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# Copyright 2024 Stability AI and The HuggingFace Team. All rights reserved. | |
# | |
# Licensed under the Apache License, Version 2.0 (the "License"); | |
# you may not use this file except in compliance with the License. | |
# You may obtain a copy of the License at | |
# | |
# http://www.apache.org/licenses/LICENSE-2.0 | |
# | |
# Unless required by applicable law or agreed to in writing, software | |
# distributed under the License is distributed on an "AS IS" BASIS, | |
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. | |
# See the License for the specific language governing permissions and | |
# limitations under the License. | |
import inspect | |
from typing import Callable, List, Optional, Union | |
import torch | |
from transformers import ( | |
T5EncoderModel, | |
T5Tokenizer, | |
T5TokenizerFast, | |
) | |
from ...models import AutoencoderOobleck, StableAudioDiTModel | |
from ...models.embeddings import get_1d_rotary_pos_embed | |
from ...schedulers import EDMDPMSolverMultistepScheduler | |
from ...utils import ( | |
logging, | |
replace_example_docstring, | |
) | |
from ...utils.torch_utils import randn_tensor | |
from ..pipeline_utils import AudioPipelineOutput, DiffusionPipeline | |
from .modeling_stable_audio import StableAudioProjectionModel | |
logger = logging.get_logger(__name__) # pylint: disable=invalid-name | |
EXAMPLE_DOC_STRING = """ | |
Examples: | |
```py | |
>>> import scipy | |
>>> import torch | |
>>> import soundfile as sf | |
>>> from diffusers import StableAudioPipeline | |
>>> repo_id = "stabilityai/stable-audio-open-1.0" | |
>>> pipe = StableAudioPipeline.from_pretrained(repo_id, torch_dtype=torch.float16) | |
>>> pipe = pipe.to("cuda") | |
>>> # define the prompts | |
>>> prompt = "The sound of a hammer hitting a wooden surface." | |
>>> negative_prompt = "Low quality." | |
>>> # set the seed for generator | |
>>> generator = torch.Generator("cuda").manual_seed(0) | |
>>> # run the generation | |
>>> audio = pipe( | |
... prompt, | |
... negative_prompt=negative_prompt, | |
... num_inference_steps=200, | |
... audio_end_in_s=10.0, | |
... num_waveforms_per_prompt=3, | |
... generator=generator, | |
... ).audios | |
>>> output = audio[0].T.float().cpu().numpy() | |
>>> sf.write("hammer.wav", output, pipe.vae.sampling_rate) | |
``` | |
""" | |
class StableAudioPipeline(DiffusionPipeline): | |
r""" | |
Pipeline for text-to-audio generation using StableAudio. | |
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 ([`AutoencoderOobleck`]): | |
Variational Auto-Encoder (VAE) model to encode and decode images to and from latent representations. | |
text_encoder ([`~transformers.T5EncoderModel`]): | |
Frozen text-encoder. StableAudio uses the encoder of | |
[T5](https://huggingface.co/docs/transformers/model_doc/t5#transformers.T5EncoderModel), specifically the | |
[google-t5/t5-base](https://huggingface.co/google-t5/t5-base) variant. | |
projection_model ([`StableAudioProjectionModel`]): | |
A trained model used to linearly project the hidden-states from the text encoder model and the start and | |
end seconds. The projected hidden-states from the encoder and the conditional seconds are concatenated to | |
give the input to the transformer model. | |
tokenizer ([`~transformers.T5Tokenizer`]): | |
Tokenizer to tokenize text for the frozen text-encoder. | |
transformer ([`StableAudioDiTModel`]): | |
A `StableAudioDiTModel` to denoise the encoded audio latents. | |
scheduler ([`EDMDPMSolverMultistepScheduler`]): | |
A scheduler to be used in combination with `transformer` to denoise the encoded audio latents. | |
""" | |
model_cpu_offload_seq = "text_encoder->projection_model->transformer->vae" | |
def __init__( | |
self, | |
vae: AutoencoderOobleck, | |
text_encoder: T5EncoderModel, | |
projection_model: StableAudioProjectionModel, | |
tokenizer: Union[T5Tokenizer, T5TokenizerFast], | |
transformer: StableAudioDiTModel, | |
scheduler: EDMDPMSolverMultistepScheduler, | |
): | |
super().__init__() | |
self.register_modules( | |
vae=vae, | |
text_encoder=text_encoder, | |
projection_model=projection_model, | |
tokenizer=tokenizer, | |
transformer=transformer, | |
scheduler=scheduler, | |
) | |
self.rotary_embed_dim = self.transformer.config.attention_head_dim // 2 | |
# Copied from diffusers.pipelines.pipeline_utils.StableDiffusionMixin.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.pipeline_utils.StableDiffusionMixin.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 encode_prompt( | |
self, | |
prompt, | |
device, | |
do_classifier_free_guidance, | |
negative_prompt=None, | |
prompt_embeds: Optional[torch.Tensor] = None, | |
negative_prompt_embeds: Optional[torch.Tensor] = None, | |
attention_mask: Optional[torch.LongTensor] = None, | |
negative_attention_mask: Optional[torch.LongTensor] = None, | |
): | |
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] | |
if prompt_embeds is None: | |
# 1. Tokenize text | |
text_inputs = self.tokenizer( | |
prompt, | |
padding="max_length", | |
max_length=self.tokenizer.model_max_length, | |
truncation=True, | |
return_tensors="pt", | |
) | |
text_input_ids = text_inputs.input_ids | |
attention_mask = text_inputs.attention_mask | |
untruncated_ids = self.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 = self.tokenizer.batch_decode( | |
untruncated_ids[:, self.tokenizer.model_max_length - 1 : -1] | |
) | |
logger.warning( | |
f"The following part of your input was truncated because {self.text_encoder.config.model_type} can " | |
f"only handle sequences up to {self.tokenizer.model_max_length} tokens: {removed_text}" | |
) | |
text_input_ids = text_input_ids.to(device) | |
attention_mask = attention_mask.to(device) | |
# 2. Text encoder forward | |
self.text_encoder.eval() | |
prompt_embeds = self.text_encoder( | |
text_input_ids, | |
attention_mask=attention_mask, | |
) | |
prompt_embeds = prompt_embeds[0] | |
if do_classifier_free_guidance and negative_prompt is not None: | |
uncond_tokens: List[str] | |
if 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 | |
# 1. Tokenize text | |
uncond_input = self.tokenizer( | |
uncond_tokens, | |
padding="max_length", | |
max_length=self.tokenizer.model_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) | |
# 2. Text encoder forward | |
self.text_encoder.eval() | |
negative_prompt_embeds = self.text_encoder( | |
uncond_input_ids, | |
attention_mask=negative_attention_mask, | |
) | |
negative_prompt_embeds = negative_prompt_embeds[0] | |
if negative_attention_mask is not None: | |
# set the masked tokens to the null embed | |
negative_prompt_embeds = torch.where( | |
negative_attention_mask.to(torch.bool).unsqueeze(2), negative_prompt_embeds, 0.0 | |
) | |
# 3. Project prompt_embeds and negative_prompt_embeds | |
if do_classifier_free_guidance and negative_prompt_embeds is not None: | |
# For classifier free guidance, we need to do two forward passes. | |
# Here we concatenate the negative and text embeddings into a single batch | |
# to avoid doing two forward passes | |
prompt_embeds = torch.cat([negative_prompt_embeds, prompt_embeds]) | |
if attention_mask is not None and negative_attention_mask is None: | |
negative_attention_mask = torch.ones_like(attention_mask) | |
elif attention_mask is None and negative_attention_mask is not None: | |
attention_mask = torch.ones_like(negative_attention_mask) | |
if attention_mask is not None: | |
attention_mask = torch.cat([negative_attention_mask, attention_mask]) | |
prompt_embeds = self.projection_model( | |
text_hidden_states=prompt_embeds, | |
).text_hidden_states | |
if attention_mask is not None: | |
prompt_embeds = prompt_embeds * attention_mask.unsqueeze(-1).to(prompt_embeds.dtype) | |
prompt_embeds = prompt_embeds * attention_mask.unsqueeze(-1).to(prompt_embeds.dtype) | |
return prompt_embeds | |
def encode_duration( | |
self, | |
audio_start_in_s, | |
audio_end_in_s, | |
device, | |
do_classifier_free_guidance, | |
batch_size, | |
): | |
audio_start_in_s = audio_start_in_s if isinstance(audio_start_in_s, list) else [audio_start_in_s] | |
audio_end_in_s = audio_end_in_s if isinstance(audio_end_in_s, list) else [audio_end_in_s] | |
if len(audio_start_in_s) == 1: | |
audio_start_in_s = audio_start_in_s * batch_size | |
if len(audio_end_in_s) == 1: | |
audio_end_in_s = audio_end_in_s * batch_size | |
# Cast the inputs to floats | |
audio_start_in_s = [float(x) for x in audio_start_in_s] | |
audio_start_in_s = torch.tensor(audio_start_in_s).to(device) | |
audio_end_in_s = [float(x) for x in audio_end_in_s] | |
audio_end_in_s = torch.tensor(audio_end_in_s).to(device) | |
projection_output = self.projection_model( | |
start_seconds=audio_start_in_s, | |
end_seconds=audio_end_in_s, | |
) | |
seconds_start_hidden_states = projection_output.seconds_start_hidden_states | |
seconds_end_hidden_states = projection_output.seconds_end_hidden_states | |
# For classifier free guidance, we need to do two forward passes. | |
# Here we repeat the audio hidden states to avoid doing two forward passes | |
if do_classifier_free_guidance: | |
seconds_start_hidden_states = torch.cat([seconds_start_hidden_states, seconds_start_hidden_states], dim=0) | |
seconds_end_hidden_states = torch.cat([seconds_end_hidden_states, seconds_end_hidden_states], dim=0) | |
return seconds_start_hidden_states, seconds_end_hidden_states | |
# 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_start_in_s, | |
audio_end_in_s, | |
callback_steps, | |
negative_prompt=None, | |
prompt_embeds=None, | |
negative_prompt_embeds=None, | |
attention_mask=None, | |
negative_attention_mask=None, | |
initial_audio_waveforms=None, | |
initial_audio_sampling_rate=None, | |
): | |
if audio_end_in_s < audio_start_in_s: | |
raise ValueError( | |
f"`audio_end_in_s={audio_end_in_s}' must be higher than 'audio_start_in_s={audio_start_in_s}` but " | |
) | |
if ( | |
audio_start_in_s < self.projection_model.config.min_value | |
or audio_start_in_s > self.projection_model.config.max_value | |
): | |
raise ValueError( | |
f"`audio_start_in_s` must be greater than or equal to {self.projection_model.config.min_value}, and lower than or equal to {self.projection_model.config.max_value} but " | |
f"is {audio_start_in_s}." | |
) | |
if ( | |
audio_end_in_s < self.projection_model.config.min_value | |
or audio_end_in_s > self.projection_model.config.max_value | |
): | |
raise ValueError( | |
f"`audio_end_in_s` must be greater than or equal to {self.projection_model.config.min_value}, and lower than or equal to {self.projection_model.config.max_value} but " | |
f"is {audio_end_in_s}." | |
) | |
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): | |
raise ValueError( | |
"Provide either `prompt`, or `prompt_embeds`. Cannot leave" | |
"`prompt` undefined without specifying `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." | |
) | |
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 initial_audio_sampling_rate is None and initial_audio_waveforms is not None: | |
raise ValueError( | |
"`initial_audio_waveforms' is provided but the sampling rate is not. Make sure to pass `initial_audio_sampling_rate`." | |
) | |
if initial_audio_sampling_rate is not None and initial_audio_sampling_rate != self.vae.sampling_rate: | |
raise ValueError( | |
f"`initial_audio_sampling_rate` must be {self.vae.hop_length}' but is `{initial_audio_sampling_rate}`." | |
"Make sure to resample the `initial_audio_waveforms` and to correct the sampling rate. " | |
) | |
def prepare_latents( | |
self, | |
batch_size, | |
num_channels_vae, | |
sample_size, | |
dtype, | |
device, | |
generator, | |
latents=None, | |
initial_audio_waveforms=None, | |
num_waveforms_per_prompt=None, | |
audio_channels=None, | |
): | |
shape = (batch_size, num_channels_vae, sample_size) | |
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 | |
# encode the initial audio for use by the model | |
if initial_audio_waveforms is not None: | |
# check dimension | |
if initial_audio_waveforms.ndim == 2: | |
initial_audio_waveforms = initial_audio_waveforms.unsqueeze(1) | |
elif initial_audio_waveforms.ndim != 3: | |
raise ValueError( | |
f"`initial_audio_waveforms` must be of shape `(batch_size, num_channels, audio_length)` or `(batch_size, audio_length)` but has `{initial_audio_waveforms.ndim}` dimensions" | |
) | |
audio_vae_length = self.transformer.config.sample_size * self.vae.hop_length | |
audio_shape = (batch_size // num_waveforms_per_prompt, audio_channels, audio_vae_length) | |
# check num_channels | |
if initial_audio_waveforms.shape[1] == 1 and audio_channels == 2: | |
initial_audio_waveforms = initial_audio_waveforms.repeat(1, 2, 1) | |
elif initial_audio_waveforms.shape[1] == 2 and audio_channels == 1: | |
initial_audio_waveforms = initial_audio_waveforms.mean(1, keepdim=True) | |
if initial_audio_waveforms.shape[:2] != audio_shape[:2]: | |
raise ValueError( | |
f"`initial_audio_waveforms` must be of shape `(batch_size, num_channels, audio_length)` or `(batch_size, audio_length)` but is of shape `{initial_audio_waveforms.shape}`" | |
) | |
# crop or pad | |
audio_length = initial_audio_waveforms.shape[-1] | |
if audio_length < audio_vae_length: | |
logger.warning( | |
f"The provided input waveform is shorter ({audio_length}) than the required audio length ({audio_vae_length}) of the model and will thus be padded." | |
) | |
elif audio_length > audio_vae_length: | |
logger.warning( | |
f"The provided input waveform is longer ({audio_length}) than the required audio length ({audio_vae_length}) of the model and will thus be cropped." | |
) | |
audio = initial_audio_waveforms.new_zeros(audio_shape) | |
audio[:, :, : min(audio_length, audio_vae_length)] = initial_audio_waveforms[:, :, :audio_vae_length] | |
encoded_audio = self.vae.encode(audio).latent_dist.sample(generator) | |
encoded_audio = encoded_audio.repeat((num_waveforms_per_prompt, 1, 1)) | |
latents = encoded_audio + latents | |
return latents | |
def __call__( | |
self, | |
prompt: Union[str, List[str]] = None, | |
audio_end_in_s: Optional[float] = None, | |
audio_start_in_s: Optional[float] = 0.0, | |
num_inference_steps: int = 100, | |
guidance_scale: float = 7.0, | |
negative_prompt: Optional[Union[str, List[str]]] = None, | |
num_waveforms_per_prompt: Optional[int] = 1, | |
eta: float = 0.0, | |
generator: Optional[Union[torch.Generator, List[torch.Generator]]] = None, | |
latents: Optional[torch.Tensor] = None, | |
initial_audio_waveforms: Optional[torch.Tensor] = None, | |
initial_audio_sampling_rate: Optional[torch.Tensor] = None, | |
prompt_embeds: Optional[torch.Tensor] = None, | |
negative_prompt_embeds: Optional[torch.Tensor] = None, | |
attention_mask: Optional[torch.LongTensor] = None, | |
negative_attention_mask: Optional[torch.LongTensor] = None, | |
return_dict: bool = True, | |
callback: Optional[Callable[[int, int, torch.Tensor], None]] = None, | |
callback_steps: Optional[int] = 1, | |
output_type: Optional[str] = "pt", | |
): | |
r""" | |
The call function to the pipeline for generation. | |
Args: | |
prompt (`str` or `List[str]`, *optional*): | |
The prompt or prompts to guide audio generation. If not defined, you need to pass `prompt_embeds`. | |
audio_end_in_s (`float`, *optional*, defaults to 47.55): | |
Audio end index in seconds. | |
audio_start_in_s (`float`, *optional*, defaults to 0): | |
Audio start index in seconds. | |
num_inference_steps (`int`, *optional*, defaults to 100): | |
The number of denoising steps. More denoising steps usually lead to a higher quality audio at the | |
expense of slower inference. | |
guidance_scale (`float`, *optional*, defaults to 7.0): | |
A higher guidance scale value encourages the model to generate audio that is closely linked to the text | |
`prompt` at the expense of lower sound quality. Guidance scale is enabled when `guidance_scale > 1`. | |
negative_prompt (`str` or `List[str]`, *optional*): | |
The prompt or prompts to guide what to not include in audio generation. If not defined, you need to | |
pass `negative_prompt_embeds` instead. Ignored when not using guidance (`guidance_scale < 1`). | |
num_waveforms_per_prompt (`int`, *optional*, defaults to 1): | |
The number of waveforms to generate per prompt. | |
eta (`float`, *optional*, defaults to 0.0): | |
Corresponds to parameter eta (η) from the [DDIM](https://arxiv.org/abs/2010.02502) paper. Only applies | |
to the [`~schedulers.DDIMScheduler`], and is ignored in other schedulers. | |
generator (`torch.Generator` or `List[torch.Generator]`, *optional*): | |
A [`torch.Generator`](https://pytorch.org/docs/stable/generated/torch.Generator.html) to make | |
generation deterministic. | |
latents (`torch.Tensor`, *optional*): | |
Pre-generated noisy latents sampled from a Gaussian distribution, to be used as inputs for audio | |
generation. Can be used to tweak the same generation with different prompts. If not provided, a latents | |
tensor is generated by sampling using the supplied random `generator`. | |
initial_audio_waveforms (`torch.Tensor`, *optional*): | |
Optional initial audio waveforms to use as the initial audio waveform for generation. Must be of shape | |
`(batch_size, num_channels, audio_length)` or `(batch_size, audio_length)`, where `batch_size` | |
corresponds to the number of prompts passed to the model. | |
initial_audio_sampling_rate (`int`, *optional*): | |
Sampling rate of the `initial_audio_waveforms`, if they are provided. Must be the same as the model. | |
prompt_embeds (`torch.Tensor`, *optional*): | |
Pre-computed text embeddings from the text encoder 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.Tensor`, *optional*): | |
Pre-computed negative text embeddings from the text encoder 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_text_audio_duration_embeds`. | |
return_dict (`bool`, *optional*, defaults to `True`): | |
Whether or not to return a [`~pipelines.stable_diffusion.StableDiffusionPipelineOutput`] instead of a | |
plain tuple. | |
callback (`Callable`, *optional*): | |
A function that calls every `callback_steps` steps during inference. The function is called with the | |
following arguments: `callback(step: int, timestep: int, latents: torch.Tensor)`. | |
callback_steps (`int`, *optional*, defaults to 1): | |
The frequency at which the `callback` function is called. If not specified, the callback is called at | |
every step. | |
output_type (`str`, *optional*, defaults to `"pt"`): | |
The output format of the generated audio. Choose between `"np"` to return a NumPy `np.ndarray` or | |
`"pt"` to return a PyTorch `torch.Tensor` object. Set to `"latent"` to return the latent diffusion | |
model (LDM) output. | |
Examples: | |
Returns: | |
[`~pipelines.stable_diffusion.StableDiffusionPipelineOutput`] or `tuple`: | |
If `return_dict` is `True`, [`~pipelines.stable_diffusion.StableDiffusionPipelineOutput`] is returned, | |
otherwise a `tuple` is returned where the first element is a list with the generated audio. | |
""" | |
# 0. Convert audio input length from seconds to latent length | |
downsample_ratio = self.vae.hop_length | |
max_audio_length_in_s = self.transformer.config.sample_size * downsample_ratio / self.vae.config.sampling_rate | |
if audio_end_in_s is None: | |
audio_end_in_s = max_audio_length_in_s | |
if audio_end_in_s - audio_start_in_s > max_audio_length_in_s: | |
raise ValueError( | |
f"The total audio length requested ({audio_end_in_s-audio_start_in_s}s) is longer than the model maximum possible length ({max_audio_length_in_s}). Make sure that 'audio_end_in_s-audio_start_in_s<={max_audio_length_in_s}'." | |
) | |
waveform_start = int(audio_start_in_s * self.vae.config.sampling_rate) | |
waveform_end = int(audio_end_in_s * self.vae.config.sampling_rate) | |
waveform_length = int(self.transformer.config.sample_size) | |
# 1. Check inputs. Raise error if not correct | |
self.check_inputs( | |
prompt, | |
audio_start_in_s, | |
audio_end_in_s, | |
callback_steps, | |
negative_prompt, | |
prompt_embeds, | |
negative_prompt_embeds, | |
attention_mask, | |
negative_attention_mask, | |
initial_audio_waveforms, | |
initial_audio_sampling_rate, | |
) | |
# 2. Define call parameters | |
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] | |
device = self._execution_device | |
# here `guidance_scale` is defined analog to the guidance weight `w` of equation (2) | |
# of the Imagen paper: https://arxiv.org/pdf/2205.11487.pdf . `guidance_scale = 1` | |
# corresponds to doing no classifier free guidance. | |
do_classifier_free_guidance = guidance_scale > 1.0 | |
# 3. Encode input prompt | |
prompt_embeds = self.encode_prompt( | |
prompt, | |
device, | |
do_classifier_free_guidance, | |
negative_prompt, | |
prompt_embeds, | |
negative_prompt_embeds, | |
attention_mask, | |
negative_attention_mask, | |
) | |
# Encode duration | |
seconds_start_hidden_states, seconds_end_hidden_states = self.encode_duration( | |
audio_start_in_s, | |
audio_end_in_s, | |
device, | |
do_classifier_free_guidance and (negative_prompt is not None or negative_prompt_embeds is not None), | |
batch_size, | |
) | |
# Create text_audio_duration_embeds and audio_duration_embeds | |
text_audio_duration_embeds = torch.cat( | |
[prompt_embeds, seconds_start_hidden_states, seconds_end_hidden_states], dim=1 | |
) | |
audio_duration_embeds = torch.cat([seconds_start_hidden_states, seconds_end_hidden_states], dim=2) | |
# In case of classifier free guidance without negative prompt, we need to create unconditional embeddings and | |
# to concatenate it to the embeddings | |
if do_classifier_free_guidance and negative_prompt_embeds is None and negative_prompt is None: | |
negative_text_audio_duration_embeds = torch.zeros_like( | |
text_audio_duration_embeds, device=text_audio_duration_embeds.device | |
) | |
text_audio_duration_embeds = torch.cat( | |
[negative_text_audio_duration_embeds, text_audio_duration_embeds], dim=0 | |
) | |
audio_duration_embeds = torch.cat([audio_duration_embeds, audio_duration_embeds], dim=0) | |
bs_embed, seq_len, hidden_size = text_audio_duration_embeds.shape | |
# duplicate audio_duration_embeds and text_audio_duration_embeds for each generation per prompt, using mps friendly method | |
text_audio_duration_embeds = text_audio_duration_embeds.repeat(1, num_waveforms_per_prompt, 1) | |
text_audio_duration_embeds = text_audio_duration_embeds.view( | |
bs_embed * num_waveforms_per_prompt, seq_len, hidden_size | |
) | |
audio_duration_embeds = audio_duration_embeds.repeat(1, num_waveforms_per_prompt, 1) | |
audio_duration_embeds = audio_duration_embeds.view( | |
bs_embed * num_waveforms_per_prompt, -1, audio_duration_embeds.shape[-1] | |
) | |
# 4. Prepare timesteps | |
self.scheduler.set_timesteps(num_inference_steps, device=device) | |
timesteps = self.scheduler.timesteps | |
# 5. Prepare latent variables | |
num_channels_vae = self.transformer.config.in_channels | |
latents = self.prepare_latents( | |
batch_size * num_waveforms_per_prompt, | |
num_channels_vae, | |
waveform_length, | |
text_audio_duration_embeds.dtype, | |
device, | |
generator, | |
latents, | |
initial_audio_waveforms, | |
num_waveforms_per_prompt, | |
audio_channels=self.vae.config.audio_channels, | |
) | |
# 6. Prepare extra step kwargs | |
extra_step_kwargs = self.prepare_extra_step_kwargs(generator, eta) | |
# 7. Prepare rotary positional embedding | |
rotary_embedding = get_1d_rotary_pos_embed( | |
self.rotary_embed_dim, | |
latents.shape[2] + audio_duration_embeds.shape[1], | |
use_real=True, | |
repeat_interleave_real=False, | |
) | |
# 8. Denoising loop | |
num_warmup_steps = len(timesteps) - num_inference_steps * self.scheduler.order | |
with self.progress_bar(total=num_inference_steps) as progress_bar: | |
for i, t in enumerate(timesteps): | |
# expand the latents if we are doing classifier free guidance | |
latent_model_input = torch.cat([latents] * 2) if do_classifier_free_guidance else latents | |
latent_model_input = self.scheduler.scale_model_input(latent_model_input, t) | |
# predict the noise residual | |
noise_pred = self.transformer( | |
latent_model_input, | |
t.unsqueeze(0), | |
encoder_hidden_states=text_audio_duration_embeds, | |
global_hidden_states=audio_duration_embeds, | |
rotary_embedding=rotary_embedding, | |
return_dict=False, | |
)[0] | |
# perform guidance | |
if do_classifier_free_guidance: | |
noise_pred_uncond, noise_pred_text = noise_pred.chunk(2) | |
noise_pred = noise_pred_uncond + guidance_scale * (noise_pred_text - noise_pred_uncond) | |
# compute the previous noisy sample x_t -> x_t-1 | |
latents = self.scheduler.step(noise_pred, t, latents, **extra_step_kwargs).prev_sample | |
# call the callback, if provided | |
if i == len(timesteps) - 1 or ((i + 1) > num_warmup_steps and (i + 1) % self.scheduler.order == 0): | |
progress_bar.update() | |
if callback is not None and i % callback_steps == 0: | |
step_idx = i // getattr(self.scheduler, "order", 1) | |
callback(step_idx, t, latents) | |
# 9. Post-processing | |
if not output_type == "latent": | |
audio = self.vae.decode(latents).sample | |
else: | |
return AudioPipelineOutput(audios=latents) | |
audio = audio[:, :, waveform_start:waveform_end] | |
if output_type == "np": | |
audio = audio.cpu().float().numpy() | |
self.maybe_free_model_hooks() | |
if not return_dict: | |
return (audio,) | |
return AudioPipelineOutput(audios=audio) | |