoTokenizer.from_pretrained(tokenizer_config.path)
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# VRAM Management
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pipe.vram_management_enabled = pipe.check_vram_management_state()
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return pipe
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@torch.no_grad()
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def __call__(
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self,
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caption: str,
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lyrics: str = "",
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duration: float = 160,
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bpm: int = None,
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keyscale: str = "",
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timesignature: str = "",
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vocal_language: str = "zh",
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instrumental: bool = False,
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inference_steps: int = 8,
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guidance_scale: float = 3.0,
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seed: int = None,
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):
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# Format text prompt with metadata
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text_prompt = self._format_text_prompt(caption, bpm, keyscale, timesignature, duration)
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lyrics_text = self._format_lyrics(lyrics, vocal_language, instrumental)
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# Tokenize
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text_inputs = self.tokenizer(
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text_prompt,
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return_tensors="pt",
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padding=True,
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truncation=True,
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max_length=512,
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).to(self.device)
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lyrics_inputs = self.tokenizer(
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lyrics_text,
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return_tensors="pt",
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padding=True,
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truncation=True,
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max_length=2048,
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).to(self.device)
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# Encode text and lyrics
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text_outputs = self.text_encoder(
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input_ids=text_inputs["input_ids"],
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attention_mask=text_inputs["attention_mask"],
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)
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lyrics_outputs = self.text_encoder(
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input_ids=lyrics_inputs["input_ids"],
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attention_mask=lyrics_inputs["attention_mask"],
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)
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# Get hidden states
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text_hidden_states = text_outputs.last_hidden_state
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lyric_hidden_states = lyrics_outputs.last_hidden_state
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# Prepare generation parameters
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latent_frames = int(duration * 46.875) # 48000 / 1024 ≈ 46.875 Hz
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# For text2music task, use silence_latent as src_latents
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# silence_latent will be tokenized/detokenized to get lm_hints_25Hz (127 dims)
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# which will be used as context for generation
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if self.silence_latent is not None:
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# Slice or pad silence_latent to match latent_frames
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if self.silence_latent.shape[1] >= latent_frames:
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src_latents = self.silence_latent[:, :latent_frames, :].to(device=self.device, dtype=self.torch_dtype)
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else:
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# Pad with zeros if silence_latent is shorter
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pad_len = latent_frames - self.silence_latent.shape[1]
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src_latents = torch.cat([
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self.silence_latent.to(device=self.device, dtype=self.torch_dtype),
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torch.zeros(1, pad_len, self.src_latent_channels, device=self.device, dtype=self.torch_dtype)
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], dim=1)
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else:
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# Fallback: create random latents if silence_latent is not loaded
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src_latents = torch.randn(1, latent_frames, self.src_latent_channels,
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device=self.device, dtype=self.torch_dtype)
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# Create attention mask
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attention_mask = torch.ones(1, latent_frames, device=self.device, dtype=self.torch_dtype)
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# Use silence_latent for the silence_latent parameter as well
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silence_latent = src_latents
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# Chunk masks and is_covers (for text2music, these are all zeros)
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# chunk_masks shape: [batch, latent_frames, 1]
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chunk_masks = torch.zeros(1, latent_frames, 1, device=self.device, dtype=self.torch_dtype)
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is_covers = torch.zeros(1, device=self.device, dtype=self.torch_dtype)
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# Reference audio (empty for text2music)
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# For text2music mode, we need empty reference audio
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# refer_audio_acoustic_hidden_states_packed: [batch, num_segments, hidden_dim]
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# refer_audio_order_mask: [num_segments] - indicates which batch each segment belongs to
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refer_audio_acoustic_hidden_states_packed = torch.zeros(1, 1, 64, device=self.device, dtype=self.torch_dtype)
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refer_audio_order_mask = torch.zeros(1, device=self.device, dtype=torch.long) # 1-d tensor
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# Generate audio latents using DiT model
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generation_result = self.dit.model.generate_audio(
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text_hidden_states=text_hidden_states,
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text_attention_mask=text_inputs["attention_mask"],
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lyric_hidden_states=lyric_hidden_states,
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lyric_attention_mask=lyrics_inputs["attention_mask"],
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refer_audio_acoustic_hidden_states_packed=refer_audio_acoustic_hidden_states_packed,
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refer_audio_order_mask=refer_audio_order_mask,
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src_latents=src_latents,
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chunk_masks=chunk_masks,
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is_covers=is_covers,
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silence_latent=silence_latent,
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attention_mask=attention_mask,
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seed=seed if seed is not None else 42,
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fix_nfe=inference_steps,
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shift=guidance_scale,
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)
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# Extract target latents from result dictionary
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generated_latents = generation_result["target_latents"]
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# Decode latents to audio
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# generated_latents shape: [batch, latent_frames, 64]
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# VAE expects: [batch, latent_frames, 64]
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audio_output = self.vae.decode(generated_latents, return_dict=True)
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audio = audio_output.sample
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# Post-process audio
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audio = self._postprocess_audio(audio)
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self.load_models_to_device([])
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return audio
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def _format_text_prompt(self, caption, bpm, keyscale, timesignature, duration):
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"""Format text prompt with metadata"""
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prompt = "# Instruction\nFill the audio semantic mask based on the given conditions:\n\n"
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prompt += f"# Caption\n{caption}\n\n"
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prompt += "# Metas\n"
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if bpm:
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prompt += f"- bpm: {bpm}\n"
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if timesignature:
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prompt += f"- timesignature: {timesignature}\n"
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if keyscale:
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prompt += f"- keyscale: {keyscale}\n"
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prompt += f"- duration: {int(duration)} seconds\n"
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prompt += "<|endoftext|>"
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return prompt
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def _format_lyrics(self, lyrics, vocal_language, instrumental):
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"""Format lyrics with language"""
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if instrumental or not lyrics:
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lyrics = "[Instrumental]"
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lyrics_text = f"# Languages\n{vocal_language}\n\n# Lyric\n{lyrics}<|endoftext|>"
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return lyrics_text
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def _postprocess_audio(self, audio):
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"""Post-process audio tensor"""
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# Ensure audio is on CPU and in float32
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audio = audio.to(device="cpu", dtype=torch.float32)
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# Normalize to [-1, 1]
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max_val = torch.abs(audio).max()
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if max_val > 0:
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audio = audio / max_val
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return audio
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