mirror of
https://github.com/modelscope/DiffSynth-Studio.git
synced 2026-04-02 15:48:11 +00:00
add acestep models
This commit is contained in:
217
diffsynth/pipelines/ace_step_audio.py
Normal file
217
diffsynth/pipelines/ace_step_audio.py
Normal file
@@ -0,0 +1,217 @@
|
||||
import torch, math
|
||||
from PIL import Image
|
||||
from typing import Union
|
||||
from tqdm import tqdm
|
||||
from einops import rearrange
|
||||
import numpy as np
|
||||
from math import prod
|
||||
from transformers import AutoTokenizer
|
||||
|
||||
from ..core.device.npu_compatible_device import get_device_type
|
||||
from ..diffusion import FlowMatchScheduler
|
||||
from ..core import ModelConfig, gradient_checkpoint_forward
|
||||
from ..diffusion.base_pipeline import BasePipeline, PipelineUnit, ControlNetInput
|
||||
from ..utils.lora.merge import merge_lora
|
||||
|
||||
from ..core.device.npu_compatible_device import get_device_type
|
||||
from ..core import ModelConfig
|
||||
from ..diffusion.base_pipeline import BasePipeline
|
||||
from ..models.ace_step_text_encoder import AceStepTextEncoder
|
||||
from ..models.ace_step_vae import AceStepVAE
|
||||
from ..models.ace_step_dit import AceStepConditionGenerationModelWrapper
|
||||
|
||||
|
||||
class AceStepAudioPipeline(BasePipeline):
|
||||
def __init__(self, device=get_device_type(), torch_dtype=torch.bfloat16):
|
||||
super().__init__(device=device, torch_dtype=torch_dtype)
|
||||
self.text_encoder: AceStepTextEncoder = None
|
||||
self.dit: AceStepConditionGenerationModelWrapper = None
|
||||
self.vae: AceStepVAE = None
|
||||
|
||||
self.scheduler = FlowMatchScheduler()
|
||||
self.tokenizer: AutoTokenizer = None
|
||||
self.in_iteration_models = ("dit",)
|
||||
self.units = []
|
||||
|
||||
@staticmethod
|
||||
def from_pretrained(
|
||||
torch_dtype: torch.dtype = torch.bfloat16,
|
||||
device: Union[str, torch.device] = get_device_type(),
|
||||
model_configs: list[ModelConfig] = [],
|
||||
tokenizer_config: ModelConfig = ModelConfig(model_id="ACE-Step/Ace-Step1.5", origin_file_pattern="Qwen3-Embedding-0.6B"),
|
||||
vram_limit: float = None,
|
||||
):
|
||||
# Initialize pipeline
|
||||
pipe = AceStepAudioPipeline(device=device, torch_dtype=torch_dtype)
|
||||
model_pool = pipe.download_and_load_models(model_configs, vram_limit)
|
||||
|
||||
# Fetch models
|
||||
pipe.text_encoder = model_pool.fetch_model("ace_step_text_encoder")
|
||||
pipe.dit = model_pool.fetch_model("ace_step_dit")
|
||||
pipe.vae = model_pool.fetch_model("ace_step_vae")
|
||||
if tokenizer_config is not None:
|
||||
tokenizer_config.download_if_necessary()
|
||||
pipe.tokenizer = AutoTokenizer.from_pretrained(tokenizer_config.path)
|
||||
|
||||
# VRAM Management
|
||||
pipe.vram_management_enabled = pipe.check_vram_management_state()
|
||||
return pipe
|
||||
|
||||
@torch.no_grad()
|
||||
def __call__(
|
||||
self,
|
||||
caption: str,
|
||||
lyrics: str = "",
|
||||
duration: float = 160,
|
||||
bpm: int = None,
|
||||
keyscale: str = "",
|
||||
timesignature: str = "",
|
||||
vocal_language: str = "zh",
|
||||
instrumental: bool = False,
|
||||
inference_steps: int = 8,
|
||||
guidance_scale: float = 3.0,
|
||||
seed: int = None,
|
||||
):
|
||||
# Format text prompt with metadata
|
||||
text_prompt = self._format_text_prompt(caption, bpm, keyscale, timesignature, duration)
|
||||
lyrics_text = self._format_lyrics(lyrics, vocal_language, instrumental)
|
||||
|
||||
# Tokenize
|
||||
text_inputs = self.tokenizer(
|
||||
text_prompt,
|
||||
return_tensors="pt",
|
||||
padding=True,
|
||||
truncation=True,
|
||||
max_length=512,
|
||||
).to(self.device)
|
||||
|
||||
lyrics_inputs = self.tokenizer(
|
||||
lyrics_text,
|
||||
return_tensors="pt",
|
||||
padding=True,
|
||||
truncation=True,
|
||||
max_length=2048,
|
||||
).to(self.device)
|
||||
|
||||
# Encode text and lyrics
|
||||
text_outputs = self.text_encoder(
|
||||
input_ids=text_inputs["input_ids"],
|
||||
attention_mask=text_inputs["attention_mask"],
|
||||
)
|
||||
|
||||
lyrics_outputs = self.text_encoder(
|
||||
input_ids=lyrics_inputs["input_ids"],
|
||||
attention_mask=lyrics_inputs["attention_mask"],
|
||||
)
|
||||
|
||||
# Get hidden states
|
||||
text_hidden_states = text_outputs.last_hidden_state
|
||||
lyric_hidden_states = lyrics_outputs.last_hidden_state
|
||||
|
||||
# Prepare generation parameters
|
||||
latent_frames = int(duration * 46.875) # 48000 / 1024 ≈ 46.875 Hz
|
||||
|
||||
# For text2music task, use silence_latent as src_latents
|
||||
# silence_latent will be tokenized/detokenized to get lm_hints_25Hz (127 dims)
|
||||
# which will be used as context for generation
|
||||
if self.silence_latent is not None:
|
||||
# Slice or pad silence_latent to match latent_frames
|
||||
if self.silence_latent.shape[1] >= latent_frames:
|
||||
src_latents = self.silence_latent[:, :latent_frames, :].to(device=self.device, dtype=self.torch_dtype)
|
||||
else:
|
||||
# Pad with zeros if silence_latent is shorter
|
||||
pad_len = latent_frames - self.silence_latent.shape[1]
|
||||
src_latents = torch.cat([
|
||||
self.silence_latent.to(device=self.device, dtype=self.torch_dtype),
|
||||
torch.zeros(1, pad_len, self.src_latent_channels, device=self.device, dtype=self.torch_dtype)
|
||||
], dim=1)
|
||||
else:
|
||||
# Fallback: create random latents if silence_latent is not loaded
|
||||
src_latents = torch.randn(1, latent_frames, self.src_latent_channels,
|
||||
device=self.device, dtype=self.torch_dtype)
|
||||
|
||||
# Create attention mask
|
||||
attention_mask = torch.ones(1, latent_frames, device=self.device, dtype=self.torch_dtype)
|
||||
|
||||
# Use silence_latent for the silence_latent parameter as well
|
||||
silence_latent = src_latents
|
||||
|
||||
# Chunk masks and is_covers (for text2music, these are all zeros)
|
||||
# chunk_masks shape: [batch, latent_frames, 1]
|
||||
chunk_masks = torch.zeros(1, latent_frames, 1, device=self.device, dtype=self.torch_dtype)
|
||||
is_covers = torch.zeros(1, device=self.device, dtype=self.torch_dtype)
|
||||
|
||||
# Reference audio (empty for text2music)
|
||||
# For text2music mode, we need empty reference audio
|
||||
# refer_audio_acoustic_hidden_states_packed: [batch, num_segments, hidden_dim]
|
||||
# refer_audio_order_mask: [num_segments] - indicates which batch each segment belongs to
|
||||
refer_audio_acoustic_hidden_states_packed = torch.zeros(1, 1, 64, device=self.device, dtype=self.torch_dtype)
|
||||
refer_audio_order_mask = torch.zeros(1, device=self.device, dtype=torch.long) # 1-d tensor
|
||||
|
||||
# Generate audio latents using DiT model
|
||||
generation_result = self.dit.model.generate_audio(
|
||||
text_hidden_states=text_hidden_states,
|
||||
text_attention_mask=text_inputs["attention_mask"],
|
||||
lyric_hidden_states=lyric_hidden_states,
|
||||
lyric_attention_mask=lyrics_inputs["attention_mask"],
|
||||
refer_audio_acoustic_hidden_states_packed=refer_audio_acoustic_hidden_states_packed,
|
||||
refer_audio_order_mask=refer_audio_order_mask,
|
||||
src_latents=src_latents,
|
||||
chunk_masks=chunk_masks,
|
||||
is_covers=is_covers,
|
||||
silence_latent=silence_latent,
|
||||
attention_mask=attention_mask,
|
||||
seed=seed if seed is not None else 42,
|
||||
fix_nfe=inference_steps,
|
||||
shift=guidance_scale,
|
||||
)
|
||||
|
||||
# Extract target latents from result dictionary
|
||||
generated_latents = generation_result["target_latents"]
|
||||
|
||||
# Decode latents to audio
|
||||
# generated_latents shape: [batch, latent_frames, 64]
|
||||
# VAE expects: [batch, latent_frames, 64]
|
||||
audio_output = self.vae.decode(generated_latents, return_dict=True)
|
||||
audio = audio_output.sample
|
||||
|
||||
# Post-process audio
|
||||
audio = self._postprocess_audio(audio)
|
||||
|
||||
self.load_models_to_device([])
|
||||
return audio
|
||||
|
||||
def _format_text_prompt(self, caption, bpm, keyscale, timesignature, duration):
|
||||
"""Format text prompt with metadata"""
|
||||
prompt = "# Instruction\nFill the audio semantic mask based on the given conditions:\n\n"
|
||||
prompt += f"# Caption\n{caption}\n\n"
|
||||
prompt += "# Metas\n"
|
||||
if bpm:
|
||||
prompt += f"- bpm: {bpm}\n"
|
||||
if timesignature:
|
||||
prompt += f"- timesignature: {timesignature}\n"
|
||||
if keyscale:
|
||||
prompt += f"- keyscale: {keyscale}\n"
|
||||
prompt += f"- duration: {int(duration)} seconds\n"
|
||||
prompt += "<|endoftext|>"
|
||||
return prompt
|
||||
|
||||
def _format_lyrics(self, lyrics, vocal_language, instrumental):
|
||||
"""Format lyrics with language"""
|
||||
if instrumental or not lyrics:
|
||||
lyrics = "[Instrumental]"
|
||||
|
||||
lyrics_text = f"# Languages\n{vocal_language}\n\n# Lyric\n{lyrics}<|endoftext|>"
|
||||
return lyrics_text
|
||||
|
||||
def _postprocess_audio(self, audio):
|
||||
"""Post-process audio tensor"""
|
||||
# Ensure audio is on CPU and in float32
|
||||
audio = audio.to(device="cpu", dtype=torch.float32)
|
||||
|
||||
# Normalize to [-1, 1]
|
||||
max_val = torch.abs(audio).max()
|
||||
if max_val > 0:
|
||||
audio = audio / max_val
|
||||
|
||||
return audio
|
||||
Reference in New Issue
Block a user