Onnx导出注意事项

重要的事情说三遍:导出Onnx时,请重新克隆整个仓库!!!导出Onnx时,请重新克隆整个仓库!!!导出Onnx时,请重新克隆整个仓库!!!
32k
Ναρουσέ·μ·γιουμεμί·Χινακάννα 3 years ago committed by 大蒟蒻
commit d0a2ca4497

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# SoftVC VITS Singing Voice Conversion
## Updates
> According to incomplete statistics, it seems that training with multiple speakers may lead to **worsened leaking of voice timbre**. It is not recommended to train models with more than 5 speakers. The current suggestion is to try to train models with only a single speaker if you want to achieve a voice timbre that is more similar to the target.
> Fixed the issue with unwanted staccato, improving audio quality by a decent amount.\
> The 2.0 version has been moved to the 2.0 branch.\
> Version 3.0 uses the code structure of FreeVC, which isn't compatible with older versions.\
> Compared to [DiffSVC](https://github.com/prophesier/diff-svc) , diffsvc performs much better when the training data is of extremely high quality, but this repository may perform better on datasets with lower quality. Additionally, this repository is much faster in terms of inference speed compared to diffsvc.
## Model Overview
A singing voice coversion (SVC) model, using the SoftVC encoder to extract features from the input audio, sent into VITS along with the F0 to replace the original input to acheive a voice conversion effect. Additionally, changing the vocoder to [NSF HiFiGAN](https://github.com/openvpi/DiffSinger/tree/refactor/modules/nsf_hifigan) to fix the issue with unwanted staccato.
## Notice
+ The current branch is the 32kHz version, which requires less vram during inferencing, as well as faster inferencing speeds, and datasets for said branch take up less disk space. Thus the 32 kHz branch is recommended for use.
+ If you want to train 48 kHz variant models, switch to the [main branch](https://github.com/innnky/so-vits-svc/tree/main).
## Colab notebook script for dataset creation and training.
[colab training notebook](https://colab.research.google.com/drive/1rCUOOVG7-XQlVZuWRAj5IpGrMM8t07pE?usp=sharing)
## Required models
+ soft vc hubert[hubert-soft-0d54a1f4.pt](https://github.com/bshall/hubert/releases/download/v0.1/hubert-soft-0d54a1f4.pt)
+ Place under `hubert`.
+ Pretrained models [G_0.pth](https://huggingface.co/innnky/sovits_pretrained/resolve/main/G_0.pth) and [D_0.pth](https://huggingface.co/innnky/sovits_pretrained/resolve/main/D_0.pth)
+ Place under `logs/32k`.
+ Pretrained models are required, because from experiments, training from scratch can be rather unpredictable to say the least, and training with a pretrained model can greatly improve training speeds.
+ The pretrained model includes云灏, 即霜, 辉宇·星AI, 派蒙, and 绫地宁宁, covering the common ranges of both male and female voices, and so it can be seen as a rather universal pretrained model.
+ The pretrained model exludes the `optimizer speaker_embedding` section, rendering it only usable for pretraining and incapable of inferencing with.
```shell
# For simple downloading.
# hubert
wget -P hubert/ https://github.com/bshall/hubert/releases/download/v0.1/hubert-soft-0d54a1f4.pt
# G&D pretrained models
wget -P logs/32k/ https://huggingface.co/innnky/sovits_pretrained/resolve/main/G_0.pth
wget -P logs/32k/ https://huggingface.co/innnky/sovits_pretrained/resolve/main/D_0.pth
```
## Dataset preparation
All that is required is that the data be put under the `dataset_raw` folder in the structure format provided below.
```shell
dataset_raw
├───speaker0
│ ├───xxx1-xxx1.wav
│ ├───...
│ └───Lxx-0xx8.wav
└───speaker1
├───xx2-0xxx2.wav
├───...
└───xxx7-xxx007.wav
```
## Data pre-processing.
1. Resample to 32khz
```shell
python resample.py
```
2. Automatically sort out training set, validation set, test set, and automatically generate configuration files.
```shell
python preprocess_flist_config.py
# Notice.
# The n_speakers value in the config will be set automatically according to the amount of speakers in the dataset.
# To reserve space for additionally added speakers in the dataset, the n_speakers value will be be set to twice the actual amount.
# If you want even more space for adding more data, you can edit the n_speakers value in the config after runing this step.
# This can not be changed after training starts.
```
3. Generate hubert and F0 features/
```shell
python preprocess_hubert_f0.py
```
After running the step above, the `dataset` folder will contain all the pre-processed data, you can delete the `dataset_raw` folder after that.
## Training.
```shell
python train.py -c configs/config.json -m 32k
```
## Inferencing.
Use [inference_main.py](inference_main.py)
+ Edit `model_path` to your newest checkpoint.
+ Place the input audio under the `raw` folder.
+ Change `clean_names` to the output file name.
+ Use `trans` to edit the pitch shifting amount (semitones).
+ Change `spk_list` to the speaker name.

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MIT License
Copyright (c) 2021 Jingyi Li
Permission is hereby granted, free of charge, to any person obtaining a copy
of this software and associated documentation files (the "Software"), to deal
in the Software without restriction, including without limitation the rights
to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
copies of the Software, and to permit persons to whom the Software is
furnished to do so, subject to the following conditions:
The above copyright notice and this permission notice shall be included in all
copies or substantial portions of the Software.
THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
SOFTWARE.

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# SoftVC VITS Singing Voice Conversion
## English docs
[英语资料](Eng_docs.md)
## Update
> 据不完全统计,多说话人似乎会导致**音色泄漏加重**不建议训练超过5人的模型目前的建议是如果想炼出来更像目标音色**尽可能炼单说话人的**\
> 断音问题已解决,音质提升了不少\
> 2.0版本已经移至 sovits_2.0分支\
> 3.0版本使用FreeVC的代码结构与旧版本不通用\
> 与[DiffSVC](https://github.com/prophesier/diff-svc) 相比在训练数据质量非常高时diffsvc有着更好的表现对于质量差一些的数据集本仓库可能会有更好的表现此外本仓库推理速度上比diffsvc快很多
## 模型简介
歌声音色转换模型通过SoftVC内容编码器提取源音频语音特征与F0同时输入VITS替换原本的文本输入达到歌声转换的效果。同时更换声码器为 [NSF HiFiGAN](https://github.com/openvpi/DiffSinger/tree/refactor/modules/nsf_hifigan) 解决断音问题
## 注意
+ 当前分支是32khz版本的分支32khz模型推理更快显存占用大幅减小数据集所占硬盘空间也大幅降低推荐训练该版本模型
+ 如果要训练48khz的模型请切换到[main分支](https://github.com/innnky/so-vits-svc/tree/main)
## 预先下载的模型文件
+ soft vc hubert[hubert-soft-0d54a1f4.pt](https://github.com/bshall/hubert/releases/download/v0.1/hubert-soft-0d54a1f4.pt)
+ 放在hubert目录下
+ 预训练底模文件 [G_0.pth](https://huggingface.co/innnky/sovits_pretrained/resolve/main/G_0.pth) 与 [D_0.pth](https://huggingface.co/innnky/sovits_pretrained/resolve/main/D_0.pth)
+ 放在logs/32k 目录下
+ 预训练底模为必选项,因为据测试从零开始训练有概率不收敛,同时底模也能加快训练速度
+ 预训练底模训练数据集包含云灏 即霜 辉宇·星AI 派蒙 绫地宁宁,覆盖男女生常见音域,可以认为是相对通用的底模
+ 底模删除了optimizer speaker_embedding 等无关权重, 只可以用于初始化训练,无法用于推理
+ 该底模和48khz底模通用
```shell
# 一键下载
# hubert
wget -P hubert/ https://github.com/bshall/hubert/releases/download/v0.1/hubert-soft-0d54a1f4.pt
# G与D预训练模型
wget -P logs/32k/ https://huggingface.co/innnky/sovits_pretrained/resolve/main/G_0.pth
wget -P logs/32k/ https://huggingface.co/innnky/sovits_pretrained/resolve/main/D_0.pth
```
## colab一键数据集制作、训练脚本
[一键colab](https://colab.research.google.com/drive/1_-gh9i-wCPNlRZw6pYF-9UufetcVrGBX?usp=sharing)
## 数据集准备
仅需要以以下文件结构将数据集放入dataset_raw目录即可
```shell
dataset_raw
├───speaker0
│ ├───xxx1-xxx1.wav
│ ├───...
│ └───Lxx-0xx8.wav
└───speaker1
├───xx2-0xxx2.wav
├───...
└───xxx7-xxx007.wav
```
## 数据预处理
1. 重采样至 32khz
```shell
python resample.py
```
2. 自动划分训练集 验证集 测试集 以及自动生成配置文件
```shell
python preprocess_flist_config.py
# 注意
# 自动生成的配置文件中说话人数量n_speakers会自动按照数据集中的人数而定
# 为了给之后添加说话人留下一定空间n_speakers自动设置为 当前数据集人数乘2
# 如果想多留一些空位可以在此步骤后 自行修改生成的config.json中n_speakers数量
# 一旦模型开始训练后此项不可再更改
```
3. 生成hubert与f0
```shell
python preprocess_hubert_f0.py
```
执行完以上步骤后 dataset 目录便是预处理完成的数据可以删除dataset_raw文件夹了
## 训练
```shell
python train.py -c configs/config.json -m 32k
```
## 推理
使用 [inference_main.py](inference_main.py)
+ 更改model_path为你自己训练的最新模型记录点
+ 将待转换的音频放在raw文件夹下
+ clean_names 写待转换的音频名称
+ trans 填写变调半音数量
+ spk_list 填写合成的说话人名称
## Onnx导出
### 重要的事情说三遍导出Onnx时请重新克隆整个仓库导出Onnx时请重新克隆整个仓库导出Onnx时请重新克隆整个仓库
使用 [onnx_export.py](onnx_export.py)
+ 新建文件夹checkpoints 并打开
+ 在checkpoints文件夹中新建一个文件夹作为项目文件夹文件夹名为你的项目名称
+ 将你的模型更名为model.pth配置文件更名为config.json并放置到刚才创建的文件夹下
+ 将 [onnx_export.py](onnx_export.py) 中path = "NyaruTaffy" 的 "NyaruTaffy" 修改为你的项目名称
+ 运行 [onnx_export.py](onnx_export.py)
+ 等待执行完毕在你的项目文件夹下会生成一个model.onnx即为导出的模型
+ 注意若想导出48K模型请按照以下步骤修改文件或者直接使用48K.py
+ 请打开[model_onnx.py](model_onnx.py)将其中最后一个class的hps中32000改为48000
+ 请打开[nvSTFT](/vdecoder/hifigan/nvSTFT.py)将其中所有32000改为48000
### Onnx模型支持的UI
+ [MoeSS](https://github.com/NaruseMioShirakana/MoeSS)
+ 我去除了所有的训练用函数和一切复杂的转置一行都没有保留因为我认为只有去除了这些东西才知道你用的是Onnx
## GradioWebUI
使用 [sovits_gradio.py](sovits_gradio.py)
+ 新建文件夹checkpoints 并打开
+ 在checkpoints文件夹中新建一个文件夹作为项目文件夹文件夹名为你的项目名称
+ 将你的模型更名为model.pth配置文件更名为config.json并放置到刚才创建的文件夹下
+ 运行 [sovits_gradio.py](sovits_gradio.py)

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import os
import argparse
from tqdm import tqdm
from random import shuffle
import json
if __name__ == "__main__":
parser = argparse.ArgumentParser()
parser.add_argument("--train_list", type=str, default="./filelists/train.txt", help="path to train list")
parser.add_argument("--val_list", type=str, default="./filelists/val.txt", help="path to val list")
parser.add_argument("--test_list", type=str, default="./filelists/test.txt", help="path to test list")
parser.add_argument("--source_dir", type=str, default="./dataset/32k", help="path to source dir")
args = parser.parse_args()
previous_config = json.load(open("configs/config.json", "rb"))
train = []
val = []
test = []
idx = 0
spk_dict = previous_config["spk"]
spk_id = max([i for i in spk_dict.values()]) + 1
for speaker in tqdm(os.listdir(args.source_dir)):
if speaker not in spk_dict.keys():
spk_dict[speaker] = spk_id
spk_id += 1
wavs = [os.path.join(args.source_dir, speaker, i)for i in os.listdir(os.path.join(args.source_dir, speaker))]
wavs = [i for i in wavs if i.endswith("wav")]
shuffle(wavs)
train += wavs[2:-10]
val += wavs[:2]
test += wavs[-10:]
assert previous_config["model"]["n_speakers"] > len(spk_dict.keys())
shuffle(train)
shuffle(val)
shuffle(test)
print("Writing", args.train_list)
with open(args.train_list, "w") as f:
for fname in tqdm(train):
wavpath = fname
f.write(wavpath + "\n")
print("Writing", args.val_list)
with open(args.val_list, "w") as f:
for fname in tqdm(val):
wavpath = fname
f.write(wavpath + "\n")
print("Writing", args.test_list)
with open(args.test_list, "w") as f:
for fname in tqdm(test):
wavpath = fname
f.write(wavpath + "\n")
previous_config["spk"] = spk_dict
print("Writing configs/config.json")
with open("configs/config.json", "w") as f:
json.dump(previous_config, f, indent=2)

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import copy
import math
import numpy as np
import torch
from torch import nn
from torch.nn import functional as F
import commons
import modules
from modules import LayerNorm
class Encoder(nn.Module):
def __init__(self, hidden_channels, filter_channels, n_heads, n_layers, kernel_size=1, p_dropout=0., window_size=4, **kwargs):
super().__init__()
self.hidden_channels = hidden_channels
self.filter_channels = filter_channels
self.n_heads = n_heads
self.n_layers = n_layers
self.kernel_size = kernel_size
self.p_dropout = p_dropout
self.window_size = window_size
self.drop = nn.Dropout(p_dropout)
self.attn_layers = nn.ModuleList()
self.norm_layers_1 = nn.ModuleList()
self.ffn_layers = nn.ModuleList()
self.norm_layers_2 = nn.ModuleList()
for i in range(self.n_layers):
self.attn_layers.append(MultiHeadAttention(hidden_channels, hidden_channels, n_heads, p_dropout=p_dropout, window_size=window_size))
self.norm_layers_1.append(LayerNorm(hidden_channels))
self.ffn_layers.append(FFN(hidden_channels, hidden_channels, filter_channels, kernel_size, p_dropout=p_dropout))
self.norm_layers_2.append(LayerNorm(hidden_channels))
def forward(self, x, x_mask):
attn_mask = x_mask.unsqueeze(2) * x_mask.unsqueeze(-1)
x = x * x_mask
for i in range(self.n_layers):
y = self.attn_layers[i](x, x, attn_mask)
y = self.drop(y)
x = self.norm_layers_1[i](x + y)
y = self.ffn_layers[i](x, x_mask)
y = self.drop(y)
x = self.norm_layers_2[i](x + y)
x = x * x_mask
return x
class Decoder(nn.Module):
def __init__(self, hidden_channels, filter_channels, n_heads, n_layers, kernel_size=1, p_dropout=0., proximal_bias=False, proximal_init=True, **kwargs):
super().__init__()
self.hidden_channels = hidden_channels
self.filter_channels = filter_channels
self.n_heads = n_heads
self.n_layers = n_layers
self.kernel_size = kernel_size
self.p_dropout = p_dropout
self.proximal_bias = proximal_bias
self.proximal_init = proximal_init
self.drop = nn.Dropout(p_dropout)
self.self_attn_layers = nn.ModuleList()
self.norm_layers_0 = nn.ModuleList()
self.encdec_attn_layers = nn.ModuleList()
self.norm_layers_1 = nn.ModuleList()
self.ffn_layers = nn.ModuleList()
self.norm_layers_2 = nn.ModuleList()
for i in range(self.n_layers):
self.self_attn_layers.append(MultiHeadAttention(hidden_channels, hidden_channels, n_heads, p_dropout=p_dropout, proximal_bias=proximal_bias, proximal_init=proximal_init))
self.norm_layers_0.append(LayerNorm(hidden_channels))
self.encdec_attn_layers.append(MultiHeadAttention(hidden_channels, hidden_channels, n_heads, p_dropout=p_dropout))
self.norm_layers_1.append(LayerNorm(hidden_channels))
self.ffn_layers.append(FFN(hidden_channels, hidden_channels, filter_channels, kernel_size, p_dropout=p_dropout, causal=True))
self.norm_layers_2.append(LayerNorm(hidden_channels))
def forward(self, x, x_mask, h, h_mask):
"""
x: decoder input
h: encoder output
"""
self_attn_mask = commons.subsequent_mask(x_mask.size(2)).to(device=x.device, dtype=x.dtype)
encdec_attn_mask = h_mask.unsqueeze(2) * x_mask.unsqueeze(-1)
x = x * x_mask
for i in range(self.n_layers):
y = self.self_attn_layers[i](x, x, self_attn_mask)
y = self.drop(y)
x = self.norm_layers_0[i](x + y)
y = self.encdec_attn_layers[i](x, h, encdec_attn_mask)
y = self.drop(y)
x = self.norm_layers_1[i](x + y)
y = self.ffn_layers[i](x, x_mask)
y = self.drop(y)
x = self.norm_layers_2[i](x + y)
x = x * x_mask
return x
class MultiHeadAttention(nn.Module):
def __init__(self, channels, out_channels, n_heads, p_dropout=0., window_size=None, heads_share=True, block_length=None, proximal_bias=False, proximal_init=False):
super().__init__()
assert channels % n_heads == 0
self.channels = channels
self.out_channels = out_channels
self.n_heads = n_heads
self.p_dropout = p_dropout
self.window_size = window_size
self.heads_share = heads_share
self.block_length = block_length
self.proximal_bias = proximal_bias
self.proximal_init = proximal_init
self.attn = None
self.k_channels = channels // n_heads
self.conv_q = nn.Conv1d(channels, channels, 1)
self.conv_k = nn.Conv1d(channels, channels, 1)
self.conv_v = nn.Conv1d(channels, channels, 1)
self.conv_o = nn.Conv1d(channels, out_channels, 1)
self.drop = nn.Dropout(p_dropout)
if window_size is not None:
n_heads_rel = 1 if heads_share else n_heads
rel_stddev = self.k_channels**-0.5
self.emb_rel_k = nn.Parameter(torch.randn(n_heads_rel, window_size * 2 + 1, self.k_channels) * rel_stddev)
self.emb_rel_v = nn.Parameter(torch.randn(n_heads_rel, window_size * 2 + 1, self.k_channels) * rel_stddev)
nn.init.xavier_uniform_(self.conv_q.weight)
nn.init.xavier_uniform_(self.conv_k.weight)
nn.init.xavier_uniform_(self.conv_v.weight)
if proximal_init:
with torch.no_grad():
self.conv_k.weight.copy_(self.conv_q.weight)
self.conv_k.bias.copy_(self.conv_q.bias)
def forward(self, x, c, attn_mask=None):
q = self.conv_q(x)
k = self.conv_k(c)
v = self.conv_v(c)
x, self.attn = self.attention(q, k, v, mask=attn_mask)
x = self.conv_o(x)
return x
def attention(self, query, key, value, mask=None):
# reshape [b, d, t] -> [b, n_h, t, d_k]
b, d, t_s, t_t = (*key.size(), query.size(2))
query = query.view(b, self.n_heads, self.k_channels, t_t).transpose(2, 3)
key = key.view(b, self.n_heads, self.k_channels, t_s).transpose(2, 3)
value = value.view(b, self.n_heads, self.k_channels, t_s).transpose(2, 3)
scores = torch.matmul(query / math.sqrt(self.k_channels), key.transpose(-2, -1))
if self.window_size is not None:
assert t_s == t_t, "Relative attention is only available for self-attention."
key_relative_embeddings = self._get_relative_embeddings(self.emb_rel_k, t_s)
rel_logits = self._matmul_with_relative_keys(query /math.sqrt(self.k_channels), key_relative_embeddings)
scores_local = self._relative_position_to_absolute_position(rel_logits)
scores = scores + scores_local
if self.proximal_bias:
assert t_s == t_t, "Proximal bias is only available for self-attention."
scores = scores + self._attention_bias_proximal(t_s).to(device=scores.device, dtype=scores.dtype)
if mask is not None:
scores = scores.masked_fill(mask == 0, -1e4)
if self.block_length is not None:
assert t_s == t_t, "Local attention is only available for self-attention."
block_mask = torch.ones_like(scores).triu(-self.block_length).tril(self.block_length)
scores = scores.masked_fill(block_mask == 0, -1e4)
p_attn = F.softmax(scores, dim=-1) # [b, n_h, t_t, t_s]
p_attn = self.drop(p_attn)
output = torch.matmul(p_attn, value)
if self.window_size is not None:
relative_weights = self._absolute_position_to_relative_position(p_attn)
value_relative_embeddings = self._get_relative_embeddings(self.emb_rel_v, t_s)
output = output + self._matmul_with_relative_values(relative_weights, value_relative_embeddings)
output = output.transpose(2, 3).contiguous().view(b, d, t_t) # [b, n_h, t_t, d_k] -> [b, d, t_t]
return output, p_attn
def _matmul_with_relative_values(self, x, y):
"""
x: [b, h, l, m]
y: [h or 1, m, d]
ret: [b, h, l, d]
"""
ret = torch.matmul(x, y.unsqueeze(0))
return ret
def _matmul_with_relative_keys(self, x, y):
"""
x: [b, h, l, d]
y: [h or 1, m, d]
ret: [b, h, l, m]
"""
ret = torch.matmul(x, y.unsqueeze(0).transpose(-2, -1))
return ret
def _get_relative_embeddings(self, relative_embeddings, length):
max_relative_position = 2 * self.window_size + 1
# Pad first before slice to avoid using cond ops.
pad_length = max(length - (self.window_size + 1), 0)
slice_start_position = max((self.window_size + 1) - length, 0)
slice_end_position = slice_start_position + 2 * length - 1
if pad_length > 0:
padded_relative_embeddings = F.pad(
relative_embeddings,
commons.convert_pad_shape([[0, 0], [pad_length, pad_length], [0, 0]]))
else:
padded_relative_embeddings = relative_embeddings
used_relative_embeddings = padded_relative_embeddings[:,slice_start_position:slice_end_position]
return used_relative_embeddings
def _relative_position_to_absolute_position(self, x):
"""
x: [b, h, l, 2*l-1]
ret: [b, h, l, l]
"""
batch, heads, length, _ = x.size()
# Concat columns of pad to shift from relative to absolute indexing.
x = F.pad(x, commons.convert_pad_shape([[0,0],[0,0],[0,0],[0,1]]))
# Concat extra elements so to add up to shape (len+1, 2*len-1).
x_flat = x.view([batch, heads, length * 2 * length])
x_flat = F.pad(x_flat, commons.convert_pad_shape([[0,0],[0,0],[0,length-1]]))
# Reshape and slice out the padded elements.
x_final = x_flat.view([batch, heads, length+1, 2*length-1])[:, :, :length, length-1:]
return x_final
def _absolute_position_to_relative_position(self, x):
"""
x: [b, h, l, l]
ret: [b, h, l, 2*l-1]
"""
batch, heads, length, _ = x.size()
# padd along column
x = F.pad(x, commons.convert_pad_shape([[0, 0], [0, 0], [0, 0], [0, length-1]]))
x_flat = x.view([batch, heads, length**2 + length*(length -1)])
# add 0's in the beginning that will skew the elements after reshape
x_flat = F.pad(x_flat, commons.convert_pad_shape([[0, 0], [0, 0], [length, 0]]))
x_final = x_flat.view([batch, heads, length, 2*length])[:,:,:,1:]
return x_final
def _attention_bias_proximal(self, length):
"""Bias for self-attention to encourage attention to close positions.
Args:
length: an integer scalar.
Returns:
a Tensor with shape [1, 1, length, length]
"""
r = torch.arange(length, dtype=torch.float32)
diff = torch.unsqueeze(r, 0) - torch.unsqueeze(r, 1)
return torch.unsqueeze(torch.unsqueeze(-torch.log1p(torch.abs(diff)), 0), 0)
class FFN(nn.Module):
def __init__(self, in_channels, out_channels, filter_channels, kernel_size, p_dropout=0., activation=None, causal=False):
super().__init__()
self.in_channels = in_channels
self.out_channels = out_channels
self.filter_channels = filter_channels
self.kernel_size = kernel_size
self.p_dropout = p_dropout
self.activation = activation
self.causal = causal
if causal:
self.padding = self._causal_padding
else:
self.padding = self._same_padding
self.conv_1 = nn.Conv1d(in_channels, filter_channels, kernel_size)
self.conv_2 = nn.Conv1d(filter_channels, out_channels, kernel_size)
self.drop = nn.Dropout(p_dropout)
def forward(self, x, x_mask):
x = self.conv_1(self.padding(x * x_mask))
if self.activation == "gelu":
x = x * torch.sigmoid(1.702 * x)
else:
x = torch.relu(x)
x = self.drop(x)
x = self.conv_2(self.padding(x * x_mask))
return x * x_mask
def _causal_padding(self, x):
if self.kernel_size == 1:
return x
pad_l = self.kernel_size - 1
pad_r = 0
padding = [[0, 0], [0, 0], [pad_l, pad_r]]
x = F.pad(x, commons.convert_pad_shape(padding))
return x
def _same_padding(self, x):
if self.kernel_size == 1:
return x
pad_l = (self.kernel_size - 1) // 2
pad_r = self.kernel_size // 2
padding = [[0, 0], [0, 0], [pad_l, pad_r]]
x = F.pad(x, commons.convert_pad_shape(padding))
return x

@ -0,0 +1,188 @@
import math
import numpy as np
import torch
from torch import nn
from torch.nn import functional as F
def slice_pitch_segments(x, ids_str, segment_size=4):
ret = torch.zeros_like(x[:, :segment_size])
for i in range(x.size(0)):
idx_str = ids_str[i]
idx_end = idx_str + segment_size
ret[i] = x[i, idx_str:idx_end]
return ret
def rand_slice_segments_with_pitch(x, pitch, x_lengths=None, segment_size=4):
b, d, t = x.size()
if x_lengths is None:
x_lengths = t
ids_str_max = x_lengths - segment_size + 1
ids_str = (torch.rand([b]).to(device=x.device) * ids_str_max).to(dtype=torch.long)
ret = slice_segments(x, ids_str, segment_size)
ret_pitch = slice_pitch_segments(pitch, ids_str, segment_size)
return ret, ret_pitch, ids_str
def init_weights(m, mean=0.0, std=0.01):
classname = m.__class__.__name__
if classname.find("Conv") != -1:
m.weight.data.normal_(mean, std)
def get_padding(kernel_size, dilation=1):
return int((kernel_size*dilation - dilation)/2)
def convert_pad_shape(pad_shape):
l = pad_shape[::-1]
pad_shape = [item for sublist in l for item in sublist]
return pad_shape
def intersperse(lst, item):
result = [item] * (len(lst) * 2 + 1)
result[1::2] = lst
return result
def kl_divergence(m_p, logs_p, m_q, logs_q):
"""KL(P||Q)"""
kl = (logs_q - logs_p) - 0.5
kl += 0.5 * (torch.exp(2. * logs_p) + ((m_p - m_q)**2)) * torch.exp(-2. * logs_q)
return kl
def rand_gumbel(shape):
"""Sample from the Gumbel distribution, protect from overflows."""
uniform_samples = torch.rand(shape) * 0.99998 + 0.00001
return -torch.log(-torch.log(uniform_samples))
def rand_gumbel_like(x):
g = rand_gumbel(x.size()).to(dtype=x.dtype, device=x.device)
return g
def slice_segments(x, ids_str, segment_size=4):
ret = torch.zeros_like(x[:, :, :segment_size])
for i in range(x.size(0)):
idx_str = ids_str[i]
idx_end = idx_str + segment_size
ret[i] = x[i, :, idx_str:idx_end]
return ret
def rand_slice_segments(x, x_lengths=None, segment_size=4):
b, d, t = x.size()
if x_lengths is None:
x_lengths = t
ids_str_max = x_lengths - segment_size + 1
ids_str = (torch.rand([b]).to(device=x.device) * ids_str_max).to(dtype=torch.long)
ret = slice_segments(x, ids_str, segment_size)
return ret, ids_str
def rand_spec_segments(x, x_lengths=None, segment_size=4):
b, d, t = x.size()
if x_lengths is None:
x_lengths = t
ids_str_max = x_lengths - segment_size
ids_str = (torch.rand([b]).to(device=x.device) * ids_str_max).to(dtype=torch.long)
ret = slice_segments(x, ids_str, segment_size)
return ret, ids_str
def get_timing_signal_1d(
length, channels, min_timescale=1.0, max_timescale=1.0e4):
position = torch.arange(length, dtype=torch.float)
num_timescales = channels // 2
log_timescale_increment = (
math.log(float(max_timescale) / float(min_timescale)) /
(num_timescales - 1))
inv_timescales = min_timescale * torch.exp(
torch.arange(num_timescales, dtype=torch.float) * -log_timescale_increment)
scaled_time = position.unsqueeze(0) * inv_timescales.unsqueeze(1)
signal = torch.cat([torch.sin(scaled_time), torch.cos(scaled_time)], 0)
signal = F.pad(signal, [0, 0, 0, channels % 2])
signal = signal.view(1, channels, length)
return signal
def add_timing_signal_1d(x, min_timescale=1.0, max_timescale=1.0e4):
b, channels, length = x.size()
signal = get_timing_signal_1d(length, channels, min_timescale, max_timescale)
return x + signal.to(dtype=x.dtype, device=x.device)
def cat_timing_signal_1d(x, min_timescale=1.0, max_timescale=1.0e4, axis=1):
b, channels, length = x.size()
signal = get_timing_signal_1d(length, channels, min_timescale, max_timescale)
return torch.cat([x, signal.to(dtype=x.dtype, device=x.device)], axis)
def subsequent_mask(length):
mask = torch.tril(torch.ones(length, length)).unsqueeze(0).unsqueeze(0)
return mask
@torch.jit.script
def fused_add_tanh_sigmoid_multiply(input_a, input_b, n_channels):
n_channels_int = n_channels[0]
in_act = input_a + input_b
t_act = torch.tanh(in_act[:, :n_channels_int, :])
s_act = torch.sigmoid(in_act[:, n_channels_int:, :])
acts = t_act * s_act
return acts
def convert_pad_shape(pad_shape):
l = pad_shape[::-1]
pad_shape = [item for sublist in l for item in sublist]
return pad_shape
def shift_1d(x):
x = F.pad(x, convert_pad_shape([[0, 0], [0, 0], [1, 0]]))[:, :, :-1]
return x
def sequence_mask(length, max_length=None):
if max_length is None:
max_length = length.max()
x = torch.arange(max_length, dtype=length.dtype, device=length.device)
return x.unsqueeze(0) < length.unsqueeze(1)
def generate_path(duration, mask):
"""
duration: [b, 1, t_x]
mask: [b, 1, t_y, t_x]
"""
device = duration.device
b, _, t_y, t_x = mask.shape
cum_duration = torch.cumsum(duration, -1)
cum_duration_flat = cum_duration.view(b * t_x)
path = sequence_mask(cum_duration_flat, t_y).to(mask.dtype)
path = path.view(b, t_x, t_y)
path = path - F.pad(path, convert_pad_shape([[0, 0], [1, 0], [0, 0]]))[:, :-1]
path = path.unsqueeze(1).transpose(2,3) * mask
return path
def clip_grad_value_(parameters, clip_value, norm_type=2):
if isinstance(parameters, torch.Tensor):
parameters = [parameters]
parameters = list(filter(lambda p: p.grad is not None, parameters))
norm_type = float(norm_type)
if clip_value is not None:
clip_value = float(clip_value)
total_norm = 0
for p in parameters:
param_norm = p.grad.data.norm(norm_type)
total_norm += param_norm.item() ** norm_type
if clip_value is not None:
p.grad.data.clamp_(min=-clip_value, max=clip_value)
total_norm = total_norm ** (1. / norm_type)
return total_norm

@ -0,0 +1 @@
使config

@ -0,0 +1,154 @@
import time
import os
import random
import numpy as np
import torch
import torch.utils.data
import commons
from mel_processing import spectrogram_torch, spec_to_mel_torch
from utils import load_wav_to_torch, load_filepaths_and_text, transform
# import h5py
"""Multi speaker version"""
class TextAudioSpeakerLoader(torch.utils.data.Dataset):
"""
1) loads audio, speaker_id, text pairs
2) normalizes text and converts them to sequences of integers
3) computes spectrograms from audio files.
"""
def __init__(self, audiopaths, hparams):
self.audiopaths = load_filepaths_and_text(audiopaths)
self.max_wav_value = hparams.data.max_wav_value
self.sampling_rate = hparams.data.sampling_rate
self.filter_length = hparams.data.filter_length
self.hop_length = hparams.data.hop_length
self.win_length = hparams.data.win_length
self.sampling_rate = hparams.data.sampling_rate
self.use_sr = hparams.train.use_sr
self.spec_len = hparams.train.max_speclen
self.spk_map = hparams.spk
random.seed(1234)
random.shuffle(self.audiopaths)
def get_audio(self, filename):
filename = filename.replace("\\", "/")
audio, sampling_rate = load_wav_to_torch(filename)
if sampling_rate != self.sampling_rate:
raise ValueError("{} SR doesn't match target {} SR".format(
sampling_rate, self.sampling_rate))
audio_norm = audio / self.max_wav_value
audio_norm = audio_norm.unsqueeze(0)
spec_filename = filename.replace(".wav", ".spec.pt")
if os.path.exists(spec_filename):
spec = torch.load(spec_filename)
else:
spec = spectrogram_torch(audio_norm, self.filter_length,
self.sampling_rate, self.hop_length, self.win_length,
center=False)
spec = torch.squeeze(spec, 0)
torch.save(spec, spec_filename)
spk = filename.split("/")[-2]
spk = torch.LongTensor([self.spk_map[spk]])
c = torch.load(filename + ".soft.pt").squeeze(0)
c = torch.repeat_interleave(c, repeats=2, dim=1)
f0 = np.load(filename + ".f0.npy")
f0 = torch.FloatTensor(f0)
lmin = min(c.size(-1), spec.size(-1), f0.shape[0])
assert abs(c.size(-1) - spec.size(-1)) < 4, (c.size(-1), spec.size(-1), f0.shape, filename)
assert abs(lmin - spec.size(-1)) < 4, (c.size(-1), spec.size(-1), f0.shape)
assert abs(lmin - c.size(-1)) < 4, (c.size(-1), spec.size(-1), f0.shape)
spec, c, f0 = spec[:, :lmin], c[:, :lmin], f0[:lmin]
audio_norm = audio_norm[:, :lmin * self.hop_length]
_spec, _c, _audio_norm, _f0 = spec, c, audio_norm, f0
while spec.size(-1) < self.spec_len:
spec = torch.cat((spec, _spec), -1)
c = torch.cat((c, _c), -1)
f0 = torch.cat((f0, _f0), -1)
audio_norm = torch.cat((audio_norm, _audio_norm), -1)
start = random.randint(0, spec.size(-1) - self.spec_len)
end = start + self.spec_len
spec = spec[:, start:end]
c = c[:, start:end]
f0 = f0[start:end]
audio_norm = audio_norm[:, start * self.hop_length:end * self.hop_length]
return c, f0, spec, audio_norm, spk
def __getitem__(self, index):
return self.get_audio(self.audiopaths[index][0])
def __len__(self):
return len(self.audiopaths)
class EvalDataLoader(torch.utils.data.Dataset):
"""
1) loads audio, speaker_id, text pairs
2) normalizes text and converts them to sequences of integers
3) computes spectrograms from audio files.
"""
def __init__(self, audiopaths, hparams):
self.audiopaths = load_filepaths_and_text(audiopaths)
self.max_wav_value = hparams.data.max_wav_value
self.sampling_rate = hparams.data.sampling_rate
self.filter_length = hparams.data.filter_length
self.hop_length = hparams.data.hop_length
self.win_length = hparams.data.win_length
self.sampling_rate = hparams.data.sampling_rate
self.use_sr = hparams.train.use_sr
self.audiopaths = self.audiopaths[:5]
self.spk_map = hparams.spk
def get_audio(self, filename):
filename = filename.replace("\\", "/")
audio, sampling_rate = load_wav_to_torch(filename)
if sampling_rate != self.sampling_rate:
raise ValueError("{} SR doesn't match target {} SR".format(
sampling_rate, self.sampling_rate))
audio_norm = audio / self.max_wav_value
audio_norm = audio_norm.unsqueeze(0)
spec_filename = filename.replace(".wav", ".spec.pt")
if os.path.exists(spec_filename):
spec = torch.load(spec_filename)
else:
spec = spectrogram_torch(audio_norm, self.filter_length,
self.sampling_rate, self.hop_length, self.win_length,
center=False)
spec = torch.squeeze(spec, 0)
torch.save(spec, spec_filename)
spk = filename.split("/")[-2]
spk = torch.LongTensor([self.spk_map[spk]])
c = torch.load(filename + ".soft.pt").squeeze(0)
c = torch.repeat_interleave(c, repeats=2, dim=1)
f0 = np.load(filename + ".f0.npy")
f0 = torch.FloatTensor(f0)
lmin = min(c.size(-1), spec.size(-1), f0.shape[0])
assert abs(c.size(-1) - spec.size(-1)) < 4, (c.size(-1), spec.size(-1), f0.shape)
assert abs(f0.shape[0] - spec.shape[-1]) < 4, (c.size(-1), spec.size(-1), f0.shape)
spec, c, f0 = spec[:, :lmin], c[:, :lmin], f0[:lmin]
audio_norm = audio_norm[:, :lmin * self.hop_length]
return c, f0, spec, audio_norm, spk
def __getitem__(self, index):
return self.get_audio(self.audiopaths[index][0])
def __len__(self):
return len(self.audiopaths)

@ -0,0 +1,20 @@
数据集准备
raw
├───speaker0
│ ├───xxx1-xxx1.wav
│ ├───...
│ └───Lxx-0xx8.wav
└───speaker1
├───xx2-0xxx2.wav
├───...
└───xxx7-xxx007.wav
此外还需要编辑config.json
"n_speakers": 10
"spk":{
"speaker0": 0,
"speaker1": 1,
}

@ -0,0 +1,7 @@
./dataset/32k/yunhao/001829.wav
./dataset/32k/yunhao/001827.wav
./dataset/32k/jishuang/000104.wav
./dataset/32k/nen/kne110_005.wav
./dataset/32k/nen/kne110_004.wav
./dataset/32k/jishuang/000223.wav
./dataset/32k/yunhao/001828.wav

@ -0,0 +1,6 @@
./dataset/32k/nen/kne110_005.wav
./dataset/32k/yunhao/001827.wav
./dataset/32k/jishuang/000104.wav
./dataset/32k/jishuang/000223.wav
./dataset/32k/nen/kne110_004.wav
./dataset/32k/yunhao/001828.wav

@ -0,0 +1,56 @@
import io
import logging
import soundfile
import torch
import torchaudio
from flask import Flask, request, send_file
from flask_cors import CORS
from inference.infer_tool import Svc, RealTimeVC
app = Flask(__name__)
CORS(app)
logging.getLogger('numba').setLevel(logging.WARNING)
@app.route("/voiceChangeModel", methods=["POST"])
def voice_change_model():
request_form = request.form
wave_file = request.files.get("sample", None)
# 变调信息
f_pitch_change = float(request_form.get("fPitchChange", 0))
# DAW所需的采样率
daw_sample = int(float(request_form.get("sampleRate", 0)))
speaker_id = int(float(request_form.get("sSpeakId", 0)))
# http获得wav文件并转换
input_wav_path = io.BytesIO(wave_file.read())
# 模型推理
if raw_infer:
out_audio, out_sr = svc_model.infer(speaker_id, f_pitch_change, input_wav_path)
tar_audio = torchaudio.functional.resample(out_audio, svc_model.target_sample, daw_sample)
else:
out_audio = svc.process(svc_model, speaker_id, f_pitch_change, input_wav_path)
tar_audio = torchaudio.functional.resample(torch.from_numpy(out_audio), svc_model.target_sample, daw_sample)
# 返回音频
out_wav_path = io.BytesIO()
soundfile.write(out_wav_path, tar_audio.cpu().numpy(), daw_sample, format="wav")
out_wav_path.seek(0)
return send_file(out_wav_path, download_name="temp.wav", as_attachment=True)
if __name__ == '__main__':
# 启用则为直接切片合成False为交叉淡化方式
# vst插件调整0.3-0.5s切片时间可以降低延迟,直接切片方法会有连接处爆音、交叉淡化会有轻微重叠声音
# 自行选择能接受的方法或将vst最大切片时间调整为1s此处设为Ture延迟大音质稳定一些
raw_infer = True
# 每个模型和config是唯一对应的
model_name = "logs/32k/G_174000-Copy1.pth"
config_name = "configs/config.json"
svc_model = Svc(model_name, config_name)
svc = RealTimeVC()
# 此处与vst插件对应不建议更改
app.run(port=6842, host="0.0.0.0", debug=False, threaded=False)

@ -0,0 +1,222 @@
import copy
import random
from typing import Optional, Tuple
import torch
import torch.nn as nn
import torch.nn.functional as t_func
from torch.nn.modules.utils import consume_prefix_in_state_dict_if_present
class Hubert(nn.Module):
def __init__(self, num_label_embeddings: int = 100, mask: bool = True):
super().__init__()
self._mask = mask
self.feature_extractor = FeatureExtractor()
self.feature_projection = FeatureProjection()
self.positional_embedding = PositionalConvEmbedding()
self.norm = nn.LayerNorm(768)
self.dropout = nn.Dropout(0.1)
self.encoder = TransformerEncoder(
nn.TransformerEncoderLayer(
768, 12, 3072, activation="gelu", batch_first=True
),
12,
)
self.proj = nn.Linear(768, 256)
self.masked_spec_embed = nn.Parameter(torch.FloatTensor(768).uniform_())
self.label_embedding = nn.Embedding(num_label_embeddings, 256)
def mask(self, x: torch.Tensor) -> Tuple[torch.Tensor, torch.Tensor]:
mask = None
if self.training and self._mask:
mask = _compute_mask((x.size(0), x.size(1)), 0.8, 10, x.device, 2)
x[mask] = self.masked_spec_embed.to(x.dtype)
return x, mask
def encode(
self, x: torch.Tensor, layer: Optional[int] = None
) -> Tuple[torch.Tensor, torch.Tensor]:
x = self.feature_extractor(x)
x = self.feature_projection(x.transpose(1, 2))
x, mask = self.mask(x)
x = x + self.positional_embedding(x)
x = self.dropout(self.norm(x))
x = self.encoder(x, output_layer=layer)
return x, mask
def logits(self, x: torch.Tensor) -> torch.Tensor:
logits = torch.cosine_similarity(
x.unsqueeze(2),
self.label_embedding.weight.unsqueeze(0).unsqueeze(0),
dim=-1,
)
return logits / 0.1
def forward(self, x: torch.Tensor) -> Tuple[torch.Tensor, torch.Tensor]:
x, mask = self.encode(x)
x = self.proj(x)
logits = self.logits(x)
return logits, mask
class HubertSoft(Hubert):
def __init__(self):
super().__init__()
@torch.inference_mode()
def units(self, wav: torch.Tensor) -> torch.Tensor:
wav = t_func.pad(wav, ((400 - 320) // 2, (400 - 320) // 2))
x, _ = self.encode(wav)
return self.proj(x)
class FeatureExtractor(nn.Module):
def __init__(self):
super().__init__()
self.conv0 = nn.Conv1d(1, 512, 10, 5, bias=False)
self.norm0 = nn.GroupNorm(512, 512)
self.conv1 = nn.Conv1d(512, 512, 3, 2, bias=False)
self.conv2 = nn.Conv1d(512, 512, 3, 2, bias=False)
self.conv3 = nn.Conv1d(512, 512, 3, 2, bias=False)
self.conv4 = nn.Conv1d(512, 512, 3, 2, bias=False)
self.conv5 = nn.Conv1d(512, 512, 2, 2, bias=False)
self.conv6 = nn.Conv1d(512, 512, 2, 2, bias=False)
def forward(self, x: torch.Tensor) -> torch.Tensor:
x = t_func.gelu(self.norm0(self.conv0(x)))
x = t_func.gelu(self.conv1(x))
x = t_func.gelu(self.conv2(x))
x = t_func.gelu(self.conv3(x))
x = t_func.gelu(self.conv4(x))
x = t_func.gelu(self.conv5(x))
x = t_func.gelu(self.conv6(x))
return x
class FeatureProjection(nn.Module):
def __init__(self):
super().__init__()
self.norm = nn.LayerNorm(512)
self.projection = nn.Linear(512, 768)
self.dropout = nn.Dropout(0.1)
def forward(self, x: torch.Tensor) -> torch.Tensor:
x = self.norm(x)
x = self.projection(x)
x = self.dropout(x)
return x
class PositionalConvEmbedding(nn.Module):
def __init__(self):
super().__init__()
self.conv = nn.Conv1d(
768,
768,
kernel_size=128,
padding=128 // 2,
groups=16,
)
self.conv = nn.utils.weight_norm(self.conv, name="weight", dim=2)
def forward(self, x: torch.Tensor) -> torch.Tensor:
x = self.conv(x.transpose(1, 2))
x = t_func.gelu(x[:, :, :-1])
return x.transpose(1, 2)
class TransformerEncoder(nn.Module):
def __init__(
self, encoder_layer: nn.TransformerEncoderLayer, num_layers: int
) -> None:
super(TransformerEncoder, self).__init__()
self.layers = nn.ModuleList(
[copy.deepcopy(encoder_layer) for _ in range(num_layers)]
)
self.num_layers = num_layers
def forward(
self,
src: torch.Tensor,
mask: torch.Tensor = None,
src_key_padding_mask: torch.Tensor = None,
output_layer: Optional[int] = None,
) -> torch.Tensor:
output = src
for layer in self.layers[:output_layer]:
output = layer(
output, src_mask=mask, src_key_padding_mask=src_key_padding_mask
)
return output
def _compute_mask(
shape: Tuple[int, int],
mask_prob: float,
mask_length: int,
device: torch.device,
min_masks: int = 0,
) -> torch.Tensor:
batch_size, sequence_length = shape
if mask_length < 1:
raise ValueError("`mask_length` has to be bigger than 0.")
if mask_length > sequence_length:
raise ValueError(
f"`mask_length` has to be smaller than `sequence_length`, but got `mask_length`: {mask_length} and `sequence_length`: {sequence_length}`"
)
# compute number of masked spans in batch
num_masked_spans = int(mask_prob * sequence_length / mask_length + random.random())
num_masked_spans = max(num_masked_spans, min_masks)
# make sure num masked indices <= sequence_length
if num_masked_spans * mask_length > sequence_length:
num_masked_spans = sequence_length // mask_length
# SpecAugment mask to fill
mask = torch.zeros((batch_size, sequence_length), device=device, dtype=torch.bool)
# uniform distribution to sample from, make sure that offset samples are < sequence_length
uniform_dist = torch.ones(
(batch_size, sequence_length - (mask_length - 1)), device=device
)
# get random indices to mask
mask_indices = torch.multinomial(uniform_dist, num_masked_spans)
# expand masked indices to masked spans
mask_indices = (
mask_indices.unsqueeze(dim=-1)
.expand((batch_size, num_masked_spans, mask_length))
.reshape(batch_size, num_masked_spans * mask_length)
)
offsets = (
torch.arange(mask_length, device=device)[None, None, :]
.expand((batch_size, num_masked_spans, mask_length))
.reshape(batch_size, num_masked_spans * mask_length)
)
mask_idxs = mask_indices + offsets
# scatter indices to mask
mask = mask.scatter(1, mask_idxs, True)
return mask
def hubert_soft(
path: str,
) -> HubertSoft:
r"""HuBERT-Soft from `"A Comparison of Discrete and Soft Speech Units for Improved Voice Conversion"`.
Args:
path (str): path of a pretrained model
"""
hubert = HubertSoft()
checkpoint = torch.load(path)
consume_prefix_in_state_dict_if_present(checkpoint, "module.")
hubert.load_state_dict(checkpoint)
hubert.eval()
return hubert

@ -0,0 +1,217 @@
import copy
import random
from typing import Optional, Tuple
import torch
import torch.nn as nn
import torch.nn.functional as t_func
from torch.nn.modules.utils import consume_prefix_in_state_dict_if_present
class Hubert(nn.Module):
def __init__(self, num_label_embeddings: int = 100, mask: bool = True):
super().__init__()
self._mask = mask
self.feature_extractor = FeatureExtractor()
self.feature_projection = FeatureProjection()
self.positional_embedding = PositionalConvEmbedding()
self.norm = nn.LayerNorm(768)
self.dropout = nn.Dropout(0.1)
self.encoder = TransformerEncoder(
nn.TransformerEncoderLayer(
768, 12, 3072, activation="gelu", batch_first=True
),
12,
)
self.proj = nn.Linear(768, 256)
self.masked_spec_embed = nn.Parameter(torch.FloatTensor(768).uniform_())
self.label_embedding = nn.Embedding(num_label_embeddings, 256)
def mask(self, x: torch.Tensor) -> Tuple[torch.Tensor, torch.Tensor]:
mask = None
if self.training and self._mask:
mask = _compute_mask((x.size(0), x.size(1)), 0.8, 10, x.device, 2)
x[mask] = self.masked_spec_embed.to(x.dtype)
return x, mask
def encode(
self, x: torch.Tensor, layer: Optional[int] = None
) -> Tuple[torch.Tensor, torch.Tensor]:
x = self.feature_extractor(x)
x = self.feature_projection(x.transpose(1, 2))
x, mask = self.mask(x)
x = x + self.positional_embedding(x)
x = self.dropout(self.norm(x))
x = self.encoder(x, output_layer=layer)
return x, mask
def logits(self, x: torch.Tensor) -> torch.Tensor:
logits = torch.cosine_similarity(
x.unsqueeze(2),
self.label_embedding.weight.unsqueeze(0).unsqueeze(0),
dim=-1,
)
return logits / 0.1
class HubertSoft(Hubert):
def __init__(self):
super().__init__()
def units(self, wav: torch.Tensor) -> torch.Tensor:
wav = t_func.pad(wav, ((400 - 320) // 2, (400 - 320) // 2))
x, _ = self.encode(wav)
return self.proj(x)
def forward(self, x):
return self.units(x)
class FeatureExtractor(nn.Module):
def __init__(self):
super().__init__()
self.conv0 = nn.Conv1d(1, 512, 10, 5, bias=False)
self.norm0 = nn.GroupNorm(512, 512)
self.conv1 = nn.Conv1d(512, 512, 3, 2, bias=False)
self.conv2 = nn.Conv1d(512, 512, 3, 2, bias=False)
self.conv3 = nn.Conv1d(512, 512, 3, 2, bias=False)
self.conv4 = nn.Conv1d(512, 512, 3, 2, bias=False)
self.conv5 = nn.Conv1d(512, 512, 2, 2, bias=False)
self.conv6 = nn.Conv1d(512, 512, 2, 2, bias=False)
def forward(self, x: torch.Tensor) -> torch.Tensor:
x = t_func.gelu(self.norm0(self.conv0(x)))
x = t_func.gelu(self.conv1(x))
x = t_func.gelu(self.conv2(x))
x = t_func.gelu(self.conv3(x))
x = t_func.gelu(self.conv4(x))
x = t_func.gelu(self.conv5(x))
x = t_func.gelu(self.conv6(x))
return x
class FeatureProjection(nn.Module):
def __init__(self):
super().__init__()
self.norm = nn.LayerNorm(512)
self.projection = nn.Linear(512, 768)
self.dropout = nn.Dropout(0.1)
def forward(self, x: torch.Tensor) -> torch.Tensor:
x = self.norm(x)
x = self.projection(x)
x = self.dropout(x)
return x
class PositionalConvEmbedding(nn.Module):
def __init__(self):
super().__init__()
self.conv = nn.Conv1d(
768,
768,
kernel_size=128,
padding=128 // 2,
groups=16,
)
self.conv = nn.utils.weight_norm(self.conv, name="weight", dim=2)
def forward(self, x: torch.Tensor) -> torch.Tensor:
x = self.conv(x.transpose(1, 2))
x = t_func.gelu(x[:, :, :-1])
return x.transpose(1, 2)
class TransformerEncoder(nn.Module):
def __init__(
self, encoder_layer: nn.TransformerEncoderLayer, num_layers: int
) -> None:
super(TransformerEncoder, self).__init__()
self.layers = nn.ModuleList(
[copy.deepcopy(encoder_layer) for _ in range(num_layers)]
)
self.num_layers = num_layers
def forward(
self,
src: torch.Tensor,
mask: torch.Tensor = None,
src_key_padding_mask: torch.Tensor = None,
output_layer: Optional[int] = None,
) -> torch.Tensor:
output = src
for layer in self.layers[:output_layer]:
output = layer(
output, src_mask=mask, src_key_padding_mask=src_key_padding_mask
)
return output
def _compute_mask(
shape: Tuple[int, int],
mask_prob: float,
mask_length: int,
device: torch.device,
min_masks: int = 0,
) -> torch.Tensor:
batch_size, sequence_length = shape
if mask_length < 1:
raise ValueError("`mask_length` has to be bigger than 0.")
if mask_length > sequence_length:
raise ValueError(
f"`mask_length` has to be smaller than `sequence_length`, but got `mask_length`: {mask_length} and `sequence_length`: {sequence_length}`"
)
# compute number of masked spans in batch
num_masked_spans = int(mask_prob * sequence_length / mask_length + random.random())
num_masked_spans = max(num_masked_spans, min_masks)
# make sure num masked indices <= sequence_length
if num_masked_spans * mask_length > sequence_length:
num_masked_spans = sequence_length // mask_length
# SpecAugment mask to fill
mask = torch.zeros((batch_size, sequence_length), device=device, dtype=torch.bool)
# uniform distribution to sample from, make sure that offset samples are < sequence_length
uniform_dist = torch.ones(
(batch_size, sequence_length - (mask_length - 1)), device=device
)
# get random indices to mask
mask_indices = torch.multinomial(uniform_dist, num_masked_spans)
# expand masked indices to masked spans
mask_indices = (
mask_indices.unsqueeze(dim=-1)
.expand((batch_size, num_masked_spans, mask_length))
.reshape(batch_size, num_masked_spans * mask_length)
)
offsets = (
torch.arange(mask_length, device=device)[None, None, :]
.expand((batch_size, num_masked_spans, mask_length))
.reshape(batch_size, num_masked_spans * mask_length)
)
mask_idxs = mask_indices + offsets
# scatter indices to mask
mask = mask.scatter(1, mask_idxs, True)
return mask
def hubert_soft(
path: str,
) -> HubertSoft:
r"""HuBERT-Soft from `"A Comparison of Discrete and Soft Speech Units for Improved Voice Conversion"`.
Args:
path (str): path of a pretrained model
"""
hubert = HubertSoft()
checkpoint = torch.load(path)
consume_prefix_in_state_dict_if_present(checkpoint, "module.")
hubert.load_state_dict(checkpoint)
hubert.eval()
return hubert

@ -0,0 +1,327 @@
import hashlib
import json
import logging
import os
import time
from pathlib import Path
import librosa
import maad
import numpy as np
# import onnxruntime
import parselmouth
import soundfile
import torch
import torchaudio
from hubert import hubert_model
import utils
from models import SynthesizerTrn
logging.getLogger('matplotlib').setLevel(logging.WARNING)
def read_temp(file_name):
if not os.path.exists(file_name):
with open(file_name, "w") as f:
f.write(json.dumps({"info": "temp_dict"}))
return {}
else:
try:
with open(file_name, "r") as f:
data = f.read()
data_dict = json.loads(data)
if os.path.getsize(file_name) > 50 * 1024 * 1024:
f_name = file_name.replace("\\", "/").split("/")[-1]
print(f"clean {f_name}")
for wav_hash in list(data_dict.keys()):
if int(time.time()) - int(data_dict[wav_hash]["time"]) > 14 * 24 * 3600:
del data_dict[wav_hash]
except Exception as e:
print(e)
print(f"{file_name} error,auto rebuild file")
data_dict = {"info": "temp_dict"}
return data_dict
def write_temp(file_name, data):
with open(file_name, "w") as f:
f.write(json.dumps(data))
def timeit(func):
def run(*args, **kwargs):
t = time.time()
res = func(*args, **kwargs)
print('executing \'%s\' costed %.3fs' % (func.__name__, time.time() - t))
return res
return run
def format_wav(audio_path):
if Path(audio_path).suffix == '.wav':
return
raw_audio, raw_sample_rate = librosa.load(audio_path, mono=True, sr=None)
soundfile.write(Path(audio_path).with_suffix(".wav"), raw_audio, raw_sample_rate)
def get_end_file(dir_path, end):
file_lists = []
for root, dirs, files in os.walk(dir_path):
files = [f for f in files if f[0] != '.']
dirs[:] = [d for d in dirs if d[0] != '.']
for f_file in files:
if f_file.endswith(end):
file_lists.append(os.path.join(root, f_file).replace("\\", "/"))
return file_lists
def get_md5(content):
return hashlib.new("md5", content).hexdigest()
def resize2d_f0(x, target_len):
source = np.array(x)
source[source < 0.001] = np.nan
target = np.interp(np.arange(0, len(source) * target_len, len(source)) / target_len, np.arange(0, len(source)),
source)
res = np.nan_to_num(target)
return res
def get_f0(x, p_len,f0_up_key=0):
time_step = 160 / 16000 * 1000
f0_min = 50
f0_max = 1100
f0_mel_min = 1127 * np.log(1 + f0_min / 700)
f0_mel_max = 1127 * np.log(1 + f0_max / 700)
f0 = parselmouth.Sound(x, 16000).to_pitch_ac(
time_step=time_step / 1000, voicing_threshold=0.6,
pitch_floor=f0_min, pitch_ceiling=f0_max).selected_array['frequency']
if len(f0) > p_len:
f0 = f0[:p_len]
pad_size=(p_len - len(f0) + 1) // 2
if(pad_size>0 or p_len - len(f0) - pad_size>0):
f0 = np.pad(f0,[[pad_size,p_len - len(f0) - pad_size]], mode='constant')
f0 *= pow(2, f0_up_key / 12)
f0_mel = 1127 * np.log(1 + f0 / 700)
f0_mel[f0_mel > 0] = (f0_mel[f0_mel > 0] - f0_mel_min) * 254 / (f0_mel_max - f0_mel_min) + 1
f0_mel[f0_mel <= 1] = 1
f0_mel[f0_mel > 255] = 255
f0_coarse = np.rint(f0_mel).astype(np.int)
return f0_coarse, f0
def clean_pitch(input_pitch):
num_nan = np.sum(input_pitch == 1)
if num_nan / len(input_pitch) > 0.9:
input_pitch[input_pitch != 1] = 1
return input_pitch
def plt_pitch(input_pitch):
input_pitch = input_pitch.astype(float)
input_pitch[input_pitch == 1] = np.nan
return input_pitch
def f0_to_pitch(ff):
f0_pitch = 69 + 12 * np.log2(ff / 440)
return f0_pitch
def fill_a_to_b(a, b):
if len(a) < len(b):
for _ in range(0, len(b) - len(a)):
a.append(a[0])
def mkdir(paths: list):
for path in paths:
if not os.path.exists(path):
os.mkdir(path)
class Svc(object):
def __init__(self, net_g_path, config_path, hubert_path="hubert/hubert-soft-0d54a1f4.pt",
onnx=False):
self.onnx = onnx
self.net_g_path = net_g_path
self.hubert_path = hubert_path
self.dev = torch.device("cuda" if torch.cuda.is_available() else "cpu")
self.net_g_ms = None
self.hps_ms = utils.get_hparams_from_file(config_path)
self.target_sample = self.hps_ms.data.sampling_rate
self.hop_size = self.hps_ms.data.hop_length
self.speakers = {}
for spk, sid in self.hps_ms.spk.items():
self.speakers[sid] = spk
self.spk2id = self.hps_ms.spk
# 加载hubert
self.hubert_soft = hubert_model.hubert_soft(hubert_path)
if torch.cuda.is_available():
self.hubert_soft = self.hubert_soft.cuda()
self.load_model()
def load_model(self):
# 获取模型配置
if self.onnx:
raise NotImplementedError
# self.net_g_ms = SynthesizerTrnForONNX(
# 178,
# self.hps_ms.data.filter_length // 2 + 1,
# self.hps_ms.train.segment_size // self.hps_ms.data.hop_length,
# n_speakers=self.hps_ms.data.n_speakers,
# **self.hps_ms.model)
# _ = utils.load_checkpoint(self.net_g_path, self.net_g_ms, None)
else:
self.net_g_ms = SynthesizerTrn(
self.hps_ms.data.filter_length // 2 + 1,
self.hps_ms.train.segment_size // self.hps_ms.data.hop_length,
**self.hps_ms.model)
_ = utils.load_checkpoint(self.net_g_path, self.net_g_ms, None)
if "half" in self.net_g_path and torch.cuda.is_available():
_ = self.net_g_ms.half().eval().to(self.dev)
else:
_ = self.net_g_ms.eval().to(self.dev)
def get_units(self, source, sr):
source = source.unsqueeze(0).to(self.dev)
with torch.inference_mode():
start = time.time()
units = self.hubert_soft.units(source)
use_time = time.time() - start
print("hubert use time:{}".format(use_time))
return units
def get_unit_pitch(self, in_path, tran):
source, sr = torchaudio.load(in_path)
source = torchaudio.functional.resample(source, sr, 16000)
if len(source.shape) == 2 and source.shape[1] >= 2:
source = torch.mean(source, dim=0).unsqueeze(0)
soft = self.get_units(source, sr).squeeze(0).cpu().numpy()
f0_coarse, f0 = get_f0(source.cpu().numpy()[0], soft.shape[0]*2, tran)
return soft, f0
def infer(self, speaker_id, tran, raw_path):
if type(speaker_id) == str:
speaker_id = self.spk2id[speaker_id]
sid = torch.LongTensor([int(speaker_id)]).to(self.dev).unsqueeze(0)
soft, pitch = self.get_unit_pitch(raw_path, tran)
f0 = torch.FloatTensor(clean_pitch(pitch)).unsqueeze(0).to(self.dev)
if "half" in self.net_g_path and torch.cuda.is_available():
stn_tst = torch.HalfTensor(soft)
else:
stn_tst = torch.FloatTensor(soft)
with torch.no_grad():
x_tst = stn_tst.unsqueeze(0).to(self.dev)
start = time.time()
x_tst = torch.repeat_interleave(x_tst, repeats=2, dim=1).transpose(1, 2)
audio = self.net_g_ms.infer(x_tst, f0=f0, g=sid)[0,0].data.float()
use_time = time.time() - start
print("vits use time:{}".format(use_time))
return audio, audio.shape[-1]
# class SvcONNXInferModel(object):
# def __init__(self, hubert_onnx, vits_onnx, config_path):
# self.config_path = config_path
# self.vits_onnx = vits_onnx
# self.hubert_onnx = hubert_onnx
# self.hubert_onnx_session = onnxruntime.InferenceSession(hubert_onnx, providers=['CUDAExecutionProvider', ])
# self.inspect_onnx(self.hubert_onnx_session)
# self.vits_onnx_session = onnxruntime.InferenceSession(vits_onnx, providers=['CUDAExecutionProvider', ])
# self.inspect_onnx(self.vits_onnx_session)
# self.hps_ms = utils.get_hparams_from_file(self.config_path)
# self.target_sample = self.hps_ms.data.sampling_rate
# self.feature_input = FeatureInput(self.hps_ms.data.sampling_rate, self.hps_ms.data.hop_length)
#
# @staticmethod
# def inspect_onnx(session):
# for i in session.get_inputs():
# print("name:{}\tshape:{}\tdtype:{}".format(i.name, i.shape, i.type))
# for i in session.get_outputs():
# print("name:{}\tshape:{}\tdtype:{}".format(i.name, i.shape, i.type))
#
# def infer(self, speaker_id, tran, raw_path):
# sid = np.array([int(speaker_id)], dtype=np.int64)
# soft, pitch = self.get_unit_pitch(raw_path, tran)
# pitch = np.expand_dims(pitch, axis=0).astype(np.int64)
# stn_tst = soft
# x_tst = np.expand_dims(stn_tst, axis=0)
# x_tst_lengths = np.array([stn_tst.shape[0]], dtype=np.int64)
# # 使用ONNX Runtime进行推理
# start = time.time()
# audio = self.vits_onnx_session.run(output_names=["audio"],
# input_feed={
# "hidden_unit": x_tst,
# "lengths": x_tst_lengths,
# "pitch": pitch,
# "sid": sid,
# })[0][0, 0]
# use_time = time.time() - start
# print("vits_onnx_session.run time:{}".format(use_time))
# audio = torch.from_numpy(audio)
# return audio, audio.shape[-1]
#
# def get_units(self, source, sr):
# source = torchaudio.functional.resample(source, sr, 16000)
# if len(source.shape) == 2 and source.shape[1] >= 2:
# source = torch.mean(source, dim=0).unsqueeze(0)
# source = source.unsqueeze(0)
# # 使用ONNX Runtime进行推理
# start = time.time()
# units = self.hubert_onnx_session.run(output_names=["embed"],
# input_feed={"source": source.numpy()})[0]
# use_time = time.time() - start
# print("hubert_onnx_session.run time:{}".format(use_time))
# return units
#
# def transcribe(self, source, sr, length, transform):
# feature_pit = self.feature_input.compute_f0(source, sr)
# feature_pit = feature_pit * 2 ** (transform / 12)
# feature_pit = resize2d_f0(feature_pit, length)
# coarse_pit = self.feature_input.coarse_f0(feature_pit)
# return coarse_pit
#
# def get_unit_pitch(self, in_path, tran):
# source, sr = torchaudio.load(in_path)
# soft = self.get_units(source, sr).squeeze(0)
# input_pitch = self.transcribe(source.numpy()[0], sr, soft.shape[0], tran)
# return soft, input_pitch
class RealTimeVC:
def __init__(self):
self.last_chunk = None
self.last_o = None
self.chunk_len = 16000 # 区块长度
self.pre_len = 3840 # 交叉淡化长度640的倍数
"""输入输出都是1维numpy 音频波形数组"""
def process(self, svc_model, speaker_id, f_pitch_change, input_wav_path):
audio, sr = torchaudio.load(input_wav_path)
audio = audio.cpu().numpy()[0]
temp_wav = io.BytesIO()
if self.last_chunk is None:
input_wav_path.seek(0)
audio, sr = svc_model.infer(speaker_id, f_pitch_change, input_wav_path)
audio = audio.cpu().numpy()
self.last_chunk = audio[-self.pre_len:]
self.last_o = audio
return audio[-self.chunk_len:]
else:
audio = np.concatenate([self.last_chunk, audio])
soundfile.write(temp_wav, audio, sr, format="wav")
temp_wav.seek(0)
audio, sr = svc_model.infer(speaker_id, f_pitch_change, temp_wav)
audio = audio.cpu().numpy()
ret = maad.util.crossfade(self.last_o, audio, self.pre_len)
self.last_chunk = audio[-self.pre_len:]
self.last_o = audio
return ret[self.chunk_len:2 * self.chunk_len]

@ -0,0 +1,160 @@
import hashlib
import json
import logging
import os
import time
from pathlib import Path
import io
import librosa
import maad
import numpy as np
from inference import slicer
import parselmouth
import soundfile
import torch
import torchaudio
from hubert import hubert_model
import utils
from models import SynthesizerTrn
logging.getLogger('numba').setLevel(logging.WARNING)
logging.getLogger('matplotlib').setLevel(logging.WARNING)
def resize2d_f0(x, target_len):
source = np.array(x)
source[source < 0.001] = np.nan
target = np.interp(np.arange(0, len(source) * target_len, len(source)) / target_len, np.arange(0, len(source)),
source)
res = np.nan_to_num(target)
return res
def get_f0(x, p_len,f0_up_key=0):
time_step = 160 / 16000 * 1000
f0_min = 50
f0_max = 1100
f0_mel_min = 1127 * np.log(1 + f0_min / 700)
f0_mel_max = 1127 * np.log(1 + f0_max / 700)
f0 = parselmouth.Sound(x, 16000).to_pitch_ac(
time_step=time_step / 1000, voicing_threshold=0.6,
pitch_floor=f0_min, pitch_ceiling=f0_max).selected_array['frequency']
pad_size=(p_len - len(f0) + 1) // 2
if(pad_size>0 or p_len - len(f0) - pad_size>0):
f0 = np.pad(f0,[[pad_size,p_len - len(f0) - pad_size]], mode='constant')
f0 *= pow(2, f0_up_key / 12)
f0_mel = 1127 * np.log(1 + f0 / 700)
f0_mel[f0_mel > 0] = (f0_mel[f0_mel > 0] - f0_mel_min) * 254 / (f0_mel_max - f0_mel_min) + 1
f0_mel[f0_mel <= 1] = 1
f0_mel[f0_mel > 255] = 255
f0_coarse = np.rint(f0_mel).astype(np.int)
return f0_coarse, f0
def clean_pitch(input_pitch):
num_nan = np.sum(input_pitch == 1)
if num_nan / len(input_pitch) > 0.9:
input_pitch[input_pitch != 1] = 1
return input_pitch
def plt_pitch(input_pitch):
input_pitch = input_pitch.astype(float)
input_pitch[input_pitch == 1] = np.nan
return input_pitch
def f0_to_pitch(ff):
f0_pitch = 69 + 12 * np.log2(ff / 440)
return f0_pitch
def fill_a_to_b(a, b):
if len(a) < len(b):
for _ in range(0, len(b) - len(a)):
a.append(a[0])
def mkdir(paths: list):
for path in paths:
if not os.path.exists(path):
os.mkdir(path)
class VitsSvc(object):
def __init__(self):
self.device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
self.SVCVITS = None
self.hps = None
self.speakers = None
self.hubert_soft = hubert_model.hubert_soft("hubert/model.pt")
def set_device(self, device):
self.device = torch.device(device)
self.hubert_soft.to(self.device)
if self.SVCVITS != None:
self.SVCVITS.to(self.device)
def loadCheckpoint(self, path):
self.hps = utils.get_hparams_from_file(f"checkpoints/{path}/config.json")
self.SVCVITS = SynthesizerTrn(
self.hps.data.filter_length // 2 + 1,
self.hps.train.segment_size // self.hps.data.hop_length,
**self.hps.model)
_ = utils.load_checkpoint(f"checkpoints/{path}/model.pth", self.SVCVITS, None)
_ = self.SVCVITS.eval().to(self.device)
self.speakers = self.hps.spk
def get_units(self, source, sr):
source = source.unsqueeze(0).to(self.device)
with torch.inference_mode():
units = self.hubert_soft.units(source)
return units
def get_unit_pitch(self, in_path, tran):
source, sr = torchaudio.load(in_path)
source = torchaudio.functional.resample(source, sr, 16000)
if len(source.shape) == 2 and source.shape[1] >= 2:
source = torch.mean(source, dim=0).unsqueeze(0)
soft = self.get_units(source, sr).squeeze(0).cpu().numpy()
f0_coarse, f0 = get_f0(source.cpu().numpy()[0], soft.shape[0]*2, tran)
return soft, f0
def infer(self, speaker_id, tran, raw_path):
speaker_id = self.speakers[speaker_id]
sid = torch.LongTensor([int(speaker_id)]).to(self.device).unsqueeze(0)
soft, pitch = self.get_unit_pitch(raw_path, tran)
f0 = torch.FloatTensor(clean_pitch(pitch)).unsqueeze(0).to(self.device)
stn_tst = torch.FloatTensor(soft)
with torch.no_grad():
x_tst = stn_tst.unsqueeze(0).to(self.device)
x_tst = torch.repeat_interleave(x_tst, repeats=2, dim=1).transpose(1, 2)
audio = self.SVCVITS.infer(x_tst, f0=f0, g=sid)[0,0].data.float()
return audio, audio.shape[-1]
def inference(self,srcaudio,chara,tran,slice_db):
sampling_rate, audio = srcaudio
audio = (audio / np.iinfo(audio.dtype).max).astype(np.float32)
if len(audio.shape) > 1:
audio = librosa.to_mono(audio.transpose(1, 0))
if sampling_rate != 16000:
audio = librosa.resample(audio, orig_sr=sampling_rate, target_sr=16000)
soundfile.write("tmpwav.wav", audio, 16000, format="wav")
chunks = slicer.cut("tmpwav.wav", db_thresh=slice_db)
audio_data, audio_sr = slicer.chunks2audio("tmpwav.wav", chunks)
audio = []
for (slice_tag, data) in audio_data:
length = int(np.ceil(len(data) / audio_sr * self.hps.data.sampling_rate))
raw_path = io.BytesIO()
soundfile.write(raw_path, data, audio_sr, format="wav")
raw_path.seek(0)
if slice_tag:
_audio = np.zeros(length)
else:
out_audio, out_sr = self.infer(chara, tran, raw_path)
_audio = out_audio.cpu().numpy()
audio.extend(list(_audio))
audio = (np.array(audio) * 32768.0).astype('int16')
return (self.hps.data.sampling_rate,audio)

@ -0,0 +1,142 @@
import librosa
import torch
import torchaudio
class Slicer:
def __init__(self,
sr: int,
threshold: float = -40.,
min_length: int = 5000,
min_interval: int = 300,
hop_size: int = 20,
max_sil_kept: int = 5000):
if not min_length >= min_interval >= hop_size:
raise ValueError('The following condition must be satisfied: min_length >= min_interval >= hop_size')
if not max_sil_kept >= hop_size:
raise ValueError('The following condition must be satisfied: max_sil_kept >= hop_size')
min_interval = sr * min_interval / 1000
self.threshold = 10 ** (threshold / 20.)
self.hop_size = round(sr * hop_size / 1000)
self.win_size = min(round(min_interval), 4 * self.hop_size)
self.min_length = round(sr * min_length / 1000 / self.hop_size)
self.min_interval = round(min_interval / self.hop_size)
self.max_sil_kept = round(sr * max_sil_kept / 1000 / self.hop_size)
def _apply_slice(self, waveform, begin, end):
if len(waveform.shape) > 1:
return waveform[:, begin * self.hop_size: min(waveform.shape[1], end * self.hop_size)]
else:
return waveform[begin * self.hop_size: min(waveform.shape[0], end * self.hop_size)]
# @timeit
def slice(self, waveform):
if len(waveform.shape) > 1:
samples = librosa.to_mono(waveform)
else:
samples = waveform
if samples.shape[0] <= self.min_length:
return {"0": {"slice": False, "split_time": f"0,{len(waveform)}"}}
rms_list = librosa.feature.rms(y=samples, frame_length=self.win_size, hop_length=self.hop_size).squeeze(0)
sil_tags = []
silence_start = None
clip_start = 0
for i, rms in enumerate(rms_list):
# Keep looping while frame is silent.
if rms < self.threshold:
# Record start of silent frames.
if silence_start is None:
silence_start = i
continue
# Keep looping while frame is not silent and silence start has not been recorded.
if silence_start is None:
continue
# Clear recorded silence start if interval is not enough or clip is too short
is_leading_silence = silence_start == 0 and i > self.max_sil_kept
need_slice_middle = i - silence_start >= self.min_interval and i - clip_start >= self.min_length
if not is_leading_silence and not need_slice_middle:
silence_start = None
continue
# Need slicing. Record the range of silent frames to be removed.
if i - silence_start <= self.max_sil_kept:
pos = rms_list[silence_start: i + 1].argmin() + silence_start
if silence_start == 0:
sil_tags.append((0, pos))
else:
sil_tags.append((pos, pos))
clip_start = pos
elif i - silence_start <= self.max_sil_kept * 2:
pos = rms_list[i - self.max_sil_kept: silence_start + self.max_sil_kept + 1].argmin()
pos += i - self.max_sil_kept
pos_l = rms_list[silence_start: silence_start + self.max_sil_kept + 1].argmin() + silence_start
pos_r = rms_list[i - self.max_sil_kept: i + 1].argmin() + i - self.max_sil_kept
if silence_start == 0:
sil_tags.append((0, pos_r))
clip_start = pos_r
else:
sil_tags.append((min(pos_l, pos), max(pos_r, pos)))
clip_start = max(pos_r, pos)
else:
pos_l = rms_list[silence_start: silence_start + self.max_sil_kept + 1].argmin() + silence_start
pos_r = rms_list[i - self.max_sil_kept: i + 1].argmin() + i - self.max_sil_kept
if silence_start == 0:
sil_tags.append((0, pos_r))
else:
sil_tags.append((pos_l, pos_r))
clip_start = pos_r
silence_start = None
# Deal with trailing silence.
total_frames = rms_list.shape[0]
if silence_start is not None and total_frames - silence_start >= self.min_interval:
silence_end = min(total_frames, silence_start + self.max_sil_kept)
pos = rms_list[silence_start: silence_end + 1].argmin() + silence_start
sil_tags.append((pos, total_frames + 1))
# Apply and return slices.
if len(sil_tags) == 0:
return {"0": {"slice": False, "split_time": f"0,{len(waveform)}"}}
else:
chunks = []
# 第一段静音并非从头开始,补上有声片段
if sil_tags[0][0]:
chunks.append(
{"slice": False, "split_time": f"0,{min(waveform.shape[0], sil_tags[0][0] * self.hop_size)}"})
for i in range(0, len(sil_tags)):
# 标识有声片段(跳过第一段)
if i:
chunks.append({"slice": False,
"split_time": f"{sil_tags[i - 1][1] * self.hop_size},{min(waveform.shape[0], sil_tags[i][0] * self.hop_size)}"})
# 标识所有静音片段
chunks.append({"slice": True,
"split_time": f"{sil_tags[i][0] * self.hop_size},{min(waveform.shape[0], sil_tags[i][1] * self.hop_size)}"})
# 最后一段静音并非结尾,补上结尾片段
if sil_tags[-1][1] * self.hop_size < len(waveform):
chunks.append({"slice": False, "split_time": f"{sil_tags[-1][1] * self.hop_size},{len(waveform)}"})
chunk_dict = {}
for i in range(len(chunks)):
chunk_dict[str(i)] = chunks[i]
return chunk_dict
def cut(audio_path, db_thresh=-30, min_len=5000):
audio, sr = librosa.load(audio_path, sr=None)
slicer = Slicer(
sr=sr,
threshold=db_thresh,
min_length=min_len
)
chunks = slicer.slice(audio)
return chunks
def chunks2audio(audio_path, chunks):
chunks = dict(chunks)
audio, sr = torchaudio.load(audio_path)
if len(audio.shape) == 2 and audio.shape[1] >= 2:
audio = torch.mean(audio, dim=0).unsqueeze(0)
audio = audio.cpu().numpy()[0]
result = []
for k, v in chunks.items():
tag = v["split_time"].split(",")
if tag[0] != tag[1]:
result.append((v["slice"], audio[int(tag[0]):int(tag[1])]))
return result, sr

@ -0,0 +1,56 @@
import io
import logging
import time
from pathlib import Path
import librosa
import numpy as np
import soundfile
from inference import infer_tool
from inference import slicer
from inference.infer_tool import Svc
logging.getLogger('numba').setLevel(logging.WARNING)
chunks_dict = infer_tool.read_temp("inference/chunks_temp.json")
model_path = "logs/32k/G_174000-Copy1.pth"
config_path = "configs/config.json"
svc_model = Svc(model_path, config_path)
infer_tool.mkdir(["raw", "results"])
# 支持多个wav文件放在raw文件夹下
clean_names = ["君の知らない物語-src"]
trans = [-5] # 音高调整,支持正负(半音)
spk_list = ['yunhao'] # 每次同时合成多语者音色
slice_db = -40 # 默认-40嘈杂的音频可以-30干声保留呼吸可以-50
wav_format = 'flac' # 音频输出格式
infer_tool.fill_a_to_b(trans, clean_names)
for clean_name, tran in zip(clean_names, trans):
raw_audio_path = f"raw/{clean_name}"
if "." not in raw_audio_path:
raw_audio_path += ".wav"
infer_tool.format_wav(raw_audio_path)
wav_path = Path(raw_audio_path).with_suffix('.wav')
chunks = slicer.cut(wav_path, db_thresh=slice_db)
audio_data, audio_sr = slicer.chunks2audio(wav_path, chunks)
for spk in spk_list:
audio = []
for (slice_tag, data) in audio_data:
print(f'#=====segment start, {round(len(data) / audio_sr, 3)}s======')
length = int(np.ceil(len(data) / audio_sr * svc_model.target_sample))
raw_path = io.BytesIO()
soundfile.write(raw_path, data, audio_sr, format="wav")
raw_path.seek(0)
if slice_tag:
print('jump empty segment')
_audio = np.zeros(length)
else:
out_audio, out_sr = svc_model.infer(spk, tran, raw_path)
_audio = out_audio.cpu().numpy()
audio.extend(list(_audio))
res_path = f'./results/{clean_name}_{tran}key_{spk}.{wav_format}'
soundfile.write(res_path, audio, svc_model.target_sample, format=wav_format)

@ -0,0 +1,61 @@
import torch
from torch.nn import functional as F
import commons
def feature_loss(fmap_r, fmap_g):
loss = 0
for dr, dg in zip(fmap_r, fmap_g):
for rl, gl in zip(dr, dg):
rl = rl.float().detach()
gl = gl.float()
loss += torch.mean(torch.abs(rl - gl))
return loss * 2
def discriminator_loss(disc_real_outputs, disc_generated_outputs):
loss = 0
r_losses = []
g_losses = []
for dr, dg in zip(disc_real_outputs, disc_generated_outputs):
dr = dr.float()
dg = dg.float()
r_loss = torch.mean((1-dr)**2)
g_loss = torch.mean(dg**2)
loss += (r_loss + g_loss)
r_losses.append(r_loss.item())
g_losses.append(g_loss.item())
return loss, r_losses, g_losses
def generator_loss(disc_outputs):
loss = 0
gen_losses = []
for dg in disc_outputs:
dg = dg.float()
l = torch.mean((1-dg)**2)
gen_losses.append(l)
loss += l
return loss, gen_losses
def kl_loss(z_p, logs_q, m_p, logs_p, z_mask):
"""
z_p, logs_q: [b, h, t_t]
m_p, logs_p: [b, h, t_t]
"""
z_p = z_p.float()
logs_q = logs_q.float()
m_p = m_p.float()
logs_p = logs_p.float()
z_mask = z_mask.float()
#print(logs_p)
kl = logs_p - logs_q - 0.5
kl += 0.5 * ((z_p - m_p)**2) * torch.exp(-2. * logs_p)
kl = torch.sum(kl * z_mask)
l = kl / torch.sum(z_mask)
return l

@ -0,0 +1,112 @@
import math
import os
import random
import torch
from torch import nn
import torch.nn.functional as F
import torch.utils.data
import numpy as np
import librosa
import librosa.util as librosa_util
from librosa.util import normalize, pad_center, tiny
from scipy.signal import get_window
from scipy.io.wavfile import read
from librosa.filters import mel as librosa_mel_fn
MAX_WAV_VALUE = 32768.0
def dynamic_range_compression_torch(x, C=1, clip_val=1e-5):
"""
PARAMS
------
C: compression factor
"""
return torch.log(torch.clamp(x, min=clip_val) * C)
def dynamic_range_decompression_torch(x, C=1):
"""
PARAMS
------
C: compression factor used to compress
"""
return torch.exp(x) / C
def spectral_normalize_torch(magnitudes):
output = dynamic_range_compression_torch(magnitudes)
return output
def spectral_de_normalize_torch(magnitudes):
output = dynamic_range_decompression_torch(magnitudes)
return output
mel_basis = {}
hann_window = {}
def spectrogram_torch(y, n_fft, sampling_rate, hop_size, win_size, center=False):
if torch.min(y) < -1.:
print('min value is ', torch.min(y))
if torch.max(y) > 1.:
print('max value is ', torch.max(y))
global hann_window
dtype_device = str(y.dtype) + '_' + str(y.device)
wnsize_dtype_device = str(win_size) + '_' + dtype_device
if wnsize_dtype_device not in hann_window:
hann_window[wnsize_dtype_device] = torch.hann_window(win_size).to(dtype=y.dtype, device=y.device)
y = torch.nn.functional.pad(y.unsqueeze(1), (int((n_fft-hop_size)/2), int((n_fft-hop_size)/2)), mode='reflect')
y = y.squeeze(1)
spec = torch.stft(y, n_fft, hop_length=hop_size, win_length=win_size, window=hann_window[wnsize_dtype_device],
center=center, pad_mode='reflect', normalized=False, onesided=True, return_complex=False)
spec = torch.sqrt(spec.pow(2).sum(-1) + 1e-6)
return spec
def spec_to_mel_torch(spec, n_fft, num_mels, sampling_rate, fmin, fmax):
global mel_basis
dtype_device = str(spec.dtype) + '_' + str(spec.device)
fmax_dtype_device = str(fmax) + '_' + dtype_device
if fmax_dtype_device not in mel_basis:
mel = librosa_mel_fn(sr=sampling_rate, n_fft=n_fft, n_mels=num_mels, fmin=fmin, fmax=fmax)
mel_basis[fmax_dtype_device] = torch.from_numpy(mel).to(dtype=spec.dtype, device=spec.device)
spec = torch.matmul(mel_basis[fmax_dtype_device], spec)
spec = spectral_normalize_torch(spec)
return spec
def mel_spectrogram_torch(y, n_fft, num_mels, sampling_rate, hop_size, win_size, fmin, fmax, center=False):
if torch.min(y) < -1.:
print('min value is ', torch.min(y))
if torch.max(y) > 1.:
print('max value is ', torch.max(y))
global mel_basis, hann_window
dtype_device = str(y.dtype) + '_' + str(y.device)
fmax_dtype_device = str(fmax) + '_' + dtype_device
wnsize_dtype_device = str(win_size) + '_' + dtype_device
if fmax_dtype_device not in mel_basis:
mel = librosa_mel_fn(sr=sampling_rate, n_fft=n_fft, n_mels=num_mels, fmin=fmin, fmax=fmax)
mel_basis[fmax_dtype_device] = torch.from_numpy(mel).to(dtype=y.dtype, device=y.device)
if wnsize_dtype_device not in hann_window:
hann_window[wnsize_dtype_device] = torch.hann_window(win_size).to(dtype=y.dtype, device=y.device)
y = torch.nn.functional.pad(y.unsqueeze(1), (int((n_fft-hop_size)/2), int((n_fft-hop_size)/2)), mode='reflect')
y = y.squeeze(1)
spec = torch.stft(y, n_fft, hop_length=hop_size, win_length=win_size, window=hann_window[wnsize_dtype_device],
center=center, pad_mode='reflect', normalized=False, onesided=True, return_complex=False)
spec = torch.sqrt(spec.pow(2).sum(-1) + 1e-6)
spec = torch.matmul(mel_basis[fmax_dtype_device], spec)
spec = spectral_normalize_torch(spec)
return spec

@ -0,0 +1,328 @@
import copy
import math
import torch
from torch import nn
from torch.nn import functional as F
import attentions
import commons
import modules
from torch.nn import Conv1d, ConvTranspose1d, AvgPool1d, Conv2d
from torch.nn.utils import weight_norm, remove_weight_norm, spectral_norm
from commons import init_weights, get_padding
from vdecoder.hifigan.models import Generator
from utils import f0_to_coarse
class ResidualCouplingBlock(nn.Module):
def __init__(self,
channels,
hidden_channels,
kernel_size,
dilation_rate,
n_layers,
n_flows=4,
gin_channels=0):
super().__init__()
self.channels = channels
self.hidden_channels = hidden_channels
self.kernel_size = kernel_size
self.dilation_rate = dilation_rate
self.n_layers = n_layers
self.n_flows = n_flows
self.gin_channels = gin_channels
self.flows = nn.ModuleList()
for i in range(n_flows):
self.flows.append(modules.ResidualCouplingLayer(channels, hidden_channels, kernel_size, dilation_rate, n_layers, gin_channels=gin_channels, mean_only=True))
self.flows.append(modules.Flip())
def forward(self, x, x_mask, g=None, reverse=False):
if not reverse:
for flow in self.flows:
x, _ = flow(x, x_mask, g=g, reverse=reverse)
else:
for flow in reversed(self.flows):
x = flow(x, x_mask, g=g, reverse=reverse)
return x
class Encoder(nn.Module):
def __init__(self,
in_channels,
out_channels,
hidden_channels,
kernel_size,
dilation_rate,
n_layers,
gin_channels=0):
super().__init__()
self.in_channels = in_channels
self.out_channels = out_channels
self.hidden_channels = hidden_channels
self.kernel_size = kernel_size
self.dilation_rate = dilation_rate
self.n_layers = n_layers
self.gin_channels = gin_channels
self.pre = nn.Conv1d(in_channels, hidden_channels, 1)
self.enc = modules.WN(hidden_channels, kernel_size, dilation_rate, n_layers, gin_channels=gin_channels)
self.proj = nn.Conv1d(hidden_channels, out_channels * 2, 1)
def forward(self, x, x_lengths, g=None):
# print(x.shape,x_lengths.shape)
x_mask = torch.unsqueeze(commons.sequence_mask(x_lengths, x.size(2)), 1).to(x.dtype)
x = self.pre(x) * x_mask
x = self.enc(x, x_mask, g=g)
stats = self.proj(x) * x_mask
m, logs = torch.split(stats, self.out_channels, dim=1)
z = (m + torch.randn_like(m) * torch.exp(logs)) * x_mask
return z, m, logs, x_mask
class TextEncoder(nn.Module):
def __init__(self,
in_channels,
out_channels,
hidden_channels,
kernel_size,
dilation_rate,
n_layers,
gin_channels=0,
filter_channels=None,
n_heads=None,
p_dropout=None):
super().__init__()
self.in_channels = in_channels
self.out_channels = out_channels
self.hidden_channels = hidden_channels
self.kernel_size = kernel_size
self.dilation_rate = dilation_rate
self.n_layers = n_layers
self.gin_channels = gin_channels
self.pre = nn.Conv1d(in_channels, hidden_channels, 1)
self.proj = nn.Conv1d(hidden_channels, out_channels * 2, 1)
self.f0_emb = nn.Embedding(256, hidden_channels)
self.enc_ = attentions.Encoder(
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size,
p_dropout)
def forward(self, x, x_lengths, f0=None):
x_mask = torch.unsqueeze(commons.sequence_mask(x_lengths, x.size(2)), 1).to(x.dtype)
x = self.pre(x) * x_mask
x = x + self.f0_emb(f0.long()).transpose(1,2)
x = self.enc_(x * x_mask, x_mask)
stats = self.proj(x) * x_mask
m, logs = torch.split(stats, self.out_channels, dim=1)
z = (m + torch.randn_like(m) * torch.exp(logs)) * x_mask
return z, m, logs, x_mask
class DiscriminatorP(torch.nn.Module):
def __init__(self, period, kernel_size=5, stride=3, use_spectral_norm=False):
super(DiscriminatorP, self).__init__()
self.period = period
self.use_spectral_norm = use_spectral_norm
norm_f = weight_norm if use_spectral_norm == False else spectral_norm
self.convs = nn.ModuleList([
norm_f(Conv2d(1, 32, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))),
norm_f(Conv2d(32, 128, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))),
norm_f(Conv2d(128, 512, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))),
norm_f(Conv2d(512, 1024, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))),
norm_f(Conv2d(1024, 1024, (kernel_size, 1), 1, padding=(get_padding(kernel_size, 1), 0))),
])
self.conv_post = norm_f(Conv2d(1024, 1, (3, 1), 1, padding=(1, 0)))
def forward(self, x):
fmap = []
# 1d to 2d
b, c, t = x.shape
if t % self.period != 0: # pad first
n_pad = self.period - (t % self.period)
x = F.pad(x, (0, n_pad), "reflect")
t = t + n_pad
x = x.view(b, c, t // self.period, self.period)
for l in self.convs:
x = l(x)
x = F.leaky_relu(x, modules.LRELU_SLOPE)
fmap.append(x)
x = self.conv_post(x)
fmap.append(x)
x = torch.flatten(x, 1, -1)
return x, fmap
class DiscriminatorS(torch.nn.Module):
def __init__(self, use_spectral_norm=False):
super(DiscriminatorS, self).__init__()
norm_f = weight_norm if use_spectral_norm == False else spectral_norm
self.convs = nn.ModuleList([
norm_f(Conv1d(1, 16, 15, 1, padding=7)),
norm_f(Conv1d(16, 64, 41, 4, groups=4, padding=20)),
norm_f(Conv1d(64, 256, 41, 4, groups=16, padding=20)),
norm_f(Conv1d(256, 1024, 41, 4, groups=64, padding=20)),
norm_f(Conv1d(1024, 1024, 41, 4, groups=256, padding=20)),
norm_f(Conv1d(1024, 1024, 5, 1, padding=2)),
])
self.conv_post = norm_f(Conv1d(1024, 1, 3, 1, padding=1))
def forward(self, x):
fmap = []
for l in self.convs:
x = l(x)
x = F.leaky_relu(x, modules.LRELU_SLOPE)
fmap.append(x)
x = self.conv_post(x)
fmap.append(x)
x = torch.flatten(x, 1, -1)
return x, fmap
class MultiPeriodDiscriminator(torch.nn.Module):
def __init__(self, use_spectral_norm=False):
super(MultiPeriodDiscriminator, self).__init__()
periods = [2,3,5,7,11]
discs = [DiscriminatorS(use_spectral_norm=use_spectral_norm)]
discs = discs + [DiscriminatorP(i, use_spectral_norm=use_spectral_norm) for i in periods]
self.discriminators = nn.ModuleList(discs)
def forward(self, y, y_hat):
y_d_rs = []
y_d_gs = []
fmap_rs = []
fmap_gs = []
for i, d in enumerate(self.discriminators):
y_d_r, fmap_r = d(y)
y_d_g, fmap_g = d(y_hat)
y_d_rs.append(y_d_r)
y_d_gs.append(y_d_g)
fmap_rs.append(fmap_r)
fmap_gs.append(fmap_g)
return y_d_rs, y_d_gs, fmap_rs, fmap_gs
class SpeakerEncoder(torch.nn.Module):
def __init__(self, mel_n_channels=80, model_num_layers=3, model_hidden_size=256, model_embedding_size=256):
super(SpeakerEncoder, self).__init__()
self.lstm = nn.LSTM(mel_n_channels, model_hidden_size, model_num_layers, batch_first=True)
self.linear = nn.Linear(model_hidden_size, model_embedding_size)
self.relu = nn.ReLU()
def forward(self, mels):
self.lstm.flatten_parameters()
_, (hidden, _) = self.lstm(mels)
embeds_raw = self.relu(self.linear(hidden[-1]))
return embeds_raw / torch.norm(embeds_raw, dim=1, keepdim=True)
def compute_partial_slices(self, total_frames, partial_frames, partial_hop):
mel_slices = []
for i in range(0, total_frames-partial_frames, partial_hop):
mel_range = torch.arange(i, i+partial_frames)
mel_slices.append(mel_range)
return mel_slices
def embed_utterance(self, mel, partial_frames=128, partial_hop=64):
mel_len = mel.size(1)
last_mel = mel[:,-partial_frames:]
if mel_len > partial_frames:
mel_slices = self.compute_partial_slices(mel_len, partial_frames, partial_hop)
mels = list(mel[:,s] for s in mel_slices)
mels.append(last_mel)
mels = torch.stack(tuple(mels), 0).squeeze(1)
with torch.no_grad():
partial_embeds = self(mels)
embed = torch.mean(partial_embeds, axis=0).unsqueeze(0)
#embed = embed / torch.linalg.norm(embed, 2)
else:
with torch.no_grad():
embed = self(last_mel)
return embed
class SynthesizerTrn(nn.Module):
"""
Synthesizer for Training
"""
def __init__(self,
spec_channels,
segment_size,
inter_channels,
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size,
p_dropout,
resblock,
resblock_kernel_sizes,
resblock_dilation_sizes,
upsample_rates,
upsample_initial_channel,
upsample_kernel_sizes,
gin_channels,
ssl_dim,
n_speakers,
**kwargs):
super().__init__()
self.spec_channels = spec_channels
self.inter_channels = inter_channels
self.hidden_channels = hidden_channels
self.filter_channels = filter_channels
self.n_heads = n_heads
self.n_layers = n_layers
self.kernel_size = kernel_size
self.p_dropout = p_dropout
self.resblock = resblock
self.resblock_kernel_sizes = resblock_kernel_sizes
self.resblock_dilation_sizes = resblock_dilation_sizes
self.upsample_rates = upsample_rates
self.upsample_initial_channel = upsample_initial_channel
self.upsample_kernel_sizes = upsample_kernel_sizes
self.segment_size = segment_size
self.gin_channels = gin_channels
self.ssl_dim = ssl_dim
self.emb_g = nn.Embedding(n_speakers, gin_channels)
self.enc_p_ = TextEncoder(ssl_dim, inter_channels, hidden_channels, 5, 1, 16,0, filter_channels, n_heads, p_dropout)
hps = {
"sampling_rate": 32000,
"inter_channels": 192,
"resblock": "1",
"resblock_kernel_sizes": [3, 7, 11],
"resblock_dilation_sizes": [[1, 3, 5], [1, 3, 5], [1, 3, 5]],
"upsample_rates": [10, 8, 2, 2],
"upsample_initial_channel": 512,
"upsample_kernel_sizes": [16, 16, 4, 4],
"gin_channels": 256,
}
self.dec = Generator(h=hps)
self.enc_q = Encoder(spec_channels, inter_channels, hidden_channels, 5, 1, 16, gin_channels=gin_channels)
self.flow = ResidualCouplingBlock(inter_channels, hidden_channels, 5, 1, 4, gin_channels=gin_channels)
def forward(self, c, c_lengths, f0, g=None):
g = self.emb_g(g.unsqueeze(0)).transpose(1,2)
z_p, m_p, logs_p, c_mask = self.enc_p_(c.transpose(1,2), c_lengths, f0=f0_to_coarse(f0))
z = self.flow(z_p, c_mask, g=g, reverse=True)
o = self.dec(z * c_mask, g=g, f0=f0.float())
return o

@ -0,0 +1,328 @@
import copy
import math
import torch
from torch import nn
from torch.nn import functional as F
import attentions
import commons
import modules
from torch.nn import Conv1d, ConvTranspose1d, AvgPool1d, Conv2d
from torch.nn.utils import weight_norm, remove_weight_norm, spectral_norm
from commons import init_weights, get_padding
from vdecoder.hifigan.models import Generator
from utils import f0_to_coarse
class ResidualCouplingBlock(nn.Module):
def __init__(self,
channels,
hidden_channels,
kernel_size,
dilation_rate,
n_layers,
n_flows=4,
gin_channels=0):
super().__init__()
self.channels = channels
self.hidden_channels = hidden_channels
self.kernel_size = kernel_size
self.dilation_rate = dilation_rate
self.n_layers = n_layers
self.n_flows = n_flows
self.gin_channels = gin_channels
self.flows = nn.ModuleList()
for i in range(n_flows):
self.flows.append(modules.ResidualCouplingLayer(channels, hidden_channels, kernel_size, dilation_rate, n_layers, gin_channels=gin_channels, mean_only=True))
self.flows.append(modules.Flip())
def forward(self, x, x_mask, g=None, reverse=False):
if not reverse:
for flow in self.flows:
x, _ = flow(x, x_mask, g=g, reverse=reverse)
else:
for flow in reversed(self.flows):
x = flow(x, x_mask, g=g, reverse=reverse)
return x
class Encoder(nn.Module):
def __init__(self,
in_channels,
out_channels,
hidden_channels,
kernel_size,
dilation_rate,
n_layers,
gin_channels=0):
super().__init__()
self.in_channels = in_channels
self.out_channels = out_channels
self.hidden_channels = hidden_channels
self.kernel_size = kernel_size
self.dilation_rate = dilation_rate
self.n_layers = n_layers
self.gin_channels = gin_channels
self.pre = nn.Conv1d(in_channels, hidden_channels, 1)
self.enc = modules.WN(hidden_channels, kernel_size, dilation_rate, n_layers, gin_channels=gin_channels)
self.proj = nn.Conv1d(hidden_channels, out_channels * 2, 1)
def forward(self, x, x_lengths, g=None):
# print(x.shape,x_lengths.shape)
x_mask = torch.unsqueeze(commons.sequence_mask(x_lengths, x.size(2)), 1).to(x.dtype)
x = self.pre(x) * x_mask
x = self.enc(x, x_mask, g=g)
stats = self.proj(x) * x_mask
m, logs = torch.split(stats, self.out_channels, dim=1)
z = (m + torch.randn_like(m) * torch.exp(logs)) * x_mask
return z, m, logs, x_mask
class TextEncoder(nn.Module):
def __init__(self,
in_channels,
out_channels,
hidden_channels,
kernel_size,
dilation_rate,
n_layers,
gin_channels=0,
filter_channels=None,
n_heads=None,
p_dropout=None):
super().__init__()
self.in_channels = in_channels
self.out_channels = out_channels
self.hidden_channels = hidden_channels
self.kernel_size = kernel_size
self.dilation_rate = dilation_rate
self.n_layers = n_layers
self.gin_channels = gin_channels
self.pre = nn.Conv1d(in_channels, hidden_channels, 1)
self.proj = nn.Conv1d(hidden_channels, out_channels * 2, 1)
self.f0_emb = nn.Embedding(256, hidden_channels)
self.enc_ = attentions.Encoder(
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size,
p_dropout)
def forward(self, x, x_lengths, f0=None):
x_mask = torch.unsqueeze(commons.sequence_mask(x_lengths, x.size(2)), 1).to(x.dtype)
x = self.pre(x) * x_mask
x = x + self.f0_emb(f0.long()).transpose(1,2)
x = self.enc_(x * x_mask, x_mask)
stats = self.proj(x) * x_mask
m, logs = torch.split(stats, self.out_channels, dim=1)
z = (m + torch.randn_like(m) * torch.exp(logs)) * x_mask
return z, m, logs, x_mask
class DiscriminatorP(torch.nn.Module):
def __init__(self, period, kernel_size=5, stride=3, use_spectral_norm=False):
super(DiscriminatorP, self).__init__()
self.period = period
self.use_spectral_norm = use_spectral_norm
norm_f = weight_norm if use_spectral_norm == False else spectral_norm
self.convs = nn.ModuleList([
norm_f(Conv2d(1, 32, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))),
norm_f(Conv2d(32, 128, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))),
norm_f(Conv2d(128, 512, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))),
norm_f(Conv2d(512, 1024, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))),
norm_f(Conv2d(1024, 1024, (kernel_size, 1), 1, padding=(get_padding(kernel_size, 1), 0))),
])
self.conv_post = norm_f(Conv2d(1024, 1, (3, 1), 1, padding=(1, 0)))
def forward(self, x):
fmap = []
# 1d to 2d
b, c, t = x.shape
if t % self.period != 0: # pad first
n_pad = self.period - (t % self.period)
x = F.pad(x, (0, n_pad), "reflect")
t = t + n_pad
x = x.view(b, c, t // self.period, self.period)
for l in self.convs:
x = l(x)
x = F.leaky_relu(x, modules.LRELU_SLOPE)
fmap.append(x)
x = self.conv_post(x)
fmap.append(x)
x = torch.flatten(x, 1, -1)
return x, fmap
class DiscriminatorS(torch.nn.Module):
def __init__(self, use_spectral_norm=False):
super(DiscriminatorS, self).__init__()
norm_f = weight_norm if use_spectral_norm == False else spectral_norm
self.convs = nn.ModuleList([
norm_f(Conv1d(1, 16, 15, 1, padding=7)),
norm_f(Conv1d(16, 64, 41, 4, groups=4, padding=20)),
norm_f(Conv1d(64, 256, 41, 4, groups=16, padding=20)),
norm_f(Conv1d(256, 1024, 41, 4, groups=64, padding=20)),
norm_f(Conv1d(1024, 1024, 41, 4, groups=256, padding=20)),
norm_f(Conv1d(1024, 1024, 5, 1, padding=2)),
])
self.conv_post = norm_f(Conv1d(1024, 1, 3, 1, padding=1))
def forward(self, x):
fmap = []
for l in self.convs:
x = l(x)
x = F.leaky_relu(x, modules.LRELU_SLOPE)
fmap.append(x)
x = self.conv_post(x)
fmap.append(x)
x = torch.flatten(x, 1, -1)
return x, fmap
class MultiPeriodDiscriminator(torch.nn.Module):
def __init__(self, use_spectral_norm=False):
super(MultiPeriodDiscriminator, self).__init__()
periods = [2,3,5,7,11]
discs = [DiscriminatorS(use_spectral_norm=use_spectral_norm)]
discs = discs + [DiscriminatorP(i, use_spectral_norm=use_spectral_norm) for i in periods]
self.discriminators = nn.ModuleList(discs)
def forward(self, y, y_hat):
y_d_rs = []
y_d_gs = []
fmap_rs = []
fmap_gs = []
for i, d in enumerate(self.discriminators):
y_d_r, fmap_r = d(y)
y_d_g, fmap_g = d(y_hat)
y_d_rs.append(y_d_r)
y_d_gs.append(y_d_g)
fmap_rs.append(fmap_r)
fmap_gs.append(fmap_g)
return y_d_rs, y_d_gs, fmap_rs, fmap_gs
class SpeakerEncoder(torch.nn.Module):
def __init__(self, mel_n_channels=80, model_num_layers=3, model_hidden_size=256, model_embedding_size=256):
super(SpeakerEncoder, self).__init__()
self.lstm = nn.LSTM(mel_n_channels, model_hidden_size, model_num_layers, batch_first=True)
self.linear = nn.Linear(model_hidden_size, model_embedding_size)
self.relu = nn.ReLU()
def forward(self, mels):
self.lstm.flatten_parameters()
_, (hidden, _) = self.lstm(mels)
embeds_raw = self.relu(self.linear(hidden[-1]))
return embeds_raw / torch.norm(embeds_raw, dim=1, keepdim=True)
def compute_partial_slices(self, total_frames, partial_frames, partial_hop):
mel_slices = []
for i in range(0, total_frames-partial_frames, partial_hop):
mel_range = torch.arange(i, i+partial_frames)
mel_slices.append(mel_range)
return mel_slices
def embed_utterance(self, mel, partial_frames=128, partial_hop=64):
mel_len = mel.size(1)
last_mel = mel[:,-partial_frames:]
if mel_len > partial_frames:
mel_slices = self.compute_partial_slices(mel_len, partial_frames, partial_hop)
mels = list(mel[:,s] for s in mel_slices)
mels.append(last_mel)
mels = torch.stack(tuple(mels), 0).squeeze(1)
with torch.no_grad():
partial_embeds = self(mels)
embed = torch.mean(partial_embeds, axis=0).unsqueeze(0)
#embed = embed / torch.linalg.norm(embed, 2)
else:
with torch.no_grad():
embed = self(last_mel)
return embed
class SynthesizerTrn(nn.Module):
"""
Synthesizer for Training
"""
def __init__(self,
spec_channels,
segment_size,
inter_channels,
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size,
p_dropout,
resblock,
resblock_kernel_sizes,
resblock_dilation_sizes,
upsample_rates,
upsample_initial_channel,
upsample_kernel_sizes,
gin_channels,
ssl_dim,
n_speakers,
**kwargs):
super().__init__()
self.spec_channels = spec_channels
self.inter_channels = inter_channels
self.hidden_channels = hidden_channels
self.filter_channels = filter_channels
self.n_heads = n_heads
self.n_layers = n_layers
self.kernel_size = kernel_size
self.p_dropout = p_dropout
self.resblock = resblock
self.resblock_kernel_sizes = resblock_kernel_sizes
self.resblock_dilation_sizes = resblock_dilation_sizes
self.upsample_rates = upsample_rates
self.upsample_initial_channel = upsample_initial_channel
self.upsample_kernel_sizes = upsample_kernel_sizes
self.segment_size = segment_size
self.gin_channels = gin_channels
self.ssl_dim = ssl_dim
self.emb_g = nn.Embedding(n_speakers, gin_channels)
self.enc_p_ = TextEncoder(ssl_dim, inter_channels, hidden_channels, 5, 1, 16,0, filter_channels, n_heads, p_dropout)
hps = {
"sampling_rate": 48000,
"inter_channels": 192,
"resblock": "1",
"resblock_kernel_sizes": [3, 7, 11],
"resblock_dilation_sizes": [[1, 3, 5], [1, 3, 5], [1, 3, 5]],
"upsample_rates": [10, 8, 2, 2],
"upsample_initial_channel": 512,
"upsample_kernel_sizes": [16, 16, 4, 4],
"gin_channels": 256,
}
self.dec = Generator(h=hps)
self.enc_q = Encoder(spec_channels, inter_channels, hidden_channels, 5, 1, 16, gin_channels=gin_channels)
self.flow = ResidualCouplingBlock(inter_channels, hidden_channels, 5, 1, 4, gin_channels=gin_channels)
def forward(self, c, c_lengths, f0, g=None):
g = self.emb_g(g.unsqueeze(0)).transpose(1,2)
z_p, m_p, logs_p, c_mask = self.enc_p_(c.transpose(1,2), c_lengths, f0=f0_to_coarse(f0))
z = self.flow(z_p, c_mask, g=g, reverse=True)
o = self.dec(z * c_mask, g=g, f0=f0.float())
return o

@ -0,0 +1,351 @@
import copy
import math
import torch
from torch import nn
from torch.nn import functional as F
import attentions
import commons
import modules
from torch.nn import Conv1d, ConvTranspose1d, AvgPool1d, Conv2d
from torch.nn.utils import weight_norm, remove_weight_norm, spectral_norm
from commons import init_weights, get_padding
from vdecoder.hifigan.models import Generator
from utils import f0_to_coarse
class ResidualCouplingBlock(nn.Module):
def __init__(self,
channels,
hidden_channels,
kernel_size,
dilation_rate,
n_layers,
n_flows=4,
gin_channels=0):
super().__init__()
self.channels = channels
self.hidden_channels = hidden_channels
self.kernel_size = kernel_size
self.dilation_rate = dilation_rate
self.n_layers = n_layers
self.n_flows = n_flows
self.gin_channels = gin_channels
self.flows = nn.ModuleList()
for i in range(n_flows):
self.flows.append(modules.ResidualCouplingLayer(channels, hidden_channels, kernel_size, dilation_rate, n_layers, gin_channels=gin_channels, mean_only=True))
self.flows.append(modules.Flip())
def forward(self, x, x_mask, g=None, reverse=False):
if not reverse:
for flow in self.flows:
x, _ = flow(x, x_mask, g=g, reverse=reverse)
else:
for flow in reversed(self.flows):
x = flow(x, x_mask, g=g, reverse=reverse)
return x
class Encoder(nn.Module):
def __init__(self,
in_channels,
out_channels,
hidden_channels,
kernel_size,
dilation_rate,
n_layers,
gin_channels=0):
super().__init__()
self.in_channels = in_channels
self.out_channels = out_channels
self.hidden_channels = hidden_channels
self.kernel_size = kernel_size
self.dilation_rate = dilation_rate
self.n_layers = n_layers
self.gin_channels = gin_channels
self.pre = nn.Conv1d(in_channels, hidden_channels, 1)
self.enc = modules.WN(hidden_channels, kernel_size, dilation_rate, n_layers, gin_channels=gin_channels)
self.proj = nn.Conv1d(hidden_channels, out_channels * 2, 1)
def forward(self, x, x_lengths, g=None):
# print(x.shape,x_lengths.shape)
x_mask = torch.unsqueeze(commons.sequence_mask(x_lengths, x.size(2)), 1).to(x.dtype)
x = self.pre(x) * x_mask
x = self.enc(x, x_mask, g=g)
stats = self.proj(x) * x_mask
m, logs = torch.split(stats, self.out_channels, dim=1)
z = (m + torch.randn_like(m) * torch.exp(logs)) * x_mask
return z, m, logs, x_mask
class TextEncoder(nn.Module):
def __init__(self,
in_channels,
out_channels,
hidden_channels,
kernel_size,
dilation_rate,
n_layers,
gin_channels=0,
filter_channels=None,
n_heads=None,
p_dropout=None):
super().__init__()
self.in_channels = in_channels
self.out_channels = out_channels
self.hidden_channels = hidden_channels
self.kernel_size = kernel_size
self.dilation_rate = dilation_rate
self.n_layers = n_layers
self.gin_channels = gin_channels
self.pre = nn.Conv1d(in_channels, hidden_channels, 1)
self.proj = nn.Conv1d(hidden_channels, out_channels * 2, 1)
self.f0_emb = nn.Embedding(256, hidden_channels)
self.enc_ = attentions.Encoder(
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size,
p_dropout)
def forward(self, x, x_lengths, f0=None):
x_mask = torch.unsqueeze(commons.sequence_mask(x_lengths, x.size(2)), 1).to(x.dtype)
x = self.pre(x) * x_mask
x = x + self.f0_emb(f0).transpose(1,2)
x = self.enc_(x * x_mask, x_mask)
stats = self.proj(x) * x_mask
m, logs = torch.split(stats, self.out_channels, dim=1)
z = (m + torch.randn_like(m) * torch.exp(logs)) * x_mask
return z, m, logs, x_mask
class DiscriminatorP(torch.nn.Module):
def __init__(self, period, kernel_size=5, stride=3, use_spectral_norm=False):
super(DiscriminatorP, self).__init__()
self.period = period
self.use_spectral_norm = use_spectral_norm
norm_f = weight_norm if use_spectral_norm == False else spectral_norm
self.convs = nn.ModuleList([
norm_f(Conv2d(1, 32, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))),
norm_f(Conv2d(32, 128, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))),
norm_f(Conv2d(128, 512, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))),
norm_f(Conv2d(512, 1024, (kernel_size, 1), (stride, 1), padding=(get_padding(kernel_size, 1), 0))),
norm_f(Conv2d(1024, 1024, (kernel_size, 1), 1, padding=(get_padding(kernel_size, 1), 0))),
])
self.conv_post = norm_f(Conv2d(1024, 1, (3, 1), 1, padding=(1, 0)))
def forward(self, x):
fmap = []
# 1d to 2d
b, c, t = x.shape
if t % self.period != 0: # pad first
n_pad = self.period - (t % self.period)
x = F.pad(x, (0, n_pad), "reflect")
t = t + n_pad
x = x.view(b, c, t // self.period, self.period)
for l in self.convs:
x = l(x)
x = F.leaky_relu(x, modules.LRELU_SLOPE)
fmap.append(x)
x = self.conv_post(x)
fmap.append(x)
x = torch.flatten(x, 1, -1)
return x, fmap
class DiscriminatorS(torch.nn.Module):
def __init__(self, use_spectral_norm=False):
super(DiscriminatorS, self).__init__()
norm_f = weight_norm if use_spectral_norm == False else spectral_norm
self.convs = nn.ModuleList([
norm_f(Conv1d(1, 16, 15, 1, padding=7)),
norm_f(Conv1d(16, 64, 41, 4, groups=4, padding=20)),
norm_f(Conv1d(64, 256, 41, 4, groups=16, padding=20)),
norm_f(Conv1d(256, 1024, 41, 4, groups=64, padding=20)),
norm_f(Conv1d(1024, 1024, 41, 4, groups=256, padding=20)),
norm_f(Conv1d(1024, 1024, 5, 1, padding=2)),
])
self.conv_post = norm_f(Conv1d(1024, 1, 3, 1, padding=1))
def forward(self, x):
fmap = []
for l in self.convs:
x = l(x)
x = F.leaky_relu(x, modules.LRELU_SLOPE)
fmap.append(x)
x = self.conv_post(x)
fmap.append(x)
x = torch.flatten(x, 1, -1)
return x, fmap
class MultiPeriodDiscriminator(torch.nn.Module):
def __init__(self, use_spectral_norm=False):
super(MultiPeriodDiscriminator, self).__init__()
periods = [2,3,5,7,11]
discs = [DiscriminatorS(use_spectral_norm=use_spectral_norm)]
discs = discs + [DiscriminatorP(i, use_spectral_norm=use_spectral_norm) for i in periods]
self.discriminators = nn.ModuleList(discs)
def forward(self, y, y_hat):
y_d_rs = []
y_d_gs = []
fmap_rs = []
fmap_gs = []
for i, d in enumerate(self.discriminators):
y_d_r, fmap_r = d(y)
y_d_g, fmap_g = d(y_hat)
y_d_rs.append(y_d_r)
y_d_gs.append(y_d_g)
fmap_rs.append(fmap_r)
fmap_gs.append(fmap_g)
return y_d_rs, y_d_gs, fmap_rs, fmap_gs
class SpeakerEncoder(torch.nn.Module):
def __init__(self, mel_n_channels=80, model_num_layers=3, model_hidden_size=256, model_embedding_size=256):
super(SpeakerEncoder, self).__init__()
self.lstm = nn.LSTM(mel_n_channels, model_hidden_size, model_num_layers, batch_first=True)
self.linear = nn.Linear(model_hidden_size, model_embedding_size)
self.relu = nn.ReLU()
def forward(self, mels):
self.lstm.flatten_parameters()
_, (hidden, _) = self.lstm(mels)
embeds_raw = self.relu(self.linear(hidden[-1]))
return embeds_raw / torch.norm(embeds_raw, dim=1, keepdim=True)
def compute_partial_slices(self, total_frames, partial_frames, partial_hop):
mel_slices = []
for i in range(0, total_frames-partial_frames, partial_hop):
mel_range = torch.arange(i, i+partial_frames)
mel_slices.append(mel_range)
return mel_slices
def embed_utterance(self, mel, partial_frames=128, partial_hop=64):
mel_len = mel.size(1)
last_mel = mel[:,-partial_frames:]
if mel_len > partial_frames:
mel_slices = self.compute_partial_slices(mel_len, partial_frames, partial_hop)
mels = list(mel[:,s] for s in mel_slices)
mels.append(last_mel)
mels = torch.stack(tuple(mels), 0).squeeze(1)
with torch.no_grad():
partial_embeds = self(mels)
embed = torch.mean(partial_embeds, axis=0).unsqueeze(0)
#embed = embed / torch.linalg.norm(embed, 2)
else:
with torch.no_grad():
embed = self(last_mel)
return embed
class SynthesizerTrn(nn.Module):
"""
Synthesizer for Training
"""
def __init__(self,
spec_channels,
segment_size,
inter_channels,
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size,
p_dropout,
resblock,
resblock_kernel_sizes,
resblock_dilation_sizes,
upsample_rates,
upsample_initial_channel,
upsample_kernel_sizes,
gin_channels,
ssl_dim,
n_speakers,
**kwargs):
super().__init__()
self.spec_channels = spec_channels
self.inter_channels = inter_channels
self.hidden_channels = hidden_channels
self.filter_channels = filter_channels
self.n_heads = n_heads
self.n_layers = n_layers
self.kernel_size = kernel_size
self.p_dropout = p_dropout
self.resblock = resblock
self.resblock_kernel_sizes = resblock_kernel_sizes
self.resblock_dilation_sizes = resblock_dilation_sizes
self.upsample_rates = upsample_rates
self.upsample_initial_channel = upsample_initial_channel
self.upsample_kernel_sizes = upsample_kernel_sizes
self.segment_size = segment_size
self.gin_channels = gin_channels
self.ssl_dim = ssl_dim
self.emb_g = nn.Embedding(n_speakers, gin_channels)
self.enc_p_ = TextEncoder(ssl_dim, inter_channels, hidden_channels, 5, 1, 16,0, filter_channels, n_heads, p_dropout)
hps = {
"sampling_rate": 32000,
"inter_channels": 192,
"resblock": "1",
"resblock_kernel_sizes": [3, 7, 11],
"resblock_dilation_sizes": [[1, 3, 5], [1, 3, 5], [1, 3, 5]],
"upsample_rates": [10, 8, 2, 2],
"upsample_initial_channel": 512,
"upsample_kernel_sizes": [16, 16, 4, 4],
"gin_channels": 256,
}
self.dec = Generator(h=hps)
self.enc_q = Encoder(spec_channels, inter_channels, hidden_channels, 5, 1, 16, gin_channels=gin_channels)
self.flow = ResidualCouplingBlock(inter_channels, hidden_channels, 5, 1, 4, gin_channels=gin_channels)
def forward(self, c, f0, spec, g=None, mel=None, c_lengths=None, spec_lengths=None):
if c_lengths == None:
c_lengths = (torch.ones(c.size(0)) * c.size(-1)).to(c.device)
if spec_lengths == None:
spec_lengths = (torch.ones(spec.size(0)) * spec.size(-1)).to(spec.device)
g = self.emb_g(g).transpose(1,2)
z_ptemp, m_p, logs_p, _ = self.enc_p_(c, c_lengths, f0=f0_to_coarse(f0))
z, m_q, logs_q, spec_mask = self.enc_q(spec, spec_lengths, g=g)
z_p = self.flow(z, spec_mask, g=g)
z_slice, pitch_slice, ids_slice = commons.rand_slice_segments_with_pitch(z, f0, spec_lengths, self.segment_size)
# o = self.dec(z_slice, g=g)
o = self.dec(z_slice, g=g, f0=pitch_slice)
return o, ids_slice, spec_mask, (z, z_p, m_p, logs_p, m_q, logs_q)
def infer(self, c, f0, g=None, mel=None, c_lengths=None):
if c_lengths == None:
c_lengths = (torch.ones(c.size(0)) * c.size(-1)).to(c.device)
g = self.emb_g(g).transpose(1,2)
z_p, m_p, logs_p, c_mask = self.enc_p_(c, c_lengths, f0=f0_to_coarse(f0))
z = self.flow(z_p, c_mask, g=g, reverse=True)
o = self.dec(z * c_mask, g=g, f0=f0)
return o

@ -0,0 +1,342 @@
import copy
import math
import numpy as np
import scipy
import torch
from torch import nn
from torch.nn import functional as F
from torch.nn import Conv1d, ConvTranspose1d, AvgPool1d, Conv2d
from torch.nn.utils import weight_norm, remove_weight_norm
import commons
from commons import init_weights, get_padding
LRELU_SLOPE = 0.1
class LayerNorm(nn.Module):
def __init__(self, channels, eps=1e-5):
super().__init__()
self.channels = channels
self.eps = eps
self.gamma = nn.Parameter(torch.ones(channels))
self.beta = nn.Parameter(torch.zeros(channels))
def forward(self, x):
x = x.transpose(1, -1)
x = F.layer_norm(x, (self.channels,), self.gamma, self.beta, self.eps)
return x.transpose(1, -1)
class ConvReluNorm(nn.Module):
def __init__(self, in_channels, hidden_channels, out_channels, kernel_size, n_layers, p_dropout):
super().__init__()
self.in_channels = in_channels
self.hidden_channels = hidden_channels
self.out_channels = out_channels
self.kernel_size = kernel_size
self.n_layers = n_layers
self.p_dropout = p_dropout
assert n_layers > 1, "Number of layers should be larger than 0."
self.conv_layers = nn.ModuleList()
self.norm_layers = nn.ModuleList()
self.conv_layers.append(nn.Conv1d(in_channels, hidden_channels, kernel_size, padding=kernel_size//2))
self.norm_layers.append(LayerNorm(hidden_channels))
self.relu_drop = nn.Sequential(
nn.ReLU(),
nn.Dropout(p_dropout))
for _ in range(n_layers-1):
self.conv_layers.append(nn.Conv1d(hidden_channels, hidden_channels, kernel_size, padding=kernel_size//2))
self.norm_layers.append(LayerNorm(hidden_channels))
self.proj = nn.Conv1d(hidden_channels, out_channels, 1)
self.proj.weight.data.zero_()
self.proj.bias.data.zero_()
def forward(self, x, x_mask):
x_org = x
for i in range(self.n_layers):
x = self.conv_layers[i](x * x_mask)
x = self.norm_layers[i](x)
x = self.relu_drop(x)
x = x_org + self.proj(x)
return x * x_mask
class DDSConv(nn.Module):
"""
Dialted and Depth-Separable Convolution
"""
def __init__(self, channels, kernel_size, n_layers, p_dropout=0.):
super().__init__()
self.channels = channels
self.kernel_size = kernel_size
self.n_layers = n_layers
self.p_dropout = p_dropout
self.drop = nn.Dropout(p_dropout)
self.convs_sep = nn.ModuleList()
self.convs_1x1 = nn.ModuleList()
self.norms_1 = nn.ModuleList()
self.norms_2 = nn.ModuleList()
for i in range(n_layers):
dilation = kernel_size ** i
padding = (kernel_size * dilation - dilation) // 2
self.convs_sep.append(nn.Conv1d(channels, channels, kernel_size,
groups=channels, dilation=dilation, padding=padding
))
self.convs_1x1.append(nn.Conv1d(channels, channels, 1))
self.norms_1.append(LayerNorm(channels))
self.norms_2.append(LayerNorm(channels))
def forward(self, x, x_mask, g=None):
if g is not None:
x = x + g
for i in range(self.n_layers):
y = self.convs_sep[i](x * x_mask)
y = self.norms_1[i](y)
y = F.gelu(y)
y = self.convs_1x1[i](y)
y = self.norms_2[i](y)
y = F.gelu(y)
y = self.drop(y)
x = x + y
return x * x_mask
class WN(torch.nn.Module):
def __init__(self, hidden_channels, kernel_size, dilation_rate, n_layers, gin_channels=0, p_dropout=0):
super(WN, self).__init__()
assert(kernel_size % 2 == 1)
self.hidden_channels =hidden_channels
self.kernel_size = kernel_size,
self.dilation_rate = dilation_rate
self.n_layers = n_layers
self.gin_channels = gin_channels
self.p_dropout = p_dropout
self.in_layers = torch.nn.ModuleList()
self.res_skip_layers = torch.nn.ModuleList()
self.drop = nn.Dropout(p_dropout)
if gin_channels != 0:
cond_layer = torch.nn.Conv1d(gin_channels, 2*hidden_channels*n_layers, 1)
self.cond_layer = torch.nn.utils.weight_norm(cond_layer, name='weight')
for i in range(n_layers):
dilation = dilation_rate ** i
padding = int((kernel_size * dilation - dilation) / 2)
in_layer = torch.nn.Conv1d(hidden_channels, 2*hidden_channels, kernel_size,
dilation=dilation, padding=padding)
in_layer = torch.nn.utils.weight_norm(in_layer, name='weight')
self.in_layers.append(in_layer)
# last one is not necessary
if i < n_layers - 1:
res_skip_channels = 2 * hidden_channels
else:
res_skip_channels = hidden_channels
res_skip_layer = torch.nn.Conv1d(hidden_channels, res_skip_channels, 1)
res_skip_layer = torch.nn.utils.weight_norm(res_skip_layer, name='weight')
self.res_skip_layers.append(res_skip_layer)
def forward(self, x, x_mask, g=None, **kwargs):
output = torch.zeros_like(x)
n_channels_tensor = torch.IntTensor([self.hidden_channels])
if g is not None:
g = self.cond_layer(g)
for i in range(self.n_layers):
x_in = self.in_layers[i](x)
if g is not None:
cond_offset = i * 2 * self.hidden_channels
g_l = g[:,cond_offset:cond_offset+2*self.hidden_channels,:]
else:
g_l = torch.zeros_like(x_in)
acts = commons.fused_add_tanh_sigmoid_multiply(
x_in,
g_l,
n_channels_tensor)
acts = self.drop(acts)
res_skip_acts = self.res_skip_layers[i](acts)
if i < self.n_layers - 1:
res_acts = res_skip_acts[:,:self.hidden_channels,:]
x = (x + res_acts) * x_mask
output = output + res_skip_acts[:,self.hidden_channels:,:]
else:
output = output + res_skip_acts
return output * x_mask
def remove_weight_norm(self):
if self.gin_channels != 0:
torch.nn.utils.remove_weight_norm(self.cond_layer)
for l in self.in_layers:
torch.nn.utils.remove_weight_norm(l)
for l in self.res_skip_layers:
torch.nn.utils.remove_weight_norm(l)
class ResBlock1(torch.nn.Module):
def __init__(self, channels, kernel_size=3, dilation=(1, 3, 5)):
super(ResBlock1, self).__init__()
self.convs1 = nn.ModuleList([
weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[0],
padding=get_padding(kernel_size, dilation[0]))),
weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[1],
padding=get_padding(kernel_size, dilation[1]))),
weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[2],
padding=get_padding(kernel_size, dilation[2])))
])
self.convs1.apply(init_weights)
self.convs2 = nn.ModuleList([
weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=1,
padding=get_padding(kernel_size, 1))),
weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=1,
padding=get_padding(kernel_size, 1))),
weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=1,
padding=get_padding(kernel_size, 1)))
])
self.convs2.apply(init_weights)
def forward(self, x, x_mask=None):
for c1, c2 in zip(self.convs1, self.convs2):
xt = F.leaky_relu(x, LRELU_SLOPE)
if x_mask is not None:
xt = xt * x_mask
xt = c1(xt)
xt = F.leaky_relu(xt, LRELU_SLOPE)
if x_mask is not None:
xt = xt * x_mask
xt = c2(xt)
x = xt + x
if x_mask is not None:
x = x * x_mask
return x
def remove_weight_norm(self):
for l in self.convs1:
remove_weight_norm(l)
for l in self.convs2:
remove_weight_norm(l)
class ResBlock2(torch.nn.Module):
def __init__(self, channels, kernel_size=3, dilation=(1, 3)):
super(ResBlock2, self).__init__()
self.convs = nn.ModuleList([
weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[0],
padding=get_padding(kernel_size, dilation[0]))),
weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[1],
padding=get_padding(kernel_size, dilation[1])))
])
self.convs.apply(init_weights)
def forward(self, x, x_mask=None):
for c in self.convs:
xt = F.leaky_relu(x, LRELU_SLOPE)
if x_mask is not None:
xt = xt * x_mask
xt = c(xt)
x = xt + x
if x_mask is not None:
x = x * x_mask
return x
def remove_weight_norm(self):
for l in self.convs:
remove_weight_norm(l)
class Log(nn.Module):
def forward(self, x, x_mask, reverse=False, **kwargs):
if not reverse:
y = torch.log(torch.clamp_min(x, 1e-5)) * x_mask
logdet = torch.sum(-y, [1, 2])
return y, logdet
else:
x = torch.exp(x) * x_mask
return x
class Flip(nn.Module):
def forward(self, x, *args, reverse=False, **kwargs):
x = torch.flip(x, [1])
if not reverse:
logdet = torch.zeros(x.size(0)).to(dtype=x.dtype, device=x.device)
return x, logdet
else:
return x
class ElementwiseAffine(nn.Module):
def __init__(self, channels):
super().__init__()
self.channels = channels
self.m = nn.Parameter(torch.zeros(channels,1))
self.logs = nn.Parameter(torch.zeros(channels,1))
def forward(self, x, x_mask, reverse=False, **kwargs):
if not reverse:
y = self.m + torch.exp(self.logs) * x
y = y * x_mask
logdet = torch.sum(self.logs * x_mask, [1,2])
return y, logdet
else:
x = (x - self.m) * torch.exp(-self.logs) * x_mask
return x
class ResidualCouplingLayer(nn.Module):
def __init__(self,
channels,
hidden_channels,
kernel_size,
dilation_rate,
n_layers,
p_dropout=0,
gin_channels=0,
mean_only=False):
assert channels % 2 == 0, "channels should be divisible by 2"
super().__init__()
self.channels = channels
self.hidden_channels = hidden_channels
self.kernel_size = kernel_size
self.dilation_rate = dilation_rate
self.n_layers = n_layers
self.half_channels = channels // 2
self.mean_only = mean_only
self.pre = nn.Conv1d(self.half_channels, hidden_channels, 1)
self.enc = WN(hidden_channels, kernel_size, dilation_rate, n_layers, p_dropout=p_dropout, gin_channels=gin_channels)
self.post = nn.Conv1d(hidden_channels, self.half_channels * (2 - mean_only), 1)
self.post.weight.data.zero_()
self.post.bias.data.zero_()
def forward(self, x, x_mask, g=None, reverse=False):
x0, x1 = torch.split(x, [self.half_channels]*2, 1)
h = self.pre(x0) * x_mask
h = self.enc(h, x_mask, g=g)
stats = self.post(h) * x_mask
if not self.mean_only:
m, logs = torch.split(stats, [self.half_channels]*2, 1)
else:
m = stats
logs = torch.zeros_like(m)
if not reverse:
x1 = m + x1 * torch.exp(logs) * x_mask
x = torch.cat([x0, x1], 1)
logdet = torch.sum(logs, [1,2])
return x, logdet
else:
x1 = (x1 - m) * torch.exp(-logs) * x_mask
x = torch.cat([x0, x1], 1)
return x

@ -0,0 +1,73 @@
import argparse
import time
import numpy as np
import onnx
from onnxsim import simplify
import onnxruntime as ort
import onnxoptimizer
import torch
from model_onnx import SynthesizerTrn
import utils
from hubert import hubert_model_onnx
def main(HubertExport,NetExport):
path = "NyaruTaffy"
if(HubertExport):
device = torch.device("cuda")
hubert_soft = hubert_model_onnx.hubert_soft("hubert/model.pt")
test_input = torch.rand(1, 1, 16000)
input_names = ["source"]
output_names = ["embed"]
torch.onnx.export(hubert_soft.to(device),
test_input.to(device),
"hubert3.0.onnx",
dynamic_axes={
"source": {
2: "sample_length"
}
},
verbose=False,
opset_version=13,
input_names=input_names,
output_names=output_names)
if(NetExport):
device = torch.device("cuda")
hps = utils.get_hparams_from_file(f"checkpoints/{path}/config.json")
SVCVITS = SynthesizerTrn(
hps.data.filter_length // 2 + 1,
hps.train.segment_size // hps.data.hop_length,
**hps.model)
_ = utils.load_checkpoint(f"checkpoints/{path}/model.pth", SVCVITS, None)
_ = SVCVITS.eval().to(device)
for i in SVCVITS.parameters():
i.requires_grad = False
test_hidden_unit = torch.rand(1, 50, 256)
test_lengths = torch.LongTensor([50])
test_pitch = torch.rand(1, 50)
test_sid = torch.LongTensor([0])
input_names = ["hidden_unit", "lengths", "pitch", "sid"]
output_names = ["audio", ]
SVCVITS.eval()
torch.onnx.export(SVCVITS,
(
test_hidden_unit.to(device),
test_lengths.to(device),
test_pitch.to(device),
test_sid.to(device)
),
f"checkpoints/{path}/model.onnx",
dynamic_axes={
"hidden_unit": [0, 1],
"pitch": [1]
},
do_constant_folding=False,
opset_version=16,
verbose=False,
input_names=input_names,
output_names=output_names)
if __name__ == '__main__':
main(False,True)

@ -0,0 +1,73 @@
import argparse
import time
import numpy as np
import onnx
from onnxsim import simplify
import onnxruntime as ort
import onnxoptimizer
import torch
from model_onnx_48k import SynthesizerTrn
import utils
from hubert import hubert_model_onnx
def main(HubertExport,NetExport):
path = "NyaruTaffy"
if(HubertExport):
device = torch.device("cuda")
hubert_soft = hubert_model_onnx.hubert_soft("hubert/model.pt")
test_input = torch.rand(1, 1, 16000)
input_names = ["source"]
output_names = ["embed"]
torch.onnx.export(hubert_soft.to(device),
test_input.to(device),
"hubert3.0.onnx",
dynamic_axes={
"source": {
2: "sample_length"
}
},
verbose=False,
opset_version=13,
input_names=input_names,
output_names=output_names)
if(NetExport):
device = torch.device("cuda")
hps = utils.get_hparams_from_file(f"checkpoints/{path}/config.json")
SVCVITS = SynthesizerTrn(
hps.data.filter_length // 2 + 1,
hps.train.segment_size // hps.data.hop_length,
**hps.model)
_ = utils.load_checkpoint(f"checkpoints/{path}/model.pth", SVCVITS, None)
_ = SVCVITS.eval().to(device)
for i in SVCVITS.parameters():
i.requires_grad = False
test_hidden_unit = torch.rand(1, 50, 256)
test_lengths = torch.LongTensor([50])
test_pitch = torch.rand(1, 50)
test_sid = torch.LongTensor([0])
input_names = ["hidden_unit", "lengths", "pitch", "sid"]
output_names = ["audio", ]
SVCVITS.eval()
torch.onnx.export(SVCVITS,
(
test_hidden_unit.to(device),
test_lengths.to(device),
test_pitch.to(device),
test_sid.to(device)
),
f"checkpoints/{path}/model.onnx",
dynamic_axes={
"hidden_unit": [0, 1],
"pitch": [1]
},
do_constant_folding=False,
opset_version=16,
verbose=False,
input_names=input_names,
output_names=output_names)
if __name__ == '__main__':
main(False,True)

@ -0,0 +1,125 @@
import os
import argparse
import re
from tqdm import tqdm
from random import shuffle
import json
config_template = {
"train": {
"log_interval": 200,
"eval_interval": 1000,
"seed": 1234,
"epochs": 10000,
"learning_rate": 1e-4,
"betas": [0.8, 0.99],
"eps": 1e-9,
"batch_size": 12,
"fp16_run": False,
"lr_decay": 0.999875,
"segment_size": 17920,
"init_lr_ratio": 1,
"warmup_epochs": 0,
"c_mel": 45,
"c_kl": 1.0,
"use_sr": True,
"max_speclen": 384,
"port": "8001"
},
"data": {
"training_files":"filelists/train.txt",
"validation_files":"filelists/val.txt",
"max_wav_value": 32768.0,
"sampling_rate": 32000,
"filter_length": 1280,
"hop_length": 320,
"win_length": 1280,
"n_mel_channels": 80,
"mel_fmin": 0.0,
"mel_fmax": None
},
"model": {
"inter_channels": 192,
"hidden_channels": 192,
"filter_channels": 768,
"n_heads": 2,
"n_layers": 6,
"kernel_size": 3,
"p_dropout": 0.1,
"resblock": "1",
"resblock_kernel_sizes": [3,7,11],
"resblock_dilation_sizes": [[1,3,5], [1,3,5], [1,3,5]],
"upsample_rates": [10,8,2,2],
"upsample_initial_channel": 512,
"upsample_kernel_sizes": [16,16,4,4],
"n_layers_q": 3,
"use_spectral_norm": False,
"gin_channels": 256,
"ssl_dim": 256,
"n_speakers": 0,
},
"spk":{
"nen": 0,
"paimon": 1,
"yunhao": 2
}
}
pattern = re.compile(r'^[\.a-zA-Z0-9_\/]+$')
if __name__ == "__main__":
parser = argparse.ArgumentParser()
parser.add_argument("--train_list", type=str, default="./filelists/train.txt", help="path to train list")
parser.add_argument("--val_list", type=str, default="./filelists/val.txt", help="path to val list")
parser.add_argument("--test_list", type=str, default="./filelists/test.txt", help="path to test list")
parser.add_argument("--source_dir", type=str, default="./dataset/32k", help="path to source dir")
args = parser.parse_args()
train = []
val = []
test = []
idx = 0
spk_dict = {}
spk_id = 0
for speaker in tqdm(os.listdir(args.source_dir)):
spk_dict[speaker] = spk_id
spk_id += 1
wavs = ["/".join([args.source_dir, speaker, i]) for i in os.listdir(os.path.join(args.source_dir, speaker))]
for wavpath in wavs:
if not pattern.match(wavpath):
print(f"warning文件名{wavpath}中包含非字母数字下划线,可能会导致错误。(也可能不会)")
if len(wavs) < 10:
print(f"warning{speaker}数据集数量小于10条请补充数据")
wavs = [i for i in wavs if i.endswith("wav")]
shuffle(wavs)
train += wavs[2:-2]
val += wavs[:2]
test += wavs[-2:]
n_speakers = len(spk_dict.keys())*2
shuffle(train)
shuffle(val)
shuffle(test)
print("Writing", args.train_list)
with open(args.train_list, "w") as f:
for fname in tqdm(train):
wavpath = fname
f.write(wavpath + "\n")
print("Writing", args.val_list)
with open(args.val_list, "w") as f:
for fname in tqdm(val):
wavpath = fname
f.write(wavpath + "\n")
print("Writing", args.test_list)
with open(args.test_list, "w") as f:
for fname in tqdm(test):
wavpath = fname
f.write(wavpath + "\n")
config_template["model"]["n_speakers"] = n_speakers
config_template["spk"] = spk_dict
print("Writing configs/config.json")
with open("configs/config.json", "w") as f:
json.dump(config_template, f, indent=2)

@ -0,0 +1,106 @@
import os
import argparse
import torch
import json
from glob import glob
from pyworld import pyworld
from tqdm import tqdm
from scipy.io import wavfile
import utils
from mel_processing import mel_spectrogram_torch
#import h5py
import logging
logging.getLogger('numba').setLevel(logging.WARNING)
import parselmouth
import librosa
import numpy as np
def get_f0(path,p_len=None, f0_up_key=0):
x, _ = librosa.load(path, 32000)
if p_len is None:
p_len = x.shape[0]//320
else:
assert abs(p_len-x.shape[0]//320) < 3, (path, p_len, x.shape)
time_step = 320 / 32000 * 1000
f0_min = 50
f0_max = 1100
f0_mel_min = 1127 * np.log(1 + f0_min / 700)
f0_mel_max = 1127 * np.log(1 + f0_max / 700)
f0 = parselmouth.Sound(x, 32000).to_pitch_ac(
time_step=time_step / 1000, voicing_threshold=0.6,
pitch_floor=f0_min, pitch_ceiling=f0_max).selected_array['frequency']
pad_size=(p_len - len(f0) + 1) // 2
if(pad_size>0 or p_len - len(f0) - pad_size>0):
f0 = np.pad(f0,[[pad_size,p_len - len(f0) - pad_size]], mode='constant')
f0bak = f0.copy()
f0 *= pow(2, f0_up_key / 12)
f0_mel = 1127 * np.log(1 + f0 / 700)
f0_mel[f0_mel > 0] = (f0_mel[f0_mel > 0] - f0_mel_min) * 254 / (f0_mel_max - f0_mel_min) + 1
f0_mel[f0_mel <= 1] = 1
f0_mel[f0_mel > 255] = 255
f0_coarse = np.rint(f0_mel).astype(np.int)
return f0_coarse, f0bak
def resize2d(x, target_len):
source = np.array(x)
source[source<0.001] = np.nan
target = np.interp(np.arange(0, len(source)*target_len, len(source))/ target_len, np.arange(0, len(source)), source)
res = np.nan_to_num(target)
return res
def compute_f0(path, c_len):
x, sr = librosa.load(path, sr=32000)
f0, t = pyworld.dio(
x.astype(np.double),
fs=sr,
f0_ceil=800,
frame_period=1000 * 320 / sr,
)
f0 = pyworld.stonemask(x.astype(np.double), f0, t, 32000)
for index, pitch in enumerate(f0):
f0[index] = round(pitch, 1)
assert abs(c_len - x.shape[0]//320) < 3, (c_len, f0.shape)
return None, resize2d(f0, c_len)
def process(filename):
print(filename)
save_name = filename+".soft.pt"
if not os.path.exists(save_name):
devive = torch.device("cuda" if torch.cuda.is_available() else "cpu")
wav, _ = librosa.load(filename, sr=16000)
wav = torch.from_numpy(wav).unsqueeze(0).to(devive)
c = utils.get_hubert_content(hmodel, wav)
torch.save(c.cpu(), save_name)
else:
c = torch.load(save_name)
f0path = filename+".f0.npy"
if not os.path.exists(f0path):
cf0, f0 = compute_f0(filename, c.shape[-1] * 2)
np.save(f0path, f0)
if __name__ == "__main__":
parser = argparse.ArgumentParser()
parser.add_argument("--in_dir", type=str, default="dataset/32k", help="path to input dir")
args = parser.parse_args()
print("Loading hubert for content...")
hmodel = utils.get_hubert_model(0 if torch.cuda.is_available() else None)
print("Loaded hubert.")
filenames = glob(f'{args.in_dir}/*/*.wav', recursive=True)#[:10]
for filename in tqdm(filenames):
process(filename)

@ -0,0 +1,21 @@
Flask==2.1.2
Flask_Cors==3.0.10
gradio==3.4.1
numpy==1.19.2
playsound==1.3.0
PyAudio==0.2.12
pydub==0.25.1
pyworld==0.3.0
requests==2.28.1
scipy==1.7.3
sounddevice==0.4.5
SoundFile==0.10.3.post1
starlette==0.19.1
torch==1.10.0+cu113
torchaudio==0.10.0+cu113
tqdm==4.63.0
scikit-maad
praat-parselmouth
onnx
onnxsim
onnxoptimizer

@ -0,0 +1,48 @@
import os
import argparse
import librosa
import numpy as np
from multiprocessing import Pool, cpu_count
from scipy.io import wavfile
from tqdm import tqdm
def process(item):
spkdir, wav_name, args = item
# speaker 's5', 'p280', 'p315' are excluded,
speaker = spkdir.replace("\\", "/").split("/")[-1]
wav_path = os.path.join(args.in_dir, speaker, wav_name)
if os.path.exists(wav_path) and '.wav' in wav_path:
os.makedirs(os.path.join(args.out_dir2, speaker), exist_ok=True)
wav, sr = librosa.load(wav_path, None)
wav, _ = librosa.effects.trim(wav, top_db=20)
peak = np.abs(wav).max()
if peak > 1.0:
wav = 0.98 * wav / peak
wav2 = librosa.resample(wav, orig_sr=sr, target_sr=args.sr2)
wav2 /= max(wav2.max(), -wav2.min())
save_name = wav_name
save_path2 = os.path.join(args.out_dir2, speaker, save_name)
wavfile.write(
save_path2,
args.sr2,
(wav2 * np.iinfo(np.int16).max).astype(np.int16)
)
if __name__ == "__main__":
parser = argparse.ArgumentParser()
parser.add_argument("--sr2", type=int, default=32000, help="sampling rate")
parser.add_argument("--in_dir", type=str, default="./dataset_raw", help="path to source dir")
parser.add_argument("--out_dir2", type=str, default="./dataset/32k", help="path to target dir")
args = parser.parse_args()
processs = cpu_count()-2 if cpu_count() >4 else 1
pool = Pool(processes=processs)
for speaker in os.listdir(args.in_dir):
spk_dir = os.path.join(args.in_dir, speaker)
if os.path.isdir(spk_dir):
print(spk_dir)
for _ in tqdm(pool.imap_unordered(process, [(spk_dir, i, args) for i in os.listdir(spk_dir) if i.endswith("wav")])):
pass

@ -0,0 +1,47 @@
from inference.infer_tool_grad import VitsSvc
import gradio as gr
import os
class VitsGradio:
def __init__(self):
self.so = VitsSvc()
self.lspk = []
self.modelPaths = []
for root,dirs,files in os.walk("checkpoints"):
for dir in dirs:
self.modelPaths.append(dir)
with gr.Blocks() as self.Vits:
with gr.Tab("VoiceConversion"):
with gr.Row(visible=False) as self.VoiceConversion:
with gr.Column():
with gr.Row():
with gr.Column():
self.srcaudio = gr.Audio(label = "输入音频")
self.btnVC = gr.Button("说话人转换")
with gr.Column():
self.dsid = gr.Dropdown(label = "目标角色", choices = self.lspk)
self.tran = gr.Slider(label = "升降调", maximum = 60, minimum = -60, step = 1, value = 0)
self.th = gr.Slider(label = "切片阈值", maximum = 32767, minimum = -32768, step = 0.1, value = -40)
with gr.Row():
self.VCOutputs = gr.Audio()
self.btnVC.click(self.so.inference, inputs=[self.srcaudio,self.dsid,self.tran,self.th], outputs=[self.VCOutputs])
with gr.Tab("SelectModel"):
with gr.Column():
modelstrs = gr.Dropdown(label = "模型", choices = self.modelPaths, value = self.modelPaths[0], type = "value")
devicestrs = gr.Dropdown(label = "设备", choices = ["cpu","cuda"], value = "cpu", type = "value")
btnMod = gr.Button("载入模型")
btnMod.click(self.loadModel, inputs=[modelstrs,devicestrs], outputs = [self.dsid,self.VoiceConversion])
def loadModel(self, path, device):
self.lspk = []
self.so.set_device(device)
self.so.loadCheckpoint(path)
for spk, sid in self.so.hps.spk.items():
self.lspk.append(spk)
VChange = gr.update(visible = True)
SDChange = gr.update(choices = self.lspk, value = self.lspk[0])
return [SDChange,VChange]
grVits = VitsGradio()
grVits.Vits.launch()

@ -0,0 +1,22 @@
from data_utils import TextAudioSpeakerLoader, EvalDataLoader
import json
from tqdm import tqdm
from utils import HParams
config_path = 'configs/config.json'
with open(config_path, "r") as f:
data = f.read()
config = json.loads(data)
hps = HParams(**config)
train_dataset = TextAudioSpeakerLoader("filelists/train.txt", hps)
test_dataset = TextAudioSpeakerLoader("filelists/test.txt", hps)
eval_dataset = TextAudioSpeakerLoader("filelists/val.txt", hps)
for _ in tqdm(train_dataset):
pass
for _ in tqdm(eval_dataset):
pass
for _ in tqdm(test_dataset):
pass

@ -0,0 +1,281 @@
import logging
logging.getLogger('matplotlib').setLevel(logging.WARNING)
import os
import json
import argparse
import itertools
import math
import torch
from torch import nn, optim
from torch.nn import functional as F
from torch.utils.data import DataLoader
from torch.utils.tensorboard import SummaryWriter
import torch.multiprocessing as mp
import torch.distributed as dist
from torch.nn.parallel import DistributedDataParallel as DDP
from torch.cuda.amp import autocast, GradScaler
import commons
import utils
from data_utils import TextAudioSpeakerLoader, EvalDataLoader
from models import (
SynthesizerTrn,
MultiPeriodDiscriminator,
)
from losses import (
kl_loss,
generator_loss, discriminator_loss, feature_loss
)
from mel_processing import mel_spectrogram_torch, spec_to_mel_torch
torch.backends.cudnn.benchmark = True
global_step = 0
# os.environ['TORCH_DISTRIBUTED_DEBUG'] = 'INFO'
def main():
"""Assume Single Node Multi GPUs Training Only"""
assert torch.cuda.is_available(), "CPU training is not allowed."
hps = utils.get_hparams()
n_gpus = torch.cuda.device_count()
os.environ['MASTER_ADDR'] = 'localhost'
os.environ['MASTER_PORT'] = hps.train.port
mp.spawn(run, nprocs=n_gpus, args=(n_gpus, hps,))
def run(rank, n_gpus, hps):
global global_step
if rank == 0:
logger = utils.get_logger(hps.model_dir)
logger.info(hps)
utils.check_git_hash(hps.model_dir)
writer = SummaryWriter(log_dir=hps.model_dir)
writer_eval = SummaryWriter(log_dir=os.path.join(hps.model_dir, "eval"))
dist.init_process_group(backend='nccl', init_method='env://', world_size=n_gpus, rank=rank)
torch.manual_seed(hps.train.seed)
torch.cuda.set_device(rank)
train_dataset = TextAudioSpeakerLoader(hps.data.training_files, hps)
train_loader = DataLoader(train_dataset, num_workers=8, shuffle=False, pin_memory=True,
batch_size=hps.train.batch_size)
if rank == 0:
eval_dataset = EvalDataLoader(hps.data.validation_files, hps)
eval_loader = DataLoader(eval_dataset, num_workers=1, shuffle=False,
batch_size=1, pin_memory=False,
drop_last=False)
net_g = SynthesizerTrn(
hps.data.filter_length // 2 + 1,
hps.train.segment_size // hps.data.hop_length,
**hps.model).cuda(rank)
net_d = MultiPeriodDiscriminator(hps.model.use_spectral_norm).cuda(rank)
optim_g = torch.optim.AdamW(
net_g.parameters(),
hps.train.learning_rate,
betas=hps.train.betas,
eps=hps.train.eps)
optim_d = torch.optim.AdamW(
net_d.parameters(),
hps.train.learning_rate,
betas=hps.train.betas,
eps=hps.train.eps)
net_g = DDP(net_g, device_ids=[rank]) # , find_unused_parameters=True)
net_d = DDP(net_d, device_ids=[rank])
try:
_, _, _, epoch_str = utils.load_checkpoint(utils.latest_checkpoint_path(hps.model_dir, "G_*.pth"), net_g,
optim_g)
_, _, _, epoch_str = utils.load_checkpoint(utils.latest_checkpoint_path(hps.model_dir, "D_*.pth"), net_d,
optim_d)
global_step = (epoch_str - 1) * len(train_loader)
except:
epoch_str = 1
global_step = 0
scheduler_g = torch.optim.lr_scheduler.ExponentialLR(optim_g, gamma=hps.train.lr_decay, last_epoch=epoch_str - 2)
scheduler_d = torch.optim.lr_scheduler.ExponentialLR(optim_d, gamma=hps.train.lr_decay, last_epoch=epoch_str - 2)
scaler = GradScaler(enabled=hps.train.fp16_run)
for epoch in range(epoch_str, hps.train.epochs + 1):
if rank == 0:
train_and_evaluate(rank, epoch, hps, [net_g, net_d], [optim_g, optim_d], [scheduler_g, scheduler_d], scaler,
[train_loader, eval_loader], logger, [writer, writer_eval])
else:
train_and_evaluate(rank, epoch, hps, [net_g, net_d], [optim_g, optim_d], [scheduler_g, scheduler_d], scaler,
[train_loader, None], None, None)
scheduler_g.step()
scheduler_d.step()
def train_and_evaluate(rank, epoch, hps, nets, optims, schedulers, scaler, loaders, logger, writers):
net_g, net_d = nets
optim_g, optim_d = optims
scheduler_g, scheduler_d = schedulers
train_loader, eval_loader = loaders
if writers is not None:
writer, writer_eval = writers
# train_loader.batch_sampler.set_epoch(epoch)
global global_step
net_g.train()
net_d.train()
for batch_idx, items in enumerate(train_loader):
c, f0, spec, y, spk = items
g = spk.cuda(rank, non_blocking=True)
spec, y = spec.cuda(rank, non_blocking=True), y.cuda(rank, non_blocking=True)
c = c.cuda(rank, non_blocking=True)
f0 = f0.cuda(rank, non_blocking=True)
mel = spec_to_mel_torch(
spec,
hps.data.filter_length,
hps.data.n_mel_channels,
hps.data.sampling_rate,
hps.data.mel_fmin,
hps.data.mel_fmax)
with autocast(enabled=hps.train.fp16_run):
y_hat, ids_slice, z_mask, \
(z, z_p, m_p, logs_p, m_q, logs_q) = net_g(c, f0, spec, g=g, mel=mel)
y_mel = commons.slice_segments(mel, ids_slice, hps.train.segment_size // hps.data.hop_length)
y_hat_mel = mel_spectrogram_torch(
y_hat.squeeze(1),
hps.data.filter_length,
hps.data.n_mel_channels,
hps.data.sampling_rate,
hps.data.hop_length,
hps.data.win_length,
hps.data.mel_fmin,
hps.data.mel_fmax
)
y = commons.slice_segments(y, ids_slice * hps.data.hop_length, hps.train.segment_size) # slice
# Discriminator
y_d_hat_r, y_d_hat_g, _, _ = net_d(y, y_hat.detach())
with autocast(enabled=False):
loss_disc, losses_disc_r, losses_disc_g = discriminator_loss(y_d_hat_r, y_d_hat_g)
loss_disc_all = loss_disc
optim_d.zero_grad()
scaler.scale(loss_disc_all).backward()
scaler.unscale_(optim_d)
grad_norm_d = commons.clip_grad_value_(net_d.parameters(), None)
scaler.step(optim_d)
with autocast(enabled=hps.train.fp16_run):
# Generator
y_d_hat_r, y_d_hat_g, fmap_r, fmap_g = net_d(y, y_hat)
with autocast(enabled=False):
loss_mel = F.l1_loss(y_mel, y_hat_mel) * hps.train.c_mel
loss_kl = kl_loss(z_p, logs_q, m_p, logs_p, z_mask) * hps.train.c_kl
loss_fm = feature_loss(fmap_r, fmap_g)
loss_gen, losses_gen = generator_loss(y_d_hat_g)
loss_gen_all = loss_gen + loss_fm + loss_mel + loss_kl
optim_g.zero_grad()
scaler.scale(loss_gen_all).backward()
scaler.unscale_(optim_g)
grad_norm_g = commons.clip_grad_value_(net_g.parameters(), None)
scaler.step(optim_g)
scaler.update()
if rank == 0:
if global_step % hps.train.log_interval == 0:
lr = optim_g.param_groups[0]['lr']
losses = [loss_disc, loss_gen, loss_fm, loss_mel, loss_kl]
logger.info('Train Epoch: {} [{:.0f}%]'.format(
epoch,
100. * batch_idx / len(train_loader)))
logger.info([x.item() for x in losses] + [global_step, lr])
scalar_dict = {"loss/g/total": loss_gen_all, "loss/d/total": loss_disc_all, "learning_rate": lr,
"grad_norm_d": grad_norm_d, "grad_norm_g": grad_norm_g}
scalar_dict.update({"loss/g/fm": loss_fm, "loss/g/mel": loss_mel, "loss/g/kl": loss_kl})
scalar_dict.update({"loss/g/{}".format(i): v for i, v in enumerate(losses_gen)})
scalar_dict.update({"loss/d_r/{}".format(i): v for i, v in enumerate(losses_disc_r)})
scalar_dict.update({"loss/d_g/{}".format(i): v for i, v in enumerate(losses_disc_g)})
image_dict = {
"slice/mel_org": utils.plot_spectrogram_to_numpy(y_mel[0].data.cpu().numpy()),
"slice/mel_gen": utils.plot_spectrogram_to_numpy(y_hat_mel[0].data.cpu().numpy()),
"all/mel": utils.plot_spectrogram_to_numpy(mel[0].data.cpu().numpy()),
}
utils.summarize(
writer=writer,
global_step=global_step,
images=image_dict,
scalars=scalar_dict
)
if global_step % hps.train.eval_interval == 0:
evaluate(hps, net_g, eval_loader, writer_eval)
utils.save_checkpoint(net_g, optim_g, hps.train.learning_rate, epoch,
os.path.join(hps.model_dir, "G_{}.pth".format(global_step)))
utils.save_checkpoint(net_d, optim_d, hps.train.learning_rate, epoch,
os.path.join(hps.model_dir, "D_{}.pth".format(global_step)))
global_step += 1
if rank == 0:
logger.info('====> Epoch: {}'.format(epoch))
def evaluate(hps, generator, eval_loader, writer_eval):
generator.eval()
image_dict = {}
audio_dict = {}
with torch.no_grad():
for batch_idx, items in enumerate(eval_loader):
c, f0, spec, y, spk = items
g = spk[:1].cuda(0)
spec, y = spec[:1].cuda(0), y[:1].cuda(0)
c = c[:1].cuda(0)
f0 = f0[:1].cuda(0)
mel = spec_to_mel_torch(
spec,
hps.data.filter_length,
hps.data.n_mel_channels,
hps.data.sampling_rate,
hps.data.mel_fmin,
hps.data.mel_fmax)
y_hat = generator.module.infer(c, f0, g=g, mel=mel)
y_hat_mel = mel_spectrogram_torch(
y_hat.squeeze(1).float(),
hps.data.filter_length,
hps.data.n_mel_channels,
hps.data.sampling_rate,
hps.data.hop_length,
hps.data.win_length,
hps.data.mel_fmin,
hps.data.mel_fmax
)
audio_dict.update({
f"gen/audio_{batch_idx}": y_hat[0],
f"gt/audio_{batch_idx}": y[0]
})
image_dict.update({
f"gen/mel": utils.plot_spectrogram_to_numpy(y_hat_mel[0].cpu().numpy()),
"gt/mel": utils.plot_spectrogram_to_numpy(mel[0].cpu().numpy())
})
utils.summarize(
writer=writer_eval,
global_step=global_step,
images=image_dict,
audios=audio_dict,
audio_sampling_rate=hps.data.sampling_rate
)
generator.train()
if __name__ == "__main__":
main()

@ -0,0 +1,355 @@
import os
import glob
import re
import sys
import argparse
import logging
import json
import subprocess
import librosa
import numpy as np
import torchaudio
from scipy.io.wavfile import read
import torch
import torchvision
from torch.nn import functional as F
from commons import sequence_mask
from hubert import hubert_model
MATPLOTLIB_FLAG = False
logging.basicConfig(stream=sys.stdout, level=logging.DEBUG)
logger = logging
f0_bin = 256
f0_max = 1100.0
f0_min = 50.0
f0_mel_min = 1127 * np.log(1 + f0_min / 700)
f0_mel_max = 1127 * np.log(1 + f0_max / 700)
def f0_to_coarse(f0):
is_torch = isinstance(f0, torch.Tensor)
f0_mel = 1127 * (1 + f0 / 700).log() if is_torch else 1127 * np.log(1 + f0 / 700)
f0_mel[f0_mel > 0] = (f0_mel[f0_mel > 0] - f0_mel_min) * (f0_bin - 2) / (f0_mel_max - f0_mel_min) + 1
f0_mel[f0_mel <= 1] = 1
f0_mel[f0_mel > f0_bin - 1] = f0_bin - 1
f0_coarse = (f0_mel + 0.5).long() if is_torch else np.rint(f0_mel).astype(np.int)
assert f0_coarse.max() <= 255 and f0_coarse.min() >= 1, (f0_coarse.max(), f0_coarse.min())
return f0_coarse
def get_hubert_model(rank=None):
hubert_soft = hubert_model.hubert_soft("hubert/hubert-soft-0d54a1f4.pt")
if rank is not None:
hubert_soft = hubert_soft.cuda(rank)
return hubert_soft
def get_hubert_content(hmodel, y=None, path=None):
if path is not None:
source, sr = torchaudio.load(path)
source = torchaudio.functional.resample(source, sr, 16000)
if len(source.shape) == 2 and source.shape[1] >= 2:
source = torch.mean(source, dim=0).unsqueeze(0)
else:
source = y
source = source.unsqueeze(0)
with torch.inference_mode():
units = hmodel.units(source)
return units.transpose(1,2)
def get_content(cmodel, y):
with torch.no_grad():
c = cmodel.extract_features(y.squeeze(1))[0]
c = c.transpose(1, 2)
return c
def transform(mel, height): # 68-92
#r = np.random.random()
#rate = r * 0.3 + 0.85 # 0.85-1.15
#height = int(mel.size(-2) * rate)
tgt = torchvision.transforms.functional.resize(mel, (height, mel.size(-1)))
if height >= mel.size(-2):
return tgt[:, :mel.size(-2), :]
else:
silence = tgt[:,-1:,:].repeat(1,mel.size(-2)-height,1)
silence += torch.randn_like(silence) / 10
return torch.cat((tgt, silence), 1)
def stretch(mel, width): # 0.5-2
return torchvision.transforms.functional.resize(mel, (mel.size(-2), width))
def load_checkpoint(checkpoint_path, model, optimizer=None):
assert os.path.isfile(checkpoint_path)
checkpoint_dict = torch.load(checkpoint_path, map_location='cpu')
iteration = checkpoint_dict['iteration']
learning_rate = checkpoint_dict['learning_rate']
if iteration is None:
iteration = 1
if learning_rate is None:
learning_rate = 0.0002
if optimizer is not None and checkpoint_dict['optimizer'] is not None:
optimizer.load_state_dict(checkpoint_dict['optimizer'])
saved_state_dict = checkpoint_dict['model']
if hasattr(model, 'module'):
state_dict = model.module.state_dict()
else:
state_dict = model.state_dict()
new_state_dict= {}
for k, v in state_dict.items():
try:
new_state_dict[k] = saved_state_dict[k]
except:
logger.info("%s is not in the checkpoint" % k)
new_state_dict[k] = v
if hasattr(model, 'module'):
model.module.load_state_dict(new_state_dict)
else:
model.load_state_dict(new_state_dict)
logger.info("Loaded checkpoint '{}' (iteration {})" .format(
checkpoint_path, iteration))
return model, optimizer, learning_rate, iteration
def save_checkpoint(model, optimizer, learning_rate, iteration, checkpoint_path):
logger.info("Saving model and optimizer state at iteration {} to {}".format(
iteration, checkpoint_path))
if hasattr(model, 'module'):
state_dict = model.module.state_dict()
else:
state_dict = model.state_dict()
torch.save({'model': state_dict,
'iteration': iteration,
'optimizer': optimizer.state_dict(),
'learning_rate': learning_rate}, checkpoint_path)
clean_ckpt = False
if clean_ckpt:
clean_checkpoints(path_to_models='logs/32k/', n_ckpts_to_keep=3, sort_by_time=True)
def clean_checkpoints(path_to_models='logs/48k/', n_ckpts_to_keep=2, sort_by_time=True):
"""Freeing up space by deleting saved ckpts
Arguments:
path_to_models -- Path to the model directory
n_ckpts_to_keep -- Number of ckpts to keep, excluding G_0.pth and D_0.pth
sort_by_time -- True -> chronologically delete ckpts
False -> lexicographically delete ckpts
"""
ckpts_files = [f for f in os.listdir(path_to_models) if os.path.isfile(os.path.join(path_to_models, f))]
name_key = (lambda _f: int(re.compile('._(\d+)\.pth').match(_f).group(1)))
time_key = (lambda _f: os.path.getmtime(os.path.join(path_to_models, _f)))
sort_key = time_key if sort_by_time else name_key
x_sorted = lambda _x: sorted([f for f in ckpts_files if f.startswith(_x) and not f.endswith('_0.pth')], key=sort_key)
to_del = [os.path.join(path_to_models, fn) for fn in
(x_sorted('G')[:-n_ckpts_to_keep] + x_sorted('D')[:-n_ckpts_to_keep])]
del_info = lambda fn: logger.info(f".. Free up space by deleting ckpt {fn}")
del_routine = lambda x: [os.remove(x), del_info(x)]
rs = [del_routine(fn) for fn in to_del]
def summarize(writer, global_step, scalars={}, histograms={}, images={}, audios={}, audio_sampling_rate=22050):
for k, v in scalars.items():
writer.add_scalar(k, v, global_step)
for k, v in histograms.items():
writer.add_histogram(k, v, global_step)
for k, v in images.items():
writer.add_image(k, v, global_step, dataformats='HWC')
for k, v in audios.items():
writer.add_audio(k, v, global_step, audio_sampling_rate)
def latest_checkpoint_path(dir_path, regex="G_*.pth"):
f_list = glob.glob(os.path.join(dir_path, regex))
f_list.sort(key=lambda f: int("".join(filter(str.isdigit, f))))
x = f_list[-1]
print(x)
return x
def plot_spectrogram_to_numpy(spectrogram):
global MATPLOTLIB_FLAG
if not MATPLOTLIB_FLAG:
import matplotlib
matplotlib.use("Agg")
MATPLOTLIB_FLAG = True
mpl_logger = logging.getLogger('matplotlib')
mpl_logger.setLevel(logging.WARNING)
import matplotlib.pylab as plt
import numpy as np
fig, ax = plt.subplots(figsize=(10,2))
im = ax.imshow(spectrogram, aspect="auto", origin="lower",
interpolation='none')
plt.colorbar(im, ax=ax)
plt.xlabel("Frames")
plt.ylabel("Channels")
plt.tight_layout()
fig.canvas.draw()
data = np.fromstring(fig.canvas.tostring_rgb(), dtype=np.uint8, sep='')
data = data.reshape(fig.canvas.get_width_height()[::-1] + (3,))
plt.close()
return data
def plot_alignment_to_numpy(alignment, info=None):
global MATPLOTLIB_FLAG
if not MATPLOTLIB_FLAG:
import matplotlib
matplotlib.use("Agg")
MATPLOTLIB_FLAG = True
mpl_logger = logging.getLogger('matplotlib')
mpl_logger.setLevel(logging.WARNING)
import matplotlib.pylab as plt
import numpy as np
fig, ax = plt.subplots(figsize=(6, 4))
im = ax.imshow(alignment.transpose(), aspect='auto', origin='lower',
interpolation='none')
fig.colorbar(im, ax=ax)
xlabel = 'Decoder timestep'
if info is not None:
xlabel += '\n\n' + info
plt.xlabel(xlabel)
plt.ylabel('Encoder timestep')
plt.tight_layout()
fig.canvas.draw()
data = np.fromstring(fig.canvas.tostring_rgb(), dtype=np.uint8, sep='')
data = data.reshape(fig.canvas.get_width_height()[::-1] + (3,))
plt.close()
return data
def load_wav_to_torch(full_path):
sampling_rate, data = read(full_path)
return torch.FloatTensor(data.astype(np.float32)), sampling_rate
def load_filepaths_and_text(filename, split="|"):
with open(filename, encoding='utf-8') as f:
filepaths_and_text = [line.strip().split(split) for line in f]
return filepaths_and_text
def get_hparams(init=True):
parser = argparse.ArgumentParser()
parser.add_argument('-c', '--config', type=str, default="./configs/base.json",
help='JSON file for configuration')
parser.add_argument('-m', '--model', type=str, required=True,
help='Model name')
args = parser.parse_args()
model_dir = os.path.join("./logs", args.model)
if not os.path.exists(model_dir):
os.makedirs(model_dir)
config_path = args.config
config_save_path = os.path.join(model_dir, "config.json")
if init:
with open(config_path, "r") as f:
data = f.read()
with open(config_save_path, "w") as f:
f.write(data)
else:
with open(config_save_path, "r") as f:
data = f.read()
config = json.loads(data)
hparams = HParams(**config)
hparams.model_dir = model_dir
return hparams
def get_hparams_from_dir(model_dir):
config_save_path = os.path.join(model_dir, "config.json")
with open(config_save_path, "r") as f:
data = f.read()
config = json.loads(data)
hparams =HParams(**config)
hparams.model_dir = model_dir
return hparams
def get_hparams_from_file(config_path):
with open(config_path, "r") as f:
data = f.read()
config = json.loads(data)
hparams =HParams(**config)
return hparams
def check_git_hash(model_dir):
source_dir = os.path.dirname(os.path.realpath(__file__))
if not os.path.exists(os.path.join(source_dir, ".git")):
logger.warn("{} is not a git repository, therefore hash value comparison will be ignored.".format(
source_dir
))
return
cur_hash = subprocess.getoutput("git rev-parse HEAD")
path = os.path.join(model_dir, "githash")
if os.path.exists(path):
saved_hash = open(path).read()
if saved_hash != cur_hash:
logger.warn("git hash values are different. {}(saved) != {}(current)".format(
saved_hash[:8], cur_hash[:8]))
else:
open(path, "w").write(cur_hash)
def get_logger(model_dir, filename="train.log"):
global logger
logger = logging.getLogger(os.path.basename(model_dir))
logger.setLevel(logging.DEBUG)
formatter = logging.Formatter("%(asctime)s\t%(name)s\t%(levelname)s\t%(message)s")
if not os.path.exists(model_dir):
os.makedirs(model_dir)
h = logging.FileHandler(os.path.join(model_dir, filename))
h.setLevel(logging.DEBUG)
h.setFormatter(formatter)
logger.addHandler(h)
return logger
class HParams():
def __init__(self, **kwargs):
for k, v in kwargs.items():
if type(v) == dict:
v = HParams(**v)
self[k] = v
def keys(self):
return self.__dict__.keys()
def items(self):
return self.__dict__.items()
def values(self):
return self.__dict__.values()
def __len__(self):
return len(self.__dict__)
def __getitem__(self, key):
return getattr(self, key)
def __setitem__(self, key, value):
return setattr(self, key, value)
def __contains__(self, key):
return key in self.__dict__
def __repr__(self):
return self.__dict__.__repr__()

@ -0,0 +1,15 @@
import os
import shutil
class AttrDict(dict):
def __init__(self, *args, **kwargs):
super(AttrDict, self).__init__(*args, **kwargs)
self.__dict__ = self
def build_env(config, config_name, path):
t_path = os.path.join(path, config_name)
if config != t_path:
os.makedirs(path, exist_ok=True)
shutil.copyfile(config, os.path.join(path, config_name))

@ -0,0 +1,503 @@
import os
import json
from .env import AttrDict
import numpy as np
import torch
import torch.nn.functional as F
import torch.nn as nn
from torch.nn import Conv1d, ConvTranspose1d, AvgPool1d, Conv2d
from torch.nn.utils import weight_norm, remove_weight_norm, spectral_norm
from .utils import init_weights, get_padding
LRELU_SLOPE = 0.1
def load_model(model_path, device='cuda'):
config_file = os.path.join(os.path.split(model_path)[0], 'config.json')
with open(config_file) as f:
data = f.read()
global h
json_config = json.loads(data)
h = AttrDict(json_config)
generator = Generator(h).to(device)
cp_dict = torch.load(model_path)
generator.load_state_dict(cp_dict['generator'])
generator.eval()
generator.remove_weight_norm()
del cp_dict
return generator, h
class ResBlock1(torch.nn.Module):
def __init__(self, h, channels, kernel_size=3, dilation=(1, 3, 5)):
super(ResBlock1, self).__init__()
self.h = h
self.convs1 = nn.ModuleList([
weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[0],
padding=get_padding(kernel_size, dilation[0]))),
weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[1],
padding=get_padding(kernel_size, dilation[1]))),
weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[2],
padding=get_padding(kernel_size, dilation[2])))
])
self.convs1.apply(init_weights)
self.convs2 = nn.ModuleList([
weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=1,
padding=get_padding(kernel_size, 1))),
weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=1,
padding=get_padding(kernel_size, 1))),
weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=1,
padding=get_padding(kernel_size, 1)))
])
self.convs2.apply(init_weights)
def forward(self, x):
for c1, c2 in zip(self.convs1, self.convs2):
xt = F.leaky_relu(x, LRELU_SLOPE)
xt = c1(xt)
xt = F.leaky_relu(xt, LRELU_SLOPE)
xt = c2(xt)
x = xt + x
return x
def remove_weight_norm(self):
for l in self.convs1:
remove_weight_norm(l)
for l in self.convs2:
remove_weight_norm(l)
class ResBlock2(torch.nn.Module):
def __init__(self, h, channels, kernel_size=3, dilation=(1, 3)):
super(ResBlock2, self).__init__()
self.h = h
self.convs = nn.ModuleList([
weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[0],
padding=get_padding(kernel_size, dilation[0]))),
weight_norm(Conv1d(channels, channels, kernel_size, 1, dilation=dilation[1],
padding=get_padding(kernel_size, dilation[1])))
])
self.convs.apply(init_weights)
def forward(self, x):
for c in self.convs:
xt = F.leaky_relu(x, LRELU_SLOPE)
xt = c(xt)
x = xt + x
return x
def remove_weight_norm(self):
for l in self.convs:
remove_weight_norm(l)
def padDiff(x):
return F.pad(F.pad(x, (0,0,-1,1), 'constant', 0) - x, (0,0,0,-1), 'constant', 0)
class SineGen(torch.nn.Module):
""" Definition of sine generator
SineGen(samp_rate, harmonic_num = 0,
sine_amp = 0.1, noise_std = 0.003,
voiced_threshold = 0,
flag_for_pulse=False)
samp_rate: sampling rate in Hz
harmonic_num: number of harmonic overtones (default 0)
sine_amp: amplitude of sine-wavefrom (default 0.1)
noise_std: std of Gaussian noise (default 0.003)
voiced_thoreshold: F0 threshold for U/V classification (default 0)
flag_for_pulse: this SinGen is used inside PulseGen (default False)
Note: when flag_for_pulse is True, the first time step of a voiced
segment is always sin(np.pi) or cos(0)
"""
def __init__(self, samp_rate, harmonic_num=0,
sine_amp=0.1, noise_std=0.003,
voiced_threshold=0,
flag_for_pulse=False):
super(SineGen, self).__init__()
self.sine_amp = sine_amp
self.noise_std = noise_std
self.harmonic_num = harmonic_num
self.dim = self.harmonic_num + 1
self.sampling_rate = samp_rate
self.voiced_threshold = voiced_threshold
self.flag_for_pulse = flag_for_pulse
def _f02uv(self, f0):
# generate uv signal
uv = (f0 > self.voiced_threshold).type(torch.float32)
return uv
def _f02sine(self, f0_values):
""" f0_values: (batchsize, length, dim)
where dim indicates fundamental tone and overtones
"""
# convert to F0 in rad. The interger part n can be ignored
# because 2 * np.pi * n doesn't affect phase
rad_values = (f0_values / self.sampling_rate) % 1
# initial phase noise (no noise for fundamental component)
rand_ini = torch.rand(f0_values.shape[0], f0_values.shape[2], \
device=f0_values.device)
rand_ini[:, 0] = 0
rad_values[:, 0, :] = rad_values[:, 0, :] + rand_ini
# instantanouse phase sine[t] = sin(2*pi \sum_i=1 ^{t} rad)
if not self.flag_for_pulse:
# for normal case
# To prevent torch.cumsum numerical overflow,
# it is necessary to add -1 whenever \sum_k=1^n rad_value_k > 1.
# Buffer tmp_over_one_idx indicates the time step to add -1.
# This will not change F0 of sine because (x-1) * 2*pi = x * 2*pi
tmp_over_one = torch.cumsum(rad_values, 1) % 1
tmp_over_one_idx = (padDiff(tmp_over_one)) < 0
cumsum_shift = torch.zeros_like(rad_values)
cumsum_shift[:, 1:, :] = tmp_over_one_idx * -1.0
sines = torch.sin(torch.cumsum(rad_values + cumsum_shift, dim=1)
* 2 * np.pi)
else:
# If necessary, make sure that the first time step of every
# voiced segments is sin(pi) or cos(0)
# This is used for pulse-train generation
# identify the last time step in unvoiced segments
uv = self._f02uv(f0_values)
uv_1 = torch.roll(uv, shifts=-1, dims=1)
uv_1[:, -1, :] = 1
u_loc = (uv < 1) * (uv_1 > 0)
# get the instantanouse phase
tmp_cumsum = torch.cumsum(rad_values, dim=1)
# different batch needs to be processed differently
for idx in range(f0_values.shape[0]):
temp_sum = tmp_cumsum[idx, u_loc[idx, :, 0], :]
temp_sum[1:, :] = temp_sum[1:, :] - temp_sum[0:-1, :]
# stores the accumulation of i.phase within
# each voiced segments
tmp_cumsum[idx, :, :] = 0
tmp_cumsum[idx, u_loc[idx, :, 0], :] = temp_sum
# rad_values - tmp_cumsum: remove the accumulation of i.phase
# within the previous voiced segment.
i_phase = torch.cumsum(rad_values - tmp_cumsum, dim=1)
# get the sines
sines = torch.cos(i_phase * 2 * np.pi)
return sines
def forward(self, f0):
""" sine_tensor, uv = forward(f0)
input F0: tensor(batchsize=1, length, dim=1)
f0 for unvoiced steps should be 0
output sine_tensor: tensor(batchsize=1, length, dim)
output uv: tensor(batchsize=1, length, 1)
"""
with torch.no_grad():
f0_buf = torch.zeros(f0.shape[0], f0.shape[1], self.dim,
device=f0.device)
# fundamental component
fn = torch.multiply(f0, torch.FloatTensor([[range(1, self.harmonic_num + 2)]]).to(f0.device))
# generate sine waveforms
sine_waves = self._f02sine(fn) * self.sine_amp
# generate uv signal
# uv = torch.ones(f0.shape)
# uv = uv * (f0 > self.voiced_threshold)
uv = self._f02uv(f0)
# noise: for unvoiced should be similar to sine_amp
# std = self.sine_amp/3 -> max value ~ self.sine_amp
# . for voiced regions is self.noise_std
noise_amp = uv * self.noise_std + (1 - uv) * self.sine_amp / 3
noise = noise_amp * torch.randn_like(sine_waves)
# first: set the unvoiced part to 0 by uv
# then: additive noise
sine_waves = sine_waves * uv + noise
return sine_waves, uv, noise
class SourceModuleHnNSF(torch.nn.Module):
""" SourceModule for hn-nsf
SourceModule(sampling_rate, harmonic_num=0, sine_amp=0.1,
add_noise_std=0.003, voiced_threshod=0)
sampling_rate: sampling_rate in Hz
harmonic_num: number of harmonic above F0 (default: 0)
sine_amp: amplitude of sine source signal (default: 0.1)
add_noise_std: std of additive Gaussian noise (default: 0.003)
note that amplitude of noise in unvoiced is decided
by sine_amp
voiced_threshold: threhold to set U/V given F0 (default: 0)
Sine_source, noise_source = SourceModuleHnNSF(F0_sampled)
F0_sampled (batchsize, length, 1)
Sine_source (batchsize, length, 1)
noise_source (batchsize, length 1)
uv (batchsize, length, 1)
"""
def __init__(self, sampling_rate, harmonic_num=0, sine_amp=0.1,
add_noise_std=0.003, voiced_threshod=0):
super(SourceModuleHnNSF, self).__init__()
self.sine_amp = sine_amp
self.noise_std = add_noise_std
# to produce sine waveforms
self.l_sin_gen = SineGen(sampling_rate, harmonic_num,
sine_amp, add_noise_std, voiced_threshod)
# to merge source harmonics into a single excitation
self.l_linear = torch.nn.Linear(harmonic_num + 1, 1)
self.l_tanh = torch.nn.Tanh()
def forward(self, x):
"""
Sine_source, noise_source = SourceModuleHnNSF(F0_sampled)
F0_sampled (batchsize, length, 1)
Sine_source (batchsize, length, 1)
noise_source (batchsize, length 1)
"""
# source for harmonic branch
sine_wavs, uv, _ = self.l_sin_gen(x)
sine_merge = self.l_tanh(self.l_linear(sine_wavs))
# source for noise branch, in the same shape as uv
noise = torch.randn_like(uv) * self.sine_amp / 3
return sine_merge, noise, uv
class Generator(torch.nn.Module):
def __init__(self, h):
super(Generator, self).__init__()
self.h = h
self.num_kernels = len(h["resblock_kernel_sizes"])
self.num_upsamples = len(h["upsample_rates"])
self.f0_upsamp = torch.nn.Upsample(scale_factor=np.prod(h["upsample_rates"]))
self.m_source = SourceModuleHnNSF(
sampling_rate=h["sampling_rate"],
harmonic_num=8)
self.noise_convs = nn.ModuleList()
self.conv_pre = weight_norm(Conv1d(h["inter_channels"], h["upsample_initial_channel"], 7, 1, padding=3))
resblock = ResBlock1 if h["resblock"] == '1' else ResBlock2
self.ups = nn.ModuleList()
for i, (u, k) in enumerate(zip(h["upsample_rates"], h["upsample_kernel_sizes"])):
c_cur = h["upsample_initial_channel"] // (2 ** (i + 1))
self.ups.append(weight_norm(
ConvTranspose1d(h["upsample_initial_channel"] // (2 ** i), h["upsample_initial_channel"] // (2 ** (i + 1)),
k, u, padding=(k - u) // 2)))
if i + 1 < len(h["upsample_rates"]): #
stride_f0 = np.prod(h["upsample_rates"][i + 1:])
self.noise_convs.append(Conv1d(
1, c_cur, kernel_size=stride_f0 * 2, stride=stride_f0, padding=stride_f0 // 2))
else:
self.noise_convs.append(Conv1d(1, c_cur, kernel_size=1))
self.resblocks = nn.ModuleList()
for i in range(len(self.ups)):
ch = h["upsample_initial_channel"] // (2 ** (i + 1))
for j, (k, d) in enumerate(zip(h["resblock_kernel_sizes"], h["resblock_dilation_sizes"])):
self.resblocks.append(resblock(h, ch, k, d))
self.conv_post = weight_norm(Conv1d(ch, 1, 7, 1, padding=3))
self.ups.apply(init_weights)
self.conv_post.apply(init_weights)
self.cond = nn.Conv1d(h['gin_channels'], h['upsample_initial_channel'], 1)
def forward(self, x, f0, g=None):
# print(1,x.shape,f0.shape,f0[:, None].shape)
f0 = self.f0_upsamp(f0[:, None]).transpose(1, 2) # bs,n,t
# print(2,f0.shape)
har_source, noi_source, uv = self.m_source(f0)
har_source = har_source.transpose(1, 2)
x = self.conv_pre(x)
x = x + self.cond(g)
# print(124,x.shape,har_source.shape)
for i in range(self.num_upsamples):
x = F.leaky_relu(x, LRELU_SLOPE)
# print(3,x.shape)
x = self.ups[i](x)
x_source = self.noise_convs[i](har_source)
# print(4,x_source.shape,har_source.shape,x.shape)
x = x + x_source
xs = None
for j in range(self.num_kernels):
if xs is None:
xs = self.resblocks[i * self.num_kernels + j](x)
else:
xs += self.resblocks[i * self.num_kernels + j](x)
x = xs / self.num_kernels
x = F.leaky_relu(x)
x = self.conv_post(x)
x = torch.tanh(x)
return x
def remove_weight_norm(self):
print('Removing weight norm...')
for l in self.ups:
remove_weight_norm(l)
for l in self.resblocks:
l.remove_weight_norm()
remove_weight_norm(self.conv_pre)
remove_weight_norm(self.conv_post)
class DiscriminatorP(torch.nn.Module):
def __init__(self, period, kernel_size=5, stride=3, use_spectral_norm=False):
super(DiscriminatorP, self).__init__()
self.period = period
norm_f = weight_norm if use_spectral_norm == False else spectral_norm
self.convs = nn.ModuleList([
norm_f(Conv2d(1, 32, (kernel_size, 1), (stride, 1), padding=(get_padding(5, 1), 0))),
norm_f(Conv2d(32, 128, (kernel_size, 1), (stride, 1), padding=(get_padding(5, 1), 0))),
norm_f(Conv2d(128, 512, (kernel_size, 1), (stride, 1), padding=(get_padding(5, 1), 0))),
norm_f(Conv2d(512, 1024, (kernel_size, 1), (stride, 1), padding=(get_padding(5, 1), 0))),
norm_f(Conv2d(1024, 1024, (kernel_size, 1), 1, padding=(2, 0))),
])
self.conv_post = norm_f(Conv2d(1024, 1, (3, 1), 1, padding=(1, 0)))
def forward(self, x):
fmap = []
# 1d to 2d
b, c, t = x.shape
if t % self.period != 0: # pad first
n_pad = self.period - (t % self.period)
x = F.pad(x, (0, n_pad), "reflect")
t = t + n_pad
x = x.view(b, c, t // self.period, self.period)
for l in self.convs:
x = l(x)
x = F.leaky_relu(x, LRELU_SLOPE)
fmap.append(x)
x = self.conv_post(x)
fmap.append(x)
x = torch.flatten(x, 1, -1)
return x, fmap
class MultiPeriodDiscriminator(torch.nn.Module):
def __init__(self, periods=None):
super(MultiPeriodDiscriminator, self).__init__()
self.periods = periods if periods is not None else [2, 3, 5, 7, 11]
self.discriminators = nn.ModuleList()
for period in self.periods:
self.discriminators.append(DiscriminatorP(period))
def forward(self, y, y_hat):
y_d_rs = []
y_d_gs = []
fmap_rs = []
fmap_gs = []
for i, d in enumerate(self.discriminators):
y_d_r, fmap_r = d(y)
y_d_g, fmap_g = d(y_hat)
y_d_rs.append(y_d_r)
fmap_rs.append(fmap_r)
y_d_gs.append(y_d_g)
fmap_gs.append(fmap_g)
return y_d_rs, y_d_gs, fmap_rs, fmap_gs
class DiscriminatorS(torch.nn.Module):
def __init__(self, use_spectral_norm=False):
super(DiscriminatorS, self).__init__()
norm_f = weight_norm if use_spectral_norm == False else spectral_norm
self.convs = nn.ModuleList([
norm_f(Conv1d(1, 128, 15, 1, padding=7)),
norm_f(Conv1d(128, 128, 41, 2, groups=4, padding=20)),
norm_f(Conv1d(128, 256, 41, 2, groups=16, padding=20)),
norm_f(Conv1d(256, 512, 41, 4, groups=16, padding=20)),
norm_f(Conv1d(512, 1024, 41, 4, groups=16, padding=20)),
norm_f(Conv1d(1024, 1024, 41, 1, groups=16, padding=20)),
norm_f(Conv1d(1024, 1024, 5, 1, padding=2)),
])
self.conv_post = norm_f(Conv1d(1024, 1, 3, 1, padding=1))
def forward(self, x):
fmap = []
for l in self.convs:
x = l(x)
x = F.leaky_relu(x, LRELU_SLOPE)
fmap.append(x)
x = self.conv_post(x)
fmap.append(x)
x = torch.flatten(x, 1, -1)
return x, fmap
class MultiScaleDiscriminator(torch.nn.Module):
def __init__(self):
super(MultiScaleDiscriminator, self).__init__()
self.discriminators = nn.ModuleList([
DiscriminatorS(use_spectral_norm=True),
DiscriminatorS(),
DiscriminatorS(),
])
self.meanpools = nn.ModuleList([
AvgPool1d(4, 2, padding=2),
AvgPool1d(4, 2, padding=2)
])
def forward(self, y, y_hat):
y_d_rs = []
y_d_gs = []
fmap_rs = []
fmap_gs = []
for i, d in enumerate(self.discriminators):
if i != 0:
y = self.meanpools[i - 1](y)
y_hat = self.meanpools[i - 1](y_hat)
y_d_r, fmap_r = d(y)
y_d_g, fmap_g = d(y_hat)
y_d_rs.append(y_d_r)
fmap_rs.append(fmap_r)
y_d_gs.append(y_d_g)
fmap_gs.append(fmap_g)
return y_d_rs, y_d_gs, fmap_rs, fmap_gs
def feature_loss(fmap_r, fmap_g):
loss = 0
for dr, dg in zip(fmap_r, fmap_g):
for rl, gl in zip(dr, dg):
loss += torch.mean(torch.abs(rl - gl))
return loss * 2
def discriminator_loss(disc_real_outputs, disc_generated_outputs):
loss = 0
r_losses = []
g_losses = []
for dr, dg in zip(disc_real_outputs, disc_generated_outputs):
r_loss = torch.mean((1 - dr) ** 2)
g_loss = torch.mean(dg ** 2)
loss += (r_loss + g_loss)
r_losses.append(r_loss.item())
g_losses.append(g_loss.item())
return loss, r_losses, g_losses
def generator_loss(disc_outputs):
loss = 0
gen_losses = []
for dg in disc_outputs:
l = torch.mean((1 - dg) ** 2)
gen_losses.append(l)
loss += l
return loss, gen_losses

@ -0,0 +1,111 @@
import math
import os
os.environ["LRU_CACHE_CAPACITY"] = "3"
import random
import torch
import torch.utils.data
import numpy as np
import librosa
from librosa.util import normalize
from librosa.filters import mel as librosa_mel_fn
from scipy.io.wavfile import read
import soundfile as sf
def load_wav_to_torch(full_path, target_sr=None, return_empty_on_exception=False):
sampling_rate = None
try:
data, sampling_rate = sf.read(full_path, always_2d=True)# than soundfile.
except Exception as ex:
print(f"'{full_path}' failed to load.\nException:")
print(ex)
if return_empty_on_exception:
return [], sampling_rate or target_sr or 32000
else:
raise Exception(ex)
if len(data.shape) > 1:
data = data[:, 0]
assert len(data) > 2# check duration of audio file is > 2 samples (because otherwise the slice operation was on the wrong dimension)
if np.issubdtype(data.dtype, np.integer): # if audio data is type int
max_mag = -np.iinfo(data.dtype).min # maximum magnitude = min possible value of intXX
else: # if audio data is type fp32
max_mag = max(np.amax(data), -np.amin(data))
max_mag = (2**31)+1 if max_mag > (2**15) else ((2**15)+1 if max_mag > 1.01 else 1.0) # data should be either 16-bit INT, 32-bit INT or [-1 to 1] float32
data = torch.FloatTensor(data.astype(np.float32))/max_mag
if (torch.isinf(data) | torch.isnan(data)).any() and return_empty_on_exception:# resample will crash with inf/NaN inputs. return_empty_on_exception will return empty arr instead of except
return [], sampling_rate or target_sr or 32000
if target_sr is not None and sampling_rate != target_sr:
data = torch.from_numpy(librosa.core.resample(data.numpy(), orig_sr=sampling_rate, target_sr=target_sr))
sampling_rate = target_sr
return data, sampling_rate
def dynamic_range_compression(x, C=1, clip_val=1e-5):
return np.log(np.clip(x, a_min=clip_val, a_max=None) * C)
def dynamic_range_decompression(x, C=1):
return np.exp(x) / C
def dynamic_range_compression_torch(x, C=1, clip_val=1e-5):
return torch.log(torch.clamp(x, min=clip_val) * C)
def dynamic_range_decompression_torch(x, C=1):
return torch.exp(x) / C
class STFT():
def __init__(self, sr=22050, n_mels=80, n_fft=1024, win_size=1024, hop_length=256, fmin=20, fmax=11025, clip_val=1e-5):
self.target_sr = sr
self.n_mels = n_mels
self.n_fft = n_fft
self.win_size = win_size
self.hop_length = hop_length
self.fmin = fmin
self.fmax = fmax
self.clip_val = clip_val
self.mel_basis = {}
self.hann_window = {}
def get_mel(self, y, center=False):
sampling_rate = self.target_sr
n_mels = self.n_mels
n_fft = self.n_fft
win_size = self.win_size
hop_length = self.hop_length
fmin = self.fmin
fmax = self.fmax
clip_val = self.clip_val
if torch.min(y) < -1.:
print('min value is ', torch.min(y))
if torch.max(y) > 1.:
print('max value is ', torch.max(y))
if fmax not in self.mel_basis:
mel = librosa_mel_fn(sr=sampling_rate, n_fft=n_fft, n_mels=n_mels, fmin=fmin, fmax=fmax)
self.mel_basis[str(fmax)+'_'+str(y.device)] = torch.from_numpy(mel).float().to(y.device)
self.hann_window[str(y.device)] = torch.hann_window(self.win_size).to(y.device)
y = torch.nn.functional.pad(y.unsqueeze(1), (int((n_fft-hop_length)/2), int((n_fft-hop_length)/2)), mode='reflect')
y = y.squeeze(1)
spec = torch.stft(y, n_fft, hop_length=hop_length, win_length=win_size, window=self.hann_window[str(y.device)],
center=center, pad_mode='reflect', normalized=False, onesided=True)
# print(111,spec)
spec = torch.sqrt(spec.pow(2).sum(-1)+(1e-9))
# print(222,spec)
spec = torch.matmul(self.mel_basis[str(fmax)+'_'+str(y.device)], spec)
# print(333,spec)
spec = dynamic_range_compression_torch(spec, clip_val=clip_val)
# print(444,spec)
return spec
def __call__(self, audiopath):
audio, sr = load_wav_to_torch(audiopath, target_sr=self.target_sr)
spect = self.get_mel(audio.unsqueeze(0)).squeeze(0)
return spect
stft = STFT()

@ -0,0 +1,68 @@
import glob
import os
import matplotlib
import torch
from torch.nn.utils import weight_norm
matplotlib.use("Agg")
import matplotlib.pylab as plt
def plot_spectrogram(spectrogram):
fig, ax = plt.subplots(figsize=(10, 2))
im = ax.imshow(spectrogram, aspect="auto", origin="lower",
interpolation='none')
plt.colorbar(im, ax=ax)
fig.canvas.draw()
plt.close()
return fig
def init_weights(m, mean=0.0, std=0.01):
classname = m.__class__.__name__
if classname.find("Conv") != -1:
m.weight.data.normal_(mean, std)
def apply_weight_norm(m):
classname = m.__class__.__name__
if classname.find("Conv") != -1:
weight_norm(m)
def get_padding(kernel_size, dilation=1):
return int((kernel_size*dilation - dilation)/2)
def load_checkpoint(filepath, device):
assert os.path.isfile(filepath)
print("Loading '{}'".format(filepath))
checkpoint_dict = torch.load(filepath, map_location=device)
print("Complete.")
return checkpoint_dict
def save_checkpoint(filepath, obj):
print("Saving checkpoint to {}".format(filepath))
torch.save(obj, filepath)
print("Complete.")
def del_old_checkpoints(cp_dir, prefix, n_models=2):
pattern = os.path.join(cp_dir, prefix + '????????')
cp_list = glob.glob(pattern) # get checkpoint paths
cp_list = sorted(cp_list)# sort by iter
if len(cp_list) > n_models: # if more than n_models models are found
for cp in cp_list[:-n_models]:# delete the oldest models other than lastest n_models
open(cp, 'w').close()# empty file contents
os.unlink(cp)# delete file (move to trash when using Colab)
def scan_checkpoint(cp_dir, prefix):
pattern = os.path.join(cp_dir, prefix + '????????')
cp_list = glob.glob(pattern)
if len(cp_list) == 0:
return None
return sorted(cp_list)[-1]
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