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| from audio_encoder.AudioMAE import AudioMAEConditionCTPoolRand, extract_kaldi_fbank_feature | |
| import torchaudio | |
| import torchaudio.transforms as T | |
| import torch.nn.functional as F | |
| import inspect | |
| from typing import Any, Callable, Dict, List, Optional, Union | |
| from APadapter.ap_adapter.attention_processor import AttnProcessor2_0, IPAttnProcessor2_0 | |
| import random | |
| import os | |
| import scipy | |
| import safetensors | |
| import numpy as np | |
| import torch | |
| from transformers import ( | |
| ClapFeatureExtractor, | |
| ClapModel, | |
| GPT2Model, | |
| RobertaTokenizer, | |
| RobertaTokenizerFast, | |
| SpeechT5HifiGan, | |
| T5EncoderModel, | |
| T5Tokenizer, | |
| T5TokenizerFast, | |
| ) | |
| from diffusers.loaders import AttnProcsLayers | |
| from diffusers import AutoencoderKL | |
| from diffusers.schedulers import KarrasDiffusionSchedulers | |
| from diffusers.utils import ( | |
| is_accelerate_available, | |
| is_accelerate_version, | |
| is_librosa_available, | |
| logging, | |
| replace_example_docstring, | |
| ) | |
| from diffusers.utils.torch_utils import randn_tensor | |
| from diffusers.pipelines.pipeline_utils import AudioPipelineOutput, DiffusionPipeline | |
| from .modeling_audioldm2 import AudioLDM2ProjectionModel, AudioLDM2UNet2DConditionModel | |
| from diffusers.loaders import TextualInversionLoaderMixin | |
| from tqdm import tqdm # for progress bar | |
| from utils.lora_utils_successed_ver1 import train_lora, load_lora, wav_to_mel | |
| from utils.model_utils import slerp, do_replace_attn | |
| from utils.alpha_scheduler import AlphaScheduler | |
| from audioldm.utils import default_audioldm_config | |
| from audioldm.audio import TacotronSTFT, read_wav_file | |
| from audioldm.audio.tools import get_mel_from_wav, _pad_spec, normalize_wav, pad_wav | |
| if is_librosa_available(): | |
| import librosa | |
| import warnings | |
| import matplotlib.pyplot as plt | |
| from .pipeline_audioldm2 import AudioLDM2Pipeline | |
| pipeline_trained = AudioLDM2Pipeline.from_pretrained("cvssp/audioldm2-large", torch_dtype=torch.float32) | |
| pipeline_trained = pipeline_trained.to("cuda") | |
| layer_num = 0 | |
| cross = [None, None, 768, 768, 1024, 1024, None, None] | |
| unet = pipeline_trained.unet | |
| attn_procs = {} | |
| for name in unet.attn_processors.keys(): | |
| cross_attention_dim = None if name.endswith("attn1.processor") else unet.config.cross_attention_dim | |
| if name.startswith("mid_block"): | |
| hidden_size = unet.config.block_out_channels[-1] | |
| elif name.startswith("up_blocks"): | |
| block_id = int(name[len("up_blocks.")]) | |
| hidden_size = list(reversed(unet.config.block_out_channels))[block_id] | |
| elif name.startswith("down_blocks"): | |
| block_id = int(name[len("down_blocks.")]) | |
| hidden_size = unet.config.block_out_channels[block_id] | |
| if cross_attention_dim is None: | |
| attn_procs[name] = AttnProcessor2_0() | |
| else: | |
| cross_attention_dim = cross[layer_num % 8] | |
| layer_num += 1 | |
| if cross_attention_dim == 768: | |
| attn_procs[name] = IPAttnProcessor2_0( | |
| hidden_size=hidden_size, | |
| name=name, | |
| cross_attention_dim=cross_attention_dim, | |
| scale=0.5, | |
| num_tokens=8, | |
| do_copy=False | |
| ).to("cuda", dtype=torch.float32) | |
| else: | |
| attn_procs[name] = AttnProcessor2_0() | |
| state_dict = torch.load('/Data/home/Dennis/DeepMIR-2024/Final_Project/AP-adapter/pytorch_model.bin', map_location="cuda") | |
| for name, processor in attn_procs.items(): | |
| if hasattr(processor, 'to_v_ip') or hasattr(processor, 'to_k_ip'): | |
| weight_name_v = name + ".to_v_ip.weight" | |
| weight_name_k = name + ".to_k_ip.weight" | |
| processor.to_v_ip.weight = torch.nn.Parameter(state_dict[weight_name_v].half()) | |
| processor.to_k_ip.weight = torch.nn.Parameter(state_dict[weight_name_k].half()) | |
| unet.set_attn_processor(attn_procs) | |
| unet.to("cuda", dtype=torch.float32) | |
| def visualize_mel_spectrogram(mel_spect_tensor, output_path=None): | |
| mel_spect_array = mel_spect_tensor.squeeze().transpose(1, 0).detach().cpu().numpy() | |
| plt.figure(figsize=(10, 5)) | |
| plt.imshow(mel_spect_array, aspect='auto', origin='lower', cmap='magma') | |
| plt.colorbar(label="Log-Mel Energy") | |
| plt.title("Mel-Spectrogram") | |
| plt.xlabel("Time") | |
| plt.ylabel("Mel Frequency Bins") | |
| plt.tight_layout() | |
| if output_path: | |
| plt.savefig(output_path, dpi=300) | |
| print(f"Mel-spectrogram saved to {output_path}") | |
| else: | |
| plt.show() | |
| warnings.filterwarnings("ignore", category=FutureWarning) | |
| warnings.filterwarnings("ignore", category=UserWarning) | |
| logger = logging.get_logger(__name__) # pylint: disable=invalid-name | |
| class StoreProcessor(): | |
| def __init__(self, original_processor, value_dict, name): | |
| self.original_processor = original_processor | |
| self.value_dict = value_dict | |
| self.name = name | |
| self.value_dict[self.name] = dict() | |
| self.id = 0 | |
| def __call__(self, attn, hidden_states, *args, encoder_hidden_states=None, attention_mask=None, **kwargs): | |
| # Is self attention | |
| if encoder_hidden_states is None: | |
| # 將 hidden_states 存入 value_dict 中,名稱為 self.name | |
| # 如果輸入沒有 encoder_hidden_states,表示是自注意力層,則將輸入的 hidden_states 儲存在 value_dict 中。 | |
| # print(f'In StoreProcessor: {self.name} {self.id}') | |
| self.value_dict[self.name][self.id] = hidden_states.detach() | |
| self.id += 1 | |
| # 調用原始處理器,執行正常的注意力操作 | |
| res = self.original_processor(attn, hidden_states, *args, | |
| encoder_hidden_states=encoder_hidden_states, | |
| attention_mask=attention_mask, | |
| **kwargs) | |
| return res | |
| class LoadProcessor(): | |
| def __init__(self, original_processor, name, aud1_dict, aud2_dict, alpha, beta=0, lamd=0.6): | |
| super().__init__() | |
| self.original_processor = original_processor | |
| self.name = name | |
| self.aud1_dict = aud1_dict | |
| self.aud2_dict = aud2_dict | |
| self.alpha = alpha | |
| self.beta = beta | |
| self.lamd = lamd | |
| self.id = 0 | |
| def __call__(self, attn, hidden_states, *args, encoder_hidden_states=None, attention_mask=None, **kwargs): | |
| # Is self attention | |
| # 判斷是否是自注意力(self-attention) | |
| if encoder_hidden_states is None: | |
| # 如果當前索引小於 10 倍的 self.lamd,使用自定義的混合邏輯 | |
| if self.id < 10 * self.lamd: | |
| map0 = self.aud1_dict[self.name][self.id] | |
| map1 = self.aud2_dict[self.name][self.id] | |
| cross_map = self.beta * hidden_states + \ | |
| (1 - self.beta) * ((1 - self.alpha) * map0 + self.alpha * map1) | |
| # 調用原始處理器,將 cross_map 作為 encoder_hidden_states 傳入 | |
| res = self.original_processor(attn, hidden_states, *args, | |
| encoder_hidden_states=cross_map, | |
| attention_mask=attention_mask, | |
| **kwargs) | |
| else: | |
| # 否則,使用原始的 encoder_hidden_states(可能為 None) | |
| res = self.original_processor(attn, hidden_states, *args, | |
| encoder_hidden_states=encoder_hidden_states, | |
| attention_mask=attention_mask, | |
| **kwargs) | |
| self.id += 1 | |
| # 如果索引到達 self.aud1_dict[self.name] 的長度,重置索引為 0 | |
| if self.id == len(self.aud1_dict[self.name]): | |
| self.id = 0 | |
| else: | |
| # 如果是跨注意力(encoder_hidden_states 不為 None),直接使用原始處理器 | |
| res = self.original_processor(attn, hidden_states, *args, | |
| encoder_hidden_states=encoder_hidden_states, | |
| attention_mask=attention_mask, | |
| **kwargs) | |
| return res | |
| def prepare_inputs_for_generation( | |
| inputs_embeds, | |
| attention_mask=None, | |
| past_key_values=None, | |
| **kwargs,): | |
| if past_key_values is not None: | |
| # only last token for inputs_embeds if past is defined in kwargs | |
| inputs_embeds = inputs_embeds[:, -1:] | |
| return { | |
| "inputs_embeds": inputs_embeds, | |
| "attention_mask": attention_mask, | |
| "past_key_values": past_key_values, | |
| "use_cache": kwargs.get("use_cache"), | |
| } | |
| class AudioLDM2MorphPipeline(DiffusionPipeline,TextualInversionLoaderMixin): | |
| r""" | |
| Pipeline for text-to-audio generation using AudioLDM2. | |
| This model inherits from [`DiffusionPipeline`]. Check the superclass documentation for the generic methods | |
| implemented for all pipelines (downloading, saving, running on a particular device, etc.). | |
| Args: | |
| vae ([`AutoencoderKL`]): | |
| Variational Auto-Encoder (VAE) model to encode and decode images to and from latent representations. | |
| text_encoder ([`~transformers.ClapModel`]): | |
| First frozen text-encoder. AudioLDM2 uses the joint audio-text embedding model | |
| [CLAP](https://huggingface.co/docs/transformers/model_doc/clap#transformers.CLAPTextModelWithProjection), | |
| specifically the [laion/clap-htsat-unfused](https://huggingface.co/laion/clap-htsat-unfused) variant. The | |
| text branch is used to encode the text prompt to a prompt embedding. The full audio-text model is used to | |
| rank generated waveforms against the text prompt by computing similarity scores. | |
| text_encoder_2 ([`~transformers.T5EncoderModel`]): | |
| Second frozen text-encoder. AudioLDM2 uses the encoder of | |
| [T5](https://huggingface.co/docs/transformers/model_doc/t5#transformers.T5EncoderModel), specifically the | |
| [google/flan-t5-large](https://huggingface.co/google/flan-t5-large) variant. | |
| projection_model ([`AudioLDM2ProjectionModel`]): | |
| A trained model used to linearly project the hidden-states from the first and second text encoder models | |
| and insert learned SOS and EOS token embeddings. The projected hidden-states from the two text encoders are | |
| concatenated to give the input to the language model. | |
| language_model ([`~transformers.GPT2Model`]): | |
| An auto-regressive language model used to generate a sequence of hidden-states conditioned on the projected | |
| outputs from the two text encoders. | |
| tokenizer ([`~transformers.RobertaTokenizer`]): | |
| Tokenizer to tokenize text for the first frozen text-encoder. | |
| tokenizer_2 ([`~transformers.T5Tokenizer`]): | |
| Tokenizer to tokenize text for the second frozen text-encoder. | |
| feature_extractor ([`~transformers.ClapFeatureExtractor`]): | |
| Feature extractor to pre-process generated audio waveforms to log-mel spectrograms for automatic scoring. | |
| unet ([`UNet2DConditionModel`]): | |
| A `UNet2DConditionModel` to denoise the encoded audio latents. | |
| scheduler ([`SchedulerMixin`]): | |
| A scheduler to be used in combination with `unet` to denoise the encoded audio latents. Can be one of | |
| [`DDIMScheduler`], [`LMSDiscreteScheduler`], or [`PNDMScheduler`]. | |
| vocoder ([`~transformers.SpeechT5HifiGan`]): | |
| Vocoder of class `SpeechT5HifiGan` to convert the mel-spectrogram latents to the final audio waveform. | |
| """ | |
| def __init__( | |
| self, | |
| vae: AutoencoderKL, | |
| text_encoder: ClapModel, | |
| text_encoder_2: T5EncoderModel, | |
| projection_model: AudioLDM2ProjectionModel, | |
| language_model: GPT2Model, | |
| tokenizer: Union[RobertaTokenizer, RobertaTokenizerFast], | |
| tokenizer_2: Union[T5Tokenizer, T5TokenizerFast], | |
| feature_extractor: ClapFeatureExtractor, | |
| unet: AudioLDM2UNet2DConditionModel, | |
| scheduler: KarrasDiffusionSchedulers, | |
| vocoder: SpeechT5HifiGan, | |
| ): | |
| super().__init__() | |
| self.register_modules( | |
| vae=vae, | |
| text_encoder=text_encoder, | |
| text_encoder_2=text_encoder_2, | |
| projection_model=projection_model, | |
| language_model=language_model, | |
| tokenizer=tokenizer, | |
| tokenizer_2=tokenizer_2, | |
| feature_extractor=feature_extractor, | |
| unet=unet, | |
| scheduler=scheduler, | |
| vocoder=vocoder, | |
| ) | |
| self.vae_scale_factor = 2 ** (len(self.vae.config.block_out_channels) - 1) | |
| self.aud1_dict = dict() | |
| self.aud2_dict = dict() | |
| # Copied from diffusers.pipelines.stable_diffusion.pipeline_stable_diffusion.StableDiffusionPipeline.enable_vae_slicing | |
| def enable_vae_slicing(self): | |
| r""" | |
| Enable sliced VAE decoding. When this option is enabled, the VAE will split the input tensor in slices to | |
| compute decoding in several steps. This is useful to save some memory and allow larger batch sizes. | |
| """ | |
| self.vae.enable_slicing() | |
| # Copied from diffusers.pipelines.stable_diffusion.pipeline_stable_diffusion.StableDiffusionPipeline.disable_vae_slicing | |
| def disable_vae_slicing(self): | |
| r""" | |
| Disable sliced VAE decoding. If `enable_vae_slicing` was previously enabled, this method will go back to | |
| computing decoding in one step. | |
| """ | |
| self.vae.disable_slicing() | |
| def enable_model_cpu_offload(self, gpu_id=0): | |
| r""" | |
| Offloads all models to CPU using accelerate, reducing memory usage with a low impact on performance. Compared | |
| to `enable_sequential_cpu_offload`, this method moves one whole model at a time to the GPU when its `forward` | |
| method is called, and the model remains in GPU until the next model runs. Memory savings are lower than with | |
| `enable_sequential_cpu_offload`, but performance is much better due to the iterative execution of the `unet`. | |
| """ | |
| if is_accelerate_available() and is_accelerate_version(">=", "0.17.0.dev0"): | |
| from accelerate import cpu_offload_with_hook | |
| else: | |
| raise ImportError("`enable_model_cpu_offload` requires `accelerate v0.17.0` or higher.") | |
| device = torch.device(f"cuda:{gpu_id}") | |
| if self.device.type != "cpu": | |
| self.to("cpu", silence_dtype_warnings=True) | |
| torch.cuda.empty_cache() # otherwise we don't see the memory savings (but they probably exist) | |
| model_sequence = [ | |
| self.text_encoder.text_model, | |
| self.text_encoder.text_projection, | |
| self.text_encoder_2, | |
| self.projection_model, | |
| self.language_model, | |
| self.unet, | |
| self.vae, | |
| self.vocoder, | |
| self.text_encoder, | |
| ] | |
| hook = None | |
| for cpu_offloaded_model in model_sequence: | |
| _, hook = cpu_offload_with_hook(cpu_offloaded_model, device, prev_module_hook=hook) | |
| # We'll offload the last model manually. | |
| self.final_offload_hook = hook | |
| def generate_language_model( | |
| self, | |
| inputs_embeds: torch.Tensor = None, | |
| max_new_tokens: int = 512, | |
| **model_kwargs, | |
| ): | |
| """ | |
| Generates a sequence of hidden-states from the language model, conditioned on the embedding inputs. | |
| Parameters: | |
| inputs_embeds (`torch.FloatTensor` of shape `(batch_size, sequence_length, hidden_size)`): | |
| The sequence used as a prompt for the generation. | |
| max_new_tokens (`int`): | |
| Number of new tokens to generate. | |
| model_kwargs (`Dict[str, Any]`, *optional*): | |
| Ad hoc parametrization of additional model-specific kwargs that will be forwarded to the `forward` | |
| function of the model. | |
| Return: | |
| `inputs_embeds (`torch.FloatTensor` of shape `(batch_size, sequence_length, hidden_size)`): | |
| The sequence of generated hidden-states. | |
| """ | |
| max_new_tokens = max_new_tokens if max_new_tokens is not None else self.language_model.config.max_new_tokens | |
| model_kwargs = self.language_model._get_initial_cache_position(inputs_embeds, model_kwargs) | |
| for _ in range(max_new_tokens): | |
| # prepare model inputs | |
| model_inputs = prepare_inputs_for_generation(inputs_embeds, **model_kwargs) | |
| # forward pass to get next hidden states | |
| output = self.language_model(**model_inputs, return_dict=True) | |
| next_hidden_states = output.last_hidden_state | |
| # Update the model input | |
| inputs_embeds = torch.cat([inputs_embeds, next_hidden_states[:, -1:, :]], dim=1) | |
| # Update generated hidden states, model inputs, and length for next step | |
| model_kwargs = self.language_model._update_model_kwargs_for_generation(output, model_kwargs) | |
| return inputs_embeds[:, -max_new_tokens:, :] | |
| def encode_prompt( | |
| self, | |
| prompt, | |
| device, | |
| num_waveforms_per_prompt, | |
| do_classifier_free_guidance, | |
| negative_prompt=None, | |
| prompt_embeds: Optional[torch.FloatTensor] = None, | |
| negative_prompt_embeds: Optional[torch.FloatTensor] = None, | |
| generated_prompt_embeds: Optional[torch.FloatTensor] = None, | |
| negative_generated_prompt_embeds: Optional[torch.FloatTensor] = None, | |
| attention_mask: Optional[torch.LongTensor] = None, | |
| negative_attention_mask: Optional[torch.LongTensor] = None, | |
| max_new_tokens: Optional[int] = None, | |
| ): | |
| r""" | |
| Encodes the prompt into text encoder hidden states. | |
| Args: | |
| prompt (`str` or `List[str]`, *optional*): | |
| prompt to be encoded | |
| device (`torch.device`): | |
| torch device | |
| num_waveforms_per_prompt (`int`): | |
| number of waveforms that should be generated per prompt | |
| do_classifier_free_guidance (`bool`): | |
| whether to use classifier free guidance or not | |
| negative_prompt (`str` or `List[str]`, *optional*): | |
| The prompt or prompts not to guide the audio generation. If not defined, one has to pass | |
| `negative_prompt_embeds` instead. Ignored when not using guidance (i.e., ignored if `guidance_scale` is | |
| less than `1`). | |
| prompt_embeds (`torch.FloatTensor`, *optional*): | |
| Pre-computed text embeddings from the Flan T5 model. Can be used to easily tweak text inputs, *e.g.* | |
| prompt weighting. If not provided, text embeddings will be computed from `prompt` input argument. | |
| negative_prompt_embeds (`torch.FloatTensor`, *optional*): | |
| Pre-computed negative text embeddings from the Flan T5 model. Can be used to easily tweak text inputs, | |
| *e.g.* prompt weighting. If not provided, negative_prompt_embeds will be computed from | |
| `negative_prompt` input argument. | |
| generated_prompt_embeds (`torch.FloatTensor`, *optional*): | |
| Pre-generated text embeddings from the GPT2 langauge model. Can be used to easily tweak text inputs, | |
| *e.g.* prompt weighting. If not provided, text embeddings will be generated from `prompt` input | |
| argument. | |
| negative_generated_prompt_embeds (`torch.FloatTensor`, *optional*): | |
| Pre-generated negative text embeddings from the GPT2 language model. Can be used to easily tweak text | |
| inputs, *e.g.* prompt weighting. If not provided, negative_prompt_embeds will be computed from | |
| `negative_prompt` input argument. | |
| attention_mask (`torch.LongTensor`, *optional*): | |
| Pre-computed attention mask to be applied to the `prompt_embeds`. If not provided, attention mask will | |
| be computed from `prompt` input argument. | |
| negative_attention_mask (`torch.LongTensor`, *optional*): | |
| Pre-computed attention mask to be applied to the `negative_prompt_embeds`. If not provided, attention | |
| mask will be computed from `negative_prompt` input argument. | |
| max_new_tokens (`int`, *optional*, defaults to None): | |
| The number of new tokens to generate with the GPT2 language model. | |
| Returns: | |
| prompt_embeds (`torch.FloatTensor`): | |
| Text embeddings from the Flan T5 model. | |
| attention_mask (`torch.LongTensor`): | |
| Attention mask to be applied to the `prompt_embeds`. | |
| generated_prompt_embeds (`torch.FloatTensor`): | |
| Text embeddings generated from the GPT2 langauge model. | |
| Example: | |
| ```python | |
| >>> import scipy | |
| >>> import torch | |
| >>> from diffusers import AudioLDM2Pipeline | |
| >>> repo_id = "cvssp/audioldm2" | |
| >>> pipe = AudioLDM2Pipeline.from_pretrained(repo_id, torch_dtype=torch.float16) | |
| >>> pipe = pipe.to("cuda") | |
| >>> # Get text embedding vectors | |
| >>> prompt_embeds, attention_mask, generated_prompt_embeds = pipe.encode_prompt( | |
| ... prompt="Techno music with a strong, upbeat tempo and high melodic riffs", | |
| ... device="cuda", | |
| ... do_classifier_free_guidance=True, | |
| ... ) | |
| >>> # Pass text embeddings to pipeline for text-conditional audio generation | |
| >>> audio = pipe( | |
| ... prompt_embeds=prompt_embeds, | |
| ... attention_mask=attention_mask, | |
| ... generated_prompt_embeds=generated_prompt_embeds, | |
| ... num_inference_steps=200, | |
| ... audio_length_in_s=10.0, | |
| ... ).audios[0] | |
| >>> # save generated audio sample | |
| >>> scipy.io.wavfile.write("techno.wav", rate=16000, data=audio) | |
| ```""" | |
| # print("prompt",prompt) | |
| if prompt is not None and isinstance(prompt, str): | |
| batch_size = 1 | |
| elif prompt is not None and isinstance(prompt, list): | |
| batch_size = len(prompt) | |
| else: | |
| batch_size = prompt_embeds.shape[0] | |
| # Define tokenizers and text encoders | |
| tokenizers = [self.tokenizer, self.tokenizer_2] | |
| text_encoders = [self.text_encoder, self.text_encoder_2] | |
| if prompt_embeds is None: | |
| prompt_embeds_list = [] | |
| attention_mask_list = [] | |
| for tokenizer, text_encoder in zip(tokenizers, text_encoders): | |
| text_inputs = tokenizer( | |
| prompt, | |
| padding="max_length" if isinstance(tokenizer, (RobertaTokenizer, RobertaTokenizerFast)) else True, | |
| max_length=tokenizer.model_max_length, | |
| truncation=True, | |
| return_tensors="pt", | |
| ) | |
| text_input_ids = text_inputs.input_ids | |
| attention_mask = text_inputs.attention_mask | |
| untruncated_ids = tokenizer(prompt, padding="longest", return_tensors="pt").input_ids | |
| if untruncated_ids.shape[-1] >= text_input_ids.shape[-1] and not torch.equal( | |
| text_input_ids, untruncated_ids | |
| ): | |
| removed_text = tokenizer.batch_decode(untruncated_ids[:, tokenizer.model_max_length - 1 : -1]) | |
| logger.warning( | |
| f"The following part of your input was truncated because {text_encoder.config.model_type} can " | |
| f"only handle sequences up to {tokenizer.model_max_length} tokens: {removed_text}" | |
| ) | |
| text_input_ids = text_input_ids.to(device) | |
| attention_mask = attention_mask.to(device) | |
| if text_encoder.config.model_type == "clap": | |
| prompt_embeds = text_encoder.get_text_features( | |
| text_input_ids, | |
| attention_mask=attention_mask, | |
| ) | |
| # append the seq-len dim: (bs, hidden_size) -> (bs, seq_len, hidden_size) | |
| prompt_embeds = prompt_embeds[:, None, :] | |
| # make sure that we attend to this single hidden-state | |
| attention_mask = attention_mask.new_ones((batch_size, 1)) | |
| else: | |
| prompt_embeds = text_encoder( | |
| text_input_ids, | |
| attention_mask=attention_mask, | |
| ) | |
| prompt_embeds = prompt_embeds[0] | |
| prompt_embeds_list.append(prompt_embeds) | |
| attention_mask_list.append(attention_mask) | |
| projection_output = self.projection_model( | |
| hidden_states=prompt_embeds_list[0], | |
| hidden_states_1=prompt_embeds_list[1], | |
| attention_mask=attention_mask_list[0], | |
| attention_mask_1=attention_mask_list[1], | |
| ) | |
| projected_prompt_embeds = projection_output.hidden_states | |
| projected_attention_mask = projection_output.attention_mask | |
| generated_prompt_embeds = self.generate_language_model( | |
| projected_prompt_embeds, | |
| attention_mask=projected_attention_mask, | |
| max_new_tokens=max_new_tokens, | |
| ) | |
| prompt_embeds = prompt_embeds.to(dtype=self.text_encoder_2.dtype, device=device) | |
| attention_mask = ( | |
| attention_mask.to(device=device) | |
| if attention_mask is not None | |
| else torch.ones(prompt_embeds.shape[:2], dtype=torch.long, device=device) | |
| ) | |
| generated_prompt_embeds = generated_prompt_embeds.to(dtype=self.language_model.dtype, device=device) | |
| bs_embed, seq_len, hidden_size = prompt_embeds.shape | |
| # duplicate text embeddings for each generation per prompt, using mps friendly method | |
| prompt_embeds = prompt_embeds.repeat(1, num_waveforms_per_prompt, 1) | |
| prompt_embeds = prompt_embeds.view(bs_embed * num_waveforms_per_prompt, seq_len, hidden_size) | |
| # duplicate attention mask for each generation per prompt | |
| attention_mask = attention_mask.repeat(1, num_waveforms_per_prompt) | |
| attention_mask = attention_mask.view(bs_embed * num_waveforms_per_prompt, seq_len) | |
| bs_embed, seq_len, hidden_size = generated_prompt_embeds.shape | |
| # duplicate generated embeddings for each generation per prompt, using mps friendly method | |
| generated_prompt_embeds = generated_prompt_embeds.repeat(1, num_waveforms_per_prompt, 1) | |
| generated_prompt_embeds = generated_prompt_embeds.view( | |
| bs_embed * num_waveforms_per_prompt, seq_len, hidden_size | |
| ) | |
| # get unconditional embeddings for classifier free guidance | |
| if do_classifier_free_guidance and negative_prompt_embeds is None: | |
| uncond_tokens: List[str] | |
| if negative_prompt is None: | |
| uncond_tokens = [""] * batch_size | |
| elif type(prompt) is not type(negative_prompt): | |
| raise TypeError( | |
| f"`negative_prompt` should be the same type to `prompt`, but got {type(negative_prompt)} !=" | |
| f" {type(prompt)}." | |
| ) | |
| elif isinstance(negative_prompt, str): | |
| uncond_tokens = [negative_prompt] | |
| elif batch_size != len(negative_prompt): | |
| raise ValueError( | |
| f"`negative_prompt`: {negative_prompt} has batch size {len(negative_prompt)}, but `prompt`:" | |
| f" {prompt} has batch size {batch_size}. Please make sure that passed `negative_prompt` matches" | |
| " the batch size of `prompt`." | |
| ) | |
| else: | |
| uncond_tokens = negative_prompt | |
| negative_prompt_embeds_list = [] | |
| negative_attention_mask_list = [] | |
| max_length = prompt_embeds.shape[1] | |
| for tokenizer, text_encoder in zip(tokenizers, text_encoders): | |
| uncond_input = tokenizer( | |
| uncond_tokens, | |
| padding="max_length", | |
| max_length=tokenizer.model_max_length | |
| if isinstance(tokenizer, (RobertaTokenizer, RobertaTokenizerFast)) | |
| else max_length, | |
| truncation=True, | |
| return_tensors="pt", | |
| ) | |
| uncond_input_ids = uncond_input.input_ids.to(device) | |
| negative_attention_mask = uncond_input.attention_mask.to(device) | |
| if text_encoder.config.model_type == "clap": | |
| negative_prompt_embeds = text_encoder.get_text_features( | |
| uncond_input_ids, | |
| attention_mask=negative_attention_mask, | |
| ) | |
| # append the seq-len dim: (bs, hidden_size) -> (bs, seq_len, hidden_size) | |
| negative_prompt_embeds = negative_prompt_embeds[:, None, :] | |
| # make sure that we attend to this single hidden-state | |
| negative_attention_mask = negative_attention_mask.new_ones((batch_size, 1)) | |
| else: | |
| negative_prompt_embeds = text_encoder( | |
| uncond_input_ids, | |
| attention_mask=negative_attention_mask, | |
| ) | |
| negative_prompt_embeds = negative_prompt_embeds[0] | |
| negative_prompt_embeds_list.append(negative_prompt_embeds) | |
| negative_attention_mask_list.append(negative_attention_mask) | |
| projection_output = self.projection_model( | |
| hidden_states=negative_prompt_embeds_list[0], | |
| hidden_states_1=negative_prompt_embeds_list[1], | |
| attention_mask=negative_attention_mask_list[0], | |
| attention_mask_1=negative_attention_mask_list[1], | |
| ) | |
| negative_projected_prompt_embeds = projection_output.hidden_states | |
| negative_projected_attention_mask = projection_output.attention_mask | |
| negative_generated_prompt_embeds = self.generate_language_model( | |
| negative_projected_prompt_embeds, | |
| attention_mask=negative_projected_attention_mask, | |
| max_new_tokens=max_new_tokens, | |
| ) | |
| if do_classifier_free_guidance: | |
| seq_len = negative_prompt_embeds.shape[1] | |
| negative_prompt_embeds = negative_prompt_embeds.to(dtype=self.text_encoder_2.dtype, device=device) | |
| negative_attention_mask = ( | |
| negative_attention_mask.to(device=device) | |
| if negative_attention_mask is not None | |
| else torch.ones(negative_prompt_embeds.shape[:2], dtype=torch.long, device=device) | |
| ) | |
| negative_generated_prompt_embeds = negative_generated_prompt_embeds.to( | |
| dtype=self.language_model.dtype, device=device | |
| ) | |
| # duplicate unconditional embeddings for each generation per prompt, using mps friendly method | |
| negative_prompt_embeds = negative_prompt_embeds.repeat(1, num_waveforms_per_prompt, 1) | |
| negative_prompt_embeds = negative_prompt_embeds.view(batch_size * num_waveforms_per_prompt, seq_len, -1) | |
| # duplicate unconditional attention mask for each generation per prompt | |
| negative_attention_mask = negative_attention_mask.repeat(1, num_waveforms_per_prompt) | |
| negative_attention_mask = negative_attention_mask.view(batch_size * num_waveforms_per_prompt, seq_len) | |
| # duplicate unconditional generated embeddings for each generation per prompt | |
| seq_len = negative_generated_prompt_embeds.shape[1] | |
| negative_generated_prompt_embeds = negative_generated_prompt_embeds.repeat(1, num_waveforms_per_prompt, 1) | |
| negative_generated_prompt_embeds = negative_generated_prompt_embeds.view( | |
| batch_size * num_waveforms_per_prompt, seq_len, -1 | |
| ) | |
| # For classifier free guidance, we need to do two forward passes. | |
| # Here we concatenate the unconditional and text embeddings into a single batch | |
| # to avoid doing two forward passes | |
| prompt_embeds = torch.cat([negative_prompt_embeds, prompt_embeds]) | |
| attention_mask = torch.cat([negative_attention_mask, attention_mask]) | |
| generated_prompt_embeds = torch.cat([negative_generated_prompt_embeds, generated_prompt_embeds]) | |
| return prompt_embeds, attention_mask, generated_prompt_embeds | |
| # Copied from diffusers.pipelines.audioldm.pipeline_audioldm.AudioLDMPipeline.mel_spectrogram_to_waveform | |
| def mel_spectrogram_to_waveform(self, mel_spectrogram): | |
| if mel_spectrogram.dim() == 4: | |
| mel_spectrogram = mel_spectrogram.squeeze(1) | |
| waveform = self.vocoder(mel_spectrogram) | |
| # we always cast to float32 as this does not cause significant overhead and is compatible with bfloat16 | |
| waveform = waveform.cpu().float() | |
| return waveform | |
| def score_waveforms(self, text, audio, num_waveforms_per_prompt, device, dtype): | |
| if not is_librosa_available(): | |
| logger.info( | |
| "Automatic scoring of the generated audio waveforms against the input prompt text requires the " | |
| "`librosa` package to resample the generated waveforms. Returning the audios in the order they were " | |
| "generated. To enable automatic scoring, install `librosa` with: `pip install librosa`." | |
| ) | |
| return audio | |
| inputs = self.tokenizer(text, return_tensors="pt", padding=True) | |
| resampled_audio = librosa.resample( | |
| audio.numpy(), orig_sr=self.vocoder.config.sampling_rate, target_sr=self.feature_extractor.sampling_rate | |
| ) | |
| inputs["input_features"] = self.feature_extractor( | |
| list(resampled_audio), return_tensors="pt", sampling_rate=self.feature_extractor.sampling_rate | |
| ).input_features.type(dtype) | |
| inputs = inputs.to(device) | |
| # compute the audio-text similarity score using the CLAP model | |
| logits_per_text = self.text_encoder(**inputs).logits_per_text | |
| # sort by the highest matching generations per prompt | |
| indices = torch.argsort(logits_per_text, dim=1, descending=True)[:, :num_waveforms_per_prompt] | |
| audio = torch.index_select(audio, 0, indices.reshape(-1).cpu()) | |
| return audio | |
| # Copied from diffusers.pipelines.stable_diffusion.pipeline_stable_diffusion.StableDiffusionPipeline.prepare_extra_step_kwargs | |
| def prepare_extra_step_kwargs(self, generator, eta): | |
| # prepare extra kwargs for the scheduler step, since not all schedulers have the same signature | |
| # eta (η) is only used with the DDIMScheduler, it will be ignored for other schedulers. | |
| # eta corresponds to η in DDIM paper: https://arxiv.org/abs/2010.02502 | |
| # and should be between [0, 1] | |
| accepts_eta = "eta" in set(inspect.signature(self.scheduler.step).parameters.keys()) | |
| extra_step_kwargs = {} | |
| if accepts_eta: | |
| extra_step_kwargs["eta"] = eta | |
| # check if the scheduler accepts generator | |
| accepts_generator = "generator" in set(inspect.signature(self.scheduler.step).parameters.keys()) | |
| if accepts_generator: | |
| extra_step_kwargs["generator"] = generator | |
| return extra_step_kwargs | |
| def check_inputs( | |
| self, | |
| prompt, | |
| audio_length_in_s, | |
| vocoder_upsample_factor, | |
| callback_steps, | |
| negative_prompt=None, | |
| prompt_embeds=None, | |
| negative_prompt_embeds=None, | |
| generated_prompt_embeds=None, | |
| negative_generated_prompt_embeds=None, | |
| attention_mask=None, | |
| negative_attention_mask=None,): | |
| min_audio_length_in_s = vocoder_upsample_factor * self.vae_scale_factor | |
| if audio_length_in_s < min_audio_length_in_s: | |
| raise ValueError( | |
| f"`audio_length_in_s` has to be a positive value greater than or equal to {min_audio_length_in_s}, but " | |
| f"is {audio_length_in_s}." | |
| ) | |
| if self.vocoder.config.model_in_dim % self.vae_scale_factor != 0: | |
| raise ValueError( | |
| f"The number of frequency bins in the vocoder's log-mel spectrogram has to be divisible by the " | |
| f"VAE scale factor, but got {self.vocoder.config.model_in_dim} bins and a scale factor of " | |
| f"{self.vae_scale_factor}." | |
| ) | |
| if (callback_steps is None) or ( | |
| callback_steps is not None and (not isinstance(callback_steps, int) or callback_steps <= 0) | |
| ): | |
| raise ValueError( | |
| f"`callback_steps` has to be a positive integer but is {callback_steps} of type" | |
| f" {type(callback_steps)}." | |
| ) | |
| if prompt is not None and prompt_embeds is not None: | |
| raise ValueError( | |
| f"Cannot forward both `prompt`: {prompt} and `prompt_embeds`: {prompt_embeds}. Please make sure to" | |
| " only forward one of the two." | |
| ) | |
| elif prompt is None and (prompt_embeds is None or generated_prompt_embeds is None): | |
| raise ValueError( | |
| "Provide either `prompt`, or `prompt_embeds` and `generated_prompt_embeds`. Cannot leave " | |
| "`prompt` undefined without specifying both `prompt_embeds` and `generated_prompt_embeds`." | |
| ) | |
| elif prompt is not None and (not isinstance(prompt, str) and not isinstance(prompt, list)): | |
| raise ValueError(f"`prompt` has to be of type `str` or `list` but is {type(prompt)}") | |
| if negative_prompt is not None and negative_prompt_embeds is not None: | |
| raise ValueError( | |
| f"Cannot forward both `negative_prompt`: {negative_prompt} and `negative_prompt_embeds`:" | |
| f" {negative_prompt_embeds}. Please make sure to only forward one of the two." | |
| ) | |
| elif negative_prompt_embeds is not None and negative_generated_prompt_embeds is None: | |
| raise ValueError( | |
| "Cannot forward `negative_prompt_embeds` without `negative_generated_prompt_embeds`. Ensure that" | |
| "both arguments are specified" | |
| ) | |
| if prompt_embeds is not None and negative_prompt_embeds is not None: | |
| if prompt_embeds.shape != negative_prompt_embeds.shape: | |
| raise ValueError( | |
| "`prompt_embeds` and `negative_prompt_embeds` must have the same shape when passed directly, but" | |
| f" got: `prompt_embeds` {prompt_embeds.shape} != `negative_prompt_embeds`" | |
| f" {negative_prompt_embeds.shape}." | |
| ) | |
| if attention_mask is not None and attention_mask.shape != prompt_embeds.shape[:2]: | |
| raise ValueError( | |
| "`attention_mask should have the same batch size and sequence length as `prompt_embeds`, but got:" | |
| f"`attention_mask: {attention_mask.shape} != `prompt_embeds` {prompt_embeds.shape}" | |
| ) | |
| if generated_prompt_embeds is not None and negative_generated_prompt_embeds is not None: | |
| if generated_prompt_embeds.shape != negative_generated_prompt_embeds.shape: | |
| raise ValueError( | |
| "`generated_prompt_embeds` and `negative_generated_prompt_embeds` must have the same shape when " | |
| f"passed directly, but got: `generated_prompt_embeds` {generated_prompt_embeds.shape} != " | |
| f"`negative_generated_prompt_embeds` {negative_generated_prompt_embeds.shape}." | |
| ) | |
| if ( | |
| negative_attention_mask is not None | |
| and negative_attention_mask.shape != negative_prompt_embeds.shape[:2] | |
| ): | |
| raise ValueError( | |
| "`attention_mask should have the same batch size and sequence length as `prompt_embeds`, but got:" | |
| f"`attention_mask: {negative_attention_mask.shape} != `prompt_embeds` {negative_prompt_embeds.shape}" | |
| ) | |
| # Copied from diffusers.pipelines.stable_diffusion.pipeline_stable_diffusion.StableDiffusionPipeline.prepare_latents with width->self.vocoder.config.model_in_dim | |
| def prepare_latents(self, batch_size, num_channels_latents, height, dtype, device, generator, latents=None): | |
| shape = ( | |
| batch_size, | |
| num_channels_latents, | |
| height // self.vae_scale_factor, | |
| self.vocoder.config.model_in_dim // self.vae_scale_factor, | |
| ) | |
| if isinstance(generator, list) and len(generator) != batch_size: | |
| raise ValueError( | |
| f"You have passed a list of generators of length {len(generator)}, but requested an effective batch" | |
| f" size of {batch_size}. Make sure the batch size matches the length of the generators." | |
| ) | |
| if latents is None: | |
| latents = randn_tensor(shape, generator=generator, device=device, dtype=dtype) | |
| else: | |
| latents = latents.to(device) | |
| # scale the initial noise by the standard deviation required by the scheduler | |
| latents = latents * self.scheduler.init_noise_sigma | |
| return latents | |
| def pre_check(self, audio_length_in_s, prompt, callback_steps, negative_prompt): | |
| """ | |
| Step 0: Convert audio input length from seconds to spectrogram height | |
| Step 1. Check inputs. Raise error if not correct | |
| """ | |
| vocoder_upsample_factor = np.prod(self.vocoder.config.upsample_rates) / self.vocoder.config.sampling_rate | |
| if audio_length_in_s is None: | |
| audio_length_in_s = self.unet.config.sample_size * self.vae_scale_factor * vocoder_upsample_factor | |
| height = int(audio_length_in_s / vocoder_upsample_factor) | |
| original_waveform_length = int(audio_length_in_s * self.vocoder.config.sampling_rate) | |
| if height % self.vae_scale_factor != 0: | |
| height = int(np.ceil(height / self.vae_scale_factor)) * self.vae_scale_factor | |
| logger.info( | |
| f"Audio length in seconds {audio_length_in_s} is increased to {height * vocoder_upsample_factor} " | |
| f"so that it can be handled by the model. It will be cut to {audio_length_in_s} after the " | |
| f"denoising process." | |
| ) | |
| # 1. Check inputs. Raise error if not correct | |
| self.check_inputs( | |
| prompt, | |
| audio_length_in_s, | |
| vocoder_upsample_factor, | |
| callback_steps, | |
| negative_prompt, | |
| ) | |
| return height, original_waveform_length | |
| def encode_prompt_for_2_sources(self, prompt_1, prompt_2, negative_prompt_1, negative_prompt_2, max_new_tokens, device, num_waveforms_per_prompt, do_classifier_free_guidance): | |
| prompt_embeds_1, attention_mask_1, generated_prompt_embeds_1 = self.encode_prompt( | |
| prompt_1, | |
| device, | |
| num_waveforms_per_prompt, | |
| do_classifier_free_guidance, | |
| negative_prompt_1, | |
| max_new_tokens=max_new_tokens, | |
| ) | |
| prompt_embeds_2, attention_mask_2, generated_prompt_embeds_2 = self.encode_prompt( | |
| prompt_2, | |
| device, | |
| num_waveforms_per_prompt, | |
| do_classifier_free_guidance, | |
| negative_prompt_2, | |
| max_new_tokens=max_new_tokens, | |
| ) | |
| return [prompt_embeds_1, attention_mask_1, generated_prompt_embeds_1], [prompt_embeds_2, attention_mask_2, generated_prompt_embeds_2] | |
| def process_encoded_prompt(self, encoded_prompt, audio_file, time_pooling, freq_pooling): | |
| prompt_embeds, attention_mask, generated_prompt_embeds = encoded_prompt | |
| waveform, sr = torchaudio.load(audio_file) | |
| fbank = torch.zeros((1024, 128)) | |
| ta_kaldi_fbank = extract_kaldi_fbank_feature(waveform, sr, fbank) | |
| # print("ta_kaldi_fbank.shape",ta_kaldi_fbank.shape) | |
| mel_spect_tensor = ta_kaldi_fbank.unsqueeze(0) | |
| model = AudioMAEConditionCTPoolRand().cuda() | |
| model.eval() | |
| LOA_embed = model(mel_spect_tensor, time_pool=time_pooling, freq_pool=freq_pooling) | |
| uncond_LOA_embed = model(torch.zeros_like(mel_spect_tensor), time_pool=time_pooling, freq_pool=freq_pooling) | |
| LOA_embeds = LOA_embed[0] | |
| uncond_LOA_embeds = uncond_LOA_embed[0] | |
| bs_embed, seq_len, _ = LOA_embeds.shape | |
| num = prompt_embeds.shape[0] // 2 | |
| LOA_embeds = LOA_embeds.view(bs_embed , seq_len, -1) | |
| LOA_embeds = LOA_embeds.repeat(num, 1, 1) | |
| uncond_LOA_embeds = uncond_LOA_embeds.view(bs_embed , seq_len, -1) | |
| uncond_LOA_embeds = uncond_LOA_embeds.repeat(num, 1, 1) | |
| negative_g, g = generated_prompt_embeds.chunk(2) | |
| uncond = torch.cat([negative_g, uncond_LOA_embeds], dim=1) | |
| cond = torch.cat([g, LOA_embeds], dim=1) | |
| generated_prompt_embeds = torch.cat([uncond, cond], dim=0) | |
| model_dtype = next(self.unet.parameters()).dtype | |
| # Convert your tensor to the same dtype as the model | |
| generated_prompt_embeds = generated_prompt_embeds.to(model_dtype) | |
| return prompt_embeds, attention_mask, generated_prompt_embeds | |
| def aud2latent(self, audio_path, audio_length_in_s): | |
| DEVICE = torch.device( | |
| "cuda") if torch.cuda.is_available() else torch.device("cpu") | |
| # waveform, sr = torchaudio.load(audio_path) | |
| # fbank = torch.zeros((height, 64)) | |
| # ta_kaldi_fbank = extract_kaldi_fbank_feature(waveform, sr, fbank, num_mels=64) | |
| # mel_spect_tensor = ta_kaldi_fbank.unsqueeze(0).unsqueeze(0) | |
| mel_spect_tensor = wav_to_mel(audio_path, duration=audio_length_in_s).unsqueeze(0) | |
| output_path = audio_path.replace('.wav', '_fbank.png') | |
| visualize_mel_spectrogram(mel_spect_tensor, output_path) | |
| mel_spect_tensor = mel_spect_tensor.to(next(self.vae.parameters()).dtype) | |
| # print(f'mel_spect_tensor dtype: {mel_spect_tensor.dtype}') | |
| # print(f'self.vae dtype: {next(self.vae.parameters()).dtype}') | |
| latents = self.vae.encode(mel_spect_tensor.to(DEVICE))['latent_dist'].mean | |
| return latents | |
| def ddim_inversion(self, start_latents, prompt_embeds, attention_mask, generated_prompt_embeds, guidance_scale,num_inference_steps): | |
| start_step = 0 | |
| num_inference_steps = num_inference_steps | |
| device = start_latents.device | |
| self.scheduler.set_timesteps(num_inference_steps, device=device) | |
| start_latents *= self.scheduler.init_noise_sigma | |
| latents = start_latents.clone() | |
| for i in tqdm(range(start_step, num_inference_steps)): | |
| t = self.scheduler.timesteps[i] | |
| latent_model_input = torch.cat([latents] * 2) if guidance_scale > 1. else latents | |
| latent_model_input = self.scheduler.scale_model_input(latent_model_input, t) | |
| noise_pred = self.unet(latent_model_input, t, encoder_hidden_states=generated_prompt_embeds, encoder_hidden_states_1=prompt_embeds, encoder_attention_mask_1=attention_mask).sample | |
| if guidance_scale > 1.: | |
| noise_pred_uncon, noise_pred_con = noise_pred.chunk(2, dim=0) | |
| noise_pred = noise_pred_uncon + guidance_scale * (noise_pred_con - noise_pred_uncon) | |
| latents = self.scheduler.step(noise_pred, t, latents).prev_sample | |
| return latents | |
| def generate_morphing_prompt(self, prompt_1, prompt_2, alpha): | |
| closer_prompt = prompt_1 if alpha <= 0.5 else prompt_2 | |
| prompt = ( | |
| f"A musical performance morphing between '{prompt_1}' and '{prompt_2}'. " | |
| f"The sound is closer to '{closer_prompt}' with an interpolation factor of alpha={alpha:.2f}, " | |
| f"where alpha=0 represents fully the {prompt_1} and alpha=1 represents fully {prompt_2}." | |
| ) | |
| return prompt | |
| def cal_latent(self,audio_length_in_s,time_pooling, freq_pooling,num_inference_steps, guidance_scale, aud_noise_1, aud_noise_2, prompt_1, prompt_2, | |
| prompt_embeds_1, attention_mask_1, generated_prompt_embeds_1, prompt_embeds_2, attention_mask_2, generated_prompt_embeds_2, | |
| alpha, original_processor,attn_processor_dict, use_morph_prompt, morphing_with_lora): | |
| latents = slerp(aud_noise_1, aud_noise_2, alpha, self.use_adain) | |
| if not use_morph_prompt: | |
| max_length = max(prompt_embeds_1.shape[1], prompt_embeds_2.shape[1]) | |
| if prompt_embeds_1.shape[1] < max_length: | |
| pad_size = max_length - prompt_embeds_1.shape[1] | |
| padding = torch.zeros( | |
| (prompt_embeds_1.shape[0], pad_size, prompt_embeds_1.shape[2]), | |
| device=prompt_embeds_1.device, | |
| dtype=prompt_embeds_1.dtype | |
| ) | |
| prompt_embeds_1 = torch.cat([prompt_embeds_1, padding], dim=1) | |
| if prompt_embeds_2.shape[1] < max_length: | |
| pad_size = max_length - prompt_embeds_2.shape[1] | |
| padding = torch.zeros( | |
| (prompt_embeds_2.shape[0], pad_size, prompt_embeds_2.shape[2]), | |
| device=prompt_embeds_2.device, | |
| dtype=prompt_embeds_2.dtype | |
| ) | |
| prompt_embeds_2 = torch.cat([prompt_embeds_2, padding], dim=1) | |
| if attention_mask_1.shape[1] < max_length: | |
| pad_size = max_length - attention_mask_1.shape[1] | |
| padding = torch.zeros( | |
| (attention_mask_1.shape[0], pad_size), | |
| device=attention_mask_1.device, | |
| dtype=attention_mask_1.dtype | |
| ) | |
| attention_mask_1 = torch.cat([attention_mask_1, padding], dim=1) | |
| if attention_mask_2.shape[1] < max_length: | |
| pad_size = max_length - attention_mask_2.shape[1] | |
| padding = torch.zeros( | |
| (attention_mask_2.shape[0], pad_size), | |
| device=attention_mask_2.device, | |
| dtype=attention_mask_2.dtype | |
| ) | |
| attention_mask_2 = torch.cat([attention_mask_2, padding], dim=1) | |
| prompt_embeds = (1 - alpha) * prompt_embeds_1 + \ | |
| alpha * prompt_embeds_2 | |
| generated_prompt_embeds = (1 - alpha) * generated_prompt_embeds_1 + \ | |
| alpha * generated_prompt_embeds_2 | |
| attention_mask = attention_mask_1 if alpha < 0.5 else attention_mask_2 | |
| # attention_mask = attention_mask_1 & attention_mask_2 | |
| # attention_mask = attention_mask_1 | attention_mask_2 | |
| # attention_mask = (1 - alpha) * attention_mask_1 + alpha * attention_mask_2 | |
| # attention_mask = (attention_mask > 0.5).long() | |
| if morphing_with_lora: | |
| pipeline_trained.unet.set_attn_processor(attn_processor_dict) | |
| waveform = pipeline_trained( | |
| time_pooling= time_pooling, | |
| freq_pooling= freq_pooling, | |
| latents = latents, | |
| num_inference_steps= num_inference_steps, | |
| guidance_scale= guidance_scale, | |
| num_waveforms_per_prompt= 1, | |
| audio_length_in_s=audio_length_in_s, | |
| prompt_embeds = prompt_embeds.chunk(2)[1], | |
| negative_prompt_embeds = prompt_embeds.chunk(2)[0], | |
| generated_prompt_embeds = generated_prompt_embeds.chunk(2)[1], | |
| negative_generated_prompt_embeds = generated_prompt_embeds.chunk(2)[0], | |
| attention_mask = attention_mask.chunk(2)[1], | |
| negative_attention_mask = attention_mask.chunk(2)[0], | |
| ).audios[0] | |
| if morphing_with_lora: | |
| pipeline_trained.unet.set_attn_processor(original_processor) | |
| else: | |
| latent_model_input = latents | |
| morphing_prompt = self.generate_morphing_prompt(prompt_1, prompt_2, alpha) | |
| if morphing_with_lora: | |
| pipeline_trained.unet.set_attn_processor(attn_processor_dict) | |
| waveform = pipeline_trained( | |
| time_pooling= time_pooling, | |
| freq_pooling= freq_pooling, | |
| latents = latent_model_input, | |
| num_inference_steps= num_inference_steps, | |
| guidance_scale= guidance_scale, | |
| num_waveforms_per_prompt= 1, | |
| audio_length_in_s=audio_length_in_s, | |
| prompt= morphing_prompt, | |
| negative_prompt= 'Low quality', | |
| ).audios[0] | |
| if morphing_with_lora: | |
| pipeline_trained.unet.set_attn_processor(original_processor) | |
| return waveform | |
| def __call__( | |
| self, | |
| audio_file = None, | |
| audio_file2 = None, | |
| save_lora_dir = "./lora", | |
| load_lora_path_1 = None, | |
| load_lora_path_2 = None, | |
| lora_steps = 200, | |
| lora_lr = 2e-4, | |
| lora_rank = 16, | |
| time_pooling = 8, | |
| freq_pooling = 8, | |
| audio_length_in_s: Optional[float] = None, | |
| prompt_1: Union[str, List[str]] = None, | |
| prompt_2: Union[str, List[str]] = None, | |
| negative_prompt_1: Optional[Union[str, List[str]]] = None, | |
| negative_prompt_2: Optional[Union[str, List[str]]] = None, | |
| use_lora: bool = True, | |
| use_adain: bool = True, | |
| use_reschedule: bool = True, | |
| output_path: Optional[str] = None, | |
| num_inference_steps: int = 200, | |
| guidance_scale: float = 7.5, | |
| num_waveforms_per_prompt: Optional[int] = 1, | |
| attn_beta=0, | |
| lamd=0.6, | |
| fix_lora=None, | |
| save_intermediates=True, | |
| num_frames=50, | |
| max_new_tokens: Optional[int] = None, | |
| callback_steps: Optional[int] = 1, | |
| noisy_latent_with_lora=False, | |
| morphing_with_lora=False, | |
| use_morph_prompt=False, | |
| ): | |
| # 0. Load the pre-trained AP-adapter model | |
| layer_num = 0 | |
| cross = [None, None, 768, 768, 1024, 1024, None, None] | |
| attn_procs = {} | |
| for name in self.unet.attn_processors.keys(): | |
| cross_attention_dim = None if name.endswith("attn1.processor") else self.unet.config.cross_attention_dim | |
| if name.startswith("mid_block"): | |
| hidden_size = self.unet.config.block_out_channels[-1] | |
| elif name.startswith("up_blocks"): | |
| block_id = int(name[len("up_blocks.")]) | |
| hidden_size = list(reversed(self.unet.config.block_out_channels))[block_id] | |
| elif name.startswith("down_blocks"): | |
| block_id = int(name[len("down_blocks.")]) | |
| hidden_size = self.unet.config.block_out_channels[block_id] | |
| if cross_attention_dim is None: | |
| attn_procs[name] = AttnProcessor2_0() | |
| else: | |
| cross_attention_dim = cross[layer_num % 8] | |
| layer_num += 1 | |
| if cross_attention_dim == 768: | |
| attn_procs[name] = IPAttnProcessor2_0( | |
| hidden_size=hidden_size, | |
| name=name, | |
| cross_attention_dim=cross_attention_dim, | |
| scale=0.5, | |
| num_tokens=8, | |
| do_copy=False | |
| ).to("cuda", dtype=torch.float32) | |
| else: | |
| attn_procs[name] = AttnProcessor2_0() | |
| state_dict = torch.load('/Data/home/Dennis/DeepMIR-2024/Final_Project/AP-adapter/pytorch_model.bin', map_location="cuda") | |
| for name, processor in attn_procs.items(): | |
| if hasattr(processor, 'to_v_ip') or hasattr(processor, 'to_k_ip'): | |
| weight_name_v = name + ".to_v_ip.weight" | |
| weight_name_k = name + ".to_k_ip.weight" | |
| processor.to_v_ip.weight = torch.nn.Parameter(state_dict[weight_name_v].half()) | |
| processor.to_k_ip.weight = torch.nn.Parameter(state_dict[weight_name_k].half()) | |
| self.unet.set_attn_processor(attn_procs) | |
| self.vae= self.vae.to("cuda", dtype=torch.float32) | |
| self.unet = self.unet.to("cuda", dtype=torch.float32) | |
| self.language_model = self.language_model.to("cuda", dtype=torch.float32) | |
| self.projection_model = self.projection_model.to("cuda", dtype=torch.float32) | |
| self.vocoder = self.vocoder.to("cuda", dtype=torch.float32) | |
| self.text_encoder = self.text_encoder.to("cuda", dtype=torch.float32) | |
| self.text_encoder_2 = self.text_encoder_2.to("cuda", dtype=torch.float32) | |
| # 1. Pre-check | |
| height, original_waveform_length = self.pre_check(audio_length_in_s, prompt_1, callback_steps, negative_prompt_1) | |
| _, _ = self.pre_check(audio_length_in_s, prompt_2, callback_steps, negative_prompt_2) | |
| # print(f"height: {height}, original_waveform_length: {original_waveform_length}") # height: 1000, original_waveform_length: 160000 | |
| # # 2. Define call parameters | |
| device = "cuda" if torch.cuda.is_available() else "cpu" | |
| do_classifier_free_guidance = guidance_scale > 1.0 | |
| self.use_lora = use_lora | |
| self.use_adain = use_adain | |
| self.use_reschedule = use_reschedule | |
| self.output_path = output_path | |
| if self.use_lora: | |
| print("Loading lora...") | |
| if not load_lora_path_1: | |
| weight_name = f"{output_path.split('/')[-1]}_lora_0.ckpt" | |
| load_lora_path_1 = save_lora_dir + "/" + weight_name | |
| if not os.path.exists(load_lora_path_1): | |
| train_lora(audio_file ,height ,time_pooling ,freq_pooling ,prompt_1, negative_prompt_1, guidance_scale, save_lora_dir, self.tokenizer, self.tokenizer_2, | |
| self.text_encoder, self.text_encoder_2, self.language_model, self.projection_model, self.vocoder, | |
| self.vae, self.unet, self.scheduler, lora_steps, lora_lr, lora_rank, weight_name=weight_name) | |
| print(f"Load from {load_lora_path_1}.") | |
| if load_lora_path_1.endswith(".safetensors"): | |
| lora_1 = safetensors.torch.load_file( | |
| load_lora_path_1, device="cpu") | |
| else: | |
| lora_1 = torch.load(load_lora_path_1, map_location="cpu") | |
| if not load_lora_path_2: | |
| weight_name = f"{output_path.split('/')[-1]}_lora_1.ckpt" | |
| load_lora_path_2 = save_lora_dir + "/" + weight_name | |
| if not os.path.exists(load_lora_path_2): | |
| train_lora(audio_file2 ,height,time_pooling ,freq_pooling ,prompt_2, negative_prompt_2, guidance_scale, save_lora_dir, self.tokenizer, self.tokenizer_2, | |
| self.text_encoder, self.text_encoder_2, self.language_model, self.projection_model, self.vocoder, | |
| self.vae, self.unet, self.scheduler, lora_steps, lora_lr, lora_rank, weight_name=weight_name) | |
| print(f"Load from {load_lora_path_2}.") | |
| if load_lora_path_2.endswith(".safetensors"): | |
| lora_2 = safetensors.torch.load_file( | |
| load_lora_path_2, device="cpu") | |
| else: | |
| lora_2 = torch.load(load_lora_path_2, map_location="cpu") | |
| else: | |
| lora_1 = lora_2 = None | |
| # # 3. Encode input prompt | |
| encoded_prompt_1, encoded_prompt_2 = self.encode_prompt_for_2_sources(prompt_1, prompt_2, negative_prompt_1, negative_prompt_2, max_new_tokens, device, num_waveforms_per_prompt, do_classifier_free_guidance) | |
| prompt_embeds_1, attention_mask_1, generated_prompt_embeds_1 = self.process_encoded_prompt(encoded_prompt_1, audio_file, time_pooling, freq_pooling) | |
| prompt_embeds_2, attention_mask_2, generated_prompt_embeds_2 = self.process_encoded_prompt(encoded_prompt_2, audio_file2, time_pooling, freq_pooling) | |
| # 4. Prepare latent variables | |
| # For the first audio file | |
| original_processor = list(self.unet.attn_processors.values())[0] | |
| if noisy_latent_with_lora: | |
| self.unet = load_lora(self.unet, lora_1, lora_2, 0) | |
| # print(self.unet.attn_processors) | |
| # We directly use the latent representation of the audio file for VAE's decoder as the 1st ground truth | |
| audio_latent = self.aud2latent(audio_file, audio_length_in_s).to(device) | |
| # mel_spectrogram = self.vae.decode(audio_latent).sample | |
| # first_audio = self.mel_spectrogram_to_waveform(mel_spectrogram) | |
| # first_audio = first_audio[:, :original_waveform_length] | |
| # torchaudio.save(f"{self.output_path}/{0:02d}_gt.wav", first_audio, 16000) | |
| # aud_noise_1 is the noisy latent representation of the audio file 1 | |
| aud_noise_1 = self.ddim_inversion(audio_latent, prompt_embeds_1, attention_mask_1, generated_prompt_embeds_1, guidance_scale, num_inference_steps) | |
| # We use the pre-trained model to generate the audio file from the noisy latent representation | |
| # waveform = pipeline_trained( | |
| # audio_file = audio_file, | |
| # time_pooling= 2, | |
| # freq_pooling= 2, | |
| # prompt= prompt_1, | |
| # latents = aud_noise_1, | |
| # negative_prompt= negative_prompt_1, | |
| # num_inference_steps= 100, | |
| # guidance_scale= guidance_scale, | |
| # num_waveforms_per_prompt= 1, | |
| # audio_length_in_s=10, | |
| # ).audios | |
| # file_path = os.path.join(self.output_path, f"{0:02d}_gt2.wav") | |
| # scipy.io.wavfile.write(file_path, rate=16000, data=waveform[0]) | |
| # After reconstructed the audio file 1, we set the original processor back | |
| if noisy_latent_with_lora: | |
| self.unet.set_attn_processor(original_processor) | |
| # print(self.unet.attn_processors) | |
| # For the second audio file | |
| if noisy_latent_with_lora: | |
| self.unet = load_lora(self.unet, lora_1, lora_2, 1) | |
| # print(self.unet.attn_processors) | |
| # We directly use the latent representation of the audio file for VAE's decoder as the 1st ground truth | |
| audio_latent = self.aud2latent(audio_file2, audio_length_in_s) | |
| # mel_spectrogram = self.vae.decode(audio_latent).sample | |
| # last_audio = self.mel_spectrogram_to_waveform(mel_spectrogram) | |
| # last_audio = last_audio[:, :original_waveform_length] | |
| # torchaudio.save(f"{self.output_path}/{num_frames-1:02d}_gt.wav", last_audio, 16000) | |
| # aud_noise_2 is the noisy latent representation of the audio file 2 | |
| aud_noise_2 = self.ddim_inversion(audio_latent, prompt_embeds_2, attention_mask_2, generated_prompt_embeds_2, guidance_scale, num_inference_steps) | |
| # waveform = pipeline_trained( | |
| # audio_file = audio_file2, | |
| # time_pooling= 2, | |
| # freq_pooling= 2, | |
| # prompt= prompt_2, | |
| # latents = aud_noise_2, | |
| # negative_prompt= negative_prompt_2, | |
| # num_inference_steps= 100, | |
| # guidance_scale= guidance_scale, | |
| # num_waveforms_per_prompt= 1, | |
| # audio_length_in_s=10, | |
| # ).audios | |
| # file_path = os.path.join(self.output_path, f"{num_frames-1:02d}_gt2.wav") | |
| # scipy.io.wavfile.write(file_path, rate=16000, data=waveform[0]) | |
| if noisy_latent_with_lora: | |
| self.unet.set_attn_processor(original_processor) | |
| # print(self.unet.attn_processors) | |
| # After reconstructed the audio file 1, we set the original processor back | |
| original_processor = list(self.unet.attn_processors.values())[0] | |
| def morph(alpha_list, desc): | |
| audios = [] | |
| # if attn_beta is not None: | |
| if self.use_lora: | |
| self.unet = load_lora( | |
| self.unet, lora_1, lora_2, 0 if fix_lora is None else fix_lora) | |
| attn_processor_dict = {} | |
| # print(self.unet.attn_processors) | |
| for k in self.unet.attn_processors.keys(): | |
| # print(k) | |
| if do_replace_attn(k): | |
| # print(f"Since the key starts with *up*, we replace the processor with StoreProcessor.") | |
| if self.use_lora: | |
| attn_processor_dict[k] = StoreProcessor(self.unet.attn_processors[k], | |
| self.aud1_dict, k) | |
| else: | |
| attn_processor_dict[k] = StoreProcessor(original_processor, | |
| self.aud1_dict, k) | |
| else: | |
| attn_processor_dict[k] = self.unet.attn_processors[k] | |
| # print(attn_processor_dict) | |
| # print(attn_processor_dict) | |
| # print(self.unet.attn_processors) | |
| # self.unet.set_attn_processor(attn_processor_dict) | |
| # print(self.unet.attn_processors) | |
| first_audio = self.cal_latent( | |
| audio_length_in_s, | |
| time_pooling, | |
| freq_pooling, | |
| num_inference_steps, | |
| guidance_scale, | |
| aud_noise_1, | |
| aud_noise_2, | |
| prompt_1, | |
| prompt_2, | |
| prompt_embeds_1, | |
| attention_mask_1, | |
| generated_prompt_embeds_1, | |
| prompt_embeds_2, | |
| attention_mask_2, | |
| generated_prompt_embeds_2, | |
| alpha_list[0], | |
| original_processor, | |
| attn_processor_dict, | |
| use_morph_prompt, | |
| morphing_with_lora | |
| ) | |
| self.unet.set_attn_processor(original_processor) | |
| file_path = os.path.join(self.output_path, f"{0:02d}.wav") | |
| scipy.io.wavfile.write(file_path, rate=16000, data=first_audio) | |
| if self.use_lora: | |
| self.unet = load_lora( | |
| self.unet, lora_1, lora_2, 1 if fix_lora is None else fix_lora) | |
| attn_processor_dict = {} | |
| for k in self.unet.attn_processors.keys(): | |
| if do_replace_attn(k): | |
| # print(f"Since the key starts with *up*, we replace the processor with StoreProcessor.") | |
| if self.use_lora: | |
| attn_processor_dict[k] = StoreProcessor(self.unet.attn_processors[k], | |
| self.aud2_dict, k) | |
| else: | |
| attn_processor_dict[k] = StoreProcessor(original_processor, | |
| self.aud2_dict, k) | |
| else: | |
| attn_processor_dict[k] = self.unet.attn_processors[k] | |
| # self.unet.set_attn_processor(attn_processor_dict) | |
| last_audio = self.cal_latent( | |
| audio_length_in_s, | |
| time_pooling, | |
| freq_pooling, | |
| num_inference_steps, | |
| guidance_scale, | |
| aud_noise_1, | |
| aud_noise_2, | |
| prompt_1, | |
| prompt_2, | |
| prompt_embeds_1, | |
| attention_mask_1, | |
| generated_prompt_embeds_1, | |
| prompt_embeds_2, | |
| attention_mask_2, | |
| generated_prompt_embeds_2, | |
| alpha_list[-1], | |
| original_processor, | |
| attn_processor_dict, | |
| use_morph_prompt, | |
| morphing_with_lora | |
| ) | |
| file_path = os.path.join(self.output_path, f"{num_frames-1:02d}.wav") | |
| scipy.io.wavfile.write(file_path, rate=16000, data=last_audio) | |
| self.unet.set_attn_processor(original_processor) | |
| for i in tqdm(range(1, num_frames - 1), desc=desc): | |
| alpha = alpha_list[i] | |
| if self.use_lora: | |
| self.unet = load_lora( | |
| self.unet, lora_1, lora_2, alpha if fix_lora is None else fix_lora) | |
| attn_processor_dict = {} | |
| for k in self.unet.attn_processors.keys(): | |
| if do_replace_attn(k): | |
| if self.use_lora: | |
| attn_processor_dict[k] = LoadProcessor( | |
| self.unet.attn_processors[k], k, self.aud1_dict, self.aud2_dict, alpha, attn_beta, lamd) | |
| else: | |
| attn_processor_dict[k] = LoadProcessor( | |
| original_processor, k, self.aud1_dict, self.aud2_dict, alpha, attn_beta, lamd) | |
| else: | |
| attn_processor_dict[k] = self.unet.attn_processors[k] | |
| # self.unet.set_attn_processor(attn_processor_dict) | |
| audio = self.cal_latent( | |
| audio_length_in_s, | |
| time_pooling, | |
| freq_pooling, | |
| num_inference_steps, | |
| guidance_scale, | |
| aud_noise_1, | |
| aud_noise_2, | |
| prompt_1, | |
| prompt_2, | |
| prompt_embeds_1, | |
| attention_mask_1, | |
| generated_prompt_embeds_1, | |
| prompt_embeds_2, | |
| attention_mask_2, | |
| generated_prompt_embeds_2, | |
| alpha_list[i], | |
| original_processor, | |
| attn_processor_dict, | |
| use_morph_prompt, | |
| morphing_with_lora | |
| ) | |
| file_path = os.path.join(self.output_path, f"{i:02d}.wav") | |
| scipy.io.wavfile.write(file_path, rate=16000, data=audio) | |
| self.unet.set_attn_processor(original_processor) | |
| audios.append(audio) | |
| audios = [first_audio] + audios + [last_audio] | |
| return audios | |
| with torch.no_grad(): | |
| if self.use_reschedule: | |
| alpha_scheduler = AlphaScheduler() | |
| alpha_list = list(torch.linspace(0, 1, num_frames)) | |
| audios_pt = morph(alpha_list, "Sampling...") | |
| audios_pt = [torch.tensor(aud).unsqueeze(0) | |
| for aud in audios_pt] | |
| alpha_scheduler.from_imgs(audios_pt) | |
| alpha_list = alpha_scheduler.get_list() | |
| audios = morph(alpha_list, "Reschedule...") | |
| else: | |
| alpha_list = list(torch.linspace(0, 1, num_frames)) | |
| audios = morph(alpha_list, "Sampling...") | |
| return audios | |