# modified from https://github.com/yangdongchao/SoundStorm/blob/master/soundstorm/s1/AR/models/t2s_model.py # reference: https://github.com/lifeiteng/vall-e import torch from tqdm import tqdm from AR.models.utils import make_pad_mask from AR.models.utils import ( topk_sampling, sample, logits_to_probs, multinomial_sample_one_no_sync, dpo_loss, make_reject_y, get_batch_logps ) from AR.modules.embedding import SinePositionalEmbedding from AR.modules.embedding import TokenEmbedding from AR.modules.transformer import LayerNorm from AR.modules.transformer import TransformerEncoder from AR.modules.transformer import TransformerEncoderLayer from torch import nn from torch.nn import functional as F from torchmetrics.classification import MulticlassAccuracy default_config = { "embedding_dim": 512, "hidden_dim": 512, "num_head": 8, "num_layers": 12, "num_codebook": 8, "p_dropout": 0.0, "vocab_size": 1024 + 1, "phoneme_vocab_size": 512, "EOS": 1024, } class Text2SemanticDecoder(nn.Module): def __init__(self, config, norm_first=False, top_k=3): super(Text2SemanticDecoder, self).__init__() self.model_dim = config["model"]["hidden_dim"] self.embedding_dim = config["model"]["embedding_dim"] self.num_head = config["model"]["head"] self.num_layers = config["model"]["n_layer"] self.norm_first = norm_first self.vocab_size = config["model"]["vocab_size"] self.phoneme_vocab_size = config["model"]["phoneme_vocab_size"] self.p_dropout = config["model"]["dropout"] self.EOS = config["model"]["EOS"] self.norm_first = norm_first assert self.EOS == self.vocab_size - 1 # should be same as num of kmeans bin # assert self.EOS == 1024 self.bert_proj = nn.Linear(1024, self.embedding_dim) self.ar_text_embedding = TokenEmbedding( self.embedding_dim, self.phoneme_vocab_size, self.p_dropout ) self.ar_text_position = SinePositionalEmbedding( self.embedding_dim, dropout=0.1, scale=False, alpha=True ) self.ar_audio_embedding = TokenEmbedding( self.embedding_dim, self.vocab_size, self.p_dropout ) self.ar_audio_position = SinePositionalEmbedding( self.embedding_dim, dropout=0.1, scale=False, alpha=True ) self.h = TransformerEncoder( TransformerEncoderLayer( d_model=self.model_dim, nhead=self.num_head, dim_feedforward=self.model_dim * 4, dropout=0.1, batch_first=True, norm_first=norm_first, ), num_layers=self.num_layers, norm=LayerNorm(self.model_dim) if norm_first else None, ) self.ar_predict_layer = nn.Linear(self.model_dim, self.vocab_size, bias=False) self.loss_fct = nn.CrossEntropyLoss(reduction="sum") self.ar_accuracy_metric = MulticlassAccuracy( self.vocab_size, top_k=top_k, average="micro", multidim_average="global", ignore_index=self.EOS, ) def make_input_data(self, x, x_lens, y, y_lens, bert_feature): x = self.ar_text_embedding(x) x = x + self.bert_proj(bert_feature.transpose(1, 2)) x = self.ar_text_position(x) x_mask = make_pad_mask(x_lens) y_mask = make_pad_mask(y_lens) y_mask_int = y_mask.type(torch.int64) codes = y.type(torch.int64) * (1 - y_mask_int) # Training # AR Decoder y, targets = self.pad_y_eos(codes, y_mask_int, eos_id=self.EOS) x_len = x_lens.max() y_len = y_lens.max() y_emb = self.ar_audio_embedding(y) y_pos = self.ar_audio_position(y_emb) xy_padding_mask = torch.concat([x_mask, y_mask], dim=1) ar_xy_padding_mask = xy_padding_mask x_attn_mask = F.pad( torch.zeros((x_len, x_len), dtype=torch.bool, device=x.device), (0, y_len), value=True, ) y_attn_mask = F.pad( torch.triu( torch.ones(y_len, y_len, dtype=torch.bool, device=x.device), diagonal=1, ), (x_len, 0), value=False, ) xy_attn_mask = torch.concat([x_attn_mask, y_attn_mask], dim=0) bsz, src_len = x.shape[0], x_len + y_len _xy_padding_mask = ( ar_xy_padding_mask.view(bsz, 1, 1, src_len) .expand(-1, self.num_head, -1, -1) .reshape(bsz * self.num_head, 1, src_len) ) xy_attn_mask = xy_attn_mask.logical_or(_xy_padding_mask) new_attn_mask = torch.zeros_like(xy_attn_mask, dtype=x.dtype) new_attn_mask.masked_fill_(xy_attn_mask, float("-inf")) xy_attn_mask = new_attn_mask # x 和完整的 y 一次性输入模型 xy_pos = torch.concat([x, y_pos], dim=1) return xy_pos, xy_attn_mask, targets def forward(self, x, x_lens, y, y_lens, bert_feature): """ x: phoneme_ids y: semantic_ids """ reject_y, reject_y_lens = make_reject_y(y, y_lens) xy_pos, xy_attn_mask, targets = self.make_input_data(x, x_lens, y, y_lens, bert_feature) xy_dec, _ = self.h( (xy_pos, None), mask=xy_attn_mask, ) x_len = x_lens.max() logits = self.ar_predict_layer(xy_dec[:, x_len:]) ###### DPO ############# reject_xy_pos, reject_xy_attn_mask, reject_targets = self.make_input_data(x, x_lens, reject_y, reject_y_lens, bert_feature) reject_xy_dec, _ = self.h( (reject_xy_pos, None), mask=reject_xy_attn_mask, ) x_len = x_lens.max() reject_logits = self.ar_predict_layer(reject_xy_dec[:, x_len:]) # loss # from feiteng: 每次 duration 越多, 梯度更新也应该更多, 所以用 sum loss_1 = F.cross_entropy(logits.permute(0, 2, 1), targets, reduction="sum") acc = self.ar_accuracy_metric(logits.permute(0, 2, 1).detach(), targets).item() A_logits, R_logits = get_batch_logps(logits, reject_logits, targets, reject_targets) loss_2, _, _ = dpo_loss(A_logits, R_logits, 0, 0, 0.2, reference_free=True) loss = loss_1 + loss_2 return loss, acc def forward_old(self, x, x_lens, y, y_lens, bert_feature): """ x: phoneme_ids y: semantic_ids """ x = self.ar_text_embedding(x) x = x + self.bert_proj(bert_feature.transpose(1, 2)) x = self.ar_text_position(x) x_mask = make_pad_mask(x_lens) y_mask = make_pad_mask(y_lens) y_mask_int = y_mask.type(torch.int64) codes = y.type(torch.int64) * (1 - y_mask_int) # Training # AR Decoder y, targets = self.pad_y_eos(codes, y_mask_int, eos_id=self.EOS) x_len = x_lens.max() y_len = y_lens.max() y_emb = self.ar_audio_embedding(y) y_pos = self.ar_audio_position(y_emb) xy_padding_mask = torch.concat([x_mask, y_mask], dim=1) ar_xy_padding_mask = xy_padding_mask x_attn_mask = F.pad( torch.zeros((x_len, x_len), dtype=torch.bool, device=x.device), (0, y_len), value=True, ) y_attn_mask = F.pad( torch.triu( torch.ones(y_len, y_len, dtype=torch.bool, device=x.device), diagonal=1, ), (x_len, 0), value=False, ) xy_attn_mask = torch.concat([x_attn_mask, y_attn_mask], dim=0) bsz, src_len = x.shape[0], x_len + y_len _xy_padding_mask = ( ar_xy_padding_mask.view(bsz, 1, 1, src_len) .expand(-1, self.num_head, -1, -1) .reshape(bsz * self.num_head, 1, src_len) ) xy_attn_mask = xy_attn_mask.logical_or(_xy_padding_mask) new_attn_mask = torch.zeros_like(xy_attn_mask, dtype=x.dtype) new_attn_mask.masked_fill_(xy_attn_mask, float("-inf")) xy_attn_mask = new_attn_mask # x 和完整的 y 一次性输入模型 xy_pos = torch.concat([x, y_pos], dim=1) xy_dec, _ = self.h( (xy_pos, None), mask=xy_attn_mask, ) logits = self.ar_predict_layer(xy_dec[:, x_len:]).permute(0, 2, 1) # loss # from feiteng: 每次 duration 越多, 梯度更新也应该更多, 所以用 sum loss = F.cross_entropy(logits, targets, reduction="sum") acc = self.ar_accuracy_metric(logits.detach(), targets).item() return loss, acc # 需要看下这个函数和 forward 的区别以及没有 semantic 的时候 prompts 输入什么 def infer( self, x, x_lens, prompts, bert_feature, top_k: int = -100, early_stop_num: int = -1, temperature: float = 1.0, ): x = self.ar_text_embedding(x) x = x + self.bert_proj(bert_feature.transpose(1, 2)) x = self.ar_text_position(x) # AR Decoder y = prompts prefix_len = y.shape[1] x_len = x.shape[1] x_attn_mask = torch.zeros((x_len, x_len), dtype=torch.bool) stop = False for _ in tqdm(range(1500)): y_emb = self.ar_audio_embedding(y) y_pos = self.ar_audio_position(y_emb) # x 和逐渐增长的 y 一起输入给模型 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 ) xy_dec, _ = self.h( (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=1.0, temperature=temperature ) if early_stop_num != -1 and (y.shape[1] - prefix_len) > early_stop_num: print("use early stop num:", early_stop_num) stop = True if torch.argmax(logits, dim=-1)[0] == self.EOS or samples[0, 0] == self.EOS: # print(torch.argmax(logits, dim=-1)[0] == self.EOS, samples[0, 0] == self.EOS) stop = True if stop: if prompts.shape[1] == y.shape[1]: y = torch.concat([y, torch.zeros_like(samples)], dim=1) print("bad zero prediction") print(f"T2S Decoding EOS [{prefix_len} -> {y.shape[1]}]") break # 本次生成的 semantic_ids 和之前的 y 构成新的 y # print(samples.shape)#[1,1]#第一个1是bs # import os # os._exit(2333) y = torch.concat([y, samples], dim=1) return y 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 ) # 错位 return targets[:, :-1], targets[:, 1:] def infer_panel( self, x, #####全部文本token x_lens, prompts, ####参考音频token bert_feature, top_k: int = -100, top_p: int = 100, early_stop_num: int = -1, temperature: float = 1.0, ): x = self.ar_text_embedding(x) x = x + self.bert_proj(bert_feature.transpose(1, 2)) x = self.ar_text_position(x) # AR Decoder y = prompts x_len = x.shape[1] x_attn_mask = torch.zeros((x_len, x_len), dtype=torch.bool) stop = False # print(1111111,self.num_layers) cache = { "all_stage": self.num_layers, "k": [None] * self.num_layers, ###根据配置自己手写 "v": [None] * self.num_layers, # "xy_pos":None,##y_pos位置编码每次都不一样的没法缓存,每次都要重新拼xy_pos.主要还是写法原因,其实是可以历史统一一样的,但也没啥计算量就不管了 "y_emb": None, ##只需要对最新的samples求emb,再拼历史的就行 # "logits":None,###原版就已经只对结尾求再拼接了,不用管 # "xy_dec":None,###不需要,本来只需要最后一个做logits "first_infer": 1, "stage": 0, } ################### first step ########################## if y is not None: y_emb = self.ar_audio_embedding(y) y_len = y_emb.shape[1] prefix_len = y.shape[1] y_pos = self.ar_audio_position(y_emb) xy_pos = torch.concat([x, y_pos], dim=1) cache["y_emb"] = y_emb ref_free = False else: y_emb = None y_len = 0 prefix_len = 0 y_pos = None xy_pos = x y = torch.zeros(x.shape[0], 0, dtype=torch.int, device=x.device) ref_free = True x_attn_mask_pad = F.pad( x_attn_mask, (0, y_len), ###xx的纯0扩展到xx纯0+xy纯1,(x,x+y) value=True, ) y_attn_mask = F.pad( ###yy的右上1扩展到左边xy的0,(y,x+y) 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( x.device ) for idx in tqdm(range(1500)): xy_dec, _ = self.h((xy_pos, None), mask=xy_attn_mask, cache=cache) logits = self.ar_predict_layer( xy_dec[:, -1] ) ##不用改,如果用了cache的默认就是只有一帧,取最后一帧一样的 # samples = topk_sampling(logits, top_k=top_k, top_p=1.0, temperature=temperature) if(idx==0):###第一次跑不能EOS否则没有了 logits = logits[:, :-1] ###刨除1024终止符号的概率 samples = sample( logits[0], y, top_k=top_k, top_p=top_p, repetition_penalty=1.35, temperature=temperature )[0].unsqueeze(0) # 本次生成的 semantic_ids 和之前的 y 构成新的 y # print(samples.shape)#[1,1]#第一个1是bs y = torch.concat([y, samples], dim=1) if early_stop_num != -1 and (y.shape[1] - prefix_len) > early_stop_num: print("use early stop num:", early_stop_num) stop = True if torch.argmax(logits, dim=-1)[0] == self.EOS or samples[0, 0] == self.EOS: # print(torch.argmax(logits, dim=-1)[0] == self.EOS, samples[0, 0] == self.EOS) stop = True if stop: # if prompts.shape[1] == y.shape[1]: # y = torch.concat([y, torch.zeros_like(samples)], dim=1) # print("bad zero prediction") if y.shape[1]==0: y = torch.concat([y, torch.zeros_like(samples)], dim=1) print("bad zero prediction") print(f"T2S Decoding EOS [{prefix_len} -> {y.shape[1]}]") break ####################### update next step ################################### cache["first_infer"] = 0 if cache["y_emb"] is not None: y_emb = torch.cat( [cache["y_emb"], self.ar_audio_embedding(y[:, -1:])], dim = 1 ) cache["y_emb"] = y_emb y_pos = self.ar_audio_position(y_emb) xy_pos = y_pos[:, -1:] else: y_emb = self.ar_audio_embedding(y[:, -1:]) cache["y_emb"] = y_emb y_pos = self.ar_audio_position(y_emb) xy_pos = y_pos y_len = y_pos.shape[1] ###最右边一列(是错的) # xy_attn_mask=torch.ones((1, x_len+y_len), dtype=torch.bool,device=xy_pos.device) # xy_attn_mask[:,-1]=False ###最下面一行(是对的) xy_attn_mask = torch.zeros( (1, x_len + y_len), dtype=torch.bool, device=xy_pos.device ) if ref_free: return y[:, :-1], 0 return y[:, :-1], idx-1