From edf3e2aa083b87c30adc12f34a97f90ea5dd2351 Mon Sep 17 00:00:00 2001 From: Justin John <34035011+justinjohn0306@users.noreply.github.com> Date: Fri, 6 Oct 2023 14:20:27 +0530 Subject: [PATCH] Add files via upload --- voice_conversion/rvc-v2-brandon/Dockerfile | 55 + voice_conversion/rvc-v2-brandon/LICENSE | 22 + voice_conversion/rvc-v2-brandon/MDXNet.py | 279 ++ .../rvc-v2-brandon/MIT协议暨相关引用库协议 | 45 + voice_conversion/rvc-v2-brandon/README.md | 92 + ...trieval_based_Voice_Conversion_WebUI.ipynb | 384 ++ ...eval_based_Voice_Conversion_WebUI_v2.ipynb | 404 ++ voice_conversion/rvc-v2-brandon/app.py | 319 ++ voice_conversion/rvc-v2-brandon/config.py | 175 + .../rvc-v2-brandon/environment_dml.yaml | 186 + .../rvc-v2-brandon/extract_f0_print.py | 166 + .../rvc-v2-brandon/extract_f0_rmvpe.py | 132 + .../rvc-v2-brandon/extract_f0_rmvpe_dml.py | 129 + .../rvc-v2-brandon/extract_feature_print.py | 132 + .../rvc-v2-brandon/extract_locale.py | 34 + .../rvc-v2-brandon/fakeyou_infer.py | 276 ++ .../rvc-v2-brandon/fakeyou_rvc_tts_infer.py | 165 + .../rvc-v2-brandon/go-realtime-gui.bat | 2 + voice_conversion/rvc-v2-brandon/go-web.bat | 2 + voice_conversion/rvc-v2-brandon/gui_v1.py | 661 +++ voice_conversion/rvc-v2-brandon/i18n.py | 28 + voice_conversion/rvc-v2-brandon/infer-web.py | 2145 +++++++++ .../rvc-v2-brandon/infer_batch_rvc.py | 215 + voice_conversion/rvc-v2-brandon/infer_cli.py | 272 ++ voice_conversion/rvc-v2-brandon/infer_uvr5.py | 363 ++ voice_conversion/rvc-v2-brandon/poetry.lock | 3871 +++++++++++++++++ .../rvc-v2-brandon/pyproject.toml | 62 + .../rvc-v2-brandon/requirements-dml.txt | 44 + ...quirements-win-for-realtime_vc_gui-dml.txt | 29 + .../requirements-win-for-realtime_vc_gui.txt | 28 + .../rvc-v2-brandon/requirements.txt | 45 + voice_conversion/rvc-v2-brandon/run.sh | 45 + .../rvc-v2-brandon/rvc_for_realtime.py | 312 ++ .../train_nsf_sim_cache_sid_load_pretrain.py | 600 +++ .../trainset_preprocess_pipeline_print.py | 139 + voice_conversion/rvc-v2-brandon/values1.json | 1 + .../rvc-v2-brandon/vc_infer_pipeline.py | 448 ++ voice_conversion/rvc-v2-brandon/venv.sh | 1 + 38 files changed, 12308 insertions(+) create mode 100644 voice_conversion/rvc-v2-brandon/Dockerfile create mode 100644 voice_conversion/rvc-v2-brandon/LICENSE create mode 100644 voice_conversion/rvc-v2-brandon/MDXNet.py create mode 100644 voice_conversion/rvc-v2-brandon/MIT协议暨相关引用库协议 create mode 100644 voice_conversion/rvc-v2-brandon/README.md create mode 100644 voice_conversion/rvc-v2-brandon/Retrieval_based_Voice_Conversion_WebUI.ipynb create mode 100644 voice_conversion/rvc-v2-brandon/Retrieval_based_Voice_Conversion_WebUI_v2.ipynb create mode 100644 voice_conversion/rvc-v2-brandon/app.py create mode 100644 voice_conversion/rvc-v2-brandon/config.py create mode 100644 voice_conversion/rvc-v2-brandon/environment_dml.yaml create mode 100644 voice_conversion/rvc-v2-brandon/extract_f0_print.py create mode 100644 voice_conversion/rvc-v2-brandon/extract_f0_rmvpe.py create mode 100644 voice_conversion/rvc-v2-brandon/extract_f0_rmvpe_dml.py create mode 100644 voice_conversion/rvc-v2-brandon/extract_feature_print.py create mode 100644 voice_conversion/rvc-v2-brandon/extract_locale.py create mode 100644 voice_conversion/rvc-v2-brandon/fakeyou_infer.py create mode 100644 voice_conversion/rvc-v2-brandon/fakeyou_rvc_tts_infer.py create mode 100644 voice_conversion/rvc-v2-brandon/go-realtime-gui.bat create mode 100644 voice_conversion/rvc-v2-brandon/go-web.bat create mode 100644 voice_conversion/rvc-v2-brandon/gui_v1.py create mode 100644 voice_conversion/rvc-v2-brandon/i18n.py create mode 100644 voice_conversion/rvc-v2-brandon/infer-web.py create mode 100644 voice_conversion/rvc-v2-brandon/infer_batch_rvc.py create mode 100644 voice_conversion/rvc-v2-brandon/infer_cli.py create mode 100644 voice_conversion/rvc-v2-brandon/infer_uvr5.py create mode 100644 voice_conversion/rvc-v2-brandon/poetry.lock create mode 100644 voice_conversion/rvc-v2-brandon/pyproject.toml create mode 100644 voice_conversion/rvc-v2-brandon/requirements-dml.txt create mode 100644 voice_conversion/rvc-v2-brandon/requirements-win-for-realtime_vc_gui-dml.txt create mode 100644 voice_conversion/rvc-v2-brandon/requirements-win-for-realtime_vc_gui.txt create mode 100644 voice_conversion/rvc-v2-brandon/requirements.txt create mode 100644 voice_conversion/rvc-v2-brandon/run.sh create mode 100644 voice_conversion/rvc-v2-brandon/rvc_for_realtime.py create mode 100644 voice_conversion/rvc-v2-brandon/train_nsf_sim_cache_sid_load_pretrain.py create mode 100644 voice_conversion/rvc-v2-brandon/trainset_preprocess_pipeline_print.py create mode 100644 voice_conversion/rvc-v2-brandon/values1.json create mode 100644 voice_conversion/rvc-v2-brandon/vc_infer_pipeline.py create mode 100644 voice_conversion/rvc-v2-brandon/venv.sh diff --git a/voice_conversion/rvc-v2-brandon/Dockerfile b/voice_conversion/rvc-v2-brandon/Dockerfile new file mode 100644 index 0000000..f763e09 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/Dockerfile @@ -0,0 +1,55 @@ +# 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 . . \ No newline at end of file diff --git a/voice_conversion/rvc-v2-brandon/LICENSE b/voice_conversion/rvc-v2-brandon/LICENSE new file mode 100644 index 0000000..4bb30b3 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/LICENSE @@ -0,0 +1,22 @@ +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. diff --git a/voice_conversion/rvc-v2-brandon/MDXNet.py b/voice_conversion/rvc-v2-brandon/MDXNet.py new file mode 100644 index 0000000..df2c672 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/MDXNet.py @@ -0,0 +1,279 @@ +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 + +""" diff --git a/voice_conversion/rvc-v2-brandon/MIT协议暨相关引用库协议 b/voice_conversion/rvc-v2-brandon/MIT协议暨相关引用库协议 new file mode 100644 index 0000000..dbb6c6d --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/MIT协议暨相关引用库协议 @@ -0,0 +1,45 @@ +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 diff --git a/voice_conversion/rvc-v2-brandon/README.md b/voice_conversion/rvc-v2-brandon/README.md new file mode 100644 index 0000000..dc0ea30 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/README.md @@ -0,0 +1,92 @@ +# 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" +``` diff --git a/voice_conversion/rvc-v2-brandon/Retrieval_based_Voice_Conversion_WebUI.ipynb b/voice_conversion/rvc-v2-brandon/Retrieval_based_Voice_Conversion_WebUI.ipynb new file mode 100644 index 0000000..2da7bf9 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/Retrieval_based_Voice_Conversion_WebUI.ipynb @@ -0,0 +1,384 @@ +{ + "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": [] + } + ] +} diff --git a/voice_conversion/rvc-v2-brandon/Retrieval_based_Voice_Conversion_WebUI_v2.ipynb b/voice_conversion/rvc-v2-brandon/Retrieval_based_Voice_Conversion_WebUI_v2.ipynb new file mode 100644 index 0000000..c286d69 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/Retrieval_based_Voice_Conversion_WebUI_v2.ipynb @@ -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 +} diff --git a/voice_conversion/rvc-v2-brandon/app.py b/voice_conversion/rvc-v2-brandon/app.py new file mode 100644 index 0000000..9fdd212 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/app.py @@ -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() diff --git a/voice_conversion/rvc-v2-brandon/config.py b/voice_conversion/rvc-v2-brandon/config.py new file mode 100644 index 0000000..83978d8 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/config.py @@ -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 \ No newline at end of file diff --git a/voice_conversion/rvc-v2-brandon/environment_dml.yaml b/voice_conversion/rvc-v2-brandon/environment_dml.yaml new file mode 100644 index 0000000..0fb3f22 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/environment_dml.yaml @@ -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 diff --git a/voice_conversion/rvc-v2-brandon/extract_f0_print.py b/voice_conversion/rvc-v2-brandon/extract_f0_print.py new file mode 100644 index 0000000..4f6c806 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/extract_f0_print.py @@ -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() diff --git a/voice_conversion/rvc-v2-brandon/extract_f0_rmvpe.py b/voice_conversion/rvc-v2-brandon/extract_f0_rmvpe.py new file mode 100644 index 0000000..00ca16c --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/extract_f0_rmvpe.py @@ -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() diff --git a/voice_conversion/rvc-v2-brandon/extract_f0_rmvpe_dml.py b/voice_conversion/rvc-v2-brandon/extract_f0_rmvpe_dml.py new file mode 100644 index 0000000..6a46fbd --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/extract_f0_rmvpe_dml.py @@ -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() diff --git a/voice_conversion/rvc-v2-brandon/extract_feature_print.py b/voice_conversion/rvc-v2-brandon/extract_feature_print.py new file mode 100644 index 0000000..e44ca04 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/extract_feature_print.py @@ -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") diff --git a/voice_conversion/rvc-v2-brandon/extract_locale.py b/voice_conversion/rvc-v2-brandon/extract_locale.py new file mode 100644 index 0000000..460e75f --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/extract_locale.py @@ -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") diff --git a/voice_conversion/rvc-v2-brandon/fakeyou_infer.py b/voice_conversion/rvc-v2-brandon/fakeyou_infer.py new file mode 100644 index 0000000..e6de786 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/fakeyou_infer.py @@ -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) diff --git a/voice_conversion/rvc-v2-brandon/fakeyou_rvc_tts_infer.py b/voice_conversion/rvc-v2-brandon/fakeyou_rvc_tts_infer.py new file mode 100644 index 0000000..f3c45ad --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/fakeyou_rvc_tts_infer.py @@ -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) diff --git a/voice_conversion/rvc-v2-brandon/go-realtime-gui.bat b/voice_conversion/rvc-v2-brandon/go-realtime-gui.bat new file mode 100644 index 0000000..94007e2 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/go-realtime-gui.bat @@ -0,0 +1,2 @@ +runtime\python.exe gui_v1.py --pycmd runtime\python.exe +pause diff --git a/voice_conversion/rvc-v2-brandon/go-web.bat b/voice_conversion/rvc-v2-brandon/go-web.bat new file mode 100644 index 0000000..db1dec5 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/go-web.bat @@ -0,0 +1,2 @@ +runtime\python.exe infer-web.py --pycmd runtime\python.exe --port 7897 +pause diff --git a/voice_conversion/rvc-v2-brandon/gui_v1.py b/voice_conversion/rvc-v2-brandon/gui_v1.py new file mode 100644 index 0000000..3216403 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/gui_v1.py @@ -0,0 +1,661 @@ +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() diff --git a/voice_conversion/rvc-v2-brandon/i18n.py b/voice_conversion/rvc-v2-brandon/i18n.py new file mode 100644 index 0000000..d64f2ea --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/i18n.py @@ -0,0 +1,28 @@ +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) diff --git a/voice_conversion/rvc-v2-brandon/infer-web.py b/voice_conversion/rvc-v2-brandon/infer-web.py new file mode 100644 index 0000000..a162a01 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/infer-web.py @@ -0,0 +1,2145 @@ +import os +import shutil +import sys + +now_dir = os.getcwd() +sys.path.append(now_dir) +import traceback, pdb +import warnings + +import numpy as np +import torch + +os.environ["OPENBLAS_NUM_THREADS"] = "1" +os.environ["no_proxy"] = "localhost, 127.0.0.1, ::1" +import logging +import threading +from random import shuffle +from subprocess import Popen +from time import sleep + +import faiss +import ffmpeg +import gradio as gr +import soundfile as sf +from config import Config +import fairseq +from i18n import I18nAuto +from lib.infer_pack.models import ( + SynthesizerTrnMs256NSFsid, + SynthesizerTrnMs256NSFsid_nono, + SynthesizerTrnMs768NSFsid, + SynthesizerTrnMs768NSFsid_nono, +) +from lib.infer_pack.models_onnx import SynthesizerTrnMsNSFsidM +from infer_uvr5 import _audio_pre_, _audio_pre_new +from lib.audio import load_audio +from lib.train.process_ckpt import change_info, extract_small_model, merge, show_info +from vc_infer_pipeline import VC +from sklearn.cluster import MiniBatchKMeans + +logging.getLogger("numba").setLevel(logging.WARNING) + +now_dir = os.getcwd() +tmp = os.path.join(now_dir, "TEMP") +shutil.rmtree(tmp, ignore_errors=True) +shutil.rmtree( + "%s/runtime/Lib/site-packages/infer_pack" % (now_dir), ignore_errors=True +) +shutil.rmtree("%s/runtime/Lib/site-packages/uvr5_pack" % (now_dir), ignore_errors=True) +os.makedirs(tmp, exist_ok=True) +os.makedirs(os.path.join(now_dir, "logs"), exist_ok=True) +os.makedirs(os.path.join(now_dir, "weights"), exist_ok=True) +os.environ["TEMP"] = tmp +warnings.filterwarnings("ignore") +torch.manual_seed(114514) + + +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 +i18n = I18nAuto() +i18n.print() +# 判断是否有能用来训练和加速推理的N卡 +ngpu = torch.cuda.device_count() +gpu_infos = [] +mem = [] +if_gpu_ok = False + +if torch.cuda.is_available() or ngpu != 0: + for i in range(ngpu): + gpu_name = torch.cuda.get_device_name(i) + if any( + value in gpu_name.upper() + for value in [ + "10", + "16", + "20", + "30", + "40", + "A2", + "A3", + "A4", + "P4", + "A50", + "500", + "A60", + "70", + "80", + "90", + "M4", + "T4", + "TITAN", + ] + ): + # A10#A100#V100#A40#P40#M40#K80#A4500 + if_gpu_ok = True # 至少有一张能用的N卡 + gpu_infos.append("%s\t%s" % (i, gpu_name)) + mem.append( + int( + torch.cuda.get_device_properties(i).total_memory + / 1024 + / 1024 + / 1024 + + 0.4 + ) + ) +if if_gpu_ok and len(gpu_infos) > 0: + gpu_info = "\n".join(gpu_infos) + default_batch_size = min(mem) // 2 +else: + gpu_info = i18n("很遗憾您这没有能用的显卡来支持您训练") + default_batch_size = 1 +gpus = "-".join([i[0] for i in gpu_infos]) + + +class ToolButton(gr.Button, gr.components.FormComponent): + """Small button with single emoji as text, fits inside gradio forms""" + + def __init__(self, **kwargs): + super().__init__(variant="tool", **kwargs) + + def get_block_name(self): + return "button" + + +hubert_model = None + + +def load_hubert(): + global hubert_model + 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() + + +weight_root = "weights" +weight_uvr5_root = "uvr5_weights" +index_root = "logs" +names = [] +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)) +uvr5_names = [] +for name in os.listdir(weight_uvr5_root): + if name.endswith(".pth") or "onnx" in name: + uvr5_names.append(name.replace(".pth", "")) + +cpt = None + + +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, cpt + if input_audio_path is None: + return "You need to upload an audio", None + f0_up_key = int(f0_up_key) + try: + 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 not hubert_model: + 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 tgt_sr != resample_sr >= 16000: + 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) + + +def vc_multi( + sid, + dir_path, + opt_root, + paths, + f0_up_key, + f0_method, + file_index, + file_index2, + # file_big_npy, + index_rate, + filter_radius, + resample_sr, + rms_mix_rate, + protect, + format1, +): + try: + dir_path = ( + dir_path.strip(" ").strip('"').strip("\n").strip('"').strip(" ") + ) # 防止小白拷路径头尾带了空格和"和回车 + opt_root = opt_root.strip(" ").strip('"').strip("\n").strip('"').strip(" ") + os.makedirs(opt_root, exist_ok=True) + try: + if dir_path != "": + paths = [os.path.join(dir_path, name) for name in os.listdir(dir_path)] + else: + paths = [path.name for path in paths] + except: + traceback.print_exc() + paths = [path.name for path in paths] + infos = [] + for path in paths: + info, opt = vc_single( + sid, + path, + f0_up_key, + None, + f0_method, + file_index, + file_index2, + # file_big_npy, + index_rate, + filter_radius, + resample_sr, + rms_mix_rate, + protect, + ) + if "Success" in info: + try: + tgt_sr, audio_opt = opt + if format1 in ["wav", "flac"]: + sf.write( + "%s/%s.%s" % (opt_root, os.path.basename(path), format1), + audio_opt, + tgt_sr, + ) + else: + path = "%s/%s.wav" % (opt_root, os.path.basename(path)) + sf.write( + path, + audio_opt, + tgt_sr, + ) + if os.path.exists(path): + os.system( + "ffmpeg -i %s -vn %s -q:a 2 -y" + % (path, path[:-4] + ".%s" % format1) + ) + except: + info += traceback.format_exc() + infos.append("%s->%s" % (os.path.basename(path), info)) + yield "\n".join(infos) + yield "\n".join(infos) + except: + yield traceback.format_exc() + + +def uvr(model_name, inp_root, save_root_vocal, paths, save_root_ins, agg, format0): + infos = [] + try: + inp_root = inp_root.strip(" ").strip('"').strip("\n").strip('"').strip(" ") + save_root_vocal = ( + save_root_vocal.strip(" ").strip('"').strip("\n").strip('"').strip(" ") + ) + save_root_ins = ( + save_root_ins.strip(" ").strip('"').strip("\n").strip('"').strip(" ") + ) + if model_name == "onnx_dereverb_By_FoxJoy": + from MDXNet import MDXNetDereverb + + pre_fun = MDXNetDereverb(15) + else: + func = _audio_pre_ if "DeEcho" not in model_name else _audio_pre_new + pre_fun = func( + agg=int(agg), + model_path=os.path.join(weight_uvr5_root, model_name + ".pth"), + device=config.device, + is_half=config.is_half, + ) + if inp_root != "": + paths = [os.path.join(inp_root, name) for name in os.listdir(inp_root)] + else: + paths = [path.name for path in paths] + for path in paths: + inp_path = os.path.join(inp_root, path) + need_reformat = 1 + done = 0 + try: + info = ffmpeg.probe(inp_path, cmd="ffprobe") + if ( + info["streams"][0]["channels"] == 2 + and info["streams"][0]["sample_rate"] == "44100" + ): + need_reformat = 0 + pre_fun._path_audio_( + inp_path, save_root_ins, save_root_vocal, format0 + ) + done = 1 + except: + need_reformat = 1 + traceback.print_exc() + if need_reformat == 1: + tmp_path = "%s/%s.reformatted.wav" % (tmp, os.path.basename(inp_path)) + os.system( + "ffmpeg -i %s -vn -acodec pcm_s16le -ac 2 -ar 44100 %s -y" + % (inp_path, tmp_path) + ) + inp_path = tmp_path + try: + if done == 0: + pre_fun._path_audio_( + inp_path, save_root_ins, save_root_vocal, format0 + ) + infos.append("%s->Success" % (os.path.basename(inp_path))) + yield "\n".join(infos) + except: + infos.append( + "%s->%s" % (os.path.basename(inp_path), traceback.format_exc()) + ) + yield "\n".join(infos) + except: + infos.append(traceback.format_exc()) + yield "\n".join(infos) + finally: + try: + if model_name == "onnx_dereverb_By_FoxJoy": + del pre_fun.pred.model + del pre_fun.pred.model_ + else: + del pre_fun.model + del pre_fun + except: + traceback.print_exc() + print("clean_empty_cache") + if torch.cuda.is_available(): + torch.cuda.empty_cache() + yield "\n".join(infos) + + +def get_index_path_from_model(sid): + sel_index_path = "" + name = os.path.join("logs", sid.split(".")[0], "") + # print(name) + for f in index_paths: + if name in f: + # print("selected index path:", f) + sel_index_path = f + break + return sel_index_path + + +# 一个选项卡全局只能有一个音色 +def get_vc(sid, to_return_protect0, to_return_protect1): + global n_spk, tgt_sr, net_g, vc, cpt, version + if sid == "" or sid == []: + global hubert_model + if hubert_model is not 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() + return {"visible": False, "__type__": "update"},{ + "visible": True, + "value": to_return_protect0, + "__type__": "update", + },{ + "visible": True, + "value": to_return_protect1, + "__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) + if if_f0 == 0: + to_return_protect0 = to_return_protect1 = { + "visible": False, + "value": 0.33, + "__type__": "update", + } + else: + to_return_protect0 = { + "visible": True, + "value": to_return_protect0, + "__type__": "update", + } + to_return_protect1 = { + "visible": True, + "value": to_return_protect1, + "__type__": "update", + } + 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] + index={ + "value":get_index_path_from_model(sid), + "__type__": "update" + } + return ( + {"visible": True, "maximum": n_spk, "__type__": "update"}, + to_return_protect0, + to_return_protect1, + index,index + ) + + +def change_choices(): + names = [] + 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)) + return {"choices": sorted(names), "__type__": "update"}, { + "choices": sorted(index_paths), + "__type__": "update", + } + + +def clean(): + return {"value": "", "__type__": "update"} + + +sr_dict = { + "32k": 32000, + "40k": 40000, + "48k": 48000, +} + + +def if_done(done, p): + while 1: + if p.poll() is None: + sleep(0.5) + else: + break + done[0] = True + + +def if_done_multi(done, ps): + while 1: + # poll==None代表进程未结束 + # 只要有一个进程未结束都不停 + flag = 1 + for p in ps: + if p.poll() is None: + flag = 0 + sleep(0.5) + break + if flag == 1: + break + done[0] = True + + +def preprocess_dataset(trainset_dir, exp_dir, sr, n_p): + sr = sr_dict[sr] + os.makedirs("%s/logs/%s" % (now_dir, exp_dir), exist_ok=True) + f = open("%s/logs/%s/preprocess.log" % (now_dir, exp_dir), "w") + f.close() + cmd = ( + config.python_cmd + + ' trainset_preprocess_pipeline_print.py "%s" %s %s "%s/logs/%s" ' + % (trainset_dir, sr, n_p, now_dir, exp_dir) + + str(config.noparallel) + ) + print(cmd) + p = Popen(cmd, shell=True) # , stdin=PIPE, stdout=PIPE,stderr=PIPE,cwd=now_dir + ###煞笔gr, popen read都非得全跑完了再一次性读取, 不用gr就正常读一句输出一句;只能额外弄出一个文本流定时读 + done = [False] + threading.Thread( + target=if_done, + args=( + done, + p, + ), + ).start() + while 1: + with open("%s/logs/%s/preprocess.log" % (now_dir, exp_dir), "r") as f: + yield (f.read()) + sleep(1) + if done[0]: + break + with open("%s/logs/%s/preprocess.log" % (now_dir, exp_dir), "r") as f: + log = f.read() + print(log) + yield log + + +# but2.click(extract_f0,[gpus6,np7,f0method8,if_f0_3,trainset_dir4],[info2]) +def extract_f0_feature(gpus, n_p, f0method, if_f0, exp_dir, version19, gpus_rmvpe): + gpus = gpus.split("-") + os.makedirs("%s/logs/%s" % (now_dir, exp_dir), exist_ok=True) + f = open("%s/logs/%s/extract_f0_feature.log" % (now_dir, exp_dir), "w") + f.close() + if if_f0: + if f0method != "rmvpe_gpu": + cmd = config.python_cmd + ' extract_f0_print.py "%s/logs/%s" %s %s' % ( + now_dir, + exp_dir, + n_p, + f0method, + ) + print(cmd) + p = Popen( + cmd, shell=True, cwd=now_dir + ) # , stdin=PIPE, stdout=PIPE,stderr=PIPE + ###煞笔gr, popen read都非得全跑完了再一次性读取, 不用gr就正常读一句输出一句;只能额外弄出一个文本流定时读 + done = [False] + threading.Thread( + target=if_done, + args=( + done, + p, + ), + ).start() + else: + if(gpus_rmvpe!="-"): + gpus_rmvpe = gpus_rmvpe.split("-") + leng = len(gpus_rmvpe) + ps = [] + for idx, n_g in enumerate(gpus_rmvpe): + cmd = ( + config.python_cmd + + ' extract_f0_rmvpe.py %s %s %s "%s/logs/%s" %s ' + % (leng, idx, n_g, now_dir, exp_dir, config.is_half) + ) + print(cmd) + p = Popen( + cmd, shell=True, cwd=now_dir + ) # , shell=True, stdin=PIPE, stdout=PIPE, stderr=PIPE, cwd=now_dir + ps.append(p) + ###煞笔gr, popen read都非得全跑完了再一次性读取, 不用gr就正常读一句输出一句;只能额外弄出一个文本流定时读 + done = [False] + threading.Thread( + target=if_done_multi,# + args=( + done, + ps, + ), + ).start() + else: + cmd = config.python_cmd + ' extract_f0_rmvpe_dml.py "%s/logs/%s" ' % ( + now_dir, exp_dir + ) + print(cmd) + p = Popen( + cmd, shell=True, cwd=now_dir + ) # , shell=True, stdin=PIPE, stdout=PIPE, stderr=PIPE, cwd=now_dir + p.wait() + done = [True] + while 1: + with open( + "%s/logs/%s/extract_f0_feature.log" % (now_dir, exp_dir), "r" + ) as f: + yield (f.read()) + sleep(1) + if done[0]: + break + with open( + "%s/logs/%s/extract_f0_feature.log" % (now_dir, exp_dir), "r" + ) as f: + log = f.read() + print(log) + yield log + ####对不同part分别开多进程 + """ + n_part=int(sys.argv[1]) + i_part=int(sys.argv[2]) + i_gpu=sys.argv[3] + exp_dir=sys.argv[4] + os.environ["CUDA_VISIBLE_DEVICES"]=str(i_gpu) + """ + leng = len(gpus) + ps = [] + for idx, n_g in enumerate(gpus): + cmd = ( + config.python_cmd + + ' extract_feature_print.py %s %s %s %s "%s/logs/%s" %s' + % ( + config.device, + leng, + idx, + n_g, + now_dir, + exp_dir, + version19, + ) + ) + print(cmd) + p = Popen( + cmd, shell=True, cwd=now_dir + ) # , shell=True, stdin=PIPE, stdout=PIPE, stderr=PIPE, cwd=now_dir + ps.append(p) + ###煞笔gr, popen read都非得全跑完了再一次性读取, 不用gr就正常读一句输出一句;只能额外弄出一个文本流定时读 + done = [False] + threading.Thread( + target=if_done_multi, + args=( + done, + ps, + ), + ).start() + while 1: + with open("%s/logs/%s/extract_f0_feature.log" % (now_dir, exp_dir), "r") as f: + yield (f.read()) + sleep(1) + if done[0]: + break + with open("%s/logs/%s/extract_f0_feature.log" % (now_dir, exp_dir), "r") as f: + log = f.read() + print(log) + yield log + + +def change_sr2(sr2, if_f0_3, version19): + path_str = "" if version19 == "v1" else "_v2" + f0_str = "f0" if if_f0_3 else "" + if_pretrained_generator_exist = os.access( + "pretrained%s/%sG%s.pth" % (path_str, f0_str, sr2), os.F_OK + ) + if_pretrained_discriminator_exist = os.access( + "pretrained%s/%sD%s.pth" % (path_str, f0_str, sr2), os.F_OK + ) + if not if_pretrained_generator_exist: + print( + "pretrained%s/%sG%s.pth" % (path_str, f0_str, sr2), + "not exist, will not use pretrained model", + ) + if not if_pretrained_discriminator_exist: + print( + "pretrained%s/%sD%s.pth" % (path_str, f0_str, sr2), + "not exist, will not use pretrained model", + ) + return ( + "pretrained%s/%sG%s.pth" % (path_str, f0_str, sr2) + if if_pretrained_generator_exist + else "", + "pretrained%s/%sD%s.pth" % (path_str, f0_str, sr2) + if if_pretrained_discriminator_exist + else "", + ) + + +def change_version19(sr2, if_f0_3, version19): + path_str = "" if version19 == "v1" else "_v2" + if sr2 == "32k" and version19 == "v1": + sr2 = "40k" + to_return_sr2 = ( + {"choices": ["40k", "48k"], "__type__": "update", "value": sr2} + if version19 == "v1" + else {"choices": ["40k", "48k", "32k"], "__type__": "update", "value": sr2} + ) + f0_str = "f0" if if_f0_3 else "" + if_pretrained_generator_exist = os.access( + "pretrained%s/%sG%s.pth" % (path_str, f0_str, sr2), os.F_OK + ) + if_pretrained_discriminator_exist = os.access( + "pretrained%s/%sD%s.pth" % (path_str, f0_str, sr2), os.F_OK + ) + if not if_pretrained_generator_exist: + print( + "pretrained%s/%sG%s.pth" % (path_str, f0_str, sr2), + "not exist, will not use pretrained model", + ) + if not if_pretrained_discriminator_exist: + print( + "pretrained%s/%sD%s.pth" % (path_str, f0_str, sr2), + "not exist, will not use pretrained model", + ) + return ( + "pretrained%s/%sG%s.pth" % (path_str, f0_str, sr2) + if if_pretrained_generator_exist + else "", + "pretrained%s/%sD%s.pth" % (path_str, f0_str, sr2) + if if_pretrained_discriminator_exist + else "", + to_return_sr2, + ) + + +def change_f0(if_f0_3, sr2, version19): # f0method8,pretrained_G14,pretrained_D15 + path_str = "" if version19 == "v1" else "_v2" + if_pretrained_generator_exist = os.access( + "pretrained%s/f0G%s.pth" % (path_str, sr2), os.F_OK + ) + if_pretrained_discriminator_exist = os.access( + "pretrained%s/f0D%s.pth" % (path_str, sr2), os.F_OK + ) + if not if_pretrained_generator_exist: + print( + "pretrained%s/f0G%s.pth" % (path_str, sr2), + "not exist, will not use pretrained model", + ) + if not if_pretrained_discriminator_exist: + print( + "pretrained%s/f0D%s.pth" % (path_str, sr2), + "not exist, will not use pretrained model", + ) + if if_f0_3: + return ( + {"visible": True, "__type__": "update"}, + "pretrained%s/f0G%s.pth" % (path_str, sr2) + if if_pretrained_generator_exist + else "", + "pretrained%s/f0D%s.pth" % (path_str, sr2) + if if_pretrained_discriminator_exist + else "", + ) + return ( + {"visible": False, "__type__": "update"}, + ("pretrained%s/G%s.pth" % (path_str, sr2)) + if if_pretrained_generator_exist + else "", + ("pretrained%s/D%s.pth" % (path_str, sr2)) + if if_pretrained_discriminator_exist + else "", + ) + + +# but3.click(click_train,[exp_dir1,sr2,if_f0_3,save_epoch10,total_epoch11,batch_size12,if_save_latest13,pretrained_G14,pretrained_D15,gpus16]) +def click_train( + exp_dir1, + sr2, + if_f0_3, + spk_id5, + save_epoch10, + total_epoch11, + batch_size12, + if_save_latest13, + pretrained_G14, + pretrained_D15, + gpus16, + if_cache_gpu17, + if_save_every_weights18, + version19, +): + # 生成filelist + exp_dir = "%s/logs/%s" % (now_dir, exp_dir1) + os.makedirs(exp_dir, exist_ok=True) + gt_wavs_dir = "%s/0_gt_wavs" % (exp_dir) + feature_dir = ( + "%s/3_feature256" % (exp_dir) + if version19 == "v1" + else "%s/3_feature768" % (exp_dir) + ) + if if_f0_3: + f0_dir = "%s/2a_f0" % (exp_dir) + f0nsf_dir = "%s/2b-f0nsf" % (exp_dir) + names = ( + set([name.split(".")[0] for name in os.listdir(gt_wavs_dir)]) + & set([name.split(".")[0] for name in os.listdir(feature_dir)]) + & set([name.split(".")[0] for name in os.listdir(f0_dir)]) + & set([name.split(".")[0] for name in os.listdir(f0nsf_dir)]) + ) + else: + names = set([name.split(".")[0] for name in os.listdir(gt_wavs_dir)]) & set( + [name.split(".")[0] for name in os.listdir(feature_dir)] + ) + opt = [] + for name in names: + if if_f0_3: + opt.append( + "%s/%s.wav|%s/%s.npy|%s/%s.wav.npy|%s/%s.wav.npy|%s" + % ( + gt_wavs_dir.replace("\\", "\\\\"), + name, + feature_dir.replace("\\", "\\\\"), + name, + f0_dir.replace("\\", "\\\\"), + name, + f0nsf_dir.replace("\\", "\\\\"), + name, + spk_id5, + ) + ) + else: + opt.append( + "%s/%s.wav|%s/%s.npy|%s" + % ( + gt_wavs_dir.replace("\\", "\\\\"), + name, + feature_dir.replace("\\", "\\\\"), + name, + spk_id5, + ) + ) + fea_dim = 256 if version19 == "v1" else 768 + if if_f0_3: + for _ in range(2): + opt.append( + "%s/logs/mute/0_gt_wavs/mute%s.wav|%s/logs/mute/3_feature%s/mute.npy|%s/logs/mute/2a_f0/mute.wav.npy|%s/logs/mute/2b-f0nsf/mute.wav.npy|%s" + % (now_dir, sr2, now_dir, fea_dim, now_dir, now_dir, spk_id5) + ) + else: + for _ in range(2): + opt.append( + "%s/logs/mute/0_gt_wavs/mute%s.wav|%s/logs/mute/3_feature%s/mute.npy|%s" + % (now_dir, sr2, now_dir, fea_dim, spk_id5) + ) + shuffle(opt) + with open("%s/filelist.txt" % exp_dir, "w") as f: + f.write("\n".join(opt)) + print("write filelist done") + # 生成config#无需生成config + # cmd = python_cmd + " train_nsf_sim_cache_sid_load_pretrain.py -e mi-test -sr 40k -f0 1 -bs 4 -g 0 -te 10 -se 5 -pg pretrained/f0G40k.pth -pd pretrained/f0D40k.pth -l 1 -c 0" + print("use gpus:", gpus16) + if pretrained_G14 == "": + print("no pretrained Generator") + if pretrained_D15 == "": + print("no pretrained Discriminator") + if gpus16: + cmd = ( + config.python_cmd + + ' train_nsf_sim_cache_sid_load_pretrain.py -e "%s" -sr %s -f0 %s -bs %s -g %s -te %s -se %s %s %s -l %s -c %s -sw %s -v %s' + % ( + exp_dir1, + sr2, + 1 if if_f0_3 else 0, + batch_size12, + gpus16, + total_epoch11, + save_epoch10, + "-pg %s" % pretrained_G14 if pretrained_G14 != "" else "", + "-pd %s" % pretrained_D15 if pretrained_D15 != "" else "", + 1 if if_save_latest13 == i18n("是") else 0, + 1 if if_cache_gpu17 == i18n("是") else 0, + 1 if if_save_every_weights18 == i18n("是") else 0, + version19, + ) + ) + else: + cmd = ( + config.python_cmd + + ' train_nsf_sim_cache_sid_load_pretrain.py -e "%s" -sr %s -f0 %s -bs %s -te %s -se %s %s %s -l %s -c %s -sw %s -v %s' + % ( + exp_dir1, + sr2, + 1 if if_f0_3 else 0, + batch_size12, + total_epoch11, + save_epoch10, + "-pg %s" % pretrained_G14 if pretrained_G14 != "" else "", + "-pd %s" % pretrained_D15 if pretrained_D15 != "" else "", + 1 if if_save_latest13 == i18n("是") else 0, + 1 if if_cache_gpu17 == i18n("是") else 0, + 1 if if_save_every_weights18 == i18n("是") else 0, + version19, + ) + ) + print(cmd) + p = Popen(cmd, shell=True, cwd=now_dir) + p.wait() + return "训练结束, 您可查看控制台训练日志或实验文件夹下的train.log" + + +# but4.click(train_index, [exp_dir1], info3) +def train_index(exp_dir1, version19): + # exp_dir = "%s/logs/%s" % (now_dir, exp_dir1) + exp_dir = "logs/%s" % ( exp_dir1) + os.makedirs(exp_dir, exist_ok=True) + feature_dir = ( + "%s/3_feature256" % (exp_dir) + if version19 == "v1" + else "%s/3_feature768" % (exp_dir) + ) + if not os.path.exists(feature_dir): + return "请先进行特征提取!" + listdir_res = list(os.listdir(feature_dir)) + if len(listdir_res) == 0: + return "请先进行特征提取！" + infos = [] + npys = [] + for name in sorted(listdir_res): + phone = np.load("%s/%s" % (feature_dir, name)) + npys.append(phone) + big_npy = np.concatenate(npys, 0) + big_npy_idx = np.arange(big_npy.shape[0]) + np.random.shuffle(big_npy_idx) + big_npy = big_npy[big_npy_idx] + if big_npy.shape[0] > 2e5: + infos.append("Trying doing kmeans %s shape to 10k centers." % big_npy.shape[0]) + yield "\n".join(infos) + try: + big_npy = ( + MiniBatchKMeans( + n_clusters=10000, + verbose=True, + batch_size=256 * config.n_cpu, + compute_labels=False, + init="random", + ) + .fit(big_npy) + .cluster_centers_ + ) + except: + info = traceback.format_exc() + print(info) + infos.append(info) + yield "\n".join(infos) + + np.save("%s/total_fea.npy" % exp_dir, big_npy) + n_ivf = min(int(16 * np.sqrt(big_npy.shape[0])), big_npy.shape[0] // 39) + infos.append("%s,%s" % (big_npy.shape, n_ivf)) + yield "\n".join(infos) + index = faiss.index_factory(256 if version19 == "v1" else 768, "IVF%s,Flat" % n_ivf) + # index = faiss.index_factory(256if version19=="v1"else 768, "IVF%s,PQ128x4fs,RFlat"%n_ivf) + infos.append("training") + yield "\n".join(infos) + index_ivf = faiss.extract_index_ivf(index) # + index_ivf.nprobe = 1 + index.train(big_npy) + faiss.write_index( + index, + "%s/trained_IVF%s_Flat_nprobe_%s_%s_%s.index" + % (exp_dir, n_ivf, index_ivf.nprobe, exp_dir1, version19), + ) + + infos.append("adding") + yield "\n".join(infos) + batch_size_add = 8192 + for i in range(0, big_npy.shape[0], batch_size_add): + index.add(big_npy[i : i + batch_size_add]) + faiss.write_index( + index, + "%s/added_IVF%s_Flat_nprobe_%s_%s_%s.index" + % (exp_dir, n_ivf, index_ivf.nprobe, exp_dir1, version19), + ) + infos.append( + "成功构建索引，added_IVF%s_Flat_nprobe_%s_%s_%s.index" + % (n_ivf, index_ivf.nprobe, exp_dir1, version19) + ) + # faiss.write_index(index, '%s/added_IVF%s_Flat_FastScan_%s.index'%(exp_dir,n_ivf,version19)) + # infos.append("成功构建索引，added_IVF%s_Flat_FastScan_%s.index"%(n_ivf,version19)) + yield "\n".join(infos) + + +# but5.click(train1key, [exp_dir1, sr2, if_f0_3, trainset_dir4, spk_id5, gpus6, np7, f0method8, save_epoch10, total_epoch11, batch_size12, if_save_latest13, pretrained_G14, pretrained_D15, gpus16, if_cache_gpu17], info3) +def train1key( + exp_dir1, + sr2, + if_f0_3, + trainset_dir4, + spk_id5, + np7, + f0method8, + save_epoch10, + total_epoch11, + batch_size12, + if_save_latest13, + pretrained_G14, + pretrained_D15, + gpus16, + if_cache_gpu17, + if_save_every_weights18, + version19, + gpus_rmvpe, +): + infos = [] + + def get_info_str(strr): + infos.append(strr) + return "\n".join(infos) + + model_log_dir = "%s/logs/%s" % (now_dir, exp_dir1) + preprocess_log_path = "%s/preprocess.log" % model_log_dir + extract_f0_feature_log_path = "%s/extract_f0_feature.log" % model_log_dir + gt_wavs_dir = "%s/0_gt_wavs" % model_log_dir + feature_dir = ( + "%s/3_feature256" % model_log_dir + if version19 == "v1" + else "%s/3_feature768" % model_log_dir + ) + + os.makedirs(model_log_dir, exist_ok=True) + #########step1:处理数据 + open(preprocess_log_path, "w").close() + cmd = ( + config.python_cmd + + ' trainset_preprocess_pipeline_print.py "%s" %s %s "%s" ' + % (trainset_dir4, sr_dict[sr2], np7, model_log_dir) + + str(config.noparallel) + ) + yield get_info_str(i18n("step1:正在处理数据")) + yield get_info_str(cmd) + p = Popen(cmd, shell=True) + p.wait() + with open(preprocess_log_path, "r") as f: + print(f.read()) + #########step2a:提取音高 + open(extract_f0_feature_log_path, "w") + if if_f0_3: + yield get_info_str("step2a:正在提取音高") + if f0method8 != "rmvpe_gpu": + cmd = config.python_cmd + ' extract_f0_print.py "%s" %s %s' % ( + model_log_dir, + np7, + f0method8, + ) + yield get_info_str(cmd) + p = Popen(cmd, shell=True, cwd=now_dir) + p.wait() + else: + if(gpus_rmvpe!="-"): + gpus_rmvpe = gpus_rmvpe.split("-") + leng = len(gpus_rmvpe) + ps = [] + for idx, n_g in enumerate(gpus_rmvpe): + cmd = config.python_cmd + ' extract_f0_rmvpe.py %s %s %s "%s" %s ' % ( + leng, + idx, + n_g, + model_log_dir, + config.is_half, + ) + yield get_info_str(cmd) + p = Popen( + cmd, shell=True, cwd=now_dir + ) # , shell=True, stdin=PIPE, stdout=PIPE, stderr=PIPE, cwd=now_dir + ps.append(p) + for p in ps: + p.wait() + else:#dml + cmd = config.python_cmd + ' extract_f0_rmvpe_dml.py "%s" ' % ( + model_log_dir + ) + yield get_info_str(cmd) + p = Popen( + cmd, shell=True, cwd=now_dir + ) # , shell=True, stdin=PIPE, stdout=PIPE, stderr=PIPE, cwd=now_dir + p.wait() + with open(extract_f0_feature_log_path, "r") as f: + print(f.read()) + else: + yield get_info_str(i18n("step2a:无需提取音高")) + #######step2b:提取特征 + yield get_info_str(i18n("step2b:正在提取特征")) + gpus = gpus16.split("-") + leng = len(gpus) + ps = [] + for idx, n_g in enumerate(gpus): + cmd = config.python_cmd + ' extract_feature_print.py %s %s %s %s "%s" %s' % ( + config.device, + leng, + idx, + n_g, + model_log_dir, + version19, + ) + yield get_info_str(cmd) + p = Popen( + cmd, shell=True, cwd=now_dir + ) # , shell=True, stdin=PIPE, stdout=PIPE, stderr=PIPE, cwd=now_dir + ps.append(p) + for p in ps: + p.wait() + with open(extract_f0_feature_log_path, "r") as f: + print(f.read()) + #######step3a:训练模型 + yield get_info_str(i18n("step3a:正在训练模型")) + # 生成filelist + if if_f0_3: + f0_dir = "%s/2a_f0" % model_log_dir + f0nsf_dir = "%s/2b-f0nsf" % model_log_dir + names = ( + set([name.split(".")[0] for name in os.listdir(gt_wavs_dir)]) + & set([name.split(".")[0] for name in os.listdir(feature_dir)]) + & set([name.split(".")[0] for name in os.listdir(f0_dir)]) + & set([name.split(".")[0] for name in os.listdir(f0nsf_dir)]) + ) + else: + names = set([name.split(".")[0] for name in os.listdir(gt_wavs_dir)]) & set( + [name.split(".")[0] for name in os.listdir(feature_dir)] + ) + opt = [] + for name in names: + if if_f0_3: + opt.append( + "%s/%s.wav|%s/%s.npy|%s/%s.wav.npy|%s/%s.wav.npy|%s" + % ( + gt_wavs_dir.replace("\\", "\\\\"), + name, + feature_dir.replace("\\", "\\\\"), + name, + f0_dir.replace("\\", "\\\\"), + name, + f0nsf_dir.replace("\\", "\\\\"), + name, + spk_id5, + ) + ) + else: + opt.append( + "%s/%s.wav|%s/%s.npy|%s" + % ( + gt_wavs_dir.replace("\\", "\\\\"), + name, + feature_dir.replace("\\", "\\\\"), + name, + spk_id5, + ) + ) + fea_dim = 256 if version19 == "v1" else 768 + if if_f0_3: + for _ in range(2): + opt.append( + "%s/logs/mute/0_gt_wavs/mute%s.wav|%s/logs/mute/3_feature%s/mute.npy|%s/logs/mute/2a_f0/mute.wav.npy|%s/logs/mute/2b-f0nsf/mute.wav.npy|%s" + % (now_dir, sr2, now_dir, fea_dim, now_dir, now_dir, spk_id5) + ) + else: + for _ in range(2): + opt.append( + "%s/logs/mute/0_gt_wavs/mute%s.wav|%s/logs/mute/3_feature%s/mute.npy|%s" + % (now_dir, sr2, now_dir, fea_dim, spk_id5) + ) + shuffle(opt) + with open("%s/filelist.txt" % model_log_dir, "w") as f: + f.write("\n".join(opt)) + yield get_info_str("write filelist done") + if gpus16: + cmd = ( + config.python_cmd + + ' train_nsf_sim_cache_sid_load_pretrain.py -e "%s" -sr %s -f0 %s -bs %s -g %s -te %s -se %s %s %s -l %s -c %s -sw %s -v %s' + % ( + exp_dir1, + sr2, + 1 if if_f0_3 else 0, + batch_size12, + gpus16, + total_epoch11, + save_epoch10, + "-pg %s" % pretrained_G14 if pretrained_G14 != "" else "", + "-pd %s" % pretrained_D15 if pretrained_D15 != "" else "", + 1 if if_save_latest13 == i18n("是") else 0, + 1 if if_cache_gpu17 == i18n("是") else 0, + 1 if if_save_every_weights18 == i18n("是") else 0, + version19, + ) + ) + else: + cmd = ( + config.python_cmd + + ' train_nsf_sim_cache_sid_load_pretrain.py -e "%s" -sr %s -f0 %s -bs %s -te %s -se %s %s %s -l %s -c %s -sw %s -v %s' + % ( + exp_dir1, + sr2, + 1 if if_f0_3 else 0, + batch_size12, + total_epoch11, + save_epoch10, + "-pg %s" % pretrained_G14 if pretrained_G14 != "" else "", + "-pd %s" % pretrained_D15 if pretrained_D15 != "" else "", + 1 if if_save_latest13 == i18n("是") else 0, + 1 if if_cache_gpu17 == i18n("是") else 0, + 1 if if_save_every_weights18 == i18n("是") else 0, + version19, + ) + ) + yield get_info_str(cmd) + p = Popen(cmd, shell=True, cwd=now_dir) + p.wait() + yield get_info_str(i18n("训练结束, 您可查看控制台训练日志或实验文件夹下的train.log")) + #######step3b:训练索引 + npys = [] + listdir_res = list(os.listdir(feature_dir)) + for name in sorted(listdir_res): + phone = np.load("%s/%s" % (feature_dir, name)) + npys.append(phone) + big_npy = np.concatenate(npys, 0) + + big_npy_idx = np.arange(big_npy.shape[0]) + np.random.shuffle(big_npy_idx) + big_npy = big_npy[big_npy_idx] + + if big_npy.shape[0] > 2e5: + # if(1): + info = "Trying doing kmeans %s shape to 10k centers." % big_npy.shape[0] + print(info) + yield get_info_str(info) + try: + big_npy = ( + MiniBatchKMeans( + n_clusters=10000, + verbose=True, + batch_size=256 * config.n_cpu, + compute_labels=False, + init="random", + ) + .fit(big_npy) + .cluster_centers_ + ) + except: + info = traceback.format_exc() + print(info) + yield get_info_str(info) + + np.save("%s/total_fea.npy" % model_log_dir, big_npy) + n_ivf = min(int(16 * np.sqrt(big_npy.shape[0])), big_npy.shape[0] // 39) + yield get_info_str("%s,%s" % (big_npy.shape, n_ivf)) + index = faiss.index_factory(256 if version19 == "v1" else 768, "IVF%s,Flat" % n_ivf) + yield get_info_str("training index") + index_ivf = faiss.extract_index_ivf(index) # + index_ivf.nprobe = 1 + index.train(big_npy) + faiss.write_index(index,"%s/trained_IVF%s_Flat_nprobe_%s_%s_%s.index"% (model_log_dir.replace(now_dir+"/",""), n_ivf, index_ivf.nprobe, exp_dir1, version19)) + yield get_info_str("adding index") + batch_size_add = 8192 + for i in range(0, big_npy.shape[0], batch_size_add): + index.add(big_npy[i : i + batch_size_add]) + faiss.write_index( + index, + "%s/added_IVF%s_Flat_nprobe_%s_%s_%s.index" + % (model_log_dir.replace(now_dir+"/",""), n_ivf, index_ivf.nprobe, exp_dir1, version19), + ) + yield get_info_str( + "成功构建索引, added_IVF%s_Flat_nprobe_%s_%s_%s.index" + % (n_ivf, index_ivf.nprobe, exp_dir1, version19) + ) + yield get_info_str(i18n("全流程结束！")) + + +# ckpt_path2.change(change_info_,[ckpt_path2],[sr__,if_f0__]) +def change_info_(ckpt_path): + if not os.path.exists(ckpt_path.replace(os.path.basename(ckpt_path), "train.log")): + return {"__type__": "update"}, {"__type__": "update"}, {"__type__": "update"} + try: + with open( + ckpt_path.replace(os.path.basename(ckpt_path), "train.log"), "r" + ) as f: + info = eval(f.read().strip("\n").split("\n")[0].split("\t")[-1]) + sr, f0 = info["sample_rate"], info["if_f0"] + version = "v2" if ("version" in info and info["version"] == "v2") else "v1" + return sr, str(f0), version + except: + traceback.print_exc() + return {"__type__": "update"}, {"__type__": "update"}, {"__type__": "update"} + + +F0GPUVisible = config.dml == False +def change_f0_method(f0method8): + if f0method8 == "rmvpe_gpu": + visible = F0GPUVisible + else: + visible = False + return {"visible": visible, "__type__": "update"} + + +def export_onnx(ModelPath, ExportedPath): + global cpt + cpt = torch.load(ModelPath, map_location="cpu") + cpt["config"][-3] = cpt["weight"]["emb_g.weight"].shape[0] + vec_channels = 256 if cpt.get("version", "v1") == "v1" else 768 + + test_phone = torch.rand(1, 200, vec_channels) # hidden unit + test_phone_lengths = torch.tensor([200]).long() # hidden unit 长度（貌似没啥用） + test_pitch = torch.randint(size=(1, 200), low=5, high=255) # 基频（单位赫兹） + test_pitchf = torch.rand(1, 200) # nsf基频 + test_ds = torch.LongTensor([0]) # 说话人ID + test_rnd = torch.rand(1, 192, 200) # 噪声（加入随机因子） + + device = "cpu" # 导出时设备（不影响使用模型） + + net_g = SynthesizerTrnMsNSFsidM( + *cpt["config"], is_half=False, version=cpt.get("version", "v1") + ) # fp32导出（C++要支持fp16必须手动将内存重新排列所以暂时不用fp16） + net_g.load_state_dict(cpt["weight"], strict=False) + input_names = ["phone", "phone_lengths", "pitch", "pitchf", "ds", "rnd"] + output_names = [ + "audio", + ] + # net_g.construct_spkmixmap(n_speaker) 多角色混合轨道导出 + torch.onnx.export( + net_g, + ( + test_phone.to(device), + test_phone_lengths.to(device), + test_pitch.to(device), + test_pitchf.to(device), + test_ds.to(device), + test_rnd.to(device), + ), + ExportedPath, + dynamic_axes={ + "phone": [1], + "pitch": [1], + "pitchf": [1], + "rnd": [2], + }, + do_constant_folding=False, + opset_version=13, + verbose=False, + input_names=input_names, + output_names=output_names, + ) + return "Finished" + + +with gr.Blocks(title="RVC WebUI") as app: + gr.Markdown( + value=i18n( + "本软件以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. 详见根目录LICENSE." + ) + ) + with gr.Tabs(): + with gr.TabItem(i18n("模型推理")): + with gr.Row(): + sid0 = gr.Dropdown(label=i18n("推理音色"), choices=sorted(names)) + refresh_button = gr.Button(i18n("刷新音色列表和索引路径"), variant="primary") + clean_button = gr.Button(i18n("卸载音色省显存"), variant="primary") + 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, + ) + clean_button.click( + fn=clean, inputs=[], outputs=[sid0], api_name="infer_clean" + ) + with gr.Group(): + 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. ") + ) + with gr.Row(): + with gr.Column(): + vc_transform0 = gr.Number( + label=i18n("变调(整数, 半音数量, 升八度12降八度-12)"), value=0 + ) + input_audio0 = gr.Textbox( + label=i18n("输入待处理音频文件路径(默认是正确格式示例)"), + value="E:\\codes\\py39\\test-20230416b\\todo-songs\\冬之花clip1.wav", + ) + 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,rmvpe效果最好且微吃GPU" + ), + choices=["pm", "harvest", "crepe", "rmvpe"]if config.dml==False else ["pm", "harvest", "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=True, + ) + 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, + ) + refresh_button.click( + fn=change_choices, + inputs=[], + outputs=[sid0, file_index2], + api_name="infer_refresh", + ) + # file_big_npy1 = gr.Textbox( + # label=i18n("特征文件路径"), + # value="E:\\codes\py39\\vits_vc_gpu_train\\logs\\mi-test-1key\\total_fea.npy", + # interactive=True, + # ) + index_rate1 = gr.Slider( + minimum=0, + maximum=1, + label=i18n("检索特征占比"), + value=0.75, + interactive=True, + ) + with gr.Column(): + 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=0.25, + 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") + with gr.Row(): + 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, + input_audio0, + 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], + api_name="infer_convert", + ) + with gr.Group(): + gr.Markdown( + value=i18n("批量转换, 输入待转换音频文件夹, 或上传多个音频文件, 在指定文件夹(默认opt)下输出转换的音频. ") + ) + with gr.Row(): + with gr.Column(): + vc_transform1 = gr.Number( + label=i18n("变调(整数, 半音数量, 升八度12降八度-12)"), value=0 + ) + opt_input = gr.Textbox(label=i18n("指定输出文件夹"), value="opt") + f0method1 = 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,rmvpe效果最好且微吃GPU" + ), + choices=["pm", "harvest", "crepe", "rmvpe"]if config.dml==False else ["pm", "harvest", "rmvpe"], + value="pm", + interactive=True, + ) + filter_radius1 = 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_index3 = 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=True, + ) + file_index4 = gr.Dropdown( + label=i18n("自动检测index路径,下拉式选择(dropdown)"), + choices=sorted(index_paths), + interactive=True, + ) + refresh_button.click( + fn=lambda: change_choices()[1], + inputs=[], + outputs=file_index4, + api_name="infer_refresh_batch", + ) + # file_big_npy2 = gr.Textbox( + # label=i18n("特征文件路径"), + # value="E:\\codes\\py39\\vits_vc_gpu_train\\logs\\mi-test-1key\\total_fea.npy", + # interactive=True, + # ) + index_rate2 = gr.Slider( + minimum=0, + maximum=1, + label=i18n("检索特征占比"), + value=1, + interactive=True, + ) + with gr.Column(): + resample_sr1 = 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_rate1 = 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, + ) + protect1 = 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, + ) + with gr.Column(): + dir_input = gr.Textbox( + label=i18n("输入待处理音频文件夹路径(去文件管理器地址栏拷就行了)"), + value="E:\codes\py39\\test-20230416b\\todo-songs", + ) + inputs = gr.File( + file_count="multiple", label=i18n("也可批量输入音频文件, 二选一, 优先读文件夹") + ) + with gr.Row(): + format1 = gr.Radio( + label=i18n("导出文件格式"), + choices=["wav", "flac", "mp3", "m4a"], + value="flac", + interactive=True, + ) + but1 = gr.Button(i18n("转换"), variant="primary") + vc_output3 = gr.Textbox(label=i18n("输出信息")) + but1.click( + vc_multi, + [ + spk_item, + dir_input, + opt_input, + inputs, + vc_transform1, + f0method1, + file_index3, + file_index4, + # file_big_npy2, + index_rate2, + filter_radius1, + resample_sr1, + rms_mix_rate1, + protect1, + format1, + ], + [vc_output3], + api_name="infer_convert_batch", + ) + sid0.change( + fn=get_vc, + inputs=[sid0, protect0, protect1], + outputs=[spk_item, protect0, protect1, file_index2, file_index4], + ) + with gr.TabItem(i18n("伴奏人声分离&去混响&去回声")): + with gr.Group(): + gr.Markdown( + value=i18n( + "人声伴奏分离批量处理， 使用UVR5模型。
合格的文件夹路径格式举例： E:\\codes\\py39\\vits_vc_gpu\\白鹭霜华测试样例(去文件管理器地址栏拷就行了)。
模型分为三类：
1、保留人声：不带和声的音频选这个，对主人声保留比HP5更好。内置HP2和HP3两个模型，HP3可能轻微漏伴奏但对主人声保留比HP2稍微好一丁点；
2、仅保留主人声：带和声的音频选这个，对主人声可能有削弱。内置HP5一个模型；
3、去混响、去延迟模型（by FoxJoy）：
  (1)MDX-Net(onnx_dereverb):对于双通道混响是最好的选择，不能去除单通道混响；
 (234)DeEcho:去除延迟效果。Aggressive比Normal去除得更彻底，DeReverb额外去除混响，可去除单声道混响，但是对高频重的板式混响去不干净。
去混响/去延迟，附：
1、DeEcho-DeReverb模型的耗时是另外2个DeEcho模型的接近2倍；
2、MDX-Net-Dereverb模型挺慢的；
3、个人推荐的最干净的配置是先MDX-Net再DeEcho-Aggressive。" + ) + ) + with gr.Row(): + with gr.Column(): + dir_wav_input = gr.Textbox( + label=i18n("输入待处理音频文件夹路径"), + value="E:\\codes\\py39\\test-20230416b\\todo-songs\\todo-songs", + ) + wav_inputs = gr.File( + file_count="multiple", label=i18n("也可批量拖拽音频文件, 二选一, 优先读文件夹，文件夹留空则读取拖拽文件") + ) + with gr.Column(): + model_choose = gr.Dropdown(label=i18n("模型"), choices=uvr5_names) + agg = gr.Slider( + minimum=0, + maximum=20, + step=1, + label="人声提取激进程度", + value=10, + interactive=True, + visible=False, # 先不开放调整 + ) + opt_vocal_root = gr.Textbox( + label=i18n("指定输出主人声文件夹"), value="opt" + ) + opt_ins_root = gr.Textbox( + label=i18n("指定输出非主人声文件夹"), value="opt" + ) + format0 = gr.Radio( + label=i18n("导出文件格式"), + choices=["wav", "flac", "mp3", "m4a"], + value="flac", + interactive=True, + ) + but2 = gr.Button(i18n("转换"), variant="primary") + vc_output4 = gr.Textbox(label=i18n("输出信息")) + but2.click( + uvr, + [ + model_choose, + dir_wav_input, + opt_vocal_root, + wav_inputs, + opt_ins_root, + agg, + format0, + ], + [vc_output4], + api_name="uvr_convert", + ) + with gr.TabItem(i18n("训练")): + gr.Markdown( + value=i18n( + "step1: 填写实验配置. 实验数据放在logs下, 每个实验一个文件夹, 需手工输入实验名路径, 内含实验配置, 日志, 训练得到的模型文件. " + ) + ) + with gr.Row(): + exp_dir1 = gr.Textbox(label=i18n("输入实验名"), value="mi-test") + sr2 = gr.Radio( + label=i18n("目标采样率"), + choices=["40k", "48k"], + value="40k", + interactive=True, + ) + if_f0_3 = gr.Radio( + label=i18n("模型是否带音高指导(唱歌一定要, 语音可以不要)"), + choices=[True, False], + value=True, + interactive=True, + ) + version19 = gr.Radio( + label=i18n("版本"), + choices=["v1", "v2"], + value="v2", + interactive=True, + visible=True, + ) + np7 = gr.Slider( + minimum=0, + maximum=config.n_cpu, + step=1, + label=i18n("提取音高和处理数据使用的CPU进程数"), + value=int(np.ceil(config.n_cpu / 1.5)), + interactive=True, + ) + with gr.Group(): # 暂时单人的, 后面支持最多4人的#数据处理 + gr.Markdown( + value=i18n( + "step2a: 自动遍历训练文件夹下所有可解码成音频的文件并进行切片归一化, 在实验目录下生成2个wav文件夹; 暂时只支持单人训练. " + ) + ) + with gr.Row(): + trainset_dir4 = gr.Textbox( + label=i18n("输入训练文件夹路径"), value="E:\\语音音频+标注\\米津玄师\\src" + ) + spk_id5 = gr.Slider( + minimum=0, + maximum=4, + step=1, + label=i18n("请指定说话人id"), + value=0, + interactive=True, + ) + but1 = gr.Button(i18n("处理数据"), variant="primary") + info1 = gr.Textbox(label=i18n("输出信息"), value="") + but1.click( + preprocess_dataset, + [trainset_dir4, exp_dir1, sr2, np7], + [info1], + api_name="train_preprocess", + ) + with gr.Group(): + gr.Markdown(value=i18n("step2b: 使用CPU提取音高(如果模型带音高), 使用GPU提取特征(选择卡号)")) + with gr.Row(): + with gr.Column(): + gpus6 = gr.Textbox( + label=i18n("以-分隔输入使用的卡号, 例如 0-1-2 使用卡0和卡1和卡2"), + value=gpus, + interactive=True,visible=F0GPUVisible + ) + gpu_info9 = gr.Textbox(label=i18n("显卡信息"), value=gpu_info,visible=F0GPUVisible) + with gr.Column(): + f0method8 = gr.Radio( + label=i18n( + "选择音高提取算法:输入歌声可用pm提速,高质量语音但CPU差可用dio提速,harvest质量更好但慢,rmvpe效果最好且微吃CPU/GPU" + ), + choices=["pm", "harvest", "dio", "rmvpe", "rmvpe_gpu"], + value="rmvpe_gpu", + interactive=True, + ) + gpus_rmvpe = gr.Textbox( + label=i18n( + "rmvpe卡号配置：以-分隔输入使用的不同进程卡号,例如0-0-1使用在卡0上跑2个进程并在卡1上跑1个进程" + ), + value="%s-%s" % (gpus, gpus), + interactive=True, + visible=F0GPUVisible, + ) + but2 = gr.Button(i18n("特征提取"), variant="primary") + info2 = gr.Textbox(label=i18n("输出信息"), value="", max_lines=8) + f0method8.change( + fn=change_f0_method, + inputs=[f0method8], + outputs=[gpus_rmvpe], + ) + but2.click( + extract_f0_feature, + [ + gpus6, + np7, + f0method8, + if_f0_3, + exp_dir1, + version19, + gpus_rmvpe, + ], + [info2], + api_name="train_extract_f0_feature", + ) + with gr.Group(): + gr.Markdown(value=i18n("step3: 填写训练设置, 开始训练模型和索引")) + with gr.Row(): + save_epoch10 = gr.Slider( + minimum=0, + maximum=50, + step=1, + label=i18n("保存频率save_every_epoch"), + value=5, + interactive=True, + ) + total_epoch11 = gr.Slider( + minimum=0, + maximum=1000, + step=1, + label=i18n("总训练轮数total_epoch"), + value=20, + interactive=True, + ) + batch_size12 = gr.Slider( + minimum=1, + maximum=40, + step=1, + label=i18n("每张显卡的batch_size"), + value=default_batch_size, + interactive=True, + ) + if_save_latest13 = gr.Radio( + label=i18n("是否仅保存最新的ckpt文件以节省硬盘空间"), + choices=[i18n("是"), i18n("否")], + value=i18n("否"), + interactive=True, + ) + if_cache_gpu17 = gr.Radio( + label=i18n( + "是否缓存所有训练集至显存. 10min以下小数据可缓存以加速训练, 大数据缓存会炸显存也加不了多少速" + ), + choices=[i18n("是"), i18n("否")], + value=i18n("否"), + interactive=True, + ) + if_save_every_weights18 = gr.Radio( + label=i18n("是否在每次保存时间点将最终小模型保存至weights文件夹"), + choices=[i18n("是"), i18n("否")], + value=i18n("否"), + interactive=True, + ) + with gr.Row(): + pretrained_G14 = gr.Textbox( + label=i18n("加载预训练底模G路径"), + value="pretrained_v2/f0G40k.pth", + interactive=True, + ) + pretrained_D15 = gr.Textbox( + label=i18n("加载预训练底模D路径"), + value="pretrained_v2/f0D40k.pth", + interactive=True, + ) + sr2.change( + change_sr2, + [sr2, if_f0_3, version19], + [pretrained_G14, pretrained_D15], + ) + version19.change( + change_version19, + [sr2, if_f0_3, version19], + [pretrained_G14, pretrained_D15, sr2], + ) + if_f0_3.change( + change_f0, + [if_f0_3, sr2, version19], + [f0method8, pretrained_G14, pretrained_D15], + ) + gpus16 = gr.Textbox( + label=i18n("以-分隔输入使用的卡号, 例如 0-1-2 使用卡0和卡1和卡2"), + value=gpus, + interactive=True, + ) + but3 = gr.Button(i18n("训练模型"), variant="primary") + but4 = gr.Button(i18n("训练特征索引"), variant="primary") + but5 = gr.Button(i18n("一键训练"), variant="primary") + info3 = gr.Textbox(label=i18n("输出信息"), value="", max_lines=10) + but3.click( + click_train, + [ + exp_dir1, + sr2, + if_f0_3, + spk_id5, + save_epoch10, + total_epoch11, + batch_size12, + if_save_latest13, + pretrained_G14, + pretrained_D15, + gpus16, + if_cache_gpu17, + if_save_every_weights18, + version19, + ], + info3, + api_name="train_start", + ) + but4.click(train_index, [exp_dir1, version19], info3) + but5.click( + train1key, + [ + exp_dir1, + sr2, + if_f0_3, + trainset_dir4, + spk_id5, + np7, + f0method8, + save_epoch10, + total_epoch11, + batch_size12, + if_save_latest13, + pretrained_G14, + pretrained_D15, + gpus16, + if_cache_gpu17, + if_save_every_weights18, + version19, + gpus_rmvpe, + ], + info3, + api_name="train_start_all", + ) + + with gr.TabItem(i18n("ckpt处理")): + with gr.Group(): + gr.Markdown(value=i18n("模型融合, 可用于测试音色融合")) + with gr.Row(): + ckpt_a = gr.Textbox(label=i18n("A模型路径"), value="", interactive=True) + ckpt_b = gr.Textbox(label=i18n("B模型路径"), value="", interactive=True) + alpha_a = gr.Slider( + minimum=0, + maximum=1, + label=i18n("A模型权重"), + value=0.5, + interactive=True, + ) + with gr.Row(): + sr_ = gr.Radio( + label=i18n("目标采样率"), + choices=["40k", "48k"], + value="40k", + interactive=True, + ) + if_f0_ = gr.Radio( + label=i18n("模型是否带音高指导"), + choices=[i18n("是"), i18n("否")], + value=i18n("是"), + interactive=True, + ) + info__ = gr.Textbox( + label=i18n("要置入的模型信息"), value="", max_lines=8, interactive=True + ) + name_to_save0 = gr.Textbox( + label=i18n("保存的模型名不带后缀"), + value="", + max_lines=1, + interactive=True, + ) + version_2 = gr.Radio( + label=i18n("模型版本型号"), + choices=["v1", "v2"], + value="v1", + interactive=True, + ) + with gr.Row(): + but6 = gr.Button(i18n("融合"), variant="primary") + info4 = gr.Textbox(label=i18n("输出信息"), value="", max_lines=8) + but6.click( + merge, + [ + ckpt_a, + ckpt_b, + alpha_a, + sr_, + if_f0_, + info__, + name_to_save0, + version_2, + ], + info4, + api_name="ckpt_merge", + ) # def merge(path1,path2,alpha1,sr,f0,info): + with gr.Group(): + gr.Markdown(value=i18n("修改模型信息(仅支持weights文件夹下提取的小模型文件)")) + with gr.Row(): + ckpt_path0 = gr.Textbox( + label=i18n("模型路径"), value="", interactive=True + ) + info_ = gr.Textbox( + label=i18n("要改的模型信息"), value="", max_lines=8, interactive=True + ) + name_to_save1 = gr.Textbox( + label=i18n("保存的文件名, 默认空为和源文件同名"), + value="", + max_lines=8, + interactive=True, + ) + with gr.Row(): + but7 = gr.Button(i18n("修改"), variant="primary") + info5 = gr.Textbox(label=i18n("输出信息"), value="", max_lines=8) + but7.click( + change_info, + [ckpt_path0, info_, name_to_save1], + info5, + api_name="ckpt_modify", + ) + with gr.Group(): + gr.Markdown(value=i18n("查看模型信息(仅支持weights文件夹下提取的小模型文件)")) + with gr.Row(): + ckpt_path1 = gr.Textbox( + label=i18n("模型路径"), value="", interactive=True + ) + but8 = gr.Button(i18n("查看"), variant="primary") + info6 = gr.Textbox(label=i18n("输出信息"), value="", max_lines=8) + but8.click(show_info, [ckpt_path1], info6, api_name="ckpt_show") + with gr.Group(): + gr.Markdown( + value=i18n( + "模型提取(输入logs文件夹下大文件模型路径),适用于训一半不想训了模型没有自动提取保存小文件模型,或者想测试中间模型的情况" + ) + ) + with gr.Row(): + ckpt_path2 = gr.Textbox( + label=i18n("模型路径"), + value="E:\\codes\\py39\\logs\\mi-test_f0_48k\\G_23333.pth", + interactive=True, + ) + save_name = gr.Textbox( + label=i18n("保存名"), value="", interactive=True + ) + sr__ = gr.Radio( + label=i18n("目标采样率"), + choices=["32k", "40k", "48k"], + value="40k", + interactive=True, + ) + if_f0__ = gr.Radio( + label=i18n("模型是否带音高指导,1是0否"), + choices=["1", "0"], + value="1", + interactive=True, + ) + version_1 = gr.Radio( + label=i18n("模型版本型号"), + choices=["v1", "v2"], + value="v2", + interactive=True, + ) + info___ = gr.Textbox( + label=i18n("要置入的模型信息"), value="", max_lines=8, interactive=True + ) + but9 = gr.Button(i18n("提取"), variant="primary") + info7 = gr.Textbox(label=i18n("输出信息"), value="", max_lines=8) + ckpt_path2.change( + change_info_, [ckpt_path2], [sr__, if_f0__, version_1] + ) + but9.click( + extract_small_model, + [ckpt_path2, save_name, sr__, if_f0__, info___, version_1], + info7, + api_name="ckpt_extract", + ) + + with gr.TabItem(i18n("Onnx导出")): + with gr.Row(): + ckpt_dir = gr.Textbox(label=i18n("RVC模型路径"), value="", interactive=True) + with gr.Row(): + onnx_dir = gr.Textbox( + label=i18n("Onnx输出路径"), value="", interactive=True + ) + with gr.Row(): + infoOnnx = gr.Label(label="info") + with gr.Row(): + butOnnx = gr.Button(i18n("导出Onnx模型"), variant="primary") + butOnnx.click( + export_onnx, [ckpt_dir, onnx_dir], infoOnnx, api_name="export_onnx" + ) + + tab_faq = i18n("常见问题解答") + with gr.TabItem(tab_faq): + try: + if tab_faq == "常见问题解答": + with open("docs/faq.md", "r", encoding="utf8") as f: + info = f.read() + else: + with open("docs/faq_en.md", "r", encoding="utf8") as f: + info = f.read() + gr.Markdown(value=info) + except: + gr.Markdown(traceback.format_exc()) + + # with gr.TabItem(i18n("招募音高曲线前端编辑器")): + # gr.Markdown(value=i18n("加开发群联系我xxxxx")) + # with gr.TabItem(i18n("点击查看交流、问题反馈群号")): + # gr.Markdown(value=i18n("xxxxx")) + + if config.iscolab: + app.queue(concurrency_count=511, max_size=1022).launch(share=True) + else: + app.queue(concurrency_count=511, max_size=1022).launch( + server_name="0.0.0.0", + inbrowser=not config.noautoopen, + server_port=config.listen_port, + quiet=True, + ) diff --git a/voice_conversion/rvc-v2-brandon/infer_batch_rvc.py b/voice_conversion/rvc-v2-brandon/infer_batch_rvc.py new file mode 100644 index 0000000..604853f --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/infer_batch_rvc.py @@ -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) diff --git a/voice_conversion/rvc-v2-brandon/infer_cli.py b/voice_conversion/rvc-v2-brandon/infer_cli.py new file mode 100644 index 0000000..63170ae --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/infer_cli.py @@ -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, + ) diff --git a/voice_conversion/rvc-v2-brandon/infer_uvr5.py b/voice_conversion/rvc-v2-brandon/infer_uvr5.py new file mode 100644 index 0000000..0ffdb5d --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/infer_uvr5.py @@ -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 = 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a/voice_conversion/rvc-v2-brandon/requirements-dml.txt b/voice_conversion/rvc-v2-brandon/requirements-dml.txt new file mode 100644 index 0000000..c56c4b8 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/requirements-dml.txt @@ -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 diff --git a/voice_conversion/rvc-v2-brandon/requirements-win-for-realtime_vc_gui-dml.txt b/voice_conversion/rvc-v2-brandon/requirements-win-for-realtime_vc_gui-dml.txt new file mode 100644 index 0000000..6514989 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/requirements-win-for-realtime_vc_gui-dml.txt @@ -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 \ No newline at end of file diff --git a/voice_conversion/rvc-v2-brandon/requirements-win-for-realtime_vc_gui.txt b/voice_conversion/rvc-v2-brandon/requirements-win-for-realtime_vc_gui.txt new file mode 100644 index 0000000..37ca238 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/requirements-win-for-realtime_vc_gui.txt @@ -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 diff --git a/voice_conversion/rvc-v2-brandon/requirements.txt b/voice_conversion/rvc-v2-brandon/requirements.txt new file mode 100644 index 0000000..b773f44 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/requirements.txt @@ -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 diff --git a/voice_conversion/rvc-v2-brandon/run.sh b/voice_conversion/rvc-v2-brandon/run.sh new file mode 100644 index 0000000..9f7186e --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/run.sh @@ -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 \ No newline at end of file diff --git a/voice_conversion/rvc-v2-brandon/rvc_for_realtime.py b/voice_conversion/rvc-v2-brandon/rvc_for_realtime.py new file mode 100644 index 0000000..3a4af04 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/rvc_for_realtime.py @@ -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 diff --git a/voice_conversion/rvc-v2-brandon/train_nsf_sim_cache_sid_load_pretrain.py b/voice_conversion/rvc-v2-brandon/train_nsf_sim_cache_sid_load_pretrain.py new file mode 100644 index 0000000..c1bdf11 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/train_nsf_sim_cache_sid_load_pretrain.py @@ -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() diff --git a/voice_conversion/rvc-v2-brandon/trainset_preprocess_pipeline_print.py b/voice_conversion/rvc-v2-brandon/trainset_preprocess_pipeline_print.py new file mode 100644 index 0000000..d9088a2 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/trainset_preprocess_pipeline_print.py @@ -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) diff --git a/voice_conversion/rvc-v2-brandon/values1.json b/voice_conversion/rvc-v2-brandon/values1.json new file mode 100644 index 0000000..06e5417 --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/values1.json @@ -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"} \ No newline at end of file diff --git a/voice_conversion/rvc-v2-brandon/vc_infer_pipeline.py b/voice_conversion/rvc-v2-brandon/vc_infer_pipeline.py new file mode 100644 index 0000000..87ab2ef --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/vc_infer_pipeline.py @@ -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 diff --git a/voice_conversion/rvc-v2-brandon/venv.sh b/voice_conversion/rvc-v2-brandon/venv.sh new file mode 100644 index 0000000..17f58bf --- /dev/null +++ b/voice_conversion/rvc-v2-brandon/venv.sh @@ -0,0 +1 @@ +python3 -m venv .venv