1

app.py

@@ -0,0 +1,194 @@
+import gradio as gr
+import spaces
+import yaml
+import torch
+import librosa
+from diffusers import DDIMScheduler
+from transformers import AutoProcessor, ClapModel
+from model.udit import UDiT
+from vae_modules.autoencoder_wrapper import Autoencoder
+import numpy as np
+
+diffusion_config = './config/SoloAudio.yaml'
+diffusion_ckpt = './pretrained_models/soloaudio_v2.pt'
+autoencoder_path = './pretrained_models/audio-vae.pt'
+uncond_path = './pretrained_models/uncond.npz'
+sample_rate = 24000
+device = 'cuda' if torch.cuda.is_available() else 'cpu'
+
+with open(diffusion_config, 'r') as fp:
+    diff_config = yaml.safe_load(fp)
+
+v_prediction = diff_config["ddim"]["v_prediction"]
+
+clapmodel = ClapModel.from_pretrained("laion/larger_clap_general").to(device)
+processor = AutoProcessor.from_pretrained('laion/larger_clap_general')
+autoencoder = Autoencoder(autoencoder_path, 'stable_vae', quantization_first=True)
+autoencoder.eval()
+autoencoder.to(device)
+unet = UDiT(
+        **diff_config['diffwrap']['UDiT']
+    ).to(device)
+unet.load_state_dict(torch.load(diffusion_ckpt)['model'])
+unet.eval()
+
+if v_prediction:
+    print('v prediction')
+    scheduler = DDIMScheduler(**diff_config["ddim"]['diffusers'])
+else:
+    print('noise prediction')
+    scheduler = DDIMScheduler(**diff_config["ddim"]['diffusers'])
+    
+# these steps reset dtype of noise_scheduler params
+latents = torch.randn((1, 128, 128),
+                        device=device)
+noise = torch.randn(latents.shape).to(latents.device)
+timesteps = torch.randint(0, scheduler.config.num_train_timesteps,
+                            (noise.shape[0],),
+                            device=latents.device).long()
+_ = scheduler.add_noise(latents, noise, timesteps)
+
+
+def rescale_noise_cfg(noise_cfg, noise_pred_text, guidance_rescale=0.0):
+    """
+    Rescale `noise_cfg` according to `guidance_rescale`. Based on findings of [Common Diffusion Noise Schedules and
+    Sample Steps are Flawed](https://arxiv.org/pdf/2305.08891.pdf). See Section 3.4
+    """
+    std_text = noise_pred_text.std(dim=list(range(1, noise_pred_text.ndim)), keepdim=True)
+    std_cfg = noise_cfg.std(dim=list(range(1, noise_cfg.ndim)), keepdim=True)
+    # rescale the results from guidance (fixes overexposure)
+    noise_pred_rescaled = noise_cfg * (std_text / std_cfg)
+    # mix with the original results from guidance by factor guidance_rescale to avoid "plain looking" images
+    noise_cfg = guidance_rescale * noise_pred_rescaled + (1 - guidance_rescale) * noise_cfg
+    return noise_cfg
+
+@spaces.GPU
+def sample_diffusion(mixture, timbre, ddim_steps=50, eta=0, seed=2023, guidance_scale=False, guidance_rescale=0.0,):
+    with torch.no_grad():
+        scheduler.set_timesteps(ddim_steps)
+        generator = torch.Generator(device=device).manual_seed(seed)
+        # init noise
+        noise = torch.randn(mixture.shape, generator=generator, device=device)
+        pred = noise
+
+        for t in scheduler.timesteps:
+            pred = scheduler.scale_model_input(pred, t)
+            if guidance_scale:
+                uncond = torch.tensor(np.load(uncond_path)['arr_0']).unsqueeze(0).to(device)
+                pred_combined = torch.cat([pred, pred], dim=0)
+                mixture_combined = torch.cat([mixture, mixture], dim=0)
+                timbre_combined = torch.cat([timbre, uncond], dim=0)
+                output_combined = unet(x=pred_combined, timesteps=t, mixture=mixture_combined, timbre=timbre_combined)
+                output_pos, output_neg = torch.chunk(output_combined, 2, dim=0)
+
+                model_output = output_neg + guidance_scale * (output_pos - output_neg)
+                if guidance_rescale > 0.0:
+                    # avoid overexposed
+                    model_output = rescale_noise_cfg(model_output, output_pos,
+                                                    guidance_rescale=guidance_rescale)
+            else:
+                model_output = unet(x=pred, timesteps=t, mixture=mixture, timbre=timbre)
+            pred = scheduler.step(model_output=model_output, timestep=t, sample=pred,
+                                eta=eta, generator=generator).prev_sample
+
+        pred = autoencoder(embedding=pred).squeeze(1)
+
+    return pred
+
+@spaces.GPU
+def tse(gt_file_input, text_input, num_infer_steps, eta, seed, guidance_scale, guidance_rescale):
+    with torch.no_grad():
+        mixture, _ = librosa.load(gt_file_input, sr=sample_rate)
+        # Check the length of the audio in samples
+        current_length = len(mixture)
+        target_length = sample_rate * 10
+        # Cut or pad the audio to match the target length
+        if current_length > target_length:
+            # Trim the audio if it's longer than the target length
+            mixture = mixture[:target_length]
+        elif current_length < target_length:
+            # Pad the audio with zeros if it's shorter than the target length
+            padding = target_length - current_length
+            mixture = np.pad(mixture, (0, padding), mode='constant')
+        mixture = torch.tensor(mixture).unsqueeze(0).to(device)
+        mixture = autoencoder(audio=mixture.unsqueeze(1))
+
+        text_inputs = processor(
+            text=[text_input],
+            max_length=10,  # Fixed length for text
+            padding='max_length',  # Pad text to max length
+            truncation=True,  # Truncate text if it's longer than max length
+            return_tensors="pt"
+        )
+        inputs = {
+            "input_ids": text_inputs["input_ids"][0].unsqueeze(0),  # Text input IDs
+            "attention_mask": text_inputs["attention_mask"][0].unsqueeze(0),  # Attention mask for text
+        }
+        inputs = {key: value.to(device) for key, value in inputs.items()}
+        timbre = clapmodel.get_text_features(**inputs)
+
+    
+    pred = sample_diffusion(mixture, timbre, num_infer_steps, eta, seed, guidance_scale, guidance_rescale)
+    return sample_rate, pred.squeeze().cpu().numpy()
+
+
+# CSS styling (optional)
+css = """
+#col-container {
+    margin: 0 auto;
+    max-width: 1280px;
+}
+"""
+
+# Gradio Blocks layout
+with gr.Blocks(css=css, theme=gr.themes.Soft()) as demo:
+    with gr.Column(elem_id="col-container"):
+        gr.Markdown("""
+            # SoloAudio: Target Sound Extraction with Language-oriented Audio Diffusion Transformer.
+            Adjust advanced settings for more control. This space only supports a 10-second audio input now.
+            
+            Learn more about 🟣**SoloAudio** on the [SoloAudio Homepage](https://wanghelin1997.github.io/SoloAudio-Demo/).
+        """)
+
+
+        with gr.Tab("Target Sound Extraction"):
+            # Basic Input: Text prompt
+            with gr.Row():
+                gt_file_input = gr.Audio(label="Upload Audio to Extract", type="filepath", value="demo/0_mix.wav")
+                text_input = gr.Textbox(
+                    label="Text Prompt",
+                    show_label=True,
+                    max_lines=2,
+                    placeholder="Enter your prompt",
+                    container=True,
+                    value="The sound of gunshot",
+                    scale=4
+                )
+                # Run button
+                run_button = gr.Button("Extract", scale=1)
+
+            # Output Component
+            result = gr.Audio(label="Extracted Audio", type="numpy")
+
+            # Advanced settings in an Accordion
+            with gr.Accordion("Advanced Settings", open=False):
+                # Audio Length
+                guidance_scale = gr.Slider(minimum=1.0, maximum=10, step=0.1, value=3.0, label="Guidance Scale")
+                guidance_rescale = gr.Slider(minimum=0.0, maximum=1, step=0.05, value=0., label="Guidance Rescale")
+                num_infer_steps = gr.Slider(minimum=25, maximum=200, step=5, value=50, label="DDIM Steps")
+                eta = gr.Slider(minimum=0.0, maximum=1.0, step=0.1, value=0.0, label="Eta")
+                seed = gr.Slider(minimum=0, maximum=100, step=1, value=0, label="Seed")
+
+            # Define the trigger and input-output linking for generation
+            run_button.click(
+                fn=tse,
+                inputs=[gt_file_input, text_input, num_infer_steps, eta, seed, guidance_scale, guidance_rescale],
+                outputs=[result]
+            )
+            text_input.submit(fn=tse,
+                inputs=[gt_file_input, text_input, num_infer_steps, eta, seed, guidance_scale, guidance_rescale],
+                outputs=[result]
+            )
+
+    # Launch the Gradio demo
+    demo.launch()

config/SoloAudio.yaml

@@ -0,0 +1,26 @@
+version: 1.0
+
+system: "udit_rotary_v_b_1000"
+
+ddim:
+  v_prediction: true
+  diffusers:
+    num_train_timesteps: 1000
+    beta_schedule: 'scaled_linear'
+    beta_start: 0.00085
+    beta_end: 0.012
+    prediction_type: 'v_prediction'
+    rescale_betas_zero_snr: true
+    timestep_spacing: 'trailing'
+    clip_sample: false
+
+diffwrap:
+  UDiT:  
+    input_dim: 256
+    output_dim: 128
+    pos_method: 'none'
+    pos_length: 500
+    timbre_dim: 512
+    hidden_size: 768
+    depth: 12
+    num_heads: 12

model/attention.py

@@ -0,0 +1,146 @@
+import torch
+import torch.nn as nn
+import torch.nn.functional as F
+import torch.utils.checkpoint
+import einops
+from einops import rearrange, repeat
+from inspect import isfunction
+from .rotary import RotaryEmbedding
+
+
+if hasattr(nn.functional, 'scaled_dot_product_attention'):
+    ATTENTION_MODE = 'flash'
+else:
+    ATTENTION_MODE = 'math'
+print(f'attention mode is {ATTENTION_MODE}')
+
+
+def add_mask(sim, mask):
+    b, ndim = sim.shape[0], mask.ndim
+    if ndim == 3:
+        mask = rearrange(mask, "b n m -> b 1 n m")
+    if ndim == 2:
+        mask = repeat(mask, "n m -> b 1 n m", b=b)
+    max_neg_value = -torch.finfo(sim.dtype).max
+    sim = sim.masked_fill(~mask, max_neg_value)
+    return sim
+
+
+def create_mask(q_shape, k_shape, device, q_mask=None, k_mask=None):
+    def default(val, d):
+        return val if val is not None else (d() if isfunction(d) else d)
+    b, i, j, device = q_shape[0], q_shape[-2], k_shape[-2], device
+    q_mask = default(q_mask, torch.ones((b, i), device=device, dtype=torch.bool))
+    k_mask = default(k_mask, torch.ones((b, j), device=device, dtype=torch.bool))
+    attn_mask = rearrange(q_mask, 'b i -> b 1 i 1') * rearrange(k_mask, 'b j -> b 1 1 j')
+    return attn_mask
+
+
+class Attention(nn.Module):
+    def __init__(self, dim, context_dim=None, num_heads=8,
+                 qkv_bias=False, qk_scale=None, qk_norm='layernorm',
+                 attn_drop=0., proj_drop=0., rope_mode='shared'):
+        super().__init__()
+        self.num_heads = num_heads
+        head_dim = dim // num_heads
+        self.scale = qk_scale or head_dim ** -0.5
+
+        if context_dim is None:
+            self.cross_attn = False
+        else:
+            self.cross_attn = True
+
+        context_dim = dim if context_dim is None else context_dim
+
+        self.to_q = nn.Linear(dim, dim, bias=qkv_bias)
+        self.to_k = nn.Linear(context_dim, dim, bias=qkv_bias)
+        self.to_v = nn.Linear(context_dim, dim, bias=qkv_bias)
+
+        if qk_norm is None:
+            self.norm_q = nn.Identity()
+            self.norm_k = nn.Identity()
+        elif qk_norm == 'layernorm':
+            self.norm_q = nn.LayerNorm(head_dim)
+            self.norm_k = nn.LayerNorm(head_dim)
+        else:
+            raise NotImplementedError
+
+        self.attn_drop_p = attn_drop
+        self.attn_drop = nn.Dropout(attn_drop)
+        self.proj = nn.Linear(dim, dim)
+        self.proj_drop = nn.Dropout(proj_drop)
+
+        if self.cross_attn:
+            assert rope_mode == 'none'
+        self.rope_mode = rope_mode
+        if self.rope_mode == 'shared' or self.rope_mode == 'x_only':
+            self.rotary = RotaryEmbedding(dim=head_dim)
+        elif self.rope_mode == 'dual':
+            self.rotary_x = RotaryEmbedding(dim=head_dim)
+            self.rotary_c = RotaryEmbedding(dim=head_dim)
+
+    def _rotary(self, q, k, extras):
+        if self.rope_mode == 'shared':
+            q, k = self.rotary(q=q, k=k)
+        elif self.rope_mode == 'x_only':
+            q_x, k_x = self.rotary(q=q[:, :, extras:, :], k=k[:, :, extras:, :])
+            q_c, k_c = q[:, :, :extras, :], k[:, :, :extras, :]
+            q = torch.cat((q_c, q_x), dim=2)
+            k = torch.cat((k_c, k_x), dim=2)
+        elif self.rope_mode == 'dual':
+            q_x, k_x = self.rotary_x(q=q[:, :, extras:, :], k=k[:, :, extras:, :])
+            q_c, k_c = self.rotary_c(q=q[:, :, :extras, :], k=k[:, :, :extras, :])
+            q = torch.cat((q_c, q_x), dim=2)
+            k = torch.cat((k_c, k_x), dim=2)
+        elif self.rope_mode == 'none':
+            pass
+        else:
+            raise NotImplementedError
+        return q, k
+
+    def _attn(self, q, k, v, mask_binary):
+        if ATTENTION_MODE == 'flash':
+            x = F.scaled_dot_product_attention(q, k, v,
+                                               dropout_p=self.attn_drop_p,
+                                               attn_mask=mask_binary)
+            x = einops.rearrange(x, 'B H L D -> B L (H D)')
+        elif ATTENTION_MODE == 'math':
+            attn = (q @ k.transpose(-2, -1)) * self.scale
+            attn = add_mask(attn, mask_binary) if mask_binary is not None else attn
+            attn = attn.softmax(dim=-1)
+            attn = self.attn_drop(attn)
+            x = (attn @ v).transpose(1, 2)
+            x = einops.rearrange(x, 'B H L D -> B L (H D)')
+        else:
+            raise NotImplementedError
+        return x
+
+    def forward(self, x, context=None, context_mask=None, extras=0):
+        B, L, C = x.shape
+        if context is None:
+            context = x
+
+        q = self.to_q(x)
+        k = self.to_k(context)
+        v = self.to_v(context)
+
+        if context_mask is not None:
+            mask_binary = create_mask(x.shape, context.shape,
+                                      x.device, None, context_mask)
+        else:
+            mask_binary = None
+
+        q = einops.rearrange(q, 'B L (H D) -> B H L D', H=self.num_heads)
+        k = einops.rearrange(k, 'B L (H D) -> B H L D', H=self.num_heads)
+        v = einops.rearrange(v, 'B L (H D) -> B H L D', H=self.num_heads)
+
+        q = self.norm_q(q)
+        k = self.norm_k(k)
+
+        q, k = self._rotary(q, k, extras)
+
+        x = self._attn(q, k, v, mask_binary)
+
+        x = self.proj(x)
+        x = self.proj_drop(x)
+        return x

model/rotary.py

@@ -0,0 +1,91 @@
+import torch
+
+"this rope is faster than llama rope with jit script"
+
+
+def rotate_half(x):
+    x1, x2 = x.chunk(2, dim=-1)
+    return torch.cat((-x2, x1), dim=-1)
+
+
+# disable in checkpoint mode
+# @torch.jit.script
+def apply_rotary_pos_emb(x, cos, sin):
+    # NOTE: This could probably be moved to Triton
+    # Handle a possible sequence length mismatch in between q and k
+    cos = cos[:, :, : x.shape[-2], :]
+    sin = sin[:, :, : x.shape[-2], :]
+    return (x * cos) + (rotate_half(x) * sin)
+
+
+class RotaryEmbedding(torch.nn.Module):
+    """
+    The rotary position embeddings from RoFormer_ (Su et. al).
+    A crucial insight from the method is that the query and keys are
+    transformed by rotation matrices which depend on the relative positions.
+
+    Other implementations are available in the Rotary Transformer repo_ and in
+    GPT-NeoX_, GPT-NeoX was an inspiration
+
+    .. _RoFormer: https://arxiv.org/abs/2104.09864
+    .. _repo: https://github.com/ZhuiyiTechnology/roformer
+    .. _GPT-NeoX: https://github.com/EleutherAI/gpt-neox
+
+
+    .. warning: Please note that this embedding is not registered on purpose, as it is transformative
+        (it does not create the embedding dimension) and will likely be picked up (imported) on a ad-hoc basis
+    """
+
+    def __init__(self, dim: int):
+        super().__init__()
+        # Generate and save the inverse frequency buffer (non trainable)
+        inv_freq = 1.0 / (10000 ** (torch.arange(0, dim, 2).float() / dim))
+        self.register_buffer("inv_freq", inv_freq)
+        self._seq_len_cached = None
+        self._cos_cached = None
+        self._sin_cached = None
+
+    def _update_cos_sin_tables(self, x, seq_dimension=-2):
+        # expect input: B, H, L, D
+        seq_len = x.shape[seq_dimension]
+
+        # Reset the tables if the sequence length has changed,
+        # or if we're on a new device (possibly due to tracing for instance)
+        # also make sure dtype wont change
+        if (
+            seq_len != self._seq_len_cached
+            or self._cos_cached.device != x.device
+            or self._cos_cached.dtype != x.dtype
+        ):
+            self._seq_len_cached = seq_len
+            t = torch.arange(
+                x.shape[seq_dimension], device=x.device, dtype=torch.float32
+            )
+            freqs = torch.einsum("i,j->ij", t, self.inv_freq.to(x.dtype))
+            emb = torch.cat((freqs, freqs), dim=-1).to(x.device)
+
+            self._cos_cached = emb.cos()[None, None, :, :].to(x.dtype)
+            self._sin_cached = emb.sin()[None, None, :, :].to(x.dtype)
+
+        return self._cos_cached, self._sin_cached
+
+    def forward(self, q, k):
+        self._cos_cached, self._sin_cached = self._update_cos_sin_tables(
+            q.float(), seq_dimension=-2
+        )
+        if k is not None:
+            return (
+                apply_rotary_pos_emb(q.float(),
+                                     self._cos_cached,
+                                     self._sin_cached).type_as(q),
+                apply_rotary_pos_emb(k.float(),
+                                     self._cos_cached,
+                                     self._sin_cached).type_as(k),
+            )
+        else:
+            return (
+                apply_rotary_pos_emb(q.float(),
+                                     self._cos_cached,
+                                     self._sin_cached).type_as(q),
+                None
+            )

model/udit.py

@@ -0,0 +1,491 @@
+# Copyright (c) Meta Platforms, Inc. and affiliates.
+# All rights reserved.
+
+# This source code is licensed under the license found in the
+# LICENSE file in the root directory of this source tree.
+# --------------------------------------------------------
+# References:
+# GLIDE: https://github.com/openai/glide-text2im
+# MAE: https://github.com/facebookresearch/mae/blob/main/models_mae.py
+# --------------------------------------------------------
+
+import torch
+import torch.nn as nn
+import numpy as np
+import math
+import warnings
+import einops
+import torch.utils.checkpoint
+import yaml
+import torch.nn.functional as F
+from .attention import Attention
+
+
+def _no_grad_trunc_normal_(tensor, mean, std, a, b):
+    # Cut & paste from PyTorch official master until it's in a few official releases - RW
+    # Method based on https://people.sc.fsu.edu/~jburkardt/presentations/truncated_normal.pdf
+    def norm_cdf(x):
+        # Computes standard normal cumulative distribution function
+        return (1. + math.erf(x / math.sqrt(2.))) / 2.
+
+    if (mean < a - 2 * std) or (mean > b + 2 * std):
+        warnings.warn("mean is more than 2 std from [a, b] in nn.init.trunc_normal_. "
+                      "The distribution of values may be incorrect.",
+                      stacklevel=2)
+
+    with torch.no_grad():
+        # Values are generated by using a truncated uniform distribution and
+        # then using the inverse CDF for the normal distribution.
+        # Get upper and lower cdf values
+        l = norm_cdf((a - mean) / std)
+        u = norm_cdf((b - mean) / std)
+
+        # Uniformly fill tensor with values from [l, u], then translate to
+        # [2l-1, 2u-1].
+        tensor.uniform_(2 * l - 1, 2 * u - 1)
+
+        # Use inverse cdf transform for normal distribution to get truncated
+        # standard normal
+        tensor.erfinv_()
+
+        # Transform to proper mean, std
+        tensor.mul_(std * math.sqrt(2.))
+        tensor.add_(mean)
+
+        # Clamp to ensure it's in the proper range
+        tensor.clamp_(min=a, max=b)
+        return tensor
+
+
+def trunc_normal_(tensor, mean=0., std=1., a=-2., b=2.):
+    # type: (Tensor, float, float, float, float) -> Tensor
+    r"""Fills the input Tensor with values drawn from a truncated
+    normal distribution. The values are effectively drawn from the
+    normal distribution :math:`\mathcal{N}(\text{mean}, \text{std}^2)`
+    with values outside :math:`[a, b]` redrawn until they are within
+    the bounds. The method used for generating the random values works
+    best when :math:`a \leq \text{mean} \leq b`.
+    Args:
+        tensor: an n-dimensional `torch.Tensor`
+        mean: the mean of the normal distribution
+        std: the standard deviation of the normal distribution
+        a: the minimum cutoff value
+        b: the maximum cutoff value
+    Examples:
+        >>> w = torch.empty(3, 5)
+        >>> nn.init.trunc_normal_(w)
+    """
+    return _no_grad_trunc_normal_(tensor, mean, std, a, b)
+
+
+class Mlp(nn.Module):
+    def __init__(self, in_features, hidden_features=None, out_features=None, act_layer=nn.GELU, drop=0.):
+        super().__init__()
+        out_features = out_features or in_features
+        hidden_features = hidden_features or in_features
+        self.fc1 = nn.Linear(in_features, hidden_features)
+        self.act = act_layer()
+        self.fc2 = nn.Linear(hidden_features, out_features)
+        self.drop = nn.Dropout(drop)
+
+    def forward(self, x):
+        x = self.fc1(x)
+        x = self.act(x)
+        x = self.drop(x)
+        x = self.fc2(x)
+        x = self.drop(x)
+        return x
+    
+
+
+class PositionalConvEmbedding(nn.Module):
+    """
+    Relative positional embedding used in HuBERT
+    """
+
+    def __init__(self, dim=768, kernel_size=128, groups=16):
+        super().__init__()
+        self.conv = nn.Conv1d(
+            dim,
+            dim,
+            kernel_size=kernel_size,
+            padding=kernel_size // 2,
+            groups=groups,
+            bias=True
+        )
+        self.conv = nn.utils.parametrizations.weight_norm(self.conv, name="weight", dim=2)
+
+    def forward(self, x):
+        x = x.transpose(2, 1)
+        # B C T
+        x = self.conv(x)
+        x = F.gelu(x[:, :, :-1])
+        x = x.transpose(2, 1)
+        return x
+
+
+class SinusoidalPositionalEncoding(nn.Module):
+    def __init__(self, dim, length):
+        super(SinusoidalPositionalEncoding, self).__init__()
+        self.length = length
+        self.dim = dim
+        self.register_buffer('pe', self._generate_positional_encoding(length, dim))
+
+    def _generate_positional_encoding(self, length, dim):
+        pe = torch.zeros(length, dim)
+        position = torch.arange(0, length, dtype=torch.float).unsqueeze(1)
+        div_term = torch.exp(torch.arange(0, dim, 2).float() * (-math.log(10000.0) / dim))
+
+        pe[:, 0::2] = torch.sin(position * div_term)
+        pe[:, 1::2] = torch.cos(position * div_term)
+
+        pe = pe.unsqueeze(0)
+        return pe
+
+    def forward(self, x):
+        x = x + self.pe[:, :x.size(1)]
+        return x
+
+
+class PE_wrapper(nn.Module):
+    def __init__(self, dim=768, method='none', length=None):
+        super().__init__()
+        self.method = method
+        if method == 'abs':
+            # init absolute pe like UViT
+            self.length = length
+            self.abs_pe = nn.Parameter(torch.zeros(1, length, dim))
+            trunc_normal_(self.abs_pe, std=.02)
+        elif method == 'conv':
+            self.conv_pe = PositionalConvEmbedding(dim=dim)
+        elif method == 'sinu':
+            self.sinu_pe = SinusoidalPositionalEncoding(dim=dim, length=length)
+        elif method == 'none':
+            # skip pe
+            self.id = nn.Identity()
+        else:
+            raise NotImplementedError
+
+    def forward(self, x):
+        if self.method == 'abs':
+            _, L, _ = x.shape
+            assert L <= self.length
+            x = x + self.abs_pe[:, :L, :]
+        elif self.method == 'conv':
+            x = x + self.conv_pe(x)
+        elif self.method == 'sinu':
+            x = self.sinu_pe(x)
+        elif self.method == 'none':
+            x = self.id(x)
+        else:
+            raise NotImplementedError
+        return x
+    
+    
+def modulate(x, shift, scale):
+    return x * (1 + scale.unsqueeze(1)) + shift.unsqueeze(1)
+
+
+#################################################################################
+#               Embedding Layers for Timesteps and Class Labels                 #
+#################################################################################
+
+class TimestepEmbedder(nn.Module):
+    """
+    Embeds scalar timesteps into vector representations.
+    """
+    def __init__(self, hidden_size, frequency_embedding_size=256):
+        super().__init__()
+        self.mlp = nn.Sequential(
+            nn.Linear(frequency_embedding_size, hidden_size, bias=True),
+            nn.SiLU(),
+            nn.Linear(hidden_size, hidden_size, bias=True),
+        )
+        self.frequency_embedding_size = frequency_embedding_size
+
+    @staticmethod
+    def timestep_embedding(t, dim, max_period=10000):
+        """
+        Create sinusoidal timestep embeddings.
+        :param t: a 1-D Tensor of N indices, one per batch element.
+                          These may be fractional.
+        :param dim: the dimension of the output.
+        :param max_period: controls the minimum frequency of the embeddings.
+        :return: an (N, D) Tensor of positional embeddings.
+        """
+        # https://github.com/openai/glide-text2im/blob/main/glide_text2im/nn.py
+        half = dim // 2
+        freqs = torch.exp(
+            -math.log(max_period) * torch.arange(start=0, end=half, dtype=torch.float32) / half
+        ).to(device=t.device)
+        args = t[:, None].float() * freqs[None]
+        embedding = torch.cat([torch.cos(args), torch.sin(args)], dim=-1)
+        if dim % 2:
+            embedding = torch.cat([embedding, torch.zeros_like(embedding[:, :1])], dim=-1)
+        return embedding
+
+    def forward(self, t):
+        t_freq = self.timestep_embedding(t, self.frequency_embedding_size)
+        t_emb = self.mlp(t_freq)
+        return t_emb
+
+
+class LabelEmbedder(nn.Module):
+    """
+    Embeds class labels into vector representations. Also handles label dropout for classifier-free guidance.
+    """
+    def __init__(self, num_classes, hidden_size, dropout_prob):
+        super().__init__()
+        use_cfg_embedding = dropout_prob > 0
+        self.embedding_table = nn.Embedding(num_classes + use_cfg_embedding, hidden_size)
+        self.num_classes = num_classes
+        self.dropout_prob = dropout_prob
+
+    def token_drop(self, labels, force_drop_ids=None):
+        """
+        Drops labels to enable classifier-free guidance.
+        """
+        if force_drop_ids is None:
+            drop_ids = torch.rand(labels.shape[0], device=labels.device) < self.dropout_prob
+        else:
+            drop_ids = force_drop_ids == 1
+        labels = torch.where(drop_ids, self.num_classes, labels)
+        return labels
+
+    def forward(self, labels, train, force_drop_ids=None):
+        use_dropout = self.dropout_prob > 0
+        if (train and use_dropout) or (force_drop_ids is not None):
+            labels = self.token_drop(labels, force_drop_ids)
+        embeddings = self.embedding_table(labels)
+        return embeddings
+
+
+#################################################################################
+#                                 Core DiT Model                                #
+#################################################################################
+
+class DiTBlock(nn.Module):
+    """
+    A DiT block with adaptive layer norm zero (adaLN-Zero) conditioning.
+    """
+    def __init__(self, hidden_size, num_heads, mlp_ratio=4.0, skip=False, skip_norm=True, use_checkpoint=True, **block_kwargs):
+        super().__init__()
+        self.norm1 = nn.LayerNorm(hidden_size, elementwise_affine=False, eps=1e-6)
+        self.attn = Attention(hidden_size, num_heads=num_heads, qkv_bias=True, **block_kwargs)
+        self.norm2 = nn.LayerNorm(hidden_size, elementwise_affine=False, eps=1e-6)
+        mlp_hidden_dim = int(hidden_size * mlp_ratio)
+        approx_gelu = lambda: nn.GELU(approximate="tanh")
+        self.mlp = Mlp(in_features=hidden_size, hidden_features=mlp_hidden_dim, act_layer=approx_gelu, drop=0)
+        self.adaLN_modulation = nn.Sequential(
+            nn.SiLU(),
+            nn.Linear(hidden_size, 6 * hidden_size, bias=True)
+        )
+        self.skip_linear = nn.Linear(2 * hidden_size, hidden_size) if skip else None
+        self.skip_norm =  nn.LayerNorm(2 * hidden_size, elementwise_affine=False, eps=1e-6) if skip_norm else nn.Identity()
+        self.use_checkpoint = use_checkpoint
+    
+    def forward(self, x, c, skip=None):
+        if self.use_checkpoint:
+            return torch.utils.checkpoint.checkpoint(self._forward, x, c, skip)
+        else:
+            return self._forward(x, c, skip)
+
+    def _forward(self, x, c, skip=None):
+        if self.skip_linear is not None:
+            cat = torch.cat([x, skip], dim=-1)
+            cat = self.skip_norm(cat)
+            x = self.skip_linear(cat)
+        shift_msa, scale_msa, gate_msa, shift_mlp, scale_mlp, gate_mlp = self.adaLN_modulation(c).chunk(6, dim=1)
+        x = x + gate_msa.unsqueeze(1) * self.attn(modulate(self.norm1(x), shift_msa, scale_msa))
+        x = x + gate_mlp.unsqueeze(1) * self.mlp(modulate(self.norm2(x), shift_mlp, scale_mlp))
+        return x
+
+
+class FinalLayer(nn.Module):
+    """
+    The final layer of DiT.
+    """
+    def __init__(self, hidden_size, output_dim):
+        super().__init__()
+        self.norm_final = nn.LayerNorm(hidden_size, elementwise_affine=False, eps=1e-6)
+        self.linear = nn.Linear(hidden_size, output_dim, bias=True)
+        self.adaLN_modulation = nn.Sequential(
+            nn.SiLU(),
+            nn.Linear(hidden_size, 2 * hidden_size, bias=True)
+        )
+
+    def forward(self, x, c):
+        shift, scale = self.adaLN_modulation(c).chunk(2, dim=1)
+        x = modulate(self.norm_final(x), shift, scale)
+        x = self.linear(x)
+        return x
+
+
+class UDiT(nn.Module):
+    """
+    Diffusion model with a Transformer backbone.
+    """
+    def __init__(
+        self,
+        input_dim=256,
+        output_dim=128,
+        pos_method='none',
+        pos_length=500,
+        timbre_dim=512,
+        hidden_size=1152,
+        depth=28,
+        num_heads=16,
+        mlp_ratio=4.0,
+        use_checkpoint=True
+    ):
+        super().__init__()
+        self.num_heads = num_heads
+        self.input_proj = nn.Linear(input_dim, hidden_size, bias=True)
+        self.t_embedder = TimestepEmbedder(hidden_size)
+        self.pos_embed = PE_wrapper(dim=hidden_size, method=pos_method, length=pos_length)
+        self.timbre_proj = nn.Linear(timbre_dim, hidden_size, bias=True)
+
+        self.in_blocks = nn.ModuleList([
+            DiTBlock(hidden_size, num_heads, mlp_ratio=mlp_ratio, use_checkpoint=use_checkpoint) for _ in range(depth // 2)
+        ])
+        self.mid_block = DiTBlock(hidden_size, num_heads, mlp_ratio=mlp_ratio, use_checkpoint=use_checkpoint)
+        self.out_blocks = nn.ModuleList([
+            DiTBlock(hidden_size, num_heads, mlp_ratio=mlp_ratio, skip=True, use_checkpoint=use_checkpoint) for _ in range(depth // 2)
+        ])
+        
+        self.final_layer = FinalLayer(hidden_size, output_dim)
+        self.initialize_weights()
+
+    def initialize_weights(self):
+        # Initialize transformer layers:
+        def _basic_init(module):
+            if isinstance(module, nn.Linear):
+                torch.nn.init.xavier_uniform_(module.weight)
+                if module.bias is not None:
+                    nn.init.constant_(module.bias, 0)
+        self.apply(_basic_init)
+
+        # Initialize patch_embed like nn.Linear (instead of nn.Conv2d):
+        nn.init.normal_(self.input_proj.weight, std=0.02)
+        nn.init.normal_(self.timbre_proj.weight, std=0.02)
+
+        # Initialize timestep embedding MLP:
+        nn.init.normal_(self.t_embedder.mlp[0].weight, std=0.02)
+        nn.init.normal_(self.t_embedder.mlp[2].weight, std=0.02)
+
+        # Zero-out adaLN modulation layers in DiT blocks:
+        for block in self.in_blocks:
+            nn.init.constant_(self.mid_block.adaLN_modulation[-1].weight, 0)
+            nn.init.constant_(self.mid_block.adaLN_modulation[-1].bias, 0)
+        
+        nn.init.constant_(block.adaLN_modulation[-1].weight, 0)
+        nn.init.constant_(block.adaLN_modulation[-1].bias, 0)
+        
+        for block in self.out_blocks:
+            nn.init.constant_(block.adaLN_modulation[-1].weight, 0)
+            nn.init.constant_(block.adaLN_modulation[-1].bias, 0)
+
+        # Zero-out output layers:
+        nn.init.constant_(self.final_layer.adaLN_modulation[-1].weight, 0)
+        nn.init.constant_(self.final_layer.adaLN_modulation[-1].bias, 0)
+        nn.init.constant_(self.final_layer.linear.weight, 0)
+        nn.init.constant_(self.final_layer.linear.bias, 0)
+
+    def forward(self, x, timesteps, mixture, timbre):
+        """
+        Forward pass of DiT.
+        x: (N, C, H, W) tensor of spatial inputs (images or latent representations of images)
+        t: (N,) tensor of diffusion timesteps
+        y: (N,) tensor of class labels
+        """
+        x = x.transpose(2,1)
+        mixture = mixture.transpose(2,1)
+        x = self.input_proj(torch.cat((x, mixture), dim=-1))
+        x = self.pos_embed(x)
+        if not torch.is_tensor(timesteps):
+            timesteps = torch.tensor([timesteps], dtype=torch.long, device=x.device)
+        elif torch.is_tensor(timesteps) and len(timesteps.shape) == 0:
+            timesteps = timesteps[None].to(x.device)
+        t = self.t_embedder(timesteps)                   # (N, D)
+        timbre = self.timbre_proj(timbre)
+        c = t + timbre                                # (N, D)
+
+        skips = []
+        for blk in self.in_blocks:
+            x = blk(x, c)
+            skips.append(x)
+
+        x = self.mid_block(x, c)
+
+        for blk in self.out_blocks:
+            x = blk(x, c, skips.pop())
+
+        x = self.final_layer(x, c)                # (N, T, out_dim)
+        x = x.transpose(2, 1)
+        return x
+
+
+#################################################################################
+#                                   DiT Configs                                  #
+#################################################################################
+
+def DiT_XL_2(**kwargs):
+    return DiT(depth=28, hidden_size=1152, patch_size=2, num_heads=16, **kwargs)
+
+def DiT_XL_4(**kwargs):
+    return DiT(depth=28, hidden_size=1152, patch_size=4, num_heads=16, **kwargs)
+
+def DiT_XL_8(**kwargs):
+    return DiT(depth=28, hidden_size=1152, patch_size=8, num_heads=16, **kwargs)
+
+def DiT_L_2(**kwargs):
+    return DiT(depth=24, hidden_size=1024, patch_size=2, num_heads=16, **kwargs)
+
+def DiT_L_4(**kwargs):
+    return DiT(depth=24, hidden_size=1024, patch_size=4, num_heads=16, **kwargs)
+
+def DiT_L_8(**kwargs):
+    return DiT(depth=24, hidden_size=1024, patch_size=8, num_heads=16, **kwargs)
+
+def DiT_B_2(**kwargs):
+    return DiT(depth=12, hidden_size=768, patch_size=2, num_heads=12, **kwargs)
+
+def DiT_B_4(**kwargs):
+    return DiT(depth=12, hidden_size=768, patch_size=4, num_heads=12, **kwargs)
+
+def DiT_B_8(**kwargs):
+    return DiT(depth=12, hidden_size=768, patch_size=8, num_heads=12, **kwargs)
+
+def DiT_S_2(**kwargs):
+    return DiT(depth=12, hidden_size=384, patch_size=2, num_heads=6, **kwargs)
+
+def DiT_S_4(**kwargs):
+    return DiT(depth=12, hidden_size=384, patch_size=4, num_heads=6, **kwargs)
+
+def DiT_S_8(**kwargs):
+    return DiT(depth=12, hidden_size=384, patch_size=8, num_heads=6, **kwargs)
+
+
+DiT_models = {
+    'DiT-XL/2': DiT_XL_2,  'DiT-XL/4': DiT_XL_4,  'DiT-XL/8': DiT_XL_8,
+    'DiT-L/2':  DiT_L_2,   'DiT-L/4':  DiT_L_4,   'DiT-L/8':  DiT_L_8,
+    'DiT-B/2':  DiT_B_2,   'DiT-B/4':  DiT_B_4,   'DiT-B/8':  DiT_B_8,
+    'DiT-S/2':  DiT_S_2,   'DiT-S/4':  DiT_S_4,   'DiT-S/8':  DiT_S_8,
+}
+
+if __name__ == "__main__":
+    with open('/export/corpora7/HW/DPMTSE-main/src/config/DiffTSE_udit_conv_v_b_1000.yaml', 'r') as fp:
+        config = yaml.safe_load(fp)
+    device = 'cuda'
+
+    model = UDiT(
+        **config['diffwrap']['UDiT']
+    ).to(device)
+
+    x = torch.rand((1, 128, 150)).to(device)
+    t = torch.randint(0, 1000, (1, )).long().to(device)
+    mixture = torch.rand((1, 128, 150)).to(device)
+    timbre = torch.rand((1, 512)).to(device)
+
+    y = model(x, t, mixture, timbre)
+    print(y.shape)

pretrained_models/config.json

@@ -0,0 +1,122 @@
+{
+    "model_type": "autoencoder",
+    "sample_size": 12000,
+    "sample_rate": 24000,
+    "audio_channels": 1,
+    "model": {
+        "encoder": {
+            "type": "oobleck",
+            "config": {
+                "in_channels": 1,
+                "channels": 128,
+                "c_mults": [1, 2, 4, 8],
+                "strides": [2, 4, 6, 10],
+                "latent_dim": 256,
+                "use_snake": true
+            }
+        },
+        "decoder": {
+            "type": "oobleck",
+            "config": {
+                "out_channels": 1,
+                "channels": 128,
+                "c_mults": [1, 2, 4, 8],
+                "strides": [2, 4, 6, 10],
+                "latent_dim": 128,
+                "use_snake": true,
+                "final_tanh": false
+            }
+        },
+        "bottleneck": {
+            "type": "vae"
+        },
+        "latent_dim": 128,
+        "downsampling_ratio": 480,
+        "io_channels": 1
+    },
+    "training": {
+        "learning_rate": 1.5e-4,
+        "warmup_steps": 0,
+        "use_ema": false,
+        "optimizer_configs": {
+            "autoencoder": {
+                "optimizer": {
+                    "type": "AdamW",
+                    "config": {
+                        "betas": [0.8, 0.99],
+                        "lr": 1.5e-4,
+                        "weight_decay": 1e-3
+                    }
+                },
+                "scheduler": {
+                    "type": "InverseLR",
+                    "config": {
+                        "inv_gamma": 200000,
+                        "power": 0.5,
+                        "warmup": 0.999
+                    }
+                }
+            },
+            "discriminator": {
+                "optimizer": {
+                    "type": "AdamW",
+                    "config": {
+                        "betas": [0.8, 0.99],
+                        "lr": 3e-4,
+                        "weight_decay": 1e-3
+                    }
+                },
+                "scheduler": {
+                    "type": "InverseLR",
+                    "config": {
+                        "inv_gamma": 200000,
+                        "power": 0.5,
+                        "warmup": 0.999
+                    }
+                }
+            }
+        },
+        "loss_configs": {
+            "discriminator": {
+                "type": "encodec",
+                "config": {
+                    "filters": 64,
+                    "n_ffts": [1280, 640, 320, 160, 80],
+                    "hop_lengths": [320, 160, 80, 40, 20],
+                    "win_lengths": [1280, 640, 320, 160, 80]
+                },
+                "weights": {
+                    "adversarial": 0.1,
+                    "feature_matching": 5.0
+                }
+            },
+            "spectral": {
+                "type": "mrstft",
+                "config": {
+                    "fft_sizes": [1280, 640, 320, 160, 80, 40, 20],
+                    "hop_sizes": [320, 160, 80, 40, 20, 10, 5],
+                    "win_lengths": [1280, 640, 320, 160, 80, 40, 20],
+                    "perceptual_weighting": true
+                },
+                "weights": {
+                    "mrstft": 1.0
+                }
+            },
+            "time": {
+                "type": "l1",
+                "weights": {
+                    "l1": 0.0
+                }
+            },
+            "bottleneck": {
+                "type": "kl",
+                "weights": {
+                    "kl": 1e-4
+                }
+            }
+        },
+        "demo": {
+            "demo_every": 10000
+        }
+    }
+}

vae_modules/autoencoder_wrapper.py

@@ -0,0 +1,83 @@
+import torch
+import torch.nn as nn
+from .dac import DAC
+from .stable_vae import load_vae
+
+
+class Autoencoder(nn.Module):
+    def __init__(self, ckpt_path, model_type='stable_vae', quantization_first=True):
+        super(Autoencoder, self).__init__()
+        self.model_type = model_type
+        if self.model_type == 'dac':
+            model = DAC.load(ckpt_path)
+        elif self.model_type == 'stable_vae':
+            model = load_vae(ckpt_path)
+        else:
+            raise NotImplementedError(f"Model type not implemented: {self.model_type}")
+        self.ae = model.eval()
+        self.quantization_first = quantization_first
+        print(f'Autoencoder quantization first mode: {quantization_first}')
+
+    @torch.no_grad()
+    def forward(self, audio=None, embedding=None):
+        if self.model_type == 'dac':
+            return self.process_dac(audio, embedding)
+        elif self.model_type == 'encodec':
+            return self.process_encodec(audio, embedding)
+        elif self.model_type == 'stable_vae':
+            return self.process_stable_vae(audio, embedding)
+        else:
+            raise NotImplementedError(f"Model type not implemented: {self.model_type}")
+
+    def process_dac(self, audio=None, embedding=None):
+        if audio is not None:
+            z = self.ae.encoder(audio)
+            if self.quantization_first:
+                z, *_ = self.ae.quantizer(z, None)
+            return z
+        elif embedding is not None:
+            z = embedding
+            if self.quantization_first:
+                audio = self.ae.decoder(z)
+            else:
+                z, *_ = self.ae.quantizer(z, None)
+                audio = self.ae.decoder(z)
+            return audio
+        else:
+            raise ValueError("Either audio or embedding must be provided.")
+
+    def process_encodec(self, audio=None, embedding=None):
+        if audio is not None:
+            z = self.ae.encoder(audio)
+            if self.quantization_first:
+                code = self.ae.quantizer.encode(z)
+                z = self.ae.quantizer.decode(code)
+            return z
+        elif embedding is not None:
+            z = embedding
+            if self.quantization_first:
+                audio = self.ae.decoder(z)
+            else:
+                code = self.ae.quantizer.encode(z)
+                z = self.ae.quantizer.decode(code)
+                audio = self.ae.decoder(z)
+            return audio
+        else:
+            raise ValueError("Either audio or embedding must be provided.")
+
+    def process_stable_vae(self, audio=None, embedding=None):
+        if audio is not None:
+            z = self.ae.encoder(audio)
+            if self.quantization_first:
+                z = self.ae.bottleneck.encode(z)
+            return z
+        if embedding is not None:
+            z = embedding
+            if self.quantization_first:
+                audio = self.ae.decoder(z)
+            else:
+                z = self.ae.bottleneck.encode(z)
+                audio = self.ae.decoder(z)
+            return audio
+        else:
+            raise ValueError("Either audio or embedding must be provided.")

vae_modules/dac/__init__.py

@@ -0,0 +1,16 @@
+__version__ = "1.0.0"
+
+# preserved here for legacy reasons
+__model_version__ = "latest"
+
+import audiotools
+
+audiotools.ml.BaseModel.INTERN += ["dac.**"]
+audiotools.ml.BaseModel.EXTERN += ["einops"]
+
+
+from . import nn
+from . import model
+from . import utils
+from .model import DAC
+from .model import DACFile

vae_modules/dac/__main__.py

@@ -0,0 +1,36 @@
+import sys
+
+import argbind
+
+from dac.utils import download
+from dac.utils.decode import decode
+from dac.utils.encode import encode
+
+STAGES = ["encode", "decode", "download"]
+
+
+def run(stage: str):
+    """Run stages.
+
+    Parameters
+    ----------
+    stage : str
+        Stage to run
+    """
+    if stage not in STAGES:
+        raise ValueError(f"Unknown command: {stage}. Allowed commands are {STAGES}")
+    stage_fn = globals()[stage]
+
+    if stage == "download":
+        stage_fn()
+        return
+
+    stage_fn()
+
+
+if __name__ == "__main__":
+    group = sys.argv.pop(1)
+    args = argbind.parse_args(group=group)
+
+    with argbind.scope(args):
+        run(group)

vae_modules/dac/compare/encodec.py

@@ -0,0 +1,54 @@
+import torch
+from audiotools import AudioSignal
+from audiotools.ml import BaseModel
+from encodec import EncodecModel
+
+
+class Encodec(BaseModel):
+    def __init__(self, sample_rate: int = 24000, bandwidth: float = 24.0):
+        super().__init__()
+
+        if sample_rate == 24000:
+            self.model = EncodecModel.encodec_model_24khz()
+        else:
+            self.model = EncodecModel.encodec_model_48khz()
+        self.model.set_target_bandwidth(bandwidth)
+        self.sample_rate = 44100
+
+    def forward(
+        self,
+        audio_data: torch.Tensor,
+        sample_rate: int = 44100,
+        n_quantizers: int = None,
+    ):
+        signal = AudioSignal(audio_data, sample_rate)
+        signal.resample(self.model.sample_rate)
+        recons = self.model(signal.audio_data)
+        recons = AudioSignal(recons, self.model.sample_rate)
+        recons.resample(sample_rate)
+        return {"audio": recons.audio_data}
+
+
+if __name__ == "__main__":
+    import numpy as np
+    from functools import partial
+
+    model = Encodec()
+
+    for n, m in model.named_modules():
+        o = m.extra_repr()
+        p = sum([np.prod(p.size()) for p in m.parameters()])
+        fn = lambda o, p: o + f" {p/1e6:<.3f}M params."
+        setattr(m, "extra_repr", partial(fn, o=o, p=p))
+    print(model)
+    print("Total # of params: ", sum([np.prod(p.size()) for p in model.parameters()]))
+
+    length = 88200 * 2
+    x = torch.randn(1, 1, length).to(model.device)
+    x.requires_grad_(True)
+    x.retain_grad()
+
+    # Make a forward pass
+    out = model(x)["audio"]
+
+    print(x.shape, out.shape)

vae_modules/dac/model/__init__.py

@@ -0,0 +1,4 @@
+from .base import CodecMixin
+from .base import DACFile
+from .dac import DAC
+from .discriminator import Discriminator

vae_modules/dac/model/base.py

@@ -0,0 +1,294 @@
+import math
+from dataclasses import dataclass
+from pathlib import Path
+from typing import Union
+
+import numpy as np
+import torch
+import tqdm
+from audiotools import AudioSignal
+from torch import nn
+
+SUPPORTED_VERSIONS = ["1.0.0"]
+
+
+@dataclass
+class DACFile:
+    codes: torch.Tensor
+
+    # Metadata
+    chunk_length: int
+    original_length: int
+    input_db: float
+    channels: int
+    sample_rate: int
+    padding: bool
+    dac_version: str
+
+    def save(self, path):
+        artifacts = {
+            "codes": self.codes.numpy().astype(np.uint16),
+            "metadata": {
+                "input_db": self.input_db.numpy().astype(np.float32),
+                "original_length": self.original_length,
+                "sample_rate": self.sample_rate,
+                "chunk_length": self.chunk_length,
+                "channels": self.channels,
+                "padding": self.padding,
+                "dac_version": SUPPORTED_VERSIONS[-1],
+            },
+        }
+        path = Path(path).with_suffix(".dac")
+        with open(path, "wb") as f:
+            np.save(f, artifacts)
+        return path
+
+    @classmethod
+    def load(cls, path):
+        artifacts = np.load(path, allow_pickle=True)[()]
+        codes = torch.from_numpy(artifacts["codes"].astype(int))
+        if artifacts["metadata"].get("dac_version", None) not in SUPPORTED_VERSIONS:
+            raise RuntimeError(
+                f"Given file {path} can't be loaded with this version of descript-audio-codec."
+            )
+        return cls(codes=codes, **artifacts["metadata"])
+
+
+class CodecMixin:
+    @property
+    def padding(self):
+        if not hasattr(self, "_padding"):
+            self._padding = True
+        return self._padding
+
+    @padding.setter
+    def padding(self, value):
+        assert isinstance(value, bool)
+
+        layers = [
+            l for l in self.modules() if isinstance(l, (nn.Conv1d, nn.ConvTranspose1d))
+        ]
+
+        for layer in layers:
+            if value:
+                if hasattr(layer, "original_padding"):
+                    layer.padding = layer.original_padding
+            else:
+                layer.original_padding = layer.padding
+                layer.padding = tuple(0 for _ in range(len(layer.padding)))
+
+        self._padding = value
+
+    def get_delay(self):
+        # Any number works here, delay is invariant to input length
+        l_out = self.get_output_length(0)
+        L = l_out
+
+        layers = []
+        for layer in self.modules():
+            if isinstance(layer, (nn.Conv1d, nn.ConvTranspose1d)):
+                layers.append(layer)
+
+        for layer in reversed(layers):
+            d = layer.dilation[0]
+            k = layer.kernel_size[0]
+            s = layer.stride[0]
+
+            if isinstance(layer, nn.ConvTranspose1d):
+                L = ((L - d * (k - 1) - 1) / s) + 1
+            elif isinstance(layer, nn.Conv1d):
+                L = (L - 1) * s + d * (k - 1) + 1
+
+            L = math.ceil(L)
+
+        l_in = L
+
+        return (l_in - l_out) // 2
+
+    def get_output_length(self, input_length):
+        L = input_length
+        # Calculate output length
+        for layer in self.modules():
+            if isinstance(layer, (nn.Conv1d, nn.ConvTranspose1d)):
+                d = layer.dilation[0]
+                k = layer.kernel_size[0]
+                s = layer.stride[0]
+
+                if isinstance(layer, nn.Conv1d):
+                    L = ((L - d * (k - 1) - 1) / s) + 1
+                elif isinstance(layer, nn.ConvTranspose1d):
+                    L = (L - 1) * s + d * (k - 1) + 1
+
+                L = math.floor(L)
+        return L
+
+    @torch.no_grad()
+    def compress(
+        self,
+        audio_path_or_signal: Union[str, Path, AudioSignal],
+        win_duration: float = 1.0,
+        verbose: bool = False,
+        normalize_db: float = -16,
+        n_quantizers: int = None,
+    ) -> DACFile:
+        """Processes an audio signal from a file or AudioSignal object into
+        discrete codes. This function processes the signal in short windows,
+        using constant GPU memory.
+
+        Parameters
+        ----------
+        audio_path_or_signal : Union[str, Path, AudioSignal]
+            audio signal to reconstruct
+        win_duration : float, optional
+            window duration in seconds, by default 5.0
+        verbose : bool, optional
+            by default False
+        normalize_db : float, optional
+            normalize db, by default -16
+
+        Returns
+        -------
+        DACFile
+            Object containing compressed codes and metadata
+            required for decompression
+        """
+        audio_signal = audio_path_or_signal
+        if isinstance(audio_signal, (str, Path)):
+            audio_signal = AudioSignal.load_from_file_with_ffmpeg(str(audio_signal))
+
+        self.eval()
+        original_padding = self.padding
+        original_device = audio_signal.device
+
+        audio_signal = audio_signal.clone()
+        original_sr = audio_signal.sample_rate
+
+        resample_fn = audio_signal.resample
+        loudness_fn = audio_signal.loudness
+
+        # If audio is > 10 minutes long, use the ffmpeg versions
+        if audio_signal.signal_duration >= 10 * 60 * 60:
+            resample_fn = audio_signal.ffmpeg_resample
+            loudness_fn = audio_signal.ffmpeg_loudness
+
+        original_length = audio_signal.signal_length
+        resample_fn(self.sample_rate)
+        input_db = loudness_fn()
+
+        if normalize_db is not None:
+            audio_signal.normalize(normalize_db)
+        audio_signal.ensure_max_of_audio()
+
+        nb, nac, nt = audio_signal.audio_data.shape
+        audio_signal.audio_data = audio_signal.audio_data.reshape(nb * nac, 1, nt)
+        win_duration = (
+            audio_signal.signal_duration if win_duration is None else win_duration
+        )
+
+        if audio_signal.signal_duration <= win_duration:
+            # Unchunked compression (used if signal length < win duration)
+            self.padding = True
+            n_samples = nt
+            hop = nt
+        else:
+            # Chunked inference
+            self.padding = False
+            # Zero-pad signal on either side by the delay
+            audio_signal.zero_pad(self.delay, self.delay)
+            n_samples = int(win_duration * self.sample_rate)
+            # Round n_samples to nearest hop length multiple
+            n_samples = int(math.ceil(n_samples / self.hop_length) * self.hop_length)
+            hop = self.get_output_length(n_samples)
+
+        codes = []
+        range_fn = range if not verbose else tqdm.trange
+
+        for i in range_fn(0, nt, hop):
+            x = audio_signal[..., i : i + n_samples]
+            x = x.zero_pad(0, max(0, n_samples - x.shape[-1]))
+
+            audio_data = x.audio_data.to(self.device)
+            audio_data = self.preprocess(audio_data, self.sample_rate)
+            _, c, _, _, _ = self.encode(audio_data, n_quantizers)
+            codes.append(c.to(original_device))
+            chunk_length = c.shape[-1]
+
+        codes = torch.cat(codes, dim=-1)
+
+        dac_file = DACFile(
+            codes=codes,
+            chunk_length=chunk_length,
+            original_length=original_length,
+            input_db=input_db,
+            channels=nac,
+            sample_rate=original_sr,
+            padding=self.padding,
+            dac_version=SUPPORTED_VERSIONS[-1],
+        )
+
+        if n_quantizers is not None:
+            codes = codes[:, :n_quantizers, :]
+
+        self.padding = original_padding
+        return dac_file
+
+    @torch.no_grad()
+    def decompress(
+        self,
+        obj: Union[str, Path, DACFile],
+        verbose: bool = False,
+    ) -> AudioSignal:
+        """Reconstruct audio from a given .dac file
+
+        Parameters
+        ----------
+        obj : Union[str, Path, DACFile]
+            .dac file location or corresponding DACFile object.
+        verbose : bool, optional
+            Prints progress if True, by default False
+
+        Returns
+        -------
+        AudioSignal
+            Object with the reconstructed audio
+        """
+        self.eval()
+        if isinstance(obj, (str, Path)):
+            obj = DACFile.load(obj)
+
+        original_padding = self.padding
+        self.padding = obj.padding
+
+        range_fn = range if not verbose else tqdm.trange
+        codes = obj.codes
+        original_device = codes.device
+        chunk_length = obj.chunk_length
+        recons = []
+
+        for i in range_fn(0, codes.shape[-1], chunk_length):
+            c = codes[..., i : i + chunk_length].to(self.device)
+            z = self.quantizer.from_codes(c)[0]
+            r = self.decode(z)
+            recons.append(r.to(original_device))
+
+        recons = torch.cat(recons, dim=-1)
+        recons = AudioSignal(recons, self.sample_rate)
+
+        resample_fn = recons.resample
+        loudness_fn = recons.loudness
+
+        # If audio is > 10 minutes long, use the ffmpeg versions
+        if recons.signal_duration >= 10 * 60 * 60:
+            resample_fn = recons.ffmpeg_resample
+            loudness_fn = recons.ffmpeg_loudness
+
+        recons.normalize(obj.input_db)
+        resample_fn(obj.sample_rate)
+        recons = recons[..., : obj.original_length]
+        loudness_fn()
+        recons.audio_data = recons.audio_data.reshape(
+            -1, obj.channels, obj.original_length
+        )
+
+        self.padding = original_padding
+        return recons

vae_modules/dac/model/dac.py

@@ -0,0 +1,364 @@
+import math
+from typing import List
+from typing import Union
+
+import numpy as np
+import torch
+from audiotools import AudioSignal
+from audiotools.ml import BaseModel
+from torch import nn
+
+from .base import CodecMixin
+from ..nn.layers import Snake1d
+from ..nn.layers import WNConv1d
+from ..nn.layers import WNConvTranspose1d
+from ..nn.quantize import ResidualVectorQuantize
+
+
+def init_weights(m):
+    if isinstance(m, nn.Conv1d):
+        nn.init.trunc_normal_(m.weight, std=0.02)
+        nn.init.constant_(m.bias, 0)
+
+
+class ResidualUnit(nn.Module):
+    def __init__(self, dim: int = 16, dilation: int = 1):
+        super().__init__()
+        pad = ((7 - 1) * dilation) // 2
+        self.block = nn.Sequential(
+            Snake1d(dim),
+            WNConv1d(dim, dim, kernel_size=7, dilation=dilation, padding=pad),
+            Snake1d(dim),
+            WNConv1d(dim, dim, kernel_size=1),
+        )
+
+    def forward(self, x):
+        y = self.block(x)
+        pad = (x.shape[-1] - y.shape[-1]) // 2
+        if pad > 0:
+            x = x[..., pad:-pad]
+        return x + y
+
+
+class EncoderBlock(nn.Module):
+    def __init__(self, dim: int = 16, stride: int = 1):
+        super().__init__()
+        self.block = nn.Sequential(
+            ResidualUnit(dim // 2, dilation=1),
+            ResidualUnit(dim // 2, dilation=3),
+            ResidualUnit(dim // 2, dilation=9),
+            Snake1d(dim // 2),
+            WNConv1d(
+                dim // 2,
+                dim,
+                kernel_size=2 * stride,
+                stride=stride,
+                padding=math.ceil(stride / 2),
+            ),
+        )
+
+    def forward(self, x):
+        return self.block(x)
+
+
+class Encoder(nn.Module):
+    def __init__(
+        self,
+        d_model: int = 64,
+        strides: list = [2, 4, 8, 8],
+        d_latent: int = 64,
+    ):
+        super().__init__()
+        # Create first convolution
+        self.block = [WNConv1d(1, d_model, kernel_size=7, padding=3)]
+
+        # Create EncoderBlocks that double channels as they downsample by `stride`
+        for stride in strides:
+            d_model *= 2
+            self.block += [EncoderBlock(d_model, stride=stride)]
+
+        # Create last convolution
+        self.block += [
+            Snake1d(d_model),
+            WNConv1d(d_model, d_latent, kernel_size=3, padding=1),
+        ]
+
+        # Wrap black into nn.Sequential
+        self.block = nn.Sequential(*self.block)
+        self.enc_dim = d_model
+
+    def forward(self, x):
+        return self.block(x)
+
+
+class DecoderBlock(nn.Module):
+    def __init__(self, input_dim: int = 16, output_dim: int = 8, stride: int = 1):
+        super().__init__()
+        self.block = nn.Sequential(
+            Snake1d(input_dim),
+            WNConvTranspose1d(
+                input_dim,
+                output_dim,
+                kernel_size=2 * stride,
+                stride=stride,
+                padding=math.ceil(stride / 2),
+            ),
+            ResidualUnit(output_dim, dilation=1),
+            ResidualUnit(output_dim, dilation=3),
+            ResidualUnit(output_dim, dilation=9),
+        )
+
+    def forward(self, x):
+        return self.block(x)
+
+
+class Decoder(nn.Module):
+    def __init__(
+        self,
+        input_channel,
+        channels,
+        rates,
+        d_out: int = 1,
+    ):
+        super().__init__()
+
+        # Add first conv layer
+        layers = [WNConv1d(input_channel, channels, kernel_size=7, padding=3)]
+
+        # Add upsampling + MRF blocks
+        for i, stride in enumerate(rates):
+            input_dim = channels // 2**i
+            output_dim = channels // 2 ** (i + 1)
+            layers += [DecoderBlock(input_dim, output_dim, stride)]
+
+        # Add final conv layer
+        layers += [
+            Snake1d(output_dim),
+            WNConv1d(output_dim, d_out, kernel_size=7, padding=3),
+            nn.Tanh(),
+        ]
+
+        self.model = nn.Sequential(*layers)
+
+    def forward(self, x):
+        return self.model(x)
+
+
+class DAC(BaseModel, CodecMixin):
+    def __init__(
+        self,
+        encoder_dim: int = 64,
+        encoder_rates: List[int] = [2, 4, 8, 8],
+        latent_dim: int = None,
+        decoder_dim: int = 1536,
+        decoder_rates: List[int] = [8, 8, 4, 2],
+        n_codebooks: int = 9,
+        codebook_size: int = 1024,
+        codebook_dim: Union[int, list] = 8,
+        quantizer_dropout: bool = False,
+        sample_rate: int = 44100,
+    ):
+        super().__init__()
+
+        self.encoder_dim = encoder_dim
+        self.encoder_rates = encoder_rates
+        self.decoder_dim = decoder_dim
+        self.decoder_rates = decoder_rates
+        self.sample_rate = sample_rate
+
+        if latent_dim is None:
+            latent_dim = encoder_dim * (2 ** len(encoder_rates))
+
+        self.latent_dim = latent_dim
+
+        self.hop_length = np.prod(encoder_rates)
+        self.encoder = Encoder(encoder_dim, encoder_rates, latent_dim)
+
+        self.n_codebooks = n_codebooks
+        self.codebook_size = codebook_size
+        self.codebook_dim = codebook_dim
+        self.quantizer = ResidualVectorQuantize(
+            input_dim=latent_dim,
+            n_codebooks=n_codebooks,
+            codebook_size=codebook_size,
+            codebook_dim=codebook_dim,
+            quantizer_dropout=quantizer_dropout,
+        )
+
+        self.decoder = Decoder(
+            latent_dim,
+            decoder_dim,
+            decoder_rates,
+        )
+        self.sample_rate = sample_rate
+        self.apply(init_weights)
+
+        self.delay = self.get_delay()
+
+    def preprocess(self, audio_data, sample_rate):
+        if sample_rate is None:
+            sample_rate = self.sample_rate
+        assert sample_rate == self.sample_rate
+
+        length = audio_data.shape[-1]
+        right_pad = math.ceil(length / self.hop_length) * self.hop_length - length
+        audio_data = nn.functional.pad(audio_data, (0, right_pad))
+
+        return audio_data
+
+    def encode(
+        self,
+        audio_data: torch.Tensor,
+        n_quantizers: int = None,
+    ):
+        """Encode given audio data and return quantized latent codes
+
+        Parameters
+        ----------
+        audio_data : Tensor[B x 1 x T]
+            Audio data to encode
+        n_quantizers : int, optional
+            Number of quantizers to use, by default None
+            If None, all quantizers are used.
+
+        Returns
+        -------
+        dict
+            A dictionary with the following keys:
+            "z" : Tensor[B x D x T]
+                Quantized continuous representation of input
+            "codes" : Tensor[B x N x T]
+                Codebook indices for each codebook
+                (quantized discrete representation of input)
+            "latents" : Tensor[B x N*D x T]
+                Projected latents (continuous representation of input before quantization)
+            "vq/commitment_loss" : Tensor[1]
+                Commitment loss to train encoder to predict vectors closer to codebook
+                entries
+            "vq/codebook_loss" : Tensor[1]
+                Codebook loss to update the codebook
+            "length" : int
+                Number of samples in input audio
+        """
+        z = self.encoder(audio_data)
+        z, codes, latents, commitment_loss, codebook_loss = self.quantizer(
+            z, n_quantizers
+        )
+        return z, codes, latents, commitment_loss, codebook_loss
+
+    def decode(self, z: torch.Tensor):
+        """Decode given latent codes and return audio data
+
+        Parameters
+        ----------
+        z : Tensor[B x D x T]
+            Quantized continuous representation of input
+        length : int, optional
+            Number of samples in output audio, by default None
+
+        Returns
+        -------
+        dict
+            A dictionary with the following keys:
+            "audio" : Tensor[B x 1 x length]
+                Decoded audio data.
+        """
+        return self.decoder(z)
+
+    def forward(
+        self,
+        audio_data: torch.Tensor,
+        sample_rate: int = None,
+        n_quantizers: int = None,
+    ):
+        """Model forward pass
+
+        Parameters
+        ----------
+        audio_data : Tensor[B x 1 x T]
+            Audio data to encode
+        sample_rate : int, optional
+            Sample rate of audio data in Hz, by default None
+            If None, defaults to `self.sample_rate`
+        n_quantizers : int, optional
+            Number of quantizers to use, by default None.
+            If None, all quantizers are used.
+
+        Returns
+        -------
+        dict
+            A dictionary with the following keys:
+            "z" : Tensor[B x D x T]
+                Quantized continuous representation of input
+            "codes" : Tensor[B x N x T]
+                Codebook indices for each codebook
+                (quantized discrete representation of input)
+            "latents" : Tensor[B x N*D x T]
+                Projected latents (continuous representation of input before quantization)
+            "vq/commitment_loss" : Tensor[1]
+                Commitment loss to train encoder to predict vectors closer to codebook
+                entries
+            "vq/codebook_loss" : Tensor[1]
+                Codebook loss to update the codebook
+            "length" : int
+                Number of samples in input audio
+            "audio" : Tensor[B x 1 x length]
+                Decoded audio data.
+        """
+        length = audio_data.shape[-1]
+        audio_data = self.preprocess(audio_data, sample_rate)
+        z, codes, latents, commitment_loss, codebook_loss = self.encode(
+            audio_data, n_quantizers
+        )
+
+        x = self.decode(z)
+        return {
+            "audio": x[..., :length],
+            "z": z,
+            "codes": codes,
+            "latents": latents,
+            "vq/commitment_loss": commitment_loss,
+            "vq/codebook_loss": codebook_loss,
+        }
+
+
+if __name__ == "__main__":
+    import numpy as np
+    from functools import partial
+
+    model = DAC().to("cpu")
+
+    for n, m in model.named_modules():
+        o = m.extra_repr()
+        p = sum([np.prod(p.size()) for p in m.parameters()])
+        fn = lambda o, p: o + f" {p/1e6:<.3f}M params."
+        setattr(m, "extra_repr", partial(fn, o=o, p=p))
+    print(model)
+    print("Total # of params: ", sum([np.prod(p.size()) for p in model.parameters()]))
+
+    length = 88200 * 2
+    x = torch.randn(1, 1, length).to(model.device)
+    x.requires_grad_(True)
+    x.retain_grad()
+
+    # Make a forward pass
+    out = model(x)["audio"]
+    print("Input shape:", x.shape)
+    print("Output shape:", out.shape)
+
+    # Create gradient variable
+    grad = torch.zeros_like(out)
+    grad[:, :, grad.shape[-1] // 2] = 1
+
+    # Make a backward pass
+    out.backward(grad)
+
+    # Check non-zero values
+    gradmap = x.grad.squeeze(0)
+    gradmap = (gradmap != 0).sum(0)  # sum across features
+    rf = (gradmap != 0).sum()
+
+    print(f"Receptive field: {rf.item()}")
+
+    x = AudioSignal(torch.randn(1, 1, 44100 * 60), 44100)
+    model.decompress(model.compress(x, verbose=True), verbose=True)

vae_modules/dac/model/discriminator.py

@@ -0,0 +1,228 @@
+import torch
+import torch.nn as nn
+import torch.nn.functional as F
+from audiotools import AudioSignal
+from audiotools import ml
+from audiotools import STFTParams
+from einops import rearrange
+from torch.nn.utils import weight_norm
+
+
+def WNConv1d(*args, **kwargs):
+    act = kwargs.pop("act", True)
+    conv = weight_norm(nn.Conv1d(*args, **kwargs))
+    if not act:
+        return conv
+    return nn.Sequential(conv, nn.LeakyReLU(0.1))
+
+
+def WNConv2d(*args, **kwargs):
+    act = kwargs.pop("act", True)
+    conv = weight_norm(nn.Conv2d(*args, **kwargs))
+    if not act:
+        return conv
+    return nn.Sequential(conv, nn.LeakyReLU(0.1))
+
+
+class MPD(nn.Module):
+    def __init__(self, period):
+        super().__init__()
+        self.period = period
+        self.convs = nn.ModuleList(
+            [
+                WNConv2d(1, 32, (5, 1), (3, 1), padding=(2, 0)),
+                WNConv2d(32, 128, (5, 1), (3, 1), padding=(2, 0)),
+                WNConv2d(128, 512, (5, 1), (3, 1), padding=(2, 0)),
+                WNConv2d(512, 1024, (5, 1), (3, 1), padding=(2, 0)),
+                WNConv2d(1024, 1024, (5, 1), 1, padding=(2, 0)),
+            ]
+        )
+        self.conv_post = WNConv2d(
+            1024, 1, kernel_size=(3, 1), padding=(1, 0), act=False
+        )
+
+    def pad_to_period(self, x):
+        t = x.shape[-1]
+        x = F.pad(x, (0, self.period - t % self.period), mode="reflect")
+        return x
+
+    def forward(self, x):
+        fmap = []
+
+        x = self.pad_to_period(x)
+        x = rearrange(x, "b c (l p) -> b c l p", p=self.period)
+
+        for layer in self.convs:
+            x = layer(x)
+            fmap.append(x)
+
+        x = self.conv_post(x)
+        fmap.append(x)
+
+        return fmap
+
+
+class MSD(nn.Module):
+    def __init__(self, rate: int = 1, sample_rate: int = 44100):
+        super().__init__()
+        self.convs = nn.ModuleList(
+            [
+                WNConv1d(1, 16, 15, 1, padding=7),
+                WNConv1d(16, 64, 41, 4, groups=4, padding=20),
+                WNConv1d(64, 256, 41, 4, groups=16, padding=20),
+                WNConv1d(256, 1024, 41, 4, groups=64, padding=20),
+                WNConv1d(1024, 1024, 41, 4, groups=256, padding=20),
+                WNConv1d(1024, 1024, 5, 1, padding=2),
+            ]
+        )
+        self.conv_post = WNConv1d(1024, 1, 3, 1, padding=1, act=False)
+        self.sample_rate = sample_rate
+        self.rate = rate
+
+    def forward(self, x):
+        x = AudioSignal(x, self.sample_rate)
+        x.resample(self.sample_rate // self.rate)
+        x = x.audio_data
+
+        fmap = []
+
+        for l in self.convs:
+            x = l(x)
+            fmap.append(x)
+        x = self.conv_post(x)
+        fmap.append(x)
+
+        return fmap
+
+
+BANDS = [(0.0, 0.1), (0.1, 0.25), (0.25, 0.5), (0.5, 0.75), (0.75, 1.0)]
+
+
+class MRD(nn.Module):
+    def __init__(
+        self,
+        window_length: int,
+        hop_factor: float = 0.25,
+        sample_rate: int = 44100,
+        bands: list = BANDS,
+    ):
+        """Complex multi-band spectrogram discriminator.
+        Parameters
+        ----------
+        window_length : int
+            Window length of STFT.
+        hop_factor : float, optional
+            Hop factor of the STFT, defaults to ``0.25 * window_length``.
+        sample_rate : int, optional
+            Sampling rate of audio in Hz, by default 44100
+        bands : list, optional
+            Bands to run discriminator over.
+        """
+        super().__init__()
+
+        self.window_length = window_length
+        self.hop_factor = hop_factor
+        self.sample_rate = sample_rate
+        self.stft_params = STFTParams(
+            window_length=window_length,
+            hop_length=int(window_length * hop_factor),
+            match_stride=True,
+        )
+
+        n_fft = window_length // 2 + 1
+        bands = [(int(b[0] * n_fft), int(b[1] * n_fft)) for b in bands]
+        self.bands = bands
+
+        ch = 32
+        convs = lambda: nn.ModuleList(
+            [
+                WNConv2d(2, ch, (3, 9), (1, 1), padding=(1, 4)),
+                WNConv2d(ch, ch, (3, 9), (1, 2), padding=(1, 4)),
+                WNConv2d(ch, ch, (3, 9), (1, 2), padding=(1, 4)),
+                WNConv2d(ch, ch, (3, 9), (1, 2), padding=(1, 4)),
+                WNConv2d(ch, ch, (3, 3), (1, 1), padding=(1, 1)),
+            ]
+        )
+        self.band_convs = nn.ModuleList([convs() for _ in range(len(self.bands))])
+        self.conv_post = WNConv2d(ch, 1, (3, 3), (1, 1), padding=(1, 1), act=False)
+
+    def spectrogram(self, x):
+        x = AudioSignal(x, self.sample_rate, stft_params=self.stft_params)
+        x = torch.view_as_real(x.stft())
+        x = rearrange(x, "b 1 f t c -> (b 1) c t f")
+        # Split into bands
+        x_bands = [x[..., b[0] : b[1]] for b in self.bands]
+        return x_bands
+
+    def forward(self, x):
+        x_bands = self.spectrogram(x)
+        fmap = []
+
+        x = []
+        for band, stack in zip(x_bands, self.band_convs):
+            for layer in stack:
+                band = layer(band)
+                fmap.append(band)
+            x.append(band)
+
+        x = torch.cat(x, dim=-1)
+        x = self.conv_post(x)
+        fmap.append(x)
+
+        return fmap
+
+
+class Discriminator(ml.BaseModel):
+    def __init__(
+        self,
+        rates: list = [],
+        periods: list = [2, 3, 5, 7, 11],
+        fft_sizes: list = [2048, 1024, 512],
+        sample_rate: int = 44100,
+        bands: list = BANDS,
+    ):
+        """Discriminator that combines multiple discriminators.
+
+        Parameters
+        ----------
+        rates : list, optional
+            sampling rates (in Hz) to run MSD at, by default []
+            If empty, MSD is not used.
+        periods : list, optional
+            periods (of samples) to run MPD at, by default [2, 3, 5, 7, 11]
+        fft_sizes : list, optional
+            Window sizes of the FFT to run MRD at, by default [2048, 1024, 512]
+        sample_rate : int, optional
+            Sampling rate of audio in Hz, by default 44100
+        bands : list, optional
+            Bands to run MRD at, by default `BANDS`
+        """
+        super().__init__()
+        discs = []
+        discs += [MPD(p) for p in periods]
+        discs += [MSD(r, sample_rate=sample_rate) for r in rates]
+        discs += [MRD(f, sample_rate=sample_rate, bands=bands) for f in fft_sizes]
+        self.discriminators = nn.ModuleList(discs)
+
+    def preprocess(self, y):
+        # Remove DC offset
+        y = y - y.mean(dim=-1, keepdims=True)
+        # Peak normalize the volume of input audio
+        y = 0.8 * y / (y.abs().max(dim=-1, keepdim=True)[0] + 1e-9)
+        return y
+
+    def forward(self, x):
+        x = self.preprocess(x)
+        fmaps = [d(x) for d in self.discriminators]
+        return fmaps
+
+
+if __name__ == "__main__":
+    disc = Discriminator()
+    x = torch.zeros(1, 1, 44100)
+    results = disc(x)
+    for i, result in enumerate(results):
+        print(f"disc{i}")
+        for i, r in enumerate(result):
+            print(r.shape, r.mean(), r.min(), r.max())
+        print()

vae_modules/dac/nn/__init__.py

@@ -0,0 +1,3 @@
+from . import layers
+from . import loss
+from . import quantize

vae_modules/dac/nn/layers.py

@@ -0,0 +1,33 @@
+import numpy as np
+import torch
+import torch.nn as nn
+import torch.nn.functional as F
+from einops import rearrange
+from torch.nn.utils import weight_norm
+
+
+def WNConv1d(*args, **kwargs):
+    return weight_norm(nn.Conv1d(*args, **kwargs))
+
+
+def WNConvTranspose1d(*args, **kwargs):
+    return weight_norm(nn.ConvTranspose1d(*args, **kwargs))
+
+
+# Scripting this brings model speed up 1.4x
+@torch.jit.script
+def snake(x, alpha):
+    shape = x.shape
+    x = x.reshape(shape[0], shape[1], -1)
+    x = x + (alpha + 1e-9).reciprocal() * torch.sin(alpha * x).pow(2)
+    x = x.reshape(shape)
+    return x
+
+
+class Snake1d(nn.Module):
+    def __init__(self, channels):
+        super().__init__()
+        self.alpha = nn.Parameter(torch.ones(1, channels, 1))
+
+    def forward(self, x):
+        return snake(x, self.alpha)

vae_modules/dac/nn/loss.py

@@ -0,0 +1,368 @@
+import typing
+from typing import List
+
+import torch
+import torch.nn.functional as F
+from audiotools import AudioSignal
+from audiotools import STFTParams
+from torch import nn
+
+
+class L1Loss(nn.L1Loss):
+    """L1 Loss between AudioSignals. Defaults
+    to comparing ``audio_data``, but any
+    attribute of an AudioSignal can be used.
+
+    Parameters
+    ----------
+    attribute : str, optional
+        Attribute of signal to compare, defaults to ``audio_data``.
+    weight : float, optional
+        Weight of this loss, defaults to 1.0.
+
+    Implementation copied from: https://github.com/descriptinc/lyrebird-audiotools/blob/961786aa1a9d628cca0c0486e5885a457fe70c1a/audiotools/metrics/distance.py
+    """
+
+    def __init__(self, attribute: str = "audio_data", weight: float = 1.0, **kwargs):
+        self.attribute = attribute
+        self.weight = weight
+        super().__init__(**kwargs)
+
+    def forward(self, x: AudioSignal, y: AudioSignal):
+        """
+        Parameters
+        ----------
+        x : AudioSignal
+            Estimate AudioSignal
+        y : AudioSignal
+            Reference AudioSignal
+
+        Returns
+        -------
+        torch.Tensor
+            L1 loss between AudioSignal attributes.
+        """
+        if isinstance(x, AudioSignal):
+            x = getattr(x, self.attribute)
+            y = getattr(y, self.attribute)
+        return super().forward(x, y)
+
+
+class SISDRLoss(nn.Module):
+    """
+    Computes the Scale-Invariant Source-to-Distortion Ratio between a batch
+    of estimated and reference audio signals or aligned features.
+
+    Parameters
+    ----------
+    scaling : int, optional
+        Whether to use scale-invariant (True) or
+        signal-to-noise ratio (False), by default True
+    reduction : str, optional
+        How to reduce across the batch (either 'mean',
+        'sum', or none).], by default ' mean'
+    zero_mean : int, optional
+        Zero mean the references and estimates before
+        computing the loss, by default True
+    clip_min : int, optional
+        The minimum possible loss value. Helps network
+        to not focus on making already good examples better, by default None
+    weight : float, optional
+        Weight of this loss, defaults to 1.0.
+
+    Implementation copied from: https://github.com/descriptinc/lyrebird-audiotools/blob/961786aa1a9d628cca0c0486e5885a457fe70c1a/audiotools/metrics/distance.py
+    """
+
+    def __init__(
+        self,
+        scaling: int = True,
+        reduction: str = "mean",
+        zero_mean: int = True,
+        clip_min: int = None,
+        weight: float = 1.0,
+    ):
+        self.scaling = scaling
+        self.reduction = reduction
+        self.zero_mean = zero_mean
+        self.clip_min = clip_min
+        self.weight = weight
+        super().__init__()
+
+    def forward(self, x: AudioSignal, y: AudioSignal):
+        eps = 1e-8
+        # nb, nc, nt
+        if isinstance(x, AudioSignal):
+            references = x.audio_data
+            estimates = y.audio_data
+        else:
+            references = x
+            estimates = y
+
+        nb = references.shape[0]
+        references = references.reshape(nb, 1, -1).permute(0, 2, 1)
+        estimates = estimates.reshape(nb, 1, -1).permute(0, 2, 1)
+
+        # samples now on axis 1
+        if self.zero_mean:
+            mean_reference = references.mean(dim=1, keepdim=True)
+            mean_estimate = estimates.mean(dim=1, keepdim=True)
+        else:
+            mean_reference = 0
+            mean_estimate = 0
+
+        _references = references - mean_reference
+        _estimates = estimates - mean_estimate
+
+        references_projection = (_references**2).sum(dim=-2) + eps
+        references_on_estimates = (_estimates * _references).sum(dim=-2) + eps
+
+        scale = (
+            (references_on_estimates / references_projection).unsqueeze(1)
+            if self.scaling
+            else 1
+        )
+
+        e_true = scale * _references
+        e_res = _estimates - e_true
+
+        signal = (e_true**2).sum(dim=1)
+        noise = (e_res**2).sum(dim=1)
+        sdr = -10 * torch.log10(signal / noise + eps)
+
+        if self.clip_min is not None:
+            sdr = torch.clamp(sdr, min=self.clip_min)
+
+        if self.reduction == "mean":
+            sdr = sdr.mean()
+        elif self.reduction == "sum":
+            sdr = sdr.sum()
+        return sdr
+
+
+class MultiScaleSTFTLoss(nn.Module):
+    """Computes the multi-scale STFT loss from [1].
+
+    Parameters
+    ----------
+    window_lengths : List[int], optional
+        Length of each window of each STFT, by default [2048, 512]
+    loss_fn : typing.Callable, optional
+        How to compare each loss, by default nn.L1Loss()
+    clamp_eps : float, optional
+        Clamp on the log magnitude, below, by default 1e-5
+    mag_weight : float, optional
+        Weight of raw magnitude portion of loss, by default 1.0
+    log_weight : float, optional
+        Weight of log magnitude portion of loss, by default 1.0
+    pow : float, optional
+        Power to raise magnitude to before taking log, by default 2.0
+    weight : float, optional
+        Weight of this loss, by default 1.0
+    match_stride : bool, optional
+        Whether to match the stride of convolutional layers, by default False
+
+    References
+    ----------
+
+    1.  Engel, Jesse, Chenjie Gu, and Adam Roberts.
+        "DDSP: Differentiable Digital Signal Processing."
+        International Conference on Learning Representations. 2019.
+
+    Implementation copied from: https://github.com/descriptinc/lyrebird-audiotools/blob/961786aa1a9d628cca0c0486e5885a457fe70c1a/audiotools/metrics/spectral.py
+    """
+
+    def __init__(
+        self,
+        window_lengths: List[int] = [2048, 512],
+        loss_fn: typing.Callable = nn.L1Loss(),
+        clamp_eps: float = 1e-5,
+        mag_weight: float = 1.0,
+        log_weight: float = 1.0,
+        pow: float = 2.0,
+        weight: float = 1.0,
+        match_stride: bool = False,
+        window_type: str = None,
+    ):
+        super().__init__()
+        self.stft_params = [
+            STFTParams(
+                window_length=w,
+                hop_length=w // 4,
+                match_stride=match_stride,
+                window_type=window_type,
+            )
+            for w in window_lengths
+        ]
+        self.loss_fn = loss_fn
+        self.log_weight = log_weight
+        self.mag_weight = mag_weight
+        self.clamp_eps = clamp_eps
+        self.weight = weight
+        self.pow = pow
+
+    def forward(self, x: AudioSignal, y: AudioSignal):
+        """Computes multi-scale STFT between an estimate and a reference
+        signal.
+
+        Parameters
+        ----------
+        x : AudioSignal
+            Estimate signal
+        y : AudioSignal
+            Reference signal
+
+        Returns
+        -------
+        torch.Tensor
+            Multi-scale STFT loss.
+        """
+        loss = 0.0
+        for s in self.stft_params:
+            x.stft(s.window_length, s.hop_length, s.window_type)
+            y.stft(s.window_length, s.hop_length, s.window_type)
+            loss += self.log_weight * self.loss_fn(
+                x.magnitude.clamp(self.clamp_eps).pow(self.pow).log10(),
+                y.magnitude.clamp(self.clamp_eps).pow(self.pow).log10(),
+            )
+            loss += self.mag_weight * self.loss_fn(x.magnitude, y.magnitude)
+        return loss
+
+
+class MelSpectrogramLoss(nn.Module):
+    """Compute distance between mel spectrograms. Can be used
+    in a multi-scale way.
+
+    Parameters
+    ----------
+    n_mels : List[int]
+        Number of mels per STFT, by default [150, 80],
+    window_lengths : List[int], optional
+        Length of each window of each STFT, by default [2048, 512]
+    loss_fn : typing.Callable, optional
+        How to compare each loss, by default nn.L1Loss()
+    clamp_eps : float, optional
+        Clamp on the log magnitude, below, by default 1e-5
+    mag_weight : float, optional
+        Weight of raw magnitude portion of loss, by default 1.0
+    log_weight : float, optional
+        Weight of log magnitude portion of loss, by default 1.0
+    pow : float, optional
+        Power to raise magnitude to before taking log, by default 2.0
+    weight : float, optional
+        Weight of this loss, by default 1.0
+    match_stride : bool, optional
+        Whether to match the stride of convolutional layers, by default False
+
+    Implementation copied from: https://github.com/descriptinc/lyrebird-audiotools/blob/961786aa1a9d628cca0c0486e5885a457fe70c1a/audiotools/metrics/spectral.py
+    """
+
+    def __init__(
+        self,
+        n_mels: List[int] = [150, 80],
+        window_lengths: List[int] = [2048, 512],
+        loss_fn: typing.Callable = nn.L1Loss(),
+        clamp_eps: float = 1e-5,
+        mag_weight: float = 1.0,
+        log_weight: float = 1.0,
+        pow: float = 2.0,
+        weight: float = 1.0,
+        match_stride: bool = False,
+        mel_fmin: List[float] = [0.0, 0.0],
+        mel_fmax: List[float] = [None, None],
+        window_type: str = None,
+    ):
+        super().__init__()
+        self.stft_params = [
+            STFTParams(
+                window_length=w,
+                hop_length=w // 4,
+                match_stride=match_stride,
+                window_type=window_type,
+            )
+            for w in window_lengths
+        ]
+        self.n_mels = n_mels
+        self.loss_fn = loss_fn
+        self.clamp_eps = clamp_eps
+        self.log_weight = log_weight
+        self.mag_weight = mag_weight
+        self.weight = weight
+        self.mel_fmin = mel_fmin
+        self.mel_fmax = mel_fmax
+        self.pow = pow
+
+    def forward(self, x: AudioSignal, y: AudioSignal):
+        """Computes mel loss between an estimate and a reference
+        signal.
+
+        Parameters
+        ----------
+        x : AudioSignal
+            Estimate signal
+        y : AudioSignal
+            Reference signal
+
+        Returns
+        -------
+        torch.Tensor
+            Mel loss.
+        """
+        loss = 0.0
+        for n_mels, fmin, fmax, s in zip(
+            self.n_mels, self.mel_fmin, self.mel_fmax, self.stft_params
+        ):
+            kwargs = {
+                "window_length": s.window_length,
+                "hop_length": s.hop_length,
+                "window_type": s.window_type,
+            }
+            x_mels = x.mel_spectrogram(n_mels, mel_fmin=fmin, mel_fmax=fmax, **kwargs)
+            y_mels = y.mel_spectrogram(n_mels, mel_fmin=fmin, mel_fmax=fmax, **kwargs)
+
+            loss += self.log_weight * self.loss_fn(
+                x_mels.clamp(self.clamp_eps).pow(self.pow).log10(),
+                y_mels.clamp(self.clamp_eps).pow(self.pow).log10(),
+            )
+            loss += self.mag_weight * self.loss_fn(x_mels, y_mels)
+        return loss
+
+
+class GANLoss(nn.Module):
+    """
+    Computes a discriminator loss, given a discriminator on
+    generated waveforms/spectrograms compared to ground truth
+    waveforms/spectrograms. Computes the loss for both the
+    discriminator and the generator in separate functions.
+    """
+
+    def __init__(self, discriminator):
+        super().__init__()
+        self.discriminator = discriminator
+
+    def forward(self, fake, real):
+        d_fake = self.discriminator(fake.audio_data)
+        d_real = self.discriminator(real.audio_data)
+        return d_fake, d_real
+
+    def discriminator_loss(self, fake, real):
+        d_fake, d_real = self.forward(fake.clone().detach(), real)
+
+        loss_d = 0
+        for x_fake, x_real in zip(d_fake, d_real):
+            loss_d += torch.mean(x_fake[-1] ** 2)
+            loss_d += torch.mean((1 - x_real[-1]) ** 2)
+        return loss_d
+
+    def generator_loss(self, fake, real):
+        d_fake, d_real = self.forward(fake, real)
+
+        loss_g = 0
+        for x_fake in d_fake:
+            loss_g += torch.mean((1 - x_fake[-1]) ** 2)
+
+        loss_feature = 0
+
+        for i in range(len(d_fake)):
+            for j in range(len(d_fake[i]) - 1):
+                loss_feature += F.l1_loss(d_fake[i][j], d_real[i][j].detach())
+        return loss_g, loss_feature

vae_modules/dac/nn/quantize.py

@@ -0,0 +1,262 @@
+from typing import Union
+
+import numpy as np
+import torch
+import torch.nn as nn
+import torch.nn.functional as F
+from einops import rearrange
+from torch.nn.utils import weight_norm
+
+from .layers import WNConv1d
+
+
+class VectorQuantize(nn.Module):
+    """
+    Implementation of VQ similar to Karpathy's repo:
+    https://github.com/karpathy/deep-vector-quantization
+    Additionally uses following tricks from Improved VQGAN
+    (https://arxiv.org/pdf/2110.04627.pdf):
+        1. Factorized codes: Perform nearest neighbor lookup in low-dimensional space
+            for improved codebook usage
+        2. l2-normalized codes: Converts euclidean distance to cosine similarity which
+            improves training stability
+    """
+
+    def __init__(self, input_dim: int, codebook_size: int, codebook_dim: int):
+        super().__init__()
+        self.codebook_size = codebook_size
+        self.codebook_dim = codebook_dim
+
+        self.in_proj = WNConv1d(input_dim, codebook_dim, kernel_size=1)
+        self.out_proj = WNConv1d(codebook_dim, input_dim, kernel_size=1)
+        self.codebook = nn.Embedding(codebook_size, codebook_dim)
+
+    def forward(self, z):
+        """Quantized the input tensor using a fixed codebook and returns
+        the corresponding codebook vectors
+
+        Parameters
+        ----------
+        z : Tensor[B x D x T]
+
+        Returns
+        -------
+        Tensor[B x D x T]
+            Quantized continuous representation of input
+        Tensor[1]
+            Commitment loss to train encoder to predict vectors closer to codebook
+            entries
+        Tensor[1]
+            Codebook loss to update the codebook
+        Tensor[B x T]
+            Codebook indices (quantized discrete representation of input)
+        Tensor[B x D x T]
+            Projected latents (continuous representation of input before quantization)
+        """
+
+        # Factorized codes (ViT-VQGAN) Project input into low-dimensional space
+        z_e = self.in_proj(z)  # z_e : (B x D x T)
+        z_q, indices = self.decode_latents(z_e)
+
+        commitment_loss = F.mse_loss(z_e, z_q.detach(), reduction="none").mean([1, 2])
+        codebook_loss = F.mse_loss(z_q, z_e.detach(), reduction="none").mean([1, 2])
+
+        z_q = (
+            z_e + (z_q - z_e).detach()
+        )  # noop in forward pass, straight-through gradient estimator in backward pass
+
+        z_q = self.out_proj(z_q)
+
+        return z_q, commitment_loss, codebook_loss, indices, z_e
+
+    def embed_code(self, embed_id):
+        return F.embedding(embed_id, self.codebook.weight)
+
+    def decode_code(self, embed_id):
+        return self.embed_code(embed_id).transpose(1, 2)
+
+    def decode_latents(self, latents):
+        encodings = rearrange(latents, "b d t -> (b t) d")
+        codebook = self.codebook.weight  # codebook: (N x D)
+
+        # L2 normalize encodings and codebook (ViT-VQGAN)
+        encodings = F.normalize(encodings)
+        codebook = F.normalize(codebook)
+
+        # Compute euclidean distance with codebook
+        dist = (
+            encodings.pow(2).sum(1, keepdim=True)
+            - 2 * encodings @ codebook.t()
+            + codebook.pow(2).sum(1, keepdim=True).t()
+        )
+        indices = rearrange((-dist).max(1)[1], "(b t) -> b t", b=latents.size(0))
+        z_q = self.decode_code(indices)
+        return z_q, indices
+
+
+class ResidualVectorQuantize(nn.Module):
+    """
+    Introduced in SoundStream: An end2end neural audio codec
+    https://arxiv.org/abs/2107.03312
+    """
+
+    def __init__(
+        self,
+        input_dim: int = 512,
+        n_codebooks: int = 9,
+        codebook_size: int = 1024,
+        codebook_dim: Union[int, list] = 8,
+        quantizer_dropout: float = 0.0,
+    ):
+        super().__init__()
+        if isinstance(codebook_dim, int):
+            codebook_dim = [codebook_dim for _ in range(n_codebooks)]
+
+        self.n_codebooks = n_codebooks
+        self.codebook_dim = codebook_dim
+        self.codebook_size = codebook_size
+
+        self.quantizers = nn.ModuleList(
+            [
+                VectorQuantize(input_dim, codebook_size, codebook_dim[i])
+                for i in range(n_codebooks)
+            ]
+        )
+        self.quantizer_dropout = quantizer_dropout
+
+    def forward(self, z, n_quantizers: int = None):
+        """Quantized the input tensor using a fixed set of `n` codebooks and returns
+        the corresponding codebook vectors
+        Parameters
+        ----------
+        z : Tensor[B x D x T]
+        n_quantizers : int, optional
+            No. of quantizers to use
+            (n_quantizers < self.n_codebooks ex: for quantizer dropout)
+            Note: if `self.quantizer_dropout` is True, this argument is ignored
+                when in training mode, and a random number of quantizers is used.
+        Returns
+        -------
+        dict
+            A dictionary with the following keys:
+
+            "z" : Tensor[B x D x T]
+                Quantized continuous representation of input
+            "codes" : Tensor[B x N x T]
+                Codebook indices for each codebook
+                (quantized discrete representation of input)
+            "latents" : Tensor[B x N*D x T]
+                Projected latents (continuous representation of input before quantization)
+            "vq/commitment_loss" : Tensor[1]
+                Commitment loss to train encoder to predict vectors closer to codebook
+                entries
+            "vq/codebook_loss" : Tensor[1]
+                Codebook loss to update the codebook
+        """
+        z_q = 0
+        residual = z
+        commitment_loss = 0
+        codebook_loss = 0
+
+        codebook_indices = []
+        latents = []
+
+        if n_quantizers is None:
+            n_quantizers = self.n_codebooks
+        if self.training:
+            n_quantizers = torch.ones((z.shape[0],)) * self.n_codebooks + 1
+            dropout = torch.randint(1, self.n_codebooks + 1, (z.shape[0],))
+            n_dropout = int(z.shape[0] * self.quantizer_dropout)
+            n_quantizers[:n_dropout] = dropout[:n_dropout]
+            n_quantizers = n_quantizers.to(z.device)
+
+        for i, quantizer in enumerate(self.quantizers):
+            if self.training is False and i >= n_quantizers:
+                break
+
+            z_q_i, commitment_loss_i, codebook_loss_i, indices_i, z_e_i = quantizer(
+                residual
+            )
+
+            # Create mask to apply quantizer dropout
+            mask = (
+                torch.full((z.shape[0],), fill_value=i, device=z.device) < n_quantizers
+            )
+            z_q = z_q + z_q_i * mask[:, None, None]
+            residual = residual - z_q_i
+
+            # Sum losses
+            commitment_loss += (commitment_loss_i * mask).mean()
+            codebook_loss += (codebook_loss_i * mask).mean()
+
+            codebook_indices.append(indices_i)
+            latents.append(z_e_i)
+
+        codes = torch.stack(codebook_indices, dim=1)
+        latents = torch.cat(latents, dim=1)
+
+        return z_q, codes, latents, commitment_loss, codebook_loss
+
+    def from_codes(self, codes: torch.Tensor):
+        """Given the quantized codes, reconstruct the continuous representation
+        Parameters
+        ----------
+        codes : Tensor[B x N x T]
+            Quantized discrete representation of input
+        Returns
+        -------
+        Tensor[B x D x T]
+            Quantized continuous representation of input
+        """
+        z_q = 0.0
+        z_p = []
+        n_codebooks = codes.shape[1]
+        for i in range(n_codebooks):
+            z_p_i = self.quantizers[i].decode_code(codes[:, i, :])
+            z_p.append(z_p_i)
+
+            z_q_i = self.quantizers[i].out_proj(z_p_i)
+            z_q = z_q + z_q_i
+        return z_q, torch.cat(z_p, dim=1), codes
+
+    def from_latents(self, latents: torch.Tensor):
+        """Given the unquantized latents, reconstruct the
+        continuous representation after quantization.
+
+        Parameters
+        ----------
+        latents : Tensor[B x N x T]
+            Continuous representation of input after projection
+
+        Returns
+        -------
+        Tensor[B x D x T]
+            Quantized representation of full-projected space
+        Tensor[B x D x T]
+            Quantized representation of latent space
+        """
+        z_q = 0
+        z_p = []
+        codes = []
+        dims = np.cumsum([0] + [q.codebook_dim for q in self.quantizers])
+
+        n_codebooks = np.where(dims <= latents.shape[1])[0].max(axis=0, keepdims=True)[
+            0
+        ]
+        for i in range(n_codebooks):
+            j, k = dims[i], dims[i + 1]
+            z_p_i, codes_i = self.quantizers[i].decode_latents(latents[:, j:k, :])
+            z_p.append(z_p_i)
+            codes.append(codes_i)
+
+            z_q_i = self.quantizers[i].out_proj(z_p_i)
+            z_q = z_q + z_q_i
+
+        return z_q, torch.cat(z_p, dim=1), torch.stack(codes, dim=1)
+
+
+if __name__ == "__main__":
+    rvq = ResidualVectorQuantize(quantizer_dropout=True)
+    x = torch.randn(16, 512, 80)
+    y = rvq(x)
+    print(y["latents"].shape)

vae_modules/dac/utils/__init__.py

@@ -0,0 +1,122 @@
+from pathlib import Path
+
+import argbind
+from audiotools import ml
+
+from ..model import DAC
+
+Accelerator = ml.Accelerator
+
+__MODEL_LATEST_TAGS__ = {
+    ("44khz", "8kbps"): "0.0.1",
+    ("24khz", "8kbps"): "0.0.4",
+    ("16khz", "8kbps"): "0.0.5",
+    ("44khz", "16kbps"): "1.0.0",
+}
+
+__MODEL_URLS__ = {
+    (
+        "44khz",
+        "0.0.1",
+        "8kbps",
+    ): "https://github.com/descriptinc/descript-audio-codec/releases/download/0.0.1/weights.pth",
+    (
+        "24khz",
+        "0.0.4",
+        "8kbps",
+    ): "https://github.com/descriptinc/descript-audio-codec/releases/download/0.0.4/weights_24khz.pth",
+    (
+        "16khz",
+        "0.0.5",
+        "8kbps",
+    ): "https://github.com/descriptinc/descript-audio-codec/releases/download/0.0.5/weights_16khz.pth",
+    (
+        "44khz",
+        "1.0.0",
+        "16kbps",
+    ): "https://github.com/descriptinc/descript-audio-codec/releases/download/1.0.0/weights_44khz_16kbps.pth",
+}
+
+
+@argbind.bind(group="download", positional=True, without_prefix=True)
+def download(
+    model_type: str = "44khz", model_bitrate: str = "8kbps", tag: str = "latest"
+):
+    """
+    Function that downloads the weights file from URL if a local cache is not found.
+
+    Parameters
+    ----------
+    model_type : str
+        The type of model to download. Must be one of "44khz", "24khz", or "16khz". Defaults to "44khz".
+    model_bitrate: str
+        Bitrate of the model. Must be one of "8kbps", or "16kbps". Defaults to "8kbps".
+        Only 44khz model supports 16kbps.
+    tag : str
+        The tag of the model to download. Defaults to "latest".
+
+    Returns
+    -------
+    Path
+        Directory path required to load model via audiotools.
+    """
+    model_type = model_type.lower()
+    tag = tag.lower()
+
+    assert model_type in [
+        "44khz",
+        "24khz",
+        "16khz",
+    ], "model_type must be one of '44khz', '24khz', or '16khz'"
+
+    assert model_bitrate in [
+        "8kbps",
+        "16kbps",
+    ], "model_bitrate must be one of '8kbps', or '16kbps'"
+
+    if tag == "latest":
+        tag = __MODEL_LATEST_TAGS__[(model_type, model_bitrate)]
+
+    download_link = __MODEL_URLS__.get((model_type, tag, model_bitrate), None)
+
+    if download_link is None:
+        raise ValueError(
+            f"Could not find model with tag {tag} and model type {model_type}"
+        )
+
+    local_path = (
+        Path.home()
+        / ".cache"
+        / "descript"
+        / "dac"
+        / f"weights_{model_type}_{model_bitrate}_{tag}.pth"
+    )
+    if not local_path.exists():
+        local_path.parent.mkdir(parents=True, exist_ok=True)
+
+        # Download the model
+        import requests
+
+        response = requests.get(download_link)
+
+        if response.status_code != 200:
+            raise ValueError(
+                f"Could not download model. Received response code {response.status_code}"
+            )
+        local_path.write_bytes(response.content)
+
+    return local_path
+
+
+def load_model(
+    model_type: str = "44khz",
+    model_bitrate: str = "8kbps",
+    tag: str = "latest",
+    load_path: str = None,
+):
+    if not load_path:
+        load_path = download(
+            model_type=model_type, model_bitrate=model_bitrate, tag=tag
+        )
+    generator = DAC.load(load_path)
+    return generator

vae_modules/dac/utils/decode.py

@@ -0,0 +1,95 @@
+import warnings
+from pathlib import Path
+
+import argbind
+import numpy as np
+import torch
+from audiotools import AudioSignal
+from tqdm import tqdm
+
+from dac import DACFile
+from dac.utils import load_model
+
+warnings.filterwarnings("ignore", category=UserWarning)
+
+
+@argbind.bind(group="decode", positional=True, without_prefix=True)
+@torch.inference_mode()
+@torch.no_grad()
+def decode(
+    input: str,
+    output: str = "",
+    weights_path: str = "",
+    model_tag: str = "latest",
+    model_bitrate: str = "8kbps",
+    device: str = "cuda",
+    model_type: str = "44khz",
+    verbose: bool = False,
+):
+    """Decode audio from codes.
+
+    Parameters
+    ----------
+    input : str
+        Path to input directory or file
+    output : str, optional
+        Path to output directory, by default "".
+        If `input` is a directory, the directory sub-tree relative to `input` is re-created in `output`.
+    weights_path : str, optional
+        Path to weights file, by default "". If not specified, the weights file will be downloaded from the internet using the
+        model_tag and model_type.
+    model_tag : str, optional
+        Tag of the model to use, by default "latest". Ignored if `weights_path` is specified.
+    model_bitrate: str
+        Bitrate of the model. Must be one of "8kbps", or "16kbps". Defaults to "8kbps".
+    device : str, optional
+        Device to use, by default "cuda". If "cpu", the model will be loaded on the CPU.
+    model_type : str, optional
+        The type of model to use. Must be one of "44khz", "24khz", or "16khz". Defaults to "44khz". Ignored if `weights_path` is specified.
+    """
+    generator = load_model(
+        model_type=model_type,
+        model_bitrate=model_bitrate,
+        tag=model_tag,
+        load_path=weights_path,
+    )
+    generator.to(device)
+    generator.eval()
+
+    # Find all .dac files in input directory
+    _input = Path(input)
+    input_files = list(_input.glob("**/*.dac"))
+
+    # If input is a .dac file, add it to the list
+    if _input.suffix == ".dac":
+        input_files.append(_input)
+
+    # Create output directory
+    output = Path(output)
+    output.mkdir(parents=True, exist_ok=True)
+
+    for i in tqdm(range(len(input_files)), desc=f"Decoding files"):
+        # Load file
+        artifact = DACFile.load(input_files[i])
+
+        # Reconstruct audio from codes
+        recons = generator.decompress(artifact, verbose=verbose)
+
+        # Compute output path
+        relative_path = input_files[i].relative_to(input)
+        output_dir = output / relative_path.parent
+        if not relative_path.name:
+            output_dir = output
+            relative_path = input_files[i]
+        output_name = relative_path.with_suffix(".wav").name
+        output_path = output_dir / output_name
+        output_path.parent.mkdir(parents=True, exist_ok=True)
+
+        # Write to file
+        recons.write(output_path)
+
+
+if __name__ == "__main__":
+    args = argbind.parse_args()
+    with argbind.scope(args):
+        decode()

vae_modules/dac/utils/encode.py

@@ -0,0 +1,94 @@
+import math
+import warnings
+from pathlib import Path
+
+import argbind
+import numpy as np
+import torch
+from audiotools import AudioSignal
+from audiotools.core import util
+from tqdm import tqdm
+
+from dac.utils import load_model
+
+warnings.filterwarnings("ignore", category=UserWarning)
+
+
+@argbind.bind(group="encode", positional=True, without_prefix=True)
+@torch.inference_mode()
+@torch.no_grad()
+def encode(
+    input: str,
+    output: str = "",
+    weights_path: str = "",
+    model_tag: str = "latest",
+    model_bitrate: str = "8kbps",
+    n_quantizers: int = None,
+    device: str = "cuda",
+    model_type: str = "44khz",
+    win_duration: float = 5.0,
+    verbose: bool = False,
+):
+    """Encode audio files in input path to .dac format.
+
+    Parameters
+    ----------
+    input : str
+        Path to input audio file or directory
+    output : str, optional
+        Path to output directory, by default "". If `input` is a directory, the directory sub-tree relative to `input` is re-created in `output`.
+    weights_path : str, optional
+        Path to weights file, by default "". If not specified, the weights file will be downloaded from the internet using the
+        model_tag and model_type.
+    model_tag : str, optional
+        Tag of the model to use, by default "latest". Ignored if `weights_path` is specified.
+    model_bitrate: str
+        Bitrate of the model. Must be one of "8kbps", or "16kbps". Defaults to "8kbps".
+    n_quantizers : int, optional
+        Number of quantizers to use, by default None. If not specified, all the quantizers will be used and the model will compress at maximum bitrate.
+    device : str, optional
+        Device to use, by default "cuda"
+    model_type : str, optional
+        The type of model to use. Must be one of "44khz", "24khz", or "16khz". Defaults to "44khz". Ignored if `weights_path` is specified.
+    """
+    generator = load_model(
+        model_type=model_type,
+        model_bitrate=model_bitrate,
+        tag=model_tag,
+        load_path=weights_path,
+    )
+    generator.to(device)
+    generator.eval()
+    kwargs = {"n_quantizers": n_quantizers}
+
+    # Find all audio files in input path
+    input = Path(input)
+    audio_files = util.find_audio(input)
+
+    output = Path(output)
+    output.mkdir(parents=True, exist_ok=True)
+
+    for i in tqdm(range(len(audio_files)), desc="Encoding files"):
+        # Load file
+        signal = AudioSignal(audio_files[i])
+
+        # Encode audio to .dac format
+        artifact = generator.compress(signal, win_duration, verbose=verbose, **kwargs)
+
+        # Compute output path
+        relative_path = audio_files[i].relative_to(input)
+        output_dir = output / relative_path.parent
+        if not relative_path.name:
+            output_dir = output
+            relative_path = audio_files[i]
+        output_name = relative_path.with_suffix(".dac").name
+        output_path = output_dir / output_name
+        output_path.parent.mkdir(parents=True, exist_ok=True)
+
+        artifact.save(output_path)
+
+
+if __name__ == "__main__":
+    args = argbind.parse_args()
+    with argbind.scope(args):
+        encode()

vae_modules/stable_vae/__init__.py

@@ -0,0 +1,40 @@
+from .models.autoencoders import create_autoencoder_from_config
+import os
+import json
+import torch
+from torch.nn.utils import remove_weight_norm
+
+
+def remove_all_weight_norm(model):
+    for name, module in model.named_modules():
+        if hasattr(module, 'weight_g'):
+            remove_weight_norm(module)
+
+
+def load_vae(ckpt_path, remove_weight_norm=False):
+    config_file = os.path.join(os.path.dirname(ckpt_path), 'config.json')
+
+    # Load the model configuration
+    with open(config_file) as f:
+        model_config = json.load(f)
+
+    # Create the model from the configuration
+    model = create_autoencoder_from_config(model_config)
+
+    # Load the state dictionary from the checkpoint
+    model_dict = torch.load(ckpt_path, map_location='cpu')['state_dict']
+
+    # Strip the "autoencoder." prefix from the keys
+    model_dict = {key[len("autoencoder."):]: value for key, value in model_dict.items() if key.startswith("autoencoder.")}
+
+    # Load the state dictionary into the model
+    model.load_state_dict(model_dict)
+
+    # Remove weight normalization
+    if remove_weight_norm:
+        remove_all_weight_norm(model)
+
+    # Set the model to evaluation mode
+    model.eval()
+
+    return model

vae_modules/stable_vae/models/autoencoders.py

@@ -0,0 +1,683 @@
+import torch
+import math
+import numpy as np
+
+from torch import nn
+from torch.nn import functional as F
+from torchaudio import transforms as T
+from alias_free_torch import Activation1d
+from .nn.layers import WNConv1d, WNConvTranspose1d
+from typing import Literal, Dict, Any
+
+# from .inference.sampling import sample
+from .utils import prepare_audio
+from .blocks import SnakeBeta
+from .bottleneck import Bottleneck, DiscreteBottleneck
+from .factory import create_pretransform_from_config, create_bottleneck_from_config
+from .pretransforms import Pretransform
+
+def checkpoint(function, *args, **kwargs):
+    kwargs.setdefault("use_reentrant", False)
+    return torch.utils.checkpoint.checkpoint(function, *args, **kwargs)
+
+def get_activation(activation: Literal["elu", "snake", "none"], antialias=False, channels=None) -> nn.Module:
+    if activation == "elu":
+        act = nn.ELU()
+    elif activation == "snake":
+        act = SnakeBeta(channels)
+    elif activation == "none":
+        act = nn.Identity()
+    else:
+        raise ValueError(f"Unknown activation {activation}")
+    
+    if antialias:
+        act = Activation1d(act)
+    
+    return act
+
+class ResidualUnit(nn.Module):
+    def __init__(self, in_channels, out_channels, dilation, use_snake=False, antialias_activation=False):
+        super().__init__()
+        
+        self.dilation = dilation
+
+        padding = (dilation * (7-1)) // 2
+
+        self.layers = nn.Sequential(
+            get_activation("snake" if use_snake else "elu", antialias=antialias_activation, channels=out_channels),
+            WNConv1d(in_channels=in_channels, out_channels=out_channels,
+                      kernel_size=7, dilation=dilation, padding=padding),
+            get_activation("snake" if use_snake else "elu", antialias=antialias_activation, channels=out_channels),
+            WNConv1d(in_channels=out_channels, out_channels=out_channels,
+                      kernel_size=1)
+        )
+
+    def forward(self, x):
+        res = x
+        
+        #x = checkpoint(self.layers, x)
+        x = self.layers(x)
+
+        return x + res
+
+class EncoderBlock(nn.Module):
+    def __init__(self, in_channels, out_channels, stride, use_snake=False, antialias_activation=False):
+        super().__init__()
+
+        self.layers = nn.Sequential(
+            ResidualUnit(in_channels=in_channels,
+                         out_channels=in_channels, dilation=1, use_snake=use_snake),
+            ResidualUnit(in_channels=in_channels,
+                         out_channels=in_channels, dilation=3, use_snake=use_snake),
+            ResidualUnit(in_channels=in_channels,
+                         out_channels=in_channels, dilation=9, use_snake=use_snake),
+            get_activation("snake" if use_snake else "elu", antialias=antialias_activation, channels=in_channels),
+            WNConv1d(in_channels=in_channels, out_channels=out_channels,
+                      kernel_size=2*stride, stride=stride, padding=math.ceil(stride/2)),
+        )
+
+    def forward(self, x):
+        return self.layers(x)
+
+class DecoderBlock(nn.Module):
+    def __init__(self, in_channels, out_channels, stride, use_snake=False, antialias_activation=False, use_nearest_upsample=False):
+        super().__init__()
+
+        if use_nearest_upsample:
+            upsample_layer = nn.Sequential(
+                nn.Upsample(scale_factor=stride, mode="nearest"),
+                WNConv1d(in_channels=in_channels,
+                        out_channels=out_channels, 
+                        kernel_size=2*stride,
+                        stride=1,
+                        bias=False,
+                        padding='same')
+            )
+        else:
+            upsample_layer = WNConvTranspose1d(in_channels=in_channels,
+                               out_channels=out_channels,
+                               kernel_size=2*stride, stride=stride, padding=math.ceil(stride/2))
+
+        self.layers = nn.Sequential(
+            get_activation("snake" if use_snake else "elu", antialias=antialias_activation, channels=in_channels),
+            upsample_layer,
+            ResidualUnit(in_channels=out_channels, out_channels=out_channels,
+                         dilation=1, use_snake=use_snake),
+            ResidualUnit(in_channels=out_channels, out_channels=out_channels,
+                         dilation=3, use_snake=use_snake),
+            ResidualUnit(in_channels=out_channels, out_channels=out_channels,
+                         dilation=9, use_snake=use_snake),
+        )
+
+    def forward(self, x):
+        return self.layers(x)
+
+class OobleckEncoder(nn.Module):
+    def __init__(self, 
+                 in_channels=2, 
+                 channels=128, 
+                 latent_dim=32, 
+                 c_mults = [1, 2, 4, 8], 
+                 strides = [2, 4, 8, 8],
+                 use_snake=False,
+                 antialias_activation=False
+        ):
+        super().__init__()
+          
+        c_mults = [1] + c_mults
+
+        self.depth = len(c_mults)
+
+        layers = [
+            WNConv1d(in_channels=in_channels, out_channels=c_mults[0] * channels, kernel_size=7, padding=3)
+        ]
+        
+        for i in range(self.depth-1):
+            layers += [EncoderBlock(in_channels=c_mults[i]*channels, out_channels=c_mults[i+1]*channels, stride=strides[i], use_snake=use_snake)]
+
+        layers += [
+            get_activation("snake" if use_snake else "elu", antialias=antialias_activation, channels=c_mults[-1] * channels),
+            WNConv1d(in_channels=c_mults[-1]*channels, out_channels=latent_dim, kernel_size=3, padding=1)
+        ]
+
+        self.layers = nn.Sequential(*layers)
+
+    def forward(self, x):
+        return self.layers(x)
+
+
+class OobleckDecoder(nn.Module):
+    def __init__(self, 
+                 out_channels=2, 
+                 channels=128, 
+                 latent_dim=32, 
+                 c_mults = [1, 2, 4, 8], 
+                 strides = [2, 4, 8, 8],
+                 use_snake=False,
+                 antialias_activation=False,
+                 use_nearest_upsample=False,
+                 final_tanh=True):
+        super().__init__()
+
+        c_mults = [1] + c_mults
+        
+        self.depth = len(c_mults)
+
+        layers = [
+            WNConv1d(in_channels=latent_dim, out_channels=c_mults[-1]*channels, kernel_size=7, padding=3),
+        ]
+        
+        for i in range(self.depth-1, 0, -1):
+            layers += [DecoderBlock(
+                in_channels=c_mults[i]*channels, 
+                out_channels=c_mults[i-1]*channels, 
+                stride=strides[i-1], 
+                use_snake=use_snake, 
+                antialias_activation=antialias_activation,
+                use_nearest_upsample=use_nearest_upsample
+                )
+            ]
+
+        layers += [
+            get_activation("snake" if use_snake else "elu", antialias=antialias_activation, channels=c_mults[0] * channels),
+            WNConv1d(in_channels=c_mults[0] * channels, out_channels=out_channels, kernel_size=7, padding=3, bias=False),
+            nn.Tanh() if final_tanh else nn.Identity()
+        ]
+
+        self.layers = nn.Sequential(*layers)
+
+    def forward(self, x):
+        return self.layers(x)
+
+
+class DACEncoderWrapper(nn.Module):
+    def __init__(self, in_channels=1, **kwargs):
+        super().__init__()
+
+        from dac.model.dac import Encoder as DACEncoder
+
+        latent_dim = kwargs.pop("latent_dim", None)
+
+        encoder_out_dim = kwargs["d_model"] * (2 ** len(kwargs["strides"]))
+        self.encoder = DACEncoder(d_latent=encoder_out_dim, **kwargs)
+        self.latent_dim = latent_dim
+
+        # Latent-dim support was added to DAC after this was first written, and implemented differently, so this is for backwards compatibility
+        self.proj_out = nn.Conv1d(self.encoder.enc_dim, latent_dim, kernel_size=1) if latent_dim is not None else nn.Identity()
+
+        if in_channels != 1:
+            self.encoder.block[0] = WNConv1d(in_channels, kwargs.get("d_model", 64), kernel_size=7, padding=3)
+
+    def forward(self, x):
+        x = self.encoder(x)
+        x = self.proj_out(x)
+        return x
+
+class DACDecoderWrapper(nn.Module):
+    def __init__(self, latent_dim, out_channels=1, **kwargs):
+        super().__init__()
+
+        from dac.model.dac import Decoder as DACDecoder
+
+        self.decoder = DACDecoder(**kwargs, input_channel = latent_dim, d_out=out_channels)
+
+        self.latent_dim = latent_dim
+
+    def forward(self, x):
+        return self.decoder(x)
+
+class AudioAutoencoder(nn.Module):
+    def __init__(
+        self,
+        encoder,
+        decoder,
+        latent_dim,
+        downsampling_ratio,
+        sample_rate,
+        io_channels=2,
+        bottleneck: Bottleneck = None,
+        pretransform: Pretransform = None,
+        in_channels = None,
+        out_channels = None,
+        soft_clip = False
+    ):
+        super().__init__()
+
+        self.downsampling_ratio = downsampling_ratio
+        self.sample_rate = sample_rate
+
+        self.latent_dim = latent_dim
+        self.io_channels = io_channels
+        self.in_channels = io_channels
+        self.out_channels = io_channels
+
+        self.min_length = self.downsampling_ratio
+
+        if in_channels is not None:
+            self.in_channels = in_channels
+
+        if out_channels is not None:
+            self.out_channels = out_channels
+
+        self.bottleneck = bottleneck
+
+        self.encoder = encoder
+
+        self.decoder = decoder
+
+        self.pretransform = pretransform
+
+        self.soft_clip = soft_clip
+ 
+        self.is_discrete = self.bottleneck is not None and self.bottleneck.is_discrete
+
+    def encode(self, audio, return_info=False, skip_pretransform=False, iterate_batch=False, **kwargs):
+
+        info = {}
+
+        if self.pretransform is not None and not skip_pretransform:
+            if self.pretransform.enable_grad:
+                if iterate_batch:
+                    audios = []
+                    for i in range(audio.shape[0]):
+                        audios.append(self.pretransform.encode(audio[i:i+1]))
+                    audio = torch.cat(audios, dim=0)
+                else:
+                    audio = self.pretransform.encode(audio)
+            else:
+                with torch.no_grad():
+                    if iterate_batch:
+                        audios = []
+                        for i in range(audio.shape[0]):
+                            audios.append(self.pretransform.encode(audio[i:i+1]))
+                        audio = torch.cat(audios, dim=0)
+                    else:
+                        audio = self.pretransform.encode(audio)
+
+        if self.encoder is not None:
+            if iterate_batch:
+                latents = []
+                for i in range(audio.shape[0]):
+                    latents.append(self.encoder(audio[i:i+1]))
+                latents = torch.cat(latents, dim=0)
+            else:
+                latents = self.encoder(audio)
+        else:
+            latents = audio
+
+        if self.bottleneck is not None:
+            # TODO: Add iterate batch logic, needs to merge the info dicts
+            latents, bottleneck_info = self.bottleneck.encode(latents, return_info=True, **kwargs)
+
+            info.update(bottleneck_info)
+
+        if return_info:
+            return latents, info
+
+        return latents
+
+    def decode(self, latents, iterate_batch=False, **kwargs):
+
+        if self.bottleneck is not None:
+            if iterate_batch:
+                decoded = []
+                for i in range(latents.shape[0]):
+                    decoded.append(self.bottleneck.decode(latents[i:i+1]))
+                decoded = torch.cat(decoded, dim=0)
+            else:
+                latents = self.bottleneck.decode(latents)
+
+        if iterate_batch:
+            decoded = []
+            for i in range(latents.shape[0]):
+                decoded.append(self.decoder(latents[i:i+1]))
+            decoded = torch.cat(decoded, dim=0)
+        else:
+            decoded = self.decoder(latents, **kwargs)
+
+        if self.pretransform is not None:
+            if self.pretransform.enable_grad:
+                if iterate_batch:
+                    decodeds = []
+                    for i in range(decoded.shape[0]):
+                        decodeds.append(self.pretransform.decode(decoded[i:i+1]))
+                    decoded = torch.cat(decodeds, dim=0)
+                else:
+                    decoded = self.pretransform.decode(decoded)
+            else:
+                with torch.no_grad():
+                    if iterate_batch:
+                        decodeds = []
+                        for i in range(latents.shape[0]):
+                            decodeds.append(self.pretransform.decode(decoded[i:i+1]))
+                        decoded = torch.cat(decodeds, dim=0)
+                    else:
+                        decoded = self.pretransform.decode(decoded)
+
+        if self.soft_clip:
+            decoded = torch.tanh(decoded)
+
+        return decoded
+
+    def decode_tokens(self, tokens, **kwargs):
+        '''
+        Decode discrete tokens to audio
+        Only works with discrete autoencoders
+        '''
+
+        assert isinstance(self.bottleneck, DiscreteBottleneck), "decode_tokens only works with discrete autoencoders"
+
+        latents = self.bottleneck.decode_tokens(tokens, **kwargs)
+
+        return self.decode(latents, **kwargs)
+        
+    
+    def preprocess_audio_for_encoder(self, audio, in_sr):
+        '''
+        Preprocess single audio tensor (Channels x Length) to be compatible with the encoder.
+        If the model is mono, stereo audio will be converted to mono.
+        Audio will be silence-padded to be a multiple of the model's downsampling ratio.
+        Audio will be resampled to the model's sample rate. 
+        The output will have batch size 1 and be shape (1 x Channels x Length)
+        '''
+        return self.preprocess_audio_list_for_encoder([audio], [in_sr])
+
+    def preprocess_audio_list_for_encoder(self, audio_list, in_sr_list):
+        '''
+        Preprocess a [list] of audio (Channels x Length) into a batch tensor to be compatable with the encoder. 
+        The audio in that list can be of different lengths and channels. 
+        in_sr can be an integer or list. If it's an integer it will be assumed it is the input sample_rate for every audio.
+        All audio will be resampled to the model's sample rate. 
+        Audio will be silence-padded to the longest length, and further padded to be a multiple of the model's downsampling ratio. 
+        If the model is mono, all audio will be converted to mono. 
+        The output will be a tensor of shape (Batch x Channels x Length)
+        '''
+        batch_size = len(audio_list)
+        if isinstance(in_sr_list, int):
+            in_sr_list = [in_sr_list]*batch_size
+        assert len(in_sr_list) == batch_size, "list of sample rates must be the same length of audio_list"
+        new_audio = []
+        max_length = 0
+        # resample & find the max length
+        for i in range(batch_size):
+            audio = audio_list[i]
+            in_sr = in_sr_list[i]
+            if len(audio.shape) == 3 and audio.shape[0] == 1:
+                # batchsize 1 was given by accident. Just squeeze it.
+                audio = audio.squeeze(0)
+            elif len(audio.shape) == 1:
+                # Mono signal, channel dimension is missing, unsqueeze it in
+                audio = audio.unsqueeze(0)
+            assert len(audio.shape)==2, "Audio should be shape (Channels x Length) with no batch dimension" 
+            # Resample audio
+            if in_sr != self.sample_rate:
+                resample_tf = T.Resample(in_sr, self.sample_rate).to(audio.device)
+                audio = resample_tf(audio)
+            new_audio.append(audio)
+            if audio.shape[-1] > max_length:
+                max_length = audio.shape[-1]
+        # Pad every audio to the same length, multiple of model's downsampling ratio
+        padded_audio_length = max_length + (self.min_length - (max_length % self.min_length)) % self.min_length
+        for i in range(batch_size):
+            # Pad it & if necessary, mixdown/duplicate stereo/mono channels to support model
+            new_audio[i] = prepare_audio(new_audio[i], in_sr=in_sr, target_sr=in_sr, target_length=padded_audio_length, 
+                target_channels=self.in_channels, device=new_audio[i].device).squeeze(0)
+        # convert to tensor 
+        return torch.stack(new_audio) 
+
+    def encode_audio(self, audio, chunked=False, overlap=32, chunk_size=128, **kwargs):
+        '''
+        Encode audios into latents. Audios should already be preprocesed by preprocess_audio_for_encoder.
+        If chunked is True, split the audio into chunks of a given maximum size chunk_size, with given overlap.
+        Overlap and chunk_size params are both measured in number of latents (not audio samples) 
+        # and therefore you likely could use the same values with decode_audio. 
+        A overlap of zero will cause discontinuity artefacts. Overlap should be => receptive field size. 
+        Every autoencoder will have a different receptive field size, and thus ideal overlap.
+        You can determine it empirically by diffing unchunked vs chunked output and looking at maximum diff.
+        The final chunk may have a longer overlap in order to keep chunk_size consistent for all chunks.
+        Smaller chunk_size uses less memory, but more compute.
+        The chunk_size vs memory tradeoff isn't linear, and possibly depends on the GPU and CUDA version
+        For example, on a A6000 chunk_size 128 is overall faster than 256 and 512 even though it has more chunks
+        '''
+        if not chunked:
+            # default behavior. Encode the entire audio in parallel
+            return self.encode(audio, **kwargs)
+        else:
+            # CHUNKED ENCODING
+            # samples_per_latent is just the downsampling ratio (which is also the upsampling ratio)
+            samples_per_latent = self.downsampling_ratio
+            total_size = audio.shape[2] # in samples
+            batch_size = audio.shape[0]
+            chunk_size *= samples_per_latent # converting metric in latents to samples
+            overlap *= samples_per_latent # converting metric in latents to samples
+            hop_size = chunk_size - overlap
+            chunks = []
+            for i in range(0, total_size - chunk_size + 1, hop_size):
+                chunk = audio[:,:,i:i+chunk_size]
+                chunks.append(chunk)
+            if i+chunk_size != total_size:
+                # Final chunk
+                chunk = audio[:,:,-chunk_size:]
+                chunks.append(chunk)
+            chunks = torch.stack(chunks)
+            num_chunks = chunks.shape[0]
+            # Note: y_size might be a different value from the latent length used in diffusion training
+            # because we can encode audio of varying lengths
+            # However, the audio should've been padded to a multiple of samples_per_latent by now.
+            y_size = total_size // samples_per_latent
+            # Create an empty latent, we will populate it with chunks as we encode them
+            y_final = torch.zeros((batch_size,self.latent_dim,y_size)).to(audio.device)
+            for i in range(num_chunks):
+                x_chunk = chunks[i,:]
+                # encode the chunk
+                y_chunk = self.encode(x_chunk)
+                # figure out where to put the audio along the time domain
+                if i == num_chunks-1:
+                    # final chunk always goes at the end
+                    t_end = y_size
+                    t_start = t_end - y_chunk.shape[2]
+                else:
+                    t_start = i * hop_size // samples_per_latent
+                    t_end = t_start + chunk_size // samples_per_latent
+                #  remove the edges of the overlaps
+                ol = overlap//samples_per_latent//2
+                chunk_start = 0
+                chunk_end = y_chunk.shape[2]
+                if i > 0:
+                    # no overlap for the start of the first chunk
+                    t_start += ol
+                    chunk_start += ol
+                if i < num_chunks-1:
+                    # no overlap for the end of the last chunk
+                    t_end -= ol
+                    chunk_end -= ol
+                # paste the chunked audio into our y_final output audio
+                y_final[:,:,t_start:t_end] = y_chunk[:,:,chunk_start:chunk_end]
+            return y_final
+    
+    def decode_audio(self, latents, chunked=False, overlap=32, chunk_size=128, **kwargs):
+        '''
+        Decode latents to audio. 
+        If chunked is True, split the latents into chunks of a given maximum size chunk_size, with given overlap, both of which are measured in number of latents. 
+        A overlap of zero will cause discontinuity artefacts. Overlap should be => receptive field size. 
+        Every autoencoder will have a different receptive field size, and thus ideal overlap.
+        You can determine it empirically by diffing unchunked vs chunked audio and looking at maximum diff.
+        The final chunk may have a longer overlap in order to keep chunk_size consistent for all chunks.
+        Smaller chunk_size uses less memory, but more compute.
+        The chunk_size vs memory tradeoff isn't linear, and possibly depends on the GPU and CUDA version
+        For example, on a A6000 chunk_size 128 is overall faster than 256 and 512 even though it has more chunks
+        '''
+        if not chunked:
+            # default behavior. Decode the entire latent in parallel
+            return self.decode(latents, **kwargs)
+        else:
+            # chunked decoding
+            hop_size = chunk_size - overlap
+            total_size = latents.shape[2]
+            batch_size = latents.shape[0]
+            chunks = []
+            for i in range(0, total_size - chunk_size + 1, hop_size):
+                chunk = latents[:,:,i:i+chunk_size]
+                chunks.append(chunk)
+            if i+chunk_size != total_size:
+                # Final chunk
+                chunk = latents[:,:,-chunk_size:]
+                chunks.append(chunk)
+            chunks = torch.stack(chunks)
+            num_chunks = chunks.shape[0]
+            # samples_per_latent is just the downsampling ratio
+            samples_per_latent = self.downsampling_ratio
+            # Create an empty waveform, we will populate it with chunks as decode them
+            y_size = total_size * samples_per_latent
+            y_final = torch.zeros((batch_size,self.out_channels,y_size)).to(latents.device)
+            for i in range(num_chunks):
+                x_chunk = chunks[i,:]
+                # decode the chunk
+                y_chunk = self.decode(x_chunk)
+                # figure out where to put the audio along the time domain
+                if i == num_chunks-1:
+                    # final chunk always goes at the end
+                    t_end = y_size
+                    t_start = t_end - y_chunk.shape[2]
+                else:
+                    t_start = i * hop_size * samples_per_latent
+                    t_end = t_start + chunk_size * samples_per_latent
+                #  remove the edges of the overlaps
+                ol = (overlap//2) * samples_per_latent
+                chunk_start = 0
+                chunk_end = y_chunk.shape[2]
+                if i > 0:
+                    # no overlap for the start of the first chunk
+                    t_start += ol
+                    chunk_start += ol
+                if i < num_chunks-1:
+                    # no overlap for the end of the last chunk
+                    t_end -= ol
+                    chunk_end -= ol
+                # paste the chunked audio into our y_final output audio
+                y_final[:,:,t_start:t_end] = y_chunk[:,:,chunk_start:chunk_end]
+            return y_final
+
+    
+# AE factories
+
+def create_encoder_from_config(encoder_config: Dict[str, Any]):
+    encoder_type = encoder_config.get("type", None)
+    assert encoder_type is not None, "Encoder type must be specified"
+
+    if encoder_type == "oobleck":
+        encoder = OobleckEncoder(
+            **encoder_config["config"]
+        )
+    
+    elif encoder_type == "seanet":
+        from encodec.modules import SEANetEncoder
+        seanet_encoder_config = encoder_config["config"]
+
+        #SEANet encoder expects strides in reverse order
+        seanet_encoder_config["ratios"] = list(reversed(seanet_encoder_config.get("ratios", [2, 2, 2, 2, 2])))
+        encoder = SEANetEncoder(
+            **seanet_encoder_config
+        )
+    elif encoder_type == "dac":
+        dac_config = encoder_config["config"]
+
+        encoder = DACEncoderWrapper(**dac_config)
+    elif encoder_type == "local_attn":
+        from .local_attention import TransformerEncoder1D
+
+        local_attn_config = encoder_config["config"]
+
+        encoder = TransformerEncoder1D(
+            **local_attn_config
+        )
+    else:
+        raise ValueError(f"Unknown encoder type {encoder_type}")
+    
+    requires_grad = encoder_config.get("requires_grad", True)
+    if not requires_grad:
+        for param in encoder.parameters():
+            param.requires_grad = False
+
+    return encoder
+
+def create_decoder_from_config(decoder_config: Dict[str, Any]):
+    decoder_type = decoder_config.get("type", None)
+    assert decoder_type is not None, "Decoder type must be specified"
+
+    if decoder_type == "oobleck":
+        decoder = OobleckDecoder(
+            **decoder_config["config"]
+        )
+    elif decoder_type == "seanet":
+        from encodec.modules import SEANetDecoder
+
+        decoder = SEANetDecoder(
+            **decoder_config["config"]
+        )
+    elif decoder_type == "dac":
+        dac_config = decoder_config["config"]
+
+        decoder = DACDecoderWrapper(**dac_config)
+    elif decoder_type == "local_attn":
+        from .local_attention import TransformerDecoder1D
+
+        local_attn_config = decoder_config["config"]
+
+        decoder = TransformerDecoder1D(
+            **local_attn_config
+        )
+    else:
+        raise ValueError(f"Unknown decoder type {decoder_type}")
+    
+    requires_grad = decoder_config.get("requires_grad", True)
+    if not requires_grad:
+        for param in decoder.parameters():
+            param.requires_grad = False
+
+    return decoder
+
+def create_autoencoder_from_config(config: Dict[str, Any]):
+    
+    ae_config = config["model"]
+
+    encoder = create_encoder_from_config(ae_config["encoder"])
+    decoder = create_decoder_from_config(ae_config["decoder"])
+
+    bottleneck = ae_config.get("bottleneck", None)
+
+    latent_dim = ae_config.get("latent_dim", None)
+    assert latent_dim is not None, "latent_dim must be specified in model config"
+    downsampling_ratio = ae_config.get("downsampling_ratio", None)
+    assert downsampling_ratio is not None, "downsampling_ratio must be specified in model config"
+    io_channels = ae_config.get("io_channels", None)
+    assert io_channels is not None, "io_channels must be specified in model config"
+    sample_rate = config.get("sample_rate", None)
+    assert sample_rate is not None, "sample_rate must be specified in model config"
+
+    in_channels = ae_config.get("in_channels", None)
+    out_channels = ae_config.get("out_channels", None)
+
+    pretransform = ae_config.get("pretransform", None)
+
+    if pretransform is not None:
+        pretransform = create_pretransform_from_config(pretransform, sample_rate)
+
+    if bottleneck is not None:
+        bottleneck = create_bottleneck_from_config(bottleneck)
+
+    soft_clip = ae_config["decoder"].get("soft_clip", False)
+
+    return AudioAutoencoder(
+        encoder,
+        decoder,
+        io_channels=io_channels,
+        latent_dim=latent_dim,
+        downsampling_ratio=downsampling_ratio,
+        sample_rate=sample_rate,
+        bottleneck=bottleneck,
+        pretransform=pretransform,
+        in_channels=in_channels,
+        out_channels=out_channels,
+        soft_clip=soft_clip
+    )

vae_modules/stable_vae/models/blocks.py

@@ -0,0 +1,359 @@
+from functools import reduce
+import math
+import numpy as np
+import torch
+from torch import nn
+from torch.nn import functional as F
+
+from torch.backends.cuda import sdp_kernel
+from packaging import version
+
+from .nn.layers import Snake1d
+
+
+class ResidualBlock(nn.Module):
+    def __init__(self, main, skip=None):
+        super().__init__()
+        self.main = nn.Sequential(*main)
+        self.skip = skip if skip else nn.Identity()
+
+    def forward(self, input):
+        return self.main(input) + self.skip(input)
+
+
+class ResConvBlock(ResidualBlock):
+    def __init__(self, c_in, c_mid, c_out, is_last=False, kernel_size=5, conv_bias=True, use_snake=False):
+        skip = None if c_in == c_out else nn.Conv1d(c_in, c_out, 1, bias=False)
+        super().__init__([
+            nn.Conv1d(c_in, c_mid, kernel_size, padding=kernel_size//2, bias=conv_bias),
+            nn.GroupNorm(1, c_mid),
+            Snake1d(c_mid) if use_snake else nn.GELU(),
+            nn.Conv1d(c_mid, c_out, kernel_size, padding=kernel_size//2, bias=conv_bias),
+            nn.GroupNorm(1, c_out) if not is_last else nn.Identity(),
+            (Snake1d(c_out) if use_snake else nn.GELU()) if not is_last else nn.Identity(),
+        ], skip)
+
+
+class SelfAttention1d(nn.Module):
+    def __init__(self, c_in, n_head=1, dropout_rate=0.):
+        super().__init__()
+        assert c_in % n_head == 0
+        self.norm = nn.GroupNorm(1, c_in)
+        self.n_head = n_head
+        self.qkv_proj = nn.Conv1d(c_in, c_in * 3, 1)
+        self.out_proj = nn.Conv1d(c_in, c_in, 1)
+        self.dropout = nn.Dropout(dropout_rate, inplace=True)
+
+        self.use_flash = torch.cuda.is_available() and version.parse(torch.__version__) >= version.parse('2.0.0')
+
+        if not self.use_flash:
+            return
+
+        device_properties = torch.cuda.get_device_properties(torch.device('cuda'))
+
+        if device_properties.major == 8 and device_properties.minor == 0:
+            # Use flash attention for A100 GPUs
+            self.sdp_kernel_config = (True, False, False)
+        else:
+            # Don't use flash attention for other GPUs
+            self.sdp_kernel_config = (False, True, True)
+
+    def forward(self, input):
+        n, c, s = input.shape
+        qkv = self.qkv_proj(self.norm(input))
+        qkv = qkv.view(
+            [n, self.n_head * 3, c // self.n_head, s]).transpose(2, 3)
+        q, k, v = qkv.chunk(3, dim=1)
+        scale = k.shape[3]**-0.25
+
+        if self.use_flash:
+            with sdp_kernel(*self.sdp_kernel_config):
+                y = F.scaled_dot_product_attention(q, k, v, is_causal=False).contiguous().view([n, c, s])
+        else:
+            att = ((q * scale) @ (k.transpose(2, 3) * scale)).softmax(3)
+            y = (att @ v).transpose(2, 3).contiguous().view([n, c, s])
+
+
+        return input + self.dropout(self.out_proj(y))
+
+
+class SkipBlock(nn.Module):
+    def __init__(self, *main):
+        super().__init__()
+        self.main = nn.Sequential(*main)
+
+    def forward(self, input):
+        return torch.cat([self.main(input), input], dim=1)
+
+
+class FourierFeatures(nn.Module):
+    def __init__(self, in_features, out_features, std=1.):
+        super().__init__()
+        assert out_features % 2 == 0
+        self.weight = nn.Parameter(torch.randn(
+            [out_features // 2, in_features]) * std)
+
+    def forward(self, input):
+        f = 2 * math.pi * input @ self.weight.T
+        return torch.cat([f.cos(), f.sin()], dim=-1)
+
+
+def expand_to_planes(input, shape):
+    return input[..., None].repeat([1, 1, shape[2]])
+
+_kernels = {
+    'linear':
+        [1 / 8, 3 / 8, 3 / 8, 1 / 8],
+    'cubic': 
+        [-0.01171875, -0.03515625, 0.11328125, 0.43359375,
+        0.43359375, 0.11328125, -0.03515625, -0.01171875],
+    'lanczos3': 
+        [0.003689131001010537, 0.015056144446134567, -0.03399861603975296,
+        -0.066637322306633, 0.13550527393817902, 0.44638532400131226,
+        0.44638532400131226, 0.13550527393817902, -0.066637322306633,
+        -0.03399861603975296, 0.015056144446134567, 0.003689131001010537]
+}
+
+
+class Downsample1d(nn.Module):
+    def __init__(self, kernel='linear', pad_mode='reflect', channels_last=False):
+        super().__init__()
+        self.pad_mode = pad_mode
+        kernel_1d = torch.tensor(_kernels[kernel])
+        self.pad = kernel_1d.shape[0] // 2 - 1
+        self.register_buffer('kernel', kernel_1d)
+        self.channels_last = channels_last
+    
+    def forward(self, x):
+        if self.channels_last:
+            x = x.permute(0, 2, 1)
+        x = F.pad(x, (self.pad,) * 2, self.pad_mode)
+        weight = x.new_zeros([x.shape[1], x.shape[1], self.kernel.shape[0]])
+        indices = torch.arange(x.shape[1], device=x.device)
+        weight[indices, indices] = self.kernel.to(weight)
+        x = F.conv1d(x, weight, stride=2)
+        if self.channels_last:
+            x = x.permute(0, 2, 1)
+        return x
+
+
+class Upsample1d(nn.Module):
+    def __init__(self, kernel='linear', pad_mode='reflect', channels_last=False):
+        super().__init__()
+        self.pad_mode = pad_mode
+        kernel_1d = torch.tensor(_kernels[kernel]) * 2
+        self.pad = kernel_1d.shape[0] // 2 - 1
+        self.register_buffer('kernel', kernel_1d)
+        self.channels_last = channels_last
+    
+    def forward(self, x):
+        if self.channels_last:
+            x = x.permute(0, 2, 1)
+        x = F.pad(x, ((self.pad + 1) // 2,) * 2, self.pad_mode)
+        weight = x.new_zeros([x.shape[1], x.shape[1], self.kernel.shape[0]])
+        indices = torch.arange(x.shape[1], device=x.device)
+        weight[indices, indices] = self.kernel.to(weight)
+        x = F.conv_transpose1d(x, weight, stride=2, padding=self.pad * 2 + 1)
+        if self.channels_last:
+            x = x.permute(0, 2, 1)
+        return x
+
+
+def Downsample1d_2(
+    in_channels: int, out_channels: int, factor: int, kernel_multiplier: int = 2
+) -> nn.Module:
+    assert kernel_multiplier % 2 == 0, "Kernel multiplier must be even"
+
+    return nn.Conv1d(
+        in_channels=in_channels,
+        out_channels=out_channels,
+        kernel_size=factor * kernel_multiplier + 1,
+        stride=factor,
+        padding=factor * (kernel_multiplier // 2),
+    )
+
+
+def Upsample1d_2(
+    in_channels: int, out_channels: int, factor: int, use_nearest: bool = False
+) -> nn.Module:
+
+    if factor == 1:
+        return nn.Conv1d(
+            in_channels=in_channels, out_channels=out_channels, kernel_size=3, padding=1
+        )
+
+    if use_nearest:
+        return nn.Sequential(
+            nn.Upsample(scale_factor=factor, mode="nearest"),
+            nn.Conv1d(
+                in_channels=in_channels,
+                out_channels=out_channels,
+                kernel_size=3,
+                padding=1,
+            ),
+        )
+    else:
+        return nn.ConvTranspose1d(
+            in_channels=in_channels,
+            out_channels=out_channels,
+            kernel_size=factor * 2,
+            stride=factor,
+            padding=factor // 2 + factor % 2,
+            output_padding=factor % 2,
+        )
+
+
+def zero_init(layer):
+    nn.init.zeros_(layer.weight)
+    if layer.bias is not None:
+        nn.init.zeros_(layer.bias)
+    return layer
+
+
+def rms_norm(x, scale, eps):
+    dtype = reduce(torch.promote_types, (x.dtype, scale.dtype, torch.float32))
+    mean_sq = torch.mean(x.to(dtype)**2, dim=-1, keepdim=True)
+    scale = scale.to(dtype) * torch.rsqrt(mean_sq + eps)
+    return x * scale.to(x.dtype)
+
+#rms_norm = torch.compile(rms_norm)
+
+class AdaRMSNorm(nn.Module):
+    def __init__(self, features, cond_features, eps=1e-6):
+        super().__init__()
+        self.eps = eps
+        self.linear = zero_init(nn.Linear(cond_features, features, bias=False))
+  
+    def extra_repr(self):
+        return f"eps={self.eps},"
+
+    def forward(self, x, cond):
+        return rms_norm(x, self.linear(cond)[:, None, :] + 1, self.eps)
+
+
+def normalize(x, eps=1e-4):
+    dim = list(range(1, x.ndim))
+    n = torch.linalg.vector_norm(x, dim=dim, keepdim=True)
+    alpha = np.sqrt(n.numel() / x.numel())
+    return x / torch.add(eps, n, alpha=alpha)
+
+
+class ForcedWNConv1d(nn.Module):
+    def __init__(self, in_channels, out_channels, kernel_size=1):
+        super().__init__()
+        self.weight = nn.Parameter(torch.randn([out_channels, in_channels, kernel_size]))
+
+    def forward(self, x):
+        if self.training:
+            with torch.no_grad():
+                self.weight.copy_(normalize(self.weight))
+        
+        fan_in = self.weight[0].numel()
+
+        w = normalize(self.weight) / math.sqrt(fan_in)
+
+        return F.conv1d(x, w, padding='same')
+        
+# Kernels
+
+use_compile = True
+
+def compile(function, *args, **kwargs):
+    if not use_compile:
+        return function
+    try:
+        return torch.compile(function, *args, **kwargs)
+    except RuntimeError:
+        return function
+
+
+@compile
+def linear_geglu(x, weight, bias=None):
+    x = x @ weight.mT
+    if bias is not None:
+        x = x + bias
+    x, gate = x.chunk(2, dim=-1)
+    return x * F.gelu(gate)
+
+
+@compile
+def rms_norm(x, scale, eps):
+    dtype = reduce(torch.promote_types, (x.dtype, scale.dtype, torch.float32))
+    mean_sq = torch.mean(x.to(dtype)**2, dim=-1, keepdim=True)
+    scale = scale.to(dtype) * torch.rsqrt(mean_sq + eps)
+    return x * scale.to(x.dtype)
+
+# Layers
+
+
+class LinearGEGLU(nn.Linear):
+    def __init__(self, in_features, out_features, bias=True):
+        super().__init__(in_features, out_features * 2, bias=bias)
+        self.out_features = out_features
+
+    def forward(self, x):
+        return linear_geglu(x, self.weight, self.bias)
+
+
+class RMSNorm(nn.Module):
+    def __init__(self, shape, fix_scale = False, eps=1e-6):
+        super().__init__()
+        self.eps = eps
+
+        if fix_scale:
+            self.register_buffer("scale", torch.ones(shape))
+        else:
+            self.scale = nn.Parameter(torch.ones(shape))
+
+    def extra_repr(self):
+        return f"shape={tuple(self.scale.shape)}, eps={self.eps}"
+
+    def forward(self, x):
+        return rms_norm(x, self.scale, self.eps)
+
+
+# jit script make it 1.4x faster and save GPU memory
+@torch.jit.script
+def snake_beta(x, alpha, beta):
+    return x + (1.0 / (beta + 0.000000001)) * pow(torch.sin(x * alpha), 2)
+
+# try:
+#     snake_beta = torch.compile(snake_beta)
+# except RuntimeError:
+#     pass
+
+
+# Adapted from https://github.com/NVIDIA/BigVGAN/blob/main/activations.py under MIT license
+# License available in LICENSES/LICENSE_NVIDIA.txt
+class SnakeBeta(nn.Module):
+
+    def __init__(self, in_features, alpha=1.0, alpha_trainable=True, alpha_logscale=True):
+        super(SnakeBeta, self).__init__()
+        self.in_features = in_features
+
+        # initialize alpha
+        self.alpha_logscale = alpha_logscale
+        if self.alpha_logscale: 
+            # log scale alphas initialized to zeros
+            self.alpha = nn.Parameter(torch.zeros(in_features) * alpha)
+            self.beta = nn.Parameter(torch.zeros(in_features) * alpha)
+        else:
+            # linear scale alphas initialized to ones
+            self.alpha = nn.Parameter(torch.ones(in_features) * alpha)
+            self.beta = nn.Parameter(torch.ones(in_features) * alpha)
+
+        self.alpha.requires_grad = alpha_trainable
+        self.beta.requires_grad = alpha_trainable
+
+        # self.no_div_by_zero = 0.000000001
+
+    def forward(self, x):
+        alpha = self.alpha.unsqueeze(0).unsqueeze(-1) 
+        # line up with x to [B, C, T]
+        beta = self.beta.unsqueeze(0).unsqueeze(-1)
+        if self.alpha_logscale:
+            alpha = torch.exp(alpha)
+            beta = torch.exp(beta)
+        x = snake_beta(x, alpha, beta)
+
+        return x

vae_modules/stable_vae/models/bottleneck.py

@@ -0,0 +1,346 @@
+import torch
+from torch import nn
+from torch.nn import functional as F
+
+from einops import rearrange
+from vector_quantize_pytorch import ResidualVQ, FSQ
+from .nn.quantize import ResidualVectorQuantize as DACResidualVQ
+
+
+class Bottleneck(nn.Module):
+    def __init__(self, is_discrete: bool = False):
+        super().__init__()
+
+        self.is_discrete = is_discrete
+
+    def encode(self, x, return_info=False, **kwargs):
+        raise NotImplementedError
+
+    def decode(self, x):
+        raise NotImplementedError
+
+
+class DiscreteBottleneck(Bottleneck):
+    def __init__(self, num_quantizers, codebook_size, tokens_id):
+        super().__init__(is_discrete=True)
+
+        self.num_quantizers = num_quantizers
+        self.codebook_size = codebook_size
+        self.tokens_id = tokens_id
+
+    def decode_tokens(self, codes, **kwargs):
+        raise NotImplementedError
+
+
+class TanhBottleneck(Bottleneck):
+    def __init__(self):
+        super().__init__(is_discrete=False)
+        self.tanh = nn.Tanh()
+
+    def encode(self, x, return_info=False):
+        info = {}
+
+        x = torch.tanh(x)
+
+        if return_info:
+            return x, info
+        else:
+            return x
+
+    def decode(self, x):
+        return x
+
+
+@torch.jit.script
+def vae_sample_kl(mean, scale):
+    stdev = nn.functional.softplus(scale) + 1e-4
+    var = stdev * stdev
+    logvar = torch.log(var)
+    latents = torch.randn_like(mean) * stdev + mean
+
+    kl = (mean * mean + var - logvar - 1).sum(1).mean()
+
+    return latents, kl
+
+
+@torch.jit.script
+def vae_sample(mean, scale):
+    stdev = nn.functional.softplus(scale) + 1e-4
+    latents = torch.randn_like(mean) * stdev + mean
+    return latents
+
+
+class VAEBottleneck(Bottleneck):
+    def __init__(self):
+        super().__init__(is_discrete=False)
+
+    def encode(self, x, return_info=False, **kwargs):
+        mean, scale = x.chunk(2, dim=1)
+
+        if return_info:
+            info = {}
+            x, kl = vae_sample_kl(mean, scale)
+            info["kl"] = kl
+            return x, info
+        else:
+            x = vae_sample(mean, scale)
+            return x
+
+    def decode(self, x):
+        return x
+
+
+def compute_mean_kernel(x, y):
+    kernel_input = (x[:, None] - y[None]).pow(2).mean(2) / x.shape[-1]
+    return torch.exp(-kernel_input).mean()
+
+
+def compute_mmd(latents):
+    latents_reshaped = latents.permute(0, 2, 1).reshape(-1, latents.shape[1])
+    noise = torch.randn_like(latents_reshaped)
+
+    latents_kernel = compute_mean_kernel(latents_reshaped, latents_reshaped)
+    noise_kernel = compute_mean_kernel(noise, noise)
+    latents_noise_kernel = compute_mean_kernel(latents_reshaped, noise)
+    
+    mmd = latents_kernel + noise_kernel - 2 * latents_noise_kernel
+    return mmd.mean()
+
+
+class WassersteinBottleneck(Bottleneck):
+    def __init__(self, noise_augment_dim: int = 0):
+        super().__init__(is_discrete=False)
+
+        self.noise_augment_dim = noise_augment_dim
+    
+    def encode(self, x, return_info=False):
+        info = {}
+
+        if self.training and return_info:
+            mmd = compute_mmd(x)
+            info["mmd"] = mmd
+        
+        if return_info:
+            return x, info
+        
+        return x
+
+    def decode(self, x):
+
+        if self.noise_augment_dim > 0:
+            noise = torch.randn(x.shape[0], self.noise_augment_dim,
+                                x.shape[-1]).type_as(x)
+            x = torch.cat([x, noise], dim=1)
+
+        return x
+
+
+class L2Bottleneck(Bottleneck):
+    def __init__(self):
+        super().__init__(is_discrete=False)
+    
+    def encode(self, x, return_info=False):
+        info = {}
+
+        x = F.normalize(x, dim=1)
+
+        if return_info:
+            return x, info
+        else:
+            return x
+        
+    def decode(self, x):
+        return F.normalize(x, dim=1)
+
+
+class RVQBottleneck(DiscreteBottleneck):
+    def __init__(self, **quantizer_kwargs):
+        super().__init__(num_quantizers = quantizer_kwargs["num_quantizers"], codebook_size = quantizer_kwargs["codebook_size"], tokens_id = "quantizer_indices")
+        self.quantizer = ResidualVQ(**quantizer_kwargs)
+        self.num_quantizers = quantizer_kwargs["num_quantizers"]
+
+    def encode(self, x, return_info=False, **kwargs):
+        info = {}
+
+        x = rearrange(x, "b c n -> b n c")
+        x, indices, loss = self.quantizer(x)
+        x = rearrange(x, "b n c -> b c n")
+
+        info["quantizer_indices"] = indices
+        info["quantizer_loss"] = loss.mean()
+
+        if return_info:
+            return x, info
+        else:
+            return x
+        
+    def decode(self, x):
+        return x
+    
+    def decode_tokens(self, codes, **kwargs):
+        latents = self.quantizer.get_outputs_from_indices(codes)
+
+        return self.decode(latents, **kwargs)
+
+
+class RVQVAEBottleneck(DiscreteBottleneck):
+    def __init__(self, **quantizer_kwargs):
+        super().__init__(num_quantizers = quantizer_kwargs["num_quantizers"], codebook_size = quantizer_kwargs["codebook_size"], tokens_id = "quantizer_indices")
+        self.quantizer = ResidualVQ(**quantizer_kwargs)
+        self.num_quantizers = quantizer_kwargs["num_quantizers"]
+
+    def encode(self, x, return_info=False):
+        info = {}
+
+        x, kl = vae_sample(*x.chunk(2, dim=1))
+
+        info["kl"] = kl
+
+        x = rearrange(x, "b c n -> b n c")
+        x, indices, loss = self.quantizer(x)
+        x = rearrange(x, "b n c -> b c n")
+
+        info["quantizer_indices"] = indices
+        info["quantizer_loss"] = loss.mean()
+
+        if return_info:
+            return x, info
+        else:
+            return x
+        
+    def decode(self, x):
+        return x
+    
+    def decode_tokens(self, codes, **kwargs):
+        latents = self.quantizer.get_outputs_from_indices(codes)
+
+        return self.decode(latents, **kwargs)
+
+
+class DACRVQBottleneck(DiscreteBottleneck):
+    def __init__(self, quantize_on_decode=False, **quantizer_kwargs):
+        super().__init__(num_quantizers = quantizer_kwargs["n_codebooks"], codebook_size = quantizer_kwargs["codebook_size"], tokens_id = "codes")
+        self.quantizer = DACResidualVQ(**quantizer_kwargs)
+        self.num_quantizers = quantizer_kwargs["n_codebooks"]
+        self.quantize_on_decode = quantize_on_decode
+
+    def encode(self, x, return_info=False, **kwargs):
+        info = {}
+
+        info["pre_quantizer"] = x
+
+        if self.quantize_on_decode:
+            return x, info if return_info else x
+
+        z, codes, latents, commitment_loss, codebook_loss = self.quantizer(x, **kwargs)
+
+        output = {
+            "z": z,
+            "codes": codes,
+            "latents": latents,
+            "vq/commitment_loss": commitment_loss,
+            "vq/codebook_loss": codebook_loss,
+        }
+
+        output["vq/commitment_loss"] /= self.num_quantizers
+        output["vq/codebook_loss"] /= self.num_quantizers
+
+        info.update(output)
+
+        if return_info:
+            return output["z"], info
+        
+        return output["z"]
+    
+    def decode(self, x):
+
+        if self.quantize_on_decode:
+            x = self.quantizer(x)[0]
+
+        return x
+    
+    def decode_tokens(self, codes, **kwargs):
+        latents, _, _ = self.quantizer.from_codes(codes)
+
+        return self.decode(latents, **kwargs)
+
+
+class DACRVQVAEBottleneck(DiscreteBottleneck):
+    def __init__(self, quantize_on_decode=False, **quantizer_kwargs):
+        super().__init__(num_quantizers = quantizer_kwargs["n_codebooks"], codebook_size = quantizer_kwargs["codebook_size"], tokens_id = "codes")
+        self.quantizer = DACResidualVQ(**quantizer_kwargs)
+        self.num_quantizers = quantizer_kwargs["n_codebooks"]
+        self.quantize_on_decode = quantize_on_decode
+
+    def encode(self, x, return_info=False, n_quantizers: int = None):
+        info = {}
+
+        mean, scale = x.chunk(2, dim=1)
+
+        x, kl = vae_sample(mean, scale)
+
+        info["pre_quantizer"] = x
+        info["kl"] = kl
+
+        if self.quantize_on_decode:
+            return x, info if return_info else x
+
+        z, codes, latents, commitment_loss, codebook_loss = self.quantizer(x, n_quantizers=n_quantizers)
+
+        output = {
+            "z": z,
+            "codes": codes,
+            "latents": latents,
+            "vq/commitment_loss": commitment_loss,
+            "vq/codebook_loss": codebook_loss,
+        }
+
+        output["vq/commitment_loss"] /= self.num_quantizers
+        output["vq/codebook_loss"] /= self.num_quantizers
+
+        info.update(output)
+
+        if return_info:
+            return output["z"], info
+        
+        return output["z"]
+    
+    def decode(self, x):
+
+        if self.quantize_on_decode:
+            x = self.quantizer(x)[0]
+
+        return x
+
+    def decode_tokens(self, codes, **kwargs):
+        latents, _, _ = self.quantizer.from_codes(codes)
+
+        return self.decode(latents, **kwargs)
+
+
+class FSQBottleneck(DiscreteBottleneck):
+    def __init__(self, dim, levels):
+        super().__init__(num_quantizers = 1, codebook_size = levels ** dim, tokens_id = "quantizer_indices")
+        self.quantizer = FSQ(levels=[levels] * dim)
+
+    def encode(self, x, return_info=False):
+        info = {}
+
+        x = rearrange(x, "b c n -> b n c")
+        x, indices = self.quantizer(x)
+        x = rearrange(x, "b n c -> b c n")
+
+        info["quantizer_indices"] = indices
+
+        if return_info:
+            return x, info
+        else:
+            return x
+        
+    def decode(self, x):
+        return x
+    
+    def decode_tokens(self, tokens, **kwargs):
+        latents = self.quantizer.indices_to_codes(tokens)
+
+        return self.decode(latents, **kwargs)

vae_modules/stable_vae/models/factory.py

@@ -0,0 +1,153 @@
+import json
+
+def create_model_from_config(model_config):
+    model_type = model_config.get('model_type', None)
+
+    assert model_type is not None, 'model_type must be specified in model config'
+
+    if model_type == 'autoencoder':
+        from .autoencoders import create_autoencoder_from_config
+        return create_autoencoder_from_config(model_config)
+    elif model_type == 'diffusion_uncond':
+        from .diffusion import create_diffusion_uncond_from_config
+        return create_diffusion_uncond_from_config(model_config)
+    elif model_type == 'diffusion_cond' or model_type == 'diffusion_cond_inpaint' or model_type == "diffusion_prior":
+        from .diffusion import create_diffusion_cond_from_config
+        return create_diffusion_cond_from_config(model_config)
+    elif model_type == 'diffusion_autoencoder':
+        from .autoencoders import create_diffAE_from_config
+        return create_diffAE_from_config(model_config)
+    elif model_type == 'lm':
+        from .lm import create_audio_lm_from_config
+        return create_audio_lm_from_config(model_config)
+    else:
+        raise NotImplementedError(f'Unknown model type: {model_type}')
+
+def create_model_from_config_path(model_config_path):
+    with open(model_config_path) as f:
+        model_config = json.load(f)
+    
+    return create_model_from_config(model_config)
+
+def create_pretransform_from_config(pretransform_config, sample_rate):
+    pretransform_type = pretransform_config.get('type', None)
+
+    assert pretransform_type is not None, 'type must be specified in pretransform config'
+
+    if pretransform_type == 'autoencoder':
+        from .autoencoders import create_autoencoder_from_config
+        from .pretransforms import AutoencoderPretransform
+
+        # Create fake top-level config to pass sample rate to autoencoder constructor
+        # This is a bit of a hack but it keeps us from re-defining the sample rate in the config
+        autoencoder_config = {"sample_rate": sample_rate, "model": pretransform_config["config"]}
+        autoencoder = create_autoencoder_from_config(autoencoder_config)
+
+        scale = pretransform_config.get("scale", 1.0)
+        model_half = pretransform_config.get("model_half", False)
+        iterate_batch = pretransform_config.get("iterate_batch", False)
+        chunked = pretransform_config.get("chunked", False)
+
+        pretransform = AutoencoderPretransform(autoencoder, scale=scale, model_half=model_half, iterate_batch=iterate_batch, chunked=chunked)
+    elif pretransform_type == 'wavelet':
+        from .pretransforms import WaveletPretransform
+
+        wavelet_config = pretransform_config["config"]
+        channels = wavelet_config["channels"]
+        levels = wavelet_config["levels"]
+        wavelet = wavelet_config["wavelet"]
+
+        pretransform = WaveletPretransform(channels, levels, wavelet)
+    elif pretransform_type == 'pqmf':
+        from .pretransforms import PQMFPretransform
+        pqmf_config = pretransform_config["config"]
+        pretransform = PQMFPretransform(**pqmf_config)
+    elif pretransform_type == 'dac_pretrained':
+        from .pretransforms import PretrainedDACPretransform
+        pretrained_dac_config = pretransform_config["config"]
+        pretransform = PretrainedDACPretransform(**pretrained_dac_config)
+    elif pretransform_type == "audiocraft_pretrained":
+        from .pretransforms import AudiocraftCompressionPretransform
+
+        audiocraft_config = pretransform_config["config"]
+        pretransform = AudiocraftCompressionPretransform(**audiocraft_config)
+    else:
+        raise NotImplementedError(f'Unknown pretransform type: {pretransform_type}')
+    
+    enable_grad = pretransform_config.get('enable_grad', False)
+    pretransform.enable_grad = enable_grad
+
+    pretransform.eval().requires_grad_(pretransform.enable_grad)
+
+    return pretransform
+
+def create_bottleneck_from_config(bottleneck_config):
+    bottleneck_type = bottleneck_config.get('type', None)
+
+    assert bottleneck_type is not None, 'type must be specified in bottleneck config'
+
+    if bottleneck_type == 'tanh':
+        from .bottleneck import TanhBottleneck
+        bottleneck = TanhBottleneck()
+    elif bottleneck_type == 'vae':
+        from .bottleneck import VAEBottleneck
+        bottleneck = VAEBottleneck()
+    elif bottleneck_type == 'rvq':
+        from .bottleneck import RVQBottleneck
+
+        quantizer_params = {
+            "dim": 128,
+            "codebook_size": 1024,
+            "num_quantizers": 8,
+            "decay": 0.99,
+            "kmeans_init": True,
+            "kmeans_iters": 50,
+            "threshold_ema_dead_code": 2,
+        }
+
+        quantizer_params.update(bottleneck_config["config"])
+
+        bottleneck = RVQBottleneck(**quantizer_params)
+    elif bottleneck_type == "dac_rvq":
+        from .bottleneck import DACRVQBottleneck
+
+        bottleneck = DACRVQBottleneck(**bottleneck_config["config"])
+    
+    elif bottleneck_type == 'rvq_vae':
+        from .bottleneck import RVQVAEBottleneck
+
+        quantizer_params = {
+            "dim": 128,
+            "codebook_size": 1024,
+            "num_quantizers": 8,
+            "decay": 0.99,
+            "kmeans_init": True,
+            "kmeans_iters": 50,
+            "threshold_ema_dead_code": 2,
+        }
+
+        quantizer_params.update(bottleneck_config["config"])
+
+        bottleneck = RVQVAEBottleneck(**quantizer_params)
+        
+    elif bottleneck_type == 'dac_rvq_vae':
+        from .bottleneck import DACRVQVAEBottleneck
+        bottleneck = DACRVQVAEBottleneck(**bottleneck_config["config"])
+    elif bottleneck_type == 'l2_norm':
+        from .bottleneck import L2Bottleneck
+        bottleneck = L2Bottleneck()
+    elif bottleneck_type == "wasserstein":
+        from .bottleneck import WassersteinBottleneck
+        bottleneck = WassersteinBottleneck(**bottleneck_config.get("config", {}))
+    elif bottleneck_type == "fsq":
+        from .bottleneck import FSQBottleneck
+        bottleneck = FSQBottleneck(**bottleneck_config["config"])
+    else:
+        raise NotImplementedError(f'Unknown bottleneck type: {bottleneck_type}')
+    
+    requires_grad = bottleneck_config.get('requires_grad', True)
+    if not requires_grad:
+        for param in bottleneck.parameters():
+            param.requires_grad = False
+
+    return bottleneck

vae_modules/stable_vae/models/nn/__init__.py

@@ -0,0 +1,3 @@
+from . import layers
+from . import loss
+from . import quantize

vae_modules/stable_vae/models/nn/layers.py

@@ -0,0 +1,33 @@
+import numpy as np
+import torch
+import torch.nn as nn
+import torch.nn.functional as F
+from einops import rearrange
+from torch.nn.utils import weight_norm
+
+
+def WNConv1d(*args, **kwargs):
+    return weight_norm(nn.Conv1d(*args, **kwargs))
+
+
+def WNConvTranspose1d(*args, **kwargs):
+    return weight_norm(nn.ConvTranspose1d(*args, **kwargs))
+
+
+# Scripting this brings model speed up 1.4x
+@torch.jit.script
+def snake(x, alpha):
+    shape = x.shape
+    x = x.reshape(shape[0], shape[1], -1)
+    x = x + (alpha + 1e-9).reciprocal() * torch.sin(alpha * x).pow(2)
+    x = x.reshape(shape)
+    return x
+
+
+class Snake1d(nn.Module):
+    def __init__(self, channels):
+        super().__init__()
+        self.alpha = nn.Parameter(torch.ones(1, channels, 1))
+
+    def forward(self, x):
+        return snake(x, self.alpha)

vae_modules/stable_vae/models/nn/loss.py

@@ -0,0 +1,368 @@
+import typing
+from typing import List
+
+import torch
+import torch.nn.functional as F
+from audiotools import AudioSignal
+from audiotools import STFTParams
+from torch import nn
+
+
+class L1Loss(nn.L1Loss):
+    """L1 Loss between AudioSignals. Defaults
+    to comparing ``audio_data``, but any
+    attribute of an AudioSignal can be used.
+
+    Parameters
+    ----------
+    attribute : str, optional
+        Attribute of signal to compare, defaults to ``audio_data``.
+    weight : float, optional
+        Weight of this loss, defaults to 1.0.
+
+    Implementation copied from: https://github.com/descriptinc/lyrebird-audiotools/blob/961786aa1a9d628cca0c0486e5885a457fe70c1a/audiotools/metrics/distance.py
+    """
+
+    def __init__(self, attribute: str = "audio_data", weight: float = 1.0, **kwargs):
+        self.attribute = attribute
+        self.weight = weight
+        super().__init__(**kwargs)
+
+    def forward(self, x: AudioSignal, y: AudioSignal):
+        """
+        Parameters
+        ----------
+        x : AudioSignal
+            Estimate AudioSignal
+        y : AudioSignal
+            Reference AudioSignal
+
+        Returns
+        -------
+        torch.Tensor
+            L1 loss between AudioSignal attributes.
+        """
+        if isinstance(x, AudioSignal):
+            x = getattr(x, self.attribute)
+            y = getattr(y, self.attribute)
+        return super().forward(x, y)
+
+
+class SISDRLoss(nn.Module):
+    """
+    Computes the Scale-Invariant Source-to-Distortion Ratio between a batch
+    of estimated and reference audio signals or aligned features.
+
+    Parameters
+    ----------
+    scaling : int, optional
+        Whether to use scale-invariant (True) or
+        signal-to-noise ratio (False), by default True
+    reduction : str, optional
+        How to reduce across the batch (either 'mean',
+        'sum', or none).], by default ' mean'
+    zero_mean : int, optional
+        Zero mean the references and estimates before
+        computing the loss, by default True
+    clip_min : int, optional
+        The minimum possible loss value. Helps network
+        to not focus on making already good examples better, by default None
+    weight : float, optional
+        Weight of this loss, defaults to 1.0.
+
+    Implementation copied from: https://github.com/descriptinc/lyrebird-audiotools/blob/961786aa1a9d628cca0c0486e5885a457fe70c1a/audiotools/metrics/distance.py
+    """
+
+    def __init__(
+        self,
+        scaling: int = True,
+        reduction: str = "mean",
+        zero_mean: int = True,
+        clip_min: int = None,
+        weight: float = 1.0,
+    ):
+        self.scaling = scaling
+        self.reduction = reduction
+        self.zero_mean = zero_mean
+        self.clip_min = clip_min
+        self.weight = weight
+        super().__init__()
+
+    def forward(self, x: AudioSignal, y: AudioSignal):
+        eps = 1e-8
+        # nb, nc, nt
+        if isinstance(x, AudioSignal):
+            references = x.audio_data
+            estimates = y.audio_data
+        else:
+            references = x
+            estimates = y
+
+        nb = references.shape[0]
+        references = references.reshape(nb, 1, -1).permute(0, 2, 1)
+        estimates = estimates.reshape(nb, 1, -1).permute(0, 2, 1)
+
+        # samples now on axis 1
+        if self.zero_mean:
+            mean_reference = references.mean(dim=1, keepdim=True)
+            mean_estimate = estimates.mean(dim=1, keepdim=True)
+        else:
+            mean_reference = 0
+            mean_estimate = 0
+
+        _references = references - mean_reference
+        _estimates = estimates - mean_estimate
+
+        references_projection = (_references**2).sum(dim=-2) + eps
+        references_on_estimates = (_estimates * _references).sum(dim=-2) + eps
+
+        scale = (
+            (references_on_estimates / references_projection).unsqueeze(1)
+            if self.scaling
+            else 1
+        )
+
+        e_true = scale * _references
+        e_res = _estimates - e_true
+
+        signal = (e_true**2).sum(dim=1)
+        noise = (e_res**2).sum(dim=1)
+        sdr = -10 * torch.log10(signal / noise + eps)
+
+        if self.clip_min is not None:
+            sdr = torch.clamp(sdr, min=self.clip_min)
+
+        if self.reduction == "mean":
+            sdr = sdr.mean()
+        elif self.reduction == "sum":
+            sdr = sdr.sum()
+        return sdr
+
+
+class MultiScaleSTFTLoss(nn.Module):
+    """Computes the multi-scale STFT loss from [1].
+
+    Parameters
+    ----------
+    window_lengths : List[int], optional
+        Length of each window of each STFT, by default [2048, 512]
+    loss_fn : typing.Callable, optional
+        How to compare each loss, by default nn.L1Loss()
+    clamp_eps : float, optional
+        Clamp on the log magnitude, below, by default 1e-5
+    mag_weight : float, optional
+        Weight of raw magnitude portion of loss, by default 1.0
+    log_weight : float, optional
+        Weight of log magnitude portion of loss, by default 1.0
+    pow : float, optional
+        Power to raise magnitude to before taking log, by default 2.0
+    weight : float, optional
+        Weight of this loss, by default 1.0
+    match_stride : bool, optional
+        Whether to match the stride of convolutional layers, by default False
+
+    References
+    ----------
+
+    1.  Engel, Jesse, Chenjie Gu, and Adam Roberts.
+        "DDSP: Differentiable Digital Signal Processing."
+        International Conference on Learning Representations. 2019.
+
+    Implementation copied from: https://github.com/descriptinc/lyrebird-audiotools/blob/961786aa1a9d628cca0c0486e5885a457fe70c1a/audiotools/metrics/spectral.py
+    """
+
+    def __init__(
+        self,
+        window_lengths: List[int] = [2048, 512],
+        loss_fn: typing.Callable = nn.L1Loss(),
+        clamp_eps: float = 1e-5,
+        mag_weight: float = 1.0,
+        log_weight: float = 1.0,
+        pow: float = 2.0,
+        weight: float = 1.0,
+        match_stride: bool = False,
+        window_type: str = None,
+    ):
+        super().__init__()
+        self.stft_params = [
+            STFTParams(
+                window_length=w,
+                hop_length=w // 4,
+                match_stride=match_stride,
+                window_type=window_type,
+            )
+            for w in window_lengths
+        ]
+        self.loss_fn = loss_fn
+        self.log_weight = log_weight
+        self.mag_weight = mag_weight
+        self.clamp_eps = clamp_eps
+        self.weight = weight
+        self.pow = pow
+
+    def forward(self, x: AudioSignal, y: AudioSignal):
+        """Computes multi-scale STFT between an estimate and a reference
+        signal.
+
+        Parameters
+        ----------
+        x : AudioSignal
+            Estimate signal
+        y : AudioSignal
+            Reference signal
+
+        Returns
+        -------
+        torch.Tensor
+            Multi-scale STFT loss.
+        """
+        loss = 0.0
+        for s in self.stft_params:
+            x.stft(s.window_length, s.hop_length, s.window_type)
+            y.stft(s.window_length, s.hop_length, s.window_type)
+            loss += self.log_weight * self.loss_fn(
+                x.magnitude.clamp(self.clamp_eps).pow(self.pow).log10(),
+                y.magnitude.clamp(self.clamp_eps).pow(self.pow).log10(),
+            )
+            loss += self.mag_weight * self.loss_fn(x.magnitude, y.magnitude)
+        return loss
+
+
+class MelSpectrogramLoss(nn.Module):
+    """Compute distance between mel spectrograms. Can be used
+    in a multi-scale way.
+
+    Parameters
+    ----------
+    n_mels : List[int]
+        Number of mels per STFT, by default [150, 80],
+    window_lengths : List[int], optional
+        Length of each window of each STFT, by default [2048, 512]
+    loss_fn : typing.Callable, optional
+        How to compare each loss, by default nn.L1Loss()
+    clamp_eps : float, optional
+        Clamp on the log magnitude, below, by default 1e-5
+    mag_weight : float, optional
+        Weight of raw magnitude portion of loss, by default 1.0
+    log_weight : float, optional
+        Weight of log magnitude portion of loss, by default 1.0
+    pow : float, optional
+        Power to raise magnitude to before taking log, by default 2.0
+    weight : float, optional
+        Weight of this loss, by default 1.0
+    match_stride : bool, optional
+        Whether to match the stride of convolutional layers, by default False
+
+    Implementation copied from: https://github.com/descriptinc/lyrebird-audiotools/blob/961786aa1a9d628cca0c0486e5885a457fe70c1a/audiotools/metrics/spectral.py
+    """
+
+    def __init__(
+        self,
+        n_mels: List[int] = [150, 80],
+        window_lengths: List[int] = [2048, 512],
+        loss_fn: typing.Callable = nn.L1Loss(),
+        clamp_eps: float = 1e-5,
+        mag_weight: float = 1.0,
+        log_weight: float = 1.0,
+        pow: float = 2.0,
+        weight: float = 1.0,
+        match_stride: bool = False,
+        mel_fmin: List[float] = [0.0, 0.0],
+        mel_fmax: List[float] = [None, None],
+        window_type: str = None,
+    ):
+        super().__init__()
+        self.stft_params = [
+            STFTParams(
+                window_length=w,
+                hop_length=w // 4,
+                match_stride=match_stride,
+                window_type=window_type,
+            )
+            for w in window_lengths
+        ]
+        self.n_mels = n_mels
+        self.loss_fn = loss_fn
+        self.clamp_eps = clamp_eps
+        self.log_weight = log_weight
+        self.mag_weight = mag_weight
+        self.weight = weight
+        self.mel_fmin = mel_fmin
+        self.mel_fmax = mel_fmax
+        self.pow = pow
+
+    def forward(self, x: AudioSignal, y: AudioSignal):
+        """Computes mel loss between an estimate and a reference
+        signal.
+
+        Parameters
+        ----------
+        x : AudioSignal
+            Estimate signal
+        y : AudioSignal
+            Reference signal
+
+        Returns
+        -------
+        torch.Tensor
+            Mel loss.
+        """
+        loss = 0.0
+        for n_mels, fmin, fmax, s in zip(
+            self.n_mels, self.mel_fmin, self.mel_fmax, self.stft_params
+        ):
+            kwargs = {
+                "window_length": s.window_length,
+                "hop_length": s.hop_length,
+                "window_type": s.window_type,
+            }
+            x_mels = x.mel_spectrogram(n_mels, mel_fmin=fmin, mel_fmax=fmax, **kwargs)
+            y_mels = y.mel_spectrogram(n_mels, mel_fmin=fmin, mel_fmax=fmax, **kwargs)
+
+            loss += self.log_weight * self.loss_fn(
+                x_mels.clamp(self.clamp_eps).pow(self.pow).log10(),
+                y_mels.clamp(self.clamp_eps).pow(self.pow).log10(),
+            )
+            loss += self.mag_weight * self.loss_fn(x_mels, y_mels)
+        return loss
+
+
+class GANLoss(nn.Module):
+    """
+    Computes a discriminator loss, given a discriminator on
+    generated waveforms/spectrograms compared to ground truth
+    waveforms/spectrograms. Computes the loss for both the
+    discriminator and the generator in separate functions.
+    """
+
+    def __init__(self, discriminator):
+        super().__init__()
+        self.discriminator = discriminator
+
+    def forward(self, fake, real):
+        d_fake = self.discriminator(fake.audio_data)
+        d_real = self.discriminator(real.audio_data)
+        return d_fake, d_real
+
+    def discriminator_loss(self, fake, real):
+        d_fake, d_real = self.forward(fake.clone().detach(), real)
+
+        loss_d = 0
+        for x_fake, x_real in zip(d_fake, d_real):
+            loss_d += torch.mean(x_fake[-1] ** 2)
+            loss_d += torch.mean((1 - x_real[-1]) ** 2)
+        return loss_d
+
+    def generator_loss(self, fake, real):
+        d_fake, d_real = self.forward(fake, real)
+
+        loss_g = 0
+        for x_fake in d_fake:
+            loss_g += torch.mean((1 - x_fake[-1]) ** 2)
+
+        loss_feature = 0
+
+        for i in range(len(d_fake)):
+            for j in range(len(d_fake[i]) - 1):
+                loss_feature += F.l1_loss(d_fake[i][j], d_real[i][j].detach())
+        return loss_g, loss_feature

vae_modules/stable_vae/models/nn/quantize.py

@@ -0,0 +1,262 @@
+from typing import Union
+
+import numpy as np
+import torch
+import torch.nn as nn
+import torch.nn.functional as F
+from einops import rearrange
+from torch.nn.utils import weight_norm
+
+from .layers import WNConv1d
+
+
+class VectorQuantize(nn.Module):
+    """
+    Implementation of VQ similar to Karpathy's repo:
+    https://github.com/karpathy/deep-vector-quantization
+    Additionally uses following tricks from Improved VQGAN
+    (https://arxiv.org/pdf/2110.04627.pdf):
+        1. Factorized codes: Perform nearest neighbor lookup in low-dimensional space
+            for improved codebook usage
+        2. l2-normalized codes: Converts euclidean distance to cosine similarity which
+            improves training stability
+    """
+
+    def __init__(self, input_dim: int, codebook_size: int, codebook_dim: int):
+        super().__init__()
+        self.codebook_size = codebook_size
+        self.codebook_dim = codebook_dim
+
+        self.in_proj = WNConv1d(input_dim, codebook_dim, kernel_size=1)
+        self.out_proj = WNConv1d(codebook_dim, input_dim, kernel_size=1)
+        self.codebook = nn.Embedding(codebook_size, codebook_dim)
+
+    def forward(self, z):
+        """Quantized the input tensor using a fixed codebook and returns
+        the corresponding codebook vectors
+
+        Parameters
+        ----------
+        z : Tensor[B x D x T]
+
+        Returns
+        -------
+        Tensor[B x D x T]
+            Quantized continuous representation of input
+        Tensor[1]
+            Commitment loss to train encoder to predict vectors closer to codebook
+            entries
+        Tensor[1]
+            Codebook loss to update the codebook
+        Tensor[B x T]
+            Codebook indices (quantized discrete representation of input)
+        Tensor[B x D x T]
+            Projected latents (continuous representation of input before quantization)
+        """
+
+        # Factorized codes (ViT-VQGAN) Project input into low-dimensional space
+        z_e = self.in_proj(z)  # z_e : (B x D x T)
+        z_q, indices = self.decode_latents(z_e)
+
+        commitment_loss = F.mse_loss(z_e, z_q.detach(), reduction="none").mean([1, 2])
+        codebook_loss = F.mse_loss(z_q, z_e.detach(), reduction="none").mean([1, 2])
+
+        z_q = (
+            z_e + (z_q - z_e).detach()
+        )  # noop in forward pass, straight-through gradient estimator in backward pass
+
+        z_q = self.out_proj(z_q)
+
+        return z_q, commitment_loss, codebook_loss, indices, z_e
+
+    def embed_code(self, embed_id):
+        return F.embedding(embed_id, self.codebook.weight)
+
+    def decode_code(self, embed_id):
+        return self.embed_code(embed_id).transpose(1, 2)
+
+    def decode_latents(self, latents):
+        encodings = rearrange(latents, "b d t -> (b t) d")
+        codebook = self.codebook.weight  # codebook: (N x D)
+
+        # L2 normalize encodings and codebook (ViT-VQGAN)
+        encodings = F.normalize(encodings)
+        codebook = F.normalize(codebook)
+
+        # Compute euclidean distance with codebook
+        dist = (
+            encodings.pow(2).sum(1, keepdim=True)
+            - 2 * encodings @ codebook.t()
+            + codebook.pow(2).sum(1, keepdim=True).t()
+        )
+        indices = rearrange((-dist).max(1)[1], "(b t) -> b t", b=latents.size(0))
+        z_q = self.decode_code(indices)
+        return z_q, indices
+
+
+class ResidualVectorQuantize(nn.Module):
+    """
+    Introduced in SoundStream: An end2end neural audio codec
+    https://arxiv.org/abs/2107.03312
+    """
+
+    def __init__(
+        self,
+        input_dim: int = 512,
+        n_codebooks: int = 9,
+        codebook_size: int = 1024,
+        codebook_dim: Union[int, list] = 8,
+        quantizer_dropout: float = 0.0,
+    ):
+        super().__init__()
+        if isinstance(codebook_dim, int):
+            codebook_dim = [codebook_dim for _ in range(n_codebooks)]
+
+        self.n_codebooks = n_codebooks
+        self.codebook_dim = codebook_dim
+        self.codebook_size = codebook_size
+
+        self.quantizers = nn.ModuleList(
+            [
+                VectorQuantize(input_dim, codebook_size, codebook_dim[i])
+                for i in range(n_codebooks)
+            ]
+        )
+        self.quantizer_dropout = quantizer_dropout
+
+    def forward(self, z, n_quantizers: int = None):
+        """Quantized the input tensor using a fixed set of `n` codebooks and returns
+        the corresponding codebook vectors
+        Parameters
+        ----------
+        z : Tensor[B x D x T]
+        n_quantizers : int, optional
+            No. of quantizers to use
+            (n_quantizers < self.n_codebooks ex: for quantizer dropout)
+            Note: if `self.quantizer_dropout` is True, this argument is ignored
+                when in training mode, and a random number of quantizers is used.
+        Returns
+        -------
+        dict
+            A dictionary with the following keys:
+
+            "z" : Tensor[B x D x T]
+                Quantized continuous representation of input
+            "codes" : Tensor[B x N x T]
+                Codebook indices for each codebook
+                (quantized discrete representation of input)
+            "latents" : Tensor[B x N*D x T]
+                Projected latents (continuous representation of input before quantization)
+            "vq/commitment_loss" : Tensor[1]
+                Commitment loss to train encoder to predict vectors closer to codebook
+                entries
+            "vq/codebook_loss" : Tensor[1]
+                Codebook loss to update the codebook
+        """
+        z_q = 0
+        residual = z
+        commitment_loss = 0
+        codebook_loss = 0
+
+        codebook_indices = []
+        latents = []
+
+        if n_quantizers is None:
+            n_quantizers = self.n_codebooks
+        if self.training:
+            n_quantizers = torch.ones((z.shape[0],)) * self.n_codebooks + 1
+            dropout = torch.randint(1, self.n_codebooks + 1, (z.shape[0],))
+            n_dropout = int(z.shape[0] * self.quantizer_dropout)
+            n_quantizers[:n_dropout] = dropout[:n_dropout]
+            n_quantizers = n_quantizers.to(z.device)
+
+        for i, quantizer in enumerate(self.quantizers):
+            if self.training is False and i >= n_quantizers:
+                break
+
+            z_q_i, commitment_loss_i, codebook_loss_i, indices_i, z_e_i = quantizer(
+                residual
+            )
+
+            # Create mask to apply quantizer dropout
+            mask = (
+                torch.full((z.shape[0],), fill_value=i, device=z.device) < n_quantizers
+            )
+            z_q = z_q + z_q_i * mask[:, None, None]
+            residual = residual - z_q_i
+
+            # Sum losses
+            commitment_loss += (commitment_loss_i * mask).mean()
+            codebook_loss += (codebook_loss_i * mask).mean()
+
+            codebook_indices.append(indices_i)
+            latents.append(z_e_i)
+
+        codes = torch.stack(codebook_indices, dim=1)
+        latents = torch.cat(latents, dim=1)
+
+        return z_q, codes, latents, commitment_loss, codebook_loss
+
+    def from_codes(self, codes: torch.Tensor):
+        """Given the quantized codes, reconstruct the continuous representation
+        Parameters
+        ----------
+        codes : Tensor[B x N x T]
+            Quantized discrete representation of input
+        Returns
+        -------
+        Tensor[B x D x T]
+            Quantized continuous representation of input
+        """
+        z_q = 0.0
+        z_p = []
+        n_codebooks = codes.shape[1]
+        for i in range(n_codebooks):
+            z_p_i = self.quantizers[i].decode_code(codes[:, i, :])
+            z_p.append(z_p_i)
+
+            z_q_i = self.quantizers[i].out_proj(z_p_i)
+            z_q = z_q + z_q_i
+        return z_q, torch.cat(z_p, dim=1), codes
+
+    def from_latents(self, latents: torch.Tensor):
+        """Given the unquantized latents, reconstruct the
+        continuous representation after quantization.
+
+        Parameters
+        ----------
+        latents : Tensor[B x N x T]
+            Continuous representation of input after projection
+
+        Returns
+        -------
+        Tensor[B x D x T]
+            Quantized representation of full-projected space
+        Tensor[B x D x T]
+            Quantized representation of latent space
+        """
+        z_q = 0
+        z_p = []
+        codes = []
+        dims = np.cumsum([0] + [q.codebook_dim for q in self.quantizers])
+
+        n_codebooks = np.where(dims <= latents.shape[1])[0].max(axis=0, keepdims=True)[
+            0
+        ]
+        for i in range(n_codebooks):
+            j, k = dims[i], dims[i + 1]
+            z_p_i, codes_i = self.quantizers[i].decode_latents(latents[:, j:k, :])
+            z_p.append(z_p_i)
+            codes.append(codes_i)
+
+            z_q_i = self.quantizers[i].out_proj(z_p_i)
+            z_q = z_q + z_q_i
+
+        return z_q, torch.cat(z_p, dim=1), torch.stack(codes, dim=1)
+
+
+if __name__ == "__main__":
+    rvq = ResidualVectorQuantize(quantizer_dropout=True)
+    x = torch.randn(16, 512, 80)
+    y = rvq(x)
+    print(y["latents"].shape)

vae_modules/stable_vae/models/pretransforms.py

@@ -0,0 +1,258 @@
+import torch
+from einops import rearrange
+from torch import nn
+
+class Pretransform(nn.Module):
+    def __init__(self, enable_grad, io_channels, is_discrete):
+        super().__init__()
+
+        self.is_discrete = is_discrete
+        self.io_channels = io_channels
+        self.encoded_channels = None
+        self.downsampling_ratio = None
+
+        self.enable_grad = enable_grad
+
+    def encode(self, x):
+        raise NotImplementedError
+
+    def decode(self, z):
+        raise NotImplementedError
+    
+    def tokenize(self, x):
+        raise NotImplementedError
+    
+    def decode_tokens(self, tokens):
+        raise NotImplementedError
+
+class AutoencoderPretransform(Pretransform):
+    def __init__(self, model, scale=1.0, model_half=False, iterate_batch=False, chunked=False):
+        super().__init__(enable_grad=False, io_channels=model.io_channels, is_discrete=model.bottleneck is not None and model.bottleneck.is_discrete)
+        self.model = model
+        self.model.requires_grad_(False).eval()
+        self.scale=scale
+        self.downsampling_ratio = model.downsampling_ratio
+        self.io_channels = model.io_channels
+        self.sample_rate = model.sample_rate
+        
+        self.model_half = model_half
+        self.iterate_batch = iterate_batch
+
+        self.encoded_channels = model.latent_dim
+
+        self.chunked = chunked
+        self.num_quantizers = model.bottleneck.num_quantizers if model.bottleneck is not None and model.bottleneck.is_discrete else None
+        self.codebook_size = model.bottleneck.codebook_size if model.bottleneck is not None and model.bottleneck.is_discrete else None
+
+        if self.model_half:
+            self.model.half()
+    
+    def encode(self, x, **kwargs):
+        
+        if self.model_half:
+            x = x.half()
+            self.model.to(torch.float16)
+
+        encoded = self.model.encode_audio(x, chunked=self.chunked, iterate_batch=self.iterate_batch, **kwargs)
+
+        if self.model_half:
+            encoded = encoded.float()
+
+        return encoded / self.scale
+
+    def decode(self, z, **kwargs):
+        z = z * self.scale
+
+        if self.model_half:
+            z = z.half()
+            self.model.to(torch.float16)
+
+        decoded = self.model.decode_audio(z, chunked=self.chunked, iterate_batch=self.iterate_batch, **kwargs)
+
+        if self.model_half:
+            decoded = decoded.float()
+
+        return decoded
+    
+    def tokenize(self, x, **kwargs):
+        assert self.model.is_discrete, "Cannot tokenize with a continuous model"
+
+        _, info = self.model.encode(x, return_info = True, **kwargs)
+
+        return info[self.model.bottleneck.tokens_id]
+    
+    def decode_tokens(self, tokens, **kwargs):
+        assert self.model.is_discrete, "Cannot decode tokens with a continuous model"
+
+        return self.model.decode_tokens(tokens, **kwargs)
+    
+    def load_state_dict(self, state_dict, strict=True):
+        self.model.load_state_dict(state_dict, strict=strict)
+
+class WaveletPretransform(Pretransform):
+    def __init__(self, channels, levels, wavelet):
+        super().__init__(enable_grad=False, io_channels=channels, is_discrete=False)
+
+        from .wavelets import WaveletEncode1d, WaveletDecode1d
+
+        self.encoder = WaveletEncode1d(channels, levels, wavelet)
+        self.decoder = WaveletDecode1d(channels, levels, wavelet)
+
+        self.downsampling_ratio = 2 ** levels
+        self.io_channels = channels
+        self.encoded_channels = channels * self.downsampling_ratio
+    
+    def encode(self, x):
+        return self.encoder(x)
+    
+    def decode(self, z):
+        return self.decoder(z)
+    
+class PQMFPretransform(Pretransform):
+    def __init__(self, attenuation=100, num_bands=16):
+        # TODO: Fix PQMF to take in in-channels
+        super().__init__(enable_grad=False, io_channels=1, is_discrete=False)
+        from .pqmf import PQMF
+        self.pqmf = PQMF(attenuation, num_bands)
+
+
+    def encode(self, x):
+        # x is (Batch x Channels x Time)
+        x = self.pqmf.forward(x)
+        # pqmf.forward returns (Batch x Channels x Bands x Time)
+        # but Pretransform needs Batch x Channels x Time
+        # so concatenate channels and bands into one axis
+        return rearrange(x, "b c n t -> b (c n) t")
+
+    def decode(self, x):
+        # x is (Batch x (Channels Bands) x Time), convert back to (Batch x Channels x Bands x Time) 
+        x = rearrange(x, "b (c n) t -> b c n t", n=self.pqmf.num_bands)
+        # returns (Batch x Channels x Time) 
+        return self.pqmf.inverse(x)
+        
+class PretrainedDACPretransform(Pretransform):
+    def __init__(self, model_type="44khz", model_bitrate="8kbps", scale=1.0, quantize_on_decode: bool = True, chunked=True):
+        super().__init__(enable_grad=False, io_channels=1, is_discrete=True)
+        
+        import dac
+        
+        model_path = dac.utils.download(model_type=model_type, model_bitrate=model_bitrate)
+        
+        self.model = dac.DAC.load(model_path)
+
+        self.quantize_on_decode = quantize_on_decode
+
+        if model_type == "44khz":
+            self.downsampling_ratio = 512
+        else:
+            self.downsampling_ratio = 320
+
+        self.io_channels = 1
+
+        self.scale = scale
+
+        self.chunked = chunked
+
+        self.encoded_channels = self.model.latent_dim
+
+        self.num_quantizers = self.model.n_codebooks
+
+        self.codebook_size = self.model.codebook_size
+
+    def encode(self, x):
+
+        latents = self.model.encoder(x)
+
+        if self.quantize_on_decode:
+            output = latents
+        else:
+            z, _, _, _, _ = self.model.quantizer(latents, n_quantizers=self.model.n_codebooks)
+            output = z
+        
+        if self.scale != 1.0:
+            output = output / self.scale
+        
+        return output
+
+    def decode(self, z):
+        
+        if self.scale != 1.0:
+            z = z * self.scale
+
+        if self.quantize_on_decode:
+            z, _, _, _, _ = self.model.quantizer(z, n_quantizers=self.model.n_codebooks)
+
+        return self.model.decode(z)
+
+    def tokenize(self, x):
+        return self.model.encode(x)[1]
+    
+    def decode_tokens(self, tokens):
+        latents = self.model.quantizer.from_codes(tokens)
+        return self.model.decode(latents)
+    
+class AudiocraftCompressionPretransform(Pretransform):
+    def __init__(self, model_type="facebook/encodec_32khz", scale=1.0, quantize_on_decode: bool = True):
+        super().__init__(enable_grad=False, io_channels=1, is_discrete=True)
+        
+        try:
+            from audiocraft.models import CompressionModel
+        except ImportError:
+            raise ImportError("Audiocraft is not installed. Please install audiocraft to use Audiocraft models.")
+               
+        self.model = CompressionModel.get_pretrained(model_type)
+
+        self.quantize_on_decode = quantize_on_decode
+
+        self.downsampling_ratio = round(self.model.sample_rate / self.model.frame_rate)
+
+        self.sample_rate = self.model.sample_rate
+
+        self.io_channels = self.model.channels
+
+        self.scale = scale
+
+        #self.encoded_channels = self.model.latent_dim
+
+        self.num_quantizers = self.model.num_codebooks
+
+        self.codebook_size = self.model.cardinality
+
+        self.model.to(torch.float16).eval().requires_grad_(False)
+
+    def encode(self, x):
+
+        assert False, "Audiocraft compression models do not support continuous encoding"
+
+        # latents = self.model.encoder(x)
+
+        # if self.quantize_on_decode:
+        #     output = latents
+        # else:
+        #     z, _, _, _, _ = self.model.quantizer(latents, n_quantizers=self.model.n_codebooks)
+        #     output = z
+        
+        # if self.scale != 1.0:
+        #     output = output / self.scale
+        
+        # return output
+
+    def decode(self, z):
+        
+        assert False, "Audiocraft compression models do not support continuous decoding"
+
+        # if self.scale != 1.0:
+        #     z = z * self.scale
+
+        # if self.quantize_on_decode:
+        #     z, _, _, _, _ = self.model.quantizer(z, n_quantizers=self.model.n_codebooks)
+
+        # return self.model.decode(z)
+
+    def tokenize(self, x):
+        with torch.cuda.amp.autocast(enabled=False):
+            return self.model.encode(x.to(torch.float16))[0]
+    
+    def decode_tokens(self, tokens):
+        with torch.cuda.amp.autocast(enabled=False):
+            return self.model.decode(tokens)

vae_modules/stable_vae/models/utils.py

@@ -0,0 +1,51 @@
+import torch
+import torch.nn as nn
+from torchaudio import transforms as T
+
+
+class PadCrop(nn.Module):
+    def __init__(self, n_samples, randomize=True):
+        super().__init__()
+        self.n_samples = n_samples
+        self.randomize = randomize
+
+    def __call__(self, signal):
+        n, s = signal.shape
+        start = 0 if (not self.randomize) else torch.randint(0, max(0, s - self.n_samples) + 1, []).item()
+        end = start + self.n_samples
+        output = signal.new_zeros([n, self.n_samples])
+        output[:, :min(s, self.n_samples)] = signal[:, start:end]
+        return output
+
+
+def set_audio_channels(audio, target_channels):
+    if target_channels == 1:
+        # Convert to mono
+        audio = audio.mean(1, keepdim=True)
+    elif target_channels == 2:
+        # Convert to stereo
+        if audio.shape[1] == 1:
+            audio = audio.repeat(1, 2, 1)
+        elif audio.shape[1] > 2:
+            audio = audio[:, :2, :]
+    return audio
+
+def prepare_audio(audio, in_sr, target_sr, target_length, target_channels, device):
+    
+    audio = audio.to(device)
+
+    if in_sr != target_sr:
+        resample_tf = T.Resample(in_sr, target_sr).to(device)
+        audio = resample_tf(audio)
+
+    audio = PadCrop(target_length, randomize=False)(audio)
+
+    # Add batch dimension
+    if audio.dim() == 1:
+        audio = audio.unsqueeze(0).unsqueeze(0)
+    elif audio.dim() == 2:
+        audio = audio.unsqueeze(0)
+
+    audio = set_audio_channels(audio, target_channels)
+
+    return audio