Add back original so-vits-svc

This commit is contained in:
Brandon Thomas
2023-04-09 22:18:14 -07:00
parent 2148c77395
commit f8957a63e9
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# Vim
*~
# Model files
checkpoint_best_legacy_500.pt
logs/44k/
# Virtual environments
python/
##### BELOW WAS ORIGINAL GITIGNORE #####
#
# Created by https://www.toptal.com/developers/gitignore/api/python
# Edit at https://www.toptal.com/developers/gitignore?templates=python
### Python ###
# Byte-compiled / optimized / DLL files
__pycache__/
*.py[cod]
*$py.class
# C extensions
*.so
# Distribution / packaging
.Python
build/
develop-eggs/
dist/
downloads/
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parts/
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var/
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share/python-wheels/
*.egg-info/
.installed.cfg
*.egg
MANIFEST
# PyInstaller
# Usually these files are written by a python script from a template
# before PyInstaller builds the exe, so as to inject date/other infos into it.
*.manifest
*.spec
# Installer logs
pip-log.txt
pip-delete-this-directory.txt
# Unit test / coverage reports
htmlcov/
.tox/
.nox/
.coverage
.coverage.*
.cache
nosetests.xml
coverage.xml
*.cover
*.py,cover
.hypothesis/
.pytest_cache/
pytestdebug.log
# Translations
*.mo
*.pot
# Django stuff:
*.log
local_settings.py
db.sqlite3
db.sqlite3-journal
# Flask stuff:
instance/
.webassets-cache
# Scrapy stuff:
.scrapy
# Sphinx documentation
docs/_build/
doc/_build/
# PyBuilder
target/
# Jupyter Notebook
.ipynb_checkpoints
# IPython
profile_default/
ipython_config.py
# pyenv
.python-version
# pipenv
# According to pypa/pipenv#598, it is recommended to include Pipfile.lock in version control.
# However, in case of collaboration, if having platform-specific dependencies or dependencies
# having no cross-platform support, pipenv may install dependencies that don't work, or not
# install all needed dependencies.
#Pipfile.lock
# PEP 582; used by e.g. github.com/David-OConnor/pyflow
__pypackages__/
# Celery stuff
celerybeat-schedule
celerybeat.pid
# SageMath parsed files
*.sage.py
# Environments
.env
.venv
env/
venv/
ENV/
env.bak/
venv.bak/
# Spyder project settings
.spyderproject
.spyproject
# Rope project settings
.ropeproject
# mkdocs documentation
/site
# mypy
.mypy_cache/
.dmypy.json
dmypy.json
# Pyre type checker
.pyre/
# pytype static type analyzer
.pytype/
# End of https://www.toptal.com/developers/gitignore/api/python
dataset
dataset_raw
raw
results
inference/chunks_temp.json
logs
hubert/checkpoint_best_legacy_500.pt
configs/config.json
filelists/test.txt
filelists/train.txt
filelists/val.txt
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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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so-vits-svc
===========
Original docs are in [README\_ORIGINAL.md](./README_ORIGINAL.md).
This is a snapshot of `f22819cb8718ead5e50dea4fdbd05195c72fb1a8` (branch `origin/4.0`). It was taken
from justinjohn0306's fork at https://github.com/justinjohn0306/so-vits-svc
Dev Installation
----------------
Use our `requirements.txt` instead of the original `upstream-requirements-but-broken.txt`
Installing this may give you trouble. Brandon had trouble under Python3.10, so downgraded to Python 3.9
using the deadsnakes PPA. Furthermore, the version of pyworld originally specified refused to build;
upgrading to the latest point release fixed it.
`pyworld==0.2.5` -> `pyworld==0.2.12`
tl;dr:
- Python3.9
- pyworld 0.2.12
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# SoftVC VITS Singing Voice Conversion
## SoVITS has stopped updating and officially Archived
Some other related warehouses for follow-up maintenance and other work:
+ [so-vits-svc](https://github.com/svc-develop-team/so-vits-svc): The svc community is ready to take over and maintain sovits. If you have pr and issue, you can submit them here
+ [SoftVitsResearch](https://github.com/NaruseMioShirakana/SoftVitsResearch): Used to make some fancy functions (mainly for Onnx-MoesS)
## Terms of use
1. Please solve the authorization problem of the data set by yourself. Any problems caused by using unauthorized data sets for training, you must bear full responsibility and all consequences, and have nothing to do with sovits!
2. Any sovits-based video published to the video platform must clearly indicate in the introduction the input source singing and audio used for voice changer conversion, for example: use the video/audio released by others, and use the separated human voice as input If the source is converted, a clear link to the original video and music must be given; if the input source is converted using your own vocals or voice synthesized by other singing voice synthesis engines, you must also explain it in the introduction.
3. Any copyright infringement caused by the input source shall bear full responsibility and all consequences. When using other commercial singing voice synthesis software as the input source, please make sure to abide by the terms of use of the software. Note that the terms of use of many singing voice synthesis engines clearly indicate that they cannot be used for input source conversion!
## update
> The 4.0-v2 model has been updated. The entire process is the same as 4.0. Compared with 4.0, there are some improvements in some scenarios, but there are also some cases where there are regressions. In the [4.0-v2 branch](https://github.com/innnky/so -vits-svc/tree/4.0-v2) This is the last update of sovits \
## Model Introduction
The singing voice conversion model extracts the voice features of the source audio through the SoftVC content encoder, and inputs VITS at the same time as F0 to replace the original text input to achieve the effect of singing voice conversion. At the same time, replace the vocoder with [NSF HiFiGAN](https://github.com/openvpi/DiffSinger/tree/refactor/modules/nsf_hifigan) to solve the staccato problem
### 4.0 version update content
+ Feature input replaced with [Content Vec](https://github.com/auspicious3000/contentvec)
+ The sampling rate is uniformly used at 44100hz
+ Due to the change of parameters such as hop size and the simplification of some model structures, the video memory usage required for inference has been greatly reduced**, and the 44khz video memory usage of version 4.0 is even smaller than the 32khz of version 3.0
+ Adjusted part of the code structure
+ Dataset production and training process are consistent with 3.0, but the model is not universal at all, and the dataset also needs to be re-preprocessed
+ Added option 1: vc mode automatically predicts the pitch f0, that is, there is no need to manually input the transposition key when converting the voice, the male and female voices can be automatically converted, but only for voice conversion, the singing voice will be out of tune when converted in this mode
+ Added option 2: reduce timbre leakage through the kmeans clustering scheme, that is, make the timbre more like the target timbre
Online demo: [![Hugging Face Spaces](https://img.shields.io/badge/%F0%9F%A4%97%20Hugging%20Face-Spaces-blue)](https://huggingface.co/spaces/innnky/sovits4)
## Pre-downloaded model files
+ contentvec: [checkpoint_best_legacy_500.pt](https://ibm.box.com/s/z1wgl1stco8ffooyatzdwsqn2psd9lrr)
+ placed in the `hubert` directory
+ Pre-training base model file: [G_0.pth](https://huggingface.co/justinjohn-03/so-vits-svc-4.0/resolve/main/G_0.pth) and [D_0.pth](https://huggingface.co/justinjohn-03/so-vits-svc-4.0/resolve/main/D_0.pth)
+ placed in the `logs/44k` directory
+ The pre-trained base model training data set includes Yunhao, Jishuang, Huiyu·Xing AI Paimeng, Lingdi Ningning, covering the common vocal range of male and female girls, which can be considered as a relatively common base model
```shell
# One-click download
# contentvec
http://obs.cstcloud.cn/share/obs/sankagenkeshi/checkpoint_best_legacy_500.pt
# It can also be manually downloaded and placed in the hubert directory
# G and D pre-training model:
wget -P logs/44k/ https://huggingface.co/justinjohn-03/so-vits-svc-4.0/resolve/main/G_0.pth
wget -P logs/44k/ https://huggingface.co/justinjohn-03/so-vits-svc-4.0/resolve/main/D_0.pth
```
## Colab one-click data set production and training script
[![Open In Colab](https://colab.research.google.com/assets/colab-badge.svg)](https://colab.research.google.com/github/justinjohn0306/so-vits-svc/blob/4.0/Sovits4_0_training_inference.ipynb)
## Dataset preparation
Just put the dataset into the dataset_raw directory with the following file structure
```shell
dataset_raw
├───speaker0
│ ├───xxx1-xxx1.wav
│ ├────...
│ └───Lxx-0xx8.wav
└───speaker1
├───xx2-0xxx2.wav
├────...
└───xxx7-xxx007.wav
```
## Data preprocessing
1. Resample to 44100hz
```shell
python resample.py
```
2. Automatically divide training set, verification set, test set and automatically generate configuration files
```shell
python preprocess_flist_config.py
```
3. Generate hubert and f0
```shell
python preprocess_hubert_f0.py
```
After performing the above steps, the dataset directory is the preprocessed data, you can delete the dataset_raw folder
## train
```shell
python train.py -c configs/config.json -m 44k
```
Note: The old model will be automatically cleared during training, and only the latest 3 models will be kept. If you want to prevent overfitting, you need to manually back up the model record points, or modify the configuration file keep_ckpts 0 to never clear
## reasoning
Use [inference_main.py](inference_main.py)
As of now, the 4.0 usage method (training, reasoning) is exactly the same as 3.0, without any changes (command line support is added for reasoning)
```shell
# example
python inference_main.py -m "logs/44k/G_30400.pth" -c "configs/config.json" -n "君の知らない物语-src.wav" -t 0 -s "nen"
```
required field
+ -m, --model_path: model path.
+ -c, --config_path: configuration file path.
+ -n, --clean_names: list of wav file names, placed under the raw folder.
+ -t, --trans: pitch adjustment, support positive and negative (semitones).
+ -s, --spk_list: synthesize target speaker names.
Optional section: see next section
+ -a, --auto_predict_f0: Voice conversion automatically predicts the pitch, do not enable this when converting singing voices, it will seriously go out of tune.
+ -cm, --cluster_model_path: The path of the cluster model, if there is no training cluster, just fill it in.
+ -cr, --cluster_infer_ratio: Clustering scheme ratio, range 0-1, if no clustering model is trained, fill in 0.
## Optional
If the previous effect is satisfactory, or if you don’t understand what is going on below, then the following content can be ignored without affecting the use of the model. (These options have relatively little impact and may have some effect on some specific data, but most of the cases seem to be less obvious),
### Automatic f0 prediction
The 4.0 model training process will train an f0 predictor. For voice conversion, you can enable automatic pitch prediction. If the effect is not good, you can also use manual, but please do not enable this function when converting singing voices! ! ! It will be seriously out of tune! !
+ Set auto_predict_f0 to true in inference_main
### Cluster Timbre Leakage Control
Introduction: The clustering scheme can reduce the timbre leakage, making the model trained more like the target timbre (but it is not particularly obvious), but the simple clustering scheme will reduce the model's articulation (it will be inarticulate) (this is obvious) , this model adopts the fusion method,
You can linearly control the ratio of the clustering scheme and the non-clustering scheme, that is, you can manually adjust the ratio between "like the target tone" and "clear articulation" to find a suitable compromise point.
Using the existing steps before clustering does not require any changes, only an additional clustering model needs to be trained, although the effect is relatively limited, but the training cost is relatively low
+ Training process:
+ Use a machine with better CPU performance for training. According to my experience, it takes about 4 minutes to train each speaker on Tencent Cloud 6-core CPU to complete the training
+ Execute python cluster/train_cluster.py, the output of the model will be in logs/44k/kmeans_10000.pt
+ Reasoning process:
+ Specify cluster_model_path in inference_main
+ Specify cluster_infer_ratio in inference_main, 0 means no clustering at all, 1 means only clustering, usually set to 0.5
## Onnx export
Use [onnx_export.py](onnx_export.py)
+ Create a new folder: `checkpoints` and open it
+ Create a new folder in the `checkpoints` folder as the project folder, and the folder name is your project name, such as `aziplayer`
+ Rename your model to `model.pth`, rename the configuration file to `config.json`, and place it in the `aziplayer` folder you just created
+ Change `"NyaruTaffy"` in `path = "NyaruTaffy"` in [onnx_export.py](onnx_export.py) to your project name, `path = "aziplayer"`
+ run [onnx_export.py](onnx_export.py)
+ After the execution is completed, a `model.onnx` will be generated in your project folder, which is the exported model
### Onnx model supported UI
+ [MoeSS](https://github.com/NaruseMioShirakana/MoeSS)
+ I removed all the training functions and all complicated transpositions, not a single line, because I think you only know that you are using Onnx if you remove these things
+ Note: Please use the model provided by MoesS for the Hubert Onnx model, which cannot be exported at present (Hubert in fairseq has many operators that onnx does not support and things that involve constants, and an error will be reported when exporting or the exported model input and output shape and The results are problematic)
[Hubert4.0](https://huggingface.co/NaruseMioShirakana/MoeSS-SUBModel)
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{
"nbformat": 4,
"nbformat_minor": 0,
"metadata": {
"colab": {
"provenance": []
},
"kernelspec": {
"name": "python3",
"display_name": "Python 3"
},
"language_info": {
"name": "python"
},
"accelerator": "GPU",
"gpuClass": "standard"
},
"cells": [
{
"cell_type": "markdown",
"source": [
"# **So-Vits-SVC 4.0 Training and Testing**\n",
"___\n",
"**This colab notebook is prepared and maintained by [justinjohn03](https://github.com/justinjohn0306)**"
],
"metadata": {
"id": "XiGNXkoTSUoT"
}
},
{
"cell_type": "code",
"source": [
"#@title Check the GPU\n",
"!nvidia-smi"
],
"metadata": {
"id": "0gQcIZ8RsOkn",
"cellView": "form"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"#@title Installs mamba and updates base\n",
"\n",
"%%shell\n",
"curl -L -O https://github.com/conda-forge/miniforge/releases/latest/download/Mambaforge-Linux-x86_64.sh\n",
"chmod +x Mambaforge-Linux-x86_64.sh\n",
"bash Mambaforge-Linux-x86_64.sh -b -f -p /usr/local\n",
"mamba config --env --set always_yes true\n",
"rm Mambaforge-Linux-x86_64.sh\n",
"mamba update -n base -c defaults conda -y"
],
"metadata": {
"id": "DLF3hGn_Nmkc",
"cellView": "form"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"execution_count": null,
"metadata": {
"id": "LS0OPRkL4Pme",
"cellView": "form"
},
"outputs": [],
"source": [
"#@title Clone github repository\n",
"!git clone https://github.com/justinjohn0306/so-vits-svc -b 4.0\n",
"%cd /content/so-vits-svc"
]
},
{
"cell_type": "code",
"source": [
"#@title Creates mamba environment and activates it from environment.yaml. Takes ~8 minutes.\n",
"\n",
"%%shell\n",
"mamba env create -f environment.yaml\n",
"mamba init bash\n",
"source /root/.bashrc\n",
"source activate sovits\n",
"pip install -r requirements_colab.txt"
],
"metadata": {
"id": "zXBLkXxL4T1O",
"cellView": "form"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"#@title Download necessary model files\n",
"# 源仓库地址:[contentvec](https://github.com/auspicious3000/contentvec)\n",
"# 模型原下载链接:[checkpoint_best_legacy_500.pt](https://ibm.box.com/s/z1wgl1stco8ffooyatzdwsqn2psd9lrr)\n",
"# 由于源网盘无法提供http直链,根据mit协议,对模型进行二次分发,提供下载直链\n",
"!wget -P hubert/ https://huggingface.co/justinjohn-03/so-vits-svc-4.0/resolve/main/checkpoint_best_legacy_500.pt"
],
"metadata": {
"id": "pCqf3W0d6ify",
"cellView": "form"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "markdown",
"source": [
"# Dataset preprocessing"
],
"metadata": {
"id": "k1qadJBFehMo"
}
},
{
"cell_type": "code",
"source": [
"#@title Loading Google Drive\n",
"#@markdown load Google Drive\n",
"from google.colab import drive\n",
"drive.mount('/content/drive')"
],
"metadata": {
"id": "wmUkpUmfn_Hs",
"cellView": "form"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "markdown",
"source": [
"This processing script can preprocess multiple speakers at one time, and generate multi-speaker filelists and corresponding configuration files\n",
"\n",
"Just put your dataset into the dataset_raw directory according to the following file structure\n",
"\n",
"\n",
"```\n",
"dataset_raw\n",
"├───speaker0\n",
"│ ├───xxx1-xxx1.wav\n",
"│ ├───...\n",
"│ └───Lxx-0xx8.wav\n",
"└───speaker1\n",
" ├───xx2-0xxx2.wav\n",
" ├───...\n",
" └───xxx7-xxx007.wav\n",
"\n",
"```\n",
"\n",
"\n"
],
"metadata": {
"id": "kBlju6Q3lSM6"
}
},
{
"cell_type": "code",
"source": [
"#@title Load the packaged dataset from Google Cloud Disk for preprocessing\n",
"#@markdown **The data set of sovits3.0 no longer needs a specific file structure. Put all the wav files of the data set in the same folder, compress them into zip and upload them to Google Cloud Disk. This processing script can be used once When processing multiple datasets, please decompress each dataset in turn**\n",
"\n",
"#@markdown dataset name (**character’s English/Pinyin name**, which is the same as when creating the data folder; without zip.)\n",
"DATASETNAME = \"kiritan\" #@param {type:\"string\"}\n",
"#@markdown Compressed package path (Google disk path, don’t change this if you pass it to dataset, create a new one if there is no dataset folder)\n",
"ZIP_PATH = \"/content/drive/MyDrive/dataset/\" #@param {type:\"string\"}\n",
"ZIP_NAME = ZIP_PATH + DATASETNAME\n",
"\n",
"!unzip -d /content/so-vits-svc/dataset_raw {ZIP_NAME}.zip"
],
"metadata": {
"cellView": "form",
"id": "U05CXlAipvJR"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"#@title Resample audio to 44100hz\n",
"!python resample.py"
],
"metadata": {
"id": "_ThKTzYs5CfL",
"cellView": "form"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"#@title Divide training set and Generate configuration file\n",
"!python preprocess_flist_config.py"
],
"metadata": {
"id": "svITReeL5N8K",
"cellView": "form"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"#@title Generate hubert and f0\n",
"!python preprocess_hubert_f0.py"
],
"metadata": {
"id": "xHUXMi836DMe",
"cellView": "form"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"#@title At this point, the preprocessing of the data set is completed, and the data set and related files are saved to the dataset folder of Google Cloud Disk, which is convenient for the next training\n",
"#Compress the dataset folder\n",
"!zip -r dataset.zip /content/so-vits-svc/dataset\n",
"#@markdown Customize the name of the dataset folder under the dataset folder of Google Cloud Disk to avoid confusion\n",
"dataset_name_drive = \"kiritan_preprocessed\" #@param {type:\"string\"}\n",
"DATASET_PATH_DRIVE = \"/content/drive/MyDrive/dataset/\" + dataset_name_drive\n",
"!mkdir -p {DATASET_PATH_DRIVE}\n",
"\n",
"!cp /content/so-vits-svc/dataset.zip \"{DATASET_PATH_DRIVE}\"\n",
"!cp configs/config.json \"{DATASET_PATH_DRIVE}\"\n",
"!cp filelists/train.txt \"{DATASET_PATH_DRIVE}\"\n",
"!cp filelists/val.txt \"{DATASET_PATH_DRIVE}\""
],
"metadata": {
"id": "Wo4OTmTAUXgj",
"cellView": "form"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"#@title If the data set has been preprocessed, you can skip the preprocessing part and decompress the processed data and configuration files directly from the cloud disk\n",
"#@markdown Load the preprocessed data set from Google Cloud Disk, the folder name is the same as what you entered when you backed up\n",
"back_up_name = \"kiritan_preprocessed\" #@param {type:\"string\"}\n",
"BACK_UP_DATASET_PATH = \"/content/drive/MyDrive/dataset/\" + back_up_name\n",
"!unzip {BACK_UP_DATASET_PATH}/dataset.zip -d /\n",
"!cp {BACK_UP_DATASET_PATH}/config.json /content/so-vits-svc/configs/config.json \n",
"!cp {BACK_UP_DATASET_PATH}/val.txt filelists/val.txt\n",
"!cp {BACK_UP_DATASET_PATH}/train.txt filelists/train.txt\n",
"\n",
"\n",
"# 拷贝云盘上保存的记录点\n",
"# !cp /content/drive/MyDrive/G_800.pth logs/48k/\n",
"# !cp /content/drive/MyDrive/D_800.pth logs/48k/"
],
"metadata": {
"id": "P2G6v_6zblWK",
"cellView": "form"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "markdown",
"source": [
"# Train"
],
"metadata": {
"id": "ENoH-pShel7w"
}
},
{
"cell_type": "code",
"source": [
"#@title Choose whether to save the trained model to Google Cloud Disk and whether to use the pre-model\n",
"\n",
"#@markdown **Save the trained model file to Google Cloud Disk. After checking, you need to check and execute it when resuming training**\n",
"Save_to_drive = True #@param {type:\"boolean\"}\n",
"if Save_to_drive:\n",
" !rm -rf /content/so-vits-svc/logs/44k\n",
" !mkdir -p /content/drive/MyDrive/44k\n",
" !ln -s /content/drive/MyDrive/44k /content/so-vits-svc/logs/44k\n",
"\n",
"#@markdown **Download the pre-model for the first training and continue training after using the record points saved by yourself, no need to download again**\n",
"\n",
"#@markdown **Use the pre-model, check the box below to automatically download and enable**\n",
"pre_pth = True #@param {type:\"boolean\"}\n",
"if pre_pth:\n",
" !wget -P logs/44k/ https://huggingface.co/justinjohn-03/so-vits-svc-4.0/resolve/main/G_0.pth\n",
" !wget -P logs/44k/ https://huggingface.co/justinjohn-03/so-vits-svc-4.0/resolve/main/D_0.pth\n"
],
"metadata": {
"id": "l8J2ubh9KV5J",
"cellView": "form"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"#@title Start Training\n",
"\n",
"#@markdown **Enable tensorboard to visualize data**\n",
"tensorboard_on = True #@param {type:\"boolean\"}\n",
"if tensorboard_on:\n",
" %load_ext tensorboard\n",
" %tensorboard --logdir logs/44k\n",
"\n",
"!python train.py -c configs/config.json -m 44k\n"
],
"metadata": {
"id": "-hEFFTCfZf57",
"cellView": "form"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"#@title Manually back up the trained model files to Google Cloud Disk\n",
"#@markdown You need to check the file name of the model under the /content/so-vits-svc/logs/44k/ folder by yourself, and manually modify the file name at the end of the command below\n",
"!mv /content/so-vits-svc/logs/44k/G_1000.pth /content/drive/MyDrive\n",
"!mv /content/so-vits-svc/logs/44k/D_1000.pth /content/drive/MyDrive"
],
"metadata": {
"id": "KiNCWprSPlKH",
"cellView": "form"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "markdown",
"source": [
"# Reasoning"
],
"metadata": {
"id": "oCnbX-OT897k"
}
},
{
"cell_type": "code",
"source": [
"#@title Synthesize Audio (Inference)\n",
"#@markdown Upload the audio to the ``so-vits-svc/raw`` folder, and then set the model path, configuration file path, and synthesized audio name\n",
"\n",
"!python inference_main.py -m \"logs/44k/G_1600.pth\" -c \"configs/config.json\" -n \"君の知らない物語-src.wav\" -s kiritan\n"
],
"metadata": {
"cellView": "form",
"id": "dYnKuKTIj3z1"
},
"execution_count": null,
"outputs": []
}
]
}
+69
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@@ -0,0 +1,69 @@
import io
import os
# os.system("wget -P cvec/ https://huggingface.co/spaces/innnky/nanami/resolve/main/checkpoint_best_legacy_500.pt")
import gradio as gr
import librosa
import numpy as np
import soundfile
from inference.infer_tool import Svc
import logging
logging.getLogger('numba').setLevel(logging.WARNING)
logging.getLogger('markdown_it').setLevel(logging.WARNING)
logging.getLogger('urllib3').setLevel(logging.WARNING)
logging.getLogger('matplotlib').setLevel(logging.WARNING)
config_path = "configs/config.json"
model = Svc("logs/44k/G_114400.pth", "configs/config.json", cluster_model_path="logs/44k/kmeans_10000.pt")
def vc_fn(sid, input_audio, vc_transform, auto_f0,cluster_ratio, slice_db, noise_scale):
if input_audio is None:
return "You need to upload an audio", None
sampling_rate, audio = input_audio
# print(audio.shape,sampling_rate)
duration = audio.shape[0] / sampling_rate
if duration > 90:
return "请上传小于90s的音频,需要转换长音频请本地进行转换", None
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)
print(audio.shape)
out_wav_path = "temp.wav"
soundfile.write(out_wav_path, audio, 16000, format="wav")
print( cluster_ratio, auto_f0, noise_scale)
_audio = model.slice_inference(out_wav_path, sid, vc_transform, slice_db, cluster_ratio, auto_f0, noise_scale)
return "Success", (44100, _audio)
app = gr.Blocks()
with app:
with gr.Tabs():
with gr.TabItem("Basic"):
gr.Markdown(value="""
sovits4.0 在线demo
此demo为预训练底模在线demo,使用数据:云灏 即霜 辉宇·星AI 派蒙 绫地宁宁
""")
spks = list(model.spk2id.keys())
sid = gr.Dropdown(label="音色", choices=spks, value=spks[0])
vc_input3 = gr.Audio(label="上传音频(长度小于90秒)")
vc_transform = gr.Number(label="变调(整数,可以正负,半音数量,升高八度就是12)", value=0)
cluster_ratio = gr.Number(label="聚类模型混合比例,0-1之间,默认为0不启用聚类,能提升音色相似度,但会导致咬字下降(如果使用建议0.5左右)", value=0)
auto_f0 = gr.Checkbox(label="自动f0预测,配合聚类模型f0预测效果更好,会导致变调功能失效(仅限转换语音,歌声不要勾选此项会究极跑调)", value=False)
slice_db = gr.Number(label="切片阈值", value=-40)
noise_scale = gr.Number(label="noise_scale 建议不要动,会影响音质,玄学参数", value=0.4)
vc_submit = gr.Button("转换", variant="primary")
vc_output1 = gr.Textbox(label="Output Message")
vc_output2 = gr.Audio(label="Output Audio")
vc_submit.click(vc_fn, [sid, vc_input3, vc_transform,auto_f0,cluster_ratio, slice_db, noise_scale], [vc_output1, vc_output2])
app.launch()
@@ -0,0 +1,29 @@
import numpy as np
import torch
from sklearn.cluster import KMeans
def get_cluster_model(ckpt_path):
checkpoint = torch.load(ckpt_path)
kmeans_dict = {}
for spk, ckpt in checkpoint.items():
km = KMeans(ckpt["n_features_in_"])
km.__dict__["n_features_in_"] = ckpt["n_features_in_"]
km.__dict__["_n_threads"] = ckpt["_n_threads"]
km.__dict__["cluster_centers_"] = ckpt["cluster_centers_"]
kmeans_dict[spk] = km
return kmeans_dict
def get_cluster_result(model, x, speaker):
"""
x: np.array [t, 256]
return cluster class result
"""
return model[speaker].predict(x)
def get_cluster_center_result(model, x,speaker):
"""x: np.array [t, 256]"""
predict = model[speaker].predict(x)
return model[speaker].cluster_centers_[predict]
def get_center(model, x,speaker):
return model[speaker].cluster_centers_[x]
@@ -0,0 +1,89 @@
import os
from glob import glob
from pathlib import Path
import torch
import logging
import argparse
import torch
import numpy as np
from sklearn.cluster import KMeans, MiniBatchKMeans
import tqdm
logging.basicConfig(level=logging.INFO)
logger = logging.getLogger(__name__)
import time
import random
def train_cluster(in_dir, n_clusters, use_minibatch=True, verbose=False):
logger.info(f"Loading features from {in_dir}")
features = []
nums = 0
for path in tqdm.tqdm(in_dir.glob("*.soft.pt")):
features.append(torch.load(path).squeeze(0).numpy().T)
# print(features[-1].shape)
features = np.concatenate(features, axis=0)
print(nums, features.nbytes/ 1024**2, "MB , shape:",features.shape, features.dtype)
features = features.astype(np.float32)
logger.info(f"Clustering features of shape: {features.shape}")
t = time.time()
if use_minibatch:
kmeans = MiniBatchKMeans(n_clusters=n_clusters,verbose=verbose, batch_size=4096, max_iter=80).fit(features)
else:
kmeans = KMeans(n_clusters=n_clusters,verbose=verbose).fit(features)
print(time.time()-t, "s")
x = {
"n_features_in_": kmeans.n_features_in_,
"_n_threads": kmeans._n_threads,
"cluster_centers_": kmeans.cluster_centers_,
}
print("end")
return x
if __name__ == "__main__":
parser = argparse.ArgumentParser()
parser.add_argument('--dataset', type=Path, default="./dataset/44k",
help='path of training data directory')
parser.add_argument('--output', type=Path, default="logs/44k",
help='path of model output directory')
args = parser.parse_args()
checkpoint_dir = args.output
dataset = args.dataset
n_clusters = 10000
ckpt = {}
for spk in os.listdir(dataset):
if os.path.isdir(dataset/spk):
print(f"train kmeans for {spk}...")
in_dir = dataset/spk
x = train_cluster(in_dir, n_clusters, verbose=False)
ckpt[spk] = x
checkpoint_path = checkpoint_dir / f"kmeans_{n_clusters}.pt"
checkpoint_path.parent.mkdir(exist_ok=True, parents=True)
torch.save(
ckpt,
checkpoint_path,
)
# import cluster
# for spk in tqdm.tqdm(os.listdir("dataset")):
# if os.path.isdir(f"dataset/{spk}"):
# print(f"start kmeans inference for {spk}...")
# for feature_path in tqdm.tqdm(glob(f"dataset/{spk}/*.discrete.npy", recursive=True)):
# mel_path = feature_path.replace(".discrete.npy",".mel.npy")
# mel_spectrogram = np.load(mel_path)
# feature_len = mel_spectrogram.shape[-1]
# c = np.load(feature_path)
# c = utils.tools.repeat_expand_2d(torch.FloatTensor(c), feature_len).numpy()
# feature = c.T
# feature_class = cluster.get_cluster_result(feature, spk)
# np.save(feature_path.replace(".discrete.npy", ".discrete_class.npy"), feature_class)
+142
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import time
import os
import random
import numpy as np
import torch
import torch.utils.data
import modules.commons as commons
import utils
from modules.mel_processing import spectrogram_torch, spec_to_mel_torch
from utils import load_wav_to_torch, load_filepaths_and_text
# 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]])
f0 = np.load(filename + ".f0.npy")
f0, uv = utils.interpolate_f0(f0)
f0 = torch.FloatTensor(f0)
uv = torch.FloatTensor(uv)
c = torch.load(filename+ ".soft.pt")
c = utils.repeat_expand_2d(c.squeeze(0), f0.shape[0])
lmin = min(c.size(-1), spec.size(-1))
assert abs(c.size(-1) - spec.size(-1)) < 3, (c.size(-1), spec.size(-1), f0.shape, filename)
assert abs(audio_norm.shape[1]-lmin * self.hop_length) < 3 * self.hop_length
spec, c, f0, uv = spec[:, :lmin], c[:, :lmin], f0[:lmin], uv[:lmin]
audio_norm = audio_norm[:, :lmin * self.hop_length]
# if spec.shape[1] < 30:
# print("skip too short audio:", filename)
# return None
if spec.shape[1] > 800:
start = random.randint(0, spec.shape[1]-800)
end = start + 790
spec, c, f0, uv = spec[:, start:end], c[:, start:end], f0[start:end], uv[start:end]
audio_norm = audio_norm[:, start * self.hop_length : end * self.hop_length]
return c, f0, spec, audio_norm, spk, uv
def __getitem__(self, index):
return self.get_audio(self.audiopaths[index][0])
def __len__(self):
return len(self.audiopaths)
class TextAudioCollate:
def __call__(self, batch):
batch = [b for b in batch if b is not None]
input_lengths, ids_sorted_decreasing = torch.sort(
torch.LongTensor([x[0].shape[1] for x in batch]),
dim=0, descending=True)
max_c_len = max([x[0].size(1) for x in batch])
max_wav_len = max([x[3].size(1) for x in batch])
lengths = torch.LongTensor(len(batch))
c_padded = torch.FloatTensor(len(batch), batch[0][0].shape[0], max_c_len)
f0_padded = torch.FloatTensor(len(batch), max_c_len)
spec_padded = torch.FloatTensor(len(batch), batch[0][2].shape[0], max_c_len)
wav_padded = torch.FloatTensor(len(batch), 1, max_wav_len)
spkids = torch.LongTensor(len(batch), 1)
uv_padded = torch.FloatTensor(len(batch), max_c_len)
c_padded.zero_()
spec_padded.zero_()
f0_padded.zero_()
wav_padded.zero_()
uv_padded.zero_()
for i in range(len(ids_sorted_decreasing)):
row = batch[ids_sorted_decreasing[i]]
c = row[0]
c_padded[i, :, :c.size(1)] = c
lengths[i] = c.size(1)
f0 = row[1]
f0_padded[i, :f0.size(0)] = f0
spec = row[2]
spec_padded[i, :, :spec.size(1)] = spec
wav = row[3]
wav_padded[i, :, :wav.size(1)] = wav
spkids[i, 0] = row[4]
uv = row[5]
uv_padded[i, :uv.size(0)] = uv
return c_padded, f0_padded, spec_padded, wav_padded, spkids, lengths, uv_padded
@@ -0,0 +1,20 @@
name: sovits
channels:
- pytorch
- nvidia
- conda-forge
- fvcore
- iopath
- bottler
- defaults
dependencies:
- python=3.8
- pytorch
- torchvision
- fvcore
- iopath
- nvidiacub
- pyembree
- cupy
- cython
- pip
+56
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@@ -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 +1,244 @@
import hashlib
import io
import json
import logging
import os
import time
from pathlib import Path
from inference import slicer
import librosa
import numpy as np
# import onnxruntime
import parselmouth
import soundfile
import torch
import torchaudio
import cluster
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 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)
def pad_array(arr, target_length):
current_length = arr.shape[0]
if current_length >= target_length:
return arr
else:
pad_width = target_length - current_length
pad_left = pad_width // 2
pad_right = pad_width - pad_left
padded_arr = np.pad(arr, (pad_left, pad_right), 'constant', constant_values=(0, 0))
return padded_arr
class Svc(object):
def __init__(self, net_g_path, config_path,
device=None,
cluster_model_path="logs/44k/kmeans_10000.pt"):
self.net_g_path = net_g_path
if device is None:
self.dev = torch.device("cuda" if torch.cuda.is_available() else "cpu")
else:
self.dev = torch.device(device)
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.spk2id = self.hps_ms.spk
# 加载hubert
self.hubert_model = utils.get_hubert_model().to(self.dev)
self.load_model()
if os.path.exists(cluster_model_path):
self.cluster_model = cluster.get_cluster_model(cluster_model_path)
def load_model(self):
# 获取模型配置
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_unit_f0(self, in_path, tran, cluster_infer_ratio, speaker):
wav, sr = librosa.load(in_path, sr=self.target_sample)
f0 = utils.compute_f0_parselmouth(wav, sampling_rate=self.target_sample, hop_length=self.hop_size)
f0, uv = utils.interpolate_f0(f0)
f0 = torch.FloatTensor(f0)
uv = torch.FloatTensor(uv)
f0 = f0 * 2 ** (tran / 12)
f0 = f0.unsqueeze(0).to(self.dev)
uv = uv.unsqueeze(0).to(self.dev)
wav16k = librosa.resample(wav, orig_sr=self.target_sample, target_sr=16000)
wav16k = torch.from_numpy(wav16k).to(self.dev)
c = utils.get_hubert_content(self.hubert_model, wav_16k_tensor=wav16k)
c = utils.repeat_expand_2d(c.squeeze(0), f0.shape[1])
if cluster_infer_ratio !=0:
cluster_c = cluster.get_cluster_center_result(self.cluster_model, c.cpu().numpy().T, speaker).T
cluster_c = torch.FloatTensor(cluster_c).to(self.dev)
c = cluster_infer_ratio * cluster_c + (1 - cluster_infer_ratio) * c
c = c.unsqueeze(0)
return c, f0, uv
def infer(self, speaker, tran, raw_path,
cluster_infer_ratio=0,
auto_predict_f0=False,
noice_scale=0.4):
speaker_id = self.spk2id[speaker]
sid = torch.LongTensor([int(speaker_id)]).to(self.dev).unsqueeze(0)
c, f0, uv = self.get_unit_f0(raw_path, tran, cluster_infer_ratio, speaker)
if "half" in self.net_g_path and torch.cuda.is_available():
c = c.half()
with torch.no_grad():
start = time.time()
audio = self.net_g_ms.infer(c, f0=f0, g=sid, uv=uv, predict_f0=auto_predict_f0, noice_scale=noice_scale)[0,0].data.float()
use_time = time.time() - start
print("vits use time:{}".format(use_time))
return audio, audio.shape[-1]
def slice_inference(self,raw_audio_path, spk, tran, slice_db,cluster_infer_ratio, auto_predict_f0,noice_scale, pad_seconds=0.5):
wav_path = raw_audio_path
chunks = slicer.cut(wav_path, db_thresh=slice_db)
audio_data, audio_sr = slicer.chunks2audio(wav_path, chunks)
audio = []
for (slice_tag, data) in audio_data:
print(f'#=====segment start, {round(len(data) / audio_sr, 3)}s======')
# padd
pad_len = int(audio_sr * pad_seconds)
data = np.concatenate([np.zeros([pad_len]), data, np.zeros([pad_len])])
length = int(np.ceil(len(data) / audio_sr * self.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 = self.infer(spk, tran, raw_path,
cluster_infer_ratio=cluster_infer_ratio,
auto_predict_f0=auto_predict_f0,
noice_scale=noice_scale
)
_audio = out_audio.cpu().numpy()
pad_len = int(self.target_sample * pad_seconds)
_audio = _audio[pad_len:-pad_len]
audio.extend(list(_audio))
return np.array(audio)
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):
import maad
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 = utils.get_hubert_model()
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 +1,101 @@
import io
import logging
import time
from pathlib import Path
import librosa
import matplotlib.pyplot as plt
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")
def main():
import argparse
parser = argparse.ArgumentParser(description='sovits4 inference')
# 一定要设置的部分
parser.add_argument('-m', '--model_path', type=str, default="logs/44k/G_0.pth", help='模型路径')
parser.add_argument('-c', '--config_path', type=str, default="configs/config.json", help='配置文件路径')
parser.add_argument('-n', '--clean_names', type=str, nargs='+', default=["君の知らない物語-src.wav"], help='wav文件名列表,放在raw文件夹下')
parser.add_argument('-t', '--trans', type=int, nargs='+', default=[0], help='音高调整,支持正负(半音)')
parser.add_argument('-s', '--spk_list', type=str, nargs='+', default=['nen'], help='合成目标说话人名称')
# 可选项部分
parser.add_argument('-a', '--auto_predict_f0', action='store_true', default=False,
help='语音转换自动预测音高,转换歌声时不要打开这个会严重跑调')
parser.add_argument('-cm', '--cluster_model_path', type=str, default="logs/44k/kmeans_10000.pt", help='聚类模型路径,如果没有训练聚类则随便填')
parser.add_argument('-cr', '--cluster_infer_ratio', type=float, default=0, help='聚类方案占比,范围0-1,若没有训练聚类模型则填0即可')
# 不用动的部分
parser.add_argument('-sd', '--slice_db', type=int, default=-40, help='默认-40,嘈杂的音频可以-30,干声保留呼吸可以-50')
parser.add_argument('-d', '--device', type=str, default=None, help='推理设备,None则为自动选择cpu和gpu')
parser.add_argument('-ns', '--noice_scale', type=float, default=0.4, help='噪音级别,会影响咬字和音质,较为玄学')
parser.add_argument('-p', '--pad_seconds', type=float, default=0.5, help='推理音频pad秒数,由于未知原因开头结尾会有异响,pad一小段静音段后就不会出现')
parser.add_argument('-wf', '--wav_format', type=str, default='flac', help='音频输出格式')
args = parser.parse_args()
svc_model = Svc(args.model_path, args.config_path, args.device, args.cluster_model_path)
infer_tool.mkdir(["raw", "results"])
clean_names = args.clean_names
trans = args.trans
spk_list = args.spk_list
slice_db = args.slice_db
wav_format = args.wav_format
auto_predict_f0 = args.auto_predict_f0
cluster_infer_ratio = args.cluster_infer_ratio
noice_scale = args.noice_scale
pad_seconds = args.pad_seconds
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))
if slice_tag:
print('jump empty segment')
_audio = np.zeros(length)
else:
# padd
pad_len = int(audio_sr * pad_seconds)
data = np.concatenate([np.zeros([pad_len]), data, np.zeros([pad_len])])
raw_path = io.BytesIO()
soundfile.write(raw_path, data, audio_sr, format="wav")
raw_path.seek(0)
out_audio, out_sr = svc_model.infer(spk, tran, raw_path,
cluster_infer_ratio=cluster_infer_ratio,
auto_predict_f0=auto_predict_f0,
noice_scale=noice_scale
)
_audio = out_audio.cpu().numpy()
pad_len = int(svc_model.target_sample * pad_seconds)
_audio = _audio[pad_len:-pad_len]
audio.extend(list(infer_tool.pad_array(_audio, length)))
key = "auto" if auto_predict_f0 else f"{tran}key"
cluster_name = "" if cluster_infer_ratio == 0 else f"_{cluster_infer_ratio}"
res_path = f'./results/{clean_name}_{key}_{spk}{cluster_name}.{wav_format}'
soundfile.write(res_path, audio, svc_model.target_sample, format=wav_format)
if __name__ == '__main__':
main()
+420
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@@ -0,0 +1,420 @@
import copy
import math
import torch
from torch import nn
from torch.nn import functional as F
import modules.attentions as attentions
import modules.commons as commons
import modules.modules as modules
from torch.nn import Conv1d, ConvTranspose1d, AvgPool1d, Conv2d
from torch.nn.utils import weight_norm, remove_weight_norm, spectral_norm
import utils
from modules.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,
out_channels,
hidden_channels,
kernel_size,
n_layers,
gin_channels=0,
filter_channels=None,
n_heads=None,
p_dropout=None):
super().__init__()
self.out_channels = out_channels
self.hidden_channels = hidden_channels
self.kernel_size = kernel_size
self.n_layers = n_layers
self.gin_channels = gin_channels
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_mask, f0=None, noice_scale=1):
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) * noice_scale) * 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 F0Decoder(nn.Module):
def __init__(self,
out_channels,
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size,
p_dropout,
spk_channels=0):
super().__init__()
self.out_channels = out_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.spk_channels = spk_channels
self.prenet = nn.Conv1d(hidden_channels, hidden_channels, 3, padding=1)
self.decoder = attentions.FFT(
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size,
p_dropout)
self.proj = nn.Conv1d(hidden_channels, out_channels, 1)
self.f0_prenet = nn.Conv1d(1, hidden_channels , 3, padding=1)
self.cond = nn.Conv1d(spk_channels, hidden_channels, 1)
def forward(self, x, norm_f0, x_mask, spk_emb=None):
x = torch.detach(x)
if (spk_emb is not None):
x = x + self.cond(spk_emb)
x += self.f0_prenet(norm_f0)
x = self.prenet(x) * x_mask
x = self.decoder(x * x_mask, x_mask)
x = self.proj(x) * x_mask
return x
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,
sampling_rate=44100,
**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.pre = nn.Conv1d(ssl_dim, hidden_channels, kernel_size=5, padding=2)
self.enc_p = TextEncoder(
inter_channels,
hidden_channels,
filter_channels=filter_channels,
n_heads=n_heads,
n_layers=n_layers,
kernel_size=kernel_size,
p_dropout=p_dropout
)
hps = {
"sampling_rate": sampling_rate,
"inter_channels": inter_channels,
"resblock": resblock,
"resblock_kernel_sizes": resblock_kernel_sizes,
"resblock_dilation_sizes": resblock_dilation_sizes,
"upsample_rates": upsample_rates,
"upsample_initial_channel": upsample_initial_channel,
"upsample_kernel_sizes": upsample_kernel_sizes,
"gin_channels": gin_channels,
}
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)
self.f0_decoder = F0Decoder(
1,
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size,
p_dropout,
spk_channels=gin_channels
)
self.emb_uv = nn.Embedding(2, hidden_channels)
def forward(self, c, f0, uv, spec, g=None, c_lengths=None, spec_lengths=None):
g = self.emb_g(g).transpose(1,2)
# ssl prenet
x_mask = torch.unsqueeze(commons.sequence_mask(c_lengths, c.size(2)), 1).to(c.dtype)
x = self.pre(c) * x_mask + self.emb_uv(uv.long()).transpose(1,2)
# f0 predict
lf0 = 2595. * torch.log10(1. + f0.unsqueeze(1) / 700.) / 500
norm_lf0 = utils.normalize_f0(lf0, x_mask, uv)
pred_lf0 = self.f0_decoder(x, norm_lf0, x_mask, spk_emb=g)
# encoder
z_ptemp, m_p, logs_p, _ = self.enc_p(x, x_mask, f0=f0_to_coarse(f0))
z, m_q, logs_q, spec_mask = self.enc_q(spec, spec_lengths, g=g)
# flow
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)
# nsf decoder
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), pred_lf0, norm_lf0, lf0
def infer(self, c, f0, uv, g=None, noice_scale=0.35, predict_f0=False):
c_lengths = (torch.ones(c.size(0)) * c.size(-1)).to(c.device)
g = self.emb_g(g).transpose(1,2)
x_mask = torch.unsqueeze(commons.sequence_mask(c_lengths, c.size(2)), 1).to(c.dtype)
x = self.pre(c) * x_mask + self.emb_uv(uv.long()).transpose(1,2)
if predict_f0:
lf0 = 2595. * torch.log10(1. + f0.unsqueeze(1) / 700.) / 500
norm_lf0 = utils.normalize_f0(lf0, x_mask, uv, random_scale=False)
pred_lf0 = self.f0_decoder(x, norm_lf0, x_mask, spk_emb=g)
f0 = (700 * (torch.pow(10, pred_lf0 * 500 / 2595) - 1)).squeeze(1)
z_p, m_p, logs_p, c_mask = self.enc_p(x, x_mask, f0=f0_to_coarse(f0), noice_scale=noice_scale)
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 +1,349 @@
import copy
import math
import numpy as np
import torch
from torch import nn
from torch.nn import functional as F
import modules.commons as commons
import modules.modules as modules
from modules.modules import LayerNorm
class FFT(nn.Module):
def __init__(self, hidden_channels, filter_channels, n_heads, n_layers=1, 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.ffn_layers = nn.ModuleList()
self.norm_layers_1 = 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.ffn_layers.append(
FFN(hidden_channels, hidden_channels, filter_channels, kernel_size, p_dropout=p_dropout, causal=True))
self.norm_layers_1.append(LayerNorm(hidden_channels))
def forward(self, x, x_mask):
"""
x: decoder input
h: encoder output
"""
self_attn_mask = commons.subsequent_mask(x_mask.size(2)).to(device=x.device, dtype=x.dtype)
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.ffn_layers[i](x, x_mask)
y = self.drop(y)
x = self.norm_layers_1[i](x + y)
x = x * x_mask
return x
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 +1,61 @@
import torch
from torch.nn import functional as F
import modules.commons as 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 +1,342 @@
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 modules.commons as commons
from modules.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 +1,94 @@
import torch
from torchaudio.models.wav2vec2.utils import import_fairseq_model
from fairseq import checkpoint_utils
from onnxexport.model_onnx import SynthesizerTrn
import utils
def get_hubert_model():
vec_path = "hubert/checkpoint_best_legacy_500.pt"
print("load model(s) from {}".format(vec_path))
models, saved_cfg, task = checkpoint_utils.load_model_ensemble_and_task(
[vec_path],
suffix="",
)
model = models[0]
model.eval()
return model
def main(HubertExport, NetExport):
path = "SoVits4.0"
'''if HubertExport:
device = torch.device("cpu")
vec_path = "hubert/checkpoint_best_legacy_500.pt"
models, saved_cfg, task = checkpoint_utils.load_model_ensemble_and_task(
[vec_path],
suffix="",
)
original = models[0]
original.eval()
model = original
test_input = torch.rand(1, 1, 16000)
model(test_input)
torch.onnx.export(model,
test_input,
"hubert4.0.onnx",
export_params=True,
opset_version=16,
do_constant_folding=True,
input_names=['source'],
output_names=['embed'],
dynamic_axes={
'source':
{
2: "sample_length"
},
}
)'''
if NetExport:
device = torch.device("cpu")
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, 10, 256)
test_pitch = torch.rand(1, 10)
test_mel2ph = torch.LongTensor([0, 1, 2, 3, 4, 5, 6, 7, 8, 9]).unsqueeze(0)
test_uv = torch.ones(1, 10, dtype=torch.float32)
test_noise = torch.randn(1, 192, 10)
test_sid = torch.LongTensor([0])
input_names = ["c", "f0", "mel2ph", "uv", "noise", "sid"]
output_names = ["audio", ]
SVCVITS.eval()
torch.onnx.export(SVCVITS,
(
test_hidden_unit.to(device),
test_pitch.to(device),
test_mel2ph.to(device),
test_uv.to(device),
test_noise.to(device),
test_sid.to(device)
),
f"checkpoints/{path}/model.onnx",
dynamic_axes={
"c": [0, 1],
"f0": [1],
"mel2ph": [1],
"uv": [1],
"noise": [2],
},
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 +1,335 @@
import torch
from torch import nn
from torch.nn import functional as F
import modules.attentions as attentions
import modules.commons as commons
import modules.modules as modules
from torch.nn import Conv1d, ConvTranspose1d, AvgPool1d, Conv2d
from torch.nn.utils import weight_norm, remove_weight_norm, spectral_norm
import utils
from modules.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,
out_channels,
hidden_channels,
kernel_size,
n_layers,
gin_channels=0,
filter_channels=None,
n_heads=None,
p_dropout=None):
super().__init__()
self.out_channels = out_channels
self.hidden_channels = hidden_channels
self.kernel_size = kernel_size
self.n_layers = n_layers
self.gin_channels = gin_channels
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_mask, f0=None, z=None):
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 + z * 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 F0Decoder(nn.Module):
def __init__(self,
out_channels,
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size,
p_dropout,
spk_channels=0):
super().__init__()
self.out_channels = out_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.spk_channels = spk_channels
self.prenet = nn.Conv1d(hidden_channels, hidden_channels, 3, padding=1)
self.decoder = attentions.FFT(
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size,
p_dropout)
self.proj = nn.Conv1d(hidden_channels, out_channels, 1)
self.f0_prenet = nn.Conv1d(1, hidden_channels, 3, padding=1)
self.cond = nn.Conv1d(spk_channels, hidden_channels, 1)
def forward(self, x, norm_f0, x_mask, spk_emb=None):
x = torch.detach(x)
if spk_emb is not None:
x = x + self.cond(spk_emb)
x += self.f0_prenet(norm_f0)
x = self.prenet(x) * x_mask
x = self.decoder(x * x_mask, x_mask)
x = self.proj(x) * x_mask
return x
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,
sampling_rate=44100,
**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.pre = nn.Conv1d(ssl_dim, hidden_channels, kernel_size=5, padding=2)
self.enc_p = TextEncoder(
inter_channels,
hidden_channels,
filter_channels=filter_channels,
n_heads=n_heads,
n_layers=n_layers,
kernel_size=kernel_size,
p_dropout=p_dropout
)
hps = {
"sampling_rate": sampling_rate,
"inter_channels": inter_channels,
"resblock": resblock,
"resblock_kernel_sizes": resblock_kernel_sizes,
"resblock_dilation_sizes": resblock_dilation_sizes,
"upsample_rates": upsample_rates,
"upsample_initial_channel": upsample_initial_channel,
"upsample_kernel_sizes": upsample_kernel_sizes,
"gin_channels": gin_channels,
}
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)
self.f0_decoder = F0Decoder(
1,
hidden_channels,
filter_channels,
n_heads,
n_layers,
kernel_size,
p_dropout,
spk_channels=gin_channels
)
self.emb_uv = nn.Embedding(2, hidden_channels)
self.predict_f0 = False
def forward(self, c, f0, mel2ph, uv, noise=None, g=None):
decoder_inp = F.pad(c, [0, 0, 1, 0])
mel2ph_ = mel2ph.unsqueeze(2).repeat([1, 1, c.shape[-1]])
c = torch.gather(decoder_inp, 1, mel2ph_).transpose(1, 2) # [B, T, H]
c_lengths = (torch.ones(c.size(0)) * c.size(-1)).to(c.device)
g = g.unsqueeze(0)
g = self.emb_g(g).transpose(1, 2)
x_mask = torch.unsqueeze(commons.sequence_mask(c_lengths, c.size(2)), 1).to(c.dtype)
x = self.pre(c) * x_mask + self.emb_uv(uv.long()).transpose(1, 2)
if self.predict_f0:
lf0 = 2595. * torch.log10(1. + f0.unsqueeze(1) / 700.) / 500
norm_lf0 = utils.normalize_f0(lf0, x_mask, uv, random_scale=False)
pred_lf0 = self.f0_decoder(x, norm_lf0, x_mask, spk_emb=g)
f0 = (700 * (torch.pow(10, pred_lf0 * 500 / 2595) - 1)).squeeze(1)
z_p, m_p, logs_p, c_mask = self.enc_p(x, x_mask, f0=f0_to_coarse(f0), z=noise)
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 +1,83 @@
import os
import argparse
import re
from tqdm import tqdm
from random import shuffle
import json
import wave
config_template = json.load(open("configs/config.json"))
pattern = re.compile(r'^[\.a-zA-Z0-9_\/]+$')
def get_wav_duration(file_path):
with wave.open(file_path, 'rb') as wav_file:
# 获取音频帧数
n_frames = wav_file.getnframes()
# 获取采样率
framerate = wav_file.getframerate()
# 计算时长(秒)
duration = n_frames / float(framerate)
return duration
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/44k", 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))]
new_wavs = []
for file in wavs:
if not file.endswith("wav"):
continue
if not pattern.match(file):
print(f"warning:文件名{file}中包含非字母数字下划线,可能会导致错误。(也可能不会)")
if get_wav_duration(file) < 0.3:
print("skip too short audio:", file)
continue
new_wavs.append(file)
wavs = new_wavs
shuffle(wavs)
train += wavs[2:-2]
val += wavs[:2]
test += wavs[-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["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 +1,62 @@
import math
import multiprocessing
import os
import argparse
from random import shuffle
import torch
from glob import glob
from tqdm import tqdm
import utils
import logging
logging.getLogger('numba').setLevel(logging.WARNING)
import librosa
import numpy as np
hps = utils.get_hparams_from_file("configs/config.json")
sampling_rate = hps.data.sampling_rate
hop_length = hps.data.hop_length
def process_one(filename, hmodel):
# print(filename)
wav, sr = librosa.load(filename, sr=sampling_rate)
soft_path = filename + ".soft.pt"
if not os.path.exists(soft_path):
devive = torch.device("cuda" if torch.cuda.is_available() else "cpu")
wav16k = librosa.resample(wav, orig_sr=sampling_rate, target_sr=16000)
wav16k = torch.from_numpy(wav16k).to(devive)
c = utils.get_hubert_content(hmodel, wav_16k_tensor=wav16k)
torch.save(c.cpu(), soft_path)
f0_path = filename + ".f0.npy"
if not os.path.exists(f0_path):
f0 = utils.compute_f0_dio(wav, sampling_rate=sampling_rate, hop_length=hop_length)
np.save(f0_path, f0)
def process_batch(filenames):
print("Loading hubert for content...")
device = "cuda" if torch.cuda.is_available() else "cpu"
hmodel = utils.get_hubert_model().to(device)
print("Loaded hubert.")
for filename in tqdm(filenames):
process_one(filename, hmodel)
if __name__ == "__main__":
parser = argparse.ArgumentParser()
parser.add_argument("--in_dir", type=str, default="dataset/44k", help="path to input dir")
args = parser.parse_args()
filenames = glob(f'{args.in_dir}/*/*.wav', recursive=True) # [:10]
shuffle(filenames)
multiprocessing.set_start_method('spawn',force=True)
num_processes = 1
chunk_size = int(math.ceil(len(filenames) / num_processes))
chunks = [filenames[i:i + chunk_size] for i in range(0, len(filenames), chunk_size)]
print([len(c) for c in chunks])
processes = [multiprocessing.Process(target=process_batch, args=(chunk,)) for chunk in chunks]
for p in processes:
p.start()
@@ -0,0 +1,19 @@
Flask
Flask_Cors
gradio
numpy
#pyworld==0.2.5
pyworld==0.2.12
scipy==1.7.3
SoundFile==0.12.1
torch==1.13.1
torchaudio==0.13.1
tqdm
scikit-maad
praat-parselmouth
onnx
onnxsim
onnxoptimizer
fairseq==0.12.2
librosa==0.8.1
tensorboard
@@ -0,0 +1,9 @@
numpy==1.20.3
pyworld==0.2.5
scipy==1.7.3
SoundFile==0.12.1
tqdm
scikit-maad
praat-parselmouth
fairseq==0.12.2
librosa==0.8.1
@@ -0,0 +1,23 @@
librosa==0.9.2
fairseq==0.12.2
Flask==2.1.2
Flask_Cors==3.0.10
gradio==3.4.1
numpy==1.20.0
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==1.12.0+cu116
torchaudio==0.12.0+cu116
tqdm==4.63.0
scikit-maad
praat-parselmouth
onnx
onnxsim
onnxoptimizer
+48
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@@ -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, sr=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=44100, 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/44k", 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
+22
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@@ -0,0 +1,22 @@
from data_utils import TextAudioSpeakerLoader
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
+310
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@@ -0,0 +1,310 @@
import logging
import multiprocessing
import time
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 modules.commons as commons
import utils
from data_utils import TextAudioSpeakerLoader, TextAudioCollate
from models import (
SynthesizerTrn,
MultiPeriodDiscriminator,
)
from modules.losses import (
kl_loss,
generator_loss, discriminator_loss, feature_loss
)
from modules.mel_processing import mel_spectrogram_torch, spec_to_mel_torch
torch.backends.cudnn.benchmark = True
global_step = 0
start_time = time.time()
# 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"))
# for pytorch on win, backend use gloo
dist.init_process_group(backend= 'gloo' if os.name == 'nt' else 'nccl', init_method='env://', world_size=n_gpus, rank=rank)
torch.manual_seed(hps.train.seed)
torch.cuda.set_device(rank)
collate_fn = TextAudioCollate()
train_dataset = TextAudioSpeakerLoader(hps.data.training_files, hps)
num_workers = 5 if multiprocessing.cpu_count() > 4 else multiprocessing.cpu_count()
train_loader = DataLoader(train_dataset, num_workers=num_workers, shuffle=False, pin_memory=True,
batch_size=hps.train.batch_size, collate_fn=collate_fn)
if rank == 0:
eval_dataset = TextAudioSpeakerLoader(hps.data.validation_files, hps)
eval_loader = DataLoader(eval_dataset, num_workers=1, shuffle=False,
batch_size=1, pin_memory=False,
drop_last=False, collate_fn=collate_fn)
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])
skip_optimizer = False
try:
_, _, _, epoch_str = utils.load_checkpoint(utils.latest_checkpoint_path(hps.model_dir, "G_*.pth"), net_g,
optim_g, skip_optimizer)
_, _, _, epoch_str = utils.load_checkpoint(utils.latest_checkpoint_path(hps.model_dir, "D_*.pth"), net_d,
optim_d, skip_optimizer)
epoch_str = max(epoch_str, 1)
global_step = (epoch_str - 1) * len(train_loader)
except:
print("load old checkpoint failed...")
epoch_str = 1
global_step = 0
if skip_optimizer:
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, lengths, uv = 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)
uv = uv.cuda(rank, non_blocking=True)
lengths = lengths.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), pred_lf0, norm_lf0, lf0 = net_g(c, f0, uv, spec, g=g, c_lengths=lengths,
spec_lengths=lengths)
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_lf0 = F.mse_loss(pred_lf0, lf0)
loss_gen_all = loss_gen + loss_fm + loss_mel + loss_kl + loss_lf0
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(f"Losses: {[x.item() for x in losses]}, step: {global_step}, lr: {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,
"loss/g/lf0": loss_lf0})
# 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()),
"all/lf0": utils.plot_data_to_numpy(lf0[0, 0, :].cpu().numpy(),
pred_lf0[0, 0, :].detach().cpu().numpy()),
"all/norm_lf0": utils.plot_data_to_numpy(lf0[0, 0, :].cpu().numpy(),
norm_lf0[0, 0, :].detach().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)))
keep_ckpts = getattr(hps.train, 'keep_ckpts', 0)
if keep_ckpts > 0:
utils.clean_checkpoints(path_to_models=hps.model_dir, n_ckpts_to_keep=keep_ckpts, sort_by_time=True)
global_step += 1
if rank == 0:
global start_time
now = time.time()
durtaion = format(now - start_time, '.2f')
logger.info(f'====> Epoch: {epoch}, cost {durtaion} s')
start_time = now
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, _, uv = 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)
uv= uv[: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, uv, g=g)
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 +1,18 @@
Flask
Flask_Cors
gradio
numpy==1.20.3
pyworld==0.2.5
scipy==1.7.3
SoundFile==0.12.1
torch==1.13.1
torchaudio==0.13.1
tqdm
scikit-maad
praat-parselmouth
onnx
onnxsim
onnxoptimizer
fairseq==0.12.2
librosa==0.8.1
tensorboard
+502
View File
@@ -0,0 +1,502 @@
import os
import glob
import re
import sys
import argparse
import logging
import json
import subprocess
import random
import librosa
import numpy as np
from scipy.io.wavfile import read
import torch
from torch.nn import functional as F
from modules.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 normalize_f0(f0, random_scale=True):
# f0_norm = f0.clone() # create a copy of the input Tensor
# batch_size, _, frame_length = f0_norm.shape
# for i in range(batch_size):
# means = torch.mean(f0_norm[i, 0, :])
# if random_scale:
# factor = random.uniform(0.8, 1.2)
# else:
# factor = 1
# f0_norm[i, 0, :] = (f0_norm[i, 0, :] - means) * factor
# return f0_norm
# def normalize_f0(f0, random_scale=True):
# means = torch.mean(f0[:, 0, :], dim=1, keepdim=True)
# if random_scale:
# factor = torch.Tensor(f0.shape[0],1).uniform_(0.8, 1.2).to(f0.device)
# else:
# factor = torch.ones(f0.shape[0], 1, 1).to(f0.device)
# f0_norm = (f0 - means.unsqueeze(-1)) * factor.unsqueeze(-1)
# return f0_norm
def normalize_f0(f0, x_mask, uv, random_scale=True):
# calculate means based on x_mask
uv_sum = torch.sum(uv, dim=1, keepdim=True)
uv_sum[uv_sum == 0] = 9999
means = torch.sum(f0[:, 0, :] * uv, dim=1, keepdim=True) / uv_sum
if random_scale:
factor = torch.Tensor(f0.shape[0], 1).uniform_(0.8, 1.2).to(f0.device)
else:
factor = torch.ones(f0.shape[0], 1).to(f0.device)
# normalize f0 based on means and factor
f0_norm = (f0 - means.unsqueeze(-1)) * factor.unsqueeze(-1)
if torch.isnan(f0_norm).any():
exit(0)
return f0_norm * x_mask
def plot_data_to_numpy(x, y):
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))
plt.plot(x)
plt.plot(y)
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 interpolate_f0(f0):
'''
对F0进行插值处理
'''
data = np.reshape(f0, (f0.size, 1))
vuv_vector = np.zeros((data.size, 1), dtype=np.float32)
vuv_vector[data > 0.0] = 1.0
vuv_vector[data <= 0.0] = 0.0
ip_data = data
frame_number = data.size
last_value = 0.0
for i in range(frame_number):
if data[i] <= 0.0:
j = i + 1
for j in range(i + 1, frame_number):
if data[j] > 0.0:
break
if j < frame_number - 1:
if last_value > 0.0:
step = (data[j] - data[i - 1]) / float(j - i)
for k in range(i, j):
ip_data[k] = data[i - 1] + step * (k - i + 1)
else:
for k in range(i, j):
ip_data[k] = data[j]
else:
for k in range(i, frame_number):
ip_data[k] = last_value
else:
ip_data[i] = data[i]
last_value = data[i]
return ip_data[:,0], vuv_vector[:,0]
def compute_f0_parselmouth(wav_numpy, p_len=None, sampling_rate=44100, hop_length=512):
import parselmouth
x = wav_numpy
if p_len is None:
p_len = x.shape[0]//hop_length
else:
assert abs(p_len-x.shape[0]//hop_length) < 4, "pad length error"
time_step = hop_length / sampling_rate * 1000
f0_min = 50
f0_max = 1100
f0 = parselmouth.Sound(x, sampling_rate).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')
return f0
def resize_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 compute_f0_dio(wav_numpy, p_len=None, sampling_rate=44100, hop_length=512):
import pyworld
if p_len is None:
p_len = wav_numpy.shape[0]//hop_length
f0, t = pyworld.dio(
wav_numpy.astype(np.double),
fs=sampling_rate,
f0_ceil=800,
frame_period=1000 * hop_length / sampling_rate,
)
f0 = pyworld.stonemask(wav_numpy.astype(np.double), f0, t, sampling_rate)
for index, pitch in enumerate(f0):
f0[index] = round(pitch, 1)
return resize_f0(f0, p_len)
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():
vec_path = "hubert/checkpoint_best_legacy_500.pt"
print("load model(s) from {}".format(vec_path))
from fairseq import checkpoint_utils
models, saved_cfg, task = checkpoint_utils.load_model_ensemble_and_task(
[vec_path],
suffix="",
)
model = models[0]
model.eval()
return model
def get_hubert_content(hmodel, wav_16k_tensor):
feats = wav_16k_tensor
if feats.dim() == 2: # double channels
feats = feats.mean(-1)
assert feats.dim() == 1, feats.dim()
feats = feats.view(1, -1)
padding_mask = torch.BoolTensor(feats.shape).fill_(False)
inputs = {
"source": feats.to(wav_16k_tensor.device),
"padding_mask": padding_mask.to(wav_16k_tensor.device),
"output_layer": 9, # layer 9
}
with torch.no_grad():
logits = hmodel.extract_features(**inputs)
feats = hmodel.final_proj(logits[0])
return feats.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 load_checkpoint(checkpoint_path, model, optimizer=None, skip_optimizer=False):
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 optimizer is not None and not skip_optimizer 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:
# assert "dec" in k or "disc" in k
# print("load", k)
new_state_dict[k] = saved_state_dict[k]
assert saved_state_dict[k].shape == v.shape, (saved_state_dict[k].shape, v.shape)
except:
print("error, %s is not in the checkpoint" % k)
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)
print("load ")
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)
def clean_checkpoints(path_to_models='logs/44k/', 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
def repeat_expand_2d(content, target_len):
# content : [h, t]
src_len = content.shape[-1]
target = torch.zeros([content.shape[0], target_len], dtype=torch.float).to(content.device)
temp = torch.arange(src_len+1) * target_len / src_len
current_pos = 0
for i in range(target_len):
if i < temp[current_pos+1]:
target[:, i] = content[:, current_pos]
else:
current_pos += 1
target[:, i] = content[:, current_pos]
return target
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]