# Copyright 2023 (authors: Feiteng Li) # # 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. import random from typing import Dict, Iterator, List, Tuple, Union import numpy as np import torch import torch.nn as nn import torch.nn.functional as F # from icefall.utils import make_pad_mask # from torchmetrics.classification import MulticlassAccuracy from modules.embedding import SinePositionalEmbedding, TokenEmbedding from modules.transformer import ( AdaptiveLayerNorm, LayerNorm, TransformerDecoderLayer, TransformerEncoder, TransformerEncoderLayer, ) from .macros import NUM_AUDIO_TOKENS, NUM_TEXT_TOKENS import psutil def get_memory_usage(): process = psutil.Process() memory_info = process.memory_info() memory_used = memory_info.rss memory_used_mb = memory_used / (1024 * 1024) return memory_used_mb class Transpose(nn.Identity): """(N, T, D) -> (N, D, T)""" def forward(self, input: torch.Tensor) -> torch.Tensor: return input.transpose(1, 2) # NOTE: There are two ways to implement the model # 1) [VALL-F] standard TransformerDecoder, use x as memory # 2) [VALL-E] modified TransformerDecoder like GPT-x(e.g. causal TransformerEncoder), # use x as the prefix of decoder inputs class VALLF(nn.Module): """It implements https://arxiv.org/abs/2301.02111 "Neural Codec Language Models are Zero-Shot Text to Speech Synthesizers" """ def __init__( self, d_model: int, nhead: int, num_layers: int, norm_first: bool = True, add_prenet: bool = False, decoder_cls: Union[ nn.TransformerDecoder, nn.TransformerEncoder ] = nn.TransformerDecoder, decoder_layer_cls: Union[ TransformerDecoderLayer, TransformerEncoderLayer ] = TransformerDecoderLayer, prefix_mode: int = 0, share_embedding: bool = True, nar_scale_factor: float = 1.0, prepend_bos: bool = True, num_quantizers: int = 8, ): """ Args: d_model: The number of expected features in the input (required). nhead: The number of heads in the multiheadattention models (required). num_layers: The number of sub-decoder-layers in the decoder (required). """ super().__init__() nar_d_model = int(d_model * nar_scale_factor) self.ar_text_embedding = TokenEmbedding(d_model, NUM_TEXT_TOKENS) # W_x self.nar_text_embedding = TokenEmbedding(nar_d_model, NUM_TEXT_TOKENS) # ID NUM_AUDIO_TOKENS -> PAD # ID NUM_AUDIO_TOKENS + 1 -> BOS self.ar_audio_prepend_bos = prepend_bos self.ar_audio_embedding = TokenEmbedding( d_model, NUM_AUDIO_TOKENS + 1 + int(prepend_bos) ) # PreNet if add_prenet: self.ar_text_prenet = nn.Sequential( Transpose(), nn.Conv1d(d_model, d_model, kernel_size=5, padding="same"), nn.BatchNorm1d(d_model), nn.ReLU(), nn.Dropout(0.5), nn.Conv1d(d_model, d_model, kernel_size=5, padding="same"), nn.BatchNorm1d(d_model), nn.ReLU(), nn.Dropout(0.5), nn.Conv1d(d_model, d_model, kernel_size=5, padding="same"), nn.BatchNorm1d(d_model), nn.ReLU(), nn.Dropout(0.5), Transpose(), nn.Linear(d_model, d_model), ) self.ar_audio_prenet = nn.Sequential( nn.Linear(d_model, 256), nn.ReLU(), nn.Dropout(0.25), nn.Linear(256, 256), nn.ReLU(), nn.Dropout(0.25), nn.Linear(256, d_model), ) else: self.ar_text_prenet = nn.Identity() self.ar_audio_prenet = nn.Identity() self.ar_text_position = SinePositionalEmbedding( d_model, dropout=0.1, scale=False, alpha=True, ) self.ar_audio_position = SinePositionalEmbedding( d_model, dropout=0.1, scale=False, alpha=True, ) self.ar_decoder = decoder_cls( decoder_layer_cls( d_model, nhead, dim_feedforward=d_model * 4, dropout=0.1, batch_first=True, norm_first=norm_first, ), num_layers=num_layers, norm=LayerNorm(d_model) if norm_first else None, ) self.ar_predict_layer = nn.Linear( d_model, NUM_AUDIO_TOKENS + 1, bias=False ) self.rng = random.Random(0) self.num_heads = nhead self.prefix_mode = prefix_mode self.num_quantizers = num_quantizers assert num_quantizers >= 1 if num_quantizers > 1: self.nar_audio_embeddings = nn.ModuleList( [TokenEmbedding(nar_d_model, NUM_AUDIO_TOKENS + 1)] + [ TokenEmbedding(nar_d_model, NUM_AUDIO_TOKENS) for i in range(num_quantizers - 1) ] ) # W_a # PreNet if add_prenet: self.nar_text_prenet = nn.Sequential( Transpose(), nn.Conv1d( nar_d_model, nar_d_model, kernel_size=5, padding="same" ), nn.BatchNorm1d(nar_d_model), nn.ReLU(), nn.Dropout(0.5), nn.Conv1d( nar_d_model, nar_d_model, kernel_size=5, padding="same" ), nn.BatchNorm1d(nar_d_model), nn.ReLU(), nn.Dropout(0.5), nn.Conv1d( nar_d_model, nar_d_model, kernel_size=5, padding="same" ), nn.BatchNorm1d(nar_d_model), nn.ReLU(), nn.Dropout(0.5), Transpose(), nn.Linear(nar_d_model, nar_d_model), ) self.nar_audio_prenet = nn.Sequential( nn.Linear(nar_d_model, 256), nn.ReLU(), nn.Dropout(0.25), nn.Linear(256, 256), nn.ReLU(), nn.Dropout(0.25), nn.Linear(256, nar_d_model), ) else: self.nar_text_prenet = nn.Identity() self.nar_audio_prenet = nn.Identity() self.nar_text_position = SinePositionalEmbedding( nar_d_model, dropout=0.0, scale=False, alpha=False, ) self.nar_audio_position = SinePositionalEmbedding( nar_d_model, dropout=0.1, scale=False, alpha=False, ) self.nar_decoder = decoder_cls( decoder_layer_cls( nar_d_model, int(nhead * nar_scale_factor), dim_feedforward=nar_d_model * 4, dropout=0.1, batch_first=True, norm_first=norm_first, adaptive_layer_norm=True, ), num_layers=int(num_layers * nar_scale_factor), norm=AdaptiveLayerNorm( nar_d_model, norm=nn.LayerNorm(nar_d_model) ) if norm_first else None, ) self.nar_predict_layers = nn.ModuleList( [ nn.Linear(nar_d_model, NUM_AUDIO_TOKENS, bias=False) for i in range(num_quantizers - 1) ] ) self.nar_stage_embeddings = nn.ModuleList( [ TokenEmbedding(nar_d_model, 1) for i in range(num_quantizers - 1) ] ) if share_embedding: # We share the parameters of the output projection layer with the parameters of the acoustic embedding Wa # NOTE(Feiteng): In the experiment, this undermines accuracy # self.ar_predict_layer.weight = self.ar_audio_embedding.weight # We also share the parameters of the acoustic embedding layer and the output prediction layer, # which means the weights of the j-th prediction layer are the same as the (j + 1)-th acoustic embedding layer. for j in range(0, num_quantizers - 2): self.nar_predict_layers[ j ].weight = self.nar_audio_embeddings[j + 2].weight def stage_parameters(self, stage: int = 1) -> Iterator[nn.Parameter]: assert stage > 0 if stage == 1: for name, param in self.named_parameters(): if name.startswith("ar_"): print(f" AR parameter: {name}") yield param if stage == 2: for name, param in self.named_parameters(): if name.startswith("nar_"): print(f"NAR parameter: {name}") yield param def stage_named_parameters( self, stage: int = 1 ) -> Iterator[Tuple[str, nn.Parameter]]: assert stage > 0 if stage == 1: for pair in self.named_parameters(): if pair[0].startswith("ar_"): yield pair if stage == 2: for pair in self.named_parameters(): if pair[0].startswith("nar_"): yield pair def pad_y_eos(self, y, y_mask_int, eos_id): targets = F.pad(y, (0, 1), value=0) + eos_id * F.pad( y_mask_int, (0, 1), value=1 ) # inputs, targets if self.ar_audio_prepend_bos: return ( F.pad(targets[:, :-1], (1, 0), value=NUM_AUDIO_TOKENS + 1), targets, ) return targets[:, :-1], targets[:, 1:] def _prepare_prompts(self, y, y_lens, codes, nar_stage, y_prompts_codes, prefix_mode): # 5.1 For the NAR acoustic prompt tokens, we select a random segment waveform of 3 seconds # from the same utterance. # We implement this differently. if prefix_mode == 0: # no prefix prefix_len = 0 y_emb = self.nar_audio_embeddings[0](y) for j in range(1, nar_stage): # Formula (4) (5) y_emb = y_emb + self.nar_audio_embeddings[j](codes[..., j]) elif prefix_mode == 1: # prefix at begining int_low = (0.25 * y_lens.min()).type(torch.int64).item() prefix_len = torch.randint(0, int_low * 2, size=()).item() prefix_len = min(prefix_len, 225) # 24000/320 * 3s = 225 frames y_prompts = self.nar_audio_embeddings[0](y[:, :prefix_len]) y_emb = self.nar_audio_embeddings[0](y[:, prefix_len:]) for j in range(1, self.num_quantizers): y_prompts += self.nar_audio_embeddings[j]( codes[:, :prefix_len, j] ) if j < nar_stage: y_emb += self.nar_audio_embeddings[j]( codes[:, prefix_len:, j] ) y_emb = torch.concat([y_prompts, y_emb], axis=1) elif prefix_mode in [2, 4]: if prefix_mode == 2: # random prefix prefix_len = min(225, int(0.25 * y_lens.min().item())) y_prompts_codes = [] for b in range(codes.shape[0]): start = self.rng.randint(0, y_lens[b].item() - prefix_len) y_prompts_codes.append( torch.clone(codes[b, start : start + prefix_len]) ) codes[ b, start : start + prefix_len, nar_stage ] = NUM_AUDIO_TOKENS y_prompts_codes = torch.stack(y_prompts_codes, dim=0) else: prefix_len = y_prompts_codes.shape[1] y_prompts = self.nar_audio_embeddings[0](y_prompts_codes[..., 0]) y_emb = self.nar_audio_embeddings[0](y) for j in range(1, self.num_quantizers): y_prompts += self.nar_audio_embeddings[j]( y_prompts_codes[..., j] ) if j < nar_stage: y_emb += self.nar_audio_embeddings[j](codes[..., j]) y_emb = torch.concat([y_prompts, y_emb], axis=1) else: raise ValueError return y_emb, prefix_len def forward( self, x: torch.Tensor, x_lens: torch.Tensor, y: Union[torch.Tensor], y_lens: Union[torch.Tensor], reduction: str = "sum", train_stage: int = 0, **kwargs, ) -> Tuple[torch.Tensor, Union[torch.Tensor, None]]: raise NotImplementedError def inference( self, x: torch.Tensor, x_lens: torch.Tensor, y: torch.Tensor, enroll_x_lens: Union[torch.Tensor, None] = None, top_k: int = -100, temperature: float = 1.0, ) -> torch.Tensor: raise NotImplementedError def visualize( self, predicts: Tuple[torch.Tensor], batch: Dict[str, Union[List, torch.Tensor]], output_dir: str, limit: int = 4, ) -> None: raise NotImplementedError class VALLE(VALLF): """It implements https://arxiv.org/abs/2301.02111 "Neural Codec Language Models are Zero-Shot Text to Speech Synthesizers" """ def __init__( self, d_model: int, nhead: int, num_layers: int, norm_first: bool = True, add_prenet: bool = False, prefix_mode: int = 0, share_embedding: bool = True, nar_scale_factor: float = 1.0, **kwargs, ): """ Args: d_model: The number of expected features in the input (required). nhead: The number of heads in the multiheadattention models (required). num_layers: The number of sub-decoder-layers in the decoder (required). """ super(VALLE, self).__init__( d_model, nhead, num_layers, norm_first=norm_first, add_prenet=add_prenet, decoder_cls=TransformerEncoder, decoder_layer_cls=TransformerEncoderLayer, prefix_mode=prefix_mode, share_embedding=share_embedding, nar_scale_factor=nar_scale_factor, **kwargs, ) self.language_ID = { 'en': 0, 'zh': 1, 'ja': 2, } self.ar_language_embedding = TokenEmbedding(d_model, len(self.language_ID)) self.nar_language_embedding = TokenEmbedding(d_model, len(self.language_ID)) def forward( self, x: torch.Tensor, x_lens: torch.Tensor, y: Union[torch.Tensor], y_lens: Union[torch.Tensor], reduction: str = "sum", train_stage: int = 0, **kwargs, ): raise NotImplementedError def inference( self, x: torch.Tensor, x_lens: torch.Tensor, y: torch.Tensor, enroll_x_lens: torch.Tensor, top_k: int = -100, temperature: float = 1.0, prompt_language: str = None, text_language: str = None, ) -> torch.Tensor: """ Args: x: A 2-D tensor of shape (1, S). x_lens: A 1-D tensor of shape (1,). It contains the number of tokens in `x` before padding. y: A 3-D tensor of shape (1, T, 8). top_k: (`optional`) int The number of highest probability tokens to keep for top-k-filtering. Default to -100. temperature: (`optional`) float The value used to module the next token probabilities. Must be strictly positive. Default to 1.0. Returns: Return the predicted audio code matrix. """ assert x.ndim == 2, x.shape assert x_lens.ndim == 1, x_lens.shape assert y.ndim == 3, y.shape assert y.shape[0] == 1, y.shape assert torch.all(x_lens > 0) # NOTE: x has been padded in TextTokenCollater text = x x = self.ar_text_embedding(text) # Add language embedding prompt_language_id = torch.LongTensor(np.array([self.language_ID[prompt_language]])).to(x.device) if isinstance(text_language, str): text_language_id = torch.LongTensor(np.array([self.language_ID[text_language]])).to(x.device) elif isinstance(text_language, List): text_language_id = torch.LongTensor(np.array([self.language_ID[tl] for tl in text_language])).to(x.device) x[:, :enroll_x_lens, :] += self.ar_language_embedding(prompt_language_id) x[:, enroll_x_lens:, :] += self.ar_language_embedding(text_language_id) x = self.ar_text_prenet(x) x = self.ar_text_position(x) text_len = x_lens.max() prompts = y prefix_len = y.shape[1] # AR Decoder # TODO: Managing decoder steps avoid repetitive computation y = prompts[..., 0] if self.ar_audio_prepend_bos: y = F.pad(y, (1, 0), value=NUM_AUDIO_TOKENS + 1) x_len = x_lens.max() x_attn_mask = torch.zeros((x_len, x_len), dtype=torch.bool) kv_cache = None use_kv_caching = True while True: y_emb = self.ar_audio_embedding(y) y_emb = self.ar_audio_prenet(y_emb) y_pos = self.ar_audio_position(y_emb) xy_pos = torch.concat([x, y_pos], dim=1) y_len = y.shape[1] x_attn_mask_pad = F.pad( x_attn_mask, (0, y_len), value=True, ) y_attn_mask = F.pad( torch.triu( torch.ones(y_len, y_len, dtype=torch.bool), diagonal=1 ), (x_len, 0), value=False, ) xy_attn_mask = torch.concat( [x_attn_mask_pad, y_attn_mask], dim=0 ).to(y.device) if use_kv_caching and kv_cache is not None: xy_pos = xy_pos[:, [-1]] else: pass xy_dec, kv_cache = self.ar_decoder.infer( xy_pos, mask=xy_attn_mask, past_kv=kv_cache, use_cache=use_kv_caching, ) # xy_dec, _ = self.ar_decoder( # (xy_pos, None), # mask=xy_attn_mask, # ) logits = self.ar_predict_layer(xy_dec[:, -1]) samples = topk_sampling( logits, top_k=top_k, top_p=1, temperature=temperature ) if ( torch.argmax(logits, dim=-1)[0] == NUM_AUDIO_TOKENS or samples[0, 0] == NUM_AUDIO_TOKENS or (y.shape[1] - prompts.shape[1]) > x_lens.max() * 16 ): if prompts.shape[1] == y.shape[1]: raise SyntaxError( "well trained model shouldn't reach here." ) print(f"VALL-E EOS [{prompts.shape[1]} -> {y.shape[1]}]") memory_used = get_memory_usage() print(f"Current memory used: {memory_used:.2f} MB") break y = torch.concat([y, samples], dim=1) codes = [y[:, prefix_len + int(self.ar_audio_prepend_bos) :]] if self.num_quantizers == 1: return torch.stack(codes, dim=-1) # Non-AR Decoders y_emb = self.nar_audio_embeddings[0]( y[:, int(self.ar_audio_prepend_bos) :] ) if self.prefix_mode in [2, 4]: # Exclude enrolled_phonemes enrolled_len = enroll_x_lens.max().item() # SOS + Synthesis Text + EOS text = torch.concat( [ text[:, :1], text[:, enrolled_len - 1 :], ], dim=1, ) text_len = text_len - (enrolled_len - 2) assert text.shape[0] == 1 x = self.nar_text_embedding(text) # Add language embedding prompt_language_id = torch.LongTensor(np.array([self.language_ID[prompt_language]])).to(x.device) if isinstance(text_language, str): text_language_id = torch.LongTensor(np.array([self.language_ID[text_language]])).to(x.device) elif isinstance(text_language, List): text_language_id = torch.LongTensor(np.array([self.language_ID[tl] for tl in text_language])).to(x.device) x[:, :enroll_x_lens, :] += self.nar_language_embedding(prompt_language_id) x[:, enroll_x_lens:, :] += self.nar_language_embedding(text_language_id) x = self.nar_text_prenet(x) x = self.nar_text_position(x) if self.prefix_mode == 0: for i, (predict_layer, embedding_layer) in enumerate( zip( self.nar_predict_layers, self.nar_audio_embeddings[1:], ) ): y_pos = self.nar_audio_prenet(y_emb) y_pos = self.nar_audio_position(y_pos) xy_pos = torch.concat([x, y_pos], dim=1) xy_dec, _ = self.nar_decoder( (xy_pos, self.nar_stage_embeddings[i].weight) ) logits = predict_layer(xy_dec[:, text_len + prefix_len :]) samples = torch.argmax(logits, dim=-1) codes.append(samples) if i < self.num_quantizers - 2: y_emb[:, :prefix_len] += embedding_layer( prompts[..., i + 1] ) y_emb[:, prefix_len:] += embedding_layer(samples) else: for j in range(1, self.num_quantizers): y_emb[:, :prefix_len] += self.nar_audio_embeddings[j]( prompts[..., j] ) for i, (predict_layer, embedding_layer) in enumerate( zip( self.nar_predict_layers, self.nar_audio_embeddings[1:], ) ): y_pos = self.nar_audio_prenet(y_emb) y_pos = self.nar_audio_position(y_pos) xy_pos = torch.concat([x, y_pos], dim=1) xy_dec, _ = self.nar_decoder( (xy_pos, self.nar_stage_embeddings[i].weight) ) logits = predict_layer(xy_dec[:, text_len + prefix_len :]) samples = torch.argmax(logits, dim=-1) codes.append(samples) if i < self.num_quantizers - 2: y_emb[:, prefix_len:] += embedding_layer(samples) assert len(codes) == self.num_quantizers return torch.stack(codes, dim=-1) def continual( self, x: torch.Tensor, x_lens: torch.Tensor, y: torch.Tensor, ) -> torch.Tensor: """ Args: x: A 2-D tensor of shape (1, S). x_lens: A 1-D tensor of shape (1,). It contains the number of tokens in `x` before padding. y: A 3-D tensor of shape (1, T, 8). Returns: Return the predicted audio code matrix. """ assert x.ndim == 2, x.shape assert x_lens.ndim == 1, x_lens.shape assert y.ndim == 3, y.shape assert y.shape[0] == 1, y.shape assert torch.all(x_lens > 0) assert self.num_quantizers == 8 # NOTE: x has been padded in TextTokenCollater text = x x = self.ar_text_embedding(text) x = self.ar_text_prenet(x) x = self.ar_text_position(x) text_len = x_lens.max() prefix_len = min(int(y.shape[1] * 0.5), 3 * 75) # AR Decoder prompts = y[:, :prefix_len] codes = [y[:, prefix_len:, 0]] # Non-AR Decoders x = self.nar_text_embedding(text) x = self.nar_text_prenet(x) x = self.nar_text_position(x) y_emb = self.nar_audio_embeddings[0](y[..., 0]) if self.prefix_mode == 0: for i, (predict_layer, embedding_layer) in enumerate( zip( self.nar_predict_layers, self.nar_audio_embeddings[1:], ) ): y_pos = self.nar_audio_position(y_emb) y_pos = self.nar_audio_prenet(y_pos) xy_pos = torch.concat([x, y_pos], dim=1) xy_dec, _ = self.nar_decoder( (xy_pos, self.nar_stage_embeddings[i].weight) ) logits = predict_layer(xy_dec[:, text_len + prefix_len :]) samples = torch.argmax(logits, dim=-1) codes.append(samples) if i < 6: y_emb[:, :prefix_len] += embedding_layer( prompts[..., i + 1] ) y_emb[:, prefix_len:] += embedding_layer(samples) else: for j in range(1, 8): y_emb[:, :prefix_len] += self.nar_audio_embeddings[j]( prompts[..., j] ) for i, (predict_layer, embedding_layer) in enumerate( zip( self.nar_predict_layers, self.nar_audio_embeddings[1:], ) ): y_pos = self.nar_audio_prenet(y_emb) y_pos = self.nar_audio_position(y_pos) xy_pos = torch.concat([x, y_pos], dim=1) xy_dec, _ = self.nar_decoder( (xy_pos, self.nar_stage_embeddings[i].weight) ) logits = predict_layer(xy_dec[:, text_len + prefix_len :]) samples = torch.argmax(logits, dim=-1) codes.append(samples) if i < 6: y_emb[:, prefix_len:] += embedding_layer(samples) assert len(codes) == 8 return torch.stack(codes, dim=-1) # https://github.com/microsoft/unilm/blob/master/xtune/src/transformers/modeling_utils.py def top_k_top_p_filtering( logits, top_k=0, top_p=1.0, filter_value=-float("Inf"), min_tokens_to_keep=1 ): """Filter a distribution of logits using top-k and/or nucleus (top-p) filtering Args: logits: logits distribution shape (batch size, vocabulary size) if top_k > 0: keep only top k tokens with highest probability (top-k filtering). if top_p < 1.0: keep the top tokens with cumulative probability >= top_p (nucleus filtering). Nucleus filtering is described in Holtzman et al. (http://arxiv.org/abs/1904.09751) Make sure we keep at least min_tokens_to_keep per batch example in the output From: https://gist.github.com/thomwolf/1a5a29f6962089e871b94cbd09daf317 """ if top_k > 0: top_k = min( max(top_k, min_tokens_to_keep), logits.size(-1) ) # Safety check # Remove all tokens with a probability less than the last token of the top-k indices_to_remove = logits < torch.topk(logits, top_k)[0][..., -1, None] logits[indices_to_remove] = filter_value if top_p < 1.0: sorted_logits, sorted_indices = torch.sort(logits, descending=True) cumulative_probs = torch.cumsum( F.softmax(sorted_logits, dim=-1), dim=-1 ) # Remove tokens with cumulative probability above the threshold (token with 0 are kept) sorted_indices_to_remove = cumulative_probs > top_p if min_tokens_to_keep > 1: # Keep at least min_tokens_to_keep (set to min_tokens_to_keep-1 because we add the first one below) sorted_indices_to_remove[..., :min_tokens_to_keep] = 0 # Shift the indices to the right to keep also the first token above the threshold sorted_indices_to_remove[..., 1:] = sorted_indices_to_remove[ ..., :-1 ].clone() sorted_indices_to_remove[..., 0] = 0 # scatter sorted tensors to original indexing indices_to_remove = sorted_indices_to_remove.scatter( 1, sorted_indices, sorted_indices_to_remove ) logits[indices_to_remove] = filter_value return logits def topk_sampling(logits, top_k=10, top_p=1.0, temperature=1.0): # temperature: (`optional`) float # The value used to module the next token probabilities. Must be strictly positive. Default to 1.0. # top_k: (`optional`) int # The number of highest probability vocabulary tokens to keep for top-k-filtering. Between 1 and infinity. Default to 50. # top_p: (`optional`) float # The cumulative probability of parameter highest probability vocabulary tokens to keep for nucleus sampling. Must be between 0 and 1. Default to 1. # Temperature (higher temperature => more likely to sample low probability tokens) if temperature != 1.0: logits = logits / temperature # Top-p/top-k filtering logits = top_k_top_p_filtering(logits, top_k=top_k, top_p=top_p) # Sample token = torch.multinomial(F.softmax(logits, dim=-1), num_samples=1) return token