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sync with latest version of diffusers
Browse files- audiodiffusion/__init__.py +77 -90
- audiodiffusion/mel.py +39 -38
- notebooks/test_model.ipynb +7 -6
audiodiffusion/__init__.py
CHANGED
@@ -8,6 +8,8 @@ from tqdm.auto import tqdm
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from librosa.beat import beat_track
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from diffusers import (DiffusionPipeline, UNet2DConditionModel, DDIMScheduler,
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DDPMScheduler, AutoencoderKL)
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from .mel import Mel
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@@ -83,7 +85,8 @@ class AudioDiffusion:
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generator=generator,
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step_generator=step_generator,
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eta=eta,
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noise=noise
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return images[0], (sample_rate, audios[0])
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def generate_spectrogram_and_audio_from_audio(
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@@ -133,7 +136,8 @@ class AudioDiffusion:
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mask_end_secs=mask_end_secs,
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step_generator=step_generator,
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eta=eta,
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noise=noise
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return images[0], (sample_rate, audios[0])
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@staticmethod
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@@ -158,9 +162,7 @@ class AudioDiffusion:
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class AudioDiffusionPipeline(DiffusionPipeline):
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def __init__(self, unet: UNet2DConditionModel,
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scheduler: Union[DDIMScheduler, DDPMScheduler]):
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super().__init__()
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self.register_modules(unet=unet, scheduler=scheduler)
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@@ -170,11 +172,13 @@ class AudioDiffusionPipeline(DiffusionPipeline):
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Returns:
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Tuple: (height, width)
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"""
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input_module = self.vqvae if hasattr(self,
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# For backwards compatibility
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sample_size = (
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input_module.sample_size, input_module.sample_size)
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-
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return sample_size
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def get_default_steps(self) -> int:
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@@ -200,8 +204,11 @@ class AudioDiffusionPipeline(DiffusionPipeline):
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mask_end_secs: float = 0,
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step_generator: torch.Generator = None,
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eta: float = 0,
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noise: torch.Tensor = None
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"""Generate random mel spectrogram from audio input and convert to audio.
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Args:
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step_generator (torch.Generator): random number generator used to de-noise or None
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eta (float): parameter between 0 and 1 used with DDIM scheduler
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noise (torch.Tensor): noise tensor of shape (batch_size, 1, height, width) or None
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Returns:
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List[PIL Image]: mel spectrograms
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(float, List[np.ndarray]): sample rate and raw audios
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"""
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steps = steps or self.get_default_steps()
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@@ -229,89 +236,78 @@ class AudioDiffusionPipeline(DiffusionPipeline):
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step_generator = step_generator or generator
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# For backwards compatibility
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if type(self.unet.sample_size) == int:
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self.unet.sample_size = (self.unet.sample_size,
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-
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if noise is None:
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noise = torch.randn(
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(batch_size, self.unet.in_channels, self.unet.sample_size[0],
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-
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images = noise
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mask = None
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if audio_file is not None or raw_audio is not None:
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mel.load_audio(audio_file, raw_audio)
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input_image = mel.audio_slice_to_image(slice)
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input_image = np.frombuffer(input_image.tobytes(),
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torch.unsqueeze(input_images,
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0).to(self.device)).latent_dist.sample(
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generator=generator).cpu()[0]
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input_images = 0.18215 * input_images
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if start_step > 0:
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images[0, 0] = self.scheduler.add_noise(
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input_images, noise,
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self.scheduler.timesteps[start_step - 1])
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pixels_per_second =
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mel.get_sample_rate() / mel.x_res /
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mel.hop_length)
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mask_start = int(mask_start_secs * pixels_per_second)
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mask_end = int(mask_end_secs * pixels_per_second)
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mask = self.scheduler.add_noise(
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input_images, noise,
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torch.tensor(self.scheduler.timesteps[start_step:]))
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-
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-
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self.progress_bar(self.scheduler.timesteps[start_step:])):
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model_output = self.unet(images, t)['sample']
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if isinstance(self.scheduler, DDIMScheduler):
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images = self.scheduler.step(
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model_output=model_output,
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sample=images,
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eta=eta,
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generator=step_generator)['prev_sample']
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else:
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images = self.scheduler.step(
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model_output=model_output,
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sample=images,
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generator=step_generator)['prev_sample']
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if mask is not None:
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if mask_start > 0:
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images[:, :, :, :mask_start] = mask[:,
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step, :, :mask_start]
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if mask_end > 0:
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images[:, :, :, -mask_end:] = mask[:, step, :, -mask_end:]
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if hasattr(self,
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# 0.18215 was scaling factor used in training to ensure unit variance
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images = 1 / 0.18215 * images
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images = self.vqvae.decode(images)[
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images = (images / 2 + 0.5).clamp(0, 1)
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images = images.cpu().permute(0, 2, 3, 1).numpy()
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images = (images * 255).round().astype("uint8")
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images = list(
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map(lambda _: Image.fromarray(_[:, :, 0]), images)
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shape[3] == 1
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audios = list(map(lambda _: mel.image_to_audio(_), images))
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@torch.no_grad()
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def encode(self, images: List[Image.Image], steps: int = 50) -> np.ndarray:
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@@ -328,35 +324,30 @@ class AudioDiffusionPipeline(DiffusionPipeline):
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# Only works with DDIM as this method is deterministic
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assert isinstance(self.scheduler, DDIMScheduler)
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self.scheduler.set_timesteps(steps)
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sample = np.array(
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np.frombuffer(image.tobytes(), dtype="uint8").reshape(
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sample = ((sample / 255) * 2 - 1)
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sample = torch.Tensor(sample).to(self.device)
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for t in self.progress_bar(torch.flip(self.scheduler.timesteps,
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prev_timestep = (t - self.scheduler.num_train_timesteps //
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self.scheduler.num_inference_steps)
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alpha_prod_t = self.scheduler.alphas_cumprod[t]
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alpha_prod_t_prev = (
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-
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beta_prod_t = 1 - alpha_prod_t
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model_output = self.unet(sample, t)[
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pred_sample_direction = (1 -
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sample =
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pred_sample_direction) * alpha_prod_t_prev**(-0.5)
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sample = sample * alpha_prod_t**(0.5) + beta_prod_t**(
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0.5) * model_output
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return sample
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@staticmethod
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def slerp(x0: torch.Tensor, x1: torch.Tensor,
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alpha: float) -> torch.Tensor:
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"""Spherical Linear intERPolation
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Args:
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@@ -368,18 +359,14 @@ class AudioDiffusionPipeline(DiffusionPipeline):
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torch.Tensor: interpolated tensor
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"""
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theta = acos(
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torch.norm(x1))
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return sin((1 - alpha) * theta) * x0 / sin(theta) + sin(
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alpha * theta) * x1 / sin(theta)
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class LatentAudioDiffusionPipeline(AudioDiffusionPipeline):
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DDPMScheduler], vqvae: AutoencoderKL):
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super().__init__(unet=unet, scheduler=scheduler)
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self.register_modules(vqvae=vqvae)
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from librosa.beat import beat_track
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from diffusers import (DiffusionPipeline, UNet2DConditionModel, DDIMScheduler,
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DDPMScheduler, AutoencoderKL)
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+
from diffusers.pipeline_utils import (AudioPipelineOutput, BaseOutput,
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ImagePipelineOutput)
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from .mel import Mel
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generator=generator,
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step_generator=step_generator,
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eta=eta,
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noise=noise,
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return_dict=False)
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return images[0], (sample_rate, audios[0])
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def generate_spectrogram_and_audio_from_audio(
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mask_end_secs=mask_end_secs,
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step_generator=step_generator,
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eta=eta,
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noise=noise,
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return_dict=False)
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return images[0], (sample_rate, audios[0])
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@staticmethod
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class AudioDiffusionPipeline(DiffusionPipeline):
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def __init__(self, unet: UNet2DConditionModel, scheduler: Union[DDIMScheduler, DDPMScheduler]):
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super().__init__()
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self.register_modules(unet=unet, scheduler=scheduler)
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Returns:
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Tuple: (height, width)
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"""
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input_module = self.vqvae if hasattr(self, "vqvae") else self.unet
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# For backwards compatibility
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sample_size = (
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(input_module.sample_size, input_module.sample_size)
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if type(input_module.sample_size) == int
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else input_module.sample_size
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)
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return sample_size
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def get_default_steps(self) -> int:
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mask_end_secs: float = 0,
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step_generator: torch.Generator = None,
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eta: float = 0,
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noise: torch.Tensor = None,
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return_dict=True,
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) -> Union[
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Union[AudioPipelineOutput, ImagePipelineOutput], Tuple[List[Image.Image], Tuple[int, List[np.ndarray]]]
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]:
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"""Generate random mel spectrogram from audio input and convert to audio.
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Args:
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step_generator (torch.Generator): random number generator used to de-noise or None
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eta (float): parameter between 0 and 1 used with DDIM scheduler
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noise (torch.Tensor): noise tensor of shape (batch_size, 1, height, width) or None
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return_dict (bool): if True return AudioPipelineOutput, ImagePipelineOutput else Tuple
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Returns:
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List[PIL Image]: mel spectrograms (float, List[np.ndarray]): sample rate and raw audios
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"""
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steps = steps or self.get_default_steps()
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step_generator = step_generator or generator
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# For backwards compatibility
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if type(self.unet.sample_size) == int:
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self.unet.sample_size = (self.unet.sample_size, self.unet.sample_size)
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input_dims = self.get_input_dims()
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mel.set_resolution(x_res=input_dims[1], y_res=input_dims[0])
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if noise is None:
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noise = torch.randn(
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(batch_size, self.unet.in_channels, self.unet.sample_size[0], self.unet.sample_size[1]),
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generator=generator,
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device=self.device,
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)
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images = noise
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mask = None
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if audio_file is not None or raw_audio is not None:
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mel.load_audio(audio_file, raw_audio)
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input_image = mel.audio_slice_to_image(slice)
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input_image = np.frombuffer(input_image.tobytes(), dtype="uint8").reshape(
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(input_image.height, input_image.width)
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)
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input_image = (input_image / 255) * 2 - 1
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input_images = torch.tensor(input_image[np.newaxis, :, :], dtype=torch.float).to(self.device)
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if hasattr(self, "vqvae"):
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input_images = self.vqvae.encode(torch.unsqueeze(input_images, 0)).latent_dist.sample(
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generator=generator
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)[0]
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input_images = 0.18215 * input_images
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if start_step > 0:
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images[0, 0] = self.scheduler.add_noise(input_images, noise, self.scheduler.timesteps[start_step - 1])
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pixels_per_second = self.unet.sample_size[1] * mel.get_sample_rate() / mel.x_res / mel.hop_length
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mask_start = int(mask_start_secs * pixels_per_second)
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mask_end = int(mask_end_secs * pixels_per_second)
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mask = self.scheduler.add_noise(input_images, noise, torch.tensor(self.scheduler.timesteps[start_step:]))
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for step, t in enumerate(self.progress_bar(self.scheduler.timesteps[start_step:])):
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model_output = self.unet(images, t)["sample"]
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if isinstance(self.scheduler, DDIMScheduler):
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images = self.scheduler.step(
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model_output=model_output, timestep=t, sample=images, eta=eta, generator=step_generator
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)["prev_sample"]
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else:
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images = self.scheduler.step(
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model_output=model_output, timestep=t, sample=images, generator=step_generator
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)["prev_sample"]
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if mask is not None:
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if mask_start > 0:
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images[:, :, :, :mask_start] = mask[:, step, :, :mask_start]
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if mask_end > 0:
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images[:, :, :, -mask_end:] = mask[:, step, :, -mask_end:]
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if hasattr(self, "vqvae"):
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# 0.18215 was scaling factor used in training to ensure unit variance
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images = 1 / 0.18215 * images
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images = self.vqvae.decode(images)["sample"]
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images = (images / 2 + 0.5).clamp(0, 1)
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images = images.cpu().permute(0, 2, 3, 1).numpy()
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images = (images * 255).round().astype("uint8")
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images = list(
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map(lambda _: Image.fromarray(_[:, :, 0]), images)
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if images.shape[3] == 1
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else map(lambda _: Image.fromarray(_, mode="RGB").convert("L"), images)
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)
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audios = list(map(lambda _: mel.image_to_audio(_), images))
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if not return_dict:
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return images, (mel.get_sample_rate(), audios)
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return BaseOutput(**AudioPipelineOutput(np.array(audios)[:, np.newaxis, :]), **ImagePipelineOutput(images))
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@torch.no_grad()
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def encode(self, images: List[Image.Image], steps: int = 50) -> np.ndarray:
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# Only works with DDIM as this method is deterministic
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assert isinstance(self.scheduler, DDIMScheduler)
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self.scheduler.set_timesteps(steps)
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sample = np.array(
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[np.frombuffer(image.tobytes(), dtype="uint8").reshape((1, image.height, image.width)) for image in images]
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)
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sample = (sample / 255) * 2 - 1
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sample = torch.Tensor(sample).to(self.device)
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for t in self.progress_bar(torch.flip(self.scheduler.timesteps, (0,))):
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prev_timestep = t - self.scheduler.num_train_timesteps // self.scheduler.num_inference_steps
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alpha_prod_t = self.scheduler.alphas_cumprod[t]
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alpha_prod_t_prev = (
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self.scheduler.alphas_cumprod[prev_timestep]
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if prev_timestep >= 0
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else self.scheduler.final_alpha_cumprod
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)
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beta_prod_t = 1 - alpha_prod_t
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model_output = self.unet(sample, t)["sample"]
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pred_sample_direction = (1 - alpha_prod_t_prev) ** (0.5) * model_output
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sample = (sample - pred_sample_direction) * alpha_prod_t_prev ** (-0.5)
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sample = sample * alpha_prod_t ** (0.5) + beta_prod_t ** (0.5) * model_output
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return sample
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@staticmethod
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def slerp(x0: torch.Tensor, x1: torch.Tensor, alpha: float) -> torch.Tensor:
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"""Spherical Linear intERPolation
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Args:
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torch.Tensor: interpolated tensor
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"""
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theta = acos(torch.dot(torch.flatten(x0), torch.flatten(x1)) / torch.norm(x0) / torch.norm(x1))
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return sin((1 - alpha) * theta) * x0 / sin(theta) + sin(alpha * theta) * x1 / sin(theta)
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class LatentAudioDiffusionPipeline(AudioDiffusionPipeline):
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def __init__(
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self, unet: UNet2DConditionModel, scheduler: Union[DDIMScheduler, DDPMScheduler], vqvae: AutoencoderKL
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):
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super().__init__(unet=unet, scheduler=scheduler)
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self.register_modules(vqvae=vqvae)
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audiodiffusion/mel.py
CHANGED
@@ -1,22 +1,25 @@
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import warnings
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warnings.filterwarnings('ignore')
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import numpy as np
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from PIL import Image
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class Mel:
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"""Class to convert audio to mel spectrograms and vice versa.
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Args:
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@@ -28,17 +31,26 @@ class Mel:
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top_db (int): loudest in decibels
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n_iter (int): number of iterations for Griffin Linn mel inversion
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"""
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self.
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self.y_res = y_res
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self.sr = sample_rate
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self.n_fft = n_fft
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self.hop_length = hop_length
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self.n_mels = self.y_res
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self.slice_size = self.x_res * self.hop_length - 1
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self.top_db = top_db
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self.n_iter = n_iter
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self.audio = None
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def load_audio(self, audio_file: str = None, raw_audio: np.ndarray = None):
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"""Load audio.
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@@ -53,10 +65,7 @@ class Mel:
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# Pad with silence if necessary.
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if len(self.audio) < self.x_res * self.hop_length:
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self.audio = np.concatenate([
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self.audio,
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np.zeros((self.x_res * self.hop_length - len(self.audio), ))
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])
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def get_number_of_slices(self) -> int:
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"""Get number of slices in audio.
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@@ -75,8 +84,7 @@ class Mel:
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Returns:
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np.ndarray: audio as numpy array
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"""
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return self.audio[self.slice_size * slice:self.slice_size *
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(slice + 1)]
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def get_sample_rate(self) -> int:
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"""Get sample rate:
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@@ -95,14 +103,11 @@ class Mel:
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Returns:
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PIL Image: grayscale image of x_res x y_res
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"""
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S = librosa.feature.melspectrogram(
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hop_length=self.hop_length,
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n_mels=self.n_mels)
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log_S = librosa.power_to_db(S, ref=np.max, top_db=self.top_db)
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bytedata = (((log_S + self.top_db) * 255 / self.top_db).clip(0, 255) +
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0.5).astype(np.uint8)
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image = Image.fromarray(bytedata)
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return image
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@@ -115,14 +120,10 @@ class Mel:
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Returns:
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audio (np.ndarray): raw audio
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"""
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bytedata = np.frombuffer(image.tobytes(), dtype="uint8").reshape(
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(image.height, image.width))
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log_S = bytedata.astype("float") * self.top_db / 255 - self.top_db
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S = librosa.db_to_power(log_S)
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audio = librosa.feature.inverse.mel_to_audio(
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S,
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n_fft=self.n_fft,
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hop_length=self.hop_length,
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n_iter=self.n_iter)
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return audio
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import warnings
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warnings.filterwarnings("ignore")
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import numpy as np # noqa: E402
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import librosa # noqa: E402
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from PIL import Image # noqa: E402
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class Mel:
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def __init__(
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self,
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x_res: int = 256,
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y_res: int = 256,
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sample_rate: int = 22050,
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n_fft: int = 2048,
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hop_length: int = 512,
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top_db: int = 80,
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n_iter: int = 32,
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):
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"""Class to convert audio to mel spectrograms and vice versa.
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Args:
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top_db (int): loudest in decibels
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n_iter (int): number of iterations for Griffin Linn mel inversion
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"""
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self.hop_length = hop_length
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self.sr = sample_rate
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self.n_fft = n_fft
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self.top_db = top_db
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self.n_iter = n_iter
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self.set_resolution(x_res, y_res)
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self.audio = None
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def set_resolution(self, x_res: int, y_res: int):
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"""Set resolution.
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Args:
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x_res (int): x resolution of spectrogram (time)
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y_res (int): y resolution of spectrogram (frequency bins)
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"""
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self.x_res = x_res
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self.y_res = y_res
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self.n_mels = self.y_res
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self.slice_size = self.x_res * self.hop_length - 1
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def load_audio(self, audio_file: str = None, raw_audio: np.ndarray = None):
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"""Load audio.
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# Pad with silence if necessary.
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if len(self.audio) < self.x_res * self.hop_length:
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self.audio = np.concatenate([self.audio, np.zeros((self.x_res * self.hop_length - len(self.audio),))])
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def get_number_of_slices(self) -> int:
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"""Get number of slices in audio.
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Returns:
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np.ndarray: audio as numpy array
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"""
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return self.audio[self.slice_size * slice : self.slice_size * (slice + 1)]
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def get_sample_rate(self) -> int:
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"""Get sample rate:
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Returns:
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PIL Image: grayscale image of x_res x y_res
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"""
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S = librosa.feature.melspectrogram(
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y=self.get_audio_slice(slice), sr=self.sr, n_fft=self.n_fft, hop_length=self.hop_length, n_mels=self.n_mels
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)
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log_S = librosa.power_to_db(S, ref=np.max, top_db=self.top_db)
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bytedata = (((log_S + self.top_db) * 255 / self.top_db).clip(0, 255) + 0.5).astype(np.uint8)
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image = Image.fromarray(bytedata)
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return image
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Returns:
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audio (np.ndarray): raw audio
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"""
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bytedata = np.frombuffer(image.tobytes(), dtype="uint8").reshape((image.height, image.width))
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log_S = bytedata.astype("float") * self.top_db / 255 - self.top_db
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S = librosa.db_to_power(log_S)
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audio = librosa.feature.inverse.mel_to_audio(
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S, sr=self.sr, n_fft=self.n_fft, hop_length=self.hop_length, n_iter=self.n_iter
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)
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return audio
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notebooks/test_model.ipynb
CHANGED
@@ -61,7 +61,8 @@
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"outputs": [],
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"source": [
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"mel = Mel(x_res=256, y_res=256)\n",
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"
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]
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},
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{
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@@ -160,7 +161,7 @@
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"metadata": {},
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"outputs": [],
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"source": [
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"seed =
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"generator.manual_seed(seed)\n",
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"image, (sample_rate, audio) = audio_diffusion.generate_spectrogram_and_audio(\n",
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" generator=generator)\n",
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"overlap_samples = overlap_secs * mel.get_sample_rate()\n",
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"slice_size = mel.x_res * mel.hop_length\n",
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"stride = slice_size - overlap_samples\n",
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"generator = torch.Generator()\n",
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"seed = generator.seed()\n",
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"print(f'Seed = {seed}')\n",
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"track = np.array([])\n",
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" raw_audio=raw_audio,\n",
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" mask_start_secs=1,\n",
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" mask_end_secs=1,\n",
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" step_generator=torch.Generator())\n",
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"display(Audio(audio, rate=sample_rate))\n",
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"display(Audio(audio2, rate=sample_rate))"
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]
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"metadata": {},
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"outputs": [],
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"source": [
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"seed =
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"generator.manual_seed(seed)\n",
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"image, (sample_rate, audio) = audio_diffusion.generate_spectrogram_and_audio(\n",
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" generator=generator)\n",
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@@ -541,7 +542,7 @@
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"metadata": {},
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"outputs": [],
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"source": [
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"seed2 =
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"generator.manual_seed(seed2)\n",
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"image2, (sample_rate, audio2) = audio_diffusion.generate_spectrogram_and_audio(\n",
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" generator=generator)\n",
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"outputs": [],
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"source": [
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"mel = Mel(x_res=256, y_res=256)\n",
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"device = \"cuda\" if torch.cuda.is_available() else \"cpu\"\n",
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"generator = torch.Generator(device=device)"
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]
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},
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{
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"metadata": {},
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"outputs": [],
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"source": [
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"seed = 2391504374279719 #@param {type:\"integer\"}\n",
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"generator.manual_seed(seed)\n",
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"image, (sample_rate, audio) = audio_diffusion.generate_spectrogram_and_audio(\n",
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" generator=generator)\n",
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"overlap_samples = overlap_secs * mel.get_sample_rate()\n",
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"slice_size = mel.x_res * mel.hop_length\n",
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"stride = slice_size - overlap_samples\n",
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"generator = torch.Generator(device=device)\n",
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"seed = generator.seed()\n",
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"print(f'Seed = {seed}')\n",
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"track = np.array([])\n",
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" raw_audio=raw_audio,\n",
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" mask_start_secs=1,\n",
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" mask_end_secs=1,\n",
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" step_generator=torch.Generator(device=device))\n",
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"display(Audio(audio, rate=sample_rate))\n",
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"display(Audio(audio2, rate=sample_rate))"
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]
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"metadata": {},
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"outputs": [],
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"source": [
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"seed = 3412253600050855 #@param {type:\"integer\"}\n",
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"generator.manual_seed(seed)\n",
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"image, (sample_rate, audio) = audio_diffusion.generate_spectrogram_and_audio(\n",
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" generator=generator)\n",
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"metadata": {},
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"outputs": [],
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"source": [
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"seed2 = 7016114633369557 #@param {type:\"integer\"}\n",
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"generator.manual_seed(seed2)\n",
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"image2, (sample_rate, audio2) = audio_diffusion.generate_spectrogram_and_audio(\n",
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" generator=generator)\n",
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