import numpy as np import torch from typing import Optional, List, Tuple, NamedTuple, Union from models import PipelineWrapper import torchaudio from audioldm.utils import get_duration MAX_DURATION = 30 class PromptEmbeddings(NamedTuple): embedding_hidden_states: torch.Tensor embedding_class_lables: torch.Tensor boolean_prompt_mask: torch.Tensor def load_audio(audio_path: Union[str, np.array], fn_STFT, left: int = 0, right: int = 0, device: Optional[torch.device] = None, return_wav: bool = False, stft: bool = False, model_sr: Optional[int] = None) -> torch.Tensor: if stft: # AudioLDM/tango loading to spectrogram if type(audio_path) is str: import audioldm import audioldm.audio duration = get_duration(audio_path) if MAX_DURATION is not None: duration = min(duration, MAX_DURATION) mel, _, wav = audioldm.audio.wav_to_fbank(audio_path, target_length=int(duration * 102.4), fn_STFT=fn_STFT) mel = mel.unsqueeze(0) else: mel = audio_path c, h, w = mel.shape left = min(left, w-1) right = min(right, w - left - 1) mel = mel[:, :, left:w-right] mel = mel.unsqueeze(0).to(device) if return_wav: return mel, 16000, duration, wav return mel, model_sr, duration else: waveform, sr = torchaudio.load(audio_path) if sr != model_sr: waveform = torchaudio.functional.resample(waveform, orig_freq=sr, new_freq=model_sr) # waveform = waveform.numpy()[0, ...] def normalize_wav(waveform): waveform = waveform - torch.mean(waveform) waveform = waveform / (torch.max(torch.abs(waveform)) + 1e-8) return waveform * 0.5 waveform = normalize_wav(waveform) # waveform = waveform[None, ...] # waveform = pad_wav(waveform, segment_length) # waveform = waveform[0, ...] waveform = torch.FloatTensor(waveform) if MAX_DURATION is not None: duration = min(waveform.shape[-1] / model_sr, MAX_DURATION) waveform = waveform[:, :int(duration * model_sr)] # cut waveform duration = waveform.shape[-1] / model_sr return waveform, model_sr, duration def get_height_of_spectrogram(length: int, ldm_stable: PipelineWrapper) -> int: vocoder_upsample_factor = np.prod(ldm_stable.model.vocoder.config.upsample_rates) / \ ldm_stable.model.vocoder.config.sampling_rate if length is None: length = ldm_stable.model.unet.config.sample_size * ldm_stable.model.vae_scale_factor * \ vocoder_upsample_factor height = int(length / vocoder_upsample_factor) # original_waveform_length = int(length * ldm_stable.model.vocoder.config.sampling_rate) if height % ldm_stable.model.vae_scale_factor != 0: height = int(np.ceil(height / ldm_stable.model.vae_scale_factor)) * ldm_stable.model.vae_scale_factor print( f"Audio length in seconds {length} is increased to {height * vocoder_upsample_factor} " f"so that it can be handled by the model. It will be cut to {length} after the " f"denoising process." ) return height def get_text_embeddings(target_prompt: List[str], target_neg_prompt: List[str], ldm_stable: PipelineWrapper ) -> Tuple[torch.Tensor, PromptEmbeddings, PromptEmbeddings]: text_embeddings_hidden_states, text_embeddings_class_labels, text_embeddings_boolean_prompt_mask = \ ldm_stable.encode_text(target_prompt) uncond_embedding_hidden_states, uncond_embedding_class_lables, uncond_boolean_prompt_mask = \ ldm_stable.encode_text(target_neg_prompt) text_emb = PromptEmbeddings(embedding_hidden_states=text_embeddings_hidden_states, boolean_prompt_mask=text_embeddings_boolean_prompt_mask, embedding_class_lables=text_embeddings_class_labels) uncond_emb = PromptEmbeddings(embedding_hidden_states=uncond_embedding_hidden_states, boolean_prompt_mask=uncond_boolean_prompt_mask, embedding_class_lables=uncond_embedding_class_lables) return text_embeddings_class_labels, text_emb, uncond_emb