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| import math |
| import random |
| from collections import OrderedDict |
| from dataclasses import dataclass |
| from typing import List, Optional, Set, Tuple |
|
|
| import torch |
| import torch.distributed |
| from omegaconf import DictConfig, ListConfig, open_dict |
| from torch import nn |
|
|
| from nemo.collections.asr.models.configs import CacheAwareStreamingConfig |
| from nemo.collections.asr.parts.mixins.streaming import StreamingEncoder |
| from nemo.collections.asr.parts.submodules.causal_convs import CausalConv1D |
| from nemo.collections.asr.parts.submodules.conformer_modules import ConformerLayer |
| from nemo.collections.asr.parts.submodules.multi_head_attention import ( |
| LocalAttRelPositionalEncoding, |
| MultiHeadAttention, |
| PositionalEncoding, |
| RelPositionalEncoding, |
| RelPositionMultiHeadAttention, |
| RelPositionMultiHeadAttentionLongformer, |
| ) |
| from nemo.collections.asr.parts.submodules.subsampling import ( |
| ConvSubsampling, |
| StackingSubsampling, |
| SubsamplingReductionModule, |
| ) |
| from nemo.collections.asr.parts.utils import adapter_utils |
| from nemo.collections.asr.parts.utils.regularization_utils import compute_stochastic_depth_drop_probs |
| from nemo.core.classes.common import typecheck |
| from nemo.core.classes.exportable import Exportable |
| from nemo.core.classes.mixins import AccessMixin, adapter_mixins |
| from nemo.core.classes.module import NeuralModule |
| from nemo.core.neural_types import ( |
| AcousticEncodedRepresentation, |
| BoolType, |
| ChannelType, |
| LengthsType, |
| NeuralType, |
| SpectrogramType, |
| ) |
| from nemo.utils import logging |
|
|
| __all__ = ['ConformerEncoder', 'ConformerMultiLayerFeatureExtractor'] |
|
|
|
|
| class ConformerEncoder(NeuralModule, StreamingEncoder, Exportable, AccessMixin): |
| """ |
| The encoder for ASR model of Conformer. |
| Based on this paper: |
| 'Conformer: Convolution-augmented Transformer for Speech Recognition' by Anmol Gulati et al. |
| https://arxiv.org/abs/2005.08100 |
| |
| Args: |
| feat_in (int): the size of feature channels |
| n_layers (int): number of layers of ConformerBlock |
| d_model (int): the hidden size of the model |
| feat_out (int): the size of the output features |
| Defaults to -1 (means feat_out is d_model) |
| subsampling (str): the method of subsampling: |
| choices = ['vggnet', 'striding', 'dw-striding', 'stacking', 'stacking_norm'] |
| Defaults to striding. |
| subsampling_factor (int): the subsampling factor which should be power of 2 |
| Defaults to 4. |
| subsampling_conv_chunking_factor(int): optionally, force chunk inputs (helpful for large inputs) |
| Should be power of 2, 1 (auto-chunking, default), or -1 (no chunking) |
| subsampling_conv_channels (int): the size of the convolutions in the subsampling module |
| Defaults to -1 which would set it to d_model. |
| reduction (str, Optional): the method of reduction, choices=['pooling', 'striding']. If no value |
| is passed, then no reduction is performed and the models runs with the original 4x subsampling. |
| reduction_position (int, Optional): the index of the layer to apply reduction. If -1, apply reduction |
| at the end. |
| reduction_factor (int): the reduction factor which should be either 1 or a power of 2 |
| Defaults to 1. |
| ff_expansion_factor (int): the expansion factor in feed forward layers |
| Defaults to 4. |
| self_attention_model (str): the type of the attention layer and positional encoding. |
| |
| 'rel_pos': |
| relative positional embedding and Transformer-XL |
| 'rel_pos_local_attn': |
| relative positional embedding and Transformer-XL with local attention using |
| overlapping chunks. Attention context is determined by att_context_size parameter. |
| 'abs_pos': |
| absolute positional embedding and Transformer |
| |
| Default is rel_pos. |
| pos_emb_max_len (int): the maximum length of positional embeddings |
| Defaults to 5000 |
| n_heads (int): number of heads in multi-headed attention layers |
| Defaults to 4. |
| att_context_size (List[Union[List[int],int]]): specifies the context sizes on each side. |
| Each context size should be a list of two integers like `[100, 100]`. |
| A list of context sizes like `[[100,100]`, `[100,50]]` can also be passed. -1 means unlimited context. |
| Defaults to `[-1, -1]` |
| att_context_probs (List[float]): a list of probabilities of each one of the att_context_size |
| when a list of them is passed. If not specified, uniform distribution is being used. |
| Defaults to None |
| att_chunk_context_size (List[List[int]]): specifies the context sizes for unified (offline/streaming) ASR training. |
| It defines the range of Left, Middle, and Right context sizes for the attention mechanism. |
| At each streaming step, the context size is sampled from the range of Left, Middle, and Right context sizes. |
| Example: att_chunk_context_size=[[70],[1,2,7,13],[0,1,3,7,13]] -> sampling -> [70, 2, 3] -> attention mask generation |
| att_context_style (str): 'regular', 'chunked_limited', or 'chunked_limited_with_rc'. |
| Defaults to 'regular' |
| xscaling (bool): enables scaling the inputs to the multi-headed attention layers by `sqrt(d_model)`. |
| Defaults to True. |
| untie_biases (bool): whether to not share (untie) the bias weights between layers of Transformer-XL |
| Defaults to True. |
| conv_kernel_size (int): the size of the convolutions in the convolutional modules |
| Defaults to 31. |
| conv_norm_type (str): the type of the normalization in the convolutional modules |
| Defaults to 'batch_norm'. |
| conv_context_size (list): it can be"causal" or a list of two integers |
| while `conv_context_size[0]+conv_context_size[1]+1==conv_kernel_size`. |
| `None` means `[(conv_kernel_size-1)//2`, `(conv_kernel_size-1)//2]`, and 'causal' means |
| `[(conv_kernel_size-1), 0]`. |
| Defaults to None. |
| conv_context_style (str): 'regular' or 'dcc' |
| DCC - Dynamic Chunked Convolution that is used for unified ASR training. |
| Defaults to 'regular'. |
| conv_dual_mode (bool): specifies if convolution should be dual mode when dual_offline mode is being used. |
| When enables, the left half of the convolution kernel would get masked in streaming cases. |
| Defaults to False. |
| use_bias (bool): Use bias in all Linear and Conv1d layers from each ConformerLayer to improve |
| activation flow and stabilize training of huge models. |
| Defaults to True. |
| dropout (float): the dropout rate used in all layers except the attention layers |
| Defaults to 0.1. |
| dropout_pre_encoder (float): the dropout rate used before the encoder |
| Defaults to 0.1. |
| dropout_emb (float): the dropout rate used for the positional embeddings |
| Defaults to 0.1. |
| dropout_att (float): the dropout rate used for the attention layer |
| Defaults to 0.0. |
| stochastic_depth_drop_prob (float): if non-zero, will randomly drop |
| layers during training. The higher this value, the more often layers |
| are dropped. Defaults to 0.0. |
| stochastic_depth_mode (str): can be either "linear" or "uniform". If |
| set to "uniform", all layers have the same probability of drop. If |
| set to "linear", the drop probability grows linearly from 0 for the |
| first layer to the desired value for the final layer. Defaults to |
| "linear". |
| stochastic_depth_start_layer (int): starting layer for stochastic depth. |
| All layers before this will never be dropped. Note that drop |
| probability will be adjusted accordingly if mode is "linear" when |
| start layer is > 1. Defaults to 1. |
| global_tokens (int): number of tokens to be used for global attention. |
| Only relevant if self_attention_model is 'rel_pos_local_attn'. |
| Defaults to 0. |
| global_tokens_spacing (int): how far apart the global tokens are |
| Defaults to 1. |
| global_attn_separate (bool): whether the q, k, v layers used for global tokens should be separate. |
| Defaults to False. |
| use_pytorch_sdpa (bool): use torch sdpa instead of manual attention. |
| Defaults to False. |
| use_pytorch_sdpa_backends (list[str]): list of backend names to use in sdpa. |
| None or empty list means all backends. e.g. ["MATH"] |
| Defaults to None. |
| bypass_pre_encode: if True, skip the pre-encoder module and the `audio_signal` should be pre-encoded |
| embeddings. The `audio_signal` input supports two formats depending on the `bypass_pre_encode` |
| boolean flag. This determines the required format of the input variable `audio_signal`. |
| Defaults to `bypass_pre_encode=False`. `bypass_pre_encode=True` is used for the cases |
| where frame-level, context-independent embeddings are needed to be saved or reused. |
| (e.g., speaker cache in streaming speaker diarization) |
| sync_max_audio_length (bool): when true, performs NCCL all_reduce to allocate the same amount of memory for |
| positional encoding buffers on all GPUs. Disabling this setting may help with deadlocks in certain |
| scenarios such as model parallelism, or generally when this module is not being ran on some GPUs |
| as a part of the training step. |
| """ |
|
|
| def input_example(self, max_batch=1, max_dim=256): |
| """ |
| Generates input examples for tracing etc. |
| Returns: |
| A tuple of input examples. |
| """ |
| dev = next(self.parameters()).device |
| if self.export_cache_support: |
| window_size = max_dim |
| if self.streaming_cfg is not None: |
| if isinstance(self.streaming_cfg.chunk_size, list): |
| chunk_size = self.streaming_cfg.chunk_size[1] |
| else: |
| chunk_size = self.streaming_cfg.chunk_size |
| if isinstance(self.streaming_cfg.pre_encode_cache_size, list): |
| pre_encode_cache_size = self.streaming_cfg.pre_encode_cache_size[1] |
| else: |
| pre_encode_cache_size = self.streaming_cfg.pre_encode_cache_size |
| window_size = chunk_size + pre_encode_cache_size |
| input_example = torch.randn(max_batch, self._feat_in, window_size, device=dev) |
| input_example_length = torch.randint( |
| window_size // 4, window_size, (max_batch,), device=dev, dtype=torch.int64 |
| ) |
| cache_last_channel, cache_last_time, cache_last_channel_len = self.get_initial_cache_state( |
| batch_size=max_batch, device=dev, max_dim=max_dim |
| ) |
| all_input_example = tuple( |
| [ |
| input_example, |
| input_example_length, |
| cache_last_channel.transpose(0, 1), |
| cache_last_time.transpose(0, 1), |
| cache_last_channel_len, |
| ] |
| ) |
| else: |
| input_example = torch.randn(max_batch, self._feat_in, max_dim, device=dev) |
| input_example_length = torch.randint(max_dim // 4, max_dim, (max_batch,), device=dev, dtype=torch.int64) |
| all_input_example = tuple([input_example, input_example_length]) |
|
|
| return all_input_example |
|
|
| @property |
| def input_types(self): |
| """Returns definitions of module input ports.""" |
| return OrderedDict( |
| { |
| "audio_signal": NeuralType(('B', 'D', 'T'), SpectrogramType()), |
| "length": NeuralType(tuple('B'), LengthsType()), |
| "cache_last_channel": NeuralType(('D', 'B', 'T', 'D'), ChannelType(), optional=True), |
| "cache_last_time": NeuralType(('D', 'B', 'D', 'T'), ChannelType(), optional=True), |
| "cache_last_channel_len": NeuralType(tuple('B'), LengthsType(), optional=True), |
| "bypass_pre_encode": NeuralType(tuple(), BoolType(), optional=True), |
| } |
| ) |
|
|
| @property |
| def input_types_for_export(self): |
| """Returns definitions of module input ports.""" |
| return OrderedDict( |
| { |
| "audio_signal": NeuralType(('B', 'D', 'T'), SpectrogramType()), |
| "length": NeuralType(tuple('B'), LengthsType()), |
| "cache_last_channel": NeuralType(('B', 'D', 'T', 'D'), ChannelType(), optional=True), |
| "cache_last_time": NeuralType(('B', 'D', 'D', 'T'), ChannelType(), optional=True), |
| "cache_last_channel_len": NeuralType(tuple('B'), LengthsType(), optional=True), |
| "bypass_pre_encode": NeuralType(tuple(), BoolType(), optional=True), |
| } |
| ) |
|
|
| @property |
| def output_types(self): |
| """Returns definitions of module output ports.""" |
| return OrderedDict( |
| { |
| "outputs": NeuralType(('B', 'D', 'T'), AcousticEncodedRepresentation()), |
| "encoded_lengths": NeuralType(tuple('B'), LengthsType()), |
| "cache_last_channel_next": NeuralType(('D', 'B', 'T', 'D'), ChannelType(), optional=True), |
| "cache_last_time_next": NeuralType(('D', 'B', 'D', 'T'), ChannelType(), optional=True), |
| "cache_last_channel_next_len": NeuralType(tuple('B'), LengthsType(), optional=True), |
| } |
| ) |
|
|
| @property |
| def output_types_for_export(self): |
| """Returns definitions of module output ports.""" |
| return OrderedDict( |
| { |
| "outputs": NeuralType(('B', 'D', 'T'), AcousticEncodedRepresentation()), |
| "encoded_lengths": NeuralType(tuple('B'), LengthsType()), |
| "cache_last_channel_next": NeuralType(('B', 'D', 'T', 'D'), ChannelType(), optional=True), |
| "cache_last_time_next": NeuralType(('B', 'D', 'D', 'T'), ChannelType(), optional=True), |
| "cache_last_channel_next_len": NeuralType(tuple('B'), LengthsType(), optional=True), |
| } |
| ) |
|
|
| @property |
| def disabled_deployment_input_names(self): |
| if not self.export_cache_support: |
| return set(["cache_last_channel", "cache_last_time", "cache_last_channel_len"]) |
| else: |
| return set() |
|
|
| @property |
| def disabled_deployment_output_names(self): |
| if not self.export_cache_support: |
| return set(["cache_last_channel_next", "cache_last_time_next", "cache_last_channel_next_len"]) |
| else: |
| return set() |
|
|
| def __init__( |
| self, |
| feat_in, |
| n_layers, |
| d_model, |
| feat_out=-1, |
| causal_downsampling=False, |
| subsampling='striding', |
| subsampling_factor=4, |
| subsampling_conv_chunking_factor=1, |
| subsampling_conv_channels=-1, |
| reduction=None, |
| reduction_position=None, |
| reduction_factor=1, |
| ff_expansion_factor=4, |
| self_attention_model='rel_pos', |
| n_heads=4, |
| att_context_size=None, |
| att_context_probs=None, |
| att_chunk_context_size=None, |
| att_context_style='regular', |
| xscaling=True, |
| untie_biases=True, |
| pos_emb_max_len=5000, |
| conv_kernel_size=31, |
| conv_norm_type='batch_norm', |
| conv_context_size=None, |
| conv_context_style='regular', |
| use_bias=True, |
| dropout=0.1, |
| dropout_pre_encoder=0.1, |
| dropout_emb=0.1, |
| dropout_att=0.0, |
| stochastic_depth_drop_prob: float = 0.0, |
| stochastic_depth_mode: str = "linear", |
| stochastic_depth_start_layer: int = 1, |
| global_tokens: int = 0, |
| global_tokens_spacing: int = 1, |
| global_attn_separate: bool = False, |
| use_pytorch_sdpa: bool = False, |
| use_pytorch_sdpa_backends=None, |
| sync_max_audio_length: bool = True, |
| ): |
| super().__init__() |
| d_ff = d_model * ff_expansion_factor |
| self.d_model = d_model |
| self.n_layers = n_layers |
| self._feat_in = feat_in |
| self.att_context_style = att_context_style |
| self.subsampling_factor = subsampling_factor |
| self.subsampling_conv_chunking_factor = subsampling_conv_chunking_factor |
|
|
| self.self_attention_model = self_attention_model |
| self.global_tokens = global_tokens |
| self.global_attn_separate = global_attn_separate |
| self.global_tokens_spacing = global_tokens_spacing |
| self.use_pytorch_sdpa = use_pytorch_sdpa |
| if use_pytorch_sdpa_backends is None: |
| use_pytorch_sdpa_backends = [] |
| self.use_pytorch_sdpa_backends = use_pytorch_sdpa_backends |
| self.sync_max_audio_length = sync_max_audio_length |
|
|
| assert conv_context_style in ["regular", "dcc"], f"Invalid conv_context_style: {conv_context_style}!" |
| self.conv_context_style = conv_context_style |
| self.conv_kernel_size = conv_kernel_size |
|
|
| |
| if att_chunk_context_size is not None: |
| assert ( |
| att_context_style == "chunked_limited_with_rc" |
| ), "att_chunk_context_size is only supported for chunked_limited_with_rc attention style!" |
| assert ( |
| len(att_chunk_context_size) == 3 |
| ), "att_chunk_context_size must have 3 elements: [left_context, chunk_size, right_context]" |
| self.att_chunk_context_size = att_chunk_context_size |
| else: |
| self.att_chunk_context_size = None |
|
|
| |
| ( |
| self.att_context_size_all, |
| self.att_context_size, |
| self.att_context_probs, |
| self.conv_context_size, |
| ) = self._calc_context_sizes( |
| att_context_style=att_context_style, |
| att_context_size=att_context_size, |
| att_context_probs=att_context_probs, |
| conv_context_size=conv_context_size, |
| conv_kernel_size=conv_kernel_size, |
| ) |
|
|
| if xscaling: |
| self.xscale = math.sqrt(d_model) |
| else: |
| self.xscale = None |
|
|
| |
| if subsampling_conv_channels == -1: |
| subsampling_conv_channels = d_model |
| if subsampling and subsampling_factor > 1: |
| if subsampling in ['stacking', 'stacking_norm']: |
| |
| self.pre_encode = StackingSubsampling( |
| subsampling_factor=subsampling_factor, |
| feat_in=feat_in, |
| feat_out=d_model, |
| norm=True if subsampling == 'stacking_norm' else False, |
| ) |
| else: |
| self.pre_encode = ConvSubsampling( |
| subsampling=subsampling, |
| subsampling_factor=subsampling_factor, |
| feat_in=feat_in, |
| feat_out=d_model, |
| conv_channels=subsampling_conv_channels, |
| subsampling_conv_chunking_factor=subsampling_conv_chunking_factor, |
| activation=nn.ReLU(True), |
| is_causal=causal_downsampling, |
| ) |
| else: |
| self.pre_encode = nn.Linear(feat_in, d_model) |
|
|
| |
| if reduction and reduction_factor > 1: |
| assert reduction_position >= -1 and reduction_position < n_layers |
| self.reduction_subsampling = SubsamplingReductionModule( |
| reduction=reduction, |
| d_model=d_model, |
| reduction_factor=reduction_factor, |
| ) |
| self.reduction_position = reduction_position |
| else: |
| self.reduction_subsampling = None |
| self.reduction_position = None |
|
|
| self._feat_out = d_model |
|
|
| |
| if not untie_biases and self_attention_model == "rel_pos": |
| d_head = d_model // n_heads |
| pos_bias_u = nn.Parameter(torch.Tensor(n_heads, d_head)) |
| pos_bias_v = nn.Parameter(torch.Tensor(n_heads, d_head)) |
| nn.init.zeros_(pos_bias_u) |
| nn.init.zeros_(pos_bias_v) |
| else: |
| pos_bias_u = None |
| pos_bias_v = None |
|
|
| |
| self.pos_emb_max_len = pos_emb_max_len |
| if self_attention_model == "rel_pos": |
| self.pos_enc = RelPositionalEncoding( |
| d_model=d_model, |
| dropout_rate=dropout_pre_encoder, |
| max_len=pos_emb_max_len, |
| xscale=self.xscale, |
| dropout_rate_emb=dropout_emb, |
| ) |
| elif self_attention_model == 'rel_pos_local_attn': |
| if max(att_context_size) <= 0: |
| raise ValueError("When using local attention, context size must be set > 0") |
| self.pos_enc = LocalAttRelPositionalEncoding( |
| att_context_size=att_context_size, |
| d_model=d_model, |
| dropout_rate=dropout, |
| max_len=pos_emb_max_len, |
| xscale=self.xscale, |
| dropout_rate_emb=dropout_emb, |
| ) |
| elif self_attention_model == "abs_pos": |
| pos_bias_u = None |
| pos_bias_v = None |
| self.pos_enc = PositionalEncoding( |
| d_model=d_model, dropout_rate=dropout_pre_encoder, max_len=pos_emb_max_len, xscale=self.xscale |
| ) |
| else: |
| raise ValueError(f"Not valid self_attention_model: '{self_attention_model}'!") |
|
|
| self.layers = nn.ModuleList() |
| for i in range(n_layers): |
| layer = ConformerLayer( |
| d_model=d_model, |
| d_ff=d_ff, |
| self_attention_model=self_attention_model, |
| global_tokens=global_tokens, |
| global_tokens_spacing=global_tokens_spacing, |
| global_attn_separate=global_attn_separate, |
| n_heads=n_heads, |
| conv_kernel_size=conv_kernel_size, |
| conv_norm_type=conv_norm_type, |
| conv_context_size=self.conv_context_size, |
| dropout=dropout, |
| dropout_att=dropout_att, |
| pos_bias_u=pos_bias_u, |
| pos_bias_v=pos_bias_v, |
| att_context_size=self.att_context_size, |
| use_bias=use_bias, |
| use_pytorch_sdpa=self.use_pytorch_sdpa, |
| use_pytorch_sdpa_backends=self.use_pytorch_sdpa_backends, |
| ) |
| self.layers.append(layer) |
|
|
| if feat_out > 0 and feat_out != self._feat_out: |
| self.out_proj = nn.Linear(self._feat_out, feat_out) |
| self._feat_out = feat_out |
| else: |
| self.out_proj = None |
| self._feat_out = d_model |
| self.set_max_audio_length(self.pos_emb_max_len) |
| self.use_pad_mask = True |
|
|
| self.setup_streaming_params() |
| self.export_cache_support = False |
|
|
| self.layer_drop_probs = compute_stochastic_depth_drop_probs( |
| len(self.layers), stochastic_depth_drop_prob, stochastic_depth_mode, stochastic_depth_start_layer |
| ) |
| |
| self.interctc_capture_at_layers = None |
|
|
| def forward_for_export( |
| self, audio_signal, length, cache_last_channel=None, cache_last_time=None, cache_last_channel_len=None |
| ): |
| """ |
| Forward function for model export. Please see `forward()` for more details. |
| """ |
| if cache_last_channel is not None: |
| cache_last_channel = cache_last_channel.transpose(0, 1) |
| cache_last_time = cache_last_time.transpose(0, 1) |
|
|
| rets = self.forward_internal( |
| audio_signal, |
| length, |
| cache_last_channel=cache_last_channel, |
| cache_last_time=cache_last_time, |
| cache_last_channel_len=cache_last_channel_len, |
| ) |
| rets = self.streaming_post_process(rets, keep_all_outputs=False) |
| if len(rets) == 2: |
| return rets |
| elif rets[2] is None and rets[3] is None and rets[4] is None: |
| return (rets[0], rets[1]) |
| else: |
| return ( |
| rets[0], |
| rets[1], |
| rets[2].transpose(0, 1), |
| rets[3].transpose(0, 1), |
| rets[4], |
| ) |
|
|
| def streaming_post_process(self, rets, keep_all_outputs=True): |
| """ |
| Post-process the output of the forward function for streaming. |
| |
| Args: |
| rets: The output of the forward function. |
| keep_all_outputs: Whether to keep all outputs. |
| """ |
| if len(rets) == 2: |
| return rets[0], rets[1], None, None, None |
|
|
| (encoded, encoded_len, cache_last_channel_next, cache_last_time_next, cache_last_channel_next_len) = rets |
|
|
| if cache_last_channel_next is not None and self.streaming_cfg.last_channel_cache_size >= 0: |
| if self.streaming_cfg.last_channel_cache_size > 0: |
| cache_last_channel_next = cache_last_channel_next[ |
| :, :, -self.streaming_cfg.last_channel_cache_size :, : |
| ] |
|
|
| if self.streaming_cfg.valid_out_len > 0 and (not keep_all_outputs or self.att_context_style == "regular"): |
| encoded = encoded[:, :, : self.streaming_cfg.valid_out_len] |
| encoded_len = torch.clamp(encoded_len, max=self.streaming_cfg.valid_out_len) |
|
|
| return (encoded, encoded_len, cache_last_channel_next, cache_last_time_next, cache_last_channel_next_len) |
|
|
| @typecheck() |
| def forward( |
| self, |
| audio_signal, |
| length, |
| cache_last_channel=None, |
| cache_last_time=None, |
| cache_last_channel_len=None, |
| bypass_pre_encode=False, |
| ): |
| """ |
| Forward function for the ConformerEncoder accepting an audio signal and its corresponding length. |
| The ``audio_signal`` input supports two formats depending on ``bypass_pre_encode``: |
| |
| - ``bypass_pre_encode=False`` (default): ``audio_signal`` must be a tensor |
| containing audio features. Shape: ``(batch, feat_in, n_frames)``. |
| - ``bypass_pre_encode=True``: ``audio_signal`` must be a tensor containing |
| pre-encoded embeddings. Shape: ``(batch, n_frame, d_model)``. |
| """ |
| if not bypass_pre_encode and audio_signal.shape[-2] != self._feat_in: |
| raise ValueError( |
| f"If bypass_pre_encode is False, audio_signal should have shape " |
| f"(batch, {self._feat_in}, n_frame) but got last dimension {audio_signal.shape[-2]}." |
| ) |
| if bypass_pre_encode and audio_signal.shape[-1] != self.d_model: |
| raise ValueError( |
| f"If bypass_pre_encode is True, audio_signal should have shape " |
| f"(batch, n_frame, {self.d_model}) but got last dimension {audio_signal.shape[-1]}." |
| ) |
|
|
| if bypass_pre_encode: |
| self.update_max_seq_length(seq_length=audio_signal.size(1), device=audio_signal.device) |
| else: |
| self.update_max_seq_length(seq_length=audio_signal.size(2), device=audio_signal.device) |
| return self.forward_internal( |
| audio_signal, |
| length, |
| cache_last_channel=cache_last_channel, |
| cache_last_time=cache_last_time, |
| cache_last_channel_len=cache_last_channel_len, |
| bypass_pre_encode=bypass_pre_encode, |
| ) |
|
|
| def forward_internal( |
| self, |
| audio_signal, |
| length, |
| cache_last_channel=None, |
| cache_last_time=None, |
| cache_last_channel_len=None, |
| bypass_pre_encode=False, |
| ): |
| """ |
| The ``audio_signal`` input supports two formats depending on ``bypass_pre_encode``: |
| |
| - ``bypass_pre_encode=False`` (default): ``audio_signal`` must be a tensor |
| containing audio features. Shape: ``(batch, feat_in, n_frames)``. |
| - ``bypass_pre_encode=True``: ``audio_signal`` must be a tensor containing |
| pre-encoded embeddings. Shape: ``(batch, n_frame, d_model)``. |
| |
| ``bypass_pre_encode=True`` is used in cases where frame-level, context-independent embeddings are |
| needed to be saved or reused (e.g., speaker cache in streaming speaker diarization). |
| """ |
| if length is None: |
| length = audio_signal.new_full( |
| (audio_signal.size(0),), audio_signal.size(-1), dtype=torch.int64, device=audio_signal.device |
| ) |
|
|
| |
| |
| if self.training and len(self.att_context_size_all) > 1: |
| cur_att_context_size = random.choices(self.att_context_size_all, weights=self.att_context_probs)[0] |
| else: |
| cur_att_context_size = self.att_context_size |
|
|
| if not bypass_pre_encode: |
| audio_signal = torch.transpose(audio_signal, 1, 2) |
|
|
| if isinstance(self.pre_encode, nn.Linear): |
| audio_signal = self.pre_encode(audio_signal) |
| else: |
| audio_signal, length = self.pre_encode(x=audio_signal, lengths=length) |
| length = length.to(torch.int64) |
| |
| if self.streaming_cfg.drop_extra_pre_encoded > 0 and cache_last_channel is not None: |
| audio_signal = audio_signal[:, self.streaming_cfg.drop_extra_pre_encoded :, :] |
| length = (length - self.streaming_cfg.drop_extra_pre_encoded).clamp(min=0) |
|
|
| if self.reduction_position is not None and cache_last_channel is not None: |
| raise ValueError("Caching with reduction feature is not supported yet!") |
|
|
| max_audio_length = audio_signal.size(1) |
| if cache_last_channel is not None: |
| cache_len = self.streaming_cfg.last_channel_cache_size |
| cache_keep_size = max_audio_length - self.streaming_cfg.cache_drop_size |
| max_audio_length = max_audio_length + cache_len |
| padding_length = length + cache_len |
| offset = torch.neg(cache_last_channel_len) + cache_len |
| else: |
| padding_length = length |
| cache_last_channel_next = None |
| cache_len = 0 |
| offset = None |
|
|
| audio_signal, pos_emb = self.pos_enc(x=audio_signal, cache_len=cache_len) |
|
|
| |
| pad_mask, att_mask = self._create_masks( |
| att_context_size=cur_att_context_size, |
| padding_length=padding_length, |
| max_audio_length=max_audio_length, |
| offset=offset, |
| device=audio_signal.device, |
| ) |
|
|
| if cache_last_channel is not None: |
| pad_mask = pad_mask[:, cache_len:] |
| if att_mask is not None: |
| att_mask = att_mask[:, cache_len:] |
| |
| cache_last_time_next = [] |
| cache_last_channel_next = [] |
|
|
| for lth, (drop_prob, layer) in enumerate(zip(self.layer_drop_probs, self.layers)): |
| original_signal = audio_signal |
| if cache_last_channel is not None: |
| cache_last_channel_cur = cache_last_channel[lth] |
| cache_last_time_cur = cache_last_time[lth] |
| else: |
| cache_last_channel_cur = None |
| cache_last_time_cur = None |
| audio_signal = layer( |
| x=audio_signal, |
| att_mask=att_mask, |
| pos_emb=pos_emb, |
| pad_mask=pad_mask, |
| cache_last_channel=cache_last_channel_cur, |
| cache_last_time=cache_last_time_cur, |
| ) |
|
|
| if cache_last_channel_cur is not None: |
| (audio_signal, cache_last_channel_cur, cache_last_time_cur) = audio_signal |
| cache_last_channel_next.append(cache_last_channel_cur) |
| cache_last_time_next.append(cache_last_time_cur) |
|
|
| |
| if self.training and drop_prob > 0.0: |
| should_drop = torch.rand(1) < drop_prob |
| |
| if should_drop: |
| |
| |
| |
| audio_signal = audio_signal * 0.0 + original_signal |
| else: |
| |
| audio_signal = (audio_signal - original_signal) / (1.0 - drop_prob) + original_signal |
|
|
| if self.reduction_position == lth: |
| audio_signal, length = self.reduction_subsampling(x=audio_signal, lengths=length) |
| max_audio_length = audio_signal.size(1) |
| |
| |
| _, pos_emb = self.pos_enc(x=audio_signal, cache_len=cache_len) |
| pad_mask, att_mask = self._create_masks( |
| att_context_size=cur_att_context_size, |
| padding_length=length, |
| max_audio_length=max_audio_length, |
| offset=offset, |
| device=audio_signal.device, |
| ) |
| |
| if self.is_access_enabled(getattr(self, "model_guid", None)): |
| if self.interctc_capture_at_layers is None: |
| self.interctc_capture_at_layers = self.access_cfg.get('interctc', {}).get('capture_layers', []) |
| if lth in self.interctc_capture_at_layers: |
| lth_audio_signal = audio_signal |
| if self.out_proj is not None: |
| lth_audio_signal = self.out_proj(audio_signal) |
| |
| self.register_accessible_tensor( |
| name=f'interctc/layer_output_{lth}', tensor=torch.transpose(lth_audio_signal, 1, 2) |
| ) |
| self.register_accessible_tensor(name=f'interctc/layer_length_{lth}', tensor=length) |
|
|
| if self.out_proj is not None: |
| audio_signal = self.out_proj(audio_signal) |
|
|
| |
| if self.reduction_position == -1: |
| audio_signal, length = self.reduction_subsampling(x=audio_signal, lengths=length) |
|
|
| audio_signal = torch.transpose(audio_signal, 1, 2) |
| length = length.to(dtype=torch.int64) |
|
|
| if cache_last_channel is not None: |
| cache_last_channel_next = torch.stack(cache_last_channel_next, dim=0) |
| cache_last_time_next = torch.stack(cache_last_time_next, dim=0) |
| return ( |
| audio_signal, |
| length, |
| cache_last_channel_next, |
| cache_last_time_next, |
| torch.clamp(cache_last_channel_len + cache_keep_size, max=cache_len), |
| ) |
| else: |
| return audio_signal, length |
|
|
| def update_max_seq_length(self, seq_length: int, device): |
| """ |
| Updates the maximum sequence length for the model. |
| |
| Args: |
| seq_length (int): New maximum sequence length. |
| device (torch.device): Device to use for computations. |
| """ |
| |
| if self.sync_max_audio_length and torch.distributed.is_initialized(): |
| global_max_len = torch.tensor([seq_length], dtype=torch.float32, device=device) |
|
|
| |
| torch.distributed.all_reduce(global_max_len, op=torch.distributed.ReduceOp.MAX) |
|
|
| seq_length = global_max_len.int().item() |
|
|
| if seq_length > self.max_audio_length: |
| self.set_max_audio_length(seq_length) |
|
|
| def set_max_audio_length(self, max_audio_length): |
| """ |
| Sets maximum input length. |
| Pre-calculates internal seq_range mask. |
| |
| Args: |
| max_audio_length (int): New maximum sequence length. |
| """ |
| self.max_audio_length = max_audio_length |
| device = next(self.parameters()).device |
| dtype = next(self.parameters()).dtype |
| self.pos_enc.extend_pe(max_audio_length, device, dtype) |
|
|
| def _create_masks(self, att_context_size, padding_length, max_audio_length, offset, device): |
| if self.self_attention_model != "rel_pos_local_attn": |
| att_mask = torch.ones(1, max_audio_length, max_audio_length, dtype=torch.bool, device=device) |
|
|
| if self.att_context_style == "regular": |
| if att_context_size[0] >= 0: |
| att_mask = att_mask.triu(diagonal=-att_context_size[0]) |
| if att_context_size[1] >= 0: |
| att_mask = att_mask.tril(diagonal=att_context_size[1]) |
| elif self.att_context_style == "chunked_limited": |
| |
| if att_context_size[1] == -1: |
| if att_context_size[0] >= 0: |
| att_mask = att_mask.triu(diagonal=-att_context_size[0]) |
| else: |
| chunk_size = att_context_size[1] + 1 |
| |
| if att_context_size[0] >= 0: |
| left_chunks_num = att_context_size[0] // chunk_size |
| else: |
| left_chunks_num = 10000 |
|
|
| chunk_idx = torch.arange(0, max_audio_length, dtype=torch.int, device=att_mask.device) |
| chunk_idx = torch.div(chunk_idx, chunk_size, rounding_mode="trunc") |
| diff_chunks = chunk_idx.unsqueeze(1) - chunk_idx.unsqueeze(0) |
| chunked_limited_mask = torch.logical_and( |
| torch.le(diff_chunks, left_chunks_num), torch.ge(diff_chunks, 0) |
| ) |
| att_mask = torch.logical_and(att_mask, chunked_limited_mask.unsqueeze(0)) |
| elif self.att_context_style == "chunked_limited_with_rc" and sum(att_context_size) != -3: |
| assert ( |
| len(att_context_size) == 3 |
| ), "att_context_size must have 3 elements: [left_context, chunk_size, right_context]" |
|
|
| left_context_frames = att_context_size[0] |
| chunk_size_frames = att_context_size[1] |
| right_context_frames = att_context_size[2] |
| assert chunk_size_frames >= 1, "chunk_size_frames must be greater than 0!" |
| |
| frame_idx = torch.arange(0, max_audio_length, dtype=torch.int, device=att_mask.device) |
| chunk_idx = torch.div(frame_idx, chunk_size_frames, rounding_mode="trunc") |
|
|
| window_start = chunk_idx * chunk_size_frames - left_context_frames |
| window_start = torch.maximum(window_start, torch.zeros_like(window_start)) |
| window_end = chunk_idx * chunk_size_frames + chunk_size_frames - 1 + right_context_frames |
|
|
| window_end = torch.minimum(window_end, torch.full_like(window_end, max_audio_length - 1)) |
| |
| j_indices = frame_idx.unsqueeze(0) |
| window_start_expanded = window_start.unsqueeze(1) |
| window_end_expanded = window_end.unsqueeze(1) |
|
|
| chunked_limited_mask = torch.logical_and( |
| j_indices >= window_start_expanded, j_indices <= window_end_expanded |
| ) |
| att_mask = torch.logical_and(att_mask, chunked_limited_mask.unsqueeze(0)) |
| else: |
| att_mask = None |
|
|
| |
| pad_mask = torch.arange(0, max_audio_length, device=device).expand( |
| padding_length.size(0), -1 |
| ) < padding_length.unsqueeze(-1) |
|
|
| if offset is not None: |
| pad_mask_off = torch.arange(0, max_audio_length, device=device).expand( |
| padding_length.size(0), -1 |
| ) >= offset.unsqueeze(-1) |
| pad_mask = pad_mask_off.logical_and(pad_mask) |
|
|
| if att_mask is not None: |
| |
| pad_mask_for_att_mask = pad_mask.unsqueeze(1).repeat([1, max_audio_length, 1]) |
| pad_mask_for_att_mask = torch.logical_and(pad_mask_for_att_mask, pad_mask_for_att_mask.transpose(1, 2)) |
| |
| att_mask = att_mask[:, :max_audio_length, :max_audio_length] |
| |
| att_mask = torch.logical_and(pad_mask_for_att_mask, att_mask.to(pad_mask_for_att_mask.device)) |
| att_mask = ~att_mask |
|
|
| pad_mask = ~pad_mask |
| return pad_mask, att_mask |
|
|
| def enable_pad_mask(self, on=True): |
| """ |
| Enables or disables the pad mask and assign the boolean state `on`. |
| |
| Returns: |
| mask (bool): The current state of the pad mask. |
| """ |
| |
| mask = self.use_pad_mask |
| self.use_pad_mask = on |
| return mask |
|
|
| def _calc_context_sizes( |
| self, att_context_size, att_context_probs, att_context_style, conv_context_size, conv_kernel_size |
| ): |
| |
| if att_context_size: |
| att_context_size_all = list(att_context_size) |
| if isinstance(att_context_size_all[0], int): |
| att_context_size_all = [att_context_size_all] |
| for i, att_cs in enumerate(att_context_size_all): |
| if isinstance(att_cs, ListConfig): |
| att_context_size_all[i] = list(att_cs) |
| if att_context_style == "chunked_limited": |
| if att_cs[0] > 0 and att_cs[0] % (att_cs[1] + 1) > 0: |
| raise ValueError(f"att_context_size[{i}][0] % (att_context_size[{i}][1] + 1) should be zero!") |
| if att_cs[1] < 0 and len(att_context_size_all) <= 1: |
| raise ValueError( |
| f"Right context (att_context_size[{i}][1]) can not be unlimited for chunked_limited style!" |
| ) |
| else: |
| att_context_size_all = [[-1, -1]] |
|
|
| if att_context_style == "chunked_limited_with_rc": |
| att_context_size_all = [[-1, -1, -1]] |
|
|
| if att_context_probs: |
| if len(att_context_probs) != len(att_context_size_all): |
| raise ValueError("The size of the att_context_probs should be the same as att_context_size.") |
| att_context_probs = list(att_context_probs) |
| if sum(att_context_probs) != 1: |
| raise ValueError( |
| "The sum of numbers in att_context_probs should be equal to one to be a distribution." |
| ) |
| else: |
| att_context_probs = [1.0 / len(att_context_size_all)] * len(att_context_size_all) |
|
|
| if conv_context_size is not None: |
| if isinstance(conv_context_size, ListConfig): |
| conv_context_size = list(conv_context_size) |
| if not isinstance(conv_context_size, list) and not isinstance(conv_context_size, str): |
| raise ValueError( |
| "Invalid conv_context_size! It should be the string 'causal' or a list of two integers." |
| ) |
| if conv_context_size == "causal": |
| conv_context_size = [conv_kernel_size - 1, 0] |
| else: |
| if conv_context_size[0] + conv_context_size[1] + 1 != conv_kernel_size: |
| raise ValueError(f"Invalid conv_context_size: {self.conv_context_size}!") |
| else: |
| conv_context_size = [(conv_kernel_size - 1) // 2, (conv_kernel_size - 1) // 2] |
| return att_context_size_all, att_context_size_all[0], att_context_probs, conv_context_size |
|
|
| def set_default_att_context_size(self, att_context_size): |
| """ |
| Sets the default attention context size from `att_context_size` argument. |
| |
| Args: |
| att_context_size (list): The attention context size to be set. |
| """ |
| if att_context_size not in self.att_context_size_all: |
| logging.warning( |
| f"att_context_size={att_context_size} is not among the list of the supported " |
| f"look-aheads: {self.att_context_size_all}" |
| ) |
| if att_context_size is not None: |
| self.att_context_size = att_context_size |
|
|
| self.setup_streaming_params() |
|
|
| def setup_streaming_params( |
| self, |
| chunk_size: int = None, |
| shift_size: int = None, |
| left_chunks: int = None, |
| att_context_size: list = None, |
| max_context: int = 10000, |
| ): |
| """ |
| This function sets the needed values and parameters to perform streaming. |
| The configuration would be stored in self.streaming_cfg. |
| The streaming configuration is needed to simulate streaming inference. |
| |
| Args: |
| chunk_size (int): overrides the chunk size |
| shift_size (int): overrides the shift size for chunks |
| left_chunks (int): overrides the number of left chunks visible to each chunk |
| max_context (int): the value used for the cache size of last_channel layers |
| if left context is set to infinity (-1) |
| Defaults to -1 (means feat_out is d_model) |
| """ |
| streaming_cfg = CacheAwareStreamingConfig() |
|
|
| |
| if att_context_size is None: |
| att_context_size = self.att_context_size |
|
|
| if chunk_size is not None: |
| if chunk_size < 1: |
| raise ValueError("chunk_size needs to be a number larger or equal to one.") |
| lookahead_steps = chunk_size - 1 |
| streaming_cfg.cache_drop_size = chunk_size - shift_size |
| elif self.att_context_style == "chunked_limited": |
| lookahead_steps = att_context_size[1] |
| streaming_cfg.cache_drop_size = 0 |
| elif self.att_context_style == "chunked_limited_with_rc": |
| lookahead_steps = att_context_size[2] * self.n_layers + self.conv_context_size[1] * self.n_layers |
| streaming_cfg.cache_drop_size = 0 |
| elif self.att_context_style == "regular": |
| lookahead_steps = att_context_size[1] * self.n_layers + self.conv_context_size[1] * self.n_layers |
| streaming_cfg.cache_drop_size = lookahead_steps |
| else: |
| streaming_cfg.cache_drop_size = 0 |
| lookahead_steps = None |
|
|
| if chunk_size is None: |
| streaming_cfg.last_channel_cache_size = att_context_size[0] if att_context_size[0] >= 0 else max_context |
| else: |
| if left_chunks is None: |
| streaming_cfg.last_channel_cache_size = ( |
| att_context_size[0] if att_context_size[0] >= 0 else max_context |
| ) |
| logging.warning( |
| f"left_chunks is not set. Setting it to default: {streaming_cfg.last_channel_cache_size}." |
| ) |
| else: |
| streaming_cfg.last_channel_cache_size = left_chunks * chunk_size |
|
|
| if hasattr(self.pre_encode, "get_sampling_frames"): |
| sampling_frames = self.pre_encode.get_sampling_frames() |
| else: |
| sampling_frames = 0 |
|
|
| if isinstance(sampling_frames, list): |
| streaming_cfg.chunk_size = [ |
| sampling_frames[0] + self.subsampling_factor * lookahead_steps, |
| sampling_frames[1] + self.subsampling_factor * lookahead_steps, |
| ] |
| else: |
| streaming_cfg.chunk_size = sampling_frames * (1 + lookahead_steps) |
|
|
| if isinstance(sampling_frames, list): |
| streaming_cfg.shift_size = [ |
| sampling_frames[0] + sampling_frames[1] * (lookahead_steps - streaming_cfg.cache_drop_size), |
| sampling_frames[1] + sampling_frames[1] * (lookahead_steps - streaming_cfg.cache_drop_size), |
| ] |
| else: |
| streaming_cfg.shift_size = sampling_frames * (1 + lookahead_steps - streaming_cfg.cache_drop_size) |
|
|
| if isinstance(streaming_cfg.shift_size, list): |
| streaming_cfg.valid_out_len = ( |
| streaming_cfg.shift_size[1] - sampling_frames[1] |
| ) // self.subsampling_factor + 1 |
| else: |
| streaming_cfg.valid_out_len = streaming_cfg.shift_size // self.subsampling_factor |
|
|
| if hasattr(self.pre_encode, "get_streaming_cache_size"): |
| streaming_cfg.pre_encode_cache_size = self.pre_encode.get_streaming_cache_size() |
| else: |
| streaming_cfg.pre_encode_cache_size = 0 |
|
|
| if isinstance(streaming_cfg.pre_encode_cache_size, list): |
| if streaming_cfg.pre_encode_cache_size[1] >= 1: |
| streaming_cfg.drop_extra_pre_encoded = ( |
| 1 + (streaming_cfg.pre_encode_cache_size[1] - 1) // self.subsampling_factor |
| ) |
| else: |
| streaming_cfg.drop_extra_pre_encoded = 0 |
| else: |
| streaming_cfg.drop_extra_pre_encoded = streaming_cfg.pre_encode_cache_size // self.subsampling_factor |
|
|
| for m in self.layers.modules(): |
| if hasattr(m, "_max_cache_len"): |
| if isinstance(m, MultiHeadAttention): |
| m.cache_drop_size = streaming_cfg.cache_drop_size |
| if isinstance(m, CausalConv1D): |
| m.cache_drop_size = streaming_cfg.cache_drop_size |
|
|
| self.streaming_cfg = streaming_cfg |
|
|
| def get_initial_cache_state(self, batch_size=1, dtype=torch.float32, device=None, max_dim=0): |
| if device is None: |
| device = next(self.parameters()).device |
| if max_dim > 0: |
| create_tensor = torch.randn |
| else: |
| create_tensor = torch.zeros |
| last_time_cache_size = self.conv_context_size[0] |
| cache_last_channel = create_tensor( |
| ( |
| len(self.layers), |
| batch_size, |
| self.streaming_cfg.last_channel_cache_size, |
| self.d_model, |
| ), |
| device=device, |
| dtype=dtype, |
| ) |
| cache_last_time = create_tensor( |
| (len(self.layers), batch_size, self.d_model, last_time_cache_size), |
| device=device, |
| dtype=dtype, |
| ) |
| if max_dim > 0: |
| cache_last_channel_len = torch.randint( |
| 0, |
| min(max_dim, self.streaming_cfg.last_channel_cache_size), |
| (batch_size,), |
| device=device, |
| dtype=torch.int64, |
| ) |
| for i in range(batch_size): |
| cache_last_channel[:, i, cache_last_channel_len[i] :, :] = 0 |
| |
| if cache_last_channel_len[i] == 0: |
| cache_last_time[:, i, :, :] = 0 |
| else: |
| cache_last_channel_len = torch.zeros(batch_size, device=device, dtype=torch.int64) |
| return cache_last_channel, cache_last_time, cache_last_channel_len |
|
|
| def change_attention_model( |
| self, |
| self_attention_model: str = None, |
| att_context_size: List[int] = None, |
| update_config: bool = True, |
| device: torch.device = None, |
| ): |
| """ |
| Update the self_attention_model which changes the positional encoding and attention layers. |
| |
| Args: |
| self_attention_model (str): type of the attention layer and positional encoding |
| |
| 'rel_pos': |
| relative positional embedding and Transformer-XL |
| |
| 'rel_pos_local_attn': |
| relative positional embedding and Transformer-XL with local attention using |
| overlapping windows. Attention context is determined by att_context_size parameter. |
| |
| 'abs_pos': |
| absolute positional embedding and Transformer |
| |
| If None is provided, the self_attention_model isn't changed. Defaults to None. |
| att_context_size (List[int]): List of 2 ints corresponding to left and right attention context sizes, |
| or None to keep as it is. Defaults to None. |
| update_config (bool): Whether to update the config or not with the new attention model. |
| Defaults to True. |
| device (torch.device): If provided, new layers will be moved to the device. |
| Defaults to None. |
| """ |
|
|
| if att_context_size: |
| att_context_size = list(att_context_size) |
| else: |
| att_context_size = self.att_context_size |
|
|
| if self_attention_model is None: |
| self_attention_model = self.self_attention_model |
|
|
| if self_attention_model == 'rel_pos_local_attn' and max(att_context_size) <= 0: |
| raise ValueError("When using local attention, context size must be set > 0") |
|
|
| if self_attention_model == "rel_pos": |
| new_pos_enc = RelPositionalEncoding( |
| d_model=self._cfg.d_model, |
| dropout_rate=self._cfg.dropout, |
| max_len=self._cfg.pos_emb_max_len, |
| xscale=self.xscale, |
| dropout_rate_emb=self._cfg.dropout_emb, |
| ) |
| elif self_attention_model == 'rel_pos_local_attn': |
| new_pos_enc = LocalAttRelPositionalEncoding( |
| att_context_size=att_context_size, |
| d_model=self._cfg.d_model, |
| dropout_rate=self._cfg.dropout, |
| max_len=self._cfg.pos_emb_max_len, |
| xscale=self.xscale, |
| dropout_rate_emb=self._cfg.dropout_emb, |
| ) |
| elif self_attention_model == "abs_pos": |
| new_pos_enc = PositionalEncoding( |
| d_model=self._cfg.d_model, |
| dropout_rate=self._cfg.dropout, |
| max_len=self._cfg.pos_emb_max_len, |
| xscale=self.xscale, |
| ) |
| else: |
| raise ValueError(f"Not valid self_attention_model: '{self_attention_model}'!") |
|
|
| if device is not None: |
| new_pos_enc = new_pos_enc.to(device=device) |
| del self.pos_enc |
| self.pos_enc = new_pos_enc |
| self.self_attention_model = self_attention_model |
| self.att_context_size = att_context_size |
| self.set_max_audio_length(self.pos_emb_max_len) |
|
|
| for _, m in self.named_modules(): |
| if type(m) == ConformerLayer: |
| if self_attention_model == 'rel_pos': |
| new_attn = RelPositionMultiHeadAttention( |
| n_head=self._cfg.n_heads, |
| n_feat=self._cfg.d_model, |
| dropout_rate=self._cfg.dropout_att, |
| max_cache_len=att_context_size[0], |
| pos_bias_u=None, |
| pos_bias_v=None, |
| use_pytorch_sdpa=self.use_pytorch_sdpa, |
| use_pytorch_sdpa_backends=self.use_pytorch_sdpa_backends, |
| ) |
| elif self_attention_model == 'rel_pos_local_attn': |
| new_attn = RelPositionMultiHeadAttentionLongformer( |
| n_head=self._cfg.n_heads, |
| n_feat=self._cfg.d_model, |
| dropout_rate=self._cfg.dropout_att, |
| max_cache_len=att_context_size[0], |
| att_context_size=att_context_size, |
| pos_bias_u=None, |
| pos_bias_v=None, |
| use_pytorch_sdpa=self.use_pytorch_sdpa, |
| use_pytorch_sdpa_backends=self.use_pytorch_sdpa_backends, |
| ) |
| elif self_attention_model == 'abs_pos': |
| new_attn = MultiHeadAttention( |
| n_head=self._cfg.n_heads, |
| n_feat=self._cfg.d_model, |
| dropout_rate=self._cfg.dropout_att, |
| max_cache_len=att_context_size[0], |
| use_pytorch_sdpa=self.use_pytorch_sdpa, |
| use_pytorch_sdpa_backends=self.use_pytorch_sdpa_backends, |
| ) |
| else: |
| raise ValueError( |
| f"'{self_attention_model}' is not not a valid value for 'self_attention_model', " |
| f"valid values can be from ['rel_pos', 'rel_pos_local_attn', 'abs_pos']" |
| ) |
| if device is not None: |
| new_attn = new_attn.to(device=device) |
| new_attn.load_state_dict(m.self_attn.state_dict(), strict=False) |
| del m.self_attn |
| m.self_attn = new_attn |
| m.self_attention_model = self_attention_model |
|
|
| if update_config: |
| with open_dict(self._cfg): |
| self._cfg.self_attention_model = self_attention_model |
| self._cfg.att_context_size = att_context_size |
|
|
| def change_subsampling_conv_chunking_factor(self, subsampling_conv_chunking_factor: int): |
| """ |
| Update the conv_chunking_factor (int) |
| Default is 1 (auto) |
| Set it to -1 (disabled) or to a specific value (power of 2) if you OOM in the conv subsampling layers |
| |
| |
| Args: |
| subsampling_conv_chunking_factor (int) |
| """ |
|
|
| if not hasattr(self.pre_encode, "change_subsampling_conv_chunking_factor"): |
| logging.info("Model pre_encoder doesn't have a change_subsampling_conv_chunking_factor method ") |
| return |
|
|
| self.pre_encode.change_subsampling_conv_chunking_factor( |
| subsampling_conv_chunking_factor=subsampling_conv_chunking_factor |
| ) |
|
|
|
|
| class ConformerEncoderAdapter(ConformerEncoder, adapter_mixins.AdapterModuleMixin): |
| """This class inherits from ConformerEncoder and wraps the adapter mixin class.""" |
|
|
| |
| def add_adapter(self, name: str, cfg: dict): |
| cfg = self._update_adapter_cfg_input_dim(cfg) |
| for conformer_layer in self.layers: |
| conformer_layer.add_adapter(name, cfg) |
|
|
| def is_adapter_available(self) -> bool: |
| return any([conformer_layer.is_adapter_available() for conformer_layer in self.layers]) |
|
|
| def set_enabled_adapters(self, name: Optional[str] = None, enabled: bool = True): |
| for conformer_layer in self.layers: |
| conformer_layer.set_enabled_adapters(name=name, enabled=enabled) |
|
|
| def get_enabled_adapters(self) -> List[str]: |
| names = set([]) |
| for conformer_layer in self.layers: |
| names.update(conformer_layer.get_enabled_adapters()) |
|
|
| names = sorted(list(names)) |
| return names |
|
|
| def _update_adapter_cfg_input_dim(self, cfg: DictConfig): |
| cfg = adapter_utils.update_adapter_cfg_input_dim(self, cfg, module_dim=self.d_model) |
| return cfg |
|
|
| def get_accepted_adapter_types( |
| self, |
| ) -> Set[type]: |
| types = super().get_accepted_adapter_types() |
|
|
| if len(types) == 0: |
| self.set_accepted_adapter_types( |
| [ |
| adapter_utils.LINEAR_ADAPTER_CLASSPATH, |
| adapter_utils.MHA_ADAPTER_CLASSPATH, |
| adapter_utils.RELMHA_ADAPTER_CLASSPATH, |
| ] |
| ) |
| types = self.get_accepted_adapter_types() |
| return types |
|
|
|
|
| class ConformerMultiLayerFeatureExtractor(NeuralModule, Exportable, AccessMixin): |
| """ |
| A wrapper module that extracts features from multiple layers of a ConformerEncoder, |
| by reusing existing mechanisim for interctc loss. |
| To use it, set `layer_idx_list` to specify the indices of layers to extract from. |
| Also, you can specify an `aggretator` module to aggregate the features from different layers, |
| default not aggregating. |
| """ |
|
|
| def __init__( |
| self, |
| encoder: ConformerEncoder, |
| layer_idx_list: Optional[List[int]] = None, |
| aggregator: Optional[NeuralModule] = None, |
| detach: bool = False, |
| convert_to_cpu: bool = False, |
| ): |
| """ |
| This class is used to extract features from different layers of the ConformerEncoder. |
| Args: |
| encoder: ConformerEncoder instance. |
| layer_idx_list: List of layer indices to extract features from. If None, all layers are extracted. |
| aggregator: Aggregator instance. If None, the features are returned as a list. |
| detach: If True, the features are detached from the graph. |
| convert_to_cpu: If True, the features are converted to CPU. |
| """ |
| super().__init__() |
| self.encoder = encoder |
| self.num_layers = len(encoder.layers) |
| self.layer_idx_list = [] |
| if not layer_idx_list: |
| layer_idx_list = list(range(self.num_layers)) |
| for lid in layer_idx_list: |
| if lid < -self.num_layers or lid >= self.num_layers: |
| raise ValueError(f"Invalid layer index {lid} for ConformerEncoder with {self.num_layers} layers.") |
| if lid < 0: |
| lid = self.num_layers + lid |
| self.layer_idx_list.append(lid) |
| self.layer_idx_list.sort() |
| logging.info(f"Extracting ConformerEncoder features from layers: {self.layer_idx_list}") |
| self.enc_access_cfg = { |
| "interctc": { |
| "capture_layers": self.layer_idx_list, |
| }, |
| "detach": detach, |
| "convert_to_cpu": convert_to_cpu, |
| } |
| self.aggregator = aggregator |
|
|
| def forward( |
| self, audio_signal, length, cache_last_channel=None, cache_last_time=None, cache_last_channel_len=None |
| ) -> Tuple[torch.Tensor, torch.Tensor]: |
| """ |
| Args: |
| same interface as ConformerEncoder.forward() |
| Returns: |
| - Tuple[List[Tensor[B,D,T]], List[Tensor[B]]] if aggregator is None |
| - Tuple[Tensor[B,H,T], Tensor[B]] if aggregator is not None, where H is the hidden size of the aggregator |
| """ |
| old_access_flag = self.is_access_enabled(guid=getattr(self, "model_guid", None)) |
| self.update_access_cfg(self.enc_access_cfg, guid=getattr(self, "model_guid", None)) |
| self.set_access_enabled(access_enabled=True, guid=getattr(self, "model_guid", None)) |
|
|
| _ = self.encoder( |
| audio_signal=audio_signal, |
| length=length, |
| cache_last_channel=cache_last_channel, |
| cache_last_time=cache_last_time, |
| cache_last_channel_len=cache_last_channel_len, |
| ) |
|
|
| |
| total_registry = {} |
| for module_registry in self.get_module_registry(self.encoder).values(): |
| for key in module_registry: |
| if key.startswith("interctc/") and key in total_registry: |
| raise RuntimeError(f"layer {key} has been logged multiple times!") |
| total_registry.update(module_registry) |
|
|
| encoded_list = [] |
| encoded_len_list = [] |
| for layer_idx in self.layer_idx_list: |
| try: |
| layer_outputs = total_registry[f"interctc/layer_output_{layer_idx}"] |
| layer_lengths = total_registry[f"interctc/layer_length_{layer_idx}"] |
| except KeyError: |
| raise RuntimeError( |
| f"Intermediate layer {layer_idx} was not captured! " |
| "Check the layer index and the number of ConformerEncoder layers." |
| ) |
| if len(layer_outputs) > 1 or len(layer_lengths) > 1: |
| raise RuntimeError("Make sure encoder.forward is called exactly one time") |
| encoded_list.append(layer_outputs[0]) |
| encoded_len_list.append(layer_lengths[0]) |
|
|
| self.encoder.reset_registry() |
| self.set_access_enabled(access_enabled=old_access_flag, guid=getattr(self, "model_guid", None)) |
| |
|
|
| if self.aggregator is not None: |
| return self.aggregator(encoded_list, encoded_len_list) |
| else: |
| return encoded_list, encoded_len_list |
|
|
|
|
| |
| if adapter_mixins.get_registered_adapter(ConformerEncoder) is None: |
| adapter_mixins.register_adapter(base_class=ConformerEncoder, adapter_class=ConformerEncoderAdapter) |
|
|
|
|
| @dataclass |
| class ConformerChangeConfig: |
| """ |
| Change self_attention_model for Conformer. |
| |
| Options: |
| 'rel_pos': relative positional embedding and Transformer-XL |
| 'rel_pos_local_attn': relative positional embedding and Transformer-XL with local attention using |
| overlapping chunks. Attention context is determined by att_context_size parameter. |
| 'abs_pos': absolute positional embedding and Transformer |
| """ |
|
|
| |
| self_attention_model: Optional[str] = None |
|
|
| |
| |
| |
| att_context_size: Optional[List[int]] = None |
|
|