TTS-GPT-SoVITS / AR /models /t2s_model.py
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0317-165840
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# 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