# coding=utf-8 # Copyright 2023 Meta Platforms, Inc. and affiliates, and the HuggingFace Inc. team. All rights reserved. # # Licensed under the Apache License, Version 2.0 (the "License"); # you may not use this file except in compliance with the License. # You may obtain a copy of the License at # # http://www.apache.org/licenses/LICENSE-2.0 # # Unless required by applicable law or agreed to in writing, software # distributed under the License is distributed on an "AS IS" BASIS, # WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. # See the License for the specific language governing permissions and # limitations under the License. """ PyTorch EnCodec model.""" import math from dataclasses import dataclass from typing import List, Optional, Tuple, Union import torch import torch.utils.checkpoint from torch import nn from ...modeling_utils import PreTrainedModel from ...utils import ( ModelOutput, add_start_docstrings, add_start_docstrings_to_model_forward, logging, replace_return_docstrings, ) from .configuration_encodec import EncodecConfig logger = logging.get_logger(__name__) # General docstring _CONFIG_FOR_DOC = "EncodecConfig" from ..deprecated._archive_maps import ENCODEC_PRETRAINED_MODEL_ARCHIVE_LIST # noqa: F401, E402 @dataclass class EncodecOutput(ModelOutput): """ Args: audio_codes (`torch.LongTensor` of shape `(batch_size, nb_chunks, chunk_length)`, *optional*): Discret code embeddings computed using `model.encode`. audio_values (`torch.FlaotTensor` of shape `(batch_size, sequence_length)`, *optional*) Decoded audio values, obtained using the decoder part of Encodec. """ audio_codes: torch.LongTensor = None audio_values: torch.FloatTensor = None @dataclass class EncodecEncoderOutput(ModelOutput): """ Args: audio_codes (`torch.LongTensor` of shape `(batch_size, nb_chunks, chunk_length)`, *optional*): Discret code embeddings computed using `model.encode`. audio_scales (`torch.Tensor` of shape `(batch_size, nb_chunks)`, *optional*): Scaling factor for each `audio_codes` input. This is used to unscale each chunk of audio when decoding. """ audio_codes: torch.LongTensor = None audio_scales: torch.FloatTensor = None @dataclass class EncodecDecoderOutput(ModelOutput): """ Args: audio_values (`torch.FloatTensor` of shape `(batch_size, segment_length)`, *optional*): Decoded audio values, obtained using the decoder part of Encodec. """ audio_values: torch.FloatTensor = None class EncodecConv1d(nn.Module): """Conv1d with asymmetric or causal padding and normalization.""" def __init__( self, config, in_channels: int, out_channels: int, kernel_size: int, stride: int = 1, dilation: int = 1 ): super().__init__() self.causal = config.use_causal_conv self.pad_mode = config.pad_mode self.norm_type = config.norm_type if self.norm_type not in ["weight_norm", "time_group_norm"]: raise ValueError( f'self.norm_type must be one of `"weight_norm"`, `"time_group_norm"`), got {self.norm_type}' ) # warn user on unusual setup between dilation and stride if stride > 1 and dilation > 1: logger.warning( "EncodecConv1d has been initialized with stride > 1 and dilation > 1" f" (kernel_size={kernel_size} stride={stride}, dilation={dilation})." ) self.conv = nn.Conv1d(in_channels, out_channels, kernel_size, stride, dilation=dilation) if self.norm_type == "weight_norm": self.conv = nn.utils.weight_norm(self.conv) elif self.norm_type == "time_group_norm": self.norm = nn.GroupNorm(1, out_channels) kernel_size = self.conv.kernel_size[0] stride = torch.tensor(self.conv.stride[0], dtype=torch.int64) dilation = self.conv.dilation[0] # Effective kernel size with dilations. kernel_size = torch.tensor((kernel_size - 1) * dilation + 1, dtype=torch.int64) self.register_buffer("stride", stride, persistent=False) self.register_buffer("kernel_size", kernel_size, persistent=False) self.register_buffer("padding_total", torch.tensor(kernel_size - stride, dtype=torch.int64), persistent=False) def _get_extra_padding_for_conv1d( self, hidden_states: torch.Tensor, ) -> torch.Tensor: """See `pad_for_conv1d`.""" length = hidden_states.shape[-1] n_frames = (length - self.kernel_size + self.padding_total) / self.stride + 1 n_frames = torch.ceil(n_frames).to(torch.int64) - 1 ideal_length = n_frames * self.stride + self.kernel_size - self.padding_total return ideal_length - length @staticmethod def _pad1d(hidden_states: torch.Tensor, paddings: Tuple[int, int], mode: str = "zero", value: float = 0.0): """Tiny wrapper around torch.nn.functional.pad, just to allow for reflect padding on small input. If this is the case, we insert extra 0 padding to the right before the reflection happens. """ length = hidden_states.shape[-1] padding_left, padding_right = paddings if not mode == "reflect": return nn.functional.pad(hidden_states, paddings, mode, value) max_pad = max(padding_left, padding_right) extra_pad = 0 if length <= max_pad: extra_pad = max_pad - length + 1 hidden_states = nn.functional.pad(hidden_states, (0, extra_pad)) padded = nn.functional.pad(hidden_states, paddings, mode, value) end = padded.shape[-1] - extra_pad return padded[..., :end] def forward(self, hidden_states): extra_padding = self._get_extra_padding_for_conv1d(hidden_states) if self.causal: # Left padding for causal hidden_states = self._pad1d(hidden_states, (self.padding_total, extra_padding), mode=self.pad_mode) else: # Asymmetric padding required for odd strides padding_right = self.padding_total // 2 padding_left = self.padding_total - padding_right hidden_states = self._pad1d( hidden_states, (padding_left, padding_right + extra_padding), mode=self.pad_mode ) hidden_states = self.conv(hidden_states) if self.norm_type == "time_group_norm": hidden_states = self.norm(hidden_states) return hidden_states class EncodecConvTranspose1d(nn.Module): """ConvTranspose1d with asymmetric or causal padding and normalization.""" def __init__(self, config, in_channels: int, out_channels: int, kernel_size: int, stride: int = 1): super().__init__() self.causal = config.use_causal_conv self.trim_right_ratio = config.trim_right_ratio self.norm_type = config.norm_type if self.norm_type not in ["weight_norm", "time_group_norm"]: raise ValueError( f'self.norm_type must be one of `"weight_norm"`, `"time_group_norm"`), got {self.norm_type}' ) self.conv = nn.ConvTranspose1d(in_channels, out_channels, kernel_size, stride) if config.norm_type == "weight_norm": self.conv = nn.utils.weight_norm(self.conv) elif config.norm_type == "time_group_norm": self.norm = nn.GroupNorm(1, out_channels) if not (self.causal or self.trim_right_ratio == 1.0): raise ValueError("`trim_right_ratio` != 1.0 only makes sense for causal convolutions") def forward(self, hidden_states): kernel_size = self.conv.kernel_size[0] stride = self.conv.stride[0] padding_total = kernel_size - stride hidden_states = self.conv(hidden_states) if self.norm_type == "time_group_norm": hidden_states = self.norm(hidden_states) # We will only trim fixed padding. Extra padding from `pad_for_conv1d` would be # removed at the very end, when keeping only the right length for the output, # as removing it here would require also passing the length at the matching layer # in the encoder. if self.causal: # Trim the padding on the right according to the specified ratio # if trim_right_ratio = 1.0, trim everything from right padding_right = math.ceil(padding_total * self.trim_right_ratio) else: # Asymmetric padding required for odd strides padding_right = padding_total // 2 padding_left = padding_total - padding_right # unpad end = hidden_states.shape[-1] - padding_right hidden_states = hidden_states[..., padding_left:end] return hidden_states class EncodecLSTM(nn.Module): """ LSTM without worrying about the hidden state, nor the layout of the data. Expects input as convolutional layout. """ def __init__(self, config, dimension): super().__init__() self.lstm = nn.LSTM(dimension, dimension, config.num_lstm_layers) def forward(self, hidden_states): hidden_states = hidden_states.permute(2, 0, 1) hidden_states = self.lstm(hidden_states)[0] + hidden_states hidden_states = hidden_states.permute(1, 2, 0) return hidden_states class EncodecResnetBlock(nn.Module): """ Residual block from SEANet model as used by EnCodec. """ def __init__(self, config: EncodecConfig, dim: int, dilations: List[int]): super().__init__() kernel_sizes = (config.residual_kernel_size, 1) if len(kernel_sizes) != len(dilations): raise ValueError("Number of kernel sizes should match number of dilations") hidden = dim // config.compress block = [] for i, (kernel_size, dilation) in enumerate(zip(kernel_sizes, dilations)): in_chs = dim if i == 0 else hidden out_chs = dim if i == len(kernel_sizes) - 1 else hidden block += [nn.ELU()] block += [EncodecConv1d(config, in_chs, out_chs, kernel_size, dilation=dilation)] self.block = nn.ModuleList(block) if config.use_conv_shortcut: self.shortcut = EncodecConv1d(config, dim, dim, kernel_size=1) else: self.shortcut = nn.Identity() def forward(self, hidden_states): residual = hidden_states for layer in self.block: hidden_states = layer(hidden_states) return self.shortcut(residual) + hidden_states class EncodecEncoder(nn.Module): """SEANet encoder as used by EnCodec.""" def __init__(self, config: EncodecConfig): super().__init__() model = [EncodecConv1d(config, config.audio_channels, config.num_filters, config.kernel_size)] scaling = 1 # Downsample to raw audio scale for ratio in reversed(config.upsampling_ratios): current_scale = scaling * config.num_filters # Add residual layers for j in range(config.num_residual_layers): model += [EncodecResnetBlock(config, current_scale, [config.dilation_growth_rate**j, 1])] # Add downsampling layers model += [nn.ELU()] model += [EncodecConv1d(config, current_scale, current_scale * 2, kernel_size=ratio * 2, stride=ratio)] scaling *= 2 model += [EncodecLSTM(config, scaling * config.num_filters)] model += [nn.ELU()] model += [EncodecConv1d(config, scaling * config.num_filters, config.hidden_size, config.last_kernel_size)] self.layers = nn.ModuleList(model) def forward(self, hidden_states): for layer in self.layers: hidden_states = layer(hidden_states) return hidden_states class EncodecDecoder(nn.Module): """SEANet decoder as used by EnCodec.""" def __init__(self, config: EncodecConfig): super().__init__() scaling = int(2 ** len(config.upsampling_ratios)) model = [EncodecConv1d(config, config.hidden_size, scaling * config.num_filters, config.kernel_size)] model += [EncodecLSTM(config, scaling * config.num_filters)] # Upsample to raw audio scale for ratio in config.upsampling_ratios: current_scale = scaling * config.num_filters # Add upsampling layers model += [nn.ELU()] model += [ EncodecConvTranspose1d(config, current_scale, current_scale // 2, kernel_size=ratio * 2, stride=ratio) ] # Add residual layers for j in range(config.num_residual_layers): model += [EncodecResnetBlock(config, current_scale // 2, (config.dilation_growth_rate**j, 1))] scaling //= 2 # Add final layers model += [nn.ELU()] model += [EncodecConv1d(config, config.num_filters, config.audio_channels, config.last_kernel_size)] self.layers = nn.ModuleList(model) def forward(self, hidden_states): for layer in self.layers: hidden_states = layer(hidden_states) return hidden_states class EncodecEuclideanCodebook(nn.Module): """Codebook with Euclidean distance.""" def __init__(self, config: EncodecConfig): super().__init__() embed = torch.zeros(config.codebook_size, config.codebook_dim) self.codebook_size = config.codebook_size self.register_buffer("inited", torch.Tensor([True])) self.register_buffer("cluster_size", torch.zeros(config.codebook_size)) self.register_buffer("embed", embed) self.register_buffer("embed_avg", embed.clone()) def quantize(self, hidden_states): embed = self.embed.t() scaled_states = hidden_states.pow(2).sum(1, keepdim=True) dist = -(scaled_states - 2 * hidden_states @ embed + embed.pow(2).sum(0, keepdim=True)) embed_ind = dist.max(dim=-1).indices return embed_ind def encode(self, hidden_states): shape = hidden_states.shape # pre-process hidden_states = hidden_states.reshape((-1, shape[-1])) # quantize embed_ind = self.quantize(hidden_states) # post-process embed_ind = embed_ind.view(*shape[:-1]) return embed_ind def decode(self, embed_ind): quantize = nn.functional.embedding(embed_ind, self.embed) return quantize class EncodecVectorQuantization(nn.Module): """ Vector quantization implementation. Currently supports only euclidean distance. """ def __init__(self, config: EncodecConfig): super().__init__() self.codebook = EncodecEuclideanCodebook(config) def encode(self, hidden_states): hidden_states = hidden_states.permute(0, 2, 1) embed_in = self.codebook.encode(hidden_states) return embed_in def decode(self, embed_ind): quantize = self.codebook.decode(embed_ind) quantize = quantize.permute(0, 2, 1) return quantize class EncodecResidualVectorQuantizer(nn.Module): """Residual Vector Quantizer.""" def __init__(self, config: EncodecConfig): super().__init__() self.codebook_size = config.codebook_size self.frame_rate = config.frame_rate self.num_quantizers = config.num_quantizers self.layers = nn.ModuleList([EncodecVectorQuantization(config) for _ in range(config.num_quantizers)]) def get_num_quantizers_for_bandwidth(self, bandwidth: Optional[float] = None) -> int: """Return num_quantizers based on specified target bandwidth.""" bw_per_q = math.log2(self.codebook_size) * self.frame_rate num_quantizers = self.num_quantizers if bandwidth is not None and bandwidth > 0.0: num_quantizers = int(max(1, math.floor(bandwidth * 1000 / bw_per_q))) return num_quantizers def encode(self, embeddings: torch.Tensor, bandwidth: Optional[float] = None) -> torch.Tensor: """ Encode a given input tensor with the specified frame rate at the given bandwidth. The RVQ encode method sets the appropriate number of quantizers to use and returns indices for each quantizer. """ num_quantizers = self.get_num_quantizers_for_bandwidth(bandwidth) residual = embeddings all_indices = [] for layer in self.layers[:num_quantizers]: indices = layer.encode(residual) quantized = layer.decode(indices) residual = residual - quantized all_indices.append(indices) out_indices = torch.stack(all_indices) return out_indices def decode(self, codes: torch.Tensor) -> torch.Tensor: """Decode the given codes to the quantized representation.""" quantized_out = torch.tensor(0.0, device=codes.device) for i, indices in enumerate(codes): layer = self.layers[i] quantized = layer.decode(indices) quantized_out = quantized_out + quantized return quantized_out class EncodecPreTrainedModel(PreTrainedModel): """ An abstract class to handle weights initialization and a simple interface for downloading and loading pretrained models. """ config_class = EncodecConfig base_model_prefix = "encodec" main_input_name = "input_values" def _init_weights(self, module): """Initialize the weights""" if isinstance(module, nn.Linear): module.weight.data.normal_(mean=0.0, std=self.config.initializer_range) if module.bias is not None: module.bias.data.zero_() elif isinstance(module, (nn.LayerNorm, nn.GroupNorm)): module.bias.data.zero_() module.weight.data.fill_(1.0) elif isinstance(module, nn.Conv1d): nn.init.kaiming_normal_(module.weight) if module.bias is not None: k = math.sqrt(module.groups / (module.in_channels * module.kernel_size[0])) nn.init.uniform_(module.bias, a=-k, b=k) elif isinstance(module, nn.Embedding): module.weight.data.normal_(mean=0.0, std=self.config.initializer_range) if module.padding_idx is not None: module.weight.data[module.padding_idx].zero_() elif isinstance(module, nn.LSTM): for name, param in module.named_parameters(): if "weight" in name: nn.init.xavier_uniform_(param) elif "bias" in name: nn.init.constant_(param, 0.0) ENCODEC_START_DOCSTRING = r""" This model inherits from [`PreTrainedModel`]. Check the superclass documentation for the generic methods the library implements for all its model (such as downloading or saving, resizing the input embeddings, pruning heads etc.) This model is also a PyTorch [torch.nn.Module](https://pytorch.org/docs/stable/nn.html#torch.nn.Module) subclass. Use it as a regular PyTorch Module and refer to the PyTorch documentation for all matter related to general usage and behavior. Parameters: config ([`EncodecConfig`]): Model configuration class with all the parameters of the model. Initializing with a config file does not load the weights associated with the model, only the configuration. Check out the [`~PreTrainedModel.from_pretrained`] method to load the model weights. """ ENCODEC_INPUTS_DOCSTRING = r""" Args: input_values (`torch.FloatTensor` of shape `(batch_size, channels, sequence_length)`, *optional*): Raw audio input converted to Float and padded to the approriate length in order to be encoded using chunks of length self.chunk_length and a stride of `config.chunk_stride`. padding_mask (`torch.BoolTensor` of shape `(batch_size, channels, sequence_length)`, *optional*): Mask to avoid computing scaling factors on padding token indices (can we avoid computing conv on these+). Mask values selected in `[0, 1]`: - 1 for tokens that are **not masked**, - 0 for tokens that are **masked**. `padding_mask` should always be passed, unless the input was truncated or not padded. This is because in order to process tensors effectively, the input audio should be padded so that `input_length % stride = step` with `step = chunk_length-stride`. This ensures that all chunks are of the same shape bandwidth (`float`, *optional*): The target bandwidth. Must be one of `config.target_bandwidths`. If `None`, uses the smallest possible bandwidth. bandwidth is represented as a thousandth of what it is, e.g. 6kbps bandwidth is represented as `bandwidth == 6.0` audio_codes (`torch.LongTensor` of shape `(batch_size, nb_chunks, chunk_length)`, *optional*): Discret code embeddings computed using `model.encode`. audio_scales (`torch.Tensor` of shape `(batch_size, nb_chunks)`, *optional*): Scaling factor for each `audio_codes` input. return_dict (`bool`, *optional*): Whether or not to return a [`~utils.ModelOutput`] instead of a plain tuple. """ @add_start_docstrings( "The EnCodec neural audio codec model.", ENCODEC_START_DOCSTRING, ) class EncodecModel(EncodecPreTrainedModel): def __init__(self, config: EncodecConfig): super().__init__(config) self.config = config self.encoder = EncodecEncoder(config) self.decoder = EncodecDecoder(config) self.quantizer = EncodecResidualVectorQuantizer(config) self.bits_per_codebook = int(math.log2(self.config.codebook_size)) if 2**self.bits_per_codebook != self.config.codebook_size: raise ValueError("The codebook_size must be a power of 2.") # Initialize weights and apply final processing self.post_init() def get_encoder(self): return self.encoder def get_decoder(self): return self.decoder def _encode_frame( self, input_values: torch.Tensor, bandwidth: float, padding_mask: int ) -> Tuple[torch.Tensor, Optional[torch.Tensor]]: """ Encodes the given input using the underlying VQVAE. If `config.normalize` is set to `True` the input is first normalized. The padding mask is required to compute the correct scale. """ length = input_values.shape[-1] duration = length / self.config.sampling_rate if self.config.chunk_length_s is not None and duration > 1e-5 + self.config.chunk_length_s: raise RuntimeError(f"Duration of frame ({duration}) is longer than chunk {self.config.chunk_length_s}") scale = None if self.config.normalize: # if the padding is non zero input_values = input_values * padding_mask mono = torch.sum(input_values, 1, keepdim=True) / input_values.shape[1] scale = mono.pow(2).mean(dim=-1, keepdim=True).sqrt() + 1e-8 input_values = input_values / scale embeddings = self.encoder(input_values) codes = self.quantizer.encode(embeddings, bandwidth) codes = codes.transpose(0, 1) return codes, scale def encode( self, input_values: torch.Tensor, padding_mask: torch.Tensor = None, bandwidth: Optional[float] = None, return_dict: Optional[bool] = None, ) -> Union[Tuple[torch.Tensor, Optional[torch.Tensor]], EncodecEncoderOutput]: """ Encodes the input audio waveform into discrete codes. Args: input_values (`torch.Tensor` of shape `(batch_size, channels, sequence_length)`): Float values of the input audio waveform. padding_mask (`torch.Tensor` of shape `(batch_size, channels, sequence_length)`): Padding mask used to pad the `input_values`. bandwidth (`float`, *optional*): The target bandwidth. Must be one of `config.target_bandwidths`. If `None`, uses the smallest possible bandwidth. bandwidth is represented as a thousandth of what it is, e.g. 6kbps bandwidth is represented as bandwidth == 6.0 Returns: A list of frames containing the discrete encoded codes for the input audio waveform, along with rescaling factors for each chunk when `normalize` is True. Each frames is a tuple `(codebook, scale)`, with `codebook` of shape `[batch_size, num_codebooks, frames]`. """ return_dict = return_dict if return_dict is not None else self.config.return_dict if bandwidth is None: bandwidth = self.config.target_bandwidths[0] if bandwidth not in self.config.target_bandwidths: raise ValueError( f"This model doesn't support the bandwidth {bandwidth}. " f"Select one of {self.config.target_bandwidths}." ) _, channels, input_length = input_values.shape if channels < 1 or channels > 2: raise ValueError(f"Number of audio channels must be 1 or 2, but got {channels}") chunk_length = self.config.chunk_length if chunk_length is None: chunk_length = input_length stride = input_length else: stride = self.config.chunk_stride if padding_mask is None: padding_mask = torch.ones_like(input_values).bool() encoded_frames = [] scales = [] step = chunk_length - stride if (input_length % stride) - step != 0: raise ValueError( "The input length is not properly padded for batched chunked decoding. Make sure to pad the input correctly." ) for offset in range(0, input_length - step, stride): mask = padding_mask[..., offset : offset + chunk_length].bool() frame = input_values[:, :, offset : offset + chunk_length] encoded_frame, scale = self._encode_frame(frame, bandwidth, mask) encoded_frames.append(encoded_frame) scales.append(scale) encoded_frames = torch.stack(encoded_frames) if not return_dict: return (encoded_frames, scales) return EncodecEncoderOutput(encoded_frames, scales) @staticmethod def _linear_overlap_add(frames: List[torch.Tensor], stride: int): # Generic overlap add, with linear fade-in/fade-out, supporting complex scenario # e.g., more than 2 frames per position. # The core idea is to use a weight function that is a triangle, # with a maximum value at the middle of the chunk. # We use this weighting when summing the frames, and divide by the sum of weights # for each positions at the end. Thus: # - if a frame is the only one to cover a position, the weighting is a no-op. # - if 2 frames cover a position: # ... ... # / \/ \ # / /\ \ # S T , i.e. S offset of second frame starts, T end of first frame. # Then the weight function for each one is: (t - S), (T - t), with `t` a given offset. # After the final normalization, the weight of the second frame at position `t` is # (t - S) / (t - S + (T - t)) = (t - S) / (T - S), which is exactly what we want. # # - if more than 2 frames overlap at a given point, we hope that by induction # something sensible happens. if len(frames) == 0: raise ValueError("`frames` cannot be an empty list.") device = frames[0].device dtype = frames[0].dtype shape = frames[0].shape[:-1] total_size = stride * (len(frames) - 1) + frames[-1].shape[-1] frame_length = frames[0].shape[-1] time_vec = torch.linspace(0, 1, frame_length + 2, device=device, dtype=dtype)[1:-1] weight = 0.5 - (time_vec - 0.5).abs() sum_weight = torch.zeros(total_size, device=device, dtype=dtype) out = torch.zeros(*shape, total_size, device=device, dtype=dtype) offset: int = 0 for frame in frames: frame_length = frame.shape[-1] out[..., offset : offset + frame_length] += weight[:frame_length] * frame sum_weight[offset : offset + frame_length] += weight[:frame_length] offset += stride if sum_weight.min() == 0: raise ValueError(f"`sum_weight` minimum element must be bigger than zero: {sum_weight}`") return out / sum_weight def _decode_frame(self, codes: torch.Tensor, scale: Optional[torch.Tensor] = None) -> torch.Tensor: codes = codes.transpose(0, 1) embeddings = self.quantizer.decode(codes) outputs = self.decoder(embeddings) if scale is not None: outputs = outputs * scale.view(-1, 1, 1) return outputs def decode( self, audio_codes: torch.Tensor, audio_scales: torch.Tensor, padding_mask: Optional[torch.Tensor] = None, return_dict: Optional[bool] = None, ) -> Union[Tuple[torch.Tensor, torch.Tensor], EncodecDecoderOutput]: """ Decodes the given frames into an output audio waveform. Note that the output might be a bit bigger than the input. In that case, any extra steps at the end can be trimmed. Args: audio_codes (`torch.LongTensor` of shape `(batch_size, nb_chunks, chunk_length)`, *optional*): Discret code embeddings computed using `model.encode`. audio_scales (`torch.Tensor` of shape `(batch_size, nb_chunks)`, *optional*): Scaling factor for each `audio_codes` input. padding_mask (`torch.Tensor` of shape `(batch_size, channels, sequence_length)`): Padding mask used to pad the `input_values`. return_dict (`bool`, *optional*): Whether or not to return a [`~utils.ModelOutput`] instead of a plain tuple. """ return_dict = return_dict or self.config.return_dict chunk_length = self.config.chunk_length if chunk_length is None: if len(audio_codes) != 1: raise ValueError(f"Expected one frame, got {len(audio_codes)}") audio_values = self._decode_frame(audio_codes[0], audio_scales[0]) else: decoded_frames = [] for frame, scale in zip(audio_codes, audio_scales): frames = self._decode_frame(frame, scale) decoded_frames.append(frames) audio_values = self._linear_overlap_add(decoded_frames, self.config.chunk_stride or 1) # truncate based on padding mask if padding_mask is not None and padding_mask.shape[-1] < audio_values.shape[-1]: audio_values = audio_values[..., : padding_mask.shape[-1]] if not return_dict: return (audio_values,) return EncodecDecoderOutput(audio_values) @add_start_docstrings_to_model_forward(ENCODEC_INPUTS_DOCSTRING) @replace_return_docstrings(output_type=EncodecOutput, config_class=_CONFIG_FOR_DOC) def forward( self, input_values: torch.Tensor, padding_mask: Optional[torch.Tensor] = None, bandwidth: Optional[float] = None, audio_codes: Optional[torch.Tensor] = None, audio_scales: Optional[torch.Tensor] = None, return_dict: Optional[bool] = None, ) -> Union[Tuple[torch.Tensor, torch.Tensor], EncodecOutput]: r""" Returns: Examples: ```python >>> from datasets import load_dataset >>> from transformers import AutoProcessor, EncodecModel >>> dataset = load_dataset("hf-internal-testing/ashraq-esc50-1-dog-example") >>> audio_sample = dataset["train"]["audio"][0]["array"] >>> model_id = "facebook/encodec_24khz" >>> model = EncodecModel.from_pretrained(model_id) >>> processor = AutoProcessor.from_pretrained(model_id) >>> inputs = processor(raw_audio=audio_sample, return_tensors="pt") >>> outputs = model(**inputs) >>> audio_codes = outputs.audio_codes >>> audio_values = outputs.audio_values ```""" return_dict = return_dict or self.config.return_dict if padding_mask is None: padding_mask = torch.ones_like(input_values).bool() if audio_codes is not None and audio_scales is None: raise ValueError("You specified `audio_codes` but did not specify the `audio_scales`") if audio_scales is not None and audio_codes is None: raise ValueError("You specified `audio_scales` but did not specify the `audio_codes`") if audio_scales is None and audio_codes is None: audio_codes, audio_scales = self.encode(input_values, padding_mask, bandwidth, False) audio_values = self.decode(audio_codes, audio_scales, padding_mask, return_dict=return_dict)[0] if not return_dict: return (audio_codes, audio_values) return EncodecOutput(audio_codes=audio_codes, audio_values=audio_values)