Add files via upload

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Justin John
2023-10-06 14:20:27 +05:30
committed by GitHub
parent aae06f16a8
commit edf3e2aa08
38 changed files with 12308 additions and 0 deletions
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# syntax=docker/dockerfile:1
#FROM python:3.10-bullseye
# Alternatively,
FROM nvidia/cuda:12.0.1-runtime-ubuntu22.04
ENV DEBIAN_FRONTEND=noninteractive
# NB: ffmpeg for transcoding opus -> wav
# NB(bt,2023-05-04): Installing lsof, htop, ripgrep, and other debugging tools
# NB(bt,2023-05-04): Installing rsync to monitor time it takes for files to sync
RUN apt-get update && apt-get install -y \
build-essential \
curl \
ffmpeg \
htop \
libsndfile1 \
lsof \
nvidia-driver-515 \
python3-pip \
python3.10 \
python3.10-dev \
python3.10-venv \
ripgrep \
rsync \
vim \
wget \
--no-install-recommends \
&& apt-get clean autoclean && apt-get autoremove -y && rm -rf /var/lib/{apt,dpkg,cache,log}/
#ENV DEBIAN_FRONTEND=noninteractive
#
#RUN apt-get update && apt-get install -y \
# rsync \
# --no-install-recommends \
# && apt-get clean autoclean && apt-get autoremove -y && rm -rf /var/lib/{apt,dpkg,cache,log}/
RUN mkdir /python_install
RUN mkdir /model_code
WORKDIR /python_install
COPY requirements.txt .
# NB: Kubernetes will copy over the venv we create here.
RUN python3 -m venv python \
&& . python/bin/activate \
&& pip install -r requirements.txt
&& pip install piper-tts
# NB: The model code is uploaded next
WORKDIR /model_code
COPY . .
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MIT License
Copyright (c) 2023 liujing04
Copyright (c) 2023 源文雨
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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import soundfile as sf
import torch, pdb, os, warnings, librosa
import numpy as np
from tqdm import tqdm
import torch
dim_c = 4
class Conv_TDF_net_trim:
def __init__(
self, device, model_name, target_name, L, dim_f, dim_t, n_fft, hop=1024
):
super(Conv_TDF_net_trim, self).__init__()
self.dim_f = dim_f
self.dim_t = 2**dim_t
self.n_fft = n_fft
self.hop = hop
self.n_bins = self.n_fft // 2 + 1
self.chunk_size = hop * (self.dim_t - 1)
self.window = torch.hann_window(window_length=self.n_fft, periodic=True).to(
device
)
self.target_name = target_name
self.blender = "blender" in model_name
out_c = dim_c * 4 if target_name == "*" else dim_c
self.freq_pad = torch.zeros(
[1, out_c, self.n_bins - self.dim_f, self.dim_t]
).to(device)
self.n = L // 2
def stft(self, x):
x = x.reshape([-1, self.chunk_size])
x = torch.stft(
x,
n_fft=self.n_fft,
hop_length=self.hop,
window=self.window,
center=True,
return_complex=True,
)
x = torch.view_as_real(x)
x = x.permute([0, 3, 1, 2])
x = x.reshape([-1, 2, 2, self.n_bins, self.dim_t]).reshape(
[-1, dim_c, self.n_bins, self.dim_t]
)
return x[:, :, : self.dim_f]
def istft(self, x, freq_pad=None):
freq_pad = (
self.freq_pad.repeat([x.shape[0], 1, 1, 1])
if freq_pad is None
else freq_pad
)
x = torch.cat([x, freq_pad], -2)
c = 4 * 2 if self.target_name == "*" else 2
x = x.reshape([-1, c, 2, self.n_bins, self.dim_t]).reshape(
[-1, 2, self.n_bins, self.dim_t]
)
x = x.permute([0, 2, 3, 1])
x = x.contiguous()
x = torch.view_as_complex(x)
x = torch.istft(
x, n_fft=self.n_fft, hop_length=self.hop, window=self.window, center=True
)
return x.reshape([-1, c, self.chunk_size])
def get_models(device, dim_f, dim_t, n_fft):
return Conv_TDF_net_trim(
device=device,
model_name="Conv-TDF",
target_name="vocals",
L=11,
dim_f=dim_f,
dim_t=dim_t,
n_fft=n_fft,
)
warnings.filterwarnings("ignore")
import sys
now_dir = os.getcwd()
sys.path.append(now_dir)
from config import Config
cpu = torch.device("cpu")
device=Config().device
# if torch.cuda.is_available():
# device = torch.device("cuda:0")
# elif torch.backends.mps.is_available():
# device = torch.device("mps")
# else:
# device = torch.device("cpu")
class Predictor:
def __init__(self, args):
self.args = args
self.model_ = get_models(
device=cpu, dim_f=args.dim_f, dim_t=args.dim_t, n_fft=args.n_fft
)
import onnxruntime as ort
print(ort.get_available_providers())
self.model = ort.InferenceSession(
os.path.join(args.onnx, self.model_.target_name + ".onnx"),
providers=["CUDAExecutionProvider", "DmlExecutionProvider","CPUExecutionProvider"],
)
print("onnx load done")
def demix(self, mix):
samples = mix.shape[-1]
margin = self.args.margin
chunk_size = self.args.chunks * 44100
assert not margin == 0, "margin cannot be zero!"
if margin > chunk_size:
margin = chunk_size
segmented_mix = {}
if self.args.chunks == 0 or samples < chunk_size:
chunk_size = samples
counter = -1
for skip in range(0, samples, chunk_size):
counter += 1
s_margin = 0 if counter == 0 else margin
end = min(skip + chunk_size + margin, samples)
start = skip - s_margin
segmented_mix[skip] = mix[:, start:end].copy()
if end == samples:
break
sources = self.demix_base(segmented_mix, margin_size=margin)
"""
mix:(2,big_sample)
segmented_mix:offset->(2,small_sample)
sources:(1,2,big_sample)
"""
return sources
def demix_base(self, mixes, margin_size):
chunked_sources = []
progress_bar = tqdm(total=len(mixes))
progress_bar.set_description("Processing")
for mix in mixes:
cmix = mixes[mix]
sources = []
n_sample = cmix.shape[1]
model = self.model_
trim = model.n_fft // 2
gen_size = model.chunk_size - 2 * trim
pad = gen_size - n_sample % gen_size
mix_p = np.concatenate(
(np.zeros((2, trim)), cmix, np.zeros((2, pad)), np.zeros((2, trim))), 1
)
mix_waves = []
i = 0
while i < n_sample + pad:
waves = np.array(mix_p[:, i : i + model.chunk_size])
mix_waves.append(waves)
i += gen_size
mix_waves = torch.tensor(mix_waves, dtype=torch.float32).to(cpu)
with torch.no_grad():
_ort = self.model
spek = model.stft(mix_waves)
if self.args.denoise:
spec_pred = (
-_ort.run(None, {"input": -spek.cpu().numpy()})[0] * 0.5
+ _ort.run(None, {"input": spek.cpu().numpy()})[0] * 0.5
)
tar_waves = model.istft(torch.tensor(spec_pred))
else:
tar_waves = model.istft(
torch.tensor(_ort.run(None, {"input": spek.cpu().numpy()})[0])
)
tar_signal = (
tar_waves[:, :, trim:-trim]
.transpose(0, 1)
.reshape(2, -1)
.numpy()[:, :-pad]
)
start = 0 if mix == 0 else margin_size
end = None if mix == list(mixes.keys())[::-1][0] else -margin_size
if margin_size == 0:
end = None
sources.append(tar_signal[:, start:end])
progress_bar.update(1)
chunked_sources.append(sources)
_sources = np.concatenate(chunked_sources, axis=-1)
# del self.model
progress_bar.close()
return _sources
def prediction(self, m, vocal_root, others_root, format):
os.makedirs(vocal_root, exist_ok=True)
os.makedirs(others_root, exist_ok=True)
basename = os.path.basename(m)
mix, rate = librosa.load(m, mono=False, sr=44100)
if mix.ndim == 1:
mix = np.asfortranarray([mix, mix])
mix = mix.T
sources = self.demix(mix.T)
opt = sources[0].T
if format in ["wav", "flac"]:
sf.write(
"%s/%s_main_vocal.%s" % (vocal_root, basename, format), mix - opt, rate
)
sf.write("%s/%s_others.%s" % (others_root, basename, format), opt, rate)
else:
path_vocal = "%s/%s_main_vocal.wav" % (vocal_root, basename)
path_other = "%s/%s_others.wav" % (others_root, basename)
sf.write(path_vocal, mix - opt, rate)
sf.write(path_other, opt, rate)
if os.path.exists(path_vocal):
os.system(
"ffmpeg -i %s -vn %s -q:a 2 -y"
% (path_vocal, path_vocal[:-4] + ".%s" % format)
)
if os.path.exists(path_other):
os.system(
"ffmpeg -i %s -vn %s -q:a 2 -y"
% (path_other, path_other[:-4] + ".%s" % format)
)
class MDXNetDereverb:
def __init__(self, chunks):
self.onnx = "uvr5_weights/onnx_dereverb_By_FoxJoy"
self.shifts = 10 #'Predict with randomised equivariant stabilisation'
self.mixing = "min_mag" # ['default','min_mag','max_mag']
self.chunks = chunks
self.margin = 44100
self.dim_t = 9
self.dim_f = 3072
self.n_fft = 6144
self.denoise = True
self.pred = Predictor(self)
def _path_audio_(self, input, vocal_root, others_root, format):
self.pred.prediction(input, vocal_root, others_root, format)
if __name__ == "__main__":
dereverb = MDXNetDereverb(15)
from time import time as ttime
t0 = ttime()
dereverb._path_audio_(
"雪雪伴奏对消HP5.wav",
"vocal",
"others",
)
t1 = ttime()
print(t1 - t0)
"""
runtime\python.exe MDXNet.py
6G:
15/9:0.8G->6.8G
14:0.8G->6.5G
25:炸
half15:0.7G->6.6G,22.69s
fp32-15:0.7G->6.6G,20.85s
"""
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本软件及其相关代码以MIT协议开源,作者不对软件具备任何控制力,使用软件者、传播软件导出的声音者自负全责。
如不认可该条款,则不能使用或引用软件包内任何代码和文件。
特此授予任何获得本软件和相关文档文件(以下简称“软件”)副本的人免费使用、复制、修改、合并、出版、分发、再授权和/或销售本软件的权利,以及授予本软件所提供的人使用本软件的权利,但须符合以下条件:
上述版权声明和本许可声明应包含在软件的所有副本或实质部分中。
软件是“按原样”提供的,没有任何明示或暗示的保证,包括但不限于适销性、适用于特定目的和不侵权的保证。在任何情况下,作者或版权持有人均不承担因软件或软件的使用或其他交易而产生、产生或与之相关的任何索赔、损害赔偿或其他责任,无论是在合同诉讼、侵权诉讼还是其他诉讼中。
The LICENCEs for related libraries are as follows.
相关引用库协议如下:
ContentVec
https://github.com/auspicious3000/contentvec/blob/main/LICENSE
MIT License
VITS
https://github.com/jaywalnut310/vits/blob/main/LICENSE
MIT License
HIFIGAN
https://github.com/jik876/hifi-gan/blob/master/LICENSE
MIT License
gradio
https://github.com/gradio-app/gradio/blob/main/LICENSE
Apache License 2.0
ffmpeg
https://github.com/FFmpeg/FFmpeg/blob/master/COPYING.LGPLv3
https://github.com/BtbN/FFmpeg-Builds/releases/download/autobuild-2021-02-28-12-32/ffmpeg-n4.3.2-160-gfbb9368226-win64-lgpl-4.3.zip
LPGLv3 License
MIT License
ultimatevocalremovergui
https://github.com/Anjok07/ultimatevocalremovergui/blob/master/LICENSE
https://github.com/yang123qwe/vocal_separation_by_uvr5
MIT License
audio-slicer
https://github.com/openvpi/audio-slicer/blob/main/LICENSE
MIT License
PySimpleGUI
https://github.com/PySimpleGUI/PySimpleGUI/blob/master/license.txt
LPGLv3 License
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# TTS and RVC Inference Script
## Overview
This Python script runs both Text-To-Speech (TTS) and Retrieval-based-Voice-Conversion (RVC) inferences. It uses Piper for TTS and a custom RVC voice conversion model. The script takes a variety of command-line arguments to specify the models, configurations, and audio files to use.
## Prerequisites
- Python 3.x
- PyTorch 2.0 or higher
- Piper TTS
- ffmpeg (for audio processing)
### Installation
1. Install Python 3.x and PyTorch 2.0 or higher.
2. Install Piper TTS via pip:
```bash
pip install piper-tts
```
3. **Download Piper Voice Models**
Piper voice models can be downloaded from [Hugging Face's Piper Voices Repository](https://huggingface.co/rhasspy/piper-voices/tree/v1.0.0/en/en_US).
4. Make sure `ffmpeg` is installed and available in the system's PATH.
- For Ubuntu:
```bash
sudo apt-get install ffmpeg
```
- For macOS:
```bash
brew install ffmpeg
```
## Usage
To run the script, you need to pass in several command-line arguments.
### TTS Arguments
- `--tts_model_path`: Required. The path to the TTS model.
- `--tts_config_path`: Required. The path to the TTS configuration file.
- `--text`: Required. The text you want to convert to speech.
#### Additional TTS Options
- `--length_scale`: Optional. Phoneme length scale for TTS. Default is 1.0.
- `--noise_scale`: Optional. Noise scale for TTS. Default is 0.5.
- `--noise_w`: Optional. Phoneme width noise for TTS. Default is 0.3.
- `--sentence_silence`: Optional. Seconds of silence after each sentence in TTS. Default is 0.0.
### RVC Arguments
- `--model_path`: Path to the VC model.
- `--model_index_path`: (Optional) Path to the VC model index.
- `--hubert_model_path`: (Optional) Path to the Hubert model. Default is `hubert_base.pt`.
- `--input_audio_filename`: Path to the input audio file for VC.
- `--output_audio_filename`: Path to save the output audio file from VC.
... (Other optional arguments for fine-tuning the VC process)
`--f0up_key`: The f0 key for the input audio file.
`--index_path`: Path to the index file.
`--f0method`: F0 estimation method to be used ('pm' 'harvest' 'crepe' or rmvpe).
`--index_rate`: The rate for the index. (Search feature ratio)
`--device`: The computation device to be used ('cuda:0' for GPU or 'cpu' for CPU).
`--is_half`: Boolean value indicating whether to use half precision. Use 'True' for half-precision and 'False' for full-precision.
`--filter_radius`: The radius of the filter. (If >=3: apply median filtering to the harvested pitch results. The value represents the filter radius and can reduce breathiness.)
`--resample_sr`: The sample rate for resampling. (Resample the output audio in post-processing to the final sample rate. Set to 0 for no resampling)
`--rms_mix_rate`: The RMS mix rate. (Use the volume envelope of the input to replace or mix with the volume envelope of the output. The closer the ratio is to 1, the more the output envelope is used)
`--protect`: The protect value. (Protect voiceless consonants and breath sounds to prevent artifacts such as tearing in electronic music. Set to 0.5 to disable. Decrease the value to increase protection, but it may reduce indexing accuracy:)
### Example:
```bash
python fakeyou_rvc_tts_infer.py --tts_model_path "path/to/tts/model" --tts_config_path "path/to/tts/config" --text "Hello, World" --model_path "path/to/vc/model" --input_audio_filename "path/to/input/audio.wav" --output_audio_filename "path/to/output/audio.wav"
```
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{
"nbformat": 4,
"nbformat_minor": 0,
"metadata": {
"colab": {
"private_outputs": true,
"provenance": []
},
"kernelspec": {
"name": "python3",
"display_name": "Python 3"
},
"language_info": {
"name": "python"
},
"accelerator": "GPU",
"gpuClass": "standard"
},
"cells": [
{
"cell_type": "markdown",
"source": [
"[![Open In Colab](https://colab.research.google.com/assets/colab-badge.svg)](https://colab.research.google.com/github/RVC-Project/Retrieval-based-Voice-Conversion-WebUI/blob/main/Retrieval_based_Voice_Conversion_WebUI.ipynb)"
],
"metadata": {
"id": "ZFFCx5J80SGa"
}
},
{
"cell_type": "code",
"execution_count": null,
"metadata": {
"id": "GmFP6bN9dvOq"
},
"outputs": [],
"source": [
"# @title 查看显卡\n",
"!nvidia-smi"
]
},
{
"cell_type": "code",
"source": [
"# @title 安装依赖\n",
"!apt-get -y install build-essential python3-dev ffmpeg\n",
"!pip3 install --upgrade setuptools wheel\n",
"!pip3 install --upgrade pip\n",
"!pip3 install faiss-cpu==1.7.2 fairseq gradio==3.14.0 ffmpeg ffmpeg-python praat-parselmouth pyworld numpy==1.23.5 numba==0.56.4 librosa==0.9.2"
],
"metadata": {
"id": "wjddIFr1oS3W"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"# @title 克隆仓库\n",
"\n",
"!git clone --depth=1 -b stable https://github.com/fumiama/Retrieval-based-Voice-Conversion-WebUI\n",
"%cd /content/Retrieval-based-Voice-Conversion-WebUI\n",
"!mkdir -p pretrained uvr5_weights"
],
"metadata": {
"id": "ge_97mfpgqTm"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"# @title 更新仓库(一般无需执行)\n",
"!git pull"
],
"metadata": {
"id": "BLDEZADkvlw1"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"# @title 安装aria2\n",
"!apt -y install -qq aria2"
],
"metadata": {
"id": "pqE0PrnuRqI2"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"# @title 下载底模\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/D32k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o D32k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/D40k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o D40k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/D48k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o D48k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/G32k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o G32k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/G40k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o G40k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/G48k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o G48k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/f0D32k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o f0D32k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/f0D40k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o f0D40k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/f0D48k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o f0D48k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/f0G32k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o f0G32k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/f0G40k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o f0G40k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/f0G48k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o f0G48k.pth"
],
"metadata": {
"id": "UG3XpUwEomUz"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"# @title 下载人声分离模型\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/uvr5_weights/HP2-人声vocals+非人声instrumentals.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/uvr5_weights -o HP2-人声vocals+非人声instrumentals.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/uvr5_weights/HP5-主旋律人声vocals+其他instrumentals.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/uvr5_weights -o HP5-主旋律人声vocals+其他instrumentals.pth"
],
"metadata": {
"id": "HugjmZqZRuiF"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"# @title 下载hubert_base\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/hubert_base.pt -d /content/Retrieval-based-Voice-Conversion-WebUI -o hubert_base.pt"
],
"metadata": {
"id": "2RCaT9FTR0ej"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"# @title 挂载谷歌云盘\n",
"\n",
"from google.colab import drive\n",
"\n",
"drive.mount(\"/content/drive\")"
],
"metadata": {
"id": "jwu07JgqoFON"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"# @title 从谷歌云盘加载打包好的数据集到/content/dataset\n",
"\n",
"# @markdown 数据集位置\n",
"DATASET = (\n",
" \"/content/drive/MyDrive/dataset/lulu20230327_32k.zip\" # @param {type:\"string\"}\n",
")\n",
"\n",
"!mkdir -p /content/dataset\n",
"!unzip -d /content/dataset -B {DATASET}"
],
"metadata": {
"id": "Mwk7Q0Loqzjx"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"# @title 重命名数据集中的重名文件\n",
"!ls -a /content/dataset/\n",
"!rename 's/(\\w+)\\.(\\w+)~(\\d*)/$1_$3.$2/' /content/dataset/*.*~*"
],
"metadata": {
"id": "PDlFxWHWEynD"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"# @title 启动web\n",
"%cd /content/Retrieval-based-Voice-Conversion-WebUI\n",
"# %load_ext tensorboard\n",
"# %tensorboard --logdir /content/Retrieval-based-Voice-Conversion-WebUI/logs\n",
"!python3 infer-web.py --colab --pycmd python3"
],
"metadata": {
"id": "7vh6vphDwO0b"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"# @title 手动将训练后的模型文件备份到谷歌云盘\n",
"# @markdown 需要自己查看logs文件夹下模型的文件名,手动修改下方命令末尾的文件名\n",
"\n",
"# @markdown 模型名\n",
"MODELNAME = \"lulu\" # @param {type:\"string\"}\n",
"# @markdown 模型epoch\n",
"MODELEPOCH = 9600 # @param {type:\"integer\"}\n",
"\n",
"!cp /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/G_{MODELEPOCH}.pth /content/drive/MyDrive/{MODELNAME}_D_{MODELEPOCH}.pth\n",
"!cp /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/D_{MODELEPOCH}.pth /content/drive/MyDrive/{MODELNAME}_G_{MODELEPOCH}.pth\n",
"!cp /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/added_*.index /content/drive/MyDrive/\n",
"!cp /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/total_*.npy /content/drive/MyDrive/\n",
"\n",
"!cp /content/Retrieval-based-Voice-Conversion-WebUI/weights/{MODELNAME}.pth /content/drive/MyDrive/{MODELNAME}{MODELEPOCH}.pth"
],
"metadata": {
"id": "FgJuNeAwx5Y_"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"# @title 从谷歌云盘恢复pth\n",
"# @markdown 需要自己查看logs文件夹下模型的文件名,手动修改下方命令末尾的文件名\n",
"\n",
"# @markdown 模型名\n",
"MODELNAME = \"lulu\" # @param {type:\"string\"}\n",
"# @markdown 模型epoch\n",
"MODELEPOCH = 7500 # @param {type:\"integer\"}\n",
"\n",
"!mkdir -p /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}\n",
"\n",
"!cp /content/drive/MyDrive/{MODELNAME}_D_{MODELEPOCH}.pth /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/G_{MODELEPOCH}.pth\n",
"!cp /content/drive/MyDrive/{MODELNAME}_G_{MODELEPOCH}.pth /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/D_{MODELEPOCH}.pth\n",
"!cp /content/drive/MyDrive/*.index /content/\n",
"!cp /content/drive/MyDrive/*.npy /content/\n",
"!cp /content/drive/MyDrive/{MODELNAME}{MODELEPOCH}.pth /content/Retrieval-based-Voice-Conversion-WebUI/weights/{MODELNAME}.pth"
],
"metadata": {
"id": "OVQoLQJXS7WX"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"# @title 手动预处理(不推荐)\n",
"# @markdown 模型名\n",
"MODELNAME = \"lulu\" # @param {type:\"string\"}\n",
"# @markdown 采样率\n",
"BITRATE = 48000 # @param {type:\"integer\"}\n",
"# @markdown 使用的进程数\n",
"THREADCOUNT = 8 # @param {type:\"integer\"}\n",
"\n",
"!python3 trainset_preprocess_pipeline_print.py /content/dataset {BITRATE} {THREADCOUNT} logs/{MODELNAME} True"
],
"metadata": {
"id": "ZKAyuKb9J6dz"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"# @title 手动提取特征(不推荐)\n",
"# @markdown 模型名\n",
"MODELNAME = \"lulu\" # @param {type:\"string\"}\n",
"# @markdown 使用的进程数\n",
"THREADCOUNT = 8 # @param {type:\"integer\"}\n",
"# @markdown 音高提取算法\n",
"ALGO = \"harvest\" # @param {type:\"string\"}\n",
"\n",
"!python3 extract_f0_print.py logs/{MODELNAME} {THREADCOUNT} {ALGO}\n",
"\n",
"!python3 extract_feature_print.py cpu 1 0 0 logs/{MODELNAME}"
],
"metadata": {
"id": "CrxJqzAUKmPJ"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"# @title 手动训练(不推荐)\n",
"# @markdown 模型名\n",
"MODELNAME = \"lulu\" # @param {type:\"string\"}\n",
"# @markdown 使用的GPU\n",
"USEGPU = \"0\" # @param {type:\"string\"}\n",
"# @markdown 批大小\n",
"BATCHSIZE = 32 # @param {type:\"integer\"}\n",
"# @markdown 停止的epoch\n",
"MODELEPOCH = 3200 # @param {type:\"integer\"}\n",
"# @markdown 保存epoch间隔\n",
"EPOCHSAVE = 100 # @param {type:\"integer\"}\n",
"# @markdown 采样率\n",
"MODELSAMPLE = \"48k\" # @param {type:\"string\"}\n",
"# @markdown 是否缓存训练集\n",
"CACHEDATA = 1 # @param {type:\"integer\"}\n",
"# @markdown 是否仅保存最新的ckpt文件\n",
"ONLYLATEST = 0 # @param {type:\"integer\"}\n",
"\n",
"!python3 train_nsf_sim_cache_sid_load_pretrain.py -e lulu -sr {MODELSAMPLE} -f0 1 -bs {BATCHSIZE} -g {USEGPU} -te {MODELEPOCH} -se {EPOCHSAVE} -pg pretrained/f0G{MODELSAMPLE}.pth -pd pretrained/f0D{MODELSAMPLE}.pth -l {ONLYLATEST} -c {CACHEDATA}"
],
"metadata": {
"id": "IMLPLKOaKj58"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"# @title 删除其它pth,只留选中的(慎点,仔细看代码)\n",
"# @markdown 模型名\n",
"MODELNAME = \"lulu\" # @param {type:\"string\"}\n",
"# @markdown 选中模型epoch\n",
"MODELEPOCH = 9600 # @param {type:\"integer\"}\n",
"\n",
"!echo \"备份选中的模型。。。\"\n",
"!cp /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/G_{MODELEPOCH}.pth /content/{MODELNAME}_D_{MODELEPOCH}.pth\n",
"!cp /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/D_{MODELEPOCH}.pth /content/{MODELNAME}_G_{MODELEPOCH}.pth\n",
"\n",
"!echo \"正在删除。。。\"\n",
"!ls /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}\n",
"!rm /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/*.pth\n",
"\n",
"!echo \"恢复选中的模型。。。\"\n",
"!mv /content/{MODELNAME}_D_{MODELEPOCH}.pth /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/G_{MODELEPOCH}.pth\n",
"!mv /content/{MODELNAME}_G_{MODELEPOCH}.pth /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/D_{MODELEPOCH}.pth\n",
"\n",
"!echo \"删除完成\"\n",
"!ls /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}"
],
"metadata": {
"id": "haYA81hySuDl"
},
"execution_count": null,
"outputs": []
},
{
"cell_type": "code",
"source": [
"# @title 清除项目下所有文件,只留选中的模型(慎点,仔细看代码)\n",
"# @markdown 模型名\n",
"MODELNAME = \"lulu\" # @param {type:\"string\"}\n",
"# @markdown 选中模型epoch\n",
"MODELEPOCH = 9600 # @param {type:\"integer\"}\n",
"\n",
"!echo \"备份选中的模型。。。\"\n",
"!cp /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/G_{MODELEPOCH}.pth /content/{MODELNAME}_D_{MODELEPOCH}.pth\n",
"!cp /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/D_{MODELEPOCH}.pth /content/{MODELNAME}_G_{MODELEPOCH}.pth\n",
"\n",
"!echo \"正在删除。。。\"\n",
"!ls /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}\n",
"!rm -rf /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/*\n",
"\n",
"!echo \"恢复选中的模型。。。\"\n",
"!mv /content/{MODELNAME}_D_{MODELEPOCH}.pth /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/G_{MODELEPOCH}.pth\n",
"!mv /content/{MODELNAME}_G_{MODELEPOCH}.pth /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/D_{MODELEPOCH}.pth\n",
"\n",
"!echo \"删除完成\"\n",
"!ls /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}"
],
"metadata": {
"id": "QhSiPTVPoIRh"
},
"execution_count": null,
"outputs": []
}
]
}
@@ -0,0 +1,404 @@
{
"cells": [
{
"attachments": {},
"cell_type": "markdown",
"metadata": {
"id": "ZFFCx5J80SGa"
},
"source": [
"[![Open In Colab](https://colab.research.google.com/assets/colab-badge.svg)](https://colab.research.google.com/github/RVC-Project/Retrieval-based-Voice-Conversion-WebUI/blob/main/Retrieval_based_Voice_Conversion_WebUI_v2.ipynb)"
]
},
{
"cell_type": "code",
"execution_count": null,
"metadata": {
"id": "GmFP6bN9dvOq"
},
"outputs": [],
"source": [
"# @title 查看显卡\n",
"!nvidia-smi"
]
},
{
"cell_type": "code",
"execution_count": null,
"metadata": {
"id": "wjddIFr1oS3W"
},
"outputs": [],
"source": [
"# @title 安装依赖\n",
"!apt-get -y install build-essential python3-dev ffmpeg\n",
"!pip3 install --upgrade setuptools wheel\n",
"!pip3 install --upgrade pip\n",
"!pip3 install faiss-cpu==1.7.2 fairseq gradio==3.14.0 ffmpeg ffmpeg-python praat-parselmouth pyworld numpy==1.23.5 numba==0.56.4 librosa==0.9.2"
]
},
{
"cell_type": "code",
"execution_count": null,
"metadata": {
"id": "ge_97mfpgqTm"
},
"outputs": [],
"source": [
"# @title 克隆仓库\n",
"\n",
"!mkdir Retrieval-based-Voice-Conversion-WebUI\n",
"%cd /content/Retrieval-based-Voice-Conversion-WebUI\n",
"!git init\n",
"!git remote add origin https://github.com/RVC-Project/Retrieval-based-Voice-Conversion-WebUI.git\n",
"!git fetch origin cfd984812804ddc9247d65b14c82cd32e56c1133 --depth=1\n",
"!git reset --hard FETCH_HEAD"
]
},
{
"cell_type": "code",
"execution_count": null,
"metadata": {
"id": "BLDEZADkvlw1"
},
"outputs": [],
"source": [
"# @title 更新仓库(一般无需执行)\n",
"!git pull"
]
},
{
"cell_type": "code",
"execution_count": null,
"metadata": {
"id": "pqE0PrnuRqI2"
},
"outputs": [],
"source": [
"# @title 安装aria2\n",
"!apt -y install -qq aria2"
]
},
{
"cell_type": "code",
"execution_count": null,
"metadata": {
"id": "UG3XpUwEomUz"
},
"outputs": [],
"source": [
"# @title 下载底模\n",
"\n",
"# v1\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/D32k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o D32k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/D40k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o D40k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/D48k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o D48k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/G32k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o G32k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/G40k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o G40k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/G48k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o G48k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/f0D32k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o f0D32k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/f0D40k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o f0D40k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/f0D48k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o f0D48k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/f0G32k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o f0G32k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/f0G40k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o f0G40k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained/f0G48k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained -o f0G48k.pth\n",
"\n",
"# v2\n",
"# !aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained_v2/D32k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained_v2 -o D32k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained_v2/D40k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained_v2 -o D40k.pth\n",
"# !aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained_v2/D48k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained_v2 -o D48k.pth\n",
"# !aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained_v2/G32k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained_v2 -o G32k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained_v2/G40k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained_v2 -o G40k.pth\n",
"# !aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained_v2/G48k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained_v2 -o G48k.pth\n",
"# !aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained_v2/f0D32k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained_v2 -o f0D32k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained_v2/f0D40k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained_v2 -o f0D40k.pth\n",
"# !aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained_v2/f0D48k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained_v2 -o f0D48k.pth\n",
"# !aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained_v2/f0G32k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained_v2 -o f0G32k.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained_v2/f0G40k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained_v2 -o f0G40k.pth\n",
"# !aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/pretrained_v2/f0G48k.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/pretrained_v2 -o f0G48k.pth"
]
},
{
"cell_type": "code",
"execution_count": null,
"metadata": {
"id": "HugjmZqZRuiF"
},
"outputs": [],
"source": [
"# @title 下载人声分离模型\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/uvr5_weights/HP2-人声vocals+非人声instrumentals.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/uvr5_weights -o HP2-人声vocals+非人声instrumentals.pth\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/uvr5_weights/HP5-主旋律人声vocals+其他instrumentals.pth -d /content/Retrieval-based-Voice-Conversion-WebUI/uvr5_weights -o HP5-主旋律人声vocals+其他instrumentals.pth"
]
},
{
"cell_type": "code",
"execution_count": null,
"metadata": {
"id": "2RCaT9FTR0ej"
},
"outputs": [],
"source": [
"# @title 下载hubert_base\n",
"!aria2c --console-log-level=error -c -x 16 -s 16 -k 1M https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/hubert_base.pt -d /content/Retrieval-based-Voice-Conversion-WebUI -o hubert_base.pt"
]
},
{
"cell_type": "code",
"execution_count": null,
"metadata": {
"id": "jwu07JgqoFON"
},
"outputs": [],
"source": [
"# @title 挂载谷歌云盘\n",
"\n",
"from google.colab import drive\n",
"\n",
"drive.mount(\"/content/drive\")"
]
},
{
"cell_type": "code",
"execution_count": null,
"metadata": {
"id": "Mwk7Q0Loqzjx"
},
"outputs": [],
"source": [
"# @title 从谷歌云盘加载打包好的数据集到/content/dataset\n",
"\n",
"# @markdown 数据集位置\n",
"DATASET = (\n",
" \"/content/drive/MyDrive/dataset/lulu20230327_32k.zip\" # @param {type:\"string\"}\n",
")\n",
"\n",
"!mkdir -p /content/dataset\n",
"!unzip -d /content/dataset -B {DATASET}"
]
},
{
"cell_type": "code",
"execution_count": null,
"metadata": {
"id": "PDlFxWHWEynD"
},
"outputs": [],
"source": [
"# @title 重命名数据集中的重名文件\n",
"!ls -a /content/dataset/\n",
"!rename 's/(\\w+)\\.(\\w+)~(\\d*)/$1_$3.$2/' /content/dataset/*.*~*"
]
},
{
"cell_type": "code",
"execution_count": null,
"metadata": {
"id": "7vh6vphDwO0b"
},
"outputs": [],
"source": [
"# @title 启动web\n",
"%cd /content/Retrieval-based-Voice-Conversion-WebUI\n",
"# %load_ext tensorboard\n",
"# %tensorboard --logdir /content/Retrieval-based-Voice-Conversion-WebUI/logs\n",
"!python3 infer-web.py --colab --pycmd python3"
]
},
{
"cell_type": "code",
"execution_count": null,
"metadata": {
"id": "FgJuNeAwx5Y_"
},
"outputs": [],
"source": [
"# @title 手动将训练后的模型文件备份到谷歌云盘\n",
"# @markdown 需要自己查看logs文件夹下模型的文件名,手动修改下方命令末尾的文件名\n",
"\n",
"# @markdown 模型名\n",
"MODELNAME = \"lulu\" # @param {type:\"string\"}\n",
"# @markdown 模型epoch\n",
"MODELEPOCH = 9600 # @param {type:\"integer\"}\n",
"\n",
"!cp /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/G_{MODELEPOCH}.pth /content/drive/MyDrive/{MODELNAME}_D_{MODELEPOCH}.pth\n",
"!cp /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/D_{MODELEPOCH}.pth /content/drive/MyDrive/{MODELNAME}_G_{MODELEPOCH}.pth\n",
"!cp /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/added_*.index /content/drive/MyDrive/\n",
"!cp /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/total_*.npy /content/drive/MyDrive/\n",
"\n",
"!cp /content/Retrieval-based-Voice-Conversion-WebUI/weights/{MODELNAME}.pth /content/drive/MyDrive/{MODELNAME}{MODELEPOCH}.pth"
]
},
{
"cell_type": "code",
"execution_count": null,
"metadata": {
"id": "OVQoLQJXS7WX"
},
"outputs": [],
"source": [
"# @title 从谷歌云盘恢复pth\n",
"# @markdown 需要自己查看logs文件夹下模型的文件名,手动修改下方命令末尾的文件名\n",
"\n",
"# @markdown 模型名\n",
"MODELNAME = \"lulu\" # @param {type:\"string\"}\n",
"# @markdown 模型epoch\n",
"MODELEPOCH = 7500 # @param {type:\"integer\"}\n",
"\n",
"!mkdir -p /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}\n",
"\n",
"!cp /content/drive/MyDrive/{MODELNAME}_D_{MODELEPOCH}.pth /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/G_{MODELEPOCH}.pth\n",
"!cp /content/drive/MyDrive/{MODELNAME}_G_{MODELEPOCH}.pth /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/D_{MODELEPOCH}.pth\n",
"!cp /content/drive/MyDrive/*.index /content/\n",
"!cp /content/drive/MyDrive/*.npy /content/\n",
"!cp /content/drive/MyDrive/{MODELNAME}{MODELEPOCH}.pth /content/Retrieval-based-Voice-Conversion-WebUI/weights/{MODELNAME}.pth"
]
},
{
"cell_type": "code",
"execution_count": null,
"metadata": {
"id": "ZKAyuKb9J6dz"
},
"outputs": [],
"source": [
"# @title 手动预处理(不推荐)\n",
"# @markdown 模型名\n",
"MODELNAME = \"lulu\" # @param {type:\"string\"}\n",
"# @markdown 采样率\n",
"BITRATE = 48000 # @param {type:\"integer\"}\n",
"# @markdown 使用的进程数\n",
"THREADCOUNT = 8 # @param {type:\"integer\"}\n",
"\n",
"!python3 trainset_preprocess_pipeline_print.py /content/dataset {BITRATE} {THREADCOUNT} logs/{MODELNAME} True"
]
},
{
"cell_type": "code",
"execution_count": null,
"metadata": {
"id": "CrxJqzAUKmPJ"
},
"outputs": [],
"source": [
"# @title 手动提取特征(不推荐)\n",
"# @markdown 模型名\n",
"MODELNAME = \"lulu\" # @param {type:\"string\"}\n",
"# @markdown 使用的进程数\n",
"THREADCOUNT = 8 # @param {type:\"integer\"}\n",
"# @markdown 音高提取算法\n",
"ALGO = \"harvest\" # @param {type:\"string\"}\n",
"\n",
"!python3 extract_f0_print.py logs/{MODELNAME} {THREADCOUNT} {ALGO}\n",
"\n",
"!python3 extract_feature_print.py cpu 1 0 0 logs/{MODELNAME}"
]
},
{
"cell_type": "code",
"execution_count": null,
"metadata": {
"id": "IMLPLKOaKj58"
},
"outputs": [],
"source": [
"# @title 手动训练(不推荐)\n",
"# @markdown 模型名\n",
"MODELNAME = \"lulu\" # @param {type:\"string\"}\n",
"# @markdown 使用的GPU\n",
"USEGPU = \"0\" # @param {type:\"string\"}\n",
"# @markdown 批大小\n",
"BATCHSIZE = 32 # @param {type:\"integer\"}\n",
"# @markdown 停止的epoch\n",
"MODELEPOCH = 3200 # @param {type:\"integer\"}\n",
"# @markdown 保存epoch间隔\n",
"EPOCHSAVE = 100 # @param {type:\"integer\"}\n",
"# @markdown 采样率\n",
"MODELSAMPLE = \"48k\" # @param {type:\"string\"}\n",
"# @markdown 是否缓存训练集\n",
"CACHEDATA = 1 # @param {type:\"integer\"}\n",
"# @markdown 是否仅保存最新的ckpt文件\n",
"ONLYLATEST = 0 # @param {type:\"integer\"}\n",
"\n",
"!python3 train_nsf_sim_cache_sid_load_pretrain.py -e lulu -sr {MODELSAMPLE} -f0 1 -bs {BATCHSIZE} -g {USEGPU} -te {MODELEPOCH} -se {EPOCHSAVE} -pg pretrained/f0G{MODELSAMPLE}.pth -pd pretrained/f0D{MODELSAMPLE}.pth -l {ONLYLATEST} -c {CACHEDATA}"
]
},
{
"cell_type": "code",
"execution_count": null,
"metadata": {
"id": "haYA81hySuDl"
},
"outputs": [],
"source": [
"# @title 删除其它pth,只留选中的(慎点,仔细看代码)\n",
"# @markdown 模型名\n",
"MODELNAME = \"lulu\" # @param {type:\"string\"}\n",
"# @markdown 选中模型epoch\n",
"MODELEPOCH = 9600 # @param {type:\"integer\"}\n",
"\n",
"!echo \"备份选中的模型。。。\"\n",
"!cp /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/G_{MODELEPOCH}.pth /content/{MODELNAME}_D_{MODELEPOCH}.pth\n",
"!cp /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/D_{MODELEPOCH}.pth /content/{MODELNAME}_G_{MODELEPOCH}.pth\n",
"\n",
"!echo \"正在删除。。。\"\n",
"!ls /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}\n",
"!rm /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/*.pth\n",
"\n",
"!echo \"恢复选中的模型。。。\"\n",
"!mv /content/{MODELNAME}_D_{MODELEPOCH}.pth /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/G_{MODELEPOCH}.pth\n",
"!mv /content/{MODELNAME}_G_{MODELEPOCH}.pth /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/D_{MODELEPOCH}.pth\n",
"\n",
"!echo \"删除完成\"\n",
"!ls /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}"
]
},
{
"cell_type": "code",
"execution_count": null,
"metadata": {
"id": "QhSiPTVPoIRh"
},
"outputs": [],
"source": [
"# @title 清除项目下所有文件,只留选中的模型(慎点,仔细看代码)\n",
"# @markdown 模型名\n",
"MODELNAME = \"lulu\" # @param {type:\"string\"}\n",
"# @markdown 选中模型epoch\n",
"MODELEPOCH = 9600 # @param {type:\"integer\"}\n",
"\n",
"!echo \"备份选中的模型。。。\"\n",
"!cp /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/G_{MODELEPOCH}.pth /content/{MODELNAME}_D_{MODELEPOCH}.pth\n",
"!cp /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/D_{MODELEPOCH}.pth /content/{MODELNAME}_G_{MODELEPOCH}.pth\n",
"\n",
"!echo \"正在删除。。。\"\n",
"!ls /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}\n",
"!rm -rf /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/*\n",
"\n",
"!echo \"恢复选中的模型。。。\"\n",
"!mv /content/{MODELNAME}_D_{MODELEPOCH}.pth /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/G_{MODELEPOCH}.pth\n",
"!mv /content/{MODELNAME}_G_{MODELEPOCH}.pth /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}/D_{MODELEPOCH}.pth\n",
"\n",
"!echo \"删除完成\"\n",
"!ls /content/Retrieval-based-Voice-Conversion-WebUI/logs/{MODELNAME}"
]
}
],
"metadata": {
"accelerator": "GPU",
"colab": {
"private_outputs": true,
"provenance": []
},
"gpuClass": "standard",
"kernelspec": {
"display_name": "Python 3",
"name": "python3"
},
"language_info": {
"name": "python"
}
},
"nbformat": 4,
"nbformat_minor": 0
}
+319
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@@ -0,0 +1,319 @@
import os
import torch
# os.system("wget -P cvec/ https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/hubert_base.pt")
import gradio as gr
import librosa
import numpy as np
import logging
from fairseq import checkpoint_utils
from vc_infer_pipeline import VC
import traceback
from config import Config
from lib.infer_pack.models import (
SynthesizerTrnMs256NSFsid,
SynthesizerTrnMs256NSFsid_nono,
SynthesizerTrnMs768NSFsid,
SynthesizerTrnMs768NSFsid_nono,
)
from i18n import I18nAuto
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)
i18n = I18nAuto()
i18n.print()
config = Config()
weight_root = "weights"
weight_uvr5_root = "uvr5_weights"
index_root = "logs"
names = []
hubert_model = None
for name in os.listdir(weight_root):
if name.endswith(".pth"):
names.append(name)
index_paths = []
for root, dirs, files in os.walk(index_root, topdown=False):
for name in files:
if name.endswith(".index") and "trained" not in name:
index_paths.append("%s/%s" % (root, name))
def get_vc(sid):
global n_spk, tgt_sr, net_g, vc, cpt, version
if sid == "" or sid == []:
global hubert_model
if hubert_model != None: # 考虑到轮询, 需要加个判断看是否 sid 是由有模型切换到无模型的
print("clean_empty_cache")
del net_g, n_spk, vc, hubert_model, tgt_sr # ,cpt
hubert_model = net_g = n_spk = vc = hubert_model = tgt_sr = None
if torch.cuda.is_available():
torch.cuda.empty_cache()
###楼下不这么折腾清理不干净
if_f0 = cpt.get("f0", 1)
version = cpt.get("version", "v1")
if version == "v1":
if if_f0 == 1:
net_g = SynthesizerTrnMs256NSFsid(
*cpt["config"], is_half=config.is_half
)
else:
net_g = SynthesizerTrnMs256NSFsid_nono(*cpt["config"])
elif version == "v2":
if if_f0 == 1:
net_g = SynthesizerTrnMs768NSFsid(
*cpt["config"], is_half=config.is_half
)
else:
net_g = SynthesizerTrnMs768NSFsid_nono(*cpt["config"])
del net_g, cpt
if torch.cuda.is_available():
torch.cuda.empty_cache()
cpt = None
return {"visible": False, "__type__": "update"}
person = "%s/%s" % (weight_root, sid)
print("loading %s" % person)
cpt = torch.load(person, map_location="cpu")
tgt_sr = cpt["config"][-1]
cpt["config"][-3] = cpt["weight"]["emb_g.weight"].shape[0] # n_spk
if_f0 = cpt.get("f0", 1)
version = cpt.get("version", "v1")
if version == "v1":
if if_f0 == 1:
net_g = SynthesizerTrnMs256NSFsid(*cpt["config"], is_half=config.is_half)
else:
net_g = SynthesizerTrnMs256NSFsid_nono(*cpt["config"])
elif version == "v2":
if if_f0 == 1:
net_g = SynthesizerTrnMs768NSFsid(*cpt["config"], is_half=config.is_half)
else:
net_g = SynthesizerTrnMs768NSFsid_nono(*cpt["config"])
del net_g.enc_q
print(net_g.load_state_dict(cpt["weight"], strict=False))
net_g.eval().to(config.device)
if config.is_half:
net_g = net_g.half()
else:
net_g = net_g.float()
vc = VC(tgt_sr, config)
n_spk = cpt["config"][-3]
return {"visible": True, "maximum": n_spk, "__type__": "update"}
def load_hubert():
global hubert_model
models, _, _ = checkpoint_utils.load_model_ensemble_and_task(
["hubert_base.pt"],
suffix="",
)
hubert_model = models[0]
hubert_model = hubert_model.to(config.device)
if config.is_half:
hubert_model = hubert_model.half()
else:
hubert_model = hubert_model.float()
hubert_model.eval()
def vc_single(
sid,
input_audio_path,
f0_up_key,
f0_file,
f0_method,
file_index,
file_index2,
# file_big_npy,
index_rate,
filter_radius,
resample_sr,
rms_mix_rate,
protect,
): # spk_item, input_audio0, vc_transform0,f0_file,f0method0
global tgt_sr, net_g, vc, hubert_model, version
if input_audio_path is None:
return "You need to upload an audio", None
f0_up_key = int(f0_up_key)
try:
audio = input_audio_path[1] / 32768.0
if len(audio.shape) == 2:
audio = np.mean(audio, -1)
audio = librosa.resample(audio, orig_sr=input_audio_path[0], target_sr=16000)
audio_max = np.abs(audio).max() / 0.95
if audio_max > 1:
audio /= audio_max
times = [0, 0, 0]
if hubert_model == None:
load_hubert()
if_f0 = cpt.get("f0", 1)
file_index = (
(
file_index.strip(" ")
.strip('"')
.strip("\n")
.strip('"')
.strip(" ")
.replace("trained", "added")
)
if file_index != ""
else file_index2
) # 防止小白写错,自动帮他替换掉
# file_big_npy = (
# file_big_npy.strip(" ").strip('"').strip("\n").strip('"').strip(" ")
# )
audio_opt = vc.pipeline(
hubert_model,
net_g,
sid,
audio,
input_audio_path,
times,
f0_up_key,
f0_method,
file_index,
# file_big_npy,
index_rate,
if_f0,
filter_radius,
tgt_sr,
resample_sr,
rms_mix_rate,
version,
protect,
f0_file=f0_file,
)
if resample_sr >= 16000 and tgt_sr != resample_sr:
tgt_sr = resample_sr
index_info = (
"Using index:%s." % file_index
if os.path.exists(file_index)
else "Index not used."
)
return "Success.\n %s\nTime:\n npy:%ss, f0:%ss, infer:%ss" % (
index_info,
times[0],
times[1],
times[2],
), (tgt_sr, audio_opt)
except:
info = traceback.format_exc()
print(info)
return info, (None, None)
app = gr.Blocks()
with app:
with gr.Tabs():
with gr.TabItem("在线demo"):
gr.Markdown(
value="""
RVC 在线demo
"""
)
sid = gr.Dropdown(label=i18n("推理音色"), choices=sorted(names))
with gr.Column():
spk_item = gr.Slider(
minimum=0,
maximum=2333,
step=1,
label=i18n("请选择说话人id"),
value=0,
visible=False,
interactive=True,
)
sid.change(
fn=get_vc,
inputs=[sid],
outputs=[spk_item],
)
gr.Markdown(
value=i18n("男转女推荐+12key, 女转男推荐-12key, 如果音域爆炸导致音色失真也可以自己调整到合适音域. ")
)
vc_input3 = gr.Audio(label="上传音频(长度小于90秒)")
vc_transform0 = gr.Number(label=i18n("变调(整数, 半音数量, 升八度12降八度-12)"), value=0)
f0method0 = gr.Radio(
label=i18n("选择音高提取算法,输入歌声可用pm提速,harvest低音好但巨慢无比,crepe效果好但吃GPU"),
choices=["pm", "harvest", "crepe", "rmvpe"],
value="pm",
interactive=True,
)
filter_radius0 = gr.Slider(
minimum=0,
maximum=7,
label=i18n(">=3则使用对harvest音高识别的结果使用中值滤波,数值为滤波半径,使用可以削弱哑音"),
value=3,
step=1,
interactive=True,
)
with gr.Column():
file_index1 = gr.Textbox(
label=i18n("特征检索库文件路径,为空则使用下拉的选择结果"),
value="",
interactive=False,
visible=False,
)
file_index2 = gr.Dropdown(
label=i18n("自动检测index路径,下拉式选择(dropdown)"),
choices=sorted(index_paths),
interactive=True,
)
index_rate1 = gr.Slider(
minimum=0,
maximum=1,
label=i18n("检索特征占比"),
value=0.88,
interactive=True,
)
resample_sr0 = gr.Slider(
minimum=0,
maximum=48000,
label=i18n("后处理重采样至最终采样率,0为不进行重采样"),
value=0,
step=1,
interactive=True,
)
rms_mix_rate0 = gr.Slider(
minimum=0,
maximum=1,
label=i18n("输入源音量包络替换输出音量包络融合比例,越靠近1越使用输出包络"),
value=1,
interactive=True,
)
protect0 = gr.Slider(
minimum=0,
maximum=0.5,
label=i18n("保护清辅音和呼吸声,防止电音撕裂等artifact,拉满0.5不开启,调低加大保护力度但可能降低索引效果"),
value=0.33,
step=0.01,
interactive=True,
)
f0_file = gr.File(label=i18n("F0曲线文件, 可选, 一行一个音高, 代替默认F0及升降调"))
but0 = gr.Button(i18n("转换"), variant="primary")
vc_output1 = gr.Textbox(label=i18n("输出信息"))
vc_output2 = gr.Audio(label=i18n("输出音频(右下角三个点,点了可以下载)"))
but0.click(
vc_single,
[
spk_item,
vc_input3,
vc_transform0,
f0_file,
f0method0,
file_index1,
file_index2,
# file_big_npy1,
index_rate1,
filter_radius0,
resample_sr0,
rms_mix_rate0,
protect0,
],
[vc_output1, vc_output2],
)
app.launch()
+175
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@@ -0,0 +1,175 @@
import os
import argparse
import sys
import torch
from multiprocessing import cpu_count
def use_fp32_config():
for config_file in [
"32k.json",
"40k.json",
"48k.json",
"48k_v2.json",
"32k_v2.json",
]:
with open(f"configs/{config_file}", "r") as f:
strr = f.read().replace("true", "false")
with open(f"configs/{config_file}", "w") as f:
f.write(strr)
with open("trainset_preprocess_pipeline_print.py", "r") as f:
strr = f.read().replace("3.7", "3.0")
with open("trainset_preprocess_pipeline_print.py", "w") as f:
f.write(strr)
class Config:
def __init__(self):
self.device = "cuda:0"
self.is_half = True
self.n_cpu = 0
self.gpu_name = None
self.gpu_mem = None
(
self.python_cmd,
self.listen_port,
self.iscolab,
self.noparallel,
self.noautoopen,
self.dml
) = self.arg_parse()
self.instead = ""
self.x_pad, self.x_query, self.x_center, self.x_max = self.device_config()
@staticmethod
def arg_parse() -> tuple:
exe = sys.executable or "python"
parser = argparse.ArgumentParser()
parser.add_argument("--port", type=int, default=7865, help="Listen port")
parser.add_argument("--pycmd", type=str, default=exe, help="Python command")
parser.add_argument("--colab", action="store_true", help="Launch in colab")
parser.add_argument(
"--noparallel", action="store_true", help="Disable parallel processing"
)
parser.add_argument(
"--noautoopen",
action="store_true",
help="Do not open in browser automatically",
)
parser.add_argument(
"--dml",
action="store_true",
help="torch_dml",
)
cmd_opts = parser.parse_args()
cmd_opts.port = cmd_opts.port if 0 <= cmd_opts.port <= 65535 else 7865
return (
cmd_opts.pycmd,
cmd_opts.port,
cmd_opts.colab,
cmd_opts.noparallel,
cmd_opts.noautoopen,
cmd_opts.dml
)
# has_mps is only available in nightly pytorch (for now) and MasOS 12.3+.
# check `getattr` and try it for compatibility
@staticmethod
def has_mps() -> bool:
if not torch.backends.mps.is_available():
return False
try:
torch.zeros(1).to(torch.device("mps"))
return True
except Exception:
return False
def device_config(self) -> tuple:
if torch.cuda.is_available():
i_device = int(self.device.split(":")[-1])
self.gpu_name = torch.cuda.get_device_name(i_device)
if (
("16" in self.gpu_name and "V100" not in self.gpu_name.upper())
or "P40" in self.gpu_name.upper()
or "1060" in self.gpu_name
or "1070" in self.gpu_name
or "1080" in self.gpu_name
):
print("Found GPU", self.gpu_name, ", force to fp32")
self.is_half = False
use_fp32_config()
else:
print("Found GPU", self.gpu_name)
self.gpu_mem = int(
torch.cuda.get_device_properties(i_device).total_memory
/ 1024
/ 1024
/ 1024
+ 0.4
)
if self.gpu_mem <= 4:
with open("trainset_preprocess_pipeline_print.py", "r") as f:
strr = f.read().replace("3.7", "3.0")
with open("trainset_preprocess_pipeline_print.py", "w") as f:
f.write(strr)
elif self.has_mps():
print("No supported Nvidia GPU found")
self.device = self.instead = "mps"
self.is_half = False
use_fp32_config()
else:
print("No supported Nvidia GPU found")
self.device = self.instead = "cpu"
self.is_half = False
use_fp32_config()
if self.n_cpu == 0:
self.n_cpu = cpu_count()
if self.is_half:
# 6G显存配置
x_pad = 3
x_query = 10
x_center = 60
x_max = 65
else:
# 5G显存配置
x_pad = 1
x_query = 6
x_center = 38
x_max = 41
if self.gpu_mem is not None and self.gpu_mem <= 4:
x_pad = 1
x_query = 5
x_center = 30
x_max = 32
if self.dml:
print("use DirectML instead")
if(os.path.exists("runtime\Lib\site-packages\onnxruntime\capi\DirectML.dll")==False):
try:
os.rename("runtime\Lib\site-packages\onnxruntime", "runtime\Lib\site-packages\onnxruntime-cuda")
except:
pass
try:
os.rename("runtime\Lib\site-packages\onnxruntime-dml", "runtime\Lib\site-packages\onnxruntime")
except:
pass
import torch_directml
self.device = torch_directml.device(torch_directml.default_device())
self.is_half = False
else:
if self.instead:
print(f"use {self.instead} instead")
if(os.path.exists("runtime\Lib\site-packages\onnxruntime\capi\onnxruntime_providers_cuda.dll")==False):
try:
os.rename("runtime\Lib\site-packages\onnxruntime", "runtime\Lib\site-packages\onnxruntime-dml")
except:
pass
try:
os.rename("runtime\Lib\site-packages\onnxruntime-cuda", "runtime\Lib\site-packages\onnxruntime")
except:
pass
return x_pad, x_query, x_center, x_max
@@ -0,0 +1,186 @@
name: pydml
channels:
- pytorch
- https://mirrors.tuna.tsinghua.edu.cn/anaconda/pkgs/main
- https://mirrors.tuna.tsinghua.edu.cn/anaconda/pkgs/main/
- https://mirrors.tuna.tsinghua.edu.cn/anaconda/pkgs/free/
- https://mirrors.tuna.tsinghua.edu.cn/anaconda/cloud/conda-forge/
- defaults
- https://mirrors.tuna.tsinghua.edu.cn/anaconda/cloud/fastai/
- https://mirrors.tuna.tsinghua.edu.cn/anaconda/cloud/pytorch/
- https://mirrors.tuna.tsinghua.edu.cn/anaconda/cloud/bioconda/
dependencies:
- abseil-cpp=20211102.0=hd77b12b_0
- absl-py=1.3.0=py310haa95532_0
- aiohttp=3.8.3=py310h2bbff1b_0
- aiosignal=1.2.0=pyhd3eb1b0_0
- async-timeout=4.0.2=py310haa95532_0
- attrs=22.1.0=py310haa95532_0
- blas=1.0=mkl
- blinker=1.4=py310haa95532_0
- bottleneck=1.3.5=py310h9128911_0
- brotli=1.0.9=h2bbff1b_7
- brotli-bin=1.0.9=h2bbff1b_7
- brotlipy=0.7.0=py310h2bbff1b_1002
- bzip2=1.0.8=he774522_0
- c-ares=1.19.0=h2bbff1b_0
- ca-certificates=2023.05.30=haa95532_0
- cachetools=4.2.2=pyhd3eb1b0_0
- certifi=2023.5.7=py310haa95532_0
- cffi=1.15.1=py310h2bbff1b_3
- charset-normalizer=2.0.4=pyhd3eb1b0_0
- click=8.0.4=py310haa95532_0
- colorama=0.4.6=py310haa95532_0
- contourpy=1.0.5=py310h59b6b97_0
- cryptography=39.0.1=py310h21b164f_0
- cycler=0.11.0=pyhd3eb1b0_0
- fonttools=4.25.0=pyhd3eb1b0_0
- freetype=2.12.1=ha860e81_0
- frozenlist=1.3.3=py310h2bbff1b_0
- giflib=5.2.1=h8cc25b3_3
- glib=2.69.1=h5dc1a3c_2
- google-auth=2.6.0=pyhd3eb1b0_0
- google-auth-oauthlib=0.4.4=pyhd3eb1b0_0
- grpc-cpp=1.48.2=hf108199_0
- grpcio=1.48.2=py310hf108199_0
- gst-plugins-base=1.18.5=h9e645db_0
- gstreamer=1.18.5=hd78058f_0
- icu=58.2=ha925a31_3
- idna=3.4=py310haa95532_0
- intel-openmp=2023.1.0=h59b6b97_46319
- jpeg=9e=h2bbff1b_1
- kiwisolver=1.4.4=py310hd77b12b_0
- krb5=1.19.4=h5b6d351_0
- lerc=3.0=hd77b12b_0
- libbrotlicommon=1.0.9=h2bbff1b_7
- libbrotlidec=1.0.9=h2bbff1b_7
- libbrotlienc=1.0.9=h2bbff1b_7
- libclang=14.0.6=default_hb5a9fac_1
- libclang13=14.0.6=default_h8e68704_1
- libdeflate=1.17=h2bbff1b_0
- libffi=3.4.4=hd77b12b_0
- libiconv=1.16=h2bbff1b_2
- libogg=1.3.5=h2bbff1b_1
- libpng=1.6.39=h8cc25b3_0
- libprotobuf=3.20.3=h23ce68f_0
- libtiff=4.5.0=h6c2663c_2
- libuv=1.44.2=h2bbff1b_0
- libvorbis=1.3.7=he774522_0
- libwebp=1.2.4=hbc33d0d_1
- libwebp-base=1.2.4=h2bbff1b_1
- libxml2=2.10.3=h0ad7f3c_0
- libxslt=1.1.37=h2bbff1b_0
- lz4-c=1.9.4=h2bbff1b_0
- markdown=3.4.1=py310haa95532_0
- markupsafe=2.1.1=py310h2bbff1b_0
- matplotlib=3.7.1=py310haa95532_1
- matplotlib-base=3.7.1=py310h4ed8f06_1
- mkl=2023.1.0=h8bd8f75_46356
- mkl-service=2.4.0=py310h2bbff1b_1
- mkl_fft=1.3.6=py310h4ed8f06_1
- mkl_random=1.2.2=py310h4ed8f06_1
- multidict=6.0.2=py310h2bbff1b_0
- munkres=1.1.4=py_0
- numexpr=2.8.4=py310h2cd9be0_1
- numpy=1.24.3=py310h055cbcc_1
- numpy-base=1.24.3=py310h65a83cf_1
- oauthlib=3.2.2=py310haa95532_0
- openssl=1.1.1t=h2bbff1b_0
- packaging=23.0=py310haa95532_0
- pandas=1.5.3=py310h4ed8f06_0
- pcre=8.45=hd77b12b_0
- pillow=9.4.0=py310hd77b12b_0
- pip=23.0.1=py310haa95532_0
- ply=3.11=py310haa95532_0
- protobuf=3.20.3=py310hd77b12b_0
- pyasn1=0.4.8=pyhd3eb1b0_0
- pyasn1-modules=0.2.8=py_0
- pycparser=2.21=pyhd3eb1b0_0
- pyjwt=2.4.0=py310haa95532_0
- pyopenssl=23.0.0=py310haa95532_0
- pyparsing=3.0.9=py310haa95532_0
- pyqt=5.15.7=py310hd77b12b_0
- pyqt5-sip=12.11.0=py310hd77b12b_0
- pysocks=1.7.1=py310haa95532_0
- python=3.10.11=h966fe2a_2
- python-dateutil=2.8.2=pyhd3eb1b0_0
- pytorch-mutex=1.0=cpu
- pytz=2022.7=py310haa95532_0
- pyyaml=6.0=py310h2bbff1b_1
- qt-main=5.15.2=he8e5bd7_8
- qt-webengine=5.15.9=hb9a9bb5_5
- qtwebkit=5.212=h2bbfb41_5
- re2=2022.04.01=hd77b12b_0
- requests=2.29.0=py310haa95532_0
- requests-oauthlib=1.3.0=py_0
- rsa=4.7.2=pyhd3eb1b0_1
- setuptools=67.8.0=py310haa95532_0
- sip=6.6.2=py310hd77b12b_0
- six=1.16.0=pyhd3eb1b0_1
- sqlite=3.41.2=h2bbff1b_0
- tbb=2021.8.0=h59b6b97_0
- tensorboard=2.10.0=py310haa95532_0
- tensorboard-data-server=0.6.1=py310haa95532_0
- tensorboard-plugin-wit=1.8.1=py310haa95532_0
- tk=8.6.12=h2bbff1b_0
- toml=0.10.2=pyhd3eb1b0_0
- tornado=6.2=py310h2bbff1b_0
- tqdm=4.65.0=py310h9909e9c_0
- typing_extensions=4.5.0=py310haa95532_0
- tzdata=2023c=h04d1e81_0
- urllib3=1.26.16=py310haa95532_0
- vc=14.2=h21ff451_1
- vs2015_runtime=14.27.29016=h5e58377_2
- werkzeug=2.2.3=py310haa95532_0
- wheel=0.38.4=py310haa95532_0
- win_inet_pton=1.1.0=py310haa95532_0
- xz=5.4.2=h8cc25b3_0
- yaml=0.2.5=he774522_0
- yarl=1.8.1=py310h2bbff1b_0
- zlib=1.2.13=h8cc25b3_0
- zstd=1.5.5=hd43e919_0
- pip:
- antlr4-python3-runtime==4.8
- appdirs==1.4.4
- audioread==3.0.0
- bitarray==2.7.4
- cython==0.29.35
- decorator==5.1.1
- fairseq==0.12.2
- faiss-cpu==1.7.4
- filelock==3.12.0
- hydra-core==1.0.7
- jinja2==3.1.2
- joblib==1.2.0
- lazy-loader==0.2
- librosa==0.10.0.post2
- llvmlite==0.40.0
- lxml==4.9.2
- mpmath==1.3.0
- msgpack==1.0.5
- networkx==3.1
- noisereduce==2.0.1
- numba==0.57.0
- omegaconf==2.0.6
- opencv-python==4.7.0.72
- pooch==1.6.0
- portalocker==2.7.0
- pysimplegui==4.60.5
- pywin32==306
- pyworld==0.3.3
- regex==2023.5.5
- sacrebleu==2.3.1
- scikit-learn==1.2.2
- scipy==1.10.1
- sounddevice==0.4.6
- soundfile==0.12.1
- soxr==0.3.5
- sympy==1.12
- tabulate==0.9.0
- threadpoolctl==3.1.0
- torch==2.0.0
- torch-directml==0.2.0.dev230426
- torchaudio==2.0.1
- torchvision==0.15.1
- wget==3.2
prefix: D:\ProgramData\anaconda3_\envs\pydml
@@ -0,0 +1,166 @@
import os, traceback, sys, parselmouth
now_dir = os.getcwd()
sys.path.append(now_dir)
from lib.audio import load_audio
import pyworld
import numpy as np, logging
logging.getLogger("numba").setLevel(logging.WARNING)
from multiprocessing import Process
exp_dir = sys.argv[1]
f = open("%s/extract_f0_feature.log" % exp_dir, "a+")
def printt(strr):
print(strr)
f.write("%s\n" % strr)
f.flush()
n_p = int(sys.argv[2])
f0method = sys.argv[3]
class FeatureInput(object):
def __init__(self, samplerate=16000, hop_size=160):
self.fs = samplerate
self.hop = hop_size
self.f0_bin = 256
self.f0_max = 1100.0
self.f0_min = 50.0
self.f0_mel_min = 1127 * np.log(1 + self.f0_min / 700)
self.f0_mel_max = 1127 * np.log(1 + self.f0_max / 700)
def compute_f0(self, path, f0_method):
x = load_audio(path, self.fs)
p_len = x.shape[0] // self.hop
if f0_method == "pm":
time_step = 160 / 16000 * 1000
f0_min = 50
f0_max = 1100
f0 = (
parselmouth.Sound(x, self.fs)
.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"
)
elif f0_method == "harvest":
f0, t = pyworld.harvest(
x.astype(np.double),
fs=self.fs,
f0_ceil=self.f0_max,
f0_floor=self.f0_min,
frame_period=1000 * self.hop / self.fs,
)
f0 = pyworld.stonemask(x.astype(np.double), f0, t, self.fs)
elif f0_method == "dio":
f0, t = pyworld.dio(
x.astype(np.double),
fs=self.fs,
f0_ceil=self.f0_max,
f0_floor=self.f0_min,
frame_period=1000 * self.hop / self.fs,
)
f0 = pyworld.stonemask(x.astype(np.double), f0, t, self.fs)
elif f0_method == "rmvpe":
if hasattr(self, "model_rmvpe") == False:
from lib.rmvpe import RMVPE
print("loading rmvpe model")
self.model_rmvpe = RMVPE("rmvpe.pt", is_half=False, device="cpu")
f0 = self.model_rmvpe.infer_from_audio(x, thred=0.03)
return f0
def coarse_f0(self, f0):
f0_mel = 1127 * np.log(1 + f0 / 700)
f0_mel[f0_mel > 0] = (f0_mel[f0_mel > 0] - self.f0_mel_min) * (
self.f0_bin - 2
) / (self.f0_mel_max - self.f0_mel_min) + 1
# use 0 or 1
f0_mel[f0_mel <= 1] = 1
f0_mel[f0_mel > self.f0_bin - 1] = self.f0_bin - 1
f0_coarse = np.rint(f0_mel).astype(int)
assert f0_coarse.max() <= 255 and f0_coarse.min() >= 1, (
f0_coarse.max(),
f0_coarse.min(),
)
return f0_coarse
def go(self, paths, f0_method):
if len(paths) == 0:
printt("no-f0-todo")
else:
printt("todo-f0-%s" % len(paths))
n = max(len(paths) // 5, 1) # 每个进程最多打印5条
for idx, (inp_path, opt_path1, opt_path2) in enumerate(paths):
try:
if idx % n == 0:
printt("f0ing,now-%s,all-%s,-%s" % (idx, len(paths), inp_path))
if (
os.path.exists(opt_path1 + ".npy") == True
and os.path.exists(opt_path2 + ".npy") == True
):
continue
featur_pit = self.compute_f0(inp_path, f0_method)
np.save(
opt_path2,
featur_pit,
allow_pickle=False,
) # nsf
coarse_pit = self.coarse_f0(featur_pit)
np.save(
opt_path1,
coarse_pit,
allow_pickle=False,
) # ori
except:
printt("f0fail-%s-%s-%s" % (idx, inp_path, traceback.format_exc()))
if __name__ == "__main__":
# exp_dir=r"E:\codes\py39\dataset\mi-test"
# n_p=16
# f = open("%s/log_extract_f0.log"%exp_dir, "w")
printt(sys.argv)
featureInput = FeatureInput()
paths = []
inp_root = "%s/1_16k_wavs" % (exp_dir)
opt_root1 = "%s/2a_f0" % (exp_dir)
opt_root2 = "%s/2b-f0nsf" % (exp_dir)
os.makedirs(opt_root1, exist_ok=True)
os.makedirs(opt_root2, exist_ok=True)
for name in sorted(list(os.listdir(inp_root))):
inp_path = "%s/%s" % (inp_root, name)
if "spec" in inp_path:
continue
opt_path1 = "%s/%s" % (opt_root1, name)
opt_path2 = "%s/%s" % (opt_root2, name)
paths.append([inp_path, opt_path1, opt_path2])
ps = []
for i in range(n_p):
p = Process(
target=featureInput.go,
args=(
paths[i::n_p],
f0method,
),
)
ps.append(p)
p.start()
for i in range(n_p):
ps[i].join()
@@ -0,0 +1,132 @@
import os, traceback, sys, parselmouth
now_dir = os.getcwd()
sys.path.append(now_dir)
from lib.audio import load_audio
import pyworld
import numpy as np, logging
logging.getLogger("numba").setLevel(logging.WARNING)
n_part = int(sys.argv[1])
i_part = int(sys.argv[2])
i_gpu = sys.argv[3]
os.environ["CUDA_VISIBLE_DEVICES"] = str(i_gpu)
exp_dir = sys.argv[4]
is_half = sys.argv[5]
f = open("%s/extract_f0_feature.log" % exp_dir, "a+")
def printt(strr):
print(strr)
f.write("%s\n" % strr)
f.flush()
class FeatureInput(object):
def __init__(self, samplerate=16000, hop_size=160):
self.fs = samplerate
self.hop = hop_size
self.f0_bin = 256
self.f0_max = 1100.0
self.f0_min = 50.0
self.f0_mel_min = 1127 * np.log(1 + self.f0_min / 700)
self.f0_mel_max = 1127 * np.log(1 + self.f0_max / 700)
def compute_f0(self, path, f0_method):
x = load_audio(path, self.fs)
# p_len = x.shape[0] // self.hop
if f0_method == "rmvpe":
if hasattr(self, "model_rmvpe") == False:
from lib.rmvpe import RMVPE
print("loading rmvpe model")
self.model_rmvpe = RMVPE("rmvpe.pt", is_half=is_half, device="cuda")
f0 = self.model_rmvpe.infer_from_audio(x, thred=0.03)
return f0
def coarse_f0(self, f0):
f0_mel = 1127 * np.log(1 + f0 / 700)
f0_mel[f0_mel > 0] = (f0_mel[f0_mel > 0] - self.f0_mel_min) * (
self.f0_bin - 2
) / (self.f0_mel_max - self.f0_mel_min) + 1
# use 0 or 1
f0_mel[f0_mel <= 1] = 1
f0_mel[f0_mel > self.f0_bin - 1] = self.f0_bin - 1
f0_coarse = np.rint(f0_mel).astype(int)
assert f0_coarse.max() <= 255 and f0_coarse.min() >= 1, (
f0_coarse.max(),
f0_coarse.min(),
)
return f0_coarse
def go(self, paths, f0_method):
if len(paths) == 0:
printt("no-f0-todo")
else:
printt("todo-f0-%s" % len(paths))
n = max(len(paths) // 5, 1) # 每个进程最多打印5条
for idx, (inp_path, opt_path1, opt_path2) in enumerate(paths):
try:
if idx % n == 0:
printt("f0ing,now-%s,all-%s,-%s" % (idx, len(paths), inp_path))
if (
os.path.exists(opt_path1 + ".npy") == True
and os.path.exists(opt_path2 + ".npy") == True
):
continue
featur_pit = self.compute_f0(inp_path, f0_method)
np.save(
opt_path2,
featur_pit,
allow_pickle=False,
) # nsf
coarse_pit = self.coarse_f0(featur_pit)
np.save(
opt_path1,
coarse_pit,
allow_pickle=False,
) # ori
except:
printt("f0fail-%s-%s-%s" % (idx, inp_path, traceback.format_exc()))
if __name__ == "__main__":
# exp_dir=r"E:\codes\py39\dataset\mi-test"
# n_p=16
# f = open("%s/log_extract_f0.log"%exp_dir, "w")
printt(sys.argv)
featureInput = FeatureInput()
paths = []
inp_root = "%s/1_16k_wavs" % (exp_dir)
opt_root1 = "%s/2a_f0" % (exp_dir)
opt_root2 = "%s/2b-f0nsf" % (exp_dir)
os.makedirs(opt_root1, exist_ok=True)
os.makedirs(opt_root2, exist_ok=True)
for name in sorted(list(os.listdir(inp_root))):
inp_path = "%s/%s" % (inp_root, name)
if "spec" in inp_path:
continue
opt_path1 = "%s/%s" % (opt_root1, name)
opt_path2 = "%s/%s" % (opt_root2, name)
paths.append([inp_path, opt_path1, opt_path2])
try:
featureInput.go(paths[i_part::n_part], "rmvpe")
except:
printt("f0_all_fail-%s" % (traceback.format_exc()))
# ps = []
# for i in range(n_p):
# p = Process(
# target=featureInput.go,
# args=(
# paths[i::n_p],
# f0method,
# ),
# )
# ps.append(p)
# p.start()
# for i in range(n_p):
# ps[i].join()
@@ -0,0 +1,129 @@
import os, traceback, sys, parselmouth
now_dir = os.getcwd()
sys.path.append(now_dir)
from lib.audio import load_audio
import pyworld
import numpy as np, logging
logging.getLogger("numba").setLevel(logging.WARNING)
exp_dir = sys.argv[1]
import torch_directml
device = torch_directml.device(torch_directml.default_device())
f = open("%s/extract_f0_feature.log" % exp_dir, "a+")
def printt(strr):
print(strr)
f.write("%s\n" % strr)
f.flush()
class FeatureInput(object):
def __init__(self, samplerate=16000, hop_size=160):
self.fs = samplerate
self.hop = hop_size
self.f0_bin = 256
self.f0_max = 1100.0
self.f0_min = 50.0
self.f0_mel_min = 1127 * np.log(1 + self.f0_min / 700)
self.f0_mel_max = 1127 * np.log(1 + self.f0_max / 700)
def compute_f0(self, path, f0_method):
x = load_audio(path, self.fs)
# p_len = x.shape[0] // self.hop
if f0_method == "rmvpe":
if hasattr(self, "model_rmvpe") == False:
from lib.rmvpe import RMVPE
print("loading rmvpe model")
self.model_rmvpe = RMVPE("rmvpe.pt", is_half=False, device=device)
f0 = self.model_rmvpe.infer_from_audio(x, thred=0.03)
return f0
def coarse_f0(self, f0):
f0_mel = 1127 * np.log(1 + f0 / 700)
f0_mel[f0_mel > 0] = (f0_mel[f0_mel > 0] - self.f0_mel_min) * (
self.f0_bin - 2
) / (self.f0_mel_max - self.f0_mel_min) + 1
# use 0 or 1
f0_mel[f0_mel <= 1] = 1
f0_mel[f0_mel > self.f0_bin - 1] = self.f0_bin - 1
f0_coarse = np.rint(f0_mel).astype(int)
assert f0_coarse.max() <= 255 and f0_coarse.min() >= 1, (
f0_coarse.max(),
f0_coarse.min(),
)
return f0_coarse
def go(self, paths, f0_method):
if len(paths) == 0:
printt("no-f0-todo")
else:
printt("todo-f0-%s" % len(paths))
n = max(len(paths) // 5, 1) # 每个进程最多打印5条
for idx, (inp_path, opt_path1, opt_path2) in enumerate(paths):
try:
if idx % n == 0:
printt("f0ing,now-%s,all-%s,-%s" % (idx, len(paths), inp_path))
if (
os.path.exists(opt_path1 + ".npy") == True
and os.path.exists(opt_path2 + ".npy") == True
):
continue
featur_pit = self.compute_f0(inp_path, f0_method)
np.save(
opt_path2,
featur_pit,
allow_pickle=False,
) # nsf
coarse_pit = self.coarse_f0(featur_pit)
np.save(
opt_path1,
coarse_pit,
allow_pickle=False,
) # ori
except:
printt("f0fail-%s-%s-%s" % (idx, inp_path, traceback.format_exc()))
if __name__ == "__main__":
# exp_dir=r"E:\codes\py39\dataset\mi-test"
# n_p=16
# f = open("%s/log_extract_f0.log"%exp_dir, "w")
printt(sys.argv)
featureInput = FeatureInput()
paths = []
inp_root = "%s/1_16k_wavs" % (exp_dir)
opt_root1 = "%s/2a_f0" % (exp_dir)
opt_root2 = "%s/2b-f0nsf" % (exp_dir)
os.makedirs(opt_root1, exist_ok=True)
os.makedirs(opt_root2, exist_ok=True)
for name in sorted(list(os.listdir(inp_root))):
inp_path = "%s/%s" % (inp_root, name)
if "spec" in inp_path:
continue
opt_path1 = "%s/%s" % (opt_root1, name)
opt_path2 = "%s/%s" % (opt_root2, name)
paths.append([inp_path, opt_path1, opt_path2])
try:
featureInput.go(paths, "rmvpe")
except:
printt("f0_all_fail-%s" % (traceback.format_exc()))
# ps = []
# for i in range(n_p):
# p = Process(
# target=featureInput.go,
# args=(
# paths[i::n_p],
# f0method,
# ),
# )
# ps.append(p)
# p.start()
# for i in range(n_p):
# ps[i].join()
@@ -0,0 +1,132 @@
import os, sys, traceback
os.environ["PYTORCH_ENABLE_MPS_FALLBACK"] = "1"
os.environ["PYTORCH_MPS_HIGH_WATERMARK_RATIO"] = "0.0"
device=sys.argv[1]
n_part = int(sys.argv[2])
i_part = int(sys.argv[3])
if len(sys.argv) == 6:
exp_dir = sys.argv[4]
version = sys.argv[5]
else:
i_gpu = sys.argv[4]
exp_dir = sys.argv[5]
os.environ["CUDA_VISIBLE_DEVICES"] = str(i_gpu)
version = sys.argv[6]
import torch
import torch.nn.functional as F
import soundfile as sf
import numpy as np
import fairseq
if("privateuseone"not in device):
device = "cpu"
if torch.cuda.is_available():
device = "cuda"
elif torch.backends.mps.is_available():
device = "mps"
else:
import torch_directml
device = torch_directml.device(torch_directml.default_device())
def forward_dml(ctx, x, scale):
ctx.scale = scale
res = x.clone().detach()
return res
fairseq.modules.grad_multiply.GradMultiply.forward=forward_dml
f = open("%s/extract_f0_feature.log" % exp_dir, "a+")
def printt(strr):
print(strr)
f.write("%s\n" % strr)
f.flush()
printt(sys.argv)
model_path = "hubert_base.pt"
printt(exp_dir)
wavPath = "%s/1_16k_wavs" % exp_dir
outPath = (
"%s/3_feature256" % exp_dir if version == "v1" else "%s/3_feature768" % exp_dir
)
os.makedirs(outPath, exist_ok=True)
# wave must be 16k, hop_size=320
def readwave(wav_path, normalize=False):
wav, sr = sf.read(wav_path)
assert sr == 16000
feats = torch.from_numpy(wav).float()
if feats.dim() == 2: # double channels
feats = feats.mean(-1)
assert feats.dim() == 1, feats.dim()
if normalize:
with torch.no_grad():
feats = F.layer_norm(feats, feats.shape)
feats = feats.view(1, -1)
return feats
# HuBERT model
printt("load model(s) from {}".format(model_path))
# if hubert model is exist
if os.access(model_path, os.F_OK) == False:
printt(
"Error: Extracting is shut down because %s does not exist, you may download it from https://huggingface.co/lj1995/VoiceConversionWebUI/tree/main"
% model_path
)
exit(0)
models, saved_cfg, task = fairseq.checkpoint_utils.load_model_ensemble_and_task(
[model_path],
suffix="",
)
model = models[0]
model = model.to(device)
printt("move model to %s" % device)
if device not in ["mps", "cpu"]:
model = model.half()
model.eval()
todo = sorted(list(os.listdir(wavPath)))[i_part::n_part]
n = max(1, len(todo) // 10) # 最多打印十条
if len(todo) == 0:
printt("no-feature-todo")
else:
printt("all-feature-%s" % len(todo))
for idx, file in enumerate(todo):
try:
if file.endswith(".wav"):
wav_path = "%s/%s" % (wavPath, file)
out_path = "%s/%s" % (outPath, file.replace("wav", "npy"))
if os.path.exists(out_path):
continue
feats = readwave(wav_path, normalize=saved_cfg.task.normalize)
padding_mask = torch.BoolTensor(feats.shape).fill_(False)
inputs = {
"source": feats.half().to(device)
if device not in ["mps", "cpu"]
else feats.to(device),
"padding_mask": padding_mask.to(device),
"output_layer": 9 if version == "v1" else 12, # layer 9
}
with torch.no_grad():
logits = model.extract_features(**inputs)
feats = (
model.final_proj(logits[0]) if version == "v1" else logits[0]
)
feats = feats.squeeze(0).float().cpu().numpy()
if np.isnan(feats).sum() == 0:
np.save(out_path, feats, allow_pickle=False)
else:
printt("%s-contains nan" % file)
if idx % n == 0:
printt("now-%s,all-%s,%s,%s" % (len(todo), idx, file, feats.shape))
except:
printt(traceback.format_exc())
printt("all-feature-done")
@@ -0,0 +1,34 @@
import json
import re
# Define regular expression patterns
pattern = r"""i18n\([\s\n\t]*(["'][^"']+["'])[\s\n\t]*\)"""
# Initialize the dictionary to store key-value pairs
data = {}
def process(fn: str):
global data
with open(fn, "r", encoding="utf-8") as f:
contents = f.read()
matches = re.findall(pattern, contents)
for key in matches:
key = eval(key)
print("extract:", key)
data[key] = key
print("processing infer-web.py")
process("infer-web.py")
print("processing gui_v0.py")
process("gui_v0.py")
print("processing gui_v1.py")
process("gui_v1.py")
# Save as a JSON file
with open("./lib/i18n/zh_CN.json", "w", encoding="utf-8") as f:
json.dump(data, f, ensure_ascii=False, indent=4)
f.write("\n")
@@ -0,0 +1,276 @@
"""
Our own copy of RVC's (somewhat difficult to use) script, `infer_batch_rvc.py`.
"""
import os, sys, pdb, torch
print("Env vars:")
print(os.environ)
def print_gpu_info():
print('========================================')
print('Python interpreter', sys.executable)
print('PyTorch version', torch.__version__)
print('CUDA Available?', torch.cuda.is_available())
print('CUDA Device count', torch.cuda.device_count())
print('CUDA architectures library was compiled for', torch.cuda.get_arch_list())
#try:
# from tensorflow.python.client import device_lib
# print('local devices', str(device_lib.list_local_devices()).replace("\n", "\n "))
#except ImportError:
# print('no tensorflow - cannot list devices')
# pass
print('========================================', flush=True)
print_gpu_info()
now_dir = os.getcwd()
sys.path.append(now_dir)
import sys
import torch
import tqdm as tq
from multiprocessing import cpu_count
class Config:
def __init__(self, device, is_half):
self.device = device
self.is_half = is_half
self.n_cpu = 0
self.gpu_name = None
self.gpu_mem = None
self.x_pad, self.x_query, self.x_center, self.x_max = self.device_config()
def device_config(self) -> tuple:
if torch.cuda.is_available():
i_device = int(self.device.split(":")[-1])
self.gpu_name = torch.cuda.get_device_name(i_device)
if (
("16" in self.gpu_name and "V100" not in self.gpu_name.upper())
or "P40" in self.gpu_name.upper()
or "1060" in self.gpu_name
or "1070" in self.gpu_name
or "1080" in self.gpu_name
):
print("16系/10系显卡和P40强制单精度")
self.is_half = False
for config_file in ["32k.json", "40k.json", "48k.json"]:
with open(f"configs/{config_file}", "r") as f:
strr = f.read().replace("true", "false")
with open(f"configs/{config_file}", "w") as f:
f.write(strr)
with open("trainset_preprocess_pipeline_print.py", "r") as f:
strr = f.read().replace("3.7", "3.0")
with open("trainset_preprocess_pipeline_print.py", "w") as f:
f.write(strr)
else:
self.gpu_name = None
self.gpu_mem = int(
torch.cuda.get_device_properties(i_device).total_memory
/ 1024
/ 1024
/ 1024
+ 0.4
)
if self.gpu_mem <= 4:
with open("trainset_preprocess_pipeline_print.py", "r") as f:
strr = f.read().replace("3.7", "3.0")
with open("trainset_preprocess_pipeline_print.py", "w") as f:
f.write(strr)
elif torch.backends.mps.is_available():
print("没有发现支持的N卡, 使用MPS进行推理")
self.device = "mps"
else:
print("没有发现支持的N卡, 使用CPU进行推理")
self.device = "cpu"
self.is_half = True
if self.n_cpu == 0:
self.n_cpu = cpu_count()
if self.is_half:
# 6G显存配置
x_pad = 3
x_query = 10
x_center = 60
x_max = 65
else:
# 5G显存配置
x_pad = 1
x_query = 6
x_center = 38
x_max = 41
if self.gpu_mem != None and self.gpu_mem <= 4:
x_pad = 1
x_query = 5
x_center = 30
x_max = 32
return x_pad, x_query, x_center, x_max
import argparse
parser = argparse.ArgumentParser(description='Run VC inference')
parser.add_argument('--model_path', type=str, help='path to the .pth file', required=True)
parser.add_argument('--model_index_path', type=str, help='path to the .index file (empty string for none)', required=False, default='')
parser.add_argument('--hubert_model_path', type=str, help='path to the hubert model .pth file (default hubert_base.pt)',
required=False, default='hubert_base.pt')
parser.add_argument('--input_audio_filename', type=str, help='source audio file', required=True)
parser.add_argument('--output_audio_filename', type=str, help='result audio file', required=True)
########
# Default values are taken from the Gradio UI
########
parser.add_argument('--f0_up_key', type=str,
help='the f0 key for the input audio file(default "0")',
required=False, default='0')
parser.add_argument('--f0_method', type=str,
help='f0 estimation method to use: harvest (default), crepe, or pm',
required=False, default='harvest')
parser.add_argument('--index_rate', type=float,
help='The rate for the index (search feature ratio) (default 0.75)',
required=False, default=0.75) # Gradio default 0.75
parser.add_argument('--filter_radius', type=int,
help='The radius of the filter. If >=3, apply median filtering to harvested pitch results. Represents the filter radius and can reduce breathiness. (default 3)',
required=False, default=3) # Gradio default 3
parser.add_argument('--resample_sr', type=int,
help='The sample rate for resampling (post process). Set to zero for no resample. (default 0)',
required=False, default=0)
parser.add_argument('--rms_mix_rate', type=float,
help='RMS mix rate. Use the volume envelope of the input to replace or mix with the volume envelope of the output. The closer the ratio to 1, the more the output envelope is used (default 0.25)',
required=False, default=0.25) # Gradio default 0.25
parser.add_argument('--protect', type=float,
help='The protect value. Protect voiceless consonants and breath sounds to prevent artifacts such as tearing in electronic music. Set to 0.5 to disable. Decrease the value to increase protection, but may reduce indexing accuracy (default 0.33 ???)',
required=False, default=0.33) # Gradio default 0.33
# Device / precision
parser.add_argument('--device', type=str, help='compute device (eg. "cuda:0"), default "cuda:0".', required=False, default='cuda:0')
parser.add_argument('--is_half', type=bool, help='halfwidth tensors (default false)', required=False, default=False)
args = parser.parse_args()
f0up_key = args.f0_up_key
input_path = args.input_audio_filename
index_path = args.model_index_path
f0method = args.f0_method
opt_path = args.output_audio_filename
model_path = args.model_path
index_rate = args.index_rate
device = args.device
is_half = args.is_half
filter_radius = args.filter_radius
resample_sr = args.resample_sr
rms_mix_rate = args.rms_mix_rate
protect = args.protect
hubert_path = args.hubert_model_path
config = Config(device, is_half)
now_dir = os.getcwd()
sys.path.append(now_dir)
from vc_infer_pipeline import VC
from lib.infer_pack.models import (
SynthesizerTrnMs256NSFsid,
SynthesizerTrnMs256NSFsid_nono,
SynthesizerTrnMs768NSFsid,
SynthesizerTrnMs768NSFsid_nono,
)
from lib.audio import load_audio
from fairseq import checkpoint_utils
from scipy.io import wavfile
hubert_model = None
def load_hubert(hubert_path="hubert_base.pt"):
global hubert_model
models, saved_cfg, task = checkpoint_utils.load_model_ensemble_and_task(
[hubert_path],
suffix="",
)
hubert_model = models[0]
hubert_model = hubert_model.to(device)
if is_half:
hubert_model = hubert_model.half()
else:
hubert_model = hubert_model.float()
hubert_model.eval()
def vc_single(sid, input_audio, f0_up_key, f0_file, f0_method, file_index, index_rate, hubert_path="hubert_base.pt"):
global tgt_sr, net_g, vc, hubert_model, version
if input_audio is None:
return "You need to upload an audio", None
f0_up_key = int(f0_up_key)
audio = load_audio(input_audio, 16000)
times = [0, 0, 0]
if hubert_model == None:
load_hubert(hubert_path)
if_f0 = cpt.get("f0", 1)
# audio_opt=vc.pipeline(hubert_model,net_g,sid,audio,times,f0_up_key,f0_method,file_index,file_big_npy,index_rate,if_f0,f0_file=f0_file)
audio_opt = vc.pipeline(
hubert_model,
net_g,
sid,
audio,
input_audio,
times,
f0_up_key,
f0_method,
file_index,
index_rate,
if_f0,
filter_radius,
tgt_sr,
resample_sr,
rms_mix_rate,
version,
protect,
f0_file=f0_file,
)
print(times)
return audio_opt
def get_vc(model_path):
global n_spk, tgt_sr, net_g, vc, cpt, device, is_half, version
print("loading pth %s" % model_path)
cpt = torch.load(model_path, map_location="cpu")
tgt_sr = cpt["config"][-1]
cpt["config"][-3] = cpt["weight"]["emb_g.weight"].shape[0] # n_spk
if_f0 = cpt.get("f0", 1)
version = cpt.get("version", "v1")
if version == "v1":
if if_f0 == 1:
net_g = SynthesizerTrnMs256NSFsid(*cpt["config"], is_half=is_half)
else:
net_g = SynthesizerTrnMs256NSFsid_nono(*cpt["config"])
elif version == "v2":
if if_f0 == 1: #
net_g = SynthesizerTrnMs768NSFsid(*cpt["config"], is_half=is_half)
else:
net_g = SynthesizerTrnMs768NSFsid_nono(*cpt["config"])
del net_g.enc_q
print(net_g.load_state_dict(cpt["weight"], strict=False)) # 不加这一行清不干净,真奇葩
net_g.eval().to(device)
if is_half:
net_g = net_g.half()
else:
net_g = net_g.float()
vc = VC(tgt_sr, config)
n_spk = cpt["config"][-3]
# return {"visible": True,"maximum": n_spk, "__type__": "update"}
get_vc(model_path)
file_path = input_path
wav_opt = vc_single(
0, file_path, f0up_key, None, f0method, index_path, index_rate, hubert_path
)
out_path = opt_path
wavfile.write(out_path, tgt_sr, wav_opt)
@@ -0,0 +1,165 @@
import os, sys, pdb, torch
from piper import PiperVoice
import wave
import argparse
from scipy.io import wavfile
import tqdm as tq
from multiprocessing import cpu_count
# Environment and GPU Info
print("Env vars:")
print(os.environ)
print('========================================')
print('Python interpreter', sys.executable)
print('PyTorch version', torch.__version__)
print('CUDA Available?', torch.cuda.is_available())
print('CUDA Device count', torch.cuda.device_count())
print('CUDA architectures library was compiled for', torch.cuda.get_arch_list())
print('========================================', flush=True)
now_dir = os.getcwd()
sys.path.append(now_dir)
# Config Class
class Config:
def __init__(self, device, is_half):
self.device = device
self.is_half = is_half
self.n_cpu = 0
self.gpu_name = None
self.gpu_mem = None
self.x_pad, self.x_query, self.x_center, self.x_max = self.device_config()
def device_config(self) -> tuple:
if torch.cuda.is_available():
i_device = int(self.device.split(":")[-1])
self.gpu_name = torch.cuda.get_device_name(i_device)
elif torch.backends.mps.is_available():
self.device = "mps"
else:
self.device = "cpu"
self.is_half = True
if self.n_cpu == 0:
self.n_cpu = cpu_count()
if self.is_half:
x_pad = 3
x_query = 10
x_center = 60
x_max = 65
else:
x_pad = 1
x_query = 6
x_center = 38
x_max = 41
if self.gpu_mem != None and self.gpu_mem <= 4:
x_pad = 1
x_query = 5
x_center = 30
x_max = 32
return x_pad, x_query, x_center, x_max
# Argument Parser
parser = argparse.ArgumentParser(description='Run Piper-TTS and RVC inference')
# Arguments for Piper-TTS
parser.add_argument('--tts_model_path', type=str, required=True)
parser.add_argument('--tts_config_path', type=str, required=True)
parser.add_argument('--text', type=str, required=True)
# Additional Arguments for Piper TTS
parser.add_argument('--length_scale', type=float, required=False, default=1.0, help='Phoneme length scale for TTS')
parser.add_argument('--noise_scale', type=float, required=False, default=0.5, help='Noise scale for TTS')
parser.add_argument('--noise_w', type=float, required=False, default=0.3, help='Phoneme width noise for TTS')
parser.add_argument('--sentence_silence', type=float, required=False, default=0.0, help='Seconds of silence after each sentence in TTS')
parser.add_argument('--model_path', type=str, required=True)
parser.add_argument('--model_index_path', type=str, required=False, default='')
parser.add_argument('--hubert_model_path', type=str, required=False, default='hubert_base.pt')
parser.add_argument('--input_audio_filename', type=str, required=True)
parser.add_argument('--output_audio_filename', type=str, required=True)
parser.add_argument('--f0_up_key', type=str, required=False, default='0')
parser.add_argument('--f0_method', type=str, required=False, default='harvest')
parser.add_argument('--index_rate', type=float, required=False, default=0.75)
parser.add_argument('--filter_radius', type=int, required=False, default=3)
parser.add_argument('--resample_sr', type=int, required=False, default=0)
parser.add_argument('--rms_mix_rate', type=float, required=False, default=0.25)
parser.add_argument('--protect', type=float, required=False, default=0.33)
parser.add_argument('--device', type=str, required=False, default='cuda:0')
parser.add_argument('--is_half', type=bool, required=False, default=False)
args = parser.parse_args()
# Load Piper Voice with additional settings
voice = PiperVoice.load(model_path=args.tts_model_path, config_path=args.tts_config_path)
synthesize_args = {
"length_scale": args.length_scale,
"noise_scale": args.noise_scale,
"noise_w": args.noise_w,
"sentence_silence": args.sentence_silence,
}
with wave.open('temp.wav', "wb") as wav_file:
voice.synthesize(args.text, wav_file, **synthesize_args)
config = Config(args.device, args.is_half)
now_dir = os.getcwd()
sys.path.append(now_dir)
from lib.train.vc_infer_pipeline import VC
from lib.infer_pack.models import SynthesizerTrnMs256NSFsid, SynthesizerTrnMs256NSFsid_nono, SynthesizerTrnMs768NSFsid, SynthesizerTrnMs768NSFsid_nono
from lib.audio import load_audio
from fairseq import checkpoint_utils
hubert_model = None
def load_hubert(hubert_path="hubert_base.pt"):
global hubert_model
models, saved_cfg, task = checkpoint_utils.load_model_ensemble_and_task([hubert_path])
hubert_model = models[0]
hubert_model = hubert_model.to(args.device)
if args.is_half:
hubert_model = hubert_model.half()
def vc_single(sid, input_audio, f0_up_key, f0_file, f0_method, file_index, index_rate):
global tgt_sr, net_g, vc, hubert_model, version
if input_audio is None:
return "You need to upload an audio", None
f0_up_key = int(f0_up_key)
audio = load_audio(input_audio, 16000)
times = [0, 0, 0]
if hubert_model == None:
load_hubert()
if_f0 = cpt.get("f0", 1)
audio_opt = vc.pipeline(hubert_model, net_g, sid, audio, input_audio, times, f0_up_key, f0_method, file_index, index_rate, if_f0, args.filter_radius, tgt_sr, args.resample_sr, args.rms_mix_rate, version, args.protect, f0_file=f0_file)
print(times)
return audio_opt
def get_vc(model_path):
global tgt_sr, net_g, vc, cpt, version
cpt = torch.load(model_path, map_location="cpu")
tgt_sr = cpt["config"][-1]
cpt["config"][-3] = cpt["weight"]["emb_g.weight"].shape[0]
if_f0 = cpt.get("f0", 1)
version = cpt.get("version", "v1")
if version == "v1":
if if_f0 == 1:
net_g = SynthesizerTrnMs256NSFsid(*cpt["config"], is_half=args.is_half)
else:
net_g = SynthesizerTrnMs256NSFsid_nono(*cpt["config"])
elif version == "v2":
if if_f0 == 1:
net_g = SynthesizerTrnMs768NSFsid(*cpt["config"], is_half=args.is_half)
else:
net_g = SynthesizerTrnMs768NSFsid_nono(*cpt["config"])
del net_g.enc_q
net_g.load_state_dict(cpt["weight"], strict=False)
net_g.eval().to(args.device)
if args.is_half:
net_g = net_g.half()
vc = VC(tgt_sr, config)
get_vc(args.model_path)
file_path = args.input_audio_filename
wav_opt = vc_single(0, file_path, args.f0_up_key, None, args.f0_method, args.model_index_path, args.index_rate)
out_path = args.output_audio_filename
wavfile.write(out_path, tgt_sr, wav_opt)
@@ -0,0 +1,2 @@
runtime\python.exe gui_v1.py --pycmd runtime\python.exe
pause
@@ -0,0 +1,2 @@
runtime\python.exe infer-web.py --pycmd runtime\python.exe --port 7897
pause
+661
View File
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import os, sys,pdb
os.environ["OMP_NUM_THREADS"]="2"
if sys.platform == "darwin":
os.environ["PYTORCH_ENABLE_MPS_FALLBACK"] = "1"
now_dir = os.getcwd()
sys.path.append(now_dir)
import multiprocessing
class Harvest(multiprocessing.Process):
def __init__(self, inp_q, opt_q):
multiprocessing.Process.__init__(self)
self.inp_q = inp_q
self.opt_q = opt_q
def run(self):
import numpy as np, pyworld
while 1:
idx, x, res_f0, n_cpu, ts = self.inp_q.get()
f0, t = pyworld.harvest(
x.astype(np.double),
fs=16000,
f0_ceil=1100,
f0_floor=50,
frame_period=10,
)
res_f0[idx] = f0
if len(res_f0.keys()) >= n_cpu:
self.opt_q.put(ts)
if __name__ == "__main__":
from multiprocessing import Queue
from queue import Empty
import numpy as np
import multiprocessing
import traceback, re
import json
import PySimpleGUI as sg
import sounddevice as sd
import noisereduce as nr
from multiprocessing import cpu_count
import librosa, torch, time, threading
import torch.nn.functional as F
import torchaudio.transforms as tat
from i18n import I18nAuto
import rvc_for_realtime
i18n = I18nAuto()
device=rvc_for_realtime.config.device
# device = torch.device(
# "cuda"
# if torch.cuda.is_available()
# else ("mps" if torch.backends.mps.is_available() else "cpu")
# )
current_dir = os.getcwd()
inp_q = Queue()
opt_q = Queue()
n_cpu = min(cpu_count(), 8)
for _ in range(n_cpu):
Harvest(inp_q, opt_q).start()
class GUIConfig:
def __init__(self) -> None:
self.pth_path: str = ""
self.index_path: str = ""
self.pitch: int = 12
self.samplerate: int = 40000
self.block_time: float = 1.0 # s
self.buffer_num: int = 1
self.threhold: int = -30
self.crossfade_time: float = 0.08
self.extra_time: float = 0.04
self.I_noise_reduce = False
self.O_noise_reduce = False
self.index_rate = 0.3
self.n_cpu = min(n_cpu, 6)
self.f0method = "harvest"
self.sg_input_device = ""
self.sg_output_device = ""
class GUI:
def __init__(self) -> None:
self.config = GUIConfig()
self.flag_vc = False
self.launcher()
def load(self):
input_devices, output_devices, _, _ = self.get_devices()
try:
with open("values1.json", "r") as j:
data = json.load(j)
data["pm"] = data["f0method"] == "pm"
data["harvest"] = data["f0method"] == "harvest"
data["crepe"] = data["f0method"] == "crepe"
data["rmvpe"] = data["f0method"] == "rmvpe"
except:
with open("values1.json", "w") as j:
data = {
"pth_path": " ",
"index_path": " ",
"sg_input_device": input_devices[sd.default.device[0]],
"sg_output_device": output_devices[sd.default.device[1]],
"threhold": "-45",
"pitch": "0",
"index_rate": "0",
"block_time": "1",
"crossfade_length": "0.04",
"extra_time": "1",
"f0method": "rmvpe",
}
return data
def launcher(self):
data = self.load()
sg.theme("LightBlue3")
input_devices, output_devices, _, _ = self.get_devices()
layout = [
[
sg.Frame(
title=i18n("加载模型"),
layout=[
[
sg.Input(
default_text=data.get("pth_path", ""),
key="pth_path",
),
sg.FileBrowse(
i18n("选择.pth文件"),
initial_folder=os.path.join(os.getcwd(), "weights"),
file_types=((". pth"),),
),
],
[
sg.Input(
default_text=data.get("index_path", ""),
key="index_path",
),
sg.FileBrowse(
i18n("选择.index文件"),
initial_folder=os.path.join(os.getcwd(), "logs"),
file_types=((". index"),),
),
],
],
)
],
[
sg.Frame(
layout=[
[
sg.Text(i18n("输入设备")),
sg.Combo(
input_devices,
key="sg_input_device",
default_value=data.get("sg_input_device", ""),
),
],
[
sg.Text(i18n("输出设备")),
sg.Combo(
output_devices,
key="sg_output_device",
default_value=data.get("sg_output_device", ""),
),
],
[sg.Button(i18n("重载设备列表"), key="reload_devices")],
],
title=i18n("音频设备(请使用同种类驱动)"),
)
],
[
sg.Frame(
layout=[
[
sg.Text(i18n("响应阈值")),
sg.Slider(
range=(-60, 0),
key="threhold",
resolution=1,
orientation="h",
default_value=data.get("threhold", ""),
),
],
[
sg.Text(i18n("音调设置")),
sg.Slider(
range=(-24, 24),
key="pitch",
resolution=1,
orientation="h",
default_value=data.get("pitch", ""),
),
],
[
sg.Text(i18n("Index Rate")),
sg.Slider(
range=(0.0, 1.0),
key="index_rate",
resolution=0.01,
orientation="h",
default_value=data.get("index_rate", ""),
),
],
[
sg.Text(i18n("音高算法")),
sg.Radio(
"pm",
"f0method",
key="pm",
default=data.get("pm", "") == True,
),
sg.Radio(
"harvest",
"f0method",
key="harvest",
default=data.get("harvest", "") == True,
),
sg.Radio(
"crepe",
"f0method",
key="crepe",
default=data.get("crepe", "") == True,
),
sg.Radio(
"rmvpe",
"f0method",
key="rmvpe",
default=data.get("rmvpe", "") == True,
),
],
],
title=i18n("常规设置"),
),
sg.Frame(
layout=[
[
sg.Text(i18n("采样长度")),
sg.Slider(
range=(0.09, 2.4),
key="block_time",
resolution=0.03,
orientation="h",
default_value=data.get("block_time", ""),
),
],
[
sg.Text(i18n("harvest进程数")),
sg.Slider(
range=(1, n_cpu),
key="n_cpu",
resolution=1,
orientation="h",
default_value=data.get(
"n_cpu", min(self.config.n_cpu, n_cpu)
),
),
],
[
sg.Text(i18n("淡入淡出长度")),
sg.Slider(
range=(0.01, 0.15),
key="crossfade_length",
resolution=0.01,
orientation="h",
default_value=data.get("crossfade_length", ""),
),
],
[
sg.Text(i18n("额外推理时长")),
sg.Slider(
range=(0.05, 5.00),
key="extra_time",
resolution=0.01,
orientation="h",
default_value=data.get("extra_time", ""),
),
],
[
sg.Checkbox(i18n("输入降噪"), key="I_noise_reduce"),
sg.Checkbox(i18n("输出降噪"), key="O_noise_reduce"),
],
],
title=i18n("性能设置"),
),
],
[
sg.Button(i18n("开始音频转换"), key="start_vc"),
sg.Button(i18n("停止音频转换"), key="stop_vc"),
sg.Text(i18n("推理时间(ms):")),
sg.Text("0", key="infer_time"),
],
]
self.window = sg.Window("RVC - GUI", layout=layout)
self.event_handler()
def event_handler(self):
while True:
event, values = self.window.read()
if event == sg.WINDOW_CLOSED:
self.flag_vc = False
exit()
if event == "reload_devices":
prev_input = self.window["sg_input_device"].get()
prev_output = self.window["sg_output_device"].get()
input_devices, output_devices, _, _ = self.get_devices(update=True)
if prev_input not in input_devices:
self.config.sg_input_device = input_devices[0]
else:
self.config.sg_input_device = prev_input
self.window["sg_input_device"].Update(values=input_devices)
self.window["sg_input_device"].Update(
value=self.config.sg_input_device
)
if prev_output not in output_devices:
self.config.sg_output_device = output_devices[0]
else:
self.config.sg_output_device = prev_output
self.window["sg_output_device"].Update(values=output_devices)
self.window["sg_output_device"].Update(
value=self.config.sg_output_device
)
if event == "start_vc" and self.flag_vc == False:
if self.set_values(values) == True:
print("using_cuda:" + str(torch.cuda.is_available()))
self.start_vc()
settings = {
"pth_path": values["pth_path"],
"index_path": values["index_path"],
"sg_input_device": values["sg_input_device"],
"sg_output_device": values["sg_output_device"],
"threhold": values["threhold"],
"pitch": values["pitch"],
"index_rate": values["index_rate"],
"block_time": values["block_time"],
"crossfade_length": values["crossfade_length"],
"extra_time": values["extra_time"],
"n_cpu": values["n_cpu"],
"f0method": ["pm", "harvest", "crepe", "rmvpe"][
[
values["pm"],
values["harvest"],
values["crepe"],
values["rmvpe"],
].index(True)
],
}
with open("values1.json", "w") as j:
json.dump(settings, j)
if event == "stop_vc" and self.flag_vc == True:
self.flag_vc = False
def set_values(self, values):
if len(values["pth_path"].strip()) == 0:
sg.popup(i18n("请选择pth文件"))
return False
if len(values["index_path"].strip()) == 0:
sg.popup(i18n("请选择index文件"))
return False
pattern = re.compile("[^\x00-\x7F]+")
if pattern.findall(values["pth_path"]):
sg.popup(i18n("pth文件路径不可包含中文"))
return False
if pattern.findall(values["index_path"]):
sg.popup(i18n("index文件路径不可包含中文"))
return False
self.set_devices(values["sg_input_device"], values["sg_output_device"])
self.config.pth_path = values["pth_path"]
self.config.index_path = values["index_path"]
self.config.threhold = values["threhold"]
self.config.pitch = values["pitch"]
self.config.block_time = values["block_time"]
self.config.crossfade_time = values["crossfade_length"]
self.config.extra_time = values["extra_time"]
self.config.I_noise_reduce = values["I_noise_reduce"]
self.config.O_noise_reduce = values["O_noise_reduce"]
self.config.index_rate = values["index_rate"]
self.config.n_cpu = values["n_cpu"]
self.config.f0method = ["pm", "harvest", "crepe", "rmvpe"][
[
values["pm"],
values["harvest"],
values["crepe"],
values["rmvpe"],
].index(True)
]
return True
def start_vc(self):
torch.cuda.empty_cache()
self.flag_vc = True
self.rvc = rvc_for_realtime.RVC(
self.config.pitch,
self.config.pth_path,
self.config.index_path,
self.config.index_rate,
self.config.n_cpu,
inp_q,
opt_q,
device,
)
self.config.samplerate = self.rvc.tgt_sr
self.config.crossfade_time = min(
self.config.crossfade_time, self.config.block_time
)
self.block_frame = int(self.config.block_time * self.config.samplerate)
self.crossfade_frame = int(
self.config.crossfade_time * self.config.samplerate
)
self.sola_search_frame = int(0.01 * self.config.samplerate)
self.extra_frame = int(self.config.extra_time * self.config.samplerate)
self.zc = self.rvc.tgt_sr // 100
self.input_wav: np.ndarray = np.zeros(
int(
np.ceil(
(
self.extra_frame
+ self.crossfade_frame
+ self.sola_search_frame
+ self.block_frame
)
/ self.zc
)
* self.zc
),
dtype="float32",
)
self.output_wav_cache: torch.Tensor = torch.zeros(
int(
np.ceil(
(
self.extra_frame
+ self.crossfade_frame
+ self.sola_search_frame
+ self.block_frame
)
/ self.zc
)
* self.zc
),
device=device,
dtype=torch.float32,
)
self.pitch: np.ndarray = np.zeros(
self.input_wav.shape[0] // self.zc,
dtype="int32",
)
self.pitchf: np.ndarray = np.zeros(
self.input_wav.shape[0] // self.zc,
dtype="float64",
)
self.output_wav: torch.Tensor = torch.zeros(
self.block_frame, device=device, dtype=torch.float32
)
self.sola_buffer: torch.Tensor = torch.zeros(
self.crossfade_frame, device=device, dtype=torch.float32
)
self.fade_in_window: torch.Tensor = torch.linspace(
0.0, 1.0, steps=self.crossfade_frame, device=device, dtype=torch.float32
)
self.fade_out_window: torch.Tensor = 1 - self.fade_in_window
self.resampler = tat.Resample(
orig_freq=self.config.samplerate, new_freq=16000, dtype=torch.float32
).to(device)
thread_vc = threading.Thread(target=self.soundinput)
thread_vc.start()
def soundinput(self):
"""
接受音频输入
"""
channels = 1 if sys.platform == "darwin" else 2
with sd.Stream(
channels=channels,
callback=self.audio_callback,
blocksize=self.block_frame,
samplerate=self.config.samplerate,
dtype="float32",
):
while self.flag_vc:
time.sleep(self.config.block_time)
print("Audio block passed.")
print("ENDing VC")
def audio_callback(
self, indata: np.ndarray, outdata: np.ndarray, frames, times, status
):
"""
音频处理
"""
start_time = time.perf_counter()
indata = librosa.to_mono(indata.T)
if self.config.I_noise_reduce:
indata[:] = nr.reduce_noise(y=indata, sr=self.config.samplerate)
"""noise gate"""
frame_length = 2048
hop_length = 1024
rms = librosa.feature.rms(
y=indata, frame_length=frame_length, hop_length=hop_length
)
if self.config.threhold > -60:
db_threhold = (
librosa.amplitude_to_db(rms, ref=1.0)[0] < self.config.threhold
)
for i in range(db_threhold.shape[0]):
if db_threhold[i]:
indata[i * hop_length : (i + 1) * hop_length] = 0
self.input_wav[:] = np.append(self.input_wav[self.block_frame :], indata)
# infer
inp = torch.from_numpy(self.input_wav).to(device)
res1 = self.resampler(inp)
###55%
rate1 = self.block_frame / (
self.extra_frame
+ self.crossfade_frame
+ self.sola_search_frame
+ self.block_frame
)
rate2 = (
self.crossfade_frame + self.sola_search_frame + self.block_frame
) / (
self.extra_frame
+ self.crossfade_frame
+ self.sola_search_frame
+ self.block_frame
)
res2 = self.rvc.infer(
res1,
res1[-self.block_frame :].cpu().numpy(),
rate1,
rate2,
self.pitch,
self.pitchf,
self.config.f0method,
)
self.output_wav_cache[-res2.shape[0] :] = res2
infer_wav = self.output_wav_cache[
-self.crossfade_frame - self.sola_search_frame - self.block_frame :
]
# SOLA algorithm from https://github.com/yxlllc/DDSP-SVC
cor_nom = F.conv1d(
infer_wav[None, None, : self.crossfade_frame + self.sola_search_frame],
self.sola_buffer[None, None, :],
)
cor_den = torch.sqrt(
F.conv1d(
infer_wav[
None, None, : self.crossfade_frame + self.sola_search_frame
]
** 2,
torch.ones(1, 1, self.crossfade_frame, device=device),
)
+ 1e-8
)
if sys.platform == "darwin":
_, sola_offset = torch.max(cor_nom[0, 0] / cor_den[0, 0])
sola_offset = sola_offset.item()
else:
sola_offset = torch.argmax(cor_nom[0, 0] / cor_den[0, 0])
print("sola offset: " + str(int(sola_offset)))
self.output_wav[:] = infer_wav[sola_offset : sola_offset + self.block_frame]
self.output_wav[: self.crossfade_frame] *= self.fade_in_window
self.output_wav[: self.crossfade_frame] += self.sola_buffer[:]
# crossfade
if sola_offset < self.sola_search_frame:
self.sola_buffer[:] = (
infer_wav[
-self.sola_search_frame
- self.crossfade_frame
+ sola_offset : -self.sola_search_frame
+ sola_offset
]
* self.fade_out_window
)
else:
self.sola_buffer[:] = (
infer_wav[-self.crossfade_frame :] * self.fade_out_window
)
if self.config.O_noise_reduce:
if sys.platform == "darwin":
noise_reduced_signal = nr.reduce_noise(
y=self.output_wav[:].cpu().numpy(), sr=self.config.samplerate
)
outdata[:] = noise_reduced_signal[:, np.newaxis]
else:
outdata[:] = np.tile(
nr.reduce_noise(
y=self.output_wav[:].cpu().numpy(),
sr=self.config.samplerate,
),
(2, 1),
).T
else:
if sys.platform == "darwin":
outdata[:] = self.output_wav[:].cpu().numpy()[:, np.newaxis]
else:
outdata[:] = self.output_wav[:].repeat(2, 1).t().cpu().numpy()
total_time = time.perf_counter() - start_time
self.window["infer_time"].update(int(total_time * 1000))
print("infer time:" + str(total_time))
def get_devices(self, update: bool = True):
"""获取设备列表"""
if update:
sd._terminate()
sd._initialize()
devices = sd.query_devices()
hostapis = sd.query_hostapis()
for hostapi in hostapis:
for device_idx in hostapi["devices"]:
devices[device_idx]["hostapi_name"] = hostapi["name"]
input_devices = [
f"{d['name']} ({d['hostapi_name']})"
for d in devices
if d["max_input_channels"] > 0
]
output_devices = [
f"{d['name']} ({d['hostapi_name']})"
for d in devices
if d["max_output_channels"] > 0
]
input_devices_indices = [
d["index"] if "index" in d else d["name"]
for d in devices
if d["max_input_channels"] > 0
]
output_devices_indices = [
d["index"] if "index" in d else d["name"]
for d in devices
if d["max_output_channels"] > 0
]
return (
input_devices,
output_devices,
input_devices_indices,
output_devices_indices,
)
def set_devices(self, input_device, output_device):
"""设置输出设备"""
(
input_devices,
output_devices,
input_device_indices,
output_device_indices,
) = self.get_devices()
sd.default.device[0] = input_device_indices[
input_devices.index(input_device)
]
sd.default.device[1] = output_device_indices[
output_devices.index(output_device)
]
print("input device:" + str(sd.default.device[0]) + ":" + str(input_device))
print(
"output device:" + str(sd.default.device[1]) + ":" + str(output_device)
)
gui = GUI()
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import locale
import json
import os
def load_language_list(language):
with open(f"./lib/i18n/{language}.json", "r", encoding="utf-8") as f:
language_list = json.load(f)
return language_list
class I18nAuto:
def __init__(self, language=None):
if language in ["Auto", None]:
language = locale.getdefaultlocale()[
0
] # getlocale can't identify the system's language ((None, None))
if not os.path.exists(f"./lib/i18n/{language}.json"):
language = "en_US"
self.language = language
# print("Use Language:", language)
self.language_map = load_language_list(language)
def __call__(self, key):
return self.language_map.get(key, key)
def print(self):
print("Use Language:", self.language)
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@@ -0,0 +1,215 @@
"""
v1
runtime\python.exe myinfer-v2-0528.py 0 "E:\codes\py39\RVC-beta\todo-songs" "E:\codes\py39\logs\mi-test\added_IVF677_Flat_nprobe_7.index" harvest "E:\codes\py39\RVC-beta\output" "E:\codes\py39\test-20230416b\weights\mi-test.pth" 0.66 cuda:0 True 3 0 1 0.33
v2
runtime\python.exe myinfer-v2-0528.py 0 "E:\codes\py39\RVC-beta\todo-songs" "E:\codes\py39\test-20230416b\logs\mi-test-v2\aadded_IVF677_Flat_nprobe_1_v2.index" harvest "E:\codes\py39\RVC-beta\output_v2" "E:\codes\py39\test-20230416b\weights\mi-test-v2.pth" 0.66 cuda:0 True 3 0 1 0.33
"""
import os, sys, pdb, torch
now_dir = os.getcwd()
sys.path.append(now_dir)
import sys
import torch
import tqdm as tq
from multiprocessing import cpu_count
class Config:
def __init__(self, device, is_half):
self.device = device
self.is_half = is_half
self.n_cpu = 0
self.gpu_name = None
self.gpu_mem = None
self.x_pad, self.x_query, self.x_center, self.x_max = self.device_config()
def device_config(self) -> tuple:
if torch.cuda.is_available():
i_device = int(self.device.split(":")[-1])
self.gpu_name = torch.cuda.get_device_name(i_device)
if (
("16" in self.gpu_name and "V100" not in self.gpu_name.upper())
or "P40" in self.gpu_name.upper()
or "1060" in self.gpu_name
or "1070" in self.gpu_name
or "1080" in self.gpu_name
):
print("16系/10系显卡和P40强制单精度")
self.is_half = False
for config_file in ["32k.json", "40k.json", "48k.json"]:
with open(f"configs/{config_file}", "r") as f:
strr = f.read().replace("true", "false")
with open(f"configs/{config_file}", "w") as f:
f.write(strr)
with open("trainset_preprocess_pipeline_print.py", "r") as f:
strr = f.read().replace("3.7", "3.0")
with open("trainset_preprocess_pipeline_print.py", "w") as f:
f.write(strr)
else:
self.gpu_name = None
self.gpu_mem = int(
torch.cuda.get_device_properties(i_device).total_memory
/ 1024
/ 1024
/ 1024
+ 0.4
)
if self.gpu_mem <= 4:
with open("trainset_preprocess_pipeline_print.py", "r") as f:
strr = f.read().replace("3.7", "3.0")
with open("trainset_preprocess_pipeline_print.py", "w") as f:
f.write(strr)
elif torch.backends.mps.is_available():
print("没有发现支持的N卡, 使用MPS进行推理")
self.device = "mps"
else:
print("没有发现支持的N卡, 使用CPU进行推理")
self.device = "cpu"
self.is_half = True
if self.n_cpu == 0:
self.n_cpu = cpu_count()
if self.is_half:
# 6G显存配置
x_pad = 3
x_query = 10
x_center = 60
x_max = 65
else:
# 5G显存配置
x_pad = 1
x_query = 6
x_center = 38
x_max = 41
if self.gpu_mem != None and self.gpu_mem <= 4:
x_pad = 1
x_query = 5
x_center = 30
x_max = 32
return x_pad, x_query, x_center, x_max
f0up_key = sys.argv[1]
input_path = sys.argv[2]
index_path = sys.argv[3]
f0method = sys.argv[4] # harvest or pm
opt_path = sys.argv[5]
model_path = sys.argv[6]
index_rate = float(sys.argv[7])
device = sys.argv[8]
is_half = sys.argv[9].lower() != "false"
filter_radius = int(sys.argv[10])
resample_sr = int(sys.argv[11])
rms_mix_rate = float(sys.argv[12])
protect = float(sys.argv[13])
print(sys.argv)
config = Config(device, is_half)
now_dir = os.getcwd()
sys.path.append(now_dir)
from vc_infer_pipeline import VC
from lib.infer_pack.models import (
SynthesizerTrnMs256NSFsid,
SynthesizerTrnMs256NSFsid_nono,
SynthesizerTrnMs768NSFsid,
SynthesizerTrnMs768NSFsid_nono,
)
from lib.audio import load_audio
from fairseq import checkpoint_utils
from scipy.io import wavfile
hubert_model = None
def load_hubert():
global hubert_model
models, saved_cfg, task = checkpoint_utils.load_model_ensemble_and_task(
["hubert_base.pt"],
suffix="",
)
hubert_model = models[0]
hubert_model = hubert_model.to(device)
if is_half:
hubert_model = hubert_model.half()
else:
hubert_model = hubert_model.float()
hubert_model.eval()
def vc_single(sid, input_audio, f0_up_key, f0_file, f0_method, file_index, index_rate):
global tgt_sr, net_g, vc, hubert_model, version
if input_audio is None:
return "You need to upload an audio", None
f0_up_key = int(f0_up_key)
audio = load_audio(input_audio, 16000)
times = [0, 0, 0]
if hubert_model == None:
load_hubert()
if_f0 = cpt.get("f0", 1)
# audio_opt=vc.pipeline(hubert_model,net_g,sid,audio,times,f0_up_key,f0_method,file_index,file_big_npy,index_rate,if_f0,f0_file=f0_file)
audio_opt = vc.pipeline(
hubert_model,
net_g,
sid,
audio,
input_audio,
times,
f0_up_key,
f0_method,
file_index,
index_rate,
if_f0,
filter_radius,
tgt_sr,
resample_sr,
rms_mix_rate,
version,
protect,
f0_file=f0_file,
)
print(times)
return audio_opt
def get_vc(model_path):
global n_spk, tgt_sr, net_g, vc, cpt, device, is_half, version
print("loading pth %s" % model_path)
cpt = torch.load(model_path, map_location="cpu")
tgt_sr = cpt["config"][-1]
cpt["config"][-3] = cpt["weight"]["emb_g.weight"].shape[0] # n_spk
if_f0 = cpt.get("f0", 1)
version = cpt.get("version", "v1")
if version == "v1":
if if_f0 == 1:
net_g = SynthesizerTrnMs256NSFsid(*cpt["config"], is_half=is_half)
else:
net_g = SynthesizerTrnMs256NSFsid_nono(*cpt["config"])
elif version == "v2":
if if_f0 == 1: #
net_g = SynthesizerTrnMs768NSFsid(*cpt["config"], is_half=is_half)
else:
net_g = SynthesizerTrnMs768NSFsid_nono(*cpt["config"])
del net_g.enc_q
print(net_g.load_state_dict(cpt["weight"], strict=False)) # 不加这一行清不干净,真奇葩
net_g.eval().to(device)
if is_half:
net_g = net_g.half()
else:
net_g = net_g.float()
vc = VC(tgt_sr, config)
n_spk = cpt["config"][-3]
# return {"visible": True,"maximum": n_spk, "__type__": "update"}
get_vc(model_path)
audios = os.listdir(input_path)
for file in tq.tqdm(audios):
if file.endswith(".wav"):
file_path = input_path + "/" + file
wav_opt = vc_single(
0, file_path, f0up_key, None, f0method, index_path, index_rate
)
out_path = opt_path + "/" + file
wavfile.write(out_path, tgt_sr, wav_opt)
@@ -0,0 +1,272 @@
from scipy.io import wavfile
from fairseq import checkpoint_utils
from lib.audio import load_audio
from lib.infer_pack.models import (
SynthesizerTrnMs256NSFsid,
SynthesizerTrnMs256NSFsid_nono,
SynthesizerTrnMs768NSFsid,
SynthesizerTrnMs768NSFsid_nono,
)
from vc_infer_pipeline import VC
from multiprocessing import cpu_count
import numpy as np
import torch
import sys
import glob
import argparse
import os
import sys
import pdb
import torch
now_dir = os.getcwd()
sys.path.append(now_dir)
####
# USAGE
#
# In your Terminal or CMD or whatever
# python infer_cli.py [TRANSPOSE_VALUE] "[INPUT_PATH]" "[OUTPUT_PATH]" "[MODEL_PATH]" "[INDEX_FILE_PATH]" "[INFERENCE_DEVICE]" "[METHOD]"
using_cli = False
device = "cuda:0"
is_half = False
if len(sys.argv) > 0:
f0_up_key = int(sys.argv[1]) # transpose value
input_path = sys.argv[2]
output_path = sys.argv[3]
model_path = sys.argv[4]
file_index = sys.argv[5] # .index file
device = sys.argv[6]
f0_method = sys.argv[7] # pm or harvest or crepe
using_cli = True
# file_index2=sys.argv[8]
# index_rate=float(sys.argv[10]) #search feature ratio
# filter_radius=float(sys.argv[11]) #median filter
# resample_sr=float(sys.argv[12]) #resample audio in post processing
# rms_mix_rate=float(sys.argv[13]) #search feature
print(sys.argv)
class Config:
def __init__(self, device, is_half):
self.device = device
self.is_half = is_half
self.n_cpu = 0
self.gpu_name = None
self.gpu_mem = None
self.x_pad, self.x_query, self.x_center, self.x_max = self.device_config()
def device_config(self) -> tuple:
if torch.cuda.is_available() and device != "cpu":
i_device = int(self.device.split(":")[-1])
self.gpu_name = torch.cuda.get_device_name(i_device)
if (
("16" in self.gpu_name and "V100" not in self.gpu_name.upper())
or "P40" in self.gpu_name.upper()
or "1060" in self.gpu_name
or "1070" in self.gpu_name
or "1080" in self.gpu_name
):
print("16系/10系显卡和P40强制单精度")
self.is_half = False
for config_file in ["32k.json", "40k.json", "48k.json"]:
with open(f"configs/{config_file}", "r") as f:
strr = f.read().replace("true", "false")
with open(f"configs/{config_file}", "w") as f:
f.write(strr)
with open("trainset_preprocess_pipeline_print.py", "r") as f:
strr = f.read().replace("3.7", "3.0")
with open("trainset_preprocess_pipeline_print.py", "w") as f:
f.write(strr)
else:
self.gpu_name = None
self.gpu_mem = int(
torch.cuda.get_device_properties(i_device).total_memory
/ 1024
/ 1024
/ 1024
+ 0.4
)
if self.gpu_mem <= 4:
with open("trainset_preprocess_pipeline_print.py", "r") as f:
strr = f.read().replace("3.7", "3.0")
with open("trainset_preprocess_pipeline_print.py", "w") as f:
f.write(strr)
elif torch.backends.mps.is_available():
print("没有发现支持的N卡, 使用MPS进行推理")
self.device = "mps"
else:
print("没有发现支持的N卡, 使用CPU进行推理")
self.device = "cpu"
self.is_half = False
if self.n_cpu == 0:
self.n_cpu = cpu_count()
if self.is_half:
# 6G显存配置
x_pad = 3
x_query = 10
x_center = 60
x_max = 65
else:
# 5G显存配置
x_pad = 1
x_query = 6
x_center = 38
x_max = 41
if self.gpu_mem != None and self.gpu_mem <= 4:
x_pad = 1
x_query = 5
x_center = 30
x_max = 32
return x_pad, x_query, x_center, x_max
config = Config(device, is_half)
now_dir = os.getcwd()
sys.path.append(now_dir)
hubert_model = None
def load_hubert():
global hubert_model
models, _, _ = checkpoint_utils.load_model_ensemble_and_task(
["hubert_base.pt"],
suffix="",
)
hubert_model = models[0]
hubert_model = hubert_model.to(config.device)
if config.is_half:
hubert_model = hubert_model.half()
else:
hubert_model = hubert_model.float()
hubert_model.eval()
def vc_single(
sid=0,
input_audio_path=None,
f0_up_key=0,
f0_file=None,
f0_method="pm",
file_index="", # .index file
file_index2="",
# file_big_npy,
index_rate=1.0,
filter_radius=3,
resample_sr=0,
rms_mix_rate=1.0,
model_path="",
output_path="",
protect=0.33,
):
global tgt_sr, net_g, vc, hubert_model, version
get_vc(model_path)
if input_audio_path is None:
return "You need to upload an audio file", None
f0_up_key = int(f0_up_key)
audio = load_audio(input_audio_path, 16000)
audio_max = np.abs(audio).max() / 0.95
if audio_max > 1:
audio /= audio_max
times = [0, 0, 0]
if hubert_model == None:
load_hubert()
if_f0 = cpt.get("f0", 1)
file_index = (
(
file_index.strip(" ")
.strip('"')
.strip("\n")
.strip('"')
.strip(" ")
.replace("trained", "added")
)
if file_index != ""
else file_index2
)
audio_opt = vc.pipeline(
hubert_model,
net_g,
sid,
audio,
input_audio_path,
times,
f0_up_key,
f0_method,
file_index,
# file_big_npy,
index_rate,
if_f0,
filter_radius,
tgt_sr,
resample_sr,
rms_mix_rate,
version,
f0_file=f0_file,
protect=protect,
)
wavfile.write(output_path, tgt_sr, audio_opt)
return "processed"
def get_vc(model_path):
global n_spk, tgt_sr, net_g, vc, cpt, device, is_half, version
print("loading pth %s" % model_path)
cpt = torch.load(model_path, map_location="cpu")
tgt_sr = cpt["config"][-1]
cpt["config"][-3] = cpt["weight"]["emb_g.weight"].shape[0] # n_spk
if_f0 = cpt.get("f0", 1)
version = cpt.get("version", "v1")
if version == "v1":
if if_f0 == 1:
net_g = SynthesizerTrnMs256NSFsid(*cpt["config"], is_half=is_half)
else:
net_g = SynthesizerTrnMs256NSFsid_nono(*cpt["config"])
elif version == "v2":
if if_f0 == 1:
net_g = SynthesizerTrnMs768NSFsid(*cpt["config"], is_half=is_half)
else:
net_g = SynthesizerTrnMs768NSFsid_nono(*cpt["config"])
del net_g.enc_q
print(net_g.load_state_dict(cpt["weight"], strict=False))
net_g.eval().to(device)
if is_half:
net_g = net_g.half()
else:
net_g = net_g.float()
vc = VC(tgt_sr, config)
n_spk = cpt["config"][-3]
# return {"visible": True,"maximum": n_spk, "__type__": "update"}
if using_cli:
vc_single(
sid=0,
input_audio_path=input_path,
f0_up_key=f0_up_key,
f0_file=None,
f0_method=f0_method,
file_index=file_index,
file_index2="",
index_rate=1,
filter_radius=3,
resample_sr=0,
rms_mix_rate=0,
model_path=model_path,
output_path=output_path,
)
@@ -0,0 +1,363 @@
import os, sys, torch, warnings, pdb
now_dir = os.getcwd()
sys.path.append(now_dir)
from json import load as ll
warnings.filterwarnings("ignore")
import librosa
import importlib
import numpy as np
import hashlib, math
from tqdm import tqdm
from lib.uvr5_pack.lib_v5 import spec_utils
from lib.uvr5_pack.utils import _get_name_params, inference
from lib.uvr5_pack.lib_v5.model_param_init import ModelParameters
import soundfile as sf
from lib.uvr5_pack.lib_v5.nets_new import CascadedNet
from lib.uvr5_pack.lib_v5 import nets_61968KB as nets
class _audio_pre_:
def __init__(self, agg, model_path, device, is_half):
self.model_path = model_path
self.device = device
self.data = {
# Processing Options
"postprocess": False,
"tta": False,
# Constants
"window_size": 512,
"agg": agg,
"high_end_process": "mirroring",
}
mp = ModelParameters("lib/uvr5_pack/lib_v5/modelparams/4band_v2.json")
model = nets.CascadedASPPNet(mp.param["bins"] * 2)
cpk = torch.load(model_path, map_location="cpu")
model.load_state_dict(cpk)
model.eval()
if is_half:
model = model.half().to(device)
else:
model = model.to(device)
self.mp = mp
self.model = model
def _path_audio_(self, music_file, ins_root=None, vocal_root=None, format="flac"):
if ins_root is None and vocal_root is None:
return "No save root."
name = os.path.basename(music_file)
if ins_root is not None:
os.makedirs(ins_root, exist_ok=True)
if vocal_root is not None:
os.makedirs(vocal_root, exist_ok=True)
X_wave, y_wave, X_spec_s, y_spec_s = {}, {}, {}, {}
bands_n = len(self.mp.param["band"])
# print(bands_n)
for d in range(bands_n, 0, -1):
bp = self.mp.param["band"][d]
if d == bands_n: # high-end band
(
X_wave[d],
_,
) = librosa.core.load( # 理论上librosa读取可能对某些音频有bug,应该上ffmpeg读取,但是太麻烦了弃坑
music_file,
bp["sr"],
False,
dtype=np.float32,
res_type=bp["res_type"],
)
if X_wave[d].ndim == 1:
X_wave[d] = np.asfortranarray([X_wave[d], X_wave[d]])
else: # lower bands
X_wave[d] = librosa.core.resample(
X_wave[d + 1],
self.mp.param["band"][d + 1]["sr"],
bp["sr"],
res_type=bp["res_type"],
)
# Stft of wave source
X_spec_s[d] = spec_utils.wave_to_spectrogram_mt(
X_wave[d],
bp["hl"],
bp["n_fft"],
self.mp.param["mid_side"],
self.mp.param["mid_side_b2"],
self.mp.param["reverse"],
)
# pdb.set_trace()
if d == bands_n and self.data["high_end_process"] != "none":
input_high_end_h = (bp["n_fft"] // 2 - bp["crop_stop"]) + (
self.mp.param["pre_filter_stop"] - self.mp.param["pre_filter_start"]
)
input_high_end = X_spec_s[d][
:, bp["n_fft"] // 2 - input_high_end_h : bp["n_fft"] // 2, :
]
X_spec_m = spec_utils.combine_spectrograms(X_spec_s, self.mp)
aggresive_set = float(self.data["agg"] / 100)
aggressiveness = {
"value": aggresive_set,
"split_bin": self.mp.param["band"][1]["crop_stop"],
}
with torch.no_grad():
pred, X_mag, X_phase = inference(
X_spec_m, self.device, self.model, aggressiveness, self.data
)
# Postprocess
if self.data["postprocess"]:
pred_inv = np.clip(X_mag - pred, 0, np.inf)
pred = spec_utils.mask_silence(pred, pred_inv)
y_spec_m = pred * X_phase
v_spec_m = X_spec_m - y_spec_m
if ins_root is not None:
if self.data["high_end_process"].startswith("mirroring"):
input_high_end_ = spec_utils.mirroring(
self.data["high_end_process"], y_spec_m, input_high_end, self.mp
)
wav_instrument = spec_utils.cmb_spectrogram_to_wave(
y_spec_m, self.mp, input_high_end_h, input_high_end_
)
else:
wav_instrument = spec_utils.cmb_spectrogram_to_wave(y_spec_m, self.mp)
print("%s instruments done" % name)
if format in ["wav", "flac"]:
sf.write(
os.path.join(
ins_root,
"instrument_{}_{}.{}".format(name, self.data["agg"], format),
),
(np.array(wav_instrument) * 32768).astype("int16"),
self.mp.param["sr"],
) #
else:
path = os.path.join(
ins_root, "instrument_{}_{}.wav".format(name, self.data["agg"])
)
sf.write(
path,
(np.array(wav_instrument) * 32768).astype("int16"),
self.mp.param["sr"],
)
if os.path.exists(path):
os.system(
"ffmpeg -i %s -vn %s -q:a 2 -y"
% (path, path[:-4] + ".%s" % format)
)
if vocal_root is not None:
if self.data["high_end_process"].startswith("mirroring"):
input_high_end_ = spec_utils.mirroring(
self.data["high_end_process"], v_spec_m, input_high_end, self.mp
)
wav_vocals = spec_utils.cmb_spectrogram_to_wave(
v_spec_m, self.mp, input_high_end_h, input_high_end_
)
else:
wav_vocals = spec_utils.cmb_spectrogram_to_wave(v_spec_m, self.mp)
print("%s vocals done" % name)
if format in ["wav", "flac"]:
sf.write(
os.path.join(
vocal_root,
"vocal_{}_{}.{}".format(name, self.data["agg"], format),
),
(np.array(wav_vocals) * 32768).astype("int16"),
self.mp.param["sr"],
)
else:
path = os.path.join(
vocal_root, "vocal_{}_{}.wav".format(name, self.data["agg"])
)
sf.write(
path,
(np.array(wav_vocals) * 32768).astype("int16"),
self.mp.param["sr"],
)
if os.path.exists(path):
os.system(
"ffmpeg -i %s -vn %s -q:a 2 -y"
% (path, path[:-4] + ".%s" % format)
)
class _audio_pre_new:
def __init__(self, agg, model_path, device, is_half):
self.model_path = model_path
self.device = device
self.data = {
# Processing Options
"postprocess": False,
"tta": False,
# Constants
"window_size": 512,
"agg": agg,
"high_end_process": "mirroring",
}
mp = ModelParameters("lib/uvr5_pack/lib_v5/modelparams/4band_v3.json")
nout = 64 if "DeReverb" in model_path else 48
model = CascadedNet(mp.param["bins"] * 2, nout)
cpk = torch.load(model_path, map_location="cpu")
model.load_state_dict(cpk)
model.eval()
if is_half:
model = model.half().to(device)
else:
model = model.to(device)
self.mp = mp
self.model = model
def _path_audio_(
self, music_file, vocal_root=None, ins_root=None, format="flac"
): # 3个VR模型vocal和ins是反的
if ins_root is None and vocal_root is None:
return "No save root."
name = os.path.basename(music_file)
if ins_root is not None:
os.makedirs(ins_root, exist_ok=True)
if vocal_root is not None:
os.makedirs(vocal_root, exist_ok=True)
X_wave, y_wave, X_spec_s, y_spec_s = {}, {}, {}, {}
bands_n = len(self.mp.param["band"])
# print(bands_n)
for d in range(bands_n, 0, -1):
bp = self.mp.param["band"][d]
if d == bands_n: # high-end band
(
X_wave[d],
_,
) = librosa.core.load( # 理论上librosa读取可能对某些音频有bug,应该上ffmpeg读取,但是太麻烦了弃坑
music_file,
bp["sr"],
False,
dtype=np.float32,
res_type=bp["res_type"],
)
if X_wave[d].ndim == 1:
X_wave[d] = np.asfortranarray([X_wave[d], X_wave[d]])
else: # lower bands
X_wave[d] = librosa.core.resample(
X_wave[d + 1],
self.mp.param["band"][d + 1]["sr"],
bp["sr"],
res_type=bp["res_type"],
)
# Stft of wave source
X_spec_s[d] = spec_utils.wave_to_spectrogram_mt(
X_wave[d],
bp["hl"],
bp["n_fft"],
self.mp.param["mid_side"],
self.mp.param["mid_side_b2"],
self.mp.param["reverse"],
)
# pdb.set_trace()
if d == bands_n and self.data["high_end_process"] != "none":
input_high_end_h = (bp["n_fft"] // 2 - bp["crop_stop"]) + (
self.mp.param["pre_filter_stop"] - self.mp.param["pre_filter_start"]
)
input_high_end = X_spec_s[d][
:, bp["n_fft"] // 2 - input_high_end_h : bp["n_fft"] // 2, :
]
X_spec_m = spec_utils.combine_spectrograms(X_spec_s, self.mp)
aggresive_set = float(self.data["agg"] / 100)
aggressiveness = {
"value": aggresive_set,
"split_bin": self.mp.param["band"][1]["crop_stop"],
}
with torch.no_grad():
pred, X_mag, X_phase = inference(
X_spec_m, self.device, self.model, aggressiveness, self.data
)
# Postprocess
if self.data["postprocess"]:
pred_inv = np.clip(X_mag - pred, 0, np.inf)
pred = spec_utils.mask_silence(pred, pred_inv)
y_spec_m = pred * X_phase
v_spec_m = X_spec_m - y_spec_m
if ins_root is not None:
if self.data["high_end_process"].startswith("mirroring"):
input_high_end_ = spec_utils.mirroring(
self.data["high_end_process"], y_spec_m, input_high_end, self.mp
)
wav_instrument = spec_utils.cmb_spectrogram_to_wave(
y_spec_m, self.mp, input_high_end_h, input_high_end_
)
else:
wav_instrument = spec_utils.cmb_spectrogram_to_wave(y_spec_m, self.mp)
print("%s instruments done" % name)
if format in ["wav", "flac"]:
sf.write(
os.path.join(
ins_root,
"instrument_{}_{}.{}".format(name, self.data["agg"], format),
),
(np.array(wav_instrument) * 32768).astype("int16"),
self.mp.param["sr"],
) #
else:
path = os.path.join(
ins_root, "instrument_{}_{}.wav".format(name, self.data["agg"])
)
sf.write(
path,
(np.array(wav_instrument) * 32768).astype("int16"),
self.mp.param["sr"],
)
if os.path.exists(path):
os.system(
"ffmpeg -i %s -vn %s -q:a 2 -y"
% (path, path[:-4] + ".%s" % format)
)
if vocal_root is not None:
if self.data["high_end_process"].startswith("mirroring"):
input_high_end_ = spec_utils.mirroring(
self.data["high_end_process"], v_spec_m, input_high_end, self.mp
)
wav_vocals = spec_utils.cmb_spectrogram_to_wave(
v_spec_m, self.mp, input_high_end_h, input_high_end_
)
else:
wav_vocals = spec_utils.cmb_spectrogram_to_wave(v_spec_m, self.mp)
print("%s vocals done" % name)
if format in ["wav", "flac"]:
sf.write(
os.path.join(
vocal_root,
"vocal_{}_{}.{}".format(name, self.data["agg"], format),
),
(np.array(wav_vocals) * 32768).astype("int16"),
self.mp.param["sr"],
)
else:
path = os.path.join(
vocal_root, "vocal_{}_{}.wav".format(name, self.data["agg"])
)
sf.write(
path,
(np.array(wav_vocals) * 32768).astype("int16"),
self.mp.param["sr"],
)
if os.path.exists(path):
os.system(
"ffmpeg -i %s -vn %s -q:a 2 -y"
% (path, path[:-4] + ".%s" % format)
)
if __name__ == "__main__":
device = "cuda"
is_half = True
# model_path = "uvr5_weights/2_HP-UVR.pth"
# model_path = "uvr5_weights/VR-DeEchoDeReverb.pth"
# model_path = "uvr5_weights/VR-DeEchoNormal.pth"
model_path = "uvr5_weights/DeEchoNormal.pth"
# pre_fun = _audio_pre_(model_path=model_path, device=device, is_half=True,agg=10)
pre_fun = _audio_pre_new(model_path=model_path, device=device, is_half=True, agg=10)
audio_path = "雪雪伴奏对消HP5.wav"
save_path = "opt"
pre_fun._path_audio_(audio_path, save_path, save_path)
File diff suppressed because it is too large Load Diff
@@ -0,0 +1,62 @@
[tool.poetry]
name = "rvc-beta"
version = "0.1.0"
description = ""
authors = ["lj1995"]
license = "MIT"
[tool.poetry.dependencies]
python = "^3.8"
torch = "^2.0.0"
torchaudio = "^2.0.1"
Cython = "^0.29.34"
gradio = "^3.34.0"
future = "^0.18.3"
pydub = "^0.25.1"
soundfile = "^0.12.1"
ffmpeg-python = "^0.2.0"
tensorboardX = "^2.6"
functorch = "^2.0.0"
fairseq = "^0.12.2"
faiss-cpu = "^1.7.2"
Jinja2 = "^3.1.2"
json5 = "^0.9.11"
librosa = "0.9.1"
llvmlite = "0.39.0"
Markdown = "^3.4.3"
matplotlib = "^3.7.1"
matplotlib-inline = "^0.1.6"
numba = "0.56.4"
numpy = "1.23.5"
scipy = "1.9.3"
praat-parselmouth = "^0.4.3"
Pillow = "9.3.0"
pyworld = "^0.3.2"
resampy = "^0.4.2"
scikit-learn = "^1.2.2"
starlette = "^0.27.0"
tensorboard = "^2.12.1"
tensorboard-data-server = "^0.7.0"
tensorboard-plugin-wit = "^1.8.1"
torchgen = "^0.0.1"
tqdm = "^4.65.0"
tornado = "^6.3"
Werkzeug = "^2.2.3"
uc-micro-py = "^1.0.1"
sympy = "^1.11.1"
tabulate = "^0.9.0"
PyYAML = "^6.0"
pyasn1 = "^0.4.8"
pyasn1-modules = "^0.2.8"
fsspec = "^2023.3.0"
absl-py = "^1.4.0"
audioread = "^3.0.0"
uvicorn = "^0.21.1"
colorama = "^0.4.6"
torchcrepe = "0.0.20"
[tool.poetry.dev-dependencies]
[build-system]
requires = ["poetry-core>=1.0.0"]
build-backend = "poetry.core.masonry.api"
@@ -0,0 +1,44 @@
joblib>=1.1.0
numba==0.56.4
numpy==1.23.5
scipy==1.9.3
librosa==0.9.1
llvmlite==0.39.0
fairseq==0.12.2
faiss-cpu==1.7.3
gradio==3.14.0
Cython
pydub>=0.25.1
soundfile>=0.12.1
ffmpeg-python>=0.2.0
tensorboardX
Jinja2>=3.1.2
json5
Markdown
matplotlib>=3.7.0
matplotlib-inline>=0.1.3
praat-parselmouth>=0.4.2
Pillow>=9.1.1
resampy>=0.4.2
scikit-learn
tensorboard
tqdm>=4.63.1
tornado>=6.1
Werkzeug>=2.2.3
uc-micro-py>=1.0.1
sympy>=1.11.1
tabulate>=0.8.10
PyYAML>=6.0
pyasn1>=0.4.8
pyasn1-modules>=0.2.8
fsspec>=2022.11.0
absl-py>=1.2.0
audioread
uvicorn>=0.21.1
colorama>=0.4.5
pyworld==0.3.2
httpx==0.23.0
onnxruntime-directml
torchcrepe==0.0.20
fastapi==0.88
ffmpy==0.3.1
@@ -0,0 +1,29 @@
#1.Install torch from pytorch.org:
#torch 2.0 with cuda 11.8
#pip3 install torch torchvision torchaudio --index-url https://download.pytorch.org/whl/cu118
#torch 1.11.0 with cuda 11.3
#pip install torch==1.11.0+cu113 torchvision==0.12.0+cu113 torchaudio==0.11.0 --extra-index-url https://download.pytorch.org/whl/cu113
einops
fairseq
flask
flask_cors
gin
gin_config
librosa
local_attention
matplotlib
praat-parselmouth
pyworld
PyYAML
resampy
scikit_learn
scipy
SoundFile
tensorboard
tqdm
wave
PySimpleGUI
sounddevice
gradio
noisereduce
onnxruntime-directml
@@ -0,0 +1,28 @@
#1.Install torch from pytorch.org:
#torch 2.0 with cuda 11.8
#pip3 install torch torchvision torchaudio --index-url https://download.pytorch.org/whl/cu118
#torch 1.11.0 with cuda 11.3
#pip install torch==1.11.0+cu113 torchvision==0.12.0+cu113 torchaudio==0.11.0 --extra-index-url https://download.pytorch.org/whl/cu113
einops
fairseq
flask
flask_cors
gin
gin_config
librosa
local_attention
matplotlib
praat-parselmouth
pyworld
PyYAML
resampy
scikit_learn
scipy
SoundFile
tensorboard
tqdm
wave
PySimpleGUI
sounddevice
gradio
noisereduce
@@ -0,0 +1,45 @@
joblib>=1.1.0
numba==0.56.4
numpy==1.23.5
scipy==1.9.3
librosa==0.9.1
llvmlite==0.39.0
fairseq==0.12.2
faiss-cpu==1.7.3
gradio==3.14.0
Cython
pydub>=0.25.1
soundfile>=0.12.1
ffmpeg-python>=0.2.0
tensorboardX
Jinja2>=3.1.2
json5
Markdown
matplotlib>=3.7.0
matplotlib-inline>=0.1.3
praat-parselmouth>=0.4.2
Pillow>=9.1.1
resampy>=0.4.2
scikit-learn
tensorboard
tqdm>=4.63.1
tornado>=6.1
Werkzeug>=2.2.3
uc-micro-py>=1.0.1
sympy>=1.11.1
tabulate>=0.8.10
PyYAML>=6.0
pyasn1>=0.4.8
pyasn1-modules>=0.2.8
fsspec>=2022.11.0
absl-py>=1.2.0
audioread
uvicorn>=0.21.1
colorama>=0.4.5
pyworld==0.3.2
httpx==0.23.0
onnxruntime; sys_platform == 'darwin'
onnxruntime-gpu; sys_platform != 'darwin'
torchcrepe==0.0.20
fastapi==0.88
ffmpy==0.3.1
+45
View File
@@ -0,0 +1,45 @@
#!/bin/bash
if [[ "$(uname)" == "Darwin" ]]; then
# macOS specific env:
export PYTORCH_ENABLE_MPS_FALLBACK=1
export PYTORCH_MPS_HIGH_WATERMARK_RATIO=0.0
elif [[ "$(uname)" != "Linux" ]]; then
echo "Unsupported operating system."
exit 1
fi
requirements_file="requirements.txt"
# Check if Python 3.8 is installed
if ! command -v python3.8 &> /dev/null; then
echo "Python 3.8 not found. Attempting to install..."
if [[ "$(uname)" == "Darwin" ]] && command -v brew &> /dev/null; then
brew install python@3.8
elif [[ "$(uname)" == "Linux" ]] && command -v apt-get &> /dev/null; then
sudo apt-get update
sudo apt-get install python3.8
else
echo "Please install Python 3.8 manually."
exit 1
fi
fi
# Check if required packages are installed and install them if not
if [ -f "${requirements_file}" ]; then
installed_packages=$(python3.8 -m pip freeze)
while IFS= read -r package; do
[[ "${package}" =~ ^#.* ]] && continue
package_name=$(echo "${package}" | sed 's/[<>=!].*//')
if ! echo "${installed_packages}" | grep -q "${package_name}"; then
echo "${package_name} not found. Attempting to install..."
python3.8 -m pip install --upgrade "${package}"
fi
done < "${requirements_file}"
else
echo "${requirements_file} not found. Please ensure the requirements file with required packages exists."
exit 1
fi
# Run the main script
python3.8 infer-web.py --pycmd python3.8
@@ -0,0 +1,312 @@
import os, sys
import faiss, torch, traceback, parselmouth, numpy as np, torchcrepe, torch.nn as nn, pyworld
import fairseq
from lib.infer_pack.models import (
SynthesizerTrnMs256NSFsid,
SynthesizerTrnMs256NSFsid_nono,
SynthesizerTrnMs768NSFsid,
SynthesizerTrnMs768NSFsid_nono,
)
from time import time as ttime
import torch.nn.functional as F
import scipy.signal as signal
now_dir = os.getcwd()
sys.path.append(now_dir)
from config import Config
from multiprocessing import Manager as M
mm = M()
config = Config()
if(config.dml==True):
def forward_dml(ctx, x, scale):
ctx.scale = scale
res = x.clone().detach()
return res
fairseq.modules.grad_multiply.GradMultiply.forward=forward_dml
# config.device=torch.device("cpu")########强制cpu测试
# config.is_half=False########强制cpu测试
class RVC:
def __init__(
self, key, pth_path, index_path, index_rate, n_cpu, inp_q, opt_q, device
) -> None:
"""
初始化
"""
try:
global config
self.inp_q = inp_q
self.opt_q = opt_q
# device="cpu"########强制cpu测试
self.device = device
self.f0_up_key = key
self.time_step = 160 / 16000 * 1000
self.f0_min = 50
self.f0_max = 1100
self.f0_mel_min = 1127 * np.log(1 + self.f0_min / 700)
self.f0_mel_max = 1127 * np.log(1 + self.f0_max / 700)
self.sr = 16000
self.window = 160
self.n_cpu = n_cpu
if index_rate != 0:
self.index = faiss.read_index(index_path)
self.big_npy = self.index.reconstruct_n(0, self.index.ntotal)
print("index search enabled")
self.index_rate = index_rate
models, _, _ = fairseq.checkpoint_utils.load_model_ensemble_and_task(
["hubert_base.pt"],
suffix="",
)
hubert_model = models[0]
hubert_model = hubert_model.to(config.device)
if config.is_half:
hubert_model = hubert_model.half()
else:
hubert_model = hubert_model.float()
hubert_model.eval()
self.model = hubert_model
cpt = torch.load(pth_path, map_location="cpu")
self.tgt_sr = cpt["config"][-1]
cpt["config"][-3] = cpt["weight"]["emb_g.weight"].shape[0]
self.if_f0 = cpt.get("f0", 1)
self.version = cpt.get("version", "v1")
if self.version == "v1":
if self.if_f0 == 1:
self.net_g = SynthesizerTrnMs256NSFsid(
*cpt["config"], is_half=config.is_half
)
else:
self.net_g = SynthesizerTrnMs256NSFsid_nono(*cpt["config"])
elif self.version == "v2":
if self.if_f0 == 1:
self.net_g = SynthesizerTrnMs768NSFsid(
*cpt["config"], is_half=config.is_half
)
else:
self.net_g = SynthesizerTrnMs768NSFsid_nono(*cpt["config"])
del self.net_g.enc_q
print(self.net_g.load_state_dict(cpt["weight"], strict=False))
self.net_g.eval().to(device)
# print(2333333333,device,config.device,self.device)#net_g是device,hubert是config.device
if config.is_half:
self.net_g = self.net_g.half()
else:
self.net_g = self.net_g.float()
self.is_half = config.is_half
except:
print(traceback.format_exc())
def get_f0_post(self, f0):
f0_min = self.f0_min
f0_max = self.f0_max
f0_mel_min = 1127 * np.log(1 + f0_min / 700)
f0_mel_max = 1127 * np.log(1 + f0_max / 700)
f0bak = f0.copy()
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.int32)
return f0_coarse, f0bak
def get_f0(self, x, f0_up_key, n_cpu, method="harvest"):
n_cpu = int(n_cpu)
if method == "crepe":
return self.get_f0_crepe(x, f0_up_key)
if method == "rmvpe":
return self.get_f0_rmvpe(x, f0_up_key)
if method == "pm":
p_len = x.shape[0] // 160
f0 = (
parselmouth.Sound(x, 16000)
.to_pitch_ac(
time_step=0.01,
voicing_threshold=0.6,
pitch_floor=50,
pitch_ceiling=1100,
)
.selected_array["frequency"]
)
pad_size = (p_len - len(f0) + 1) // 2
if pad_size > 0 or p_len - len(f0) - pad_size > 0:
# print(pad_size, p_len - len(f0) - pad_size)
f0 = np.pad(
f0, [[pad_size, p_len - len(f0) - pad_size]], mode="constant"
)
f0 *= pow(2, f0_up_key / 12)
return self.get_f0_post(f0)
if n_cpu == 1:
f0, t = pyworld.harvest(
x.astype(np.double),
fs=16000,
f0_ceil=1100,
f0_floor=50,
frame_period=10,
)
f0 = signal.medfilt(f0, 3)
f0 *= pow(2, f0_up_key / 12)
return self.get_f0_post(f0)
f0bak = np.zeros(x.shape[0] // 160, dtype=np.float64)
length = len(x)
part_length = int(length / n_cpu / 160) * 160
ts = ttime()
res_f0 = mm.dict()
for idx in range(n_cpu):
tail = part_length * (idx + 1) + 320
if idx == 0:
self.inp_q.put((idx, x[:tail], res_f0, n_cpu, ts))
else:
self.inp_q.put(
(idx, x[part_length * idx - 320 : tail], res_f0, n_cpu, ts)
)
while 1:
res_ts = self.opt_q.get()
if res_ts == ts:
break
f0s = [i[1] for i in sorted(res_f0.items(), key=lambda x: x[0])]
for idx, f0 in enumerate(f0s):
if idx == 0:
f0 = f0[:-3]
elif idx != n_cpu - 1:
f0 = f0[2:-3]
else:
f0 = f0[2:-1]
f0bak[
part_length * idx // 160 : part_length * idx // 160 + f0.shape[0]
] = f0
f0bak = signal.medfilt(f0bak, 3)
f0bak *= pow(2, f0_up_key / 12)
return self.get_f0_post(f0bak)
def get_f0_crepe(self, x, f0_up_key):
if self.device.type == "privateuseone":###不支持dml,cpu又太慢用不成,拿pm顶替
return self.get_f0(x, f0_up_key,1,"pm")
audio = torch.tensor(np.copy(x))[None].float()
# print("using crepe,device:%s"%self.device)
f0, pd = torchcrepe.predict(
audio,
self.sr,
160,
self.f0_min,
self.f0_max,
"full",
batch_size=512,
# device=self.device if self.device.type!="privateuseone" else "cpu",###crepe不用半精度全部是全精度所以不愁###cpu延迟高到没法用
device=self.device,
return_periodicity=True,
)
pd = torchcrepe.filter.median(pd, 3)
f0 = torchcrepe.filter.mean(f0, 3)
f0[pd < 0.1] = 0
f0 = f0[0].cpu().numpy()
f0 *= pow(2, f0_up_key / 12)
return self.get_f0_post(f0)
def get_f0_rmvpe(self, x, f0_up_key):
if hasattr(self, "model_rmvpe") == False:
from lib.rmvpe import RMVPE
print("loading rmvpe model")
self.model_rmvpe = RMVPE(
# "rmvpe.pt", is_half=self.is_half if self.device.type!="privateuseone" else False, device=self.device if self.device.type!="privateuseone"else "cpu"####dml时强制对rmvpe用cpu跑
# "rmvpe.pt", is_half=False, device=self.device####dml配置
# "rmvpe.pt", is_half=False, device="cpu"####锁定cpu配置
"rmvpe.pt", is_half=self.is_half, device=self.device####正常逻辑
)
# self.model_rmvpe = RMVPE("aug2_58000_half.pt", is_half=self.is_half, device=self.device)
f0 = self.model_rmvpe.infer_from_audio(x, thred=0.03)
f0 *= pow(2, f0_up_key / 12)
return self.get_f0_post(f0)
def infer(
self,
feats: torch.Tensor,
indata: np.ndarray,
rate1,
rate2,
cache_pitch,
cache_pitchf,
f0method,
) -> np.ndarray:
feats = feats.view(1, -1)
if config.is_half:
feats = feats.half()
else:
feats = feats.float()
feats = feats.to(self.device)
t1 = ttime()
with torch.no_grad():
padding_mask = torch.BoolTensor(feats.shape).to(self.device).fill_(False)
inputs = {
"source": feats,
"padding_mask": padding_mask,
"output_layer": 9 if self.version == "v1" else 12,
}
logits = self.model.extract_features(**inputs)
feats = (
self.model.final_proj(logits[0]) if self.version == "v1" else logits[0]
)
t2 = ttime()
try:
if hasattr(self, "index") and self.index_rate != 0:
leng_replace_head = int(rate1 * feats[0].shape[0])
npy = feats[0][-leng_replace_head:].cpu().numpy().astype("float32")
score, ix = self.index.search(npy, k=8)
weight = np.square(1 / score)
weight /= weight.sum(axis=1, keepdims=True)
npy = np.sum(self.big_npy[ix] * np.expand_dims(weight, axis=2), axis=1)
if config.is_half:
npy = npy.astype("float16")
feats[0][-leng_replace_head:] = (
torch.from_numpy(npy).unsqueeze(0).to(self.device) * self.index_rate
+ (1 - self.index_rate) * feats[0][-leng_replace_head:]
)
else:
print("index search FAIL or disabled")
except:
traceback.print_exc()
print("index search FAIL")
feats = F.interpolate(feats.permute(0, 2, 1), scale_factor=2).permute(0, 2, 1)
t3 = ttime()
if self.if_f0 == 1:
pitch, pitchf = self.get_f0(indata, self.f0_up_key, self.n_cpu, f0method)
cache_pitch[:] = np.append(cache_pitch[pitch[:-1].shape[0] :], pitch[:-1])
cache_pitchf[:] = np.append(
cache_pitchf[pitchf[:-1].shape[0] :], pitchf[:-1]
)
p_len = min(feats.shape[1], 13000, cache_pitch.shape[0])
else:
cache_pitch, cache_pitchf = None, None
p_len = min(feats.shape[1], 13000)
t4 = ttime()
feats = feats[:, :p_len, :]
if self.if_f0 == 1:
cache_pitch = cache_pitch[:p_len]
cache_pitchf = cache_pitchf[:p_len]
cache_pitch = torch.LongTensor(cache_pitch).unsqueeze(0).to(self.device)
cache_pitchf = torch.FloatTensor(cache_pitchf).unsqueeze(0).to(self.device)
p_len = torch.LongTensor([p_len]).to(self.device)
ii = 0 # sid
sid = torch.LongTensor([ii]).to(self.device)
with torch.no_grad():
if self.if_f0 == 1:
# print(12222222222,feats.device,p_len.device,cache_pitch.device,cache_pitchf.device,sid.device,rate2)
infered_audio = (
self.net_g.infer(
feats, p_len, cache_pitch, cache_pitchf, sid, rate2
)[0][0, 0]
.data.cpu()
.float()
)
else:
infered_audio = (
self.net_g.infer(feats, p_len, sid, rate2)[0][0, 0]
.data.cpu()
.float()
)
t5 = ttime()
print("time->fea-index-f0-model:", t2 - t1, t3 - t2, t4 - t3, t5 - t4)
return infered_audio
@@ -0,0 +1,600 @@
import os, sys
now_dir = os.getcwd()
sys.path.append(os.path.join(now_dir))
from lib.train import utils
import datetime
hps = utils.get_hparams()
os.environ["CUDA_VISIBLE_DEVICES"] = hps.gpus.replace("-", ",")
n_gpus = len(hps.gpus.split("-"))
from random import shuffle, randint
import torch
torch.backends.cudnn.deterministic = False
torch.backends.cudnn.benchmark = False
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
from lib.infer_pack import commons
from time import sleep
from time import time as ttime
from lib.train.data_utils import (
TextAudioLoaderMultiNSFsid,
TextAudioLoader,
TextAudioCollateMultiNSFsid,
TextAudioCollate,
DistributedBucketSampler,
)
if hps.version == "v1":
from lib.infer_pack.models import (
SynthesizerTrnMs256NSFsid as RVC_Model_f0,
SynthesizerTrnMs256NSFsid_nono as RVC_Model_nof0,
MultiPeriodDiscriminator,
)
else:
from lib.infer_pack.models import (
SynthesizerTrnMs768NSFsid as RVC_Model_f0,
SynthesizerTrnMs768NSFsid_nono as RVC_Model_nof0,
MultiPeriodDiscriminatorV2 as MultiPeriodDiscriminator,
)
from lib.train.losses import generator_loss, discriminator_loss, feature_loss, kl_loss
from lib.train.mel_processing import mel_spectrogram_torch, spec_to_mel_torch
from lib.train.process_ckpt import savee
global_step = 0
class EpochRecorder:
def __init__(self):
self.last_time = ttime()
def record(self):
now_time = ttime()
elapsed_time = now_time - self.last_time
self.last_time = now_time
elapsed_time_str = str(datetime.timedelta(seconds=elapsed_time))
current_time = datetime.datetime.now().strftime("%Y-%m-%d %H:%M:%S")
return f"[{current_time}] | ({elapsed_time_str})"
def main():
n_gpus = torch.cuda.device_count()
if torch.cuda.is_available() == False and torch.backends.mps.is_available() == True:
n_gpus = 1
if n_gpus < 1:
# patch to unblock people without gpus. there is probably a better way.
print("NO GPU DETECTED: falling back to CPU - this may take a while")
n_gpus = 1
os.environ["MASTER_ADDR"] = "localhost"
os.environ["MASTER_PORT"] = str(randint(20000, 55555))
children = []
for i in range(n_gpus):
subproc = mp.Process(
target=run,
args=(
i,
n_gpus,
hps,
),
)
children.append(subproc)
subproc.start()
for i in range(n_gpus):
children[i].join()
def run(rank, n_gpus, hps):
global global_step
if rank == 0:
logger = utils.get_logger(hps.model_dir)
logger.info(hps)
# utils.check_git_hash(hps.model_dir)
writer = SummaryWriter(log_dir=hps.model_dir)
writer_eval = SummaryWriter(log_dir=os.path.join(hps.model_dir, "eval"))
dist.init_process_group(
backend="gloo", init_method="env://", world_size=n_gpus, rank=rank
)
torch.manual_seed(hps.train.seed)
if torch.cuda.is_available():
torch.cuda.set_device(rank)
if hps.if_f0 == 1:
train_dataset = TextAudioLoaderMultiNSFsid(hps.data.training_files, hps.data)
else:
train_dataset = TextAudioLoader(hps.data.training_files, hps.data)
train_sampler = DistributedBucketSampler(
train_dataset,
hps.train.batch_size * n_gpus,
# [100, 200, 300, 400, 500, 600, 700, 800, 900, 1000, 1200,1400], # 16s
[100, 200, 300, 400, 500, 600, 700, 800, 900], # 16s
num_replicas=n_gpus,
rank=rank,
shuffle=True,
)
# It is possible that dataloader's workers are out of shared memory. Please try to raise your shared memory limit.
# num_workers=8 -> num_workers=4
if hps.if_f0 == 1:
collate_fn = TextAudioCollateMultiNSFsid()
else:
collate_fn = TextAudioCollate()
train_loader = DataLoader(
train_dataset,
num_workers=4,
shuffle=False,
pin_memory=True,
collate_fn=collate_fn,
batch_sampler=train_sampler,
persistent_workers=True,
prefetch_factor=8,
)
if hps.if_f0 == 1:
net_g = RVC_Model_f0(
hps.data.filter_length // 2 + 1,
hps.train.segment_size // hps.data.hop_length,
**hps.model,
is_half=hps.train.fp16_run,
sr=hps.sample_rate,
)
else:
net_g = RVC_Model_nof0(
hps.data.filter_length // 2 + 1,
hps.train.segment_size // hps.data.hop_length,
**hps.model,
is_half=hps.train.fp16_run,
)
if torch.cuda.is_available():
net_g = net_g.cuda(rank)
net_d = MultiPeriodDiscriminator(hps.model.use_spectral_norm)
if torch.cuda.is_available():
net_d = net_d.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], find_unused_parameters=True)
if torch.cuda.is_available():
net_g = DDP(net_g, device_ids=[rank])
net_d = DDP(net_d, device_ids=[rank])
else:
net_g = DDP(net_g)
net_d = DDP(net_d)
try: # 如果能加载自动resume
_, _, _, epoch_str = utils.load_checkpoint(
utils.latest_checkpoint_path(hps.model_dir, "D_*.pth"), net_d, optim_d
) # D多半加载没事
if rank == 0:
logger.info("loaded D")
# _, _, _, epoch_str = utils.load_checkpoint(utils.latest_checkpoint_path(hps.model_dir, "G_*.pth"), net_g, optim_g,load_opt=0)
_, _, _, epoch_str = utils.load_checkpoint(
utils.latest_checkpoint_path(hps.model_dir, "G_*.pth"), net_g, optim_g
)
global_step = (epoch_str - 1) * len(train_loader)
# epoch_str = 1
# global_step = 0
except: # 如果首次不能加载,加载pretrain
# traceback.print_exc()
epoch_str = 1
global_step = 0
if hps.pretrainG != "":
if rank == 0:
logger.info("loaded pretrained %s" % (hps.pretrainG))
print(
net_g.module.load_state_dict(
torch.load(hps.pretrainG, map_location="cpu")["model"]
)
) ##测试不加载优化器
if hps.pretrainD != "":
if rank == 0:
logger.info("loaded pretrained %s" % (hps.pretrainD))
print(
net_d.module.load_state_dict(
torch.load(hps.pretrainD, map_location="cpu")["model"]
)
)
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)
cache = []
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, None],
logger,
[writer, writer_eval],
cache,
)
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,
cache,
)
scheduler_g.step()
scheduler_d.step()
def train_and_evaluate(
rank, epoch, hps, nets, optims, schedulers, scaler, loaders, logger, writers, cache
):
net_g, net_d = nets
optim_g, optim_d = optims
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()
# Prepare data iterator
if hps.if_cache_data_in_gpu == True:
# Use Cache
data_iterator = cache
if cache == []:
# Make new cache
for batch_idx, info in enumerate(train_loader):
# Unpack
if hps.if_f0 == 1:
(
phone,
phone_lengths,
pitch,
pitchf,
spec,
spec_lengths,
wave,
wave_lengths,
sid,
) = info
else:
(
phone,
phone_lengths,
spec,
spec_lengths,
wave,
wave_lengths,
sid,
) = info
# Load on CUDA
if torch.cuda.is_available():
phone = phone.cuda(rank, non_blocking=True)
phone_lengths = phone_lengths.cuda(rank, non_blocking=True)
if hps.if_f0 == 1:
pitch = pitch.cuda(rank, non_blocking=True)
pitchf = pitchf.cuda(rank, non_blocking=True)
sid = sid.cuda(rank, non_blocking=True)
spec = spec.cuda(rank, non_blocking=True)
spec_lengths = spec_lengths.cuda(rank, non_blocking=True)
wave = wave.cuda(rank, non_blocking=True)
wave_lengths = wave_lengths.cuda(rank, non_blocking=True)
# Cache on list
if hps.if_f0 == 1:
cache.append(
(
batch_idx,
(
phone,
phone_lengths,
pitch,
pitchf,
spec,
spec_lengths,
wave,
wave_lengths,
sid,
),
)
)
else:
cache.append(
(
batch_idx,
(
phone,
phone_lengths,
spec,
spec_lengths,
wave,
wave_lengths,
sid,
),
)
)
else:
# Load shuffled cache
shuffle(cache)
else:
# Loader
data_iterator = enumerate(train_loader)
# Run steps
epoch_recorder = EpochRecorder()
for batch_idx, info in data_iterator:
# Data
## Unpack
if hps.if_f0 == 1:
(
phone,
phone_lengths,
pitch,
pitchf,
spec,
spec_lengths,
wave,
wave_lengths,
sid,
) = info
else:
phone, phone_lengths, spec, spec_lengths, wave, wave_lengths, sid = info
## Load on CUDA
if (hps.if_cache_data_in_gpu == False) and torch.cuda.is_available():
phone = phone.cuda(rank, non_blocking=True)
phone_lengths = phone_lengths.cuda(rank, non_blocking=True)
if hps.if_f0 == 1:
pitch = pitch.cuda(rank, non_blocking=True)
pitchf = pitchf.cuda(rank, non_blocking=True)
sid = sid.cuda(rank, non_blocking=True)
spec = spec.cuda(rank, non_blocking=True)
spec_lengths = spec_lengths.cuda(rank, non_blocking=True)
wave = wave.cuda(rank, non_blocking=True)
# wave_lengths = wave_lengths.cuda(rank, non_blocking=True)
# Calculate
with autocast(enabled=hps.train.fp16_run):
if hps.if_f0 == 1:
(
y_hat,
ids_slice,
x_mask,
z_mask,
(z, z_p, m_p, logs_p, m_q, logs_q),
) = net_g(phone, phone_lengths, pitch, pitchf, spec, spec_lengths, sid)
else:
(
y_hat,
ids_slice,
x_mask,
z_mask,
(z, z_p, m_p, logs_p, m_q, logs_q),
) = net_g(phone, phone_lengths, spec, spec_lengths, sid)
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_mel = commons.slice_segments(
mel, ids_slice, hps.train.segment_size // hps.data.hop_length
)
with autocast(enabled=False):
y_hat_mel = mel_spectrogram_torch(
y_hat.float().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,
)
if hps.train.fp16_run == True:
y_hat_mel = y_hat_mel.half()
wave = commons.slice_segments(
wave, ids_slice * hps.data.hop_length, hps.train.segment_size
) # slice
# Discriminator
y_d_hat_r, y_d_hat_g, _, _ = net_d(wave, 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
)
optim_d.zero_grad()
scaler.scale(loss_disc).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(wave, y_hat)
with autocast(enabled=False):
loss_mel = F.l1_loss(y_mel, y_hat_mel) * hps.train.c_mel
loss_kl = kl_loss(z_p, logs_q, m_p, logs_p, z_mask) * hps.train.c_kl
loss_fm = feature_loss(fmap_r, fmap_g)
loss_gen, losses_gen = generator_loss(y_d_hat_g)
loss_gen_all = loss_gen + loss_fm + loss_mel + loss_kl
optim_g.zero_grad()
scaler.scale(loss_gen_all).backward()
scaler.unscale_(optim_g)
grad_norm_g = commons.clip_grad_value_(net_g.parameters(), None)
scaler.step(optim_g)
scaler.update()
if rank == 0:
if global_step % hps.train.log_interval == 0:
lr = optim_g.param_groups[0]["lr"]
logger.info(
"Train Epoch: {} [{:.0f}%]".format(
epoch, 100.0 * batch_idx / len(train_loader)
)
)
# Amor For Tensorboard display
if loss_mel > 75:
loss_mel = 75
if loss_kl > 9:
loss_kl = 9
logger.info([global_step, lr])
logger.info(
f"loss_disc={loss_disc:.3f}, loss_gen={loss_gen:.3f}, loss_fm={loss_fm:.3f},loss_mel={loss_mel:.3f}, loss_kl={loss_kl:.3f}"
)
scalar_dict = {
"loss/g/total": loss_gen_all,
"loss/d/total": loss_disc,
"learning_rate": lr,
"grad_norm_d": grad_norm_d,
"grad_norm_g": grad_norm_g,
}
scalar_dict.update(
{
"loss/g/fm": loss_fm,
"loss/g/mel": loss_mel,
"loss/g/kl": loss_kl,
}
)
scalar_dict.update(
{"loss/g/{}".format(i): v for i, v in enumerate(losses_gen)}
)
scalar_dict.update(
{"loss/d_r/{}".format(i): v for i, v in enumerate(losses_disc_r)}
)
scalar_dict.update(
{"loss/d_g/{}".format(i): v for i, v in enumerate(losses_disc_g)}
)
image_dict = {
"slice/mel_org": utils.plot_spectrogram_to_numpy(
y_mel[0].data.cpu().numpy()
),
"slice/mel_gen": utils.plot_spectrogram_to_numpy(
y_hat_mel[0].data.cpu().numpy()
),
"all/mel": utils.plot_spectrogram_to_numpy(
mel[0].data.cpu().numpy()
),
}
utils.summarize(
writer=writer,
global_step=global_step,
images=image_dict,
scalars=scalar_dict,
)
global_step += 1
# /Run steps
if epoch % hps.save_every_epoch == 0 and rank == 0:
if hps.if_latest == 0:
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)),
)
else:
utils.save_checkpoint(
net_g,
optim_g,
hps.train.learning_rate,
epoch,
os.path.join(hps.model_dir, "G_{}.pth".format(2333333)),
)
utils.save_checkpoint(
net_d,
optim_d,
hps.train.learning_rate,
epoch,
os.path.join(hps.model_dir, "D_{}.pth".format(2333333)),
)
if rank == 0 and hps.save_every_weights == "1":
if hasattr(net_g, "module"):
ckpt = net_g.module.state_dict()
else:
ckpt = net_g.state_dict()
logger.info(
"saving ckpt %s_e%s:%s"
% (
hps.name,
epoch,
savee(
ckpt,
hps.sample_rate,
hps.if_f0,
hps.name + "_e%s_s%s" % (epoch, global_step),
epoch,
hps.version,
hps,
),
)
)
if rank == 0:
logger.info("====> Epoch: {} {}".format(epoch, epoch_recorder.record()))
if epoch >= hps.total_epoch and rank == 0:
logger.info("Training is done. The program is closed.")
if hasattr(net_g, "module"):
ckpt = net_g.module.state_dict()
else:
ckpt = net_g.state_dict()
logger.info(
"saving final ckpt:%s"
% (
savee(
ckpt, hps.sample_rate, hps.if_f0, hps.name, epoch, hps.version, hps
)
)
)
sleep(1)
os._exit(2333333)
if __name__ == "__main__":
torch.multiprocessing.set_start_method("spawn")
main()
@@ -0,0 +1,139 @@
import sys, os, multiprocessing
from scipy import signal
now_dir = os.getcwd()
sys.path.append(now_dir)
print(sys.argv)
inp_root = sys.argv[1]
sr = int(sys.argv[2])
n_p = int(sys.argv[3])
exp_dir = sys.argv[4]
noparallel = sys.argv[5] == "True"
import numpy as np, os, traceback
from lib.slicer2 import Slicer
import librosa, traceback
from scipy.io import wavfile
import multiprocessing
from lib.audio import load_audio
mutex = multiprocessing.Lock()
f = open("%s/preprocess.log" % exp_dir, "a+")
def println(strr):
mutex.acquire()
print(strr)
f.write("%s\n" % strr)
f.flush()
mutex.release()
class PreProcess:
def __init__(self, sr, exp_dir):
self.slicer = Slicer(
sr=sr,
threshold=-42,
min_length=1500,
min_interval=400,
hop_size=15,
max_sil_kept=500,
)
self.sr = sr
self.bh, self.ah = signal.butter(N=5, Wn=48, btype="high", fs=self.sr)
self.per = 3.0
self.overlap = 0.3
self.tail = self.per + self.overlap
self.max = 0.9
self.alpha = 0.75
self.exp_dir = exp_dir
self.gt_wavs_dir = "%s/0_gt_wavs" % exp_dir
self.wavs16k_dir = "%s/1_16k_wavs" % exp_dir
os.makedirs(self.exp_dir, exist_ok=True)
os.makedirs(self.gt_wavs_dir, exist_ok=True)
os.makedirs(self.wavs16k_dir, exist_ok=True)
def norm_write(self, tmp_audio, idx0, idx1):
tmp_max = np.abs(tmp_audio).max()
if tmp_max > 2.5:
print("%s-%s-%s-filtered" % (idx0, idx1, tmp_max))
return
tmp_audio = (tmp_audio / tmp_max * (self.max * self.alpha)) + (
1 - self.alpha
) * tmp_audio
wavfile.write(
"%s/%s_%s.wav" % (self.gt_wavs_dir, idx0, idx1),
self.sr,
tmp_audio.astype(np.float32),
)
tmp_audio = librosa.resample(
tmp_audio, orig_sr=self.sr, target_sr=16000
) # , res_type="soxr_vhq"
wavfile.write(
"%s/%s_%s.wav" % (self.wavs16k_dir, idx0, idx1),
16000,
tmp_audio.astype(np.float32),
)
def pipeline(self, path, idx0):
try:
audio = load_audio(path, self.sr)
# zero phased digital filter cause pre-ringing noise...
# audio = signal.filtfilt(self.bh, self.ah, audio)
audio = signal.lfilter(self.bh, self.ah, audio)
idx1 = 0
for audio in self.slicer.slice(audio):
i = 0
while 1:
start = int(self.sr * (self.per - self.overlap) * i)
i += 1
if len(audio[start:]) > self.tail * self.sr:
tmp_audio = audio[start : start + int(self.per * self.sr)]
self.norm_write(tmp_audio, idx0, idx1)
idx1 += 1
else:
tmp_audio = audio[start:]
idx1 += 1
break
self.norm_write(tmp_audio, idx0, idx1)
println("%s->Suc." % path)
except:
println("%s->%s" % (path, traceback.format_exc()))
def pipeline_mp(self, infos):
for path, idx0 in infos:
self.pipeline(path, idx0)
def pipeline_mp_inp_dir(self, inp_root, n_p):
try:
infos = [
("%s/%s" % (inp_root, name), idx)
for idx, name in enumerate(sorted(list(os.listdir(inp_root))))
]
if noparallel:
for i in range(n_p):
self.pipeline_mp(infos[i::n_p])
else:
ps = []
for i in range(n_p):
p = multiprocessing.Process(
target=self.pipeline_mp, args=(infos[i::n_p],)
)
ps.append(p)
p.start()
for i in range(n_p):
ps[i].join()
except:
println("Fail. %s" % traceback.format_exc())
def preprocess_trainset(inp_root, sr, n_p, exp_dir):
pp = PreProcess(sr, exp_dir)
println("start preprocess")
println(sys.argv)
pp.pipeline_mp_inp_dir(inp_root, n_p)
println("end preprocess")
if __name__ == "__main__":
preprocess_trainset(inp_root, sr, n_p, exp_dir)
@@ -0,0 +1 @@
{"pth_path": "weights/keruanV1.pth", "index_path": "logs/kikiV1.index", "sg_input_device": "VoiceMeeter Output (VB-Audio Vo (MME)", "sg_output_device": "VoiceMeeter Aux Input (VB-Audio (MME)", "threhold": -60.0, "pitch": 0.0, "index_rate": 0.0, "block_time": 0.51, "crossfade_length": 0.15, "extra_time": 2.99, "n_cpu": 4.0, "f0method": "rmvpe"}
@@ -0,0 +1,448 @@
import numpy as np, parselmouth, torch, pdb, sys, os
from time import time as ttime
import torch.nn.functional as F
import scipy.signal as signal
import pyworld, os, traceback, faiss, librosa, torchcrepe
from scipy import signal
from functools import lru_cache
now_dir = os.getcwd()
sys.path.append(now_dir)
bh, ah = signal.butter(N=5, Wn=48, btype="high", fs=16000)
input_audio_path2wav = {}
@lru_cache
def cache_harvest_f0(input_audio_path, fs, f0max, f0min, frame_period):
audio = input_audio_path2wav[input_audio_path]
f0, t = pyworld.harvest(
audio,
fs=fs,
f0_ceil=f0max,
f0_floor=f0min,
frame_period=frame_period,
)
f0 = pyworld.stonemask(audio, f0, t, fs)
return f0
def change_rms(data1, sr1, data2, sr2, rate): # 1是输入音频,2是输出音频,rate是2的占比
# print(data1.max(),data2.max())
rms1 = librosa.feature.rms(
y=data1, frame_length=sr1 // 2 * 2, hop_length=sr1 // 2
) # 每半秒一个点
rms2 = librosa.feature.rms(y=data2, frame_length=sr2 // 2 * 2, hop_length=sr2 // 2)
rms1 = torch.from_numpy(rms1)
rms1 = F.interpolate(
rms1.unsqueeze(0), size=data2.shape[0], mode="linear"
).squeeze()
rms2 = torch.from_numpy(rms2)
rms2 = F.interpolate(
rms2.unsqueeze(0), size=data2.shape[0], mode="linear"
).squeeze()
rms2 = torch.max(rms2, torch.zeros_like(rms2) + 1e-6)
data2 *= (
torch.pow(rms1, torch.tensor(1 - rate))
* torch.pow(rms2, torch.tensor(rate - 1))
).numpy()
return data2
class VC(object):
def __init__(self, tgt_sr, config):
self.x_pad, self.x_query, self.x_center, self.x_max, self.is_half = (
config.x_pad,
config.x_query,
config.x_center,
config.x_max,
config.is_half,
)
self.sr = 16000 # hubert输入采样率
self.window = 160 # 每帧点数
self.t_pad = self.sr * self.x_pad # 每条前后pad时间
self.t_pad_tgt = tgt_sr * self.x_pad
self.t_pad2 = self.t_pad * 2
self.t_query = self.sr * self.x_query # 查询切点前后查询时间
self.t_center = self.sr * self.x_center # 查询切点位置
self.t_max = self.sr * self.x_max # 免查询时长阈值
self.device = config.device
def get_f0(
self,
input_audio_path,
x,
p_len,
f0_up_key,
f0_method,
filter_radius,
inp_f0=None,
):
global input_audio_path2wav
time_step = self.window / self.sr * 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)
if f0_method == "pm":
f0 = (
parselmouth.Sound(x, self.sr)
.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"
)
elif f0_method == "harvest":
input_audio_path2wav[input_audio_path] = x.astype(np.double)
f0 = cache_harvest_f0(input_audio_path, self.sr, f0_max, f0_min, 10)
if filter_radius > 2:
f0 = signal.medfilt(f0, 3)
elif f0_method == "crepe":
model = "full"
# Pick a batch size that doesn't cause memory errors on your gpu
batch_size = 512
# Compute pitch using first gpu
audio = torch.tensor(np.copy(x))[None].float()
f0, pd = torchcrepe.predict(
audio,
self.sr,
self.window,
f0_min,
f0_max,
model,
batch_size=batch_size,
device=self.device,
return_periodicity=True,
)
pd = torchcrepe.filter.median(pd, 3)
f0 = torchcrepe.filter.mean(f0, 3)
f0[pd < 0.1] = 0
f0 = f0[0].cpu().numpy()
elif f0_method == "rmvpe":
if hasattr(self, "model_rmvpe") == False:
from lib.rmvpe import RMVPE
print("loading rmvpe model")
self.model_rmvpe = RMVPE(
"rmvpe.pt", is_half=self.is_half, device=self.device
)
f0 = self.model_rmvpe.infer_from_audio(x, thred=0.03)
if("privateuseone"in str(self.device)):#clean ortruntime memory
del self.model_rmvpe.model
del self.model_rmvpe
print("cleaning ortruntime memory")
f0 *= pow(2, f0_up_key / 12)
# with open("test.txt","w")as f:f.write("\n".join([str(i)for i in f0.tolist()]))
tf0 = self.sr // self.window # 每秒f0点数
if inp_f0 is not None:
delta_t = np.round(
(inp_f0[:, 0].max() - inp_f0[:, 0].min()) * tf0 + 1
).astype("int16")
replace_f0 = np.interp(
list(range(delta_t)), inp_f0[:, 0] * 100, inp_f0[:, 1]
)
shape = f0[self.x_pad * tf0 : self.x_pad * tf0 + len(replace_f0)].shape[0]
f0[self.x_pad * tf0 : self.x_pad * tf0 + len(replace_f0)] = replace_f0[
:shape
]
# with open("test_opt.txt","w")as f:f.write("\n".join([str(i)for i in f0.tolist()]))
f0bak = f0.copy()
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.int32)
return f0_coarse, f0bak # 1-0
def vc(
self,
model,
net_g,
sid,
audio0,
pitch,
pitchf,
times,
index,
big_npy,
index_rate,
version,
protect,
): # ,file_index,file_big_npy
feats = torch.from_numpy(audio0)
if self.is_half:
feats = feats.half()
else:
feats = feats.float()
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).to(self.device).fill_(False)
inputs = {
"source": feats.to(self.device),
"padding_mask": padding_mask,
"output_layer": 9 if version == "v1" else 12,
}
t0 = ttime()
with torch.no_grad():
logits = model.extract_features(**inputs)
feats = model.final_proj(logits[0]) if version == "v1" else logits[0]
if protect < 0.5 and pitch != None and pitchf != None:
feats0 = feats.clone()
if (
isinstance(index, type(None)) == False
and isinstance(big_npy, type(None)) == False
and index_rate != 0
):
npy = feats[0].cpu().numpy()
if self.is_half:
npy = npy.astype("float32")
# _, I = index.search(npy, 1)
# npy = big_npy[I.squeeze()]
score, ix = index.search(npy, k=8)
weight = np.square(1 / score)
weight /= weight.sum(axis=1, keepdims=True)
npy = np.sum(big_npy[ix] * np.expand_dims(weight, axis=2), axis=1)
if self.is_half:
npy = npy.astype("float16")
feats = (
torch.from_numpy(npy).unsqueeze(0).to(self.device) * index_rate
+ (1 - index_rate) * feats
)
feats = F.interpolate(feats.permute(0, 2, 1), scale_factor=2).permute(0, 2, 1)
if protect < 0.5 and pitch != None and pitchf != None:
feats0 = F.interpolate(feats0.permute(0, 2, 1), scale_factor=2).permute(
0, 2, 1
)
t1 = ttime()
p_len = audio0.shape[0] // self.window
if feats.shape[1] < p_len:
p_len = feats.shape[1]
if pitch != None and pitchf != None:
pitch = pitch[:, :p_len]
pitchf = pitchf[:, :p_len]
if protect < 0.5 and pitch != None and pitchf != None:
pitchff = pitchf.clone()
pitchff[pitchf > 0] = 1
pitchff[pitchf < 1] = protect
pitchff = pitchff.unsqueeze(-1)
feats = feats * pitchff + feats0 * (1 - pitchff)
feats = feats.to(feats0.dtype)
p_len = torch.tensor([p_len], device=self.device).long()
with torch.no_grad():
if pitch != None and pitchf != None:
audio1 = (
(net_g.infer(feats, p_len, pitch, pitchf, sid)[0][0, 0])
.data.cpu()
.float()
.numpy()
)
else:
audio1 = (
(net_g.infer(feats, p_len, sid)[0][0, 0]).data.cpu().float().numpy()
)
del feats, p_len, padding_mask
if torch.cuda.is_available():
torch.cuda.empty_cache()
t2 = ttime()
times[0] += t1 - t0
times[2] += t2 - t1
return audio1
def pipeline(
self,
model,
net_g,
sid,
audio,
input_audio_path,
times,
f0_up_key,
f0_method,
file_index,
# file_big_npy,
index_rate,
if_f0,
filter_radius,
tgt_sr,
resample_sr,
rms_mix_rate,
version,
protect,
f0_file=None,
):
if (
file_index != ""
# and file_big_npy != ""
# and os.path.exists(file_big_npy) == True
and os.path.exists(file_index) == True
and index_rate != 0
):
try:
index = faiss.read_index(file_index)
# big_npy = np.load(file_big_npy)
big_npy = index.reconstruct_n(0, index.ntotal)
except:
traceback.print_exc()
index = big_npy = None
else:
index = big_npy = None
audio = signal.filtfilt(bh, ah, audio)
audio_pad = np.pad(audio, (self.window // 2, self.window // 2), mode="reflect")
opt_ts = []
if audio_pad.shape[0] > self.t_max:
audio_sum = np.zeros_like(audio)
for i in range(self.window):
audio_sum += audio_pad[i : i - self.window]
for t in range(self.t_center, audio.shape[0], self.t_center):
opt_ts.append(
t
- self.t_query
+ np.where(
np.abs(audio_sum[t - self.t_query : t + self.t_query])
== np.abs(audio_sum[t - self.t_query : t + self.t_query]).min()
)[0][0]
)
s = 0
audio_opt = []
t = None
t1 = ttime()
audio_pad = np.pad(audio, (self.t_pad, self.t_pad), mode="reflect")
p_len = audio_pad.shape[0] // self.window
inp_f0 = None
if hasattr(f0_file, "name") == True:
try:
with open(f0_file.name, "r") as f:
lines = f.read().strip("\n").split("\n")
inp_f0 = []
for line in lines:
inp_f0.append([float(i) for i in line.split(",")])
inp_f0 = np.array(inp_f0, dtype="float32")
except:
traceback.print_exc()
sid = torch.tensor(sid, device=self.device).unsqueeze(0).long()
pitch, pitchf = None, None
if if_f0 == 1:
pitch, pitchf = self.get_f0(
input_audio_path,
audio_pad,
p_len,
f0_up_key,
f0_method,
filter_radius,
inp_f0,
)
pitch = pitch[:p_len]
pitchf = pitchf[:p_len]
if self.device == "mps":
pitchf = pitchf.astype(np.float32)
pitch = torch.tensor(pitch, device=self.device).unsqueeze(0).long()
pitchf = torch.tensor(pitchf, device=self.device).unsqueeze(0).float()
t2 = ttime()
times[1] += t2 - t1
for t in opt_ts:
t = t // self.window * self.window
if if_f0 == 1:
audio_opt.append(
self.vc(
model,
net_g,
sid,
audio_pad[s : t + self.t_pad2 + self.window],
pitch[:, s // self.window : (t + self.t_pad2) // self.window],
pitchf[:, s // self.window : (t + self.t_pad2) // self.window],
times,
index,
big_npy,
index_rate,
version,
protect,
)[self.t_pad_tgt : -self.t_pad_tgt]
)
else:
audio_opt.append(
self.vc(
model,
net_g,
sid,
audio_pad[s : t + self.t_pad2 + self.window],
None,
None,
times,
index,
big_npy,
index_rate,
version,
protect,
)[self.t_pad_tgt : -self.t_pad_tgt]
)
s = t
if if_f0 == 1:
audio_opt.append(
self.vc(
model,
net_g,
sid,
audio_pad[t:],
pitch[:, t // self.window :] if t is not None else pitch,
pitchf[:, t // self.window :] if t is not None else pitchf,
times,
index,
big_npy,
index_rate,
version,
protect,
)[self.t_pad_tgt : -self.t_pad_tgt]
)
else:
audio_opt.append(
self.vc(
model,
net_g,
sid,
audio_pad[t:],
None,
None,
times,
index,
big_npy,
index_rate,
version,
protect,
)[self.t_pad_tgt : -self.t_pad_tgt]
)
audio_opt = np.concatenate(audio_opt)
if rms_mix_rate != 1:
audio_opt = change_rms(audio, 16000, audio_opt, tgt_sr, rms_mix_rate)
if resample_sr >= 16000 and tgt_sr != resample_sr:
audio_opt = librosa.resample(
audio_opt, orig_sr=tgt_sr, target_sr=resample_sr
)
audio_max = np.abs(audio_opt).max() / 0.99
max_int16 = 32768
if audio_max > 1:
max_int16 /= audio_max
audio_opt = (audio_opt * max_int16).astype(np.int16)
del pitch, pitchf, sid
if torch.cuda.is_available():
torch.cuda.empty_cache()
return audio_opt
+1
View File
@@ -0,0 +1 @@
python3 -m venv .venv