import math from dataclasses import dataclass from pathlib import Path from typing import Union, List import numpy as np import torch import tqdm from audiotools import AudioSignal from audiotools.ml import BaseModel from torch import nn from torch.nn.utils import weight_norm 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() audio_signal = audio_signal.to_mono() 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 if obj.input_db is not None: recons.normalize(obj.input_db) resample_fn(obj.sample_rate) if obj.original_length is not None: recons = recons[..., : obj.original_length] loudness_fn() recons.audio_data = recons.audio_data.reshape( -1, obj.channels, obj.original_length ) else: loudness_fn() self.padding = original_padding return recons 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): # print(f"{x.shape = } {x.device = } {x.dtype = }") # print(f"{self.alpha.shape = } {self.alpha.device = } {self.alpha.dtype = }") return snake(x, self.alpha) import torch.nn.functional as F from einops import rearrange 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) class AbstractDistribution: def sample(self): raise NotImplementedError() def mode(self): raise NotImplementedError() class DiracDistribution(AbstractDistribution): def __init__(self, value): self.value = value def sample(self): return self.value def mode(self): return self.value class DiagonalGaussianDistribution(object): def __init__(self, parameters, deterministic=False): self.parameters = parameters self.mean, self.logvar = torch.chunk(parameters, 2, dim=1) self.logvar = torch.clamp(self.logvar, -30.0, 20.0) self.deterministic = deterministic self.std = torch.exp(0.5 * self.logvar) self.var = torch.exp(self.logvar) if self.deterministic: self.var = self.std = torch.zeros_like(self.mean).to(device=self.parameters.device) def sample(self): x = self.mean + self.std * torch.randn(self.mean.shape).to(device=self.parameters.device) return x def kl(self, other=None): if self.deterministic: return torch.Tensor([0.0]) else: if other is None: return 0.5 * torch.mean( torch.pow(self.mean, 2) + self.var - 1.0 - self.logvar, dim=[1, 2], ) else: return 0.5 * torch.mean( torch.pow(self.mean - other.mean, 2) / other.var + self.var / other.var - 1.0 - self.logvar + other.logvar, dim=[1, 2], ) def nll(self, sample, dims=[1, 2]): if self.deterministic: return torch.Tensor([0.0]) logtwopi = np.log(2.0 * np.pi) return 0.5 * torch.sum( logtwopi + self.logvar + torch.pow(sample - self.mean, 2) / self.var, dim=dims, ) def mode(self): return self.mean def normal_kl(mean1, logvar1, mean2, logvar2): """ source: https://github.com/openai/guided-diffusion/blob/27c20a8fab9cb472df5d6bdd6c8d11c8f430b924/guided_diffusion/losses.py#L12 Compute the KL divergence between two gaussians. Shapes are automatically broadcasted, so batches can be compared to scalars, among other use cases. """ tensor = None for obj in (mean1, logvar1, mean2, logvar2): if isinstance(obj, torch.Tensor): tensor = obj break assert tensor is not None, "at least one argument must be a Tensor" # Force variances to be Tensors. Broadcasting helps convert scalars to # Tensors, but it does not work for torch.exp(). logvar1, logvar2 = [x if isinstance(x, torch.Tensor) else torch.tensor(x).to(tensor) for x in (logvar1, logvar2)] return 0.5 * ( -1.0 + logvar2 - logvar1 + torch.exp(logvar1 - logvar2) + ((mean1 - mean2) ** 2) * torch.exp(-logvar2) ) 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), output_padding=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, continuous: bool = False, ): 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 self.continuous = continuous 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) if not continuous: 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, ) else: self.quant_conv = torch.nn.Conv1d(latent_dim, 2 * latent_dim, 1) self.post_quant_conv = torch.nn.Conv1d(latent_dim, latent_dim, 1) self.decoder = Decoder( latent_dim, decoder_dim, decoder_rates, ) self.sample_rate = sample_rate self.apply(init_weights) self.delay = self.get_delay() @property def dtype(self): """Get the dtype of the model parameters.""" # Return the dtype of the first parameter found for param in self.parameters(): return param.dtype return torch.float32 # fallback @property def device(self): """Get the device of the model parameters.""" # Return the device of the first parameter found for param in self.parameters(): return param.device return torch.device('cpu') # fallback 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) # [B x D x T] if not self.continuous: z, codes, latents, commitment_loss, codebook_loss = self.quantizer(z, n_quantizers) else: z = self.quant_conv(z) # [B x 2D x T] z = DiagonalGaussianDistribution(z) codes, latents, commitment_loss, codebook_loss = None, None, 0, 0 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. """ if not self.continuous: audio = self.decoder(z) else: z = self.post_quant_conv(z) audio = self.decoder(z) return audio 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) if not self.continuous: 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, } else: posterior, _, _, _, _ = self.encode(audio_data, n_quantizers) z = posterior.sample() x = self.decode(z) kl_loss = posterior.kl() kl_loss = kl_loss.mean() return { "audio": x[..., :length], "z": z, "kl_loss": kl_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)