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 dac.nn.layers import Snake1d from dac.nn.layers import WNConv1d from dac.nn.layers import WNConvTranspose1d from dac.nn.quantize import ResidualVectorQuantize from .encodec import SConv1d, SConvTranspose1d, SLSTM 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, causal: bool = False): super().__init__() conv1d_type = SConv1d# if causal else WNConv1d pad = ((7 - 1) * dilation) // 2 self.block = nn.Sequential( Snake1d(dim), conv1d_type(dim, dim, kernel_size=7, dilation=dilation, padding=pad, causal=causal, norm='weight_norm'), Snake1d(dim), conv1d_type(dim, dim, kernel_size=1, causal=causal, norm='weight_norm'), ) 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, causal: bool = False): super().__init__() conv1d_type = SConv1d# if causal else WNConv1d self.block = nn.Sequential( ResidualUnit(dim // 2, dilation=1, causal=causal), ResidualUnit(dim // 2, dilation=3, causal=causal), ResidualUnit(dim // 2, dilation=9, causal=causal), Snake1d(dim // 2), conv1d_type( dim // 2, dim, kernel_size=2 * stride, stride=stride, padding=math.ceil(stride / 2), causal=causal, norm='weight_norm', ), ) 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, causal: bool = False, lstm: int = 2, ): super().__init__() conv1d_type = SConv1d# if causal else WNConv1d # Create first convolution self.block = [conv1d_type(1, d_model, kernel_size=7, padding=3, causal=causal, norm='weight_norm')] # Create EncoderBlocks that double channels as they downsample by `stride` for stride in strides: d_model *= 2 self.block += [EncoderBlock(d_model, stride=stride, causal=causal)] # Add LSTM if needed self.use_lstm = lstm if lstm: self.block += [SLSTM(d_model, lstm)] # Create last convolution self.block += [ Snake1d(d_model), conv1d_type(d_model, d_latent, kernel_size=3, padding=1, causal=causal, norm='weight_norm'), ] # Wrap black into nn.Sequential self.block = nn.Sequential(*self.block) self.enc_dim = d_model def forward(self, x): return self.block(x) def reset_cache(self): # recursively find all submodules named SConv1d in self.block and use their reset_cache method def reset_cache(m): if isinstance(m, SConv1d) or isinstance(m, SLSTM): m.reset_cache() return for child in m.children(): reset_cache(child) reset_cache(self.block) class DecoderBlock(nn.Module): def __init__(self, input_dim: int = 16, output_dim: int = 8, stride: int = 1, causal: bool = False): super().__init__() conv1d_type = SConvTranspose1d #if causal else WNConvTranspose1d self.block = nn.Sequential( Snake1d(input_dim), conv1d_type( input_dim, output_dim, kernel_size=2 * stride, stride=stride, padding=math.ceil(stride / 2), causal=causal, norm='weight_norm' ), ResidualUnit(output_dim, dilation=1, causal=causal), ResidualUnit(output_dim, dilation=3, causal=causal), ResidualUnit(output_dim, dilation=9, causal=causal), ) def forward(self, x): return self.block(x) class Decoder(nn.Module): def __init__( self, input_channel, channels, rates, d_out: int = 1, causal: bool = False, lstm: int = 2, ): super().__init__() conv1d_type = SConv1d# if causal else WNConv1d # Add first conv layer layers = [conv1d_type(input_channel, channels, kernel_size=7, padding=3, causal=causal, norm='weight_norm')] if lstm: layers += [SLSTM(channels, num_layers=lstm)] # 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, causal=causal)] # Add final conv layer layers += [ Snake1d(output_dim), conv1d_type(output_dim, d_out, kernel_size=7, padding=3, causal=causal, norm='weight_norm'), 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, lstm: int = 2, causal: 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 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, causal=causal, lstm=lstm) 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, lstm=lstm, causal=causal, ) 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)