Text-to-Audio
Inference Endpoints
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from transformers  import T5EncoderModel,T5TokenizerFast
import torch
from diffusers import   FluxTransformer2DModel
from torch import nn

from typing import List
from diffusers import FlowMatchEulerDiscreteScheduler
from diffusers.training_utils import compute_density_for_timestep_sampling
import copy
import torch.nn.functional as F
import numpy as np
from tqdm import tqdm

from typing import Optional,Union,List
from datasets import load_dataset, Audio
from math import pi
import inspect
import yaml



class StableAudioPositionalEmbedding(nn.Module):
    """Used for continuous time

    Adapted from stable audio open.
    
    """

    def __init__(self, dim: int):
        super().__init__()
        assert (dim % 2) == 0
        half_dim = dim // 2
        self.weights = nn.Parameter(torch.randn(half_dim))

    def forward(self, times: torch.Tensor) -> torch.Tensor:
        times = times[..., None]
        freqs = times * self.weights[None] * 2 * pi
        fouriered = torch.cat((freqs.sin(), freqs.cos()), dim=-1)
        fouriered = torch.cat((times, fouriered), dim=-1)
        return fouriered
        
class DurationEmbedder(nn.Module):
    """
    A simple linear projection model to map numbers to a latent space.

    Code is adapted from
    https://github.com/Stability-AI/stable-audio-tools

    Args:
        number_embedding_dim (`int`):
            Dimensionality of the number embeddings.
        min_value (`int`):
            The minimum value of the seconds number conditioning modules.
        max_value (`int`):
            The maximum value of the seconds number conditioning modules
        internal_dim (`int`):
            Dimensionality of the intermediate number hidden states.
    """

    def __init__(
        self,
        number_embedding_dim,
        min_value,
        max_value,
        internal_dim: Optional[int] = 256,
    ):
        super().__init__()
        self.time_positional_embedding = nn.Sequential(
            StableAudioPositionalEmbedding(internal_dim),
            nn.Linear(in_features=internal_dim + 1, out_features=number_embedding_dim),
        )

        self.number_embedding_dim = number_embedding_dim
        self.min_value = min_value
        self.max_value = max_value
        self.dtype = torch.float32 

    def forward(
        self,
        floats: torch.Tensor,
    ):
        floats = floats.clamp(self.min_value, self.max_value)

        normalized_floats = (floats - self.min_value) / (self.max_value - self.min_value)

        # Cast floats to same type as embedder
        embedder_dtype = next(self.time_positional_embedding.parameters()).dtype
        normalized_floats = normalized_floats.to(embedder_dtype)

        embedding = self.time_positional_embedding(normalized_floats)
        float_embeds = embedding.view(-1, 1, self.number_embedding_dim)

        return float_embeds


def retrieve_timesteps(
    scheduler,
    num_inference_steps: Optional[int] = None,
    device: Optional[Union[str, torch.device]] = None,
    timesteps: Optional[List[int]] = None,
    sigmas: Optional[List[float]] = None,
    **kwargs,
):

    if timesteps is not None and sigmas is not None:
        raise ValueError("Only one of `timesteps` or `sigmas` can be passed. Please choose one to set custom values")
    if timesteps is not None:
        accepts_timesteps = "timesteps" in set(inspect.signature(scheduler.set_timesteps).parameters.keys())
        if not accepts_timesteps:
            raise ValueError(
                f"The current scheduler class {scheduler.__class__}'s `set_timesteps` does not support custom"
                f" timestep schedules. Please check whether you are using the correct scheduler."
            )
        scheduler.set_timesteps(timesteps=timesteps, device=device, **kwargs)
        timesteps = scheduler.timesteps
        num_inference_steps = len(timesteps)
    elif sigmas is not None:
        accept_sigmas = "sigmas" in set(inspect.signature(scheduler.set_timesteps).parameters.keys())
        if not accept_sigmas:
            raise ValueError(
                f"The current scheduler class {scheduler.__class__}'s `set_timesteps` does not support custom"
                f" sigmas schedules. Please check whether you are using the correct scheduler."
            )
        scheduler.set_timesteps(sigmas=sigmas, device=device, **kwargs)
        timesteps = scheduler.timesteps
        num_inference_steps = len(timesteps)
    else:
        scheduler.set_timesteps(num_inference_steps, device=device, **kwargs)
        timesteps = scheduler.timesteps
    return timesteps, num_inference_steps
    


       



class TangoFlux(nn.Module):


    def __init__(self,config,initialize_reference_model=False):

        super().__init__()
        

        
        self.num_layers = config.get('num_layers', 6)
        self.num_single_layers = config.get('num_single_layers', 18)
        self.in_channels = config.get('in_channels', 64)
        self.attention_head_dim = config.get('attention_head_dim', 128)
        self.joint_attention_dim = config.get('joint_attention_dim', 1024)
        self.num_attention_heads = config.get('num_attention_heads', 8)
        self.audio_seq_len = config.get('audio_seq_len', 645)
        self.max_duration = config.get('max_duration', 30)
        self.uncondition = config.get('uncondition', False)
        self.text_encoder_name = config.get('text_encoder_name', "google/flan-t5-large")
        
        self.noise_scheduler = FlowMatchEulerDiscreteScheduler(num_train_timesteps=1000)
        self.noise_scheduler_copy = copy.deepcopy(self.noise_scheduler)
        self.max_text_seq_len = 64
        self.text_encoder = T5EncoderModel.from_pretrained(self.text_encoder_name)
        self.tokenizer = T5TokenizerFast.from_pretrained(self.text_encoder_name)
        self.text_embedding_dim = self.text_encoder.config.d_model
        
        
        self.fc = nn.Sequential(nn.Linear(self.text_embedding_dim,self.joint_attention_dim),nn.ReLU())
        self.duration_emebdder = DurationEmbedder(self.text_embedding_dim,min_value=0,max_value=self.max_duration)
        
        self.transformer = FluxTransformer2DModel(
                                     in_channels=self.in_channels,
                                     num_layers=self.num_layers,
                                     num_single_layers=self.num_single_layers,
                                     attention_head_dim=self.attention_head_dim,
                                     num_attention_heads=self.num_attention_heads,
                                     joint_attention_dim=self.joint_attention_dim,
                                     pooled_projection_dim=self.text_embedding_dim,
                                     guidance_embeds=False)

        self.beta_dpo = 2000 ## this is used for dpo training
            
            
        
        
       
        
    def get_sigmas(self,timesteps, n_dim=3, dtype=torch.float32):
        device = self.text_encoder.device
        sigmas = self.noise_scheduler_copy.sigmas.to(device=device, dtype=dtype)
        

        schedule_timesteps = self.noise_scheduler_copy.timesteps.to(device)
        timesteps = timesteps.to(device)
        step_indices = [(schedule_timesteps == t).nonzero().item() for t in timesteps]
       
        sigma = sigmas[step_indices].flatten()
        while len(sigma.shape) < n_dim:
            sigma = sigma.unsqueeze(-1)
        return sigma
    
   
    
    def encode_text_classifier_free(self, prompt: List[str], num_samples_per_prompt=1):
        device = self.text_encoder.device
        batch = self.tokenizer(
            prompt, max_length=self.tokenizer.model_max_length, padding=True, truncation=True, return_tensors="pt"
        )
        input_ids, attention_mask = batch.input_ids.to(device), batch.attention_mask.to(device)

        with torch.no_grad():
            prompt_embeds = self.text_encoder(
                input_ids=input_ids, attention_mask=attention_mask
            )[0]
                
        prompt_embeds = prompt_embeds.repeat_interleave(num_samples_per_prompt, 0)
        attention_mask = attention_mask.repeat_interleave(num_samples_per_prompt, 0)

        # get unconditional embeddings for classifier free guidance
        uncond_tokens = [""] 
        
        max_length = prompt_embeds.shape[1]
        uncond_batch = self.tokenizer(
            uncond_tokens, max_length=max_length, padding='max_length', truncation=True, return_tensors="pt",
        )
        uncond_input_ids = uncond_batch.input_ids.to(device)
        uncond_attention_mask = uncond_batch.attention_mask.to(device)
       
        with torch.no_grad():
            negative_prompt_embeds = self.text_encoder(
                input_ids=uncond_input_ids, attention_mask=uncond_attention_mask
            )[0]
                
        negative_prompt_embeds = negative_prompt_embeds.repeat_interleave(num_samples_per_prompt, 0)
        uncond_attention_mask = uncond_attention_mask.repeat_interleave(num_samples_per_prompt, 0)

        # For classifier free guidance, we need to do two forward passes.
        # 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])
        prompt_mask = torch.cat([uncond_attention_mask, attention_mask])
        boolean_prompt_mask = (prompt_mask == 1).to(device)

        return prompt_embeds, boolean_prompt_mask

    @torch.no_grad()
    def encode_text(self, prompt):
        device = self.text_encoder.device
        batch = self.tokenizer(
            prompt, max_length=self.max_text_seq_len, padding=True, truncation=True, return_tensors="pt")
        input_ids, attention_mask = batch.input_ids.to(device), batch.attention_mask.to(device)

        
       
        encoder_hidden_states = self.text_encoder(
            input_ids=input_ids, attention_mask=attention_mask)[0]
    
        boolean_encoder_mask = (attention_mask == 1).to(device)
        
        return encoder_hidden_states, boolean_encoder_mask
    
        
    def encode_duration(self,duration):
        return self.duration_emebdder(duration)


    
    @torch.no_grad()
    def inference_flow(self, prompt,
                    num_inference_steps=50,
                    timesteps=None,
                    guidance_scale=3,
                    duration=10,
                    disable_progress=False,
                    num_samples_per_prompt=1):

        '''Only tested for single inference. Haven't test for batch inference'''

        bsz = num_samples_per_prompt
        device = self.transformer.device
        scheduler = self.noise_scheduler

        if not isinstance(prompt,list):
            prompt = [prompt]
        if not isinstance(duration,torch.Tensor):
            duration = torch.tensor([duration],device=device)
        classifier_free_guidance = guidance_scale > 1.0
        duration_hidden_states = self.encode_duration(duration)
        if classifier_free_guidance:
            bsz = 2 * num_samples_per_prompt

            encoder_hidden_states, boolean_encoder_mask = self.encode_text_classifier_free(prompt, num_samples_per_prompt=num_samples_per_prompt)
            duration_hidden_states = duration_hidden_states.repeat(bsz,1,1)
        

        else:

            encoder_hidden_states, boolean_encoder_mask = self.encode_text(prompt,num_samples_per_prompt=num_samples_per_prompt)
            
        mask_expanded = boolean_encoder_mask.unsqueeze(-1).expand_as(encoder_hidden_states)
        masked_data = torch.where(mask_expanded, encoder_hidden_states, torch.tensor(float('nan')))

        pooled = torch.nanmean(masked_data, dim=1)
        pooled_projection = self.fc(pooled)

        encoder_hidden_states = torch.cat([encoder_hidden_states,duration_hidden_states],dim=1) ## (bs,seq_len,dim)
        
        sigmas = np.linspace(1.0, 1 / num_inference_steps, num_inference_steps)
        timesteps, num_inference_steps = retrieve_timesteps(
            scheduler,
            num_inference_steps,
            device,
            timesteps,
            sigmas
        )

        latents = torch.randn(num_samples_per_prompt,self.audio_seq_len,64)
        weight_dtype = latents.dtype

        progress_bar = tqdm(range(num_inference_steps), disable=disable_progress)

        txt_ids = torch.zeros(bsz,encoder_hidden_states.shape[1],3).to(device)
        audio_ids = torch.arange(self.audio_seq_len).unsqueeze(0).unsqueeze(-1).repeat(bsz,1,3).to(device)

        
        timesteps = timesteps.to(device)
        latents = latents.to(device)
        encoder_hidden_states = encoder_hidden_states.to(device)
        

        for i, t in enumerate(timesteps):
            
            latents_input = torch.cat([latents] * 2) if classifier_free_guidance else latents

        

            noise_pred = self.transformer(
                    hidden_states=latents_input,
                    # YiYi notes: divide it by 1000 for now because we scale it by 1000 in the transforme rmodel (we should not keep it but I want to keep the inputs same for the model for testing)
                    timestep=torch.tensor([t/1000],device=device),
                    guidance = None,
                    pooled_projections=pooled_projection,
                    encoder_hidden_states=encoder_hidden_states,
                    txt_ids=txt_ids,
                    img_ids=audio_ids,
                    return_dict=False,
                )[0]
            
            if 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)
            
            
            latents = scheduler.step(noise_pred, t, latents).prev_sample


        return latents

    def forward(self,
                latents,
                prompt,
                duration=torch.tensor([10]),
                sft=True
                ):


        device = latents.device
        audio_seq_length = self.audio_seq_len
        bsz = latents.shape[0]
        


        encoder_hidden_states, boolean_encoder_mask = self.encode_text(prompt)
        duration_hidden_states = self.encode_duration(duration)
            
            
        mask_expanded = boolean_encoder_mask.unsqueeze(-1).expand_as(encoder_hidden_states)
        masked_data = torch.where(mask_expanded, encoder_hidden_states, torch.tensor(float('nan')))
        pooled = torch.nanmean(masked_data, dim=1)
        pooled_projection = self.fc(pooled)

        ## Add duration hidden states to encoder hidden states
        encoder_hidden_states = torch.cat([encoder_hidden_states,duration_hidden_states],dim=1) ## (bs,seq_len,dim)

        txt_ids = torch.zeros(bsz,encoder_hidden_states.shape[1],3).to(device)
        audio_ids = torch.arange(audio_seq_length).unsqueeze(0).unsqueeze(-1).repeat(bsz,1,3).to(device)
        
        if sft:
            
            if self.uncondition:
                mask_indices = [k for k in range(len(prompt)) if random.random() < 0.1]
                if len(mask_indices) > 0:
                    encoder_hidden_states[mask_indices] = 0
            
            
            noise = torch.randn_like(latents)
            
            
            u = compute_density_for_timestep_sampling(
                    weighting_scheme='logit_normal',
                    batch_size=bsz,
                    logit_mean=0,
                    logit_std=1,
                    mode_scale=None,
                )

                
            indices = (u * self.noise_scheduler_copy.config.num_train_timesteps).long()
            timesteps = self.noise_scheduler_copy.timesteps[indices].to(device=latents.device)
            sigmas = self.get_sigmas(timesteps, n_dim=latents.ndim, dtype=latents.dtype)
            
            noisy_model_input = (1.0 - sigmas) * latents + sigmas * noise
            
            

            model_pred =  self.transformer(
                                    hidden_states=noisy_model_input,
                                    encoder_hidden_states=encoder_hidden_states,
                                    pooled_projections=pooled_projection,
                                    img_ids=audio_ids,
                                    txt_ids=txt_ids,
                                    guidance=None,
                # YiYi notes: divide it by 1000 for now because we scale it by 1000 in the transforme rmodel (we should not keep it but I want to keep the inputs same for the model for testing)
                                    timestep=timesteps/1000,
                                    return_dict=False)[0]
            
            

            target = noise - latents
            loss = torch.mean(
                        ( (model_pred.float() - target.float()) ** 2).reshape(target.shape[0], -1),
                        1,
                    )
            loss = loss.mean()
            raw_model_loss, raw_ref_loss,implicit_acc = 0,0,0 ## default this to 0 if doing sft

        else:
            encoder_hidden_states = encoder_hidden_states.repeat(2, 1, 1)
            pooled_projection = pooled_projection.repeat(2,1)
            noise = torch.randn_like(latents).chunk(2)[0].repeat(2, 1, 1) ## Have to sample same noise for preferred and rejected
            u = compute_density_for_timestep_sampling(
                    weighting_scheme='logit_normal',
                    batch_size=bsz//2,
                    logit_mean=0,
                    logit_std=1,
                    mode_scale=None,
                )

                
            indices = (u * self.noise_scheduler_copy.config.num_train_timesteps).long()
            timesteps = self.noise_scheduler_copy.timesteps[indices].to(device=latents.device)
            timesteps = timesteps.repeat(2)
            sigmas = self.get_sigmas(timesteps, n_dim=latents.ndim, dtype=latents.dtype)
            
            noisy_model_input = (1.0 - sigmas) * latents + sigmas * noise

            model_pred =  self.transformer(
                                    hidden_states=noisy_model_input,
                                    encoder_hidden_states=encoder_hidden_states,
                                    pooled_projections=pooled_projection,
                                    img_ids=audio_ids,
                                    txt_ids=txt_ids,
                                    guidance=None,
                # YiYi notes: divide it by 1000 for now because we scale it by 1000 in the transforme rmodel (we should not keep it but I want to keep the inputs same for the model for testing)
                                    timestep=timesteps/1000,
                                    return_dict=False)[0]
            target = noise - latents
            
            model_losses = F.mse_loss(model_pred.float(), target.float(), reduction="none")
            model_losses = model_losses.mean(dim=list(range(1, len(model_losses.shape))))
            model_losses_w, model_losses_l = model_losses.chunk(2)
            model_diff = model_losses_w - model_losses_l 
            raw_model_loss = 0.5 * (model_losses_w.mean() + model_losses_l.mean())

            
            with torch.no_grad():
                ref_preds = self.ref_transformer(
                                    hidden_states=noisy_model_input,
                                    encoder_hidden_states=encoder_hidden_states,
                                    pooled_projections=pooled_projection,
                                    img_ids=audio_ids,
                                    txt_ids=txt_ids,
                                    guidance=None,
                                    timestep=timesteps/1000,
                                    return_dict=False)[0]
                                
    
                ref_loss = F.mse_loss(ref_preds.float(), target.float(), reduction="none")
                ref_loss = ref_loss.mean(dim=list(range(1, len(ref_loss.shape))))
    
                ref_losses_w, ref_losses_l = ref_loss.chunk(2)
                ref_diff = ref_losses_w - ref_losses_l
                raw_ref_loss = ref_loss.mean()
                

                
                
            
            
            
            

            scale_term = -0.5 * self.beta_dpo
            inside_term = scale_term * (model_diff - ref_diff)  
            implicit_acc = (scale_term * (model_diff - ref_diff)  > 0).sum().float() / inside_term.size(0)
            loss = -1 * F.logsigmoid(inside_term).mean()  + model_losses_w.mean() 
        
        ## raw_model_loss, raw_ref_loss, implicit_acc is used to help to analyze dpo behaviour. 
        return loss, raw_model_loss, raw_ref_loss, implicit_acc