diff --git a/.dockerignore b/.dockerignore
new file mode 100644
index 0000000000000000000000000000000000000000..4eca27be6d8721a0bc619989952418132b3d59aa
--- /dev/null
+++ b/.dockerignore
@@ -0,0 +1,8 @@
+docs
+logs
+output
+reference
+SoVITS_weights
+GPT_weights
+TEMP
+.git
diff --git a/.gitignore b/.gitignore
new file mode 100644
index 0000000000000000000000000000000000000000..0bb4e0bf22a20558257328203d997569b4a83722
--- /dev/null
+++ b/.gitignore
@@ -0,0 +1,200 @@
+.DS_Store
+.vscode
+__pycache__
+*.pyc
+env
+runtime
+.idea
+output
+logs
+reference
+GPT_weights
+SoVITS_weights
+GPT_weights_v2
+SoVITS_weights_v2
+GPT_weights_v3
+SoVITS_weights_v3
+TEMP
+weight.json
+ffmpeg*
+ffprobe*
+cfg.json
+speakers.json
+ref_audios
+tools/AP_BWE_main/24kto48k/*
+!tools/AP_BWE_main/24kto48k/readme.txt
+
+# Byte-compiled / optimized / DLL files
+__pycache__/
+*.py[cod]
+*$py.class
+
+# C extensions
+*.so
+
+# Distribution / packaging
+.Python
+build/
+develop-eggs/
+dist/
+downloads/
+eggs/
+.eggs/
+lib/
+lib64/
+parts/
+sdist/
+var/
+wheels/
+share/python-wheels/
+*.egg-info/
+.installed.cfg
+*.egg
+MANIFEST
+
+# PyInstaller
+# Usually these files are written by a python script from a template
+# before PyInstaller builds the exe, so as to inject date/other infos into it.
+*.manifest
+*.spec
+
+# Installer logs
+pip-log.txt
+pip-delete-this-directory.txt
+
+# Unit test / coverage reports
+htmlcov/
+.tox/
+.nox/
+.coverage
+.coverage.*
+.cache
+nosetests.xml
+coverage.xml
+*.cover
+*.py,cover
+.hypothesis/
+.pytest_cache/
+cover/
+
+# Translations
+*.mo
+*.pot
+
+# Django stuff:
+*.log
+local_settings.py
+db.sqlite3
+db.sqlite3-journal
+
+# Flask stuff:
+instance/
+.webassets-cache
+
+# Scrapy stuff:
+.scrapy
+
+# Sphinx documentation
+docs/_build/
+
+# PyBuilder
+.pybuilder/
+target/
+
+# Jupyter Notebook
+.ipynb_checkpoints
+
+# IPython
+profile_default/
+ipython_config.py
+
+# pyenv
+# For a library or package, you might want to ignore these files since the code is
+# intended to run in multiple environments; otherwise, check them in:
+# .python-version
+
+# pipenv
+# According to pypa/pipenv#598, it is recommended to include Pipfile.lock in version control.
+# However, in case of collaboration, if having platform-specific dependencies or dependencies
+# having no cross-platform support, pipenv may install dependencies that don't work, or not
+# install all needed dependencies.
+#Pipfile.lock
+
+# UV
+# Similar to Pipfile.lock, it is generally recommended to include uv.lock in version control.
+# This is especially recommended for binary packages to ensure reproducibility, and is more
+# commonly ignored for libraries.
+#uv.lock
+
+# poetry
+# Similar to Pipfile.lock, it is generally recommended to include poetry.lock in version control.
+# This is especially recommended for binary packages to ensure reproducibility, and is more
+# commonly ignored for libraries.
+# https://python-poetry.org/docs/basic-usage/#commit-your-poetrylock-file-to-version-control
+#poetry.lock
+
+# pdm
+# Similar to Pipfile.lock, it is generally recommended to include pdm.lock in version control.
+#pdm.lock
+# pdm stores project-wide configurations in .pdm.toml, but it is recommended to not include it
+# in version control.
+# https://pdm.fming.dev/latest/usage/project/#working-with-version-control
+.pdm.toml
+.pdm-python
+.pdm-build/
+
+# PEP 582; used by e.g. github.com/David-OConnor/pyflow and github.com/pdm-project/pdm
+__pypackages__/
+
+# Celery stuff
+celerybeat-schedule
+celerybeat.pid
+
+# SageMath parsed files
+*.sage.py
+
+# Environments
+.env
+.venv
+env/
+venv/
+ENV/
+env.bak/
+venv.bak/
+
+# Spyder project settings
+.spyderproject
+.spyproject
+
+# Rope project settings
+.ropeproject
+
+# mkdocs documentation
+/site
+
+# mypy
+.mypy_cache/
+.dmypy.json
+dmypy.json
+
+# Pyre type checker
+.pyre/
+
+# pytype static type analyzer
+.pytype/
+
+# Cython debug symbols
+cython_debug/
+
+# PyCharm
+# JetBrains specific template is maintained in a separate JetBrains.gitignore that can
+# be found at https://github.com/github/gitignore/blob/main/Global/JetBrains.gitignore
+# and can be added to the global gitignore or merged into this file. For a more nuclear
+# option (not recommended) you can uncomment the following to ignore the entire idea folder.
+#.idea/
+
+# Ruff stuff:
+.ruff_cache/
+
+# PyPI configuration file
+.pypirc
diff --git a/Docker/damo.sha256 b/Docker/damo.sha256
new file mode 100644
index 0000000000000000000000000000000000000000..6e9804da4a02caf45bac51adfb9a9b75de7c16d0
--- /dev/null
+++ b/Docker/damo.sha256
@@ -0,0 +1,3 @@
+5bba782a5e9196166233b9ab12ba04cadff9ef9212b4ff6153ed9290ff679025 /workspace/tools/damo_asr/models/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch/model.pb
+b3be75be477f0780277f3bae0fe489f48718f585f3a6e45d7dd1fbb1a4255fc5 /workspace/tools/damo_asr/models/speech_fsmn_vad_zh-cn-16k-common-pytorch/model.pb
+a5818bb9d933805a916eebe41eb41648f7f9caad30b4bd59d56f3ca135421916 /workspace/tools/damo_asr/models/punc_ct-transformer_zh-cn-common-vocab272727-pytorch/model.pb
\ No newline at end of file
diff --git a/Docker/download.py b/Docker/download.py
new file mode 100644
index 0000000000000000000000000000000000000000..952423d1edc0b048db3865a0f25516b9726773ac
--- /dev/null
+++ b/Docker/download.py
@@ -0,0 +1,8 @@
+# Download moda ASR related models
+from modelscope import snapshot_download
+
+model_dir = snapshot_download(
+ "damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch", revision="v2.0.4"
+)
+model_dir = snapshot_download("damo/speech_fsmn_vad_zh-cn-16k-common-pytorch", revision="v2.0.4")
+model_dir = snapshot_download("damo/punc_ct-transformer_zh-cn-common-vocab272727-pytorch", revision="v2.0.4")
diff --git a/Docker/download.sh b/Docker/download.sh
new file mode 100644
index 0000000000000000000000000000000000000000..447e018ead5725d2a220234398161702ff12c5f8
--- /dev/null
+++ b/Docker/download.sh
@@ -0,0 +1,11 @@
+#!/usr/bin/env bash
+
+set -Eeuo pipefail
+
+echo "Downloading models..."
+
+aria2c --disable-ipv6 --input-file /workspace/Docker/links.txt --dir /workspace --continue
+
+echo "Checking SHA256..."
+
+parallel --will-cite -a /workspace/Docker/links.sha256 "echo -n {} | sha256sum -c"
diff --git a/Docker/links.sha256 b/Docker/links.sha256
new file mode 100644
index 0000000000000000000000000000000000000000..cda6dc1552e87ff035bfb27ffcac76f429c780f8
--- /dev/null
+++ b/Docker/links.sha256
@@ -0,0 +1,12 @@
+b1c1e17e9c99547a89388f72048cd6e1b41b5a18b170e86a46dfde0324d63eb1 /workspace/GPT_SoVITS/pretrained_models/s1bert25hz-2kh-longer-epoch=68e-step=50232.ckpt
+fc579c1db3c1e21b721001cf99d7a584214280df19b002e200b630a34fa06eb8 /workspace/GPT_SoVITS/pretrained_models/s2D488k.pth
+020a014e1e01e550e510f2f61fae5e5f5b6aab40f15c22f1f12f724df507e835 /workspace/GPT_SoVITS/pretrained_models/s2G488k.pth
+24164f129c66499d1346e2aa55f183250c223161ec2770c0da3d3b08cf432d3c /workspace/GPT_SoVITS/pretrained_models/chinese-hubert-base/pytorch_model.bin
+e53a693acc59ace251d143d068096ae0d7b79e4b1b503fa84c9dcf576448c1d8 /workspace/GPT_SoVITS/pretrained_models/chinese-roberta-wwm-ext-large/pytorch_model.bin
+39796caa5db18d7f9382d8ac997ac967bfd85f7761014bb807d2543cc844ef05 /workspace/tools/uvr5/uvr5_weights/HP2_all_vocals.pth
+45e6b65199e781b4a6542002699be9f19cd3d1cb7d1558bc2bfbcd84674dfe28 /workspace/tools/uvr5/uvr5_weights/HP3_all_vocals.pth
+5908891829634926119720241e8573d97cbeb8277110a7512bdb0bd7563258ee /workspace/tools/uvr5/uvr5_weights/HP5_only_main_vocal.pth
+8c8fd1582f9aabc363e47af62ddb88df6cae7e064cae75bbf041a067a5e0aee2 /workspace/tools/uvr5/uvr5_weights/VR-DeEchoAggressive.pth
+01376dd2a571bf3cb9cced680732726d2d732609d09216a610b0d110f133febe /workspace/tools/uvr5/uvr5_weights/VR-DeEchoDeReverb.pth
+56aba59db3bcdd14a14464e62f3129698ecdea62eee0f003b9360923eb3ac79e /workspace/tools/uvr5/uvr5_weights/VR-DeEchoNormal.pth
+233bb5c6aaa365e568659a0a81211746fa881f8f47f82d9e864fce1f7692db80 /workspace/tools/uvr5/uvr5_weights/onnx_dereverb_By_FoxJoy/vocals.onnx
\ No newline at end of file
diff --git a/Docker/links.txt b/Docker/links.txt
new file mode 100644
index 0000000000000000000000000000000000000000..e6603db0c027547b9bf85d1e8f54837fbfe0b076
--- /dev/null
+++ b/Docker/links.txt
@@ -0,0 +1,34 @@
+# GPT-SoVITS models
+https://huggingface.co/lj1995/GPT-SoVITS/resolve/main/s1bert25hz-2kh-longer-epoch%3D68e-step%3D50232.ckpt
+ out=GPT_SoVITS/pretrained_models/s1bert25hz-2kh-longer-epoch=68e-step=50232.ckpt
+https://huggingface.co/lj1995/GPT-SoVITS/resolve/main/s2D488k.pth
+ out=GPT_SoVITS/pretrained_models/s2D488k.pth
+https://huggingface.co/lj1995/GPT-SoVITS/resolve/main/s2G488k.pth
+ out=GPT_SoVITS/pretrained_models/s2G488k.pth
+https://huggingface.co/lj1995/GPT-SoVITS/resolve/main/chinese-hubert-base/config.json
+ out=GPT_SoVITS/pretrained_models/chinese-hubert-base/config.json
+https://huggingface.co/lj1995/GPT-SoVITS/resolve/main/chinese-hubert-base/preprocessor_config.json
+ out=GPT_SoVITS/pretrained_models/chinese-hubert-base/preprocessor_config.json
+https://huggingface.co/lj1995/GPT-SoVITS/resolve/main/chinese-hubert-base/pytorch_model.bin
+ out=GPT_SoVITS/pretrained_models/chinese-hubert-base/pytorch_model.bin
+https://huggingface.co/lj1995/GPT-SoVITS/resolve/main/chinese-roberta-wwm-ext-large/config.json
+ out=GPT_SoVITS/pretrained_models/chinese-roberta-wwm-ext-large/config.json
+https://huggingface.co/lj1995/GPT-SoVITS/resolve/main/chinese-roberta-wwm-ext-large/pytorch_model.bin
+ out=GPT_SoVITS/pretrained_models/chinese-roberta-wwm-ext-large/pytorch_model.bin
+https://huggingface.co/lj1995/GPT-SoVITS/resolve/main/chinese-roberta-wwm-ext-large/tokenizer.json
+ out=GPT_SoVITS/pretrained_models/chinese-roberta-wwm-ext-large/tokenizer.json
+# UVR5
+https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/uvr5_weights/HP2_all_vocals.pth
+ out=tools/uvr5/uvr5_weights/HP2_all_vocals.pth
+https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/uvr5_weights/HP3_all_vocals.pth
+ out=tools/uvr5/uvr5_weights/HP3_all_vocals.pth
+https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/uvr5_weights/HP5_only_main_vocal.pth
+ out=tools/uvr5/uvr5_weights/HP5_only_main_vocal.pth
+https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/uvr5_weights/VR-DeEchoAggressive.pth
+ out=tools/uvr5/uvr5_weights/VR-DeEchoAggressive.pth
+https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/uvr5_weights/VR-DeEchoDeReverb.pth
+ out=tools/uvr5/uvr5_weights/VR-DeEchoDeReverb.pth
+https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/uvr5_weights/VR-DeEchoNormal.pth
+ out=tools/uvr5/uvr5_weights/VR-DeEchoNormal.pth
+https://huggingface.co/lj1995/VoiceConversionWebUI/resolve/main/uvr5_weights/onnx_dereverb_By_FoxJoy/vocals.onnx
+ out=tools/uvr5/uvr5_weights/onnx_dereverb_By_FoxJoy/vocals.onnx
\ No newline at end of file
diff --git a/Dockerfile b/Dockerfile
new file mode 100644
index 0000000000000000000000000000000000000000..80cd9f3a106c9c997c23c0aa7c8fecab7039001b
--- /dev/null
+++ b/Dockerfile
@@ -0,0 +1,42 @@
+# Base CUDA image
+FROM cnstark/pytorch:2.0.1-py3.9.17-cuda11.8.0-ubuntu20.04
+
+LABEL maintainer="breakstring@hotmail.com"
+LABEL version="dev-20240209"
+LABEL description="Docker image for GPT-SoVITS"
+
+
+# Install 3rd party apps
+ENV DEBIAN_FRONTEND=noninteractive
+ENV TZ=Etc/UTC
+RUN apt-get update && \
+ apt-get install -y --no-install-recommends tzdata ffmpeg libsox-dev parallel aria2 git git-lfs && \
+ git lfs install && \
+ rm -rf /var/lib/apt/lists/*
+
+# Copy only requirements.txt initially to leverage Docker cache
+WORKDIR /workspace
+COPY requirements.txt /workspace/
+RUN pip install --no-cache-dir -r requirements.txt
+
+# Define a build-time argument for image type
+ARG IMAGE_TYPE=full
+
+# Conditional logic based on the IMAGE_TYPE argument
+# Always copy the Docker directory, but only use it if IMAGE_TYPE is not "elite"
+COPY ./Docker /workspace/Docker
+# elite 类型的镜像里面不包含额外的模型
+RUN if [ "$IMAGE_TYPE" != "elite" ]; then \
+ chmod +x /workspace/Docker/download.sh && \
+ /workspace/Docker/download.sh && \
+ python /workspace/Docker/download.py && \
+ python -m nltk.downloader averaged_perceptron_tagger cmudict; \
+ fi
+
+
+# Copy the rest of the application
+COPY . /workspace
+
+EXPOSE 9871 9872 9873 9874 9880
+
+CMD ["python", "webui.py"]
diff --git a/GPT_SoVITS/BigVGAN/LICENSE b/GPT_SoVITS/BigVGAN/LICENSE
new file mode 100644
index 0000000000000000000000000000000000000000..4c78361c86d4f685117d60d6623e2197fcfed706
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/LICENSE
@@ -0,0 +1,21 @@
+MIT License
+
+Copyright (c) 2024 NVIDIA CORPORATION.
+
+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/GPT_SoVITS/BigVGAN/README.md b/GPT_SoVITS/BigVGAN/README.md
new file mode 100644
index 0000000000000000000000000000000000000000..2fa70ceea647053933b913b329041ee8c41526db
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/README.md
@@ -0,0 +1,266 @@
+## BigVGAN: A Universal Neural Vocoder with Large-Scale Training
+
+#### Sang-gil Lee, Wei Ping, Boris Ginsburg, Bryan Catanzaro, Sungroh Yoon
+
+[[Paper]](https://arxiv.org/abs/2206.04658) - [[Code]](https://github.com/NVIDIA/BigVGAN) - [[Showcase]](https://bigvgan-demo.github.io/) - [[Project Page]](https://research.nvidia.com/labs/adlr/projects/bigvgan/) - [[Weights]](https://huggingface.co/collections/nvidia/bigvgan-66959df3d97fd7d98d97dc9a) - [[Demo]](https://huggingface.co/spaces/nvidia/BigVGAN)
+
+[](https://paperswithcode.com/sota/speech-synthesis-on-libritts?p=bigvgan-a-universal-neural-vocoder-with-large)
+
+
+
+## News
+- **Sep 2024 (v2.4):**
+ - We have updated the pretrained checkpoints trained for 5M steps. This is final release of the BigVGAN-v2 checkpoints.
+
+- **Jul 2024 (v2.3):**
+ - General refactor and code improvements for improved readability.
+ - Fully fused CUDA kernel of anti-alised activation (upsampling + activation + downsampling) with inference speed benchmark.
+
+- **Jul 2024 (v2.2):** The repository now includes an interactive local demo using gradio.
+
+- **Jul 2024 (v2.1):** BigVGAN is now integrated with 🤗 Hugging Face Hub with easy access to inference using pretrained checkpoints. We also provide an interactive demo on Hugging Face Spaces.
+
+- **Jul 2024 (v2):** We release BigVGAN-v2 along with pretrained checkpoints. Below are the highlights:
+ - Custom CUDA kernel for inference: we provide a fused upsampling + activation kernel written in CUDA for accelerated inference speed. Our test shows 1.5 - 3x faster speed on a single A100 GPU.
+ - Improved discriminator and loss: BigVGAN-v2 is trained using a [multi-scale sub-band CQT discriminator](https://arxiv.org/abs/2311.14957) and a [multi-scale mel spectrogram loss](https://arxiv.org/abs/2306.06546).
+ - Larger training data: BigVGAN-v2 is trained using datasets containing diverse audio types, including speech in multiple languages, environmental sounds, and instruments.
+ - We provide pretrained checkpoints of BigVGAN-v2 using diverse audio configurations, supporting up to 44 kHz sampling rate and 512x upsampling ratio.
+
+## Installation
+
+The codebase has been tested on Python `3.10` and PyTorch `2.3.1` conda packages with either `pytorch-cuda=12.1` or `pytorch-cuda=11.8`. Below is an example command to create the conda environment:
+
+```shell
+conda create -n bigvgan python=3.10 pytorch torchvision torchaudio pytorch-cuda=12.1 -c pytorch -c nvidia
+conda activate bigvgan
+```
+
+Clone the repository and install dependencies:
+
+```shell
+git clone https://github.com/NVIDIA/BigVGAN
+cd BigVGAN
+pip install -r requirements.txt
+```
+
+## Inference Quickstart using 🤗 Hugging Face Hub
+
+Below example describes how you can use BigVGAN: load the pretrained BigVGAN generator from Hugging Face Hub, compute mel spectrogram from input waveform, and generate synthesized waveform using the mel spectrogram as the model's input.
+
+```python
+device = 'cuda'
+
+import torch
+import bigvgan
+import librosa
+from meldataset import get_mel_spectrogram
+
+# instantiate the model. You can optionally set use_cuda_kernel=True for faster inference.
+model = bigvgan.BigVGAN.from_pretrained('nvidia/bigvgan_v2_24khz_100band_256x', use_cuda_kernel=False)
+
+# remove weight norm in the model and set to eval mode
+model.remove_weight_norm()
+model = model.eval().to(device)
+
+# load wav file and compute mel spectrogram
+wav_path = '/path/to/your/audio.wav'
+wav, sr = librosa.load(wav_path, sr=model.h.sampling_rate, mono=True) # wav is np.ndarray with shape [T_time] and values in [-1, 1]
+wav = torch.FloatTensor(wav).unsqueeze(0) # wav is FloatTensor with shape [B(1), T_time]
+
+# compute mel spectrogram from the ground truth audio
+mel = get_mel_spectrogram(wav, model.h).to(device) # mel is FloatTensor with shape [B(1), C_mel, T_frame]
+
+# generate waveform from mel
+with torch.inference_mode():
+ wav_gen = model(mel) # wav_gen is FloatTensor with shape [B(1), 1, T_time] and values in [-1, 1]
+wav_gen_float = wav_gen.squeeze(0).cpu() # wav_gen is FloatTensor with shape [1, T_time]
+
+# you can convert the generated waveform to 16 bit linear PCM
+wav_gen_int16 = (wav_gen_float * 32767.0).numpy().astype('int16') # wav_gen is now np.ndarray with shape [1, T_time] and int16 dtype
+```
+
+## Local gradio demo
+
+You can run a local gradio demo using below command:
+
+```python
+pip install -r demo/requirements.txt
+python demo/app.py
+```
+
+## Training
+
+Create symbolic link to the root of the dataset. The codebase uses filelist with the relative path from the dataset. Below are the example commands for LibriTTS dataset:
+
+```shell
+cd filelists/LibriTTS && \
+ln -s /path/to/your/LibriTTS/train-clean-100 train-clean-100 && \
+ln -s /path/to/your/LibriTTS/train-clean-360 train-clean-360 && \
+ln -s /path/to/your/LibriTTS/train-other-500 train-other-500 && \
+ln -s /path/to/your/LibriTTS/dev-clean dev-clean && \
+ln -s /path/to/your/LibriTTS/dev-other dev-other && \
+ln -s /path/to/your/LibriTTS/test-clean test-clean && \
+ln -s /path/to/your/LibriTTS/test-other test-other && \
+cd ../..
+```
+
+Train BigVGAN model. Below is an example command for training BigVGAN-v2 using LibriTTS dataset at 24kHz with a full 100-band mel spectrogram as input:
+
+```shell
+python train.py \
+--config configs/bigvgan_v2_24khz_100band_256x.json \
+--input_wavs_dir filelists/LibriTTS \
+--input_training_file filelists/LibriTTS/train-full.txt \
+--input_validation_file filelists/LibriTTS/val-full.txt \
+--list_input_unseen_wavs_dir filelists/LibriTTS filelists/LibriTTS \
+--list_input_unseen_validation_file filelists/LibriTTS/dev-clean.txt filelists/LibriTTS/dev-other.txt \
+--checkpoint_path exp/bigvgan_v2_24khz_100band_256x
+```
+
+## Synthesis
+
+Synthesize from BigVGAN model. Below is an example command for generating audio from the model.
+It computes mel spectrograms using wav files from `--input_wavs_dir` and saves the generated audio to `--output_dir`.
+
+```shell
+python inference.py \
+--checkpoint_file /path/to/your/bigvgan_v2_24khz_100band_256x/bigvgan_generator.pt \
+--input_wavs_dir /path/to/your/input_wav \
+--output_dir /path/to/your/output_wav
+```
+
+`inference_e2e.py` supports synthesis directly from the mel spectrogram saved in `.npy` format, with shapes `[1, channel, frame]` or `[channel, frame]`.
+It loads mel spectrograms from `--input_mels_dir` and saves the generated audio to `--output_dir`.
+
+Make sure that the STFT hyperparameters for mel spectrogram are the same as the model, which are defined in `config.json` of the corresponding model.
+
+```shell
+python inference_e2e.py \
+--checkpoint_file /path/to/your/bigvgan_v2_24khz_100band_256x/bigvgan_generator.pt \
+--input_mels_dir /path/to/your/input_mel \
+--output_dir /path/to/your/output_wav
+```
+
+## Using Custom CUDA Kernel for Synthesis
+
+You can apply the fast CUDA inference kernel by using a parameter `use_cuda_kernel` when instantiating BigVGAN:
+
+```python
+generator = BigVGAN(h, use_cuda_kernel=True)
+```
+
+You can also pass `--use_cuda_kernel` to `inference.py` and `inference_e2e.py` to enable this feature.
+
+When applied for the first time, it builds the kernel using `nvcc` and `ninja`. If the build succeeds, the kernel is saved to `alias_free_activation/cuda/build` and the model automatically loads the kernel. The codebase has been tested using CUDA `12.1`.
+
+Please make sure that both are installed in your system and `nvcc` installed in your system matches the version your PyTorch build is using.
+
+We recommend running `test_cuda_vs_torch_model.py` first to build and check the correctness of the CUDA kernel. See below example command and its output, where it returns `[Success] test CUDA fused vs. plain torch BigVGAN inference`:
+
+```python
+python tests/test_cuda_vs_torch_model.py \
+--checkpoint_file /path/to/your/bigvgan_generator.pt
+```
+
+```shell
+loading plain Pytorch BigVGAN
+...
+loading CUDA kernel BigVGAN with auto-build
+Detected CUDA files, patching ldflags
+Emitting ninja build file /path/to/your/BigVGAN/alias_free_activation/cuda/build/build.ninja..
+Building extension module anti_alias_activation_cuda...
+...
+Loading extension module anti_alias_activation_cuda...
+...
+Loading '/path/to/your/bigvgan_generator.pt'
+...
+[Success] test CUDA fused vs. plain torch BigVGAN inference
+ > mean_difference=0.0007238413265440613
+...
+```
+
+If you see `[Fail] test CUDA fused vs. plain torch BigVGAN inference`, it means that the CUDA kernel inference is incorrect. Please check if `nvcc` installed in your system is compatible with your PyTorch version.
+
+## Pretrained Models
+
+We provide the [pretrained models on Hugging Face Collections](https://huggingface.co/collections/nvidia/bigvgan-66959df3d97fd7d98d97dc9a).
+One can download the checkpoints of the generator weight (named `bigvgan_generator.pt`) and its discriminator/optimizer states (named `bigvgan_discriminator_optimizer.pt`) within the listed model repositories.
+
+| Model Name | Sampling Rate | Mel band | fmax | Upsampling Ratio | Params | Dataset | Steps | Fine-Tuned |
+|:--------------------------------------------------------------------------------------------------------:|:-------------:|:--------:|:-----:|:----------------:|:------:|:--------------------------:|:-----:|:----------:|
+| [bigvgan_v2_44khz_128band_512x](https://huggingface.co/nvidia/bigvgan_v2_44khz_128band_512x) | 44 kHz | 128 | 22050 | 512 | 122M | Large-scale Compilation | 5M | No |
+| [bigvgan_v2_44khz_128band_256x](https://huggingface.co/nvidia/bigvgan_v2_44khz_128band_256x) | 44 kHz | 128 | 22050 | 256 | 112M | Large-scale Compilation | 5M | No |
+| [bigvgan_v2_24khz_100band_256x](https://huggingface.co/nvidia/bigvgan_v2_24khz_100band_256x) | 24 kHz | 100 | 12000 | 256 | 112M | Large-scale Compilation | 5M | No |
+| [bigvgan_v2_22khz_80band_256x](https://huggingface.co/nvidia/bigvgan_v2_22khz_80band_256x) | 22 kHz | 80 | 11025 | 256 | 112M | Large-scale Compilation | 5M | No |
+| [bigvgan_v2_22khz_80band_fmax8k_256x](https://huggingface.co/nvidia/bigvgan_v2_22khz_80band_fmax8k_256x) | 22 kHz | 80 | 8000 | 256 | 112M | Large-scale Compilation | 5M | No |
+| [bigvgan_24khz_100band](https://huggingface.co/nvidia/bigvgan_24khz_100band) | 24 kHz | 100 | 12000 | 256 | 112M | LibriTTS | 5M | No |
+| [bigvgan_base_24khz_100band](https://huggingface.co/nvidia/bigvgan_base_24khz_100band) | 24 kHz | 100 | 12000 | 256 | 14M | LibriTTS | 5M | No |
+| [bigvgan_22khz_80band](https://huggingface.co/nvidia/bigvgan_22khz_80band) | 22 kHz | 80 | 8000 | 256 | 112M | LibriTTS + VCTK + LJSpeech | 5M | No |
+| [bigvgan_base_22khz_80band](https://huggingface.co/nvidia/bigvgan_base_22khz_80band) | 22 kHz | 80 | 8000 | 256 | 14M | LibriTTS + VCTK + LJSpeech | 5M | No |
+
+The paper results are based on the original 24kHz BigVGAN models (`bigvgan_24khz_100band` and `bigvgan_base_24khz_100band`) trained on LibriTTS dataset.
+We also provide 22kHz BigVGAN models with band-limited setup (i.e., fmax=8000) for TTS applications.
+Note that the checkpoints use `snakebeta` activation with log scale parameterization, which have the best overall quality.
+
+You can fine-tune the models by:
+
+1. downloading the checkpoints (both the generator weight and its discriminator/optimizer states)
+2. resuming training using your audio dataset by specifying `--checkpoint_path` that includes the checkpoints when launching `train.py`
+
+## Training Details of BigVGAN-v2
+
+Comapred to the original BigVGAN, the pretrained checkpoints of BigVGAN-v2 used `batch_size=32` with a longer `segment_size=65536` and are trained using 8 A100 GPUs.
+
+Note that the BigVGAN-v2 `json` config files in `./configs` use `batch_size=4` as default to fit in a single A100 GPU for training. You can fine-tune the models adjusting `batch_size` depending on your GPUs.
+
+When training BigVGAN-v2 from scratch with small batch size, it can potentially encounter the early divergence problem mentioned in the paper. In such case, we recommend lowering the `clip_grad_norm` value (e.g. `100`) for the early training iterations (e.g. 20000 steps) and increase the value to the default `500`.
+
+## Evaluation Results of BigVGAN-v2
+
+Below are the objective results of the 24kHz model (`bigvgan_v2_24khz_100band_256x`) obtained from the LibriTTS `dev` sets. BigVGAN-v2 shows noticeable improvements of the metrics. The model also exhibits reduced perceptual artifacts, especially for non-speech audio.
+
+| Model | Dataset | Steps | PESQ(↑) | M-STFT(↓) | MCD(↓) | Periodicity(↓) | V/UV F1(↑) |
+|:----------:|:-----------------------:|:-----:|:---------:|:----------:|:----------:|:--------------:|:----------:|
+| BigVGAN | LibriTTS | 1M | 4.027 | 0.7997 | 0.3745 | 0.1018 | 0.9598 |
+| BigVGAN | LibriTTS | 5M | 4.256 | 0.7409 | 0.2988 | 0.0809 | 0.9698 |
+| BigVGAN-v2 | Large-scale Compilation | 3M | 4.359 | 0.7134 | 0.3060 | 0.0621 | 0.9777 |
+| BigVGAN-v2 | Large-scale Compilation | 5M | **4.362** | **0.7026** | **0.2903** | **0.0593** | **0.9793** |
+
+## Speed Benchmark
+
+Below are the speed and VRAM usage benchmark results of BigVGAN from `tests/test_cuda_vs_torch_model.py`, using `bigvgan_v2_24khz_100band_256x` as a reference model.
+
+| GPU | num_mel_frame | use_cuda_kernel | Speed (kHz) | Real-time Factor | VRAM (GB) |
+|:--------------------------:|:-------------:|:---------------:|:-----------:|:----------------:|:---------:|
+| NVIDIA A100 | 256 | False | 1672.1 | 69.7x | 1.3 |
+| | | True | 3916.5 | 163.2x | 1.3 |
+| | 2048 | False | 1899.6 | 79.2x | 1.7 |
+| | | True | 5330.1 | 222.1x | 1.7 |
+| | 16384 | False | 1973.8 | 82.2x | 5.0 |
+| | | True | 5761.7 | 240.1x | 4.4 |
+| NVIDIA GeForce RTX 3080 | 256 | False | 841.1 | 35.0x | 1.3 |
+| | | True | 1598.1 | 66.6x | 1.3 |
+| | 2048 | False | 929.9 | 38.7x | 1.7 |
+| | | True | 1971.3 | 82.1x | 1.6 |
+| | 16384 | False | 943.4 | 39.3x | 5.0 |
+| | | True | 2026.5 | 84.4x | 3.9 |
+| NVIDIA GeForce RTX 2080 Ti | 256 | False | 515.6 | 21.5x | 1.3 |
+| | | True | 811.3 | 33.8x | 1.3 |
+| | 2048 | False | 576.5 | 24.0x | 1.7 |
+| | | True | 1023.0 | 42.6x | 1.5 |
+| | 16384 | False | 589.4 | 24.6x | 5.0 |
+| | | True | 1068.1 | 44.5x | 3.2 |
+
+## Acknowledgements
+
+We thank Vijay Anand Korthikanti and Kevin J. Shih for their generous support in implementing the CUDA kernel for inference.
+
+## References
+
+- [HiFi-GAN](https://github.com/jik876/hifi-gan) (for generator and multi-period discriminator)
+- [Snake](https://github.com/EdwardDixon/snake) (for periodic activation)
+- [Alias-free-torch](https://github.com/junjun3518/alias-free-torch) (for anti-aliasing)
+- [Julius](https://github.com/adefossez/julius) (for low-pass filter)
+- [UnivNet](https://github.com/mindslab-ai/univnet) (for multi-resolution discriminator)
+- [descript-audio-codec](https://github.com/descriptinc/descript-audio-codec) and [vocos](https://github.com/gemelo-ai/vocos) (for multi-band multi-scale STFT discriminator and multi-scale mel spectrogram loss)
+- [Amphion](https://github.com/open-mmlab/Amphion) (for multi-scale sub-band CQT discriminator)
diff --git a/GPT_SoVITS/BigVGAN/activations.py b/GPT_SoVITS/BigVGAN/activations.py
new file mode 100644
index 0000000000000000000000000000000000000000..abe3ad9e25c6ab3d4545c6a8c60e1f85a5a8e98e
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/activations.py
@@ -0,0 +1,122 @@
+# Implementation adapted from https://github.com/EdwardDixon/snake under the MIT license.
+# LICENSE is in incl_licenses directory.
+
+import torch
+from torch import nn, sin, pow
+from torch.nn import Parameter
+
+
+class Snake(nn.Module):
+ """
+ Implementation of a sine-based periodic activation function
+ Shape:
+ - Input: (B, C, T)
+ - Output: (B, C, T), same shape as the input
+ Parameters:
+ - alpha - trainable parameter
+ References:
+ - This activation function is from this paper by Liu Ziyin, Tilman Hartwig, Masahito Ueda:
+ https://arxiv.org/abs/2006.08195
+ Examples:
+ >>> a1 = snake(256)
+ >>> x = torch.randn(256)
+ >>> x = a1(x)
+ """
+
+ def __init__(self, in_features, alpha=1.0, alpha_trainable=True, alpha_logscale=False):
+ """
+ Initialization.
+ INPUT:
+ - in_features: shape of the input
+ - alpha: trainable parameter
+ alpha is initialized to 1 by default, higher values = higher-frequency.
+ alpha will be trained along with the rest of your model.
+ """
+ super(Snake, self).__init__()
+ self.in_features = in_features
+
+ # Initialize alpha
+ self.alpha_logscale = alpha_logscale
+ if self.alpha_logscale: # Log scale alphas initialized to zeros
+ self.alpha = Parameter(torch.zeros(in_features) * alpha)
+ else: # Linear scale alphas initialized to ones
+ self.alpha = Parameter(torch.ones(in_features) * alpha)
+
+ self.alpha.requires_grad = alpha_trainable
+
+ self.no_div_by_zero = 0.000000001
+
+ def forward(self, x):
+ """
+ Forward pass of the function.
+ Applies the function to the input elementwise.
+ Snake ∶= x + 1/a * sin^2 (xa)
+ """
+ alpha = self.alpha.unsqueeze(0).unsqueeze(-1) # Line up with x to [B, C, T]
+ if self.alpha_logscale:
+ alpha = torch.exp(alpha)
+ x = x + (1.0 / (alpha + self.no_div_by_zero)) * pow(sin(x * alpha), 2)
+
+ return x
+
+
+class SnakeBeta(nn.Module):
+ """
+ A modified Snake function which uses separate parameters for the magnitude of the periodic components
+ Shape:
+ - Input: (B, C, T)
+ - Output: (B, C, T), same shape as the input
+ Parameters:
+ - alpha - trainable parameter that controls frequency
+ - beta - trainable parameter that controls magnitude
+ References:
+ - This activation function is a modified version based on this paper by Liu Ziyin, Tilman Hartwig, Masahito Ueda:
+ https://arxiv.org/abs/2006.08195
+ Examples:
+ >>> a1 = snakebeta(256)
+ >>> x = torch.randn(256)
+ >>> x = a1(x)
+ """
+
+ def __init__(self, in_features, alpha=1.0, alpha_trainable=True, alpha_logscale=False):
+ """
+ Initialization.
+ INPUT:
+ - in_features: shape of the input
+ - alpha - trainable parameter that controls frequency
+ - beta - trainable parameter that controls magnitude
+ alpha is initialized to 1 by default, higher values = higher-frequency.
+ beta is initialized to 1 by default, higher values = higher-magnitude.
+ alpha will be trained along with the rest of your model.
+ """
+ super(SnakeBeta, self).__init__()
+ self.in_features = in_features
+
+ # Initialize alpha
+ self.alpha_logscale = alpha_logscale
+ if self.alpha_logscale: # Log scale alphas initialized to zeros
+ self.alpha = Parameter(torch.zeros(in_features) * alpha)
+ self.beta = Parameter(torch.zeros(in_features) * alpha)
+ else: # Linear scale alphas initialized to ones
+ self.alpha = Parameter(torch.ones(in_features) * alpha)
+ self.beta = Parameter(torch.ones(in_features) * alpha)
+
+ self.alpha.requires_grad = alpha_trainable
+ self.beta.requires_grad = alpha_trainable
+
+ self.no_div_by_zero = 0.000000001
+
+ def forward(self, x):
+ """
+ Forward pass of the function.
+ Applies the function to the input elementwise.
+ SnakeBeta ∶= x + 1/b * sin^2 (xa)
+ """
+ alpha = self.alpha.unsqueeze(0).unsqueeze(-1) # Line up with x to [B, C, T]
+ beta = self.beta.unsqueeze(0).unsqueeze(-1)
+ if self.alpha_logscale:
+ alpha = torch.exp(alpha)
+ beta = torch.exp(beta)
+ x = x + (1.0 / (beta + self.no_div_by_zero)) * pow(sin(x * alpha), 2)
+
+ return x
diff --git a/GPT_SoVITS/BigVGAN/bigvgan.py b/GPT_SoVITS/BigVGAN/bigvgan.py
new file mode 100644
index 0000000000000000000000000000000000000000..febdf165c354b1fa2932f27e4ef8b7b6da10e2a6
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/bigvgan.py
@@ -0,0 +1,461 @@
+# Copyright (c) 2024 NVIDIA CORPORATION.
+# Licensed under the MIT license.
+
+# Adapted from https://github.com/jik876/hifi-gan under the MIT license.
+# LICENSE is in incl_licenses directory.
+
+import os
+import json
+from pathlib import Path
+from typing import Optional, Union, Dict
+
+import torch
+import torch.nn as nn
+from torch.nn import Conv1d, ConvTranspose1d
+from torch.nn.utils import weight_norm, remove_weight_norm
+
+from . import activations
+from .utils0 import init_weights, get_padding
+from .alias_free_activation.torch.act import Activation1d as TorchActivation1d
+from .env import AttrDict
+
+from huggingface_hub import PyTorchModelHubMixin, hf_hub_download
+
+
+def load_hparams_from_json(path) -> AttrDict:
+ with open(path) as f:
+ data = f.read()
+ return AttrDict(json.loads(data))
+
+
+class AMPBlock1(torch.nn.Module):
+ """
+ AMPBlock applies Snake / SnakeBeta activation functions with trainable parameters that control periodicity, defined for each layer.
+ AMPBlock1 has additional self.convs2 that contains additional Conv1d layers with a fixed dilation=1 followed by each layer in self.convs1
+
+ Args:
+ h (AttrDict): Hyperparameters.
+ channels (int): Number of convolution channels.
+ kernel_size (int): Size of the convolution kernel. Default is 3.
+ dilation (tuple): Dilation rates for the convolutions. Each dilation layer has two convolutions. Default is (1, 3, 5).
+ activation (str): Activation function type. Should be either 'snake' or 'snakebeta'. Default is None.
+ """
+
+ def __init__(
+ self,
+ h: AttrDict,
+ channels: int,
+ kernel_size: int = 3,
+ dilation: tuple = (1, 3, 5),
+ activation: str = None,
+ ):
+ super().__init__()
+
+ self.h = h
+
+ self.convs1 = nn.ModuleList(
+ [
+ weight_norm(
+ Conv1d(
+ channels,
+ channels,
+ kernel_size,
+ stride=1,
+ dilation=d,
+ padding=get_padding(kernel_size, d),
+ )
+ )
+ for d in dilation
+ ]
+ )
+ self.convs1.apply(init_weights)
+
+ self.convs2 = nn.ModuleList(
+ [
+ weight_norm(
+ Conv1d(
+ channels,
+ channels,
+ kernel_size,
+ stride=1,
+ dilation=1,
+ padding=get_padding(kernel_size, 1),
+ )
+ )
+ for _ in range(len(dilation))
+ ]
+ )
+ self.convs2.apply(init_weights)
+
+ self.num_layers = len(self.convs1) + len(self.convs2) # Total number of conv layers
+
+ # Select which Activation1d, lazy-load cuda version to ensure backward compatibility
+ if self.h.get("use_cuda_kernel", False):
+ from .alias_free_activation.cuda.activation1d import (
+ Activation1d as CudaActivation1d,
+ )
+
+ Activation1d = CudaActivation1d
+ else:
+ Activation1d = TorchActivation1d
+
+ # Activation functions
+ if activation == "snake":
+ self.activations = nn.ModuleList(
+ [
+ Activation1d(activation=activations.Snake(channels, alpha_logscale=h.snake_logscale))
+ for _ in range(self.num_layers)
+ ]
+ )
+ elif activation == "snakebeta":
+ self.activations = nn.ModuleList(
+ [
+ Activation1d(activation=activations.SnakeBeta(channels, alpha_logscale=h.snake_logscale))
+ for _ in range(self.num_layers)
+ ]
+ )
+ else:
+ raise NotImplementedError(
+ "activation incorrectly specified. check the config file and look for 'activation'."
+ )
+
+ def forward(self, x):
+ acts1, acts2 = self.activations[::2], self.activations[1::2]
+ for c1, c2, a1, a2 in zip(self.convs1, self.convs2, acts1, acts2):
+ xt = a1(x)
+ xt = c1(xt)
+ xt = a2(xt)
+ xt = c2(xt)
+ x = xt + x
+
+ return x
+
+ def remove_weight_norm(self):
+ for l in self.convs1:
+ remove_weight_norm(l)
+ for l in self.convs2:
+ remove_weight_norm(l)
+
+
+class AMPBlock2(torch.nn.Module):
+ """
+ AMPBlock applies Snake / SnakeBeta activation functions with trainable parameters that control periodicity, defined for each layer.
+ Unlike AMPBlock1, AMPBlock2 does not contain extra Conv1d layers with fixed dilation=1
+
+ Args:
+ h (AttrDict): Hyperparameters.
+ channels (int): Number of convolution channels.
+ kernel_size (int): Size of the convolution kernel. Default is 3.
+ dilation (tuple): Dilation rates for the convolutions. Each dilation layer has two convolutions. Default is (1, 3, 5).
+ activation (str): Activation function type. Should be either 'snake' or 'snakebeta'. Default is None.
+ """
+
+ def __init__(
+ self,
+ h: AttrDict,
+ channels: int,
+ kernel_size: int = 3,
+ dilation: tuple = (1, 3, 5),
+ activation: str = None,
+ ):
+ super().__init__()
+
+ self.h = h
+
+ self.convs = nn.ModuleList(
+ [
+ weight_norm(
+ Conv1d(
+ channels,
+ channels,
+ kernel_size,
+ stride=1,
+ dilation=d,
+ padding=get_padding(kernel_size, d),
+ )
+ )
+ for d in dilation
+ ]
+ )
+ self.convs.apply(init_weights)
+
+ self.num_layers = len(self.convs) # Total number of conv layers
+
+ # Select which Activation1d, lazy-load cuda version to ensure backward compatibility
+ if self.h.get("use_cuda_kernel", False):
+ from .alias_free_activation.cuda.activation1d import (
+ Activation1d as CudaActivation1d,
+ )
+
+ Activation1d = CudaActivation1d
+ else:
+ Activation1d = TorchActivation1d
+
+ # Activation functions
+ if activation == "snake":
+ self.activations = nn.ModuleList(
+ [
+ Activation1d(activation=activations.Snake(channels, alpha_logscale=h.snake_logscale))
+ for _ in range(self.num_layers)
+ ]
+ )
+ elif activation == "snakebeta":
+ self.activations = nn.ModuleList(
+ [
+ Activation1d(activation=activations.SnakeBeta(channels, alpha_logscale=h.snake_logscale))
+ for _ in range(self.num_layers)
+ ]
+ )
+ else:
+ raise NotImplementedError(
+ "activation incorrectly specified. check the config file and look for 'activation'."
+ )
+
+ def forward(self, x):
+ for c, a in zip(self.convs, self.activations):
+ xt = a(x)
+ xt = c(xt)
+ x = xt + x
+ return x
+
+ def remove_weight_norm(self):
+ for l in self.convs:
+ remove_weight_norm(l)
+
+
+class BigVGAN(
+ torch.nn.Module,
+ PyTorchModelHubMixin,
+ # library_name="bigvgan",
+ # repo_url="https://github.com/NVIDIA/BigVGAN",
+ # docs_url="https://github.com/NVIDIA/BigVGAN/blob/main/README.md",
+ # pipeline_tag="audio-to-audio",
+ # license="mit",
+ # tags=["neural-vocoder", "audio-generation", "arxiv:2206.04658"],
+):
+ """
+ BigVGAN is a neural vocoder model that applies anti-aliased periodic activation for residual blocks (resblocks).
+ New in BigVGAN-v2: it can optionally use optimized CUDA kernels for AMP (anti-aliased multi-periodicity) blocks.
+
+ Args:
+ h (AttrDict): Hyperparameters.
+ use_cuda_kernel (bool): If set to True, loads optimized CUDA kernels for AMP. This should be used for inference only, as training is not supported with CUDA kernels.
+
+ Note:
+ - The `use_cuda_kernel` parameter should be used for inference only, as training with CUDA kernels is not supported.
+ - Ensure that the activation function is correctly specified in the hyperparameters (h.activation).
+ """
+
+ def __init__(self, h: AttrDict, use_cuda_kernel: bool = False):
+ super().__init__()
+ self.h = h
+ self.h["use_cuda_kernel"] = use_cuda_kernel
+
+ # Select which Activation1d, lazy-load cuda version to ensure backward compatibility
+ if self.h.get("use_cuda_kernel", False):
+ from .alias_free_activation.cuda.activation1d import (
+ Activation1d as CudaActivation1d,
+ )
+
+ Activation1d = CudaActivation1d
+ else:
+ Activation1d = TorchActivation1d
+
+ self.num_kernels = len(h.resblock_kernel_sizes)
+ self.num_upsamples = len(h.upsample_rates)
+
+ # Pre-conv
+ self.conv_pre = weight_norm(Conv1d(h.num_mels, h.upsample_initial_channel, 7, 1, padding=3))
+
+ # Define which AMPBlock to use. BigVGAN uses AMPBlock1 as default
+ if h.resblock == "1":
+ resblock_class = AMPBlock1
+ elif h.resblock == "2":
+ resblock_class = AMPBlock2
+ else:
+ raise ValueError(f"Incorrect resblock class specified in hyperparameters. Got {h.resblock}")
+
+ # Transposed conv-based upsamplers. does not apply anti-aliasing
+ self.ups = nn.ModuleList()
+ for i, (u, k) in enumerate(zip(h.upsample_rates, h.upsample_kernel_sizes)):
+ self.ups.append(
+ nn.ModuleList(
+ [
+ weight_norm(
+ ConvTranspose1d(
+ h.upsample_initial_channel // (2**i),
+ h.upsample_initial_channel // (2 ** (i + 1)),
+ k,
+ u,
+ padding=(k - u) // 2,
+ )
+ )
+ ]
+ )
+ )
+
+ # Residual blocks using anti-aliased multi-periodicity composition modules (AMP)
+ self.resblocks = nn.ModuleList()
+ for i in range(len(self.ups)):
+ ch = h.upsample_initial_channel // (2 ** (i + 1))
+ for j, (k, d) in enumerate(zip(h.resblock_kernel_sizes, h.resblock_dilation_sizes)):
+ self.resblocks.append(resblock_class(h, ch, k, d, activation=h.activation))
+
+ # Post-conv
+ activation_post = (
+ activations.Snake(ch, alpha_logscale=h.snake_logscale)
+ if h.activation == "snake"
+ else (activations.SnakeBeta(ch, alpha_logscale=h.snake_logscale) if h.activation == "snakebeta" else None)
+ )
+ if activation_post is None:
+ raise NotImplementedError(
+ "activation incorrectly specified. check the config file and look for 'activation'."
+ )
+
+ self.activation_post = Activation1d(activation=activation_post)
+
+ # Whether to use bias for the final conv_post. Default to True for backward compatibility
+ self.use_bias_at_final = h.get("use_bias_at_final", True)
+ self.conv_post = weight_norm(Conv1d(ch, 1, 7, 1, padding=3, bias=self.use_bias_at_final))
+
+ # Weight initialization
+ for i in range(len(self.ups)):
+ self.ups[i].apply(init_weights)
+ self.conv_post.apply(init_weights)
+
+ # Final tanh activation. Defaults to True for backward compatibility
+ self.use_tanh_at_final = h.get("use_tanh_at_final", True)
+
+ def forward(self, x):
+ # Pre-conv
+ x = self.conv_pre(x)
+
+ for i in range(self.num_upsamples):
+ # Upsampling
+ for i_up in range(len(self.ups[i])):
+ x = self.ups[i][i_up](x)
+ # AMP blocks
+ xs = None
+ for j in range(self.num_kernels):
+ if xs is None:
+ xs = self.resblocks[i * self.num_kernels + j](x)
+ else:
+ xs += self.resblocks[i * self.num_kernels + j](x)
+ x = xs / self.num_kernels
+
+ # Post-conv
+ x = self.activation_post(x)
+ x = self.conv_post(x)
+ # Final tanh activation
+ if self.use_tanh_at_final:
+ x = torch.tanh(x)
+ else:
+ x = torch.clamp(x, min=-1.0, max=1.0) # Bound the output to [-1, 1]
+
+ return x
+
+ def remove_weight_norm(self):
+ try:
+ # print("Removing weight norm...")
+ for l in self.ups:
+ for l_i in l:
+ remove_weight_norm(l_i)
+ for l in self.resblocks:
+ l.remove_weight_norm()
+ remove_weight_norm(self.conv_pre)
+ remove_weight_norm(self.conv_post)
+ except ValueError:
+ print("[INFO] Model already removed weight norm. Skipping!")
+ pass
+
+ # Additional methods for huggingface_hub support
+ def _save_pretrained(self, save_directory: Path) -> None:
+ """Save weights and config.json from a Pytorch model to a local directory."""
+
+ model_path = save_directory / "bigvgan_generator.pt"
+ torch.save({"generator": self.state_dict()}, model_path)
+
+ config_path = save_directory / "config.json"
+ with open(config_path, "w") as config_file:
+ json.dump(self.h, config_file, indent=4)
+
+ @classmethod
+ def _from_pretrained(
+ cls,
+ *,
+ model_id: str,
+ revision: str,
+ cache_dir: str,
+ force_download: bool,
+ proxies: Optional[Dict],
+ resume_download: bool,
+ local_files_only: bool,
+ token: Union[str, bool, None],
+ map_location: str = "cpu", # Additional argument
+ strict: bool = False, # Additional argument
+ use_cuda_kernel: bool = False,
+ **model_kwargs,
+ ):
+ """Load Pytorch pretrained weights and return the loaded model."""
+
+ # Download and load hyperparameters (h) used by BigVGAN
+ if os.path.isdir(model_id):
+ # print("Loading config.json from local directory")
+ config_file = os.path.join(model_id, "config.json")
+ else:
+ config_file = hf_hub_download(
+ repo_id=model_id,
+ filename="config.json",
+ revision=revision,
+ cache_dir=cache_dir,
+ force_download=force_download,
+ proxies=proxies,
+ resume_download=resume_download,
+ token=token,
+ local_files_only=local_files_only,
+ )
+ h = load_hparams_from_json(config_file)
+
+ # instantiate BigVGAN using h
+ if use_cuda_kernel:
+ print(
+ "[WARNING] You have specified use_cuda_kernel=True during BigVGAN.from_pretrained(). Only inference is supported (training is not implemented)!"
+ )
+ print(
+ "[WARNING] You need nvcc and ninja installed in your system that matches your PyTorch build is using to build the kernel. If not, the model will fail to initialize or generate incorrect waveform!"
+ )
+ print(
+ "[WARNING] For detail, see the official GitHub repository: https://github.com/NVIDIA/BigVGAN?tab=readme-ov-file#using-custom-cuda-kernel-for-synthesis"
+ )
+ model = cls(h, use_cuda_kernel=use_cuda_kernel)
+
+ # Download and load pretrained generator weight
+ if os.path.isdir(model_id):
+ # print("Loading weights from local directory")
+ model_file = os.path.join(model_id, "bigvgan_generator.pt")
+ else:
+ # print(f"Loading weights from {model_id}")
+ model_file = hf_hub_download(
+ repo_id=model_id,
+ filename="bigvgan_generator.pt",
+ revision=revision,
+ cache_dir=cache_dir,
+ force_download=force_download,
+ proxies=proxies,
+ resume_download=resume_download,
+ token=token,
+ local_files_only=local_files_only,
+ )
+
+ checkpoint_dict = torch.load(model_file, map_location=map_location)
+
+ try:
+ model.load_state_dict(checkpoint_dict["generator"])
+ except RuntimeError:
+ print(
+ "[INFO] the pretrained checkpoint does not contain weight norm. Loading the checkpoint after removing weight norm!"
+ )
+ model.remove_weight_norm()
+ model.load_state_dict(checkpoint_dict["generator"])
+
+ return model
diff --git a/GPT_SoVITS/BigVGAN/configs/bigvgan_22khz_80band.json b/GPT_SoVITS/BigVGAN/configs/bigvgan_22khz_80band.json
new file mode 100644
index 0000000000000000000000000000000000000000..64bca7846edb4e86d7ee22d9ca7a1554cf7f1042
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/configs/bigvgan_22khz_80band.json
@@ -0,0 +1,45 @@
+{
+ "resblock": "1",
+ "num_gpus": 0,
+ "batch_size": 32,
+ "learning_rate": 0.0001,
+ "adam_b1": 0.8,
+ "adam_b2": 0.99,
+ "lr_decay": 0.9999996,
+ "seed": 1234,
+
+ "upsample_rates": [4,4,2,2,2,2],
+ "upsample_kernel_sizes": [8,8,4,4,4,4],
+ "upsample_initial_channel": 1536,
+ "resblock_kernel_sizes": [3,7,11],
+ "resblock_dilation_sizes": [[1,3,5], [1,3,5], [1,3,5]],
+
+ "activation": "snakebeta",
+ "snake_logscale": true,
+
+ "resolutions": [[1024, 120, 600], [2048, 240, 1200], [512, 50, 240]],
+ "mpd_reshapes": [2, 3, 5, 7, 11],
+ "use_spectral_norm": false,
+ "discriminator_channel_mult": 1,
+
+ "segment_size": 8192,
+ "num_mels": 80,
+ "num_freq": 1025,
+ "n_fft": 1024,
+ "hop_size": 256,
+ "win_size": 1024,
+
+ "sampling_rate": 22050,
+
+ "fmin": 0,
+ "fmax": 8000,
+ "fmax_for_loss": null,
+
+ "num_workers": 4,
+
+ "dist_config": {
+ "dist_backend": "nccl",
+ "dist_url": "tcp://localhost:54321",
+ "world_size": 1
+ }
+}
diff --git a/GPT_SoVITS/BigVGAN/configs/bigvgan_24khz_100band.json b/GPT_SoVITS/BigVGAN/configs/bigvgan_24khz_100band.json
new file mode 100644
index 0000000000000000000000000000000000000000..e7f7ff08f6697a4640d8e28c0b3fe7e62d0c3fc7
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/configs/bigvgan_24khz_100band.json
@@ -0,0 +1,45 @@
+{
+ "resblock": "1",
+ "num_gpus": 0,
+ "batch_size": 32,
+ "learning_rate": 0.0001,
+ "adam_b1": 0.8,
+ "adam_b2": 0.99,
+ "lr_decay": 0.9999996,
+ "seed": 1234,
+
+ "upsample_rates": [4,4,2,2,2,2],
+ "upsample_kernel_sizes": [8,8,4,4,4,4],
+ "upsample_initial_channel": 1536,
+ "resblock_kernel_sizes": [3,7,11],
+ "resblock_dilation_sizes": [[1,3,5], [1,3,5], [1,3,5]],
+
+ "activation": "snakebeta",
+ "snake_logscale": true,
+
+ "resolutions": [[1024, 120, 600], [2048, 240, 1200], [512, 50, 240]],
+ "mpd_reshapes": [2, 3, 5, 7, 11],
+ "use_spectral_norm": false,
+ "discriminator_channel_mult": 1,
+
+ "segment_size": 8192,
+ "num_mels": 100,
+ "num_freq": 1025,
+ "n_fft": 1024,
+ "hop_size": 256,
+ "win_size": 1024,
+
+ "sampling_rate": 24000,
+
+ "fmin": 0,
+ "fmax": 12000,
+ "fmax_for_loss": null,
+
+ "num_workers": 4,
+
+ "dist_config": {
+ "dist_backend": "nccl",
+ "dist_url": "tcp://localhost:54321",
+ "world_size": 1
+ }
+}
diff --git a/GPT_SoVITS/BigVGAN/configs/bigvgan_base_22khz_80band.json b/GPT_SoVITS/BigVGAN/configs/bigvgan_base_22khz_80band.json
new file mode 100644
index 0000000000000000000000000000000000000000..fd244848308917f4df7ce49bf6b76530fd04cbc2
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/configs/bigvgan_base_22khz_80band.json
@@ -0,0 +1,45 @@
+{
+ "resblock": "1",
+ "num_gpus": 0,
+ "batch_size": 32,
+ "learning_rate": 0.0001,
+ "adam_b1": 0.8,
+ "adam_b2": 0.99,
+ "lr_decay": 0.9999996,
+ "seed": 1234,
+
+ "upsample_rates": [8,8,2,2],
+ "upsample_kernel_sizes": [16,16,4,4],
+ "upsample_initial_channel": 512,
+ "resblock_kernel_sizes": [3,7,11],
+ "resblock_dilation_sizes": [[1,3,5], [1,3,5], [1,3,5]],
+
+ "activation": "snakebeta",
+ "snake_logscale": true,
+
+ "resolutions": [[1024, 120, 600], [2048, 240, 1200], [512, 50, 240]],
+ "mpd_reshapes": [2, 3, 5, 7, 11],
+ "use_spectral_norm": false,
+ "discriminator_channel_mult": 1,
+
+ "segment_size": 8192,
+ "num_mels": 80,
+ "num_freq": 1025,
+ "n_fft": 1024,
+ "hop_size": 256,
+ "win_size": 1024,
+
+ "sampling_rate": 22050,
+
+ "fmin": 0,
+ "fmax": 8000,
+ "fmax_for_loss": null,
+
+ "num_workers": 4,
+
+ "dist_config": {
+ "dist_backend": "nccl",
+ "dist_url": "tcp://localhost:54321",
+ "world_size": 1
+ }
+}
diff --git a/GPT_SoVITS/BigVGAN/configs/bigvgan_base_24khz_100band.json b/GPT_SoVITS/BigVGAN/configs/bigvgan_base_24khz_100band.json
new file mode 100644
index 0000000000000000000000000000000000000000..0911508cac4a9346ada8c196bfcc228998da6f42
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/configs/bigvgan_base_24khz_100band.json
@@ -0,0 +1,45 @@
+{
+ "resblock": "1",
+ "num_gpus": 0,
+ "batch_size": 32,
+ "learning_rate": 0.0001,
+ "adam_b1": 0.8,
+ "adam_b2": 0.99,
+ "lr_decay": 0.9999996,
+ "seed": 1234,
+
+ "upsample_rates": [8,8,2,2],
+ "upsample_kernel_sizes": [16,16,4,4],
+ "upsample_initial_channel": 512,
+ "resblock_kernel_sizes": [3,7,11],
+ "resblock_dilation_sizes": [[1,3,5], [1,3,5], [1,3,5]],
+
+ "activation": "snakebeta",
+ "snake_logscale": true,
+
+ "resolutions": [[1024, 120, 600], [2048, 240, 1200], [512, 50, 240]],
+ "mpd_reshapes": [2, 3, 5, 7, 11],
+ "use_spectral_norm": false,
+ "discriminator_channel_mult": 1,
+
+ "segment_size": 8192,
+ "num_mels": 100,
+ "num_freq": 1025,
+ "n_fft": 1024,
+ "hop_size": 256,
+ "win_size": 1024,
+
+ "sampling_rate": 24000,
+
+ "fmin": 0,
+ "fmax": 12000,
+ "fmax_for_loss": null,
+
+ "num_workers": 4,
+
+ "dist_config": {
+ "dist_backend": "nccl",
+ "dist_url": "tcp://localhost:54321",
+ "world_size": 1
+ }
+}
diff --git a/GPT_SoVITS/BigVGAN/configs/bigvgan_v2_22khz_80band_256x.json b/GPT_SoVITS/BigVGAN/configs/bigvgan_v2_22khz_80band_256x.json
new file mode 100644
index 0000000000000000000000000000000000000000..e96bd5fdd5b99767adba7f13bfcd1f777d5c599a
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/configs/bigvgan_v2_22khz_80band_256x.json
@@ -0,0 +1,61 @@
+{
+ "resblock": "1",
+ "num_gpus": 0,
+ "batch_size": 4,
+ "learning_rate": 0.0001,
+ "adam_b1": 0.8,
+ "adam_b2": 0.99,
+ "lr_decay": 0.9999996,
+ "seed": 1234,
+
+ "upsample_rates": [4,4,2,2,2,2],
+ "upsample_kernel_sizes": [8,8,4,4,4,4],
+ "upsample_initial_channel": 1536,
+ "resblock_kernel_sizes": [3,7,11],
+ "resblock_dilation_sizes": [[1,3,5], [1,3,5], [1,3,5]],
+
+ "use_tanh_at_final": false,
+ "use_bias_at_final": false,
+
+ "activation": "snakebeta",
+ "snake_logscale": true,
+
+ "use_cqtd_instead_of_mrd": true,
+ "cqtd_filters": 128,
+ "cqtd_max_filters": 1024,
+ "cqtd_filters_scale": 1,
+ "cqtd_dilations": [1, 2, 4],
+ "cqtd_hop_lengths": [512, 256, 256],
+ "cqtd_n_octaves": [9, 9, 9],
+ "cqtd_bins_per_octaves": [24, 36, 48],
+
+ "mpd_reshapes": [2, 3, 5, 7, 11],
+ "use_spectral_norm": false,
+ "discriminator_channel_mult": 1,
+
+ "use_multiscale_melloss": true,
+ "lambda_melloss": 15,
+
+ "clip_grad_norm": 500,
+
+ "segment_size": 65536,
+ "num_mels": 80,
+ "num_freq": 1025,
+ "n_fft": 1024,
+ "hop_size": 256,
+ "win_size": 1024,
+
+ "sampling_rate": 22050,
+
+ "fmin": 0,
+ "fmax": null,
+ "fmax_for_loss": null,
+
+ "num_workers": 4,
+
+ "dist_config": {
+ "dist_backend": "nccl",
+ "dist_url": "tcp://localhost:54321",
+ "world_size": 1
+ }
+}
diff --git a/GPT_SoVITS/BigVGAN/configs/bigvgan_v2_22khz_80band_fmax8k_256x.json b/GPT_SoVITS/BigVGAN/configs/bigvgan_v2_22khz_80band_fmax8k_256x.json
new file mode 100644
index 0000000000000000000000000000000000000000..a3c9699fbe11948f4fd7e3434d2e623a00c802dd
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/configs/bigvgan_v2_22khz_80band_fmax8k_256x.json
@@ -0,0 +1,61 @@
+{
+ "resblock": "1",
+ "num_gpus": 0,
+ "batch_size": 4,
+ "learning_rate": 0.0001,
+ "adam_b1": 0.8,
+ "adam_b2": 0.99,
+ "lr_decay": 0.9999996,
+ "seed": 1234,
+
+ "upsample_rates": [4,4,2,2,2,2],
+ "upsample_kernel_sizes": [8,8,4,4,4,4],
+ "upsample_initial_channel": 1536,
+ "resblock_kernel_sizes": [3,7,11],
+ "resblock_dilation_sizes": [[1,3,5], [1,3,5], [1,3,5]],
+
+ "use_tanh_at_final": false,
+ "use_bias_at_final": false,
+
+ "activation": "snakebeta",
+ "snake_logscale": true,
+
+ "use_cqtd_instead_of_mrd": true,
+ "cqtd_filters": 128,
+ "cqtd_max_filters": 1024,
+ "cqtd_filters_scale": 1,
+ "cqtd_dilations": [1, 2, 4],
+ "cqtd_hop_lengths": [512, 256, 256],
+ "cqtd_n_octaves": [9, 9, 9],
+ "cqtd_bins_per_octaves": [24, 36, 48],
+
+ "mpd_reshapes": [2, 3, 5, 7, 11],
+ "use_spectral_norm": false,
+ "discriminator_channel_mult": 1,
+
+ "use_multiscale_melloss": true,
+ "lambda_melloss": 15,
+
+ "clip_grad_norm": 500,
+
+ "segment_size": 65536,
+ "num_mels": 80,
+ "num_freq": 1025,
+ "n_fft": 1024,
+ "hop_size": 256,
+ "win_size": 1024,
+
+ "sampling_rate": 22050,
+
+ "fmin": 0,
+ "fmax": 8000,
+ "fmax_for_loss": null,
+
+ "num_workers": 4,
+
+ "dist_config": {
+ "dist_backend": "nccl",
+ "dist_url": "tcp://localhost:54321",
+ "world_size": 1
+ }
+}
diff --git a/GPT_SoVITS/BigVGAN/configs/bigvgan_v2_24khz_100band_256x.json b/GPT_SoVITS/BigVGAN/configs/bigvgan_v2_24khz_100band_256x.json
new file mode 100644
index 0000000000000000000000000000000000000000..8057ee267c8ed80615362a41892b923a3ccd27e5
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/configs/bigvgan_v2_24khz_100band_256x.json
@@ -0,0 +1,61 @@
+{
+ "resblock": "1",
+ "num_gpus": 0,
+ "batch_size": 4,
+ "learning_rate": 0.0001,
+ "adam_b1": 0.8,
+ "adam_b2": 0.99,
+ "lr_decay": 0.9999996,
+ "seed": 1234,
+
+ "upsample_rates": [4,4,2,2,2,2],
+ "upsample_kernel_sizes": [8,8,4,4,4,4],
+ "upsample_initial_channel": 1536,
+ "resblock_kernel_sizes": [3,7,11],
+ "resblock_dilation_sizes": [[1,3,5], [1,3,5], [1,3,5]],
+
+ "use_tanh_at_final": false,
+ "use_bias_at_final": false,
+
+ "activation": "snakebeta",
+ "snake_logscale": true,
+
+ "use_cqtd_instead_of_mrd": true,
+ "cqtd_filters": 128,
+ "cqtd_max_filters": 1024,
+ "cqtd_filters_scale": 1,
+ "cqtd_dilations": [1, 2, 4],
+ "cqtd_hop_lengths": [512, 256, 256],
+ "cqtd_n_octaves": [9, 9, 9],
+ "cqtd_bins_per_octaves": [24, 36, 48],
+
+ "mpd_reshapes": [2, 3, 5, 7, 11],
+ "use_spectral_norm": false,
+ "discriminator_channel_mult": 1,
+
+ "use_multiscale_melloss": true,
+ "lambda_melloss": 15,
+
+ "clip_grad_norm": 500,
+
+ "segment_size": 65536,
+ "num_mels": 100,
+ "num_freq": 1025,
+ "n_fft": 1024,
+ "hop_size": 256,
+ "win_size": 1024,
+
+ "sampling_rate": 24000,
+
+ "fmin": 0,
+ "fmax": null,
+ "fmax_for_loss": null,
+
+ "num_workers": 4,
+
+ "dist_config": {
+ "dist_backend": "nccl",
+ "dist_url": "tcp://localhost:54321",
+ "world_size": 1
+ }
+}
diff --git a/GPT_SoVITS/BigVGAN/configs/bigvgan_v2_44khz_128band_256x.json b/GPT_SoVITS/BigVGAN/configs/bigvgan_v2_44khz_128band_256x.json
new file mode 100644
index 0000000000000000000000000000000000000000..b6999d3028e5d741ec99b16b34f153e763d0cfec
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/configs/bigvgan_v2_44khz_128band_256x.json
@@ -0,0 +1,61 @@
+{
+ "resblock": "1",
+ "num_gpus": 0,
+ "batch_size": 4,
+ "learning_rate": 0.0001,
+ "adam_b1": 0.8,
+ "adam_b2": 0.99,
+ "lr_decay": 0.9999996,
+ "seed": 1234,
+
+ "upsample_rates": [4,4,2,2,2,2],
+ "upsample_kernel_sizes": [8,8,4,4,4,4],
+ "upsample_initial_channel": 1536,
+ "resblock_kernel_sizes": [3,7,11],
+ "resblock_dilation_sizes": [[1,3,5], [1,3,5], [1,3,5]],
+
+ "use_tanh_at_final": false,
+ "use_bias_at_final": false,
+
+ "activation": "snakebeta",
+ "snake_logscale": true,
+
+ "use_cqtd_instead_of_mrd": true,
+ "cqtd_filters": 128,
+ "cqtd_max_filters": 1024,
+ "cqtd_filters_scale": 1,
+ "cqtd_dilations": [1, 2, 4],
+ "cqtd_hop_lengths": [512, 256, 256],
+ "cqtd_n_octaves": [9, 9, 9],
+ "cqtd_bins_per_octaves": [24, 36, 48],
+
+ "mpd_reshapes": [2, 3, 5, 7, 11],
+ "use_spectral_norm": false,
+ "discriminator_channel_mult": 1,
+
+ "use_multiscale_melloss": true,
+ "lambda_melloss": 15,
+
+ "clip_grad_norm": 500,
+
+ "segment_size": 65536,
+ "num_mels": 128,
+ "num_freq": 1025,
+ "n_fft": 1024,
+ "hop_size": 256,
+ "win_size": 1024,
+
+ "sampling_rate": 44100,
+
+ "fmin": 0,
+ "fmax": null,
+ "fmax_for_loss": null,
+
+ "num_workers": 4,
+
+ "dist_config": {
+ "dist_backend": "nccl",
+ "dist_url": "tcp://localhost:54321",
+ "world_size": 1
+ }
+}
diff --git a/GPT_SoVITS/BigVGAN/configs/bigvgan_v2_44khz_128band_512x.json b/GPT_SoVITS/BigVGAN/configs/bigvgan_v2_44khz_128band_512x.json
new file mode 100644
index 0000000000000000000000000000000000000000..2d7176c910ae0969f208f6d28e3f14abca2dbc7f
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/configs/bigvgan_v2_44khz_128band_512x.json
@@ -0,0 +1,61 @@
+{
+ "resblock": "1",
+ "num_gpus": 0,
+ "batch_size": 4,
+ "learning_rate": 0.0001,
+ "adam_b1": 0.8,
+ "adam_b2": 0.99,
+ "lr_decay": 0.9999996,
+ "seed": 1234,
+
+ "upsample_rates": [8,4,2,2,2,2],
+ "upsample_kernel_sizes": [16,8,4,4,4,4],
+ "upsample_initial_channel": 1536,
+ "resblock_kernel_sizes": [3,7,11],
+ "resblock_dilation_sizes": [[1,3,5], [1,3,5], [1,3,5]],
+
+ "use_tanh_at_final": false,
+ "use_bias_at_final": false,
+
+ "activation": "snakebeta",
+ "snake_logscale": true,
+
+ "use_cqtd_instead_of_mrd": true,
+ "cqtd_filters": 128,
+ "cqtd_max_filters": 1024,
+ "cqtd_filters_scale": 1,
+ "cqtd_dilations": [1, 2, 4],
+ "cqtd_hop_lengths": [512, 256, 256],
+ "cqtd_n_octaves": [9, 9, 9],
+ "cqtd_bins_per_octaves": [24, 36, 48],
+
+ "mpd_reshapes": [2, 3, 5, 7, 11],
+ "use_spectral_norm": false,
+ "discriminator_channel_mult": 1,
+
+ "use_multiscale_melloss": true,
+ "lambda_melloss": 15,
+
+ "clip_grad_norm": 500,
+
+ "segment_size": 65536,
+ "num_mels": 128,
+ "num_freq": 2049,
+ "n_fft": 2048,
+ "hop_size": 512,
+ "win_size": 2048,
+
+ "sampling_rate": 44100,
+
+ "fmin": 0,
+ "fmax": null,
+ "fmax_for_loss": null,
+
+ "num_workers": 4,
+
+ "dist_config": {
+ "dist_backend": "nccl",
+ "dist_url": "tcp://localhost:54321",
+ "world_size": 1
+ }
+}
diff --git a/GPT_SoVITS/BigVGAN/discriminators.py b/GPT_SoVITS/BigVGAN/discriminators.py
new file mode 100644
index 0000000000000000000000000000000000000000..2d44c7983955a1be15a4520f6730de272f799128
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/discriminators.py
@@ -0,0 +1,625 @@
+# Copyright (c) 2024 NVIDIA CORPORATION.
+# Licensed under the MIT license.
+
+# Adapted from https://github.com/jik876/hifi-gan under the MIT license.
+# LICENSE is in incl_licenses directory.
+
+
+import torch
+import torch.nn.functional as F
+import torch.nn as nn
+from torch.nn import Conv2d
+from torch.nn.utils import weight_norm, spectral_norm
+from torchaudio.transforms import Spectrogram, Resample
+
+from env import AttrDict
+from utils import get_padding
+import typing
+from typing import List, Tuple
+
+
+class DiscriminatorP(torch.nn.Module):
+ def __init__(
+ self,
+ h: AttrDict,
+ period: List[int],
+ kernel_size: int = 5,
+ stride: int = 3,
+ use_spectral_norm: bool = False,
+ ):
+ super().__init__()
+ self.period = period
+ self.d_mult = h.discriminator_channel_mult
+ norm_f = weight_norm if not use_spectral_norm else spectral_norm
+
+ self.convs = nn.ModuleList(
+ [
+ norm_f(
+ Conv2d(
+ 1,
+ int(32 * self.d_mult),
+ (kernel_size, 1),
+ (stride, 1),
+ padding=(get_padding(5, 1), 0),
+ )
+ ),
+ norm_f(
+ Conv2d(
+ int(32 * self.d_mult),
+ int(128 * self.d_mult),
+ (kernel_size, 1),
+ (stride, 1),
+ padding=(get_padding(5, 1), 0),
+ )
+ ),
+ norm_f(
+ Conv2d(
+ int(128 * self.d_mult),
+ int(512 * self.d_mult),
+ (kernel_size, 1),
+ (stride, 1),
+ padding=(get_padding(5, 1), 0),
+ )
+ ),
+ norm_f(
+ Conv2d(
+ int(512 * self.d_mult),
+ int(1024 * self.d_mult),
+ (kernel_size, 1),
+ (stride, 1),
+ padding=(get_padding(5, 1), 0),
+ )
+ ),
+ norm_f(
+ Conv2d(
+ int(1024 * self.d_mult),
+ int(1024 * self.d_mult),
+ (kernel_size, 1),
+ 1,
+ padding=(2, 0),
+ )
+ ),
+ ]
+ )
+ self.conv_post = norm_f(Conv2d(int(1024 * self.d_mult), 1, (3, 1), 1, padding=(1, 0)))
+
+ def forward(self, x: torch.Tensor) -> Tuple[torch.Tensor, List[torch.Tensor]]:
+ fmap = []
+
+ # 1d to 2d
+ b, c, t = x.shape
+ if t % self.period != 0: # pad first
+ n_pad = self.period - (t % self.period)
+ x = F.pad(x, (0, n_pad), "reflect")
+ t = t + n_pad
+ x = x.view(b, c, t // self.period, self.period)
+
+ for l in self.convs:
+ x = l(x)
+ x = F.leaky_relu(x, 0.1)
+ fmap.append(x)
+ x = self.conv_post(x)
+ fmap.append(x)
+ x = torch.flatten(x, 1, -1)
+
+ return x, fmap
+
+
+class MultiPeriodDiscriminator(torch.nn.Module):
+ def __init__(self, h: AttrDict):
+ super().__init__()
+ self.mpd_reshapes = h.mpd_reshapes
+ print(f"mpd_reshapes: {self.mpd_reshapes}")
+ self.discriminators = nn.ModuleList(
+ [DiscriminatorP(h, rs, use_spectral_norm=h.use_spectral_norm) for rs in self.mpd_reshapes]
+ )
+
+ def forward(
+ self, y: torch.Tensor, y_hat: torch.Tensor
+ ) -> Tuple[
+ List[torch.Tensor],
+ List[torch.Tensor],
+ List[List[torch.Tensor]],
+ List[List[torch.Tensor]],
+ ]:
+ y_d_rs = []
+ y_d_gs = []
+ fmap_rs = []
+ fmap_gs = []
+ for i, d in enumerate(self.discriminators):
+ y_d_r, fmap_r = d(y)
+ y_d_g, fmap_g = d(y_hat)
+ y_d_rs.append(y_d_r)
+ fmap_rs.append(fmap_r)
+ y_d_gs.append(y_d_g)
+ fmap_gs.append(fmap_g)
+
+ return y_d_rs, y_d_gs, fmap_rs, fmap_gs
+
+
+class DiscriminatorR(nn.Module):
+ def __init__(self, cfg: AttrDict, resolution: List[List[int]]):
+ super().__init__()
+
+ self.resolution = resolution
+ assert len(self.resolution) == 3, f"MRD layer requires list with len=3, got {self.resolution}"
+ self.lrelu_slope = 0.1
+
+ norm_f = weight_norm if cfg.use_spectral_norm == False else spectral_norm
+ if hasattr(cfg, "mrd_use_spectral_norm"):
+ print(f"[INFO] overriding MRD use_spectral_norm as {cfg.mrd_use_spectral_norm}")
+ norm_f = weight_norm if cfg.mrd_use_spectral_norm == False else spectral_norm
+ self.d_mult = cfg.discriminator_channel_mult
+ if hasattr(cfg, "mrd_channel_mult"):
+ print(f"[INFO] overriding mrd channel multiplier as {cfg.mrd_channel_mult}")
+ self.d_mult = cfg.mrd_channel_mult
+
+ self.convs = nn.ModuleList(
+ [
+ norm_f(nn.Conv2d(1, int(32 * self.d_mult), (3, 9), padding=(1, 4))),
+ norm_f(
+ nn.Conv2d(
+ int(32 * self.d_mult),
+ int(32 * self.d_mult),
+ (3, 9),
+ stride=(1, 2),
+ padding=(1, 4),
+ )
+ ),
+ norm_f(
+ nn.Conv2d(
+ int(32 * self.d_mult),
+ int(32 * self.d_mult),
+ (3, 9),
+ stride=(1, 2),
+ padding=(1, 4),
+ )
+ ),
+ norm_f(
+ nn.Conv2d(
+ int(32 * self.d_mult),
+ int(32 * self.d_mult),
+ (3, 9),
+ stride=(1, 2),
+ padding=(1, 4),
+ )
+ ),
+ norm_f(
+ nn.Conv2d(
+ int(32 * self.d_mult),
+ int(32 * self.d_mult),
+ (3, 3),
+ padding=(1, 1),
+ )
+ ),
+ ]
+ )
+ self.conv_post = norm_f(nn.Conv2d(int(32 * self.d_mult), 1, (3, 3), padding=(1, 1)))
+
+ def forward(self, x: torch.Tensor) -> Tuple[torch.Tensor, List[torch.Tensor]]:
+ fmap = []
+
+ x = self.spectrogram(x)
+ x = x.unsqueeze(1)
+ for l in self.convs:
+ x = l(x)
+ x = F.leaky_relu(x, self.lrelu_slope)
+ fmap.append(x)
+ x = self.conv_post(x)
+ fmap.append(x)
+ x = torch.flatten(x, 1, -1)
+
+ return x, fmap
+
+ def spectrogram(self, x: torch.Tensor) -> torch.Tensor:
+ n_fft, hop_length, win_length = self.resolution
+ x = F.pad(
+ x,
+ (int((n_fft - hop_length) / 2), int((n_fft - hop_length) / 2)),
+ mode="reflect",
+ )
+ x = x.squeeze(1)
+ x = torch.stft(
+ x,
+ n_fft=n_fft,
+ hop_length=hop_length,
+ win_length=win_length,
+ center=False,
+ return_complex=True,
+ )
+ x = torch.view_as_real(x) # [B, F, TT, 2]
+ mag = torch.norm(x, p=2, dim=-1) # [B, F, TT]
+
+ return mag
+
+
+class MultiResolutionDiscriminator(nn.Module):
+ def __init__(self, cfg, debug=False):
+ super().__init__()
+ self.resolutions = cfg.resolutions
+ assert len(self.resolutions) == 3, (
+ f"MRD requires list of list with len=3, each element having a list with len=3. Got {self.resolutions}"
+ )
+ self.discriminators = nn.ModuleList([DiscriminatorR(cfg, resolution) for resolution in self.resolutions])
+
+ def forward(
+ self, y: torch.Tensor, y_hat: torch.Tensor
+ ) -> Tuple[
+ List[torch.Tensor],
+ List[torch.Tensor],
+ List[List[torch.Tensor]],
+ List[List[torch.Tensor]],
+ ]:
+ y_d_rs = []
+ y_d_gs = []
+ fmap_rs = []
+ fmap_gs = []
+
+ for i, d in enumerate(self.discriminators):
+ y_d_r, fmap_r = d(x=y)
+ y_d_g, fmap_g = d(x=y_hat)
+ y_d_rs.append(y_d_r)
+ fmap_rs.append(fmap_r)
+ y_d_gs.append(y_d_g)
+ fmap_gs.append(fmap_g)
+
+ return y_d_rs, y_d_gs, fmap_rs, fmap_gs
+
+
+# Method based on descript-audio-codec: https://github.com/descriptinc/descript-audio-codec
+# Modified code adapted from https://github.com/gemelo-ai/vocos under the MIT license.
+# LICENSE is in incl_licenses directory.
+class DiscriminatorB(nn.Module):
+ def __init__(
+ self,
+ window_length: int,
+ channels: int = 32,
+ hop_factor: float = 0.25,
+ bands: Tuple[Tuple[float, float], ...] = (
+ (0.0, 0.1),
+ (0.1, 0.25),
+ (0.25, 0.5),
+ (0.5, 0.75),
+ (0.75, 1.0),
+ ),
+ ):
+ super().__init__()
+ self.window_length = window_length
+ self.hop_factor = hop_factor
+ self.spec_fn = Spectrogram(
+ n_fft=window_length,
+ hop_length=int(window_length * hop_factor),
+ win_length=window_length,
+ power=None,
+ )
+ n_fft = window_length // 2 + 1
+ bands = [(int(b[0] * n_fft), int(b[1] * n_fft)) for b in bands]
+ self.bands = bands
+ convs = lambda: nn.ModuleList(
+ [
+ weight_norm(nn.Conv2d(2, channels, (3, 9), (1, 1), padding=(1, 4))),
+ weight_norm(nn.Conv2d(channels, channels, (3, 9), (1, 2), padding=(1, 4))),
+ weight_norm(nn.Conv2d(channels, channels, (3, 9), (1, 2), padding=(1, 4))),
+ weight_norm(nn.Conv2d(channels, channels, (3, 9), (1, 2), padding=(1, 4))),
+ weight_norm(nn.Conv2d(channels, channels, (3, 3), (1, 1), padding=(1, 1))),
+ ]
+ )
+ self.band_convs = nn.ModuleList([convs() for _ in range(len(self.bands))])
+
+ self.conv_post = weight_norm(nn.Conv2d(channels, 1, (3, 3), (1, 1), padding=(1, 1)))
+
+ def spectrogram(self, x: torch.Tensor) -> List[torch.Tensor]:
+ # Remove DC offset
+ x = x - x.mean(dim=-1, keepdims=True)
+ # Peak normalize the volume of input audio
+ x = 0.8 * x / (x.abs().max(dim=-1, keepdim=True)[0] + 1e-9)
+ x = self.spec_fn(x)
+ x = torch.view_as_real(x)
+ x = x.permute(0, 3, 2, 1) # [B, F, T, C] -> [B, C, T, F]
+ # Split into bands
+ x_bands = [x[..., b[0] : b[1]] for b in self.bands]
+ return x_bands
+
+ def forward(self, x: torch.Tensor) -> Tuple[torch.Tensor, List[torch.Tensor]]:
+ x_bands = self.spectrogram(x.squeeze(1))
+ fmap = []
+ x = []
+
+ for band, stack in zip(x_bands, self.band_convs):
+ for i, layer in enumerate(stack):
+ band = layer(band)
+ band = torch.nn.functional.leaky_relu(band, 0.1)
+ if i > 0:
+ fmap.append(band)
+ x.append(band)
+
+ x = torch.cat(x, dim=-1)
+ x = self.conv_post(x)
+ fmap.append(x)
+
+ return x, fmap
+
+
+# Method based on descript-audio-codec: https://github.com/descriptinc/descript-audio-codec
+# Modified code adapted from https://github.com/gemelo-ai/vocos under the MIT license.
+# LICENSE is in incl_licenses directory.
+class MultiBandDiscriminator(nn.Module):
+ def __init__(
+ self,
+ h,
+ ):
+ """
+ Multi-band multi-scale STFT discriminator, with the architecture based on https://github.com/descriptinc/descript-audio-codec.
+ and the modified code adapted from https://github.com/gemelo-ai/vocos.
+ """
+ super().__init__()
+ # fft_sizes (list[int]): Tuple of window lengths for FFT. Defaults to [2048, 1024, 512] if not set in h.
+ self.fft_sizes = h.get("mbd_fft_sizes", [2048, 1024, 512])
+ self.discriminators = nn.ModuleList([DiscriminatorB(window_length=w) for w in self.fft_sizes])
+
+ def forward(
+ self, y: torch.Tensor, y_hat: torch.Tensor
+ ) -> Tuple[
+ List[torch.Tensor],
+ List[torch.Tensor],
+ List[List[torch.Tensor]],
+ List[List[torch.Tensor]],
+ ]:
+ y_d_rs = []
+ y_d_gs = []
+ fmap_rs = []
+ fmap_gs = []
+
+ for d in self.discriminators:
+ y_d_r, fmap_r = d(x=y)
+ y_d_g, fmap_g = d(x=y_hat)
+ y_d_rs.append(y_d_r)
+ fmap_rs.append(fmap_r)
+ y_d_gs.append(y_d_g)
+ fmap_gs.append(fmap_g)
+
+ return y_d_rs, y_d_gs, fmap_rs, fmap_gs
+
+
+# Adapted from https://github.com/open-mmlab/Amphion/blob/main/models/vocoders/gan/discriminator/mssbcqtd.py under the MIT license.
+# LICENSE is in incl_licenses directory.
+class DiscriminatorCQT(nn.Module):
+ def __init__(self, cfg: AttrDict, hop_length: int, n_octaves: int, bins_per_octave: int):
+ super().__init__()
+ self.cfg = cfg
+
+ self.filters = cfg["cqtd_filters"]
+ self.max_filters = cfg["cqtd_max_filters"]
+ self.filters_scale = cfg["cqtd_filters_scale"]
+ self.kernel_size = (3, 9)
+ self.dilations = cfg["cqtd_dilations"]
+ self.stride = (1, 2)
+
+ self.in_channels = cfg["cqtd_in_channels"]
+ self.out_channels = cfg["cqtd_out_channels"]
+ self.fs = cfg["sampling_rate"]
+ self.hop_length = hop_length
+ self.n_octaves = n_octaves
+ self.bins_per_octave = bins_per_octave
+
+ # Lazy-load
+ from nnAudio import features
+
+ self.cqt_transform = features.cqt.CQT2010v2(
+ sr=self.fs * 2,
+ hop_length=self.hop_length,
+ n_bins=self.bins_per_octave * self.n_octaves,
+ bins_per_octave=self.bins_per_octave,
+ output_format="Complex",
+ pad_mode="constant",
+ )
+
+ self.conv_pres = nn.ModuleList()
+ for _ in range(self.n_octaves):
+ self.conv_pres.append(
+ nn.Conv2d(
+ self.in_channels * 2,
+ self.in_channels * 2,
+ kernel_size=self.kernel_size,
+ padding=self.get_2d_padding(self.kernel_size),
+ )
+ )
+
+ self.convs = nn.ModuleList()
+
+ self.convs.append(
+ nn.Conv2d(
+ self.in_channels * 2,
+ self.filters,
+ kernel_size=self.kernel_size,
+ padding=self.get_2d_padding(self.kernel_size),
+ )
+ )
+
+ in_chs = min(self.filters_scale * self.filters, self.max_filters)
+ for i, dilation in enumerate(self.dilations):
+ out_chs = min((self.filters_scale ** (i + 1)) * self.filters, self.max_filters)
+ self.convs.append(
+ weight_norm(
+ nn.Conv2d(
+ in_chs,
+ out_chs,
+ kernel_size=self.kernel_size,
+ stride=self.stride,
+ dilation=(dilation, 1),
+ padding=self.get_2d_padding(self.kernel_size, (dilation, 1)),
+ )
+ )
+ )
+ in_chs = out_chs
+ out_chs = min(
+ (self.filters_scale ** (len(self.dilations) + 1)) * self.filters,
+ self.max_filters,
+ )
+ self.convs.append(
+ weight_norm(
+ nn.Conv2d(
+ in_chs,
+ out_chs,
+ kernel_size=(self.kernel_size[0], self.kernel_size[0]),
+ padding=self.get_2d_padding((self.kernel_size[0], self.kernel_size[0])),
+ )
+ )
+ )
+
+ self.conv_post = weight_norm(
+ nn.Conv2d(
+ out_chs,
+ self.out_channels,
+ kernel_size=(self.kernel_size[0], self.kernel_size[0]),
+ padding=self.get_2d_padding((self.kernel_size[0], self.kernel_size[0])),
+ )
+ )
+
+ self.activation = torch.nn.LeakyReLU(negative_slope=0.1)
+ self.resample = Resample(orig_freq=self.fs, new_freq=self.fs * 2)
+
+ self.cqtd_normalize_volume = self.cfg.get("cqtd_normalize_volume", False)
+ if self.cqtd_normalize_volume:
+ print(
+ "[INFO] cqtd_normalize_volume set to True. Will apply DC offset removal & peak volume normalization in CQTD!"
+ )
+
+ def get_2d_padding(
+ self,
+ kernel_size: typing.Tuple[int, int],
+ dilation: typing.Tuple[int, int] = (1, 1),
+ ):
+ return (
+ ((kernel_size[0] - 1) * dilation[0]) // 2,
+ ((kernel_size[1] - 1) * dilation[1]) // 2,
+ )
+
+ def forward(self, x: torch.tensor) -> Tuple[torch.Tensor, List[torch.Tensor]]:
+ fmap = []
+
+ if self.cqtd_normalize_volume:
+ # Remove DC offset
+ x = x - x.mean(dim=-1, keepdims=True)
+ # Peak normalize the volume of input audio
+ x = 0.8 * x / (x.abs().max(dim=-1, keepdim=True)[0] + 1e-9)
+
+ x = self.resample(x)
+
+ z = self.cqt_transform(x)
+
+ z_amplitude = z[:, :, :, 0].unsqueeze(1)
+ z_phase = z[:, :, :, 1].unsqueeze(1)
+
+ z = torch.cat([z_amplitude, z_phase], dim=1)
+ z = torch.permute(z, (0, 1, 3, 2)) # [B, C, W, T] -> [B, C, T, W]
+
+ latent_z = []
+ for i in range(self.n_octaves):
+ latent_z.append(
+ self.conv_pres[i](
+ z[
+ :,
+ :,
+ :,
+ i * self.bins_per_octave : (i + 1) * self.bins_per_octave,
+ ]
+ )
+ )
+ latent_z = torch.cat(latent_z, dim=-1)
+
+ for i, l in enumerate(self.convs):
+ latent_z = l(latent_z)
+
+ latent_z = self.activation(latent_z)
+ fmap.append(latent_z)
+
+ latent_z = self.conv_post(latent_z)
+
+ return latent_z, fmap
+
+
+class MultiScaleSubbandCQTDiscriminator(nn.Module):
+ def __init__(self, cfg: AttrDict):
+ super().__init__()
+
+ self.cfg = cfg
+ # Using get with defaults
+ self.cfg["cqtd_filters"] = self.cfg.get("cqtd_filters", 32)
+ self.cfg["cqtd_max_filters"] = self.cfg.get("cqtd_max_filters", 1024)
+ self.cfg["cqtd_filters_scale"] = self.cfg.get("cqtd_filters_scale", 1)
+ self.cfg["cqtd_dilations"] = self.cfg.get("cqtd_dilations", [1, 2, 4])
+ self.cfg["cqtd_in_channels"] = self.cfg.get("cqtd_in_channels", 1)
+ self.cfg["cqtd_out_channels"] = self.cfg.get("cqtd_out_channels", 1)
+ # Multi-scale params to loop over
+ self.cfg["cqtd_hop_lengths"] = self.cfg.get("cqtd_hop_lengths", [512, 256, 256])
+ self.cfg["cqtd_n_octaves"] = self.cfg.get("cqtd_n_octaves", [9, 9, 9])
+ self.cfg["cqtd_bins_per_octaves"] = self.cfg.get("cqtd_bins_per_octaves", [24, 36, 48])
+
+ self.discriminators = nn.ModuleList(
+ [
+ DiscriminatorCQT(
+ self.cfg,
+ hop_length=self.cfg["cqtd_hop_lengths"][i],
+ n_octaves=self.cfg["cqtd_n_octaves"][i],
+ bins_per_octave=self.cfg["cqtd_bins_per_octaves"][i],
+ )
+ for i in range(len(self.cfg["cqtd_hop_lengths"]))
+ ]
+ )
+
+ def forward(
+ self, y: torch.Tensor, y_hat: torch.Tensor
+ ) -> Tuple[
+ List[torch.Tensor],
+ List[torch.Tensor],
+ List[List[torch.Tensor]],
+ List[List[torch.Tensor]],
+ ]:
+ y_d_rs = []
+ y_d_gs = []
+ fmap_rs = []
+ fmap_gs = []
+
+ for disc in self.discriminators:
+ y_d_r, fmap_r = disc(y)
+ y_d_g, fmap_g = disc(y_hat)
+ y_d_rs.append(y_d_r)
+ fmap_rs.append(fmap_r)
+ y_d_gs.append(y_d_g)
+ fmap_gs.append(fmap_g)
+
+ return y_d_rs, y_d_gs, fmap_rs, fmap_gs
+
+
+class CombinedDiscriminator(nn.Module):
+ """
+ Wrapper of chaining multiple discrimiantor architectures.
+ Example: combine mbd and cqtd as a single class
+ """
+
+ def __init__(self, list_discriminator: List[nn.Module]):
+ super().__init__()
+ self.discrimiantor = nn.ModuleList(list_discriminator)
+
+ def forward(
+ self, y: torch.Tensor, y_hat: torch.Tensor
+ ) -> Tuple[
+ List[torch.Tensor],
+ List[torch.Tensor],
+ List[List[torch.Tensor]],
+ List[List[torch.Tensor]],
+ ]:
+ y_d_rs = []
+ y_d_gs = []
+ fmap_rs = []
+ fmap_gs = []
+
+ for disc in self.discrimiantor:
+ y_d_r, y_d_g, fmap_r, fmap_g = disc(y, y_hat)
+ y_d_rs.extend(y_d_r)
+ fmap_rs.extend(fmap_r)
+ y_d_gs.extend(y_d_g)
+ fmap_gs.extend(fmap_g)
+
+ return y_d_rs, y_d_gs, fmap_rs, fmap_gs
diff --git a/GPT_SoVITS/BigVGAN/env.py b/GPT_SoVITS/BigVGAN/env.py
new file mode 100644
index 0000000000000000000000000000000000000000..cf8ac6cea644c78d115dd3902b902993f366ee61
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/env.py
@@ -0,0 +1,18 @@
+# Adapted from https://github.com/jik876/hifi-gan under the MIT license.
+# LICENSE is in incl_licenses directory.
+
+import os
+import shutil
+
+
+class AttrDict(dict):
+ def __init__(self, *args, **kwargs):
+ super(AttrDict, self).__init__(*args, **kwargs)
+ self.__dict__ = self
+
+
+def build_env(config, config_name, path):
+ t_path = os.path.join(path, config_name)
+ if config != t_path:
+ os.makedirs(path, exist_ok=True)
+ shutil.copyfile(config, os.path.join(path, config_name))
diff --git a/GPT_SoVITS/BigVGAN/inference.py b/GPT_SoVITS/BigVGAN/inference.py
new file mode 100644
index 0000000000000000000000000000000000000000..5f892a3c807a7020eff7fea35179b0f6e5f991c9
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/inference.py
@@ -0,0 +1,85 @@
+# Adapted from https://github.com/jik876/hifi-gan under the MIT license.
+# LICENSE is in incl_licenses directory.
+
+from __future__ import absolute_import, division, print_function, unicode_literals
+
+import os
+import argparse
+import json
+import torch
+import librosa
+from utils import load_checkpoint
+from meldataset import get_mel_spectrogram
+from scipy.io.wavfile import write
+from env import AttrDict
+from meldataset import MAX_WAV_VALUE
+from bigvgan import BigVGAN as Generator
+
+h = None
+device = None
+torch.backends.cudnn.benchmark = False
+
+
+def inference(a, h):
+ generator = Generator(h, use_cuda_kernel=a.use_cuda_kernel).to(device)
+
+ state_dict_g = load_checkpoint(a.checkpoint_file, device)
+ generator.load_state_dict(state_dict_g["generator"])
+
+ filelist = os.listdir(a.input_wavs_dir)
+
+ os.makedirs(a.output_dir, exist_ok=True)
+
+ generator.eval()
+ generator.remove_weight_norm()
+ with torch.no_grad():
+ for i, filname in enumerate(filelist):
+ # Load the ground truth audio and resample if necessary
+ wav, sr = librosa.load(os.path.join(a.input_wavs_dir, filname), sr=h.sampling_rate, mono=True)
+ wav = torch.FloatTensor(wav).to(device)
+ # Compute mel spectrogram from the ground truth audio
+ x = get_mel_spectrogram(wav.unsqueeze(0), generator.h)
+
+ y_g_hat = generator(x)
+
+ audio = y_g_hat.squeeze()
+ audio = audio * MAX_WAV_VALUE
+ audio = audio.cpu().numpy().astype("int16")
+
+ output_file = os.path.join(a.output_dir, os.path.splitext(filname)[0] + "_generated.wav")
+ write(output_file, h.sampling_rate, audio)
+ print(output_file)
+
+
+def main():
+ print("Initializing Inference Process..")
+
+ parser = argparse.ArgumentParser()
+ parser.add_argument("--input_wavs_dir", default="test_files")
+ parser.add_argument("--output_dir", default="generated_files")
+ parser.add_argument("--checkpoint_file", required=True)
+ parser.add_argument("--use_cuda_kernel", action="store_true", default=False)
+
+ a = parser.parse_args()
+
+ config_file = os.path.join(os.path.split(a.checkpoint_file)[0], "config.json")
+ with open(config_file) as f:
+ data = f.read()
+
+ global h
+ json_config = json.loads(data)
+ h = AttrDict(json_config)
+
+ torch.manual_seed(h.seed)
+ global device
+ if torch.cuda.is_available():
+ torch.cuda.manual_seed(h.seed)
+ device = torch.device("cuda")
+ else:
+ device = torch.device("cpu")
+
+ inference(a, h)
+
+
+if __name__ == "__main__":
+ main()
diff --git a/GPT_SoVITS/BigVGAN/inference_e2e.py b/GPT_SoVITS/BigVGAN/inference_e2e.py
new file mode 100644
index 0000000000000000000000000000000000000000..9c0df77435e91935beaca365dd5fd38d76098a4a
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/inference_e2e.py
@@ -0,0 +1,100 @@
+# Adapted from https://github.com/jik876/hifi-gan under the MIT license.
+# LICENSE is in incl_licenses directory.
+
+from __future__ import absolute_import, division, print_function, unicode_literals
+
+import glob
+import os
+import numpy as np
+import argparse
+import json
+import torch
+from scipy.io.wavfile import write
+from env import AttrDict
+from meldataset import MAX_WAV_VALUE
+from bigvgan import BigVGAN as Generator
+
+h = None
+device = None
+torch.backends.cudnn.benchmark = False
+
+
+def load_checkpoint(filepath, device):
+ assert os.path.isfile(filepath)
+ print(f"Loading '{filepath}'")
+ checkpoint_dict = torch.load(filepath, map_location=device)
+ print("Complete.")
+ return checkpoint_dict
+
+
+def scan_checkpoint(cp_dir, prefix):
+ pattern = os.path.join(cp_dir, prefix + "*")
+ cp_list = glob.glob(pattern)
+ if len(cp_list) == 0:
+ return ""
+ return sorted(cp_list)[-1]
+
+
+def inference(a, h):
+ generator = Generator(h, use_cuda_kernel=a.use_cuda_kernel).to(device)
+
+ state_dict_g = load_checkpoint(a.checkpoint_file, device)
+ generator.load_state_dict(state_dict_g["generator"])
+
+ filelist = os.listdir(a.input_mels_dir)
+
+ os.makedirs(a.output_dir, exist_ok=True)
+
+ generator.eval()
+ generator.remove_weight_norm()
+ with torch.no_grad():
+ for i, filname in enumerate(filelist):
+ # Load the mel spectrogram in .npy format
+ x = np.load(os.path.join(a.input_mels_dir, filname))
+ x = torch.FloatTensor(x).to(device)
+ if len(x.shape) == 2:
+ x = x.unsqueeze(0)
+
+ y_g_hat = generator(x)
+
+ audio = y_g_hat.squeeze()
+ audio = audio * MAX_WAV_VALUE
+ audio = audio.cpu().numpy().astype("int16")
+
+ output_file = os.path.join(a.output_dir, os.path.splitext(filname)[0] + "_generated_e2e.wav")
+ write(output_file, h.sampling_rate, audio)
+ print(output_file)
+
+
+def main():
+ print("Initializing Inference Process..")
+
+ parser = argparse.ArgumentParser()
+ parser.add_argument("--input_mels_dir", default="test_mel_files")
+ parser.add_argument("--output_dir", default="generated_files_from_mel")
+ parser.add_argument("--checkpoint_file", required=True)
+ parser.add_argument("--use_cuda_kernel", action="store_true", default=False)
+
+ a = parser.parse_args()
+
+ config_file = os.path.join(os.path.split(a.checkpoint_file)[0], "config.json")
+ with open(config_file) as f:
+ data = f.read()
+
+ global h
+ json_config = json.loads(data)
+ h = AttrDict(json_config)
+
+ torch.manual_seed(h.seed)
+ global device
+ if torch.cuda.is_available():
+ torch.cuda.manual_seed(h.seed)
+ device = torch.device("cuda")
+ else:
+ device = torch.device("cpu")
+
+ inference(a, h)
+
+
+if __name__ == "__main__":
+ main()
diff --git a/GPT_SoVITS/BigVGAN/loss.py b/GPT_SoVITS/BigVGAN/loss.py
new file mode 100644
index 0000000000000000000000000000000000000000..c295a144ff7bcfc0d91d9d4676bedfa7015cdb79
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/loss.py
@@ -0,0 +1,238 @@
+# Copyright (c) 2024 NVIDIA CORPORATION.
+# Licensed under the MIT license.
+
+# Adapted from https://github.com/jik876/hifi-gan under the MIT license.
+# LICENSE is in incl_licenses directory.
+
+
+import torch
+import torch.nn as nn
+from librosa.filters import mel as librosa_mel_fn
+from scipy import signal
+
+import typing
+from typing import List, Tuple
+from collections import namedtuple
+import math
+import functools
+
+
+# Adapted from https://github.com/descriptinc/descript-audio-codec/blob/main/dac/nn/loss.py under the MIT license.
+# LICENSE is in incl_licenses directory.
+class MultiScaleMelSpectrogramLoss(nn.Module):
+ """Compute distance between mel spectrograms. Can be used
+ in a multi-scale way.
+
+ Parameters
+ ----------
+ n_mels : List[int]
+ Number of mels per STFT, by default [5, 10, 20, 40, 80, 160, 320],
+ window_lengths : List[int], optional
+ Length of each window of each STFT, by default [32, 64, 128, 256, 512, 1024, 2048]
+ loss_fn : typing.Callable, optional
+ How to compare each loss, by default nn.L1Loss()
+ clamp_eps : float, optional
+ Clamp on the log magnitude, below, by default 1e-5
+ mag_weight : float, optional
+ Weight of raw magnitude portion of loss, by default 0.0 (no ampliciation on mag part)
+ log_weight : float, optional
+ Weight of log magnitude portion of loss, by default 1.0
+ pow : float, optional
+ Power to raise magnitude to before taking log, by default 1.0
+ weight : float, optional
+ Weight of this loss, by default 1.0
+ match_stride : bool, optional
+ Whether to match the stride of convolutional layers, by default False
+
+ Implementation copied from: https://github.com/descriptinc/lyrebird-audiotools/blob/961786aa1a9d628cca0c0486e5885a457fe70c1a/audiotools/metrics/spectral.py
+ Additional code copied and modified from https://github.com/descriptinc/audiotools/blob/master/audiotools/core/audio_signal.py
+ """
+
+ def __init__(
+ self,
+ sampling_rate: int,
+ n_mels: List[int] = [5, 10, 20, 40, 80, 160, 320],
+ window_lengths: List[int] = [32, 64, 128, 256, 512, 1024, 2048],
+ loss_fn: typing.Callable = nn.L1Loss(),
+ clamp_eps: float = 1e-5,
+ mag_weight: float = 0.0,
+ log_weight: float = 1.0,
+ pow: float = 1.0,
+ weight: float = 1.0,
+ match_stride: bool = False,
+ mel_fmin: List[float] = [0, 0, 0, 0, 0, 0, 0],
+ mel_fmax: List[float] = [None, None, None, None, None, None, None],
+ window_type: str = "hann",
+ ):
+ super().__init__()
+ self.sampling_rate = sampling_rate
+
+ STFTParams = namedtuple(
+ "STFTParams",
+ ["window_length", "hop_length", "window_type", "match_stride"],
+ )
+
+ self.stft_params = [
+ STFTParams(
+ window_length=w,
+ hop_length=w // 4,
+ match_stride=match_stride,
+ window_type=window_type,
+ )
+ for w in window_lengths
+ ]
+ self.n_mels = n_mels
+ self.loss_fn = loss_fn
+ self.clamp_eps = clamp_eps
+ self.log_weight = log_weight
+ self.mag_weight = mag_weight
+ self.weight = weight
+ self.mel_fmin = mel_fmin
+ self.mel_fmax = mel_fmax
+ self.pow = pow
+
+ @staticmethod
+ @functools.lru_cache(None)
+ def get_window(
+ window_type,
+ window_length,
+ ):
+ return signal.get_window(window_type, window_length)
+
+ @staticmethod
+ @functools.lru_cache(None)
+ def get_mel_filters(sr, n_fft, n_mels, fmin, fmax):
+ return librosa_mel_fn(sr=sr, n_fft=n_fft, n_mels=n_mels, fmin=fmin, fmax=fmax)
+
+ def mel_spectrogram(
+ self,
+ wav,
+ n_mels,
+ fmin,
+ fmax,
+ window_length,
+ hop_length,
+ match_stride,
+ window_type,
+ ):
+ """
+ Mirrors AudioSignal.mel_spectrogram used by BigVGAN-v2 training from:
+ https://github.com/descriptinc/audiotools/blob/master/audiotools/core/audio_signal.py
+ """
+ B, C, T = wav.shape
+
+ if match_stride:
+ assert hop_length == window_length // 4, "For match_stride, hop must equal n_fft // 4"
+ right_pad = math.ceil(T / hop_length) * hop_length - T
+ pad = (window_length - hop_length) // 2
+ else:
+ right_pad = 0
+ pad = 0
+
+ wav = torch.nn.functional.pad(wav, (pad, pad + right_pad), mode="reflect")
+
+ window = self.get_window(window_type, window_length)
+ window = torch.from_numpy(window).to(wav.device).float()
+
+ stft = torch.stft(
+ wav.reshape(-1, T),
+ n_fft=window_length,
+ hop_length=hop_length,
+ window=window,
+ return_complex=True,
+ center=True,
+ )
+ _, nf, nt = stft.shape
+ stft = stft.reshape(B, C, nf, nt)
+ if match_stride:
+ """
+ Drop first two and last two frames, which are added, because of padding. Now num_frames * hop_length = num_samples.
+ """
+ stft = stft[..., 2:-2]
+ magnitude = torch.abs(stft)
+
+ nf = magnitude.shape[2]
+ mel_basis = self.get_mel_filters(self.sampling_rate, 2 * (nf - 1), n_mels, fmin, fmax)
+ mel_basis = torch.from_numpy(mel_basis).to(wav.device)
+ mel_spectrogram = magnitude.transpose(2, -1) @ mel_basis.T
+ mel_spectrogram = mel_spectrogram.transpose(-1, 2)
+
+ return mel_spectrogram
+
+ def forward(self, x: torch.Tensor, y: torch.Tensor) -> torch.Tensor:
+ """Computes mel loss between an estimate and a reference
+ signal.
+
+ Parameters
+ ----------
+ x : torch.Tensor
+ Estimate signal
+ y : torch.Tensor
+ Reference signal
+
+ Returns
+ -------
+ torch.Tensor
+ Mel loss.
+ """
+
+ loss = 0.0
+ for n_mels, fmin, fmax, s in zip(self.n_mels, self.mel_fmin, self.mel_fmax, self.stft_params):
+ kwargs = {
+ "n_mels": n_mels,
+ "fmin": fmin,
+ "fmax": fmax,
+ "window_length": s.window_length,
+ "hop_length": s.hop_length,
+ "match_stride": s.match_stride,
+ "window_type": s.window_type,
+ }
+
+ x_mels = self.mel_spectrogram(x, **kwargs)
+ y_mels = self.mel_spectrogram(y, **kwargs)
+ x_logmels = torch.log(x_mels.clamp(min=self.clamp_eps).pow(self.pow)) / torch.log(torch.tensor(10.0))
+ y_logmels = torch.log(y_mels.clamp(min=self.clamp_eps).pow(self.pow)) / torch.log(torch.tensor(10.0))
+
+ loss += self.log_weight * self.loss_fn(x_logmels, y_logmels)
+ loss += self.mag_weight * self.loss_fn(x_logmels, y_logmels)
+
+ return loss
+
+
+# Loss functions
+def feature_loss(fmap_r: List[List[torch.Tensor]], fmap_g: List[List[torch.Tensor]]) -> torch.Tensor:
+ loss = 0
+ for dr, dg in zip(fmap_r, fmap_g):
+ for rl, gl in zip(dr, dg):
+ loss += torch.mean(torch.abs(rl - gl))
+
+ return loss * 2 # This equates to lambda=2.0 for the feature matching loss
+
+
+def discriminator_loss(
+ disc_real_outputs: List[torch.Tensor], disc_generated_outputs: List[torch.Tensor]
+) -> Tuple[torch.Tensor, List[torch.Tensor], List[torch.Tensor]]:
+ loss = 0
+ r_losses = []
+ g_losses = []
+ for dr, dg in zip(disc_real_outputs, disc_generated_outputs):
+ r_loss = torch.mean((1 - dr) ** 2)
+ g_loss = torch.mean(dg**2)
+ loss += r_loss + g_loss
+ r_losses.append(r_loss.item())
+ g_losses.append(g_loss.item())
+
+ return loss, r_losses, g_losses
+
+
+def generator_loss(
+ disc_outputs: List[torch.Tensor],
+) -> Tuple[torch.Tensor, List[torch.Tensor]]:
+ loss = 0
+ gen_losses = []
+ for dg in disc_outputs:
+ l = torch.mean((1 - dg) ** 2)
+ gen_losses.append(l)
+ loss += l
+
+ return loss, gen_losses
diff --git a/GPT_SoVITS/BigVGAN/meldataset.py b/GPT_SoVITS/BigVGAN/meldataset.py
new file mode 100644
index 0000000000000000000000000000000000000000..dc12c9874cfb9958d6f4842cc067ffda66a390eb
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/meldataset.py
@@ -0,0 +1,370 @@
+# Copyright (c) 2024 NVIDIA CORPORATION.
+# Licensed under the MIT license.
+
+# Adapted from https://github.com/jik876/hifi-gan under the MIT license.
+# LICENSE is in incl_licenses directory.
+
+import math
+import os
+import random
+import torch
+import torch.utils.data
+import numpy as np
+import librosa
+from librosa.filters import mel as librosa_mel_fn
+import pathlib
+from tqdm import tqdm
+from typing import List, Tuple, Optional
+from .env import AttrDict
+
+MAX_WAV_VALUE = 32767.0 # NOTE: 32768.0 -1 to prevent int16 overflow (results in popping sound in corner cases)
+
+
+def dynamic_range_compression(x, C=1, clip_val=1e-5):
+ return np.log(np.clip(x, a_min=clip_val, a_max=None) * C)
+
+
+def dynamic_range_decompression(x, C=1):
+ return np.exp(x) / C
+
+
+def dynamic_range_compression_torch(x, C=1, clip_val=1e-5):
+ return torch.log(torch.clamp(x, min=clip_val) * C)
+
+
+def dynamic_range_decompression_torch(x, C=1):
+ return torch.exp(x) / C
+
+
+def spectral_normalize_torch(magnitudes):
+ return dynamic_range_compression_torch(magnitudes)
+
+
+def spectral_de_normalize_torch(magnitudes):
+ return dynamic_range_decompression_torch(magnitudes)
+
+
+mel_basis_cache = {}
+hann_window_cache = {}
+
+
+def mel_spectrogram(
+ y: torch.Tensor,
+ n_fft: int,
+ num_mels: int,
+ sampling_rate: int,
+ hop_size: int,
+ win_size: int,
+ fmin: int,
+ fmax: int = None,
+ center: bool = False,
+) -> torch.Tensor:
+ """
+ Calculate the mel spectrogram of an input signal.
+ This function uses slaney norm for the librosa mel filterbank (using librosa.filters.mel) and uses Hann window for STFT (using torch.stft).
+
+ Args:
+ y (torch.Tensor): Input signal.
+ n_fft (int): FFT size.
+ num_mels (int): Number of mel bins.
+ sampling_rate (int): Sampling rate of the input signal.
+ hop_size (int): Hop size for STFT.
+ win_size (int): Window size for STFT.
+ fmin (int): Minimum frequency for mel filterbank.
+ fmax (int): Maximum frequency for mel filterbank. If None, defaults to half the sampling rate (fmax = sr / 2.0) inside librosa_mel_fn
+ center (bool): Whether to pad the input to center the frames. Default is False.
+
+ Returns:
+ torch.Tensor: Mel spectrogram.
+ """
+ if torch.min(y) < -1.0:
+ print(f"[WARNING] Min value of input waveform signal is {torch.min(y)}")
+ if torch.max(y) > 1.0:
+ print(f"[WARNING] Max value of input waveform signal is {torch.max(y)}")
+
+ device = y.device
+ key = f"{n_fft}_{num_mels}_{sampling_rate}_{hop_size}_{win_size}_{fmin}_{fmax}_{device}"
+
+ if key not in mel_basis_cache:
+ mel = librosa_mel_fn(sr=sampling_rate, n_fft=n_fft, n_mels=num_mels, fmin=fmin, fmax=fmax)
+ mel_basis_cache[key] = torch.from_numpy(mel).float().to(device)
+ hann_window_cache[key] = torch.hann_window(win_size).to(device)
+
+ mel_basis = mel_basis_cache[key]
+ hann_window = hann_window_cache[key]
+
+ padding = (n_fft - hop_size) // 2
+ y = torch.nn.functional.pad(y.unsqueeze(1), (padding, padding), mode="reflect").squeeze(1)
+
+ spec = torch.stft(
+ y,
+ n_fft,
+ hop_length=hop_size,
+ win_length=win_size,
+ window=hann_window,
+ center=center,
+ pad_mode="reflect",
+ normalized=False,
+ onesided=True,
+ return_complex=True,
+ )
+ spec = torch.sqrt(torch.view_as_real(spec).pow(2).sum(-1) + 1e-9)
+
+ mel_spec = torch.matmul(mel_basis, spec)
+ mel_spec = spectral_normalize_torch(mel_spec)
+
+ return mel_spec
+
+
+def get_mel_spectrogram(wav, h):
+ """
+ Generate mel spectrogram from a waveform using given hyperparameters.
+
+ Args:
+ wav (torch.Tensor): Input waveform.
+ h: Hyperparameters object with attributes n_fft, num_mels, sampling_rate, hop_size, win_size, fmin, fmax.
+
+ Returns:
+ torch.Tensor: Mel spectrogram.
+ """
+ return mel_spectrogram(
+ wav,
+ h.n_fft,
+ h.num_mels,
+ h.sampling_rate,
+ h.hop_size,
+ h.win_size,
+ h.fmin,
+ h.fmax,
+ )
+
+
+def get_dataset_filelist(a):
+ training_files = []
+ validation_files = []
+ list_unseen_validation_files = []
+
+ with open(a.input_training_file, "r", encoding="utf-8") as fi:
+ training_files = [
+ os.path.join(a.input_wavs_dir, x.split("|")[0] + ".wav") for x in fi.read().split("\n") if len(x) > 0
+ ]
+ print(f"first training file: {training_files[0]}")
+
+ with open(a.input_validation_file, "r", encoding="utf-8") as fi:
+ validation_files = [
+ os.path.join(a.input_wavs_dir, x.split("|")[0] + ".wav") for x in fi.read().split("\n") if len(x) > 0
+ ]
+ print(f"first validation file: {validation_files[0]}")
+
+ for i in range(len(a.list_input_unseen_validation_file)):
+ with open(a.list_input_unseen_validation_file[i], "r", encoding="utf-8") as fi:
+ unseen_validation_files = [
+ os.path.join(a.list_input_unseen_wavs_dir[i], x.split("|")[0] + ".wav")
+ for x in fi.read().split("\n")
+ if len(x) > 0
+ ]
+ print(f"first unseen {i}th validation fileset: {unseen_validation_files[0]}")
+ list_unseen_validation_files.append(unseen_validation_files)
+
+ return training_files, validation_files, list_unseen_validation_files
+
+
+class MelDataset(torch.utils.data.Dataset):
+ def __init__(
+ self,
+ training_files: List[str],
+ hparams: AttrDict,
+ segment_size: int,
+ n_fft: int,
+ num_mels: int,
+ hop_size: int,
+ win_size: int,
+ sampling_rate: int,
+ fmin: int,
+ fmax: Optional[int],
+ split: bool = True,
+ shuffle: bool = True,
+ device: str = None,
+ fmax_loss: Optional[int] = None,
+ fine_tuning: bool = False,
+ base_mels_path: str = None,
+ is_seen: bool = True,
+ ):
+ self.audio_files = training_files
+ random.seed(1234)
+ if shuffle:
+ random.shuffle(self.audio_files)
+ self.hparams = hparams
+ self.is_seen = is_seen
+ if self.is_seen:
+ self.name = pathlib.Path(self.audio_files[0]).parts[0]
+ else:
+ self.name = "-".join(pathlib.Path(self.audio_files[0]).parts[:2]).strip("/")
+
+ self.segment_size = segment_size
+ self.sampling_rate = sampling_rate
+ self.split = split
+ self.n_fft = n_fft
+ self.num_mels = num_mels
+ self.hop_size = hop_size
+ self.win_size = win_size
+ self.fmin = fmin
+ self.fmax = fmax
+ self.fmax_loss = fmax_loss
+ self.device = device
+ self.fine_tuning = fine_tuning
+ self.base_mels_path = base_mels_path
+
+ print("[INFO] checking dataset integrity...")
+ for i in tqdm(range(len(self.audio_files))):
+ assert os.path.exists(self.audio_files[i]), f"{self.audio_files[i]} not found"
+
+ def __getitem__(self, index: int) -> Tuple[torch.Tensor, torch.Tensor, str, torch.Tensor]:
+ try:
+ filename = self.audio_files[index]
+
+ # Use librosa.load that ensures loading waveform into mono with [-1, 1] float values
+ # Audio is ndarray with shape [T_time]. Disable auto-resampling here to minimize overhead
+ # The on-the-fly resampling during training will be done only for the obtained random chunk
+ audio, source_sampling_rate = librosa.load(filename, sr=None, mono=True)
+
+ # Main logic that uses pair for training BigVGAN
+ if not self.fine_tuning:
+ if self.split: # Training step
+ # Obtain randomized audio chunk
+ if source_sampling_rate != self.sampling_rate:
+ # Adjust segment size to crop if the source sr is different
+ target_segment_size = math.ceil(self.segment_size * (source_sampling_rate / self.sampling_rate))
+ else:
+ target_segment_size = self.segment_size
+
+ # Compute upper bound index for the random chunk
+ random_chunk_upper_bound = max(0, audio.shape[0] - target_segment_size)
+
+ # Crop or pad audio to obtain random chunk with target_segment_size
+ if audio.shape[0] >= target_segment_size:
+ audio_start = random.randint(0, random_chunk_upper_bound)
+ audio = audio[audio_start : audio_start + target_segment_size]
+ else:
+ audio = np.pad(
+ audio,
+ (0, target_segment_size - audio.shape[0]),
+ mode="constant",
+ )
+
+ # Resample audio chunk to self.sampling rate
+ if source_sampling_rate != self.sampling_rate:
+ audio = librosa.resample(
+ audio,
+ orig_sr=source_sampling_rate,
+ target_sr=self.sampling_rate,
+ )
+ if audio.shape[0] > self.segment_size:
+ # trim last elements to match self.segment_size (e.g., 16385 for 44khz downsampled to 24khz -> 16384)
+ audio = audio[: self.segment_size]
+
+ else: # Validation step
+ # Resample full audio clip to target sampling rate
+ if source_sampling_rate != self.sampling_rate:
+ audio = librosa.resample(
+ audio,
+ orig_sr=source_sampling_rate,
+ target_sr=self.sampling_rate,
+ )
+ # Trim last elements to match audio length to self.hop_size * n for evaluation
+ if (audio.shape[0] % self.hop_size) != 0:
+ audio = audio[: -(audio.shape[0] % self.hop_size)]
+
+ # BigVGAN is trained using volume-normalized waveform
+ audio = librosa.util.normalize(audio) * 0.95
+
+ # Cast ndarray to torch tensor
+ audio = torch.FloatTensor(audio)
+ audio = audio.unsqueeze(0) # [B(1), self.segment_size]
+
+ # Compute mel spectrogram corresponding to audio
+ mel = mel_spectrogram(
+ audio,
+ self.n_fft,
+ self.num_mels,
+ self.sampling_rate,
+ self.hop_size,
+ self.win_size,
+ self.fmin,
+ self.fmax,
+ center=False,
+ ) # [B(1), self.num_mels, self.segment_size // self.hop_size]
+
+ # Fine-tuning logic that uses pre-computed mel. Example: Using TTS model-generated mel as input
+ else:
+ # For fine-tuning, assert that the waveform is in the defined sampling_rate
+ # Fine-tuning won't support on-the-fly resampling to be fool-proof (the dataset should have been prepared properly)
+ assert source_sampling_rate == self.sampling_rate, (
+ f"For fine_tuning, waveform must be in the spcified sampling rate {self.sampling_rate}, got {source_sampling_rate}"
+ )
+
+ # Cast ndarray to torch tensor
+ audio = torch.FloatTensor(audio)
+ audio = audio.unsqueeze(0) # [B(1), T_time]
+
+ # Load pre-computed mel from disk
+ mel = np.load(
+ os.path.join(
+ self.base_mels_path,
+ os.path.splitext(os.path.split(filename)[-1])[0] + ".npy",
+ )
+ )
+ mel = torch.from_numpy(mel)
+
+ if len(mel.shape) < 3:
+ mel = mel.unsqueeze(0) # ensure [B, C, T]
+
+ if self.split:
+ frames_per_seg = math.ceil(self.segment_size / self.hop_size)
+
+ if audio.size(1) >= self.segment_size:
+ mel_start = random.randint(0, mel.size(2) - frames_per_seg - 1)
+ mel = mel[:, :, mel_start : mel_start + frames_per_seg]
+ audio = audio[
+ :,
+ mel_start * self.hop_size : (mel_start + frames_per_seg) * self.hop_size,
+ ]
+
+ # Pad pre-computed mel and audio to match length to ensuring fine-tuning without error.
+ # NOTE: this may introduce a single-frame misalignment of the
+ # To remove possible misalignment, it is recommended to prepare the pair where the audio length is the integer multiple of self.hop_size
+ mel = torch.nn.functional.pad(mel, (0, frames_per_seg - mel.size(2)), "constant")
+ audio = torch.nn.functional.pad(audio, (0, self.segment_size - audio.size(1)), "constant")
+
+ # Compute mel_loss used by spectral regression objective. Uses self.fmax_loss instead (usually None)
+ mel_loss = mel_spectrogram(
+ audio,
+ self.n_fft,
+ self.num_mels,
+ self.sampling_rate,
+ self.hop_size,
+ self.win_size,
+ self.fmin,
+ self.fmax_loss,
+ center=False,
+ ) # [B(1), self.num_mels, self.segment_size // self.hop_size]
+
+ # Shape sanity checks
+ assert (
+ audio.shape[1] == mel.shape[2] * self.hop_size and audio.shape[1] == mel_loss.shape[2] * self.hop_size
+ ), (
+ f"Audio length must be mel frame length * hop_size. Got audio shape {audio.shape} mel shape {mel.shape} mel_loss shape {mel_loss.shape}"
+ )
+
+ return (mel.squeeze(), audio.squeeze(0), filename, mel_loss.squeeze())
+
+ # If it encounters error during loading the data, skip this sample and load random other sample to the batch
+ except Exception as e:
+ if self.fine_tuning:
+ raise e # Terminate training if it is fine-tuning. The dataset should have been prepared properly.
+ else:
+ print(f"[WARNING] Failed to load waveform, skipping! filename: {filename} Error: {e}")
+ return self[random.randrange(len(self))]
+
+ def __len__(self):
+ return len(self.audio_files)
diff --git a/GPT_SoVITS/BigVGAN/nv-modelcard++/.gitkeep b/GPT_SoVITS/BigVGAN/nv-modelcard++/.gitkeep
new file mode 100644
index 0000000000000000000000000000000000000000..8b137891791fe96927ad78e64b0aad7bded08bdc
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/nv-modelcard++/.gitkeep
@@ -0,0 +1 @@
+
diff --git a/GPT_SoVITS/BigVGAN/nv-modelcard++/bias.md b/GPT_SoVITS/BigVGAN/nv-modelcard++/bias.md
new file mode 100644
index 0000000000000000000000000000000000000000..4b388c28d09b8ca3aab5096304c52e1a5dac0e16
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/nv-modelcard++/bias.md
@@ -0,0 +1,4 @@
+| Field | Response |
+| :--------------------------------------------------------------------------------------------------------- | :--------------------------------------------------- |
+| Participation considerations from adversely impacted groups protected classes in model design and testing: | None |
+| Measures taken to mitigate against unwanted bias: | No measures taken to mitigate against unwanted bias. |
diff --git a/GPT_SoVITS/BigVGAN/nv-modelcard++/explainability.md b/GPT_SoVITS/BigVGAN/nv-modelcard++/explainability.md
new file mode 100644
index 0000000000000000000000000000000000000000..6f1a16676e438ba95f9d411a19e04a0f13409e54
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/nv-modelcard++/explainability.md
@@ -0,0 +1,13 @@
+| Field | Response |
+| :---------------------------------------------------------------------------------------------------- | :--------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
+| Intended Application & Domain: | Generating waveform from mel spectrogram. |
+| Model Type: | Convolutional Neural Network (CNN) |
+| Intended Users: | This model is intended for developers to synthesize and generate waveforms from the AI-generated mel spectrograms. |
+| Output: | Audio Waveform |
+| Describe how the model works: | Model generates audio waveform corresponding to the input mel spectrogram. |
+| Name the adversely impacted groups this has been tested to deliver comparable outcomes regardless of: | Not Applicable |
+| Technical Limitations: | This may not perform well on synthetically-generated mel spectrograms that deviate significantly from the profile of mel spectrograms on which this was trained. |
+| Verified to have met prescribed NVIDIA quality standards: | Yes |
+| Performance Metrics: | Perceptual Evaluation of Speech Quality (PESQ), Virtual Speech Quality Objective Listener (VISQOL), Multi-resolution STFT (MRSTFT), Mel cepstral distortion (MCD), Periodicity RMSE, Voice/Unvoiced F1 Score (V/UV F1) |
+| Potential Known Risks: | This model may generate low-quality or distorted soundwaves. |
+| Licensing: | https://github.com/NVIDIA/BigVGAN/blob/main/LICENSE |
diff --git a/GPT_SoVITS/BigVGAN/nv-modelcard++/overview.md b/GPT_SoVITS/BigVGAN/nv-modelcard++/overview.md
new file mode 100644
index 0000000000000000000000000000000000000000..a39cba0b49a4a32a37afa90f2baf4630dcd9cadc
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/nv-modelcard++/overview.md
@@ -0,0 +1,126 @@
+# Model Overview
+
+## Description:
+
+BigVGAN is a generative AI model specialized in synthesizing audio waveforms using Mel spectrogram as inputs.
+
+
+
+BigVGAN is a fully convolutional architecture with several upsampling blocks using transposed convolution followed by multiple residual dilated convolution layers.
+
+BigVGAN consists of a novel module, called anti-aliased multi-periodicity composition (AMP), which is specifically designed for generating waveforms. AMP is specialized in synthesizing high-frequency and periodic soundwaves drawing inspiration from audio signal processing principles.
+
+It applies a periodic activation function, called Snake, which provides an inductive bias to the architecture in generating periodic soundwaves. It also applies anti-aliasing filters to reduce undesired artifacts in the generated waveforms.
+
+This model is ready for commercial use.
+
+## References(s):
+
+- [BigVGAN: A Universal Neural Vocoder with Large-Scale Training](https://arxiv.org/abs/2206.04658)
+- [Project Page](https://research.nvidia.com/labs/adlr/projects/bigvgan/)
+- [Audio Demo](https://bigvgan-demo.github.io/)
+
+## Model Architecture:
+
+**Architecture Type:** Convolution Neural Network (CNN)
+**Network Architecture:** You can see the details of this model on this link: https://github.com/NVIDIA/BigVGAN and the related paper can be found here: https://arxiv.org/abs/2206.04658
+**Model Version:** 2.0
+
+## Input:
+
+**Input Type:** Audio
+**Input Format:** Mel Spectrogram
+**Input Parameters:** None
+**Other Properties Related to Input:** The input mel spectrogram has shape `[batch, channels, frames]`, where `channels` refers to the number of mel bands defined by the model and `frames` refers to the temporal length. The model supports arbitrary long `frames` that fits into the GPU memory.
+
+## Output:
+
+**Input Type:** Audio
+**Output Format:** Audio Waveform
+**Output Parameters:** None
+**Other Properties Related to Output:** The output audio waveform has shape `[batch, 1, time]`, where `1` refers to the mono audio channels and `time` refers to the temporal length. `time` is defined as a fixed integer multiple of input `frames`, which is an upsampling ratio of the model (`time = upsampling ratio * frames`). The output audio waveform consitutes float values with a range of `[-1, 1]`.
+
+## Software Integration:
+
+**Runtime Engine(s):** PyTorch
+
+**Supported Hardware Microarchitecture Compatibility:** NVIDIA Ampere, NVIDIA Hopper, NVIDIA Lovelace, NVIDIA Turing, NVIDIA Volta
+
+## Preferred/Supported Operating System(s):
+
+Linux
+
+## Model Version(s):
+
+v2.0
+
+## Training, Testing, and Evaluation Datasets:
+
+### Training Dataset:
+
+The dataset contains diverse audio types, including speech in multiple languages, environmental sounds, and instruments.
+
+**Links:**
+
+- [AAM: Artificial Audio Multitracks Dataset](https://zenodo.org/records/5794629)
+- [AudioCaps](https://audiocaps.github.io/)
+- [AudioSet](https://research.google.com/audioset/index.html)
+- [common-accent](https://huggingface.co/datasets/DTU54DL/common-accent)
+- [Crowd Sourced Emotional Multimodal Actors Dataset (CREMA-D)](https://ieeexplore.ieee.org/document/6849440)
+- [DCASE2017 Challenge, Task 4: Large-scale weakly supervised sound event detection for smart cars](https://dcase.community/challenge2017/task-large-scale-sound-event-detection)
+- [FSDnoisy18k](https://zenodo.org/records/2529934)
+- [Free Universal Sound Separation Dataset](https://zenodo.org/records/3694384)
+- [Greatest Hits dataset](https://andrewowens.com/vis/)
+- [GTZAN](https://ieeexplore.ieee.org/document/1021072)
+- [JL corpus](https://www.kaggle.com/datasets/tli725/jl-corpus)
+- [Medley-solos-DB: a cross-collection dataset for musical instrument recognition](https://zenodo.org/records/3464194)
+- [MUSAN: A Music, Speech, and Noise Corpus](https://www.openslr.org/17/)
+- [MusicBench](https://huggingface.co/datasets/amaai-lab/MusicBench)
+- [MusicCaps](https://www.kaggle.com/datasets/googleai/musiccaps)
+- [MusicNet](https://www.kaggle.com/datasets/imsparsh/musicnet-dataset)
+- [NSynth](https://magenta.tensorflow.org/datasets/nsynth)
+- [OnAir-Music-Dataset](https://github.com/sevagh/OnAir-Music-Dataset)
+- [Audio Piano Triads Dataset](https://zenodo.org/records/4740877)
+- [Pitch Audio Dataset (Surge synthesizer)](https://zenodo.org/records/4677097)
+- [SONYC Urban Sound Tagging (SONYC-UST): a multilabel dataset from an urban acoustic sensor network](https://zenodo.org/records/3966543)
+- [VocalSound: A Dataset for Improving Human Vocal Sounds Recognition](https://arxiv.org/abs/2205.03433)
+- [WavText5K](https://github.com/microsoft/WavText5K)
+- [CSS10: A Collection of Single Speaker Speech Datasets for 10 Languages](https://github.com/Kyubyong/css10)
+- [Hi-Fi Multi-Speaker English TTS Dataset (Hi-Fi TTS)](https://www.openslr.org/109/)
+- [IIIT-H Indic Speech Databases](http://festvox.org/databases/iiit_voices/)
+- [Libri-Light: A Benchmark for ASR with Limited or No Supervision](https://arxiv.org/abs/1912.07875)
+- [LibriTTS: A Corpus Derived from LibriSpeech for Text-to-Speech](https://www.openslr.org/60)
+- [LibriTTS-R: A Restored Multi-Speaker Text-to-Speech Corpus](https://www.openslr.org/141/)
+- [The SIWIS French Speech Synthesis Database](https://datashare.ed.ac.uk/handle/10283/2353)
+- [Crowdsourced high-quality Colombian Spanish speech data set](https://openslr.org/72/)
+- [TTS-Portuguese Corpus](https://github.com/Edresson/TTS-Portuguese-Corpus)
+- [CSTR VCTK Corpus: English Multi-speaker Corpus for CSTR Voice Cloning Toolkit](https://datashare.ed.ac.uk/handle/10283/3443)
+
+\*\* Data Collection Method by dataset
+
+- Human
+
+\*\* Labeling Method by dataset (for those with labels)
+
+- Hybrid: Automated, Human, Unknown
+
+### Evaluating Dataset:
+
+Properties: The audio generation quality of BigVGAN is evaluated using `dev` splits of the [LibriTTS dataset](https://www.openslr.org/60/) and [Hi-Fi TTS dataset](https://www.openslr.org/109/). The datasets include speech in English language with equal balance of genders.
+
+\*\* Data Collection Method by dataset
+
+- Human
+
+\*\* Labeling Method by dataset
+
+- Automated
+
+## Inference:
+
+**Engine:** PyTorch
+**Test Hardware:** NVIDIA A100 GPU
+
+## Ethical Considerations:
+
+NVIDIA believes Trustworthy AI is a shared responsibility and we have established policies and practices to enable development for a wide array of AI applications. When downloaded or used in accordance with our terms of service, developers should work with their internal model team to ensure this model meets requirements for the relevant industry and use case and addresses unforeseen product misuse. For more detailed information on ethical considerations for this model, please see the Model Card++ Explainability, Bias, Safety & Security, and Privacy Subcards. Please report security vulnerabilities or NVIDIA AI Concerns [here](https://www.nvidia.com/en-us/support/submit-security-vulnerability/).
diff --git a/GPT_SoVITS/BigVGAN/nv-modelcard++/privacy.md b/GPT_SoVITS/BigVGAN/nv-modelcard++/privacy.md
new file mode 100644
index 0000000000000000000000000000000000000000..73554a998384ca1b1050239ebd51bda46aec1878
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/nv-modelcard++/privacy.md
@@ -0,0 +1,14 @@
+| Field | Response |
+| :------------------------------------------------------------------------------------------------------------------------------------- | :--------------------------------------------- |
+| Generatable or reverse engineerable personal information? | None |
+| Protected class data used to create this model? | None |
+| Was consent obtained for any personal data used? | Not Applicable (No Personal Data) |
+| How often is dataset reviewed? | Before Release |
+| Is a mechanism in place to honor data subject right of access or deletion of personal data? | Not Applicable |
+| If personal collected for the development of the model, was it collected directly by NVIDIA? | Not Applicable |
+| If personal collected for the development of the model by NVIDIA, do you maintain or have access to disclosures made to data subjects? | Not Applicable |
+| If personal collected for the development of this AI model, was it minimized to only what was required? | Not Applicable |
+| Is data in dataset traceable? | Yes |
+| Is there provenance for all datasets used in training? | Yes |
+| Does data labeling (annotation, metadata) comply with privacy laws? | Yes |
+| Is data compliant with data subject requests for data correction or removal, if such a request was made? | No, not possible with externally-sourced data. |
diff --git a/GPT_SoVITS/BigVGAN/nv-modelcard++/safety.md b/GPT_SoVITS/BigVGAN/nv-modelcard++/safety.md
new file mode 100644
index 0000000000000000000000000000000000000000..ed30370dfedbbb49748706034a7153d54f1a668f
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/nv-modelcard++/safety.md
@@ -0,0 +1,6 @@
+| Field | Response |
+| :---------------------------------------------- | :---------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- |
+| Model Application(s): | Synethic Audio Generation |
+| Describe the life critical impact (if present). | Not Applicable |
+| Use Case Restrictions: | None |
+| Model and dataset restrictions: | The Principle of least privilege (PoLP) is applied limiting access for dataset generation and model development. Restrictions enforce dataset access during training, and dataset license constraints adhered to. |
diff --git a/GPT_SoVITS/BigVGAN/requirements.txt b/GPT_SoVITS/BigVGAN/requirements.txt
new file mode 100644
index 0000000000000000000000000000000000000000..6e61d3203966612e6ad193bbabdef10b1d3fed84
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/requirements.txt
@@ -0,0 +1,13 @@
+torch
+numpy
+librosa>=0.8.1
+scipy
+tensorboard
+soundfile
+matplotlib
+pesq
+auraloss
+tqdm
+nnAudio
+ninja
+huggingface_hub>=0.23.4
\ No newline at end of file
diff --git a/GPT_SoVITS/BigVGAN/train.py b/GPT_SoVITS/BigVGAN/train.py
new file mode 100644
index 0000000000000000000000000000000000000000..39718cdb33d2e9a88ec9b98dd2032bdce83a4231
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/train.py
@@ -0,0 +1,716 @@
+# Copyright (c) 2024 NVIDIA CORPORATION.
+# Licensed under the MIT license.
+
+# Adapted from https://github.com/jik876/hifi-gan under the MIT license.
+# LICENSE is in incl_licenses directory.
+
+
+import warnings
+
+warnings.simplefilter(action="ignore", category=FutureWarning)
+import itertools
+import os
+import time
+import argparse
+import json
+import torch
+import torch.nn.functional as F
+from torch.utils.tensorboard import SummaryWriter
+from torch.utils.data import DistributedSampler, DataLoader
+import torch.multiprocessing as mp
+from torch.distributed import init_process_group
+from torch.nn.parallel import DistributedDataParallel
+from env import AttrDict, build_env
+from meldataset import MelDataset, mel_spectrogram, get_dataset_filelist, MAX_WAV_VALUE
+
+from bigvgan import BigVGAN
+from discriminators import (
+ MultiPeriodDiscriminator,
+ MultiResolutionDiscriminator,
+ MultiBandDiscriminator,
+ MultiScaleSubbandCQTDiscriminator,
+)
+from loss import (
+ feature_loss,
+ generator_loss,
+ discriminator_loss,
+ MultiScaleMelSpectrogramLoss,
+)
+
+from utils import (
+ plot_spectrogram,
+ plot_spectrogram_clipped,
+ scan_checkpoint,
+ load_checkpoint,
+ save_checkpoint,
+ save_audio,
+)
+import torchaudio as ta
+from pesq import pesq
+from tqdm import tqdm
+import auraloss
+
+torch.backends.cudnn.benchmark = False
+
+
+def train(rank, a, h):
+ if h.num_gpus > 1:
+ # initialize distributed
+ init_process_group(
+ backend=h.dist_config["dist_backend"],
+ init_method=h.dist_config["dist_url"],
+ world_size=h.dist_config["world_size"] * h.num_gpus,
+ rank=rank,
+ )
+
+ # Set seed and device
+ torch.cuda.manual_seed(h.seed)
+ torch.cuda.set_device(rank)
+ device = torch.device(f"cuda:{rank:d}")
+
+ # Define BigVGAN generator
+ generator = BigVGAN(h).to(device)
+
+ # Define discriminators. MPD is used by default
+ mpd = MultiPeriodDiscriminator(h).to(device)
+
+ # Define additional discriminators. BigVGAN-v1 uses UnivNet's MRD as default
+ # New in BigVGAN-v2: option to switch to new discriminators: MultiBandDiscriminator / MultiScaleSubbandCQTDiscriminator
+ if h.get("use_mbd_instead_of_mrd", False): # Switch to MBD
+ print("[INFO] using MultiBandDiscriminator of BigVGAN-v2 instead of MultiResolutionDiscriminator")
+ # Variable name is kept as "mrd" for backward compatibility & minimal code change
+ mrd = MultiBandDiscriminator(h).to(device)
+ elif h.get("use_cqtd_instead_of_mrd", False): # Switch to CQTD
+ print("[INFO] using MultiScaleSubbandCQTDiscriminator of BigVGAN-v2 instead of MultiResolutionDiscriminator")
+ mrd = MultiScaleSubbandCQTDiscriminator(h).to(device)
+ else: # Fallback to original MRD in BigVGAN-v1
+ mrd = MultiResolutionDiscriminator(h).to(device)
+
+ # New in BigVGAN-v2: option to switch to multi-scale L1 mel loss
+ if h.get("use_multiscale_melloss", False):
+ print("[INFO] using multi-scale Mel l1 loss of BigVGAN-v2 instead of the original single-scale loss")
+ fn_mel_loss_multiscale = MultiScaleMelSpectrogramLoss(
+ sampling_rate=h.sampling_rate
+ ) # NOTE: accepts waveform as input
+ else:
+ fn_mel_loss_singlescale = F.l1_loss
+
+ # Print the model & number of parameters, and create or scan the latest checkpoint from checkpoints directory
+ if rank == 0:
+ print(generator)
+ print(mpd)
+ print(mrd)
+ print(f"Generator params: {sum(p.numel() for p in generator.parameters())}")
+ print(f"Discriminator mpd params: {sum(p.numel() for p in mpd.parameters())}")
+ print(f"Discriminator mrd params: {sum(p.numel() for p in mrd.parameters())}")
+ os.makedirs(a.checkpoint_path, exist_ok=True)
+ print(f"Checkpoints directory: {a.checkpoint_path}")
+
+ if os.path.isdir(a.checkpoint_path):
+ # New in v2.1: If the step prefix pattern-based checkpoints are not found, also check for renamed files in Hugging Face Hub to resume training
+ cp_g = scan_checkpoint(a.checkpoint_path, prefix="g_", renamed_file="bigvgan_generator.pt")
+ cp_do = scan_checkpoint(
+ a.checkpoint_path,
+ prefix="do_",
+ renamed_file="bigvgan_discriminator_optimizer.pt",
+ )
+
+ # Load the latest checkpoint if exists
+ steps = 0
+ if cp_g is None or cp_do is None:
+ state_dict_do = None
+ last_epoch = -1
+ else:
+ state_dict_g = load_checkpoint(cp_g, device)
+ state_dict_do = load_checkpoint(cp_do, device)
+ generator.load_state_dict(state_dict_g["generator"])
+ mpd.load_state_dict(state_dict_do["mpd"])
+ mrd.load_state_dict(state_dict_do["mrd"])
+ steps = state_dict_do["steps"] + 1
+ last_epoch = state_dict_do["epoch"]
+
+ # Initialize DDP, optimizers, and schedulers
+ if h.num_gpus > 1:
+ generator = DistributedDataParallel(generator, device_ids=[rank]).to(device)
+ mpd = DistributedDataParallel(mpd, device_ids=[rank]).to(device)
+ mrd = DistributedDataParallel(mrd, device_ids=[rank]).to(device)
+
+ optim_g = torch.optim.AdamW(generator.parameters(), h.learning_rate, betas=[h.adam_b1, h.adam_b2])
+ optim_d = torch.optim.AdamW(
+ itertools.chain(mrd.parameters(), mpd.parameters()),
+ h.learning_rate,
+ betas=[h.adam_b1, h.adam_b2],
+ )
+
+ if state_dict_do is not None:
+ optim_g.load_state_dict(state_dict_do["optim_g"])
+ optim_d.load_state_dict(state_dict_do["optim_d"])
+
+ scheduler_g = torch.optim.lr_scheduler.ExponentialLR(optim_g, gamma=h.lr_decay, last_epoch=last_epoch)
+ scheduler_d = torch.optim.lr_scheduler.ExponentialLR(optim_d, gamma=h.lr_decay, last_epoch=last_epoch)
+
+ # Define training and validation datasets
+
+ """
+ unseen_validation_filelist will contain sample filepaths outside the seen training & validation dataset
+ Example: trained on LibriTTS, validate on VCTK
+ """
+ training_filelist, validation_filelist, list_unseen_validation_filelist = get_dataset_filelist(a)
+
+ trainset = MelDataset(
+ training_filelist,
+ h,
+ h.segment_size,
+ h.n_fft,
+ h.num_mels,
+ h.hop_size,
+ h.win_size,
+ h.sampling_rate,
+ h.fmin,
+ h.fmax,
+ shuffle=False if h.num_gpus > 1 else True,
+ fmax_loss=h.fmax_for_loss,
+ device=device,
+ fine_tuning=a.fine_tuning,
+ base_mels_path=a.input_mels_dir,
+ is_seen=True,
+ )
+
+ train_sampler = DistributedSampler(trainset) if h.num_gpus > 1 else None
+
+ train_loader = DataLoader(
+ trainset,
+ num_workers=h.num_workers,
+ shuffle=False,
+ sampler=train_sampler,
+ batch_size=h.batch_size,
+ pin_memory=True,
+ drop_last=True,
+ )
+
+ if rank == 0:
+ validset = MelDataset(
+ validation_filelist,
+ h,
+ h.segment_size,
+ h.n_fft,
+ h.num_mels,
+ h.hop_size,
+ h.win_size,
+ h.sampling_rate,
+ h.fmin,
+ h.fmax,
+ False,
+ False,
+ fmax_loss=h.fmax_for_loss,
+ device=device,
+ fine_tuning=a.fine_tuning,
+ base_mels_path=a.input_mels_dir,
+ is_seen=True,
+ )
+ validation_loader = DataLoader(
+ validset,
+ num_workers=1,
+ shuffle=False,
+ sampler=None,
+ batch_size=1,
+ pin_memory=True,
+ drop_last=True,
+ )
+
+ list_unseen_validset = []
+ list_unseen_validation_loader = []
+ for i in range(len(list_unseen_validation_filelist)):
+ unseen_validset = MelDataset(
+ list_unseen_validation_filelist[i],
+ h,
+ h.segment_size,
+ h.n_fft,
+ h.num_mels,
+ h.hop_size,
+ h.win_size,
+ h.sampling_rate,
+ h.fmin,
+ h.fmax,
+ False,
+ False,
+ fmax_loss=h.fmax_for_loss,
+ device=device,
+ fine_tuning=a.fine_tuning,
+ base_mels_path=a.input_mels_dir,
+ is_seen=False,
+ )
+ unseen_validation_loader = DataLoader(
+ unseen_validset,
+ num_workers=1,
+ shuffle=False,
+ sampler=None,
+ batch_size=1,
+ pin_memory=True,
+ drop_last=True,
+ )
+ list_unseen_validset.append(unseen_validset)
+ list_unseen_validation_loader.append(unseen_validation_loader)
+
+ # Tensorboard logger
+ sw = SummaryWriter(os.path.join(a.checkpoint_path, "logs"))
+ if a.save_audio: # Also save audio to disk if --save_audio is set to True
+ os.makedirs(os.path.join(a.checkpoint_path, "samples"), exist_ok=True)
+
+ """
+ Validation loop, "mode" parameter is automatically defined as (seen or unseen)_(name of the dataset).
+ If the name of the dataset contains "nonspeech", it skips PESQ calculation to prevent errors
+ """
+
+ def validate(rank, a, h, loader, mode="seen"):
+ assert rank == 0, "validate should only run on rank=0"
+ generator.eval()
+ torch.cuda.empty_cache()
+
+ val_err_tot = 0
+ val_pesq_tot = 0
+ val_mrstft_tot = 0
+
+ # Modules for evaluation metrics
+ pesq_resampler = ta.transforms.Resample(h.sampling_rate, 16000).cuda()
+ loss_mrstft = auraloss.freq.MultiResolutionSTFTLoss(device="cuda")
+
+ if a.save_audio: # Also save audio to disk if --save_audio is set to True
+ os.makedirs(
+ os.path.join(a.checkpoint_path, "samples", f"gt_{mode}"),
+ exist_ok=True,
+ )
+ os.makedirs(
+ os.path.join(a.checkpoint_path, "samples", f"{mode}_{steps:08d}"),
+ exist_ok=True,
+ )
+
+ with torch.no_grad():
+ print(f"step {steps} {mode} speaker validation...")
+
+ # Loop over validation set and compute metrics
+ for j, batch in enumerate(tqdm(loader)):
+ x, y, _, y_mel = batch
+ y = y.to(device)
+ if hasattr(generator, "module"):
+ y_g_hat = generator.module(x.to(device))
+ else:
+ y_g_hat = generator(x.to(device))
+ y_mel = y_mel.to(device, non_blocking=True)
+ y_g_hat_mel = mel_spectrogram(
+ y_g_hat.squeeze(1),
+ h.n_fft,
+ h.num_mels,
+ h.sampling_rate,
+ h.hop_size,
+ h.win_size,
+ h.fmin,
+ h.fmax_for_loss,
+ )
+ min_t = min(y_mel.size(-1), y_g_hat_mel.size(-1))
+ val_err_tot += F.l1_loss(y_mel[..., :min_t], y_g_hat_mel[..., :min_t]).item()
+
+ # PESQ calculation. only evaluate PESQ if it's speech signal (nonspeech PESQ will error out)
+ if "nonspeech" not in mode: # Skips if the name of dataset (in mode string) contains "nonspeech"
+ # Resample to 16000 for pesq
+ y_16k = pesq_resampler(y)
+ y_g_hat_16k = pesq_resampler(y_g_hat.squeeze(1))
+ y_int_16k = (y_16k[0] * MAX_WAV_VALUE).short().cpu().numpy()
+ y_g_hat_int_16k = (y_g_hat_16k[0] * MAX_WAV_VALUE).short().cpu().numpy()
+ val_pesq_tot += pesq(16000, y_int_16k, y_g_hat_int_16k, "wb")
+
+ # MRSTFT calculation
+ min_t = min(y.size(-1), y_g_hat.size(-1))
+ val_mrstft_tot += loss_mrstft(y_g_hat[..., :min_t], y[..., :min_t]).item()
+
+ # Log audio and figures to Tensorboard
+ if j % a.eval_subsample == 0: # Subsample every nth from validation set
+ if steps >= 0:
+ sw.add_audio(f"gt_{mode}/y_{j}", y[0], steps, h.sampling_rate)
+ if a.save_audio: # Also save audio to disk if --save_audio is set to True
+ save_audio(
+ y[0],
+ os.path.join(
+ a.checkpoint_path,
+ "samples",
+ f"gt_{mode}",
+ f"{j:04d}.wav",
+ ),
+ h.sampling_rate,
+ )
+ sw.add_figure(
+ f"gt_{mode}/y_spec_{j}",
+ plot_spectrogram(x[0]),
+ steps,
+ )
+
+ sw.add_audio(
+ f"generated_{mode}/y_hat_{j}",
+ y_g_hat[0],
+ steps,
+ h.sampling_rate,
+ )
+ if a.save_audio: # Also save audio to disk if --save_audio is set to True
+ save_audio(
+ y_g_hat[0, 0],
+ os.path.join(
+ a.checkpoint_path,
+ "samples",
+ f"{mode}_{steps:08d}",
+ f"{j:04d}.wav",
+ ),
+ h.sampling_rate,
+ )
+ # Spectrogram of synthesized audio
+ y_hat_spec = mel_spectrogram(
+ y_g_hat.squeeze(1),
+ h.n_fft,
+ h.num_mels,
+ h.sampling_rate,
+ h.hop_size,
+ h.win_size,
+ h.fmin,
+ h.fmax,
+ )
+ sw.add_figure(
+ f"generated_{mode}/y_hat_spec_{j}",
+ plot_spectrogram(y_hat_spec.squeeze(0).cpu().numpy()),
+ steps,
+ )
+
+ """
+ Visualization of spectrogram difference between GT and synthesized audio, difference higher than 1 is clipped for better visualization.
+ """
+ spec_delta = torch.clamp(
+ torch.abs(x[0] - y_hat_spec.squeeze(0).cpu()),
+ min=1e-6,
+ max=1.0,
+ )
+ sw.add_figure(
+ f"delta_dclip1_{mode}/spec_{j}",
+ plot_spectrogram_clipped(spec_delta.numpy(), clip_max=1.0),
+ steps,
+ )
+
+ val_err = val_err_tot / (j + 1)
+ val_pesq = val_pesq_tot / (j + 1)
+ val_mrstft = val_mrstft_tot / (j + 1)
+ # Log evaluation metrics to Tensorboard
+ sw.add_scalar(f"validation_{mode}/mel_spec_error", val_err, steps)
+ sw.add_scalar(f"validation_{mode}/pesq", val_pesq, steps)
+ sw.add_scalar(f"validation_{mode}/mrstft", val_mrstft, steps)
+
+ generator.train()
+
+ # If the checkpoint is loaded, start with validation loop
+ if steps != 0 and rank == 0 and not a.debug:
+ if not a.skip_seen:
+ validate(
+ rank,
+ a,
+ h,
+ validation_loader,
+ mode=f"seen_{train_loader.dataset.name}",
+ )
+ for i in range(len(list_unseen_validation_loader)):
+ validate(
+ rank,
+ a,
+ h,
+ list_unseen_validation_loader[i],
+ mode=f"unseen_{list_unseen_validation_loader[i].dataset.name}",
+ )
+ # Exit the script if --evaluate is set to True
+ if a.evaluate:
+ exit()
+
+ # Main training loop
+ generator.train()
+ mpd.train()
+ mrd.train()
+ for epoch in range(max(0, last_epoch), a.training_epochs):
+ if rank == 0:
+ start = time.time()
+ print(f"Epoch: {epoch + 1}")
+
+ if h.num_gpus > 1:
+ train_sampler.set_epoch(epoch)
+
+ for i, batch in enumerate(train_loader):
+ if rank == 0:
+ start_b = time.time()
+ x, y, _, y_mel = batch
+
+ x = x.to(device, non_blocking=True)
+ y = y.to(device, non_blocking=True)
+ y_mel = y_mel.to(device, non_blocking=True)
+ y = y.unsqueeze(1)
+
+ y_g_hat = generator(x)
+ y_g_hat_mel = mel_spectrogram(
+ y_g_hat.squeeze(1),
+ h.n_fft,
+ h.num_mels,
+ h.sampling_rate,
+ h.hop_size,
+ h.win_size,
+ h.fmin,
+ h.fmax_for_loss,
+ )
+
+ optim_d.zero_grad()
+
+ # MPD
+ y_df_hat_r, y_df_hat_g, _, _ = mpd(y, y_g_hat.detach())
+ loss_disc_f, losses_disc_f_r, losses_disc_f_g = discriminator_loss(y_df_hat_r, y_df_hat_g)
+
+ # MRD
+ y_ds_hat_r, y_ds_hat_g, _, _ = mrd(y, y_g_hat.detach())
+ loss_disc_s, losses_disc_s_r, losses_disc_s_g = discriminator_loss(y_ds_hat_r, y_ds_hat_g)
+
+ loss_disc_all = loss_disc_s + loss_disc_f
+
+ # Set clip_grad_norm value
+ clip_grad_norm = h.get("clip_grad_norm", 1000.0) # Default to 1000
+
+ # Whether to freeze D for initial training steps
+ if steps >= a.freeze_step:
+ loss_disc_all.backward()
+ grad_norm_mpd = torch.nn.utils.clip_grad_norm_(mpd.parameters(), clip_grad_norm)
+ grad_norm_mrd = torch.nn.utils.clip_grad_norm_(mrd.parameters(), clip_grad_norm)
+ optim_d.step()
+ else:
+ print(f"[WARNING] skipping D training for the first {a.freeze_step} steps")
+ grad_norm_mpd = 0.0
+ grad_norm_mrd = 0.0
+
+ # Generator
+ optim_g.zero_grad()
+
+ # L1 Mel-Spectrogram Loss
+ lambda_melloss = h.get("lambda_melloss", 45.0) # Defaults to 45 in BigVGAN-v1 if not set
+ if h.get("use_multiscale_melloss", False): # uses wav for loss
+ loss_mel = fn_mel_loss_multiscale(y, y_g_hat) * lambda_melloss
+ else: # Uses mel for loss
+ loss_mel = fn_mel_loss_singlescale(y_mel, y_g_hat_mel) * lambda_melloss
+
+ # MPD loss
+ y_df_hat_r, y_df_hat_g, fmap_f_r, fmap_f_g = mpd(y, y_g_hat)
+ loss_fm_f = feature_loss(fmap_f_r, fmap_f_g)
+ loss_gen_f, losses_gen_f = generator_loss(y_df_hat_g)
+
+ # MRD loss
+ y_ds_hat_r, y_ds_hat_g, fmap_s_r, fmap_s_g = mrd(y, y_g_hat)
+ loss_fm_s = feature_loss(fmap_s_r, fmap_s_g)
+ loss_gen_s, losses_gen_s = generator_loss(y_ds_hat_g)
+
+ if steps >= a.freeze_step:
+ loss_gen_all = loss_gen_s + loss_gen_f + loss_fm_s + loss_fm_f + loss_mel
+ else:
+ print(f"[WARNING] using regression loss only for G for the first {a.freeze_step} steps")
+ loss_gen_all = loss_mel
+
+ loss_gen_all.backward()
+ grad_norm_g = torch.nn.utils.clip_grad_norm_(generator.parameters(), clip_grad_norm)
+ optim_g.step()
+
+ if rank == 0:
+ # STDOUT logging
+ if steps % a.stdout_interval == 0:
+ mel_error = loss_mel.item() / lambda_melloss # Log training mel regression loss to stdout
+ print(
+ f"Steps: {steps:d}, "
+ f"Gen Loss Total: {loss_gen_all:4.3f}, "
+ f"Mel Error: {mel_error:4.3f}, "
+ f"s/b: {time.time() - start_b:4.3f} "
+ f"lr: {optim_g.param_groups[0]['lr']:4.7f} "
+ f"grad_norm_g: {grad_norm_g:4.3f}"
+ )
+
+ # Checkpointing
+ if steps % a.checkpoint_interval == 0 and steps != 0:
+ checkpoint_path = f"{a.checkpoint_path}/g_{steps:08d}"
+ save_checkpoint(
+ checkpoint_path,
+ {"generator": (generator.module if h.num_gpus > 1 else generator).state_dict()},
+ )
+ checkpoint_path = f"{a.checkpoint_path}/do_{steps:08d}"
+ save_checkpoint(
+ checkpoint_path,
+ {
+ "mpd": (mpd.module if h.num_gpus > 1 else mpd).state_dict(),
+ "mrd": (mrd.module if h.num_gpus > 1 else mrd).state_dict(),
+ "optim_g": optim_g.state_dict(),
+ "optim_d": optim_d.state_dict(),
+ "steps": steps,
+ "epoch": epoch,
+ },
+ )
+
+ # Tensorboard summary logging
+ if steps % a.summary_interval == 0:
+ mel_error = loss_mel.item() / lambda_melloss # Log training mel regression loss to tensorboard
+ sw.add_scalar("training/gen_loss_total", loss_gen_all.item(), steps)
+ sw.add_scalar("training/mel_spec_error", mel_error, steps)
+ sw.add_scalar("training/fm_loss_mpd", loss_fm_f.item(), steps)
+ sw.add_scalar("training/gen_loss_mpd", loss_gen_f.item(), steps)
+ sw.add_scalar("training/disc_loss_mpd", loss_disc_f.item(), steps)
+ sw.add_scalar("training/grad_norm_mpd", grad_norm_mpd, steps)
+ sw.add_scalar("training/fm_loss_mrd", loss_fm_s.item(), steps)
+ sw.add_scalar("training/gen_loss_mrd", loss_gen_s.item(), steps)
+ sw.add_scalar("training/disc_loss_mrd", loss_disc_s.item(), steps)
+ sw.add_scalar("training/grad_norm_mrd", grad_norm_mrd, steps)
+ sw.add_scalar("training/grad_norm_g", grad_norm_g, steps)
+ sw.add_scalar("training/learning_rate_d", scheduler_d.get_last_lr()[0], steps)
+ sw.add_scalar("training/learning_rate_g", scheduler_g.get_last_lr()[0], steps)
+ sw.add_scalar("training/epoch", epoch + 1, steps)
+
+ # Validation
+ if steps % a.validation_interval == 0:
+ # Plot training input x so far used
+ for i_x in range(x.shape[0]):
+ sw.add_figure(
+ f"training_input/x_{i_x}",
+ plot_spectrogram(x[i_x].cpu()),
+ steps,
+ )
+ sw.add_audio(
+ f"training_input/y_{i_x}",
+ y[i_x][0],
+ steps,
+ h.sampling_rate,
+ )
+
+ # Seen and unseen speakers validation loops
+ if not a.debug and steps != 0:
+ validate(
+ rank,
+ a,
+ h,
+ validation_loader,
+ mode=f"seen_{train_loader.dataset.name}",
+ )
+ for i in range(len(list_unseen_validation_loader)):
+ validate(
+ rank,
+ a,
+ h,
+ list_unseen_validation_loader[i],
+ mode=f"unseen_{list_unseen_validation_loader[i].dataset.name}",
+ )
+ steps += 1
+
+ # BigVGAN-v2 learning rate scheduler is changed from epoch-level to step-level
+ scheduler_g.step()
+ scheduler_d.step()
+
+ if rank == 0:
+ print(f"Time taken for epoch {epoch + 1} is {int(time.time() - start)} sec\n")
+
+
+def main():
+ print("Initializing Training Process..")
+
+ parser = argparse.ArgumentParser()
+
+ parser.add_argument("--group_name", default=None)
+
+ parser.add_argument("--input_wavs_dir", default="LibriTTS")
+ parser.add_argument("--input_mels_dir", default="ft_dataset")
+ parser.add_argument("--input_training_file", default="tests/LibriTTS/train-full.txt")
+ parser.add_argument("--input_validation_file", default="tests/LibriTTS/val-full.txt")
+
+ parser.add_argument(
+ "--list_input_unseen_wavs_dir",
+ nargs="+",
+ default=["tests/LibriTTS", "tests/LibriTTS"],
+ )
+ parser.add_argument(
+ "--list_input_unseen_validation_file",
+ nargs="+",
+ default=["tests/LibriTTS/dev-clean.txt", "tests/LibriTTS/dev-other.txt"],
+ )
+
+ parser.add_argument("--checkpoint_path", default="exp/bigvgan")
+ parser.add_argument("--config", default="")
+
+ parser.add_argument("--training_epochs", default=100000, type=int)
+ parser.add_argument("--stdout_interval", default=5, type=int)
+ parser.add_argument("--checkpoint_interval", default=50000, type=int)
+ parser.add_argument("--summary_interval", default=100, type=int)
+ parser.add_argument("--validation_interval", default=50000, type=int)
+
+ parser.add_argument(
+ "--freeze_step",
+ default=0,
+ type=int,
+ help="freeze D for the first specified steps. G only uses regression loss for these steps.",
+ )
+
+ parser.add_argument("--fine_tuning", default=False, type=bool)
+
+ parser.add_argument(
+ "--debug",
+ default=False,
+ type=bool,
+ help="debug mode. skips validation loop throughout training",
+ )
+ parser.add_argument(
+ "--evaluate",
+ default=False,
+ type=bool,
+ help="only run evaluation from checkpoint and exit",
+ )
+ parser.add_argument(
+ "--eval_subsample",
+ default=5,
+ type=int,
+ help="subsampling during evaluation loop",
+ )
+ parser.add_argument(
+ "--skip_seen",
+ default=False,
+ type=bool,
+ help="skip seen dataset. useful for test set inference",
+ )
+ parser.add_argument(
+ "--save_audio",
+ default=False,
+ type=bool,
+ help="save audio of test set inference to disk",
+ )
+
+ a = parser.parse_args()
+
+ with open(a.config) as f:
+ data = f.read()
+
+ json_config = json.loads(data)
+ h = AttrDict(json_config)
+
+ build_env(a.config, "config.json", a.checkpoint_path)
+
+ torch.manual_seed(h.seed)
+ if torch.cuda.is_available():
+ torch.cuda.manual_seed(h.seed)
+ h.num_gpus = torch.cuda.device_count()
+ h.batch_size = int(h.batch_size / h.num_gpus)
+ print(f"Batch size per GPU: {h.batch_size}")
+ else:
+ pass
+
+ if h.num_gpus > 1:
+ mp.spawn(
+ train,
+ nprocs=h.num_gpus,
+ args=(
+ a,
+ h,
+ ),
+ )
+ else:
+ train(0, a, h)
+
+
+if __name__ == "__main__":
+ main()
diff --git a/GPT_SoVITS/BigVGAN/utils0.py b/GPT_SoVITS/BigVGAN/utils0.py
new file mode 100644
index 0000000000000000000000000000000000000000..da98a24cf1447778305563f8e909f30b06e06b26
--- /dev/null
+++ b/GPT_SoVITS/BigVGAN/utils0.py
@@ -0,0 +1,99 @@
+# Adapted from https://github.com/jik876/hifi-gan under the MIT license.
+# LICENSE is in incl_licenses directory.
+
+import glob
+import os
+import matplotlib
+import torch
+from torch.nn.utils import weight_norm
+
+matplotlib.use("Agg")
+import matplotlib.pylab as plt
+from .meldataset import MAX_WAV_VALUE
+from scipy.io.wavfile import write
+
+
+def plot_spectrogram(spectrogram):
+ fig, ax = plt.subplots(figsize=(10, 2))
+ im = ax.imshow(spectrogram, aspect="auto", origin="lower", interpolation="none")
+ plt.colorbar(im, ax=ax)
+
+ fig.canvas.draw()
+ plt.close()
+
+ return fig
+
+
+def plot_spectrogram_clipped(spectrogram, clip_max=2.0):
+ fig, ax = plt.subplots(figsize=(10, 2))
+ im = ax.imshow(
+ spectrogram,
+ aspect="auto",
+ origin="lower",
+ interpolation="none",
+ vmin=1e-6,
+ vmax=clip_max,
+ )
+ plt.colorbar(im, ax=ax)
+
+ fig.canvas.draw()
+ plt.close()
+
+ return fig
+
+
+def init_weights(m, mean=0.0, std=0.01):
+ classname = m.__class__.__name__
+ if classname.find("Conv") != -1:
+ m.weight.data.normal_(mean, std)
+
+
+def apply_weight_norm(m):
+ classname = m.__class__.__name__
+ if classname.find("Conv") != -1:
+ weight_norm(m)
+
+
+def get_padding(kernel_size, dilation=1):
+ return int((kernel_size * dilation - dilation) / 2)
+
+
+def load_checkpoint(filepath, device):
+ assert os.path.isfile(filepath)
+ print(f"Loading '{filepath}'")
+ checkpoint_dict = torch.load(filepath, map_location=device)
+ print("Complete.")
+ return checkpoint_dict
+
+
+def save_checkpoint(filepath, obj):
+ print(f"Saving checkpoint to {filepath}")
+ torch.save(obj, filepath)
+ print("Complete.")
+
+
+def scan_checkpoint(cp_dir, prefix, renamed_file=None):
+ # Fallback to original scanning logic first
+ pattern = os.path.join(cp_dir, prefix + "????????")
+ cp_list = glob.glob(pattern)
+
+ if len(cp_list) > 0:
+ last_checkpoint_path = sorted(cp_list)[-1]
+ print(f"[INFO] Resuming from checkpoint: '{last_checkpoint_path}'")
+ return last_checkpoint_path
+
+ # If no pattern-based checkpoints are found, check for renamed file
+ if renamed_file:
+ renamed_path = os.path.join(cp_dir, renamed_file)
+ if os.path.isfile(renamed_path):
+ print(f"[INFO] Resuming from renamed checkpoint: '{renamed_file}'")
+ return renamed_path
+
+ return None
+
+
+def save_audio(audio, path, sr):
+ # wav: torch with 1d shape
+ audio = audio * MAX_WAV_VALUE
+ audio = audio.cpu().numpy().astype("int16")
+ write(path, sr, audio)
diff --git a/GPT_SoVITS/download.py b/GPT_SoVITS/download.py
new file mode 100644
index 0000000000000000000000000000000000000000..fc4ead63bfe3c15326212a6ebabe2dac166e0ff2
--- /dev/null
+++ b/GPT_SoVITS/download.py
@@ -0,0 +1,13 @@
+import os
+import sys
+
+now_dir = os.getcwd()
+sys.path.insert(0, now_dir)
+from text.g2pw import G2PWPinyin
+
+g2pw = G2PWPinyin(
+ model_dir="GPT_SoVITS/text/G2PWModel",
+ model_source="GPT_SoVITS/pretrained_models/chinese-roberta-wwm-ext-large",
+ v_to_u=False,
+ neutral_tone_with_five=True,
+)
diff --git a/GPT_SoVITS/export_torch_script.py b/GPT_SoVITS/export_torch_script.py
new file mode 100644
index 0000000000000000000000000000000000000000..69817a3763b140430ad68907be0766823c170056
--- /dev/null
+++ b/GPT_SoVITS/export_torch_script.py
@@ -0,0 +1,861 @@
+# modified from https://github.com/yangdongchao/SoundStorm/blob/master/soundstorm/s1/AR/models/t2s_model.py
+# reference: https://github.com/lifeiteng/vall-e
+import argparse
+from typing import Optional
+from my_utils import load_audio
+import torch
+import torchaudio
+
+from torch import IntTensor, LongTensor, Tensor, nn
+from torch.nn import functional as F
+
+from transformers import AutoModelForMaskedLM, AutoTokenizer
+from feature_extractor import cnhubert
+
+from AR.models.t2s_lightning_module import Text2SemanticLightningModule
+from module.models_onnx import SynthesizerTrn
+
+from inference_webui import get_phones_and_bert
+
+import os
+import soundfile
+
+default_config = {
+ "embedding_dim": 512,
+ "hidden_dim": 512,
+ "num_head": 8,
+ "num_layers": 12,
+ "num_codebook": 8,
+ "p_dropout": 0.0,
+ "vocab_size": 1024 + 1,
+ "phoneme_vocab_size": 512,
+ "EOS": 1024,
+}
+
+
+def get_raw_t2s_model(dict_s1) -> Text2SemanticLightningModule:
+ config = dict_s1["config"]
+ config["model"]["dropout"] = float(config["model"]["dropout"])
+ t2s_model = Text2SemanticLightningModule(config, "****", is_train=False)
+ t2s_model.load_state_dict(dict_s1["weight"])
+ t2s_model = t2s_model.eval()
+ return t2s_model
+
+
+@torch.jit.script
+def logits_to_probs(
+ logits,
+ previous_tokens: Optional[torch.Tensor] = None,
+ temperature: float = 1.0,
+ top_k: Optional[int] = None,
+ top_p: Optional[int] = None,
+ repetition_penalty: float = 1.0,
+):
+ # if previous_tokens is not None:
+ # previous_tokens = previous_tokens.squeeze()
+ # print(logits.shape,previous_tokens.shape)
+ # pdb.set_trace()
+ if previous_tokens is not None and repetition_penalty != 1.0:
+ previous_tokens = previous_tokens.long()
+ score = torch.gather(logits, dim=1, index=previous_tokens)
+ score = torch.where(score < 0, score * repetition_penalty, score / repetition_penalty)
+ logits.scatter_(dim=1, index=previous_tokens, src=score)
+
+ if top_p is not None and top_p < 1.0:
+ sorted_logits, sorted_indices = torch.sort(logits, descending=True)
+ cum_probs = torch.cumsum(torch.nn.functional.softmax(sorted_logits, dim=-1), dim=-1)
+ sorted_indices_to_remove = cum_probs > top_p
+ sorted_indices_to_remove[:, 0] = False # keep at least one option
+ indices_to_remove = sorted_indices_to_remove.scatter(dim=1, index=sorted_indices, src=sorted_indices_to_remove)
+ logits = logits.masked_fill(indices_to_remove, -float("Inf"))
+
+ logits = logits / max(temperature, 1e-5)
+
+ if top_k is not None:
+ v, _ = torch.topk(logits, min(top_k, logits.size(-1)))
+ pivot = v[:, -1].unsqueeze(-1)
+ logits = torch.where(logits < pivot, -float("Inf"), logits)
+
+ probs = torch.nn.functional.softmax(logits, dim=-1)
+ return probs
+
+
+@torch.jit.script
+def multinomial_sample_one_no_sync(probs_sort):
+ # Does multinomial sampling without a cuda synchronization
+ q = torch.randn_like(probs_sort)
+ return torch.argmax(probs_sort / q, dim=-1, keepdim=True).to(dtype=torch.int)
+
+
+@torch.jit.script
+def sample(
+ logits,
+ previous_tokens,
+ temperature: float = 1.0,
+ top_k: Optional[int] = None,
+ top_p: Optional[int] = None,
+ repetition_penalty: float = 1.0,
+):
+ probs = logits_to_probs(
+ logits=logits,
+ previous_tokens=previous_tokens,
+ temperature=temperature,
+ top_k=top_k,
+ top_p=top_p,
+ repetition_penalty=repetition_penalty,
+ )
+ idx_next = multinomial_sample_one_no_sync(probs)
+ return idx_next, probs
+
+
+@torch.jit.script
+def spectrogram_torch(y: Tensor, n_fft: int, sampling_rate: int, hop_size: int, win_size: int, center: bool = False):
+ hann_window = torch.hann_window(win_size, device=y.device, dtype=y.dtype)
+ y = torch.nn.functional.pad(
+ y.unsqueeze(1),
+ (int((n_fft - hop_size) / 2), int((n_fft - hop_size) / 2)),
+ mode="reflect",
+ )
+ y = y.squeeze(1)
+ spec = torch.stft(
+ y,
+ n_fft,
+ hop_length=hop_size,
+ win_length=win_size,
+ window=hann_window,
+ center=center,
+ pad_mode="reflect",
+ normalized=False,
+ onesided=True,
+ return_complex=False,
+ )
+ spec = torch.sqrt(spec.pow(2).sum(-1) + 1e-6)
+ return spec
+
+
+class DictToAttrRecursive(dict):
+ def __init__(self, input_dict):
+ super().__init__(input_dict)
+ for key, value in input_dict.items():
+ if isinstance(value, dict):
+ value = DictToAttrRecursive(value)
+ self[key] = value
+ setattr(self, key, value)
+
+ def __getattr__(self, item):
+ try:
+ return self[item]
+ except KeyError:
+ raise AttributeError(f"Attribute {item} not found")
+
+ def __setattr__(self, key, value):
+ if isinstance(value, dict):
+ value = DictToAttrRecursive(value)
+ super(DictToAttrRecursive, self).__setitem__(key, value)
+ super().__setattr__(key, value)
+
+ def __delattr__(self, item):
+ try:
+ del self[item]
+ except KeyError:
+ raise AttributeError(f"Attribute {item} not found")
+
+
+@torch.jit.script
+class T2SMLP:
+ def __init__(self, w1, b1, w2, b2):
+ self.w1 = w1
+ self.b1 = b1
+ self.w2 = w2
+ self.b2 = b2
+
+ def forward(self, x):
+ x = F.relu(F.linear(x, self.w1, self.b1))
+ x = F.linear(x, self.w2, self.b2)
+ return x
+
+
+@torch.jit.script
+class T2SBlock:
+ def __init__(
+ self,
+ num_heads: int,
+ hidden_dim: int,
+ mlp: T2SMLP,
+ qkv_w,
+ qkv_b,
+ out_w,
+ out_b,
+ norm_w1,
+ norm_b1,
+ norm_eps1: float,
+ norm_w2,
+ norm_b2,
+ norm_eps2: float,
+ ):
+ self.num_heads = num_heads
+ self.mlp = mlp
+ self.hidden_dim: int = hidden_dim
+ self.qkv_w = qkv_w
+ self.qkv_b = qkv_b
+ self.out_w = out_w
+ self.out_b = out_b
+ self.norm_w1 = norm_w1
+ self.norm_b1 = norm_b1
+ self.norm_eps1 = norm_eps1
+ self.norm_w2 = norm_w2
+ self.norm_b2 = norm_b2
+ self.norm_eps2 = norm_eps2
+
+ self.false = torch.tensor(False, dtype=torch.bool)
+
+ @torch.jit.ignore
+ def to_mask(self, x: torch.Tensor, padding_mask: Optional[torch.Tensor]):
+ if padding_mask is None:
+ return x
+
+ if padding_mask.dtype == torch.bool:
+ return x.masked_fill(padding_mask, 0)
+ else:
+ return x * padding_mask
+
+ def process_prompt(self, x: torch.Tensor, attn_mask: torch.Tensor, padding_mask: Optional[torch.Tensor] = None):
+ q, k, v = F.linear(self.to_mask(x, padding_mask), self.qkv_w, self.qkv_b).chunk(3, dim=-1)
+
+ batch_size = q.shape[0]
+ q_len = q.shape[1]
+ kv_len = k.shape[1]
+
+ q = self.to_mask(q, padding_mask)
+ k_cache = self.to_mask(k, padding_mask)
+ v_cache = self.to_mask(v, padding_mask)
+
+ q = q.view(batch_size, q_len, self.num_heads, -1).transpose(1, 2)
+ k = k_cache.view(batch_size, kv_len, self.num_heads, -1).transpose(1, 2)
+ v = v_cache.view(batch_size, kv_len, self.num_heads, -1).transpose(1, 2)
+
+ attn = F.scaled_dot_product_attention(q, k, v, ~attn_mask)
+
+ attn = attn.permute(2, 0, 1, 3).reshape(batch_size * q_len, self.hidden_dim)
+ attn = attn.view(q_len, batch_size, self.hidden_dim).transpose(1, 0)
+ attn = F.linear(self.to_mask(attn, padding_mask), self.out_w, self.out_b)
+
+ if padding_mask is not None:
+ for i in range(batch_size):
+ # mask = padding_mask[i,:,0]
+ if self.false.device != padding_mask.device:
+ self.false = self.false.to(padding_mask.device)
+ idx = torch.where(padding_mask[i, :, 0] == self.false)[0]
+ x_item = x[i, idx, :].unsqueeze(0)
+ attn_item = attn[i, idx, :].unsqueeze(0)
+ x_item = x_item + attn_item
+ x_item = F.layer_norm(x_item, [self.hidden_dim], self.norm_w1, self.norm_b1, self.norm_eps1)
+ x_item = x_item + self.mlp.forward(x_item)
+ x_item = F.layer_norm(
+ x_item,
+ [self.hidden_dim],
+ self.norm_w2,
+ self.norm_b2,
+ self.norm_eps2,
+ )
+ x[i, idx, :] = x_item.squeeze(0)
+ x = self.to_mask(x, padding_mask)
+ else:
+ x = x + attn
+ x = F.layer_norm(x, [self.hidden_dim], self.norm_w1, self.norm_b1, self.norm_eps1)
+ x = x + self.mlp.forward(x)
+ x = F.layer_norm(
+ x,
+ [self.hidden_dim],
+ self.norm_w2,
+ self.norm_b2,
+ self.norm_eps2,
+ )
+ return x, k_cache, v_cache
+
+ def decode_next_token(self, x: torch.Tensor, k_cache: torch.Tensor, v_cache: torch.Tensor):
+ q, k, v = F.linear(x, self.qkv_w, self.qkv_b).chunk(3, dim=-1)
+
+ k_cache = torch.cat([k_cache, k], dim=1)
+ v_cache = torch.cat([v_cache, v], dim=1)
+
+ batch_size = q.shape[0]
+ q_len = q.shape[1]
+ kv_len = k_cache.shape[1]
+
+ q = q.view(batch_size, q_len, self.num_heads, -1).transpose(1, 2)
+ k = k_cache.view(batch_size, kv_len, self.num_heads, -1).transpose(1, 2)
+ v = v_cache.view(batch_size, kv_len, self.num_heads, -1).transpose(1, 2)
+
+ attn = F.scaled_dot_product_attention(q, k, v)
+
+ attn = attn.permute(2, 0, 1, 3).reshape(batch_size * q_len, self.hidden_dim)
+ attn = attn.view(q_len, batch_size, self.hidden_dim).transpose(1, 0)
+ attn = F.linear(attn, self.out_w, self.out_b)
+
+ x = x + attn
+ x = F.layer_norm(x, [self.hidden_dim], self.norm_w1, self.norm_b1, self.norm_eps1)
+ x = x + self.mlp.forward(x)
+ x = F.layer_norm(
+ x,
+ [self.hidden_dim],
+ self.norm_w2,
+ self.norm_b2,
+ self.norm_eps2,
+ )
+ return x, k_cache, v_cache
+
+
+@torch.jit.script
+class T2STransformer:
+ def __init__(self, num_blocks: int, blocks: list[T2SBlock]):
+ self.num_blocks: int = num_blocks
+ self.blocks = blocks
+
+ def process_prompt(self, x: torch.Tensor, attn_mask: torch.Tensor, padding_mask: Optional[torch.Tensor] = None):
+ k_cache: list[torch.Tensor] = []
+ v_cache: list[torch.Tensor] = []
+ for i in range(self.num_blocks):
+ x, k_cache_, v_cache_ = self.blocks[i].process_prompt(x, attn_mask, padding_mask)
+ k_cache.append(k_cache_)
+ v_cache.append(v_cache_)
+ return x, k_cache, v_cache
+
+ def decode_next_token(self, x: torch.Tensor, k_cache: list[torch.Tensor], v_cache: list[torch.Tensor]):
+ for i in range(self.num_blocks):
+ x, k_cache[i], v_cache[i] = self.blocks[i].decode_next_token(x, k_cache[i], v_cache[i])
+ return x, k_cache, v_cache
+
+
+class VitsModel(nn.Module):
+ def __init__(self, vits_path):
+ super().__init__()
+ # dict_s2 = torch.load(vits_path,map_location="cpu")
+ dict_s2 = torch.load(vits_path)
+ self.hps = dict_s2["config"]
+ if dict_s2["weight"]["enc_p.text_embedding.weight"].shape[0] == 322:
+ self.hps["model"]["version"] = "v1"
+ else:
+ self.hps["model"]["version"] = "v2"
+
+ self.hps = DictToAttrRecursive(self.hps)
+ self.hps.model.semantic_frame_rate = "25hz"
+ self.vq_model = SynthesizerTrn(
+ self.hps.data.filter_length // 2 + 1,
+ self.hps.train.segment_size // self.hps.data.hop_length,
+ n_speakers=self.hps.data.n_speakers,
+ **self.hps.model,
+ )
+ self.vq_model.eval()
+ self.vq_model.load_state_dict(dict_s2["weight"], strict=False)
+
+ def forward(self, text_seq, pred_semantic, ref_audio, speed=1.0):
+ refer = spectrogram_torch(
+ ref_audio,
+ self.hps.data.filter_length,
+ self.hps.data.sampling_rate,
+ self.hps.data.hop_length,
+ self.hps.data.win_length,
+ center=False,
+ )
+ return self.vq_model(pred_semantic, text_seq, refer, speed)[0, 0]
+
+
+class T2SModel(nn.Module):
+ def __init__(self, raw_t2s: Text2SemanticLightningModule):
+ super(T2SModel, self).__init__()
+ self.model_dim = raw_t2s.model.model_dim
+ self.embedding_dim = raw_t2s.model.embedding_dim
+ self.num_head = raw_t2s.model.num_head
+ self.num_layers = raw_t2s.model.num_layers
+ self.vocab_size = raw_t2s.model.vocab_size
+ self.phoneme_vocab_size = raw_t2s.model.phoneme_vocab_size
+ # self.p_dropout = float(raw_t2s.model.p_dropout)
+ self.EOS: int = int(raw_t2s.model.EOS)
+ self.norm_first = raw_t2s.model.norm_first
+ assert self.EOS == self.vocab_size - 1
+ self.hz = 50
+
+ self.bert_proj = raw_t2s.model.bert_proj
+ self.ar_text_embedding = raw_t2s.model.ar_text_embedding
+ self.ar_text_position = raw_t2s.model.ar_text_position
+ self.ar_audio_embedding = raw_t2s.model.ar_audio_embedding
+ self.ar_audio_position = raw_t2s.model.ar_audio_position
+
+ # self.t2s_transformer = T2STransformer(self.num_layers, blocks)
+ # self.t2s_transformer = raw_t2s.model.t2s_transformer
+
+ blocks = []
+ h = raw_t2s.model.h
+
+ for i in range(self.num_layers):
+ layer = h.layers[i]
+ t2smlp = T2SMLP(layer.linear1.weight, layer.linear1.bias, layer.linear2.weight, layer.linear2.bias)
+
+ block = T2SBlock(
+ self.num_head,
+ self.model_dim,
+ t2smlp,
+ layer.self_attn.in_proj_weight,
+ layer.self_attn.in_proj_bias,
+ layer.self_attn.out_proj.weight,
+ layer.self_attn.out_proj.bias,
+ layer.norm1.weight,
+ layer.norm1.bias,
+ layer.norm1.eps,
+ layer.norm2.weight,
+ layer.norm2.bias,
+ layer.norm2.eps,
+ )
+
+ blocks.append(block)
+
+ self.t2s_transformer = T2STransformer(self.num_layers, blocks)
+
+ # self.ar_predict_layer = nn.Linear(self.model_dim, self.vocab_size, bias=False)
+ self.ar_predict_layer = raw_t2s.model.ar_predict_layer
+ # self.loss_fct = nn.CrossEntropyLoss(reduction="sum")
+ self.max_sec = raw_t2s.config["data"]["max_sec"]
+ self.top_k = int(raw_t2s.config["inference"]["top_k"])
+ self.early_stop_num = torch.LongTensor([self.hz * self.max_sec])
+
+ def forward(
+ self,
+ prompts: LongTensor,
+ ref_seq: LongTensor,
+ text_seq: LongTensor,
+ ref_bert: torch.Tensor,
+ text_bert: torch.Tensor,
+ top_k: LongTensor,
+ ):
+ bert = torch.cat([ref_bert.T, text_bert.T], 1)
+ all_phoneme_ids = torch.cat([ref_seq, text_seq], 1)
+ bert = bert.unsqueeze(0)
+
+ x = self.ar_text_embedding(all_phoneme_ids)
+ x = x + self.bert_proj(bert.transpose(1, 2))
+ x: torch.Tensor = self.ar_text_position(x)
+
+ early_stop_num = self.early_stop_num
+
+ # [1,N,512] [1,N]
+ # y, k, v, y_emb, x_example = self.first_stage_decoder(x, prompts)
+ y = prompts
+ # x_example = x[:,:,0] * 0.0
+
+ x_len = x.shape[1]
+ x_attn_mask = torch.zeros((x_len, x_len), dtype=torch.bool)
+
+ y_emb = self.ar_audio_embedding(y)
+ y_len = y_emb.shape[1]
+ prefix_len = y.shape[1]
+ y_pos = self.ar_audio_position(y_emb)
+ xy_pos = torch.concat([x, y_pos], dim=1)
+
+ bsz = x.shape[0]
+ src_len = x_len + y_len
+ x_attn_mask_pad = F.pad(
+ x_attn_mask,
+ (0, y_len), ###xx的纯0扩展到xx纯0+xy纯1,(x,x+y)
+ value=True,
+ )
+ y_attn_mask = F.pad( ###yy的右上1扩展到左边xy的0,(y,x+y)
+ torch.triu(torch.ones(y_len, y_len, dtype=torch.bool), diagonal=1),
+ (x_len, 0),
+ value=False,
+ )
+ xy_attn_mask = (
+ torch.concat([x_attn_mask_pad, y_attn_mask], dim=0)
+ .unsqueeze(0)
+ .expand(bsz * self.num_head, -1, -1)
+ .view(bsz, self.num_head, src_len, src_len)
+ .to(device=x.device, dtype=torch.bool)
+ )
+
+ idx = 0
+ top_k = int(top_k)
+
+ xy_dec, k_cache, v_cache = self.t2s_transformer.process_prompt(xy_pos, xy_attn_mask, None)
+
+ logits = self.ar_predict_layer(xy_dec[:, -1])
+ logits = logits[:, :-1]
+ samples = sample(logits, y, top_k=top_k, top_p=1, repetition_penalty=1.35, temperature=1.0)[0]
+ y = torch.concat([y, samples], dim=1)
+ y_emb = self.ar_audio_embedding(y[:, -1:])
+ xy_pos = y_emb * self.ar_audio_position.x_scale + self.ar_audio_position.alpha * self.ar_audio_position.pe[
+ :, y_len + idx
+ ].to(dtype=y_emb.dtype, device=y_emb.device)
+
+ stop = False
+ # for idx in range(1, 50):
+ for idx in range(1, 1500):
+ # [1, N] [N_layer, N, 1, 512] [N_layer, N, 1, 512] [1, N, 512] [1] [1, N, 512] [1, N]
+ # y, k, v, y_emb, logits, samples = self.stage_decoder(y, k, v, y_emb, x_example)
+ xy_dec, k_cache, v_cache = self.t2s_transformer.decode_next_token(xy_pos, k_cache, v_cache)
+ logits = self.ar_predict_layer(xy_dec[:, -1])
+
+ if idx < 11: ###至少预测出10个token不然不给停止(0.4s)
+ logits = logits[:, :-1]
+
+ samples = sample(logits, y, top_k=top_k, top_p=1, repetition_penalty=1.35, temperature=1.0)[0]
+
+ y = torch.concat([y, samples], dim=1)
+
+ if early_stop_num != -1 and (y.shape[1] - prefix_len) > early_stop_num:
+ stop = True
+ if torch.argmax(logits, dim=-1)[0] == self.EOS or samples[0, 0] == self.EOS:
+ stop = True
+ if stop:
+ if y.shape[1] == 0:
+ y = torch.concat([y, torch.zeros_like(samples)], dim=1)
+ break
+
+ y_emb = self.ar_audio_embedding(y[:, -1:])
+ xy_pos = y_emb * self.ar_audio_position.x_scale + self.ar_audio_position.alpha * self.ar_audio_position.pe[
+ :, y_len + idx
+ ].to(dtype=y_emb.dtype, device=y_emb.device)
+
+ y[0, -1] = 0
+
+ return y[:, -idx:].unsqueeze(0)
+
+
+bert_path = os.environ.get("bert_path", "GPT_SoVITS/pretrained_models/chinese-roberta-wwm-ext-large")
+cnhubert_base_path = "GPT_SoVITS/pretrained_models/chinese-hubert-base"
+cnhubert.cnhubert_base_path = cnhubert_base_path
+
+
+@torch.jit.script
+def build_phone_level_feature(res: Tensor, word2ph: IntTensor):
+ phone_level_feature = []
+ for i in range(word2ph.shape[0]):
+ repeat_feature = res[i].repeat(word2ph[i].item(), 1)
+ phone_level_feature.append(repeat_feature)
+ phone_level_feature = torch.cat(phone_level_feature, dim=0)
+ # [sum(word2ph), 1024]
+ return phone_level_feature
+
+
+class MyBertModel(torch.nn.Module):
+ def __init__(self, bert_model):
+ super(MyBertModel, self).__init__()
+ self.bert = bert_model
+
+ def forward(
+ self, input_ids: torch.Tensor, attention_mask: torch.Tensor, token_type_ids: torch.Tensor, word2ph: IntTensor
+ ):
+ outputs = self.bert(input_ids=input_ids, attention_mask=attention_mask, token_type_ids=token_type_ids)
+ # res = torch.cat(outputs["hidden_states"][-3:-2], -1)[0][1:-1]
+ res = torch.cat(outputs[1][-3:-2], -1)[0][1:-1]
+ return build_phone_level_feature(res, word2ph)
+
+
+class SSLModel(torch.nn.Module):
+ def __init__(self):
+ super().__init__()
+ self.ssl = cnhubert.get_model().model
+
+ def forward(self, ref_audio_16k) -> torch.Tensor:
+ ssl_content = self.ssl(ref_audio_16k)["last_hidden_state"].transpose(1, 2)
+ return ssl_content
+
+
+class ExportSSLModel(torch.nn.Module):
+ def __init__(self, ssl: SSLModel):
+ super().__init__()
+ self.ssl = ssl
+
+ def forward(self, ref_audio: torch.Tensor):
+ return self.ssl(ref_audio)
+
+ @torch.jit.export
+ def resample(self, ref_audio: torch.Tensor, src_sr: int, dst_sr: int) -> torch.Tensor:
+ audio = resamplex(ref_audio, src_sr, dst_sr).float()
+ return audio
+
+
+def export_bert(output_path):
+ tokenizer = AutoTokenizer.from_pretrained(bert_path)
+
+ text = "叹息声一声接着一声传出,木兰对着房门织布.听不见织布机织布的声音,只听见木兰在叹息.问木兰在想什么?问木兰在惦记什么?木兰答道,我也没有在想什么,也没有在惦记什么."
+ ref_bert_inputs = tokenizer(text, return_tensors="pt")
+ word2ph = []
+ for c in text:
+ if c in [",", "。", ":", "?", ",", ".", "?"]:
+ word2ph.append(1)
+ else:
+ word2ph.append(2)
+ ref_bert_inputs["word2ph"] = torch.Tensor(word2ph).int()
+
+ bert_model = AutoModelForMaskedLM.from_pretrained(bert_path, output_hidden_states=True, torchscript=True)
+ my_bert_model = MyBertModel(bert_model)
+
+ ref_bert_inputs = {
+ "input_ids": ref_bert_inputs["input_ids"],
+ "attention_mask": ref_bert_inputs["attention_mask"],
+ "token_type_ids": ref_bert_inputs["token_type_ids"],
+ "word2ph": ref_bert_inputs["word2ph"],
+ }
+
+ torch._dynamo.mark_dynamic(ref_bert_inputs["input_ids"], 1)
+ torch._dynamo.mark_dynamic(ref_bert_inputs["attention_mask"], 1)
+ torch._dynamo.mark_dynamic(ref_bert_inputs["token_type_ids"], 1)
+ torch._dynamo.mark_dynamic(ref_bert_inputs["word2ph"], 0)
+
+ my_bert_model = torch.jit.trace(my_bert_model, example_kwarg_inputs=ref_bert_inputs)
+ output_path = os.path.join(output_path, "bert_model.pt")
+ my_bert_model.save(output_path)
+ print("#### exported bert ####")
+
+
+def export(gpt_path, vits_path, ref_audio_path, ref_text, output_path, export_bert_and_ssl=False, device="cpu"):
+ if not os.path.exists(output_path):
+ os.makedirs(output_path)
+ print(f"目录已创建: {output_path}")
+ else:
+ print(f"目录已存在: {output_path}")
+
+ ref_audio = torch.tensor([load_audio(ref_audio_path, 16000)]).float()
+ ssl = SSLModel()
+ if export_bert_and_ssl:
+ s = ExportSSLModel(torch.jit.trace(ssl, example_inputs=(ref_audio)))
+ ssl_path = os.path.join(output_path, "ssl_model.pt")
+ torch.jit.script(s).save(ssl_path)
+ print("#### exported ssl ####")
+ export_bert(output_path)
+ else:
+ s = ExportSSLModel(ssl)
+
+ print(f"device: {device}")
+
+ ref_seq_id, ref_bert_T, ref_norm_text = get_phones_and_bert(ref_text, "all_zh", "v2")
+ ref_seq = torch.LongTensor([ref_seq_id]).to(device)
+ ref_bert = ref_bert_T.T.to(ref_seq.device)
+ text_seq_id, text_bert_T, norm_text = get_phones_and_bert(
+ "这是一条测试语音,说什么无所谓,只是给它一个例子", "all_zh", "v2"
+ )
+ text_seq = torch.LongTensor([text_seq_id]).to(device)
+ text_bert = text_bert_T.T.to(text_seq.device)
+
+ ssl_content = ssl(ref_audio).to(device)
+
+ # vits_path = "SoVITS_weights_v2/xw_e8_s216.pth"
+ vits = VitsModel(vits_path).to(device)
+ vits.eval()
+
+ # gpt_path = "GPT_weights_v2/xw-e15.ckpt"
+ # dict_s1 = torch.load(gpt_path, map_location=device)
+ dict_s1 = torch.load(gpt_path)
+ raw_t2s = get_raw_t2s_model(dict_s1).to(device)
+ print("#### get_raw_t2s_model ####")
+ print(raw_t2s.config)
+ t2s_m = T2SModel(raw_t2s)
+ t2s_m.eval()
+ t2s = torch.jit.script(t2s_m).to(device)
+ print("#### script t2s_m ####")
+
+ print("vits.hps.data.sampling_rate:", vits.hps.data.sampling_rate)
+ gpt_sovits = GPT_SoVITS(t2s, vits).to(device)
+ gpt_sovits.eval()
+
+ ref_audio_sr = s.resample(ref_audio, 16000, 32000).to(device)
+
+ torch._dynamo.mark_dynamic(ssl_content, 2)
+ torch._dynamo.mark_dynamic(ref_audio_sr, 1)
+ torch._dynamo.mark_dynamic(ref_seq, 1)
+ torch._dynamo.mark_dynamic(text_seq, 1)
+ torch._dynamo.mark_dynamic(ref_bert, 0)
+ torch._dynamo.mark_dynamic(text_bert, 0)
+
+ top_k = torch.LongTensor([5]).to(device)
+
+ with torch.no_grad():
+ gpt_sovits_export = torch.jit.trace(
+ gpt_sovits, example_inputs=(ssl_content, ref_audio_sr, ref_seq, text_seq, ref_bert, text_bert, top_k)
+ )
+
+ gpt_sovits_path = os.path.join(output_path, "gpt_sovits_model.pt")
+ gpt_sovits_export.save(gpt_sovits_path)
+ print("#### exported gpt_sovits ####")
+
+
+@torch.jit.script
+def parse_audio(ref_audio):
+ ref_audio_16k = torchaudio.functional.resample(ref_audio, 48000, 16000).float() # .to(ref_audio.device)
+ ref_audio_sr = torchaudio.functional.resample(ref_audio, 48000, 32000).float() # .to(ref_audio.device)
+ return ref_audio_16k, ref_audio_sr
+
+
+@torch.jit.script
+def resamplex(ref_audio: torch.Tensor, src_sr: int, dst_sr: int) -> torch.Tensor:
+ return torchaudio.functional.resample(ref_audio, src_sr, dst_sr).float()
+
+
+class GPT_SoVITS(nn.Module):
+ def __init__(self, t2s: T2SModel, vits: VitsModel):
+ super().__init__()
+ self.t2s = t2s
+ self.vits = vits
+
+ def forward(
+ self,
+ ssl_content: torch.Tensor,
+ ref_audio_sr: torch.Tensor,
+ ref_seq: Tensor,
+ text_seq: Tensor,
+ ref_bert: Tensor,
+ text_bert: Tensor,
+ top_k: LongTensor,
+ speed=1.0,
+ ):
+ codes = self.vits.vq_model.extract_latent(ssl_content)
+ prompt_semantic = codes[0, 0]
+ prompts = prompt_semantic.unsqueeze(0)
+
+ pred_semantic = self.t2s(prompts, ref_seq, text_seq, ref_bert, text_bert, top_k)
+ audio = self.vits(text_seq, pred_semantic, ref_audio_sr, speed)
+ return audio
+
+
+def test():
+ parser = argparse.ArgumentParser(description="GPT-SoVITS Command Line Tool")
+ parser.add_argument("--gpt_model", required=True, help="Path to the GPT model file")
+ parser.add_argument("--sovits_model", required=True, help="Path to the SoVITS model file")
+ parser.add_argument("--ref_audio", required=True, help="Path to the reference audio file")
+ parser.add_argument("--ref_text", required=True, help="Path to the reference text file")
+ parser.add_argument("--output_path", required=True, help="Path to the output directory")
+
+ args = parser.parse_args()
+ gpt_path = args.gpt_model
+ vits_path = args.sovits_model
+ ref_audio_path = args.ref_audio
+ ref_text = args.ref_text
+
+ tokenizer = AutoTokenizer.from_pretrained(bert_path)
+ # bert_model = AutoModelForMaskedLM.from_pretrained(bert_path,output_hidden_states=True,torchscript=True)
+ # bert = MyBertModel(bert_model)
+ my_bert = torch.jit.load("onnx/bert_model.pt", map_location="cuda")
+
+ # dict_s1 = torch.load(gpt_path, map_location="cuda")
+ # raw_t2s = get_raw_t2s_model(dict_s1)
+ # t2s = T2SModel(raw_t2s)
+ # t2s.eval()
+ # t2s = torch.jit.load("onnx/xw/t2s_model.pt",map_location='cuda')
+
+ # vits_path = "SoVITS_weights_v2/xw_e8_s216.pth"
+ # vits = VitsModel(vits_path)
+ # vits.eval()
+
+ # ssl = ExportSSLModel(SSLModel()).to('cuda')
+ # ssl.eval()
+ ssl = torch.jit.load("onnx/by/ssl_model.pt", map_location="cuda")
+
+ # gpt_sovits = GPT_SoVITS(t2s,vits)
+ gpt_sovits = torch.jit.load("onnx/by/gpt_sovits_model.pt", map_location="cuda")
+
+ ref_seq_id, ref_bert_T, ref_norm_text = get_phones_and_bert(ref_text, "all_zh", "v2")
+ ref_seq = torch.LongTensor([ref_seq_id])
+ ref_bert = ref_bert_T.T.to(ref_seq.device)
+ # text_seq_id,text_bert_T,norm_text = get_phones_and_bert("昨天晚上看见征兵文书,知道君主在大规模征兵,那么多卷征兵文册,每一卷上都有父亲的名字.","all_zh",'v2')
+ text = "昨天晚上看见征兵文书,知道君主在大规模征兵,那么多卷征兵文册,每一卷上都有父亲的名字."
+
+ text_seq_id, text_bert_T, norm_text = get_phones_and_bert(text, "all_zh", "v2")
+
+ test_bert = tokenizer(text, return_tensors="pt")
+ word2ph = []
+ for c in text:
+ if c in [",", "。", ":", "?", "?", ",", "."]:
+ word2ph.append(1)
+ else:
+ word2ph.append(2)
+ test_bert["word2ph"] = torch.Tensor(word2ph).int()
+
+ test_bert = my_bert(
+ test_bert["input_ids"].to("cuda"),
+ test_bert["attention_mask"].to("cuda"),
+ test_bert["token_type_ids"].to("cuda"),
+ test_bert["word2ph"].to("cuda"),
+ )
+
+ text_seq = torch.LongTensor([text_seq_id])
+ text_bert = text_bert_T.T.to(text_seq.device)
+
+ print("text_bert:", text_bert.shape, text_bert)
+ print("test_bert:", test_bert.shape, test_bert)
+ print(torch.allclose(text_bert.to("cuda"), test_bert))
+
+ print("text_seq:", text_seq.shape)
+ print("text_bert:", text_bert.shape, text_bert.type())
+
+ # [1,N]
+ ref_audio = torch.tensor([load_audio(ref_audio_path, 16000)]).float().to("cuda")
+ print("ref_audio:", ref_audio.shape)
+
+ ref_audio_sr = ssl.resample(ref_audio, 16000, 32000)
+ print("start ssl")
+ ssl_content = ssl(ref_audio)
+
+ print("start gpt_sovits:")
+ print("ssl_content:", ssl_content.shape)
+ print("ref_audio_sr:", ref_audio_sr.shape)
+ print("ref_seq:", ref_seq.shape)
+ ref_seq = ref_seq.to("cuda")
+ print("text_seq:", text_seq.shape)
+ text_seq = text_seq.to("cuda")
+ print("ref_bert:", ref_bert.shape)
+ ref_bert = ref_bert.to("cuda")
+ print("text_bert:", text_bert.shape)
+ text_bert = text_bert.to("cuda")
+
+ top_k = torch.LongTensor([5]).to("cuda")
+
+ with torch.no_grad():
+ audio = gpt_sovits(ssl_content, ref_audio_sr, ref_seq, text_seq, ref_bert, test_bert, top_k)
+ print("start write wav")
+ soundfile.write("out.wav", audio.detach().cpu().numpy(), 32000)
+
+
+import text
+import json
+
+
+def export_symbel(version="v2"):
+ if version == "v1":
+ symbols = text._symbol_to_id_v1
+ with open("onnx/symbols_v1.json", "w") as file:
+ json.dump(symbols, file, indent=4)
+ else:
+ symbols = text._symbol_to_id_v2
+ with open("onnx/symbols_v2.json", "w") as file:
+ json.dump(symbols, file, indent=4)
+
+
+def main():
+ parser = argparse.ArgumentParser(description="GPT-SoVITS Command Line Tool")
+ parser.add_argument("--gpt_model", required=True, help="Path to the GPT model file")
+ parser.add_argument("--sovits_model", required=True, help="Path to the SoVITS model file")
+ parser.add_argument("--ref_audio", required=True, help="Path to the reference audio file")
+ parser.add_argument("--ref_text", required=True, help="Path to the reference text file")
+ parser.add_argument("--output_path", required=True, help="Path to the output directory")
+ parser.add_argument("--export_common_model", action="store_true", help="Export Bert and SSL model")
+ parser.add_argument("--device", help="Device to use")
+
+ args = parser.parse_args()
+ export(
+ gpt_path=args.gpt_model,
+ vits_path=args.sovits_model,
+ ref_audio_path=args.ref_audio,
+ ref_text=args.ref_text,
+ output_path=args.output_path,
+ device=args.device,
+ export_bert_and_ssl=args.export_common_model,
+ )
+
+
+import inference_webui
+
+if __name__ == "__main__":
+ inference_webui.is_half = False
+ inference_webui.dtype = torch.float32
+ main()
+ # test()
diff --git a/GPT_SoVITS/export_torch_script_v3.py b/GPT_SoVITS/export_torch_script_v3.py
new file mode 100644
index 0000000000000000000000000000000000000000..b34495a7adc3486ea2eb482c43c41e20ddbc7c48
--- /dev/null
+++ b/GPT_SoVITS/export_torch_script_v3.py
@@ -0,0 +1,1035 @@
+import os
+from export_torch_script import (
+ T2SModel,
+ get_raw_t2s_model,
+ resamplex,
+ spectrogram_torch,
+)
+from f5_tts.model.backbones.dit import DiT
+from inference_webui import get_phones_and_bert
+import librosa
+from module import commons
+from module.mel_processing import mel_spectrogram_torch
+from module.models_onnx import CFM, SynthesizerTrnV3
+import numpy as np
+import torch._dynamo.config
+import torchaudio
+import logging
+import uvicorn
+import torch
+import soundfile
+from librosa.filters import mel as librosa_mel_fn
+
+
+from inference_webui import get_spepc, norm_spec, resample, ssl_model
+
+logging.config.dictConfig(uvicorn.config.LOGGING_CONFIG)
+logger = logging.getLogger("uvicorn")
+
+is_half = True
+device = "cuda" if torch.cuda.is_available() else "cpu"
+now_dir = os.getcwd()
+
+
+class MelSpectrgram(torch.nn.Module):
+ def __init__(
+ self,
+ dtype,
+ device,
+ n_fft,
+ num_mels,
+ sampling_rate,
+ hop_size,
+ win_size,
+ fmin,
+ fmax,
+ center=False,
+ ):
+ super().__init__()
+ self.hann_window = torch.hann_window(1024).to(device=device, dtype=dtype)
+ mel = librosa_mel_fn(sr=sampling_rate, n_fft=n_fft, n_mels=num_mels, fmin=fmin, fmax=fmax)
+ self.mel_basis = torch.from_numpy(mel).to(dtype=dtype, device=device)
+ self.n_fft: int = n_fft
+ self.hop_size: int = hop_size
+ self.win_size: int = win_size
+ self.center: bool = center
+
+ def forward(self, y):
+ y = torch.nn.functional.pad(
+ y.unsqueeze(1),
+ (
+ int((self.n_fft - self.hop_size) / 2),
+ int((self.n_fft - self.hop_size) / 2),
+ ),
+ mode="reflect",
+ )
+ y = y.squeeze(1)
+ spec = torch.stft(
+ y,
+ self.n_fft,
+ hop_length=self.hop_size,
+ win_length=self.win_size,
+ window=self.hann_window,
+ center=self.center,
+ pad_mode="reflect",
+ normalized=False,
+ onesided=True,
+ return_complex=False,
+ )
+ spec = torch.sqrt(spec.pow(2).sum(-1) + 1e-9)
+ spec = torch.matmul(self.mel_basis, spec)
+ # spec = spectral_normalize_torch(spec)
+ spec = torch.log(torch.clamp(spec, min=1e-5))
+ return spec
+
+
+class ExportDitBlocks(torch.nn.Module):
+ def __init__(self, dit: DiT):
+ super().__init__()
+ self.transformer_blocks = dit.transformer_blocks
+ self.norm_out = dit.norm_out
+ self.proj_out = dit.proj_out
+ self.depth = dit.depth
+
+ def forward(self, x, t, mask, rope):
+ for block in self.transformer_blocks:
+ x = block(x, t, mask=mask, rope=(rope, 1.0))
+ x = self.norm_out(x, t)
+ output = self.proj_out(x)
+ return output
+
+
+class ExportDitEmbed(torch.nn.Module):
+ def __init__(self, dit: DiT):
+ super().__init__()
+ self.time_embed = dit.time_embed
+ self.d_embed = dit.d_embed
+ self.text_embed = dit.text_embed
+ self.input_embed = dit.input_embed
+ self.rotary_embed = dit.rotary_embed
+ self.rotary_embed.inv_freq.to(device)
+
+ def forward(
+ self,
+ x0: torch.Tensor, # nosied input audio # noqa: F722
+ cond0: torch.Tensor, # masked cond audio # noqa: F722
+ x_lens: torch.Tensor,
+ time: torch.Tensor, # time step # noqa: F821 F722
+ dt_base_bootstrap: torch.Tensor,
+ text0: torch.Tensor, # noqa: F722#####condition feature
+ ):
+ x = x0.transpose(2, 1)
+ cond = cond0.transpose(2, 1)
+ text = text0.transpose(2, 1)
+ mask = commons.sequence_mask(x_lens, max_length=x.size(1)).to(x.device)
+
+ t = self.time_embed(time) + self.d_embed(dt_base_bootstrap)
+ text_embed = self.text_embed(text, x.shape[1])
+ rope_t = torch.arange(x.shape[1], device=device)
+ rope, _ = self.rotary_embed(rope_t)
+ x = self.input_embed(x, cond, text_embed)
+ return x, t, mask, rope
+
+
+class ExportDiT(torch.nn.Module):
+ def __init__(self, dit: DiT):
+ super().__init__()
+ if dit != None:
+ self.embed = ExportDitEmbed(dit)
+ self.blocks = ExportDitBlocks(dit)
+ else:
+ self.embed = None
+ self.blocks = None
+
+ def forward( # x, prompt_x, x_lens, t, style,cond
+ self, # d is channel,n is T
+ x0: torch.Tensor, # nosied input audio # noqa: F722
+ cond0: torch.Tensor, # masked cond audio # noqa: F722
+ x_lens: torch.Tensor,
+ time: torch.Tensor, # time step # noqa: F821 F722
+ dt_base_bootstrap: torch.Tensor,
+ text0: torch.Tensor, # noqa: F722#####condition feature
+ ):
+ x, t, mask, rope = self.embed(x0, cond0, x_lens, time, dt_base_bootstrap, text0)
+ output = self.blocks(x, t, mask, rope)
+ return output
+
+
+class ExportCFM(torch.nn.Module):
+ def __init__(self, cfm: CFM):
+ super().__init__()
+ self.cfm = cfm
+
+ def forward(
+ self,
+ fea_ref: torch.Tensor,
+ fea_todo_chunk: torch.Tensor,
+ mel2: torch.Tensor,
+ sample_steps: torch.LongTensor,
+ ):
+ T_min = fea_ref.size(2)
+ fea = torch.cat([fea_ref, fea_todo_chunk], 2).transpose(2, 1)
+ cfm_res = self.cfm(fea, torch.LongTensor([fea.size(1)]).to(fea.device), mel2, sample_steps)
+ cfm_res = cfm_res[:, :, mel2.shape[2] :]
+ mel2 = cfm_res[:, :, -T_min:]
+ fea_ref = fea_todo_chunk[:, :, -T_min:]
+ return cfm_res, fea_ref, mel2
+
+
+mel_fn = lambda x: mel_spectrogram_torch(
+ x,
+ **{
+ "n_fft": 1024,
+ "win_size": 1024,
+ "hop_size": 256,
+ "num_mels": 100,
+ "sampling_rate": 24000,
+ "fmin": 0,
+ "fmax": None,
+ "center": False,
+ },
+)
+
+spec_min = -12
+spec_max = 2
+
+
+@torch.jit.script
+def norm_spec(x):
+ spec_min = -12
+ spec_max = 2
+ return (x - spec_min) / (spec_max - spec_min) * 2 - 1
+
+
+def denorm_spec(x):
+ spec_min = -12
+ spec_max = 2
+ return (x + 1) / 2 * (spec_max - spec_min) + spec_min
+
+
+class ExportGPTSovitsHalf(torch.nn.Module):
+ def __init__(self, hps, t2s_m: T2SModel, vq_model: SynthesizerTrnV3):
+ super().__init__()
+ self.hps = hps
+ self.t2s_m = t2s_m
+ self.vq_model = vq_model
+ self.mel2 = MelSpectrgram(
+ dtype=torch.float32,
+ device=device,
+ n_fft=1024,
+ num_mels=100,
+ sampling_rate=24000,
+ hop_size=256,
+ win_size=1024,
+ fmin=0,
+ fmax=None,
+ center=False,
+ )
+ # self.dtype = dtype
+ self.filter_length: int = hps.data.filter_length
+ self.sampling_rate: int = hps.data.sampling_rate
+ self.hop_length: int = hps.data.hop_length
+ self.win_length: int = hps.data.win_length
+
+ def forward(
+ self,
+ ssl_content,
+ ref_audio_32k: torch.FloatTensor,
+ phoneme_ids0,
+ phoneme_ids1,
+ bert1,
+ bert2,
+ top_k,
+ ):
+ refer = spectrogram_torch(
+ ref_audio_32k,
+ self.filter_length,
+ self.sampling_rate,
+ self.hop_length,
+ self.win_length,
+ center=False,
+ ).to(ssl_content.dtype)
+
+ codes = self.vq_model.extract_latent(ssl_content)
+ prompt_semantic = codes[0, 0]
+ prompt = prompt_semantic.unsqueeze(0)
+ # print('extract_latent',codes.shape,datetime.now().strftime("%Y-%m-%d %H:%M:%S"))
+
+ pred_semantic = self.t2s_m(prompt, phoneme_ids0, phoneme_ids1, bert1, bert2, top_k)
+ # print('t2s_m',pred_semantic.shape,datetime.now().strftime("%Y-%m-%d %H:%M:%S"))
+
+ ge = self.vq_model.create_ge(refer)
+ # print('create_ge',datetime.now().strftime("%Y-%m-%d %H:%M:%S"))
+
+ prompt_ = prompt.unsqueeze(0)
+ fea_ref = self.vq_model(prompt_, phoneme_ids0, ge)
+ # print('fea_ref',datetime.now().strftime("%Y-%m-%d %H:%M:%S"))
+ # print(prompt_.shape, phoneme_ids0.shape, ge.shape)
+ # print(fea_ref.shape)
+
+ ref_24k = resamplex(ref_audio_32k, 32000, 24000)
+ mel2 = norm_spec(self.mel2(ref_24k)).to(ssl_content.dtype)
+ T_min = min(mel2.shape[2], fea_ref.shape[2])
+ mel2 = mel2[:, :, :T_min]
+ fea_ref = fea_ref[:, :, :T_min]
+ if T_min > 468:
+ mel2 = mel2[:, :, -468:]
+ fea_ref = fea_ref[:, :, -468:]
+ T_min = 468
+
+ fea_todo = self.vq_model(pred_semantic, phoneme_ids1, ge)
+ # print('fea_todo',datetime.now().strftime("%Y-%m-%d %H:%M:%S"))
+ # print(pred_semantic.shape, phoneme_ids1.shape, ge.shape)
+ # print(fea_todo.shape)
+
+ return fea_ref, fea_todo, mel2
+
+
+class GPTSoVITSV3(torch.nn.Module):
+ def __init__(self, gpt_sovits_half, cfm, bigvgan):
+ super().__init__()
+ self.gpt_sovits_half = gpt_sovits_half
+ self.cfm = cfm
+ self.bigvgan = bigvgan
+
+ def forward(
+ self,
+ ssl_content,
+ ref_audio_32k: torch.FloatTensor,
+ phoneme_ids0: torch.LongTensor,
+ phoneme_ids1: torch.LongTensor,
+ bert1,
+ bert2,
+ top_k: torch.LongTensor,
+ sample_steps: torch.LongTensor,
+ ):
+ # current_time = datetime.now()
+ # print("gpt_sovits_half",current_time.strftime("%Y-%m-%d %H:%M:%S"))
+ fea_ref, fea_todo, mel2 = self.gpt_sovits_half(
+ ssl_content, ref_audio_32k, phoneme_ids0, phoneme_ids1, bert1, bert2, top_k
+ )
+ chunk_len = 934 - fea_ref.shape[2]
+ wav_gen_list = []
+ idx = 0
+ wav_gen_length = fea_todo.shape[2] * 256
+ while 1:
+ # current_time = datetime.now()
+ # print("idx:",idx,current_time.strftime("%Y-%m-%d %H:%M:%S"))
+ fea_todo_chunk = fea_todo[:, :, idx : idx + chunk_len]
+ if fea_todo_chunk.shape[-1] == 0:
+ break
+
+ # 因为导出的模型在不同shape时会重新编译还是怎么的,会卡顿10s这样,
+ # 所以在这里补0让他shape维持不变
+ # 但是这样会导致生成的音频长度不对,所以在最后截取一下。
+ # 经过 bigvgan 之后音频长度就是 fea_todo.shape[2] * 256
+ complete_len = chunk_len - fea_todo_chunk.shape[-1]
+ if complete_len != 0:
+ fea_todo_chunk = torch.cat(
+ [
+ fea_todo_chunk,
+ torch.zeros(1, 512, complete_len).to(fea_todo_chunk.device).to(fea_todo_chunk.dtype),
+ ],
+ 2,
+ )
+
+ cfm_res, fea_ref, mel2 = self.cfm(fea_ref, fea_todo_chunk, mel2, sample_steps)
+ idx += chunk_len
+
+ cfm_res = denorm_spec(cfm_res)
+ bigvgan_res = self.bigvgan(cfm_res)
+ wav_gen_list.append(bigvgan_res)
+
+ wav_gen = torch.cat(wav_gen_list, 2)
+ return wav_gen[0][0][:wav_gen_length]
+
+
+def init_bigvgan():
+ global bigvgan_model
+ from BigVGAN import bigvgan
+
+ bigvgan_model = bigvgan.BigVGAN.from_pretrained(
+ "%s/GPT_SoVITS/pretrained_models/models--nvidia--bigvgan_v2_24khz_100band_256x" % (now_dir,),
+ use_cuda_kernel=False,
+ ) # if True, RuntimeError: Ninja is required to load C++ extensions
+ # remove weight norm in the model and set to eval mode
+ bigvgan_model.remove_weight_norm()
+ bigvgan_model = bigvgan_model.eval()
+ if is_half == True:
+ bigvgan_model = bigvgan_model.half().to(device)
+ else:
+ bigvgan_model = bigvgan_model.to(device)
+
+
+class Sovits:
+ def __init__(self, vq_model: SynthesizerTrnV3, cfm: CFM, hps):
+ self.vq_model = vq_model
+ self.hps = hps
+ cfm.estimator = ExportDiT(cfm.estimator)
+ self.cfm = cfm
+
+
+class DictToAttrRecursive(dict):
+ def __init__(self, input_dict):
+ super().__init__(input_dict)
+ for key, value in input_dict.items():
+ if isinstance(value, dict):
+ value = DictToAttrRecursive(value)
+ self[key] = value
+ setattr(self, key, value)
+
+ def __getattr__(self, item):
+ try:
+ return self[item]
+ except KeyError:
+ raise AttributeError(f"Attribute {item} not found")
+
+ def __setattr__(self, key, value):
+ if isinstance(value, dict):
+ value = DictToAttrRecursive(value)
+ super(DictToAttrRecursive, self).__setitem__(key, value)
+ super().__setattr__(key, value)
+
+ def __delattr__(self, item):
+ try:
+ del self[item]
+ except KeyError:
+ raise AttributeError(f"Attribute {item} not found")
+
+
+from process_ckpt import get_sovits_version_from_path_fast, load_sovits_new
+
+
+def get_sovits_weights(sovits_path):
+ path_sovits_v3 = "GPT_SoVITS/pretrained_models/s2Gv3.pth"
+ is_exist_s2gv3 = os.path.exists(path_sovits_v3)
+
+ version, model_version, if_lora_v3 = get_sovits_version_from_path_fast(sovits_path)
+ if if_lora_v3 == True and is_exist_s2gv3 == False:
+ logger.info("SoVITS V3 底模缺失,无法加载相应 LoRA 权重")
+
+ dict_s2 = load_sovits_new(sovits_path)
+ hps = dict_s2["config"]
+ hps = DictToAttrRecursive(hps)
+ hps.model.semantic_frame_rate = "25hz"
+ if "enc_p.text_embedding.weight" not in dict_s2["weight"]:
+ hps.model.version = "v2" # v3model,v2sybomls
+ elif dict_s2["weight"]["enc_p.text_embedding.weight"].shape[0] == 322:
+ hps.model.version = "v1"
+ else:
+ hps.model.version = "v2"
+
+ if model_version == "v3":
+ hps.model.version = "v3"
+
+ logger.info(f"hps: {hps}")
+
+ vq_model = SynthesizerTrnV3(
+ hps.data.filter_length // 2 + 1,
+ hps.train.segment_size // hps.data.hop_length,
+ n_speakers=hps.data.n_speakers,
+ **hps.model,
+ )
+ # init_bigvgan()
+ model_version = hps.model.version
+ logger.info(f"模型版本: {model_version}")
+
+ if is_half == True:
+ vq_model = vq_model.half().to(device)
+ else:
+ vq_model = vq_model.to(device)
+ vq_model.load_state_dict(dict_s2["weight"], strict=False)
+ vq_model.eval()
+
+ cfm = vq_model.cfm
+ del vq_model.cfm
+
+ sovits = Sovits(vq_model, cfm, hps)
+ return sovits
+
+
+logger.info(f"torch version {torch.__version__}")
+# ssl_model = cnhubert.get_model()
+# if is_half:
+# ssl_model = ssl_model.half().to(device)
+# else:
+# ssl_model = ssl_model.to(device)
+
+
+def export_cfm(
+ e_cfm: ExportCFM,
+ mu: torch.Tensor,
+ x_lens: torch.LongTensor,
+ prompt: torch.Tensor,
+ n_timesteps: torch.IntTensor,
+ temperature=1.0,
+):
+ cfm = e_cfm.cfm
+
+ B, T = mu.size(0), mu.size(1)
+ x = torch.randn([B, cfm.in_channels, T], device=mu.device, dtype=mu.dtype) * temperature
+ print("x:", x.shape, x.dtype)
+ prompt_len = prompt.size(-1)
+ prompt_x = torch.zeros_like(x, dtype=mu.dtype)
+ prompt_x[..., :prompt_len] = prompt[..., :prompt_len]
+ x[..., :prompt_len] = 0.0
+ mu = mu.transpose(2, 1)
+
+ ntimestep = int(n_timesteps)
+
+ t = torch.tensor(0.0, dtype=x.dtype, device=x.device)
+ d = torch.tensor(1.0 / ntimestep, dtype=x.dtype, device=x.device)
+
+ t_tensor = torch.ones(x.shape[0], device=x.device, dtype=mu.dtype) * t
+ d_tensor = torch.ones(x.shape[0], device=x.device, dtype=mu.dtype) * d
+
+ print(
+ "cfm input shapes:",
+ x.shape,
+ prompt_x.shape,
+ x_lens.shape,
+ t_tensor.shape,
+ d_tensor.shape,
+ mu.shape,
+ )
+
+ print("cfm input dtypes:", x.dtype, prompt_x.dtype, x_lens.dtype, t_tensor.dtype, d_tensor.dtype, mu.dtype)
+
+ estimator: ExportDiT = torch.jit.trace(
+ cfm.estimator,
+ optimize=True,
+ example_inputs=(x, prompt_x, x_lens, t_tensor, d_tensor, mu),
+ )
+ estimator.save("onnx/ad/estimator.pt")
+ # torch.onnx.export(
+ # cfm.estimator,
+ # (x, prompt_x, x_lens, t_tensor, d_tensor, mu),
+ # "onnx/ad/dit.onnx",
+ # input_names=["x", "prompt_x", "x_lens", "t", "d", "mu"],
+ # output_names=["output"],
+ # dynamic_axes={
+ # "x": [2],
+ # "prompt_x": [2],
+ # "mu": [2],
+ # },
+ # )
+ print("save estimator ok")
+ cfm.estimator = estimator
+ export_cfm = torch.jit.script(e_cfm)
+ export_cfm.save("onnx/ad/cfm.pt")
+ # sovits.cfm = cfm
+ # cfm.save("onnx/ad/cfm.pt")
+ return export_cfm
+
+
+def export():
+ sovits = get_sovits_weights("GPT_SoVITS/pretrained_models/s2Gv3.pth")
+
+ init_bigvgan()
+
+ dict_s1 = torch.load("GPT_SoVITS/pretrained_models/s1v3.ckpt")
+ raw_t2s = get_raw_t2s_model(dict_s1).to(device)
+ print("#### get_raw_t2s_model ####")
+ print(raw_t2s.config)
+
+ if is_half:
+ raw_t2s = raw_t2s.half().to(device)
+
+ t2s_m = T2SModel(raw_t2s)
+ t2s_m.eval()
+ script_t2s = torch.jit.script(t2s_m).to(device)
+
+ hps = sovits.hps
+ ref_wav_path = "onnx/ad/ref.wav"
+ speed = 1.0
+ sample_steps = 32
+ dtype = torch.float16 if is_half == True else torch.float32
+ refer = get_spepc(hps, ref_wav_path).to(device).to(dtype)
+ zero_wav = np.zeros(
+ int(hps.data.sampling_rate * 0.3),
+ dtype=np.float16 if is_half == True else np.float32,
+ )
+
+ with torch.no_grad():
+ wav16k, sr = librosa.load(ref_wav_path, sr=16000)
+ wav16k = torch.from_numpy(wav16k)
+ zero_wav_torch = torch.from_numpy(zero_wav)
+
+ if is_half == True:
+ wav16k = wav16k.half().to(device)
+ zero_wav_torch = zero_wav_torch.half().to(device)
+ else:
+ wav16k = wav16k.to(device)
+ zero_wav_torch = zero_wav_torch.to(device)
+ wav16k = torch.cat([wav16k, zero_wav_torch])
+ ssl_content = ssl_model.model(wav16k.unsqueeze(0))["last_hidden_state"].transpose(1, 2) # .float()
+ codes = sovits.vq_model.extract_latent(ssl_content)
+ prompt_semantic = codes[0, 0]
+ prompt = prompt_semantic.unsqueeze(0).to(device)
+
+ phones1, bert1, norm_text1 = get_phones_and_bert(
+ "你这老坏蛋,我找了你这么久,真没想到在这里找到你。他说。", "all_zh", "v3"
+ )
+ phones2, bert2, norm_text2 = get_phones_and_bert(
+ "这是一个简单的示例,真没想到这么简单就完成了。The King and His Stories.Once there was a king. He likes to write stories, but his stories were not good. As people were afraid of him, they all said his stories were good.After reading them, the writer at once turned to the soldiers and said: Take me back to prison, please.",
+ "auto",
+ "v3",
+ )
+ phoneme_ids0 = torch.LongTensor(phones1).to(device).unsqueeze(0)
+ phoneme_ids1 = torch.LongTensor(phones2).to(device).unsqueeze(0)
+
+ # codes = sovits.vq_model.extract_latent(ssl_content)
+ # prompt_semantic = codes[0, 0]
+ # prompts = prompt_semantic.unsqueeze(0)
+
+ top_k = torch.LongTensor([15]).to(device)
+ print("topk", top_k)
+
+ bert1 = bert1.T.to(device)
+ bert2 = bert2.T.to(device)
+ print(
+ prompt.dtype,
+ phoneme_ids0.dtype,
+ phoneme_ids1.dtype,
+ bert1.dtype,
+ bert2.dtype,
+ top_k.dtype,
+ )
+ print(
+ prompt.shape,
+ phoneme_ids0.shape,
+ phoneme_ids1.shape,
+ bert1.shape,
+ bert2.shape,
+ top_k.shape,
+ )
+ pred_semantic = t2s_m(prompt, phoneme_ids0, phoneme_ids1, bert1, bert2, top_k)
+
+ ge = sovits.vq_model.create_ge(refer)
+ prompt_ = prompt.unsqueeze(0)
+
+ torch._dynamo.mark_dynamic(prompt_, 2)
+ torch._dynamo.mark_dynamic(phoneme_ids0, 1)
+
+ fea_ref = sovits.vq_model(prompt_, phoneme_ids0, ge)
+
+ inputs = {
+ "forward": (prompt_, phoneme_ids0, ge),
+ "extract_latent": ssl_content,
+ "create_ge": refer,
+ }
+
+ trace_vq_model = torch.jit.trace_module(sovits.vq_model, inputs, optimize=True)
+ trace_vq_model.save("onnx/ad/vq_model.pt")
+
+ print(fea_ref.shape, fea_ref.dtype, ge.shape)
+ print(prompt_.shape, phoneme_ids0.shape, ge.shape)
+
+ # vq_model = torch.jit.trace(
+ # sovits.vq_model,
+ # optimize=True,
+ # # strict=False,
+ # example_inputs=(prompt_, phoneme_ids0, ge),
+ # )
+ # vq_model = sovits.vq_model
+ vq_model = trace_vq_model
+
+ gpt_sovits_half = ExportGPTSovitsHalf(sovits.hps, script_t2s, trace_vq_model)
+ torch.jit.script(gpt_sovits_half).save("onnx/ad/gpt_sovits_v3_half.pt")
+
+ ref_audio, sr = torchaudio.load(ref_wav_path)
+ ref_audio = ref_audio.to(device).float()
+ if ref_audio.shape[0] == 2:
+ ref_audio = ref_audio.mean(0).unsqueeze(0)
+ if sr != 24000:
+ ref_audio = resample(ref_audio, sr)
+ # mel2 = mel_fn(ref_audio)
+ mel2 = norm_spec(mel_fn(ref_audio))
+ T_min = min(mel2.shape[2], fea_ref.shape[2])
+ fea_ref = fea_ref[:, :, :T_min]
+ print("fea_ref:", fea_ref.shape, T_min)
+ if T_min > 468:
+ mel2 = mel2[:, :, -468:]
+ fea_ref = fea_ref[:, :, -468:]
+ T_min = 468
+ chunk_len = 934 - T_min
+ mel2 = mel2.to(dtype)
+
+ # fea_todo, ge = sovits.vq_model(pred_semantic,y_lengths, phoneme_ids1, ge)
+ fea_todo = vq_model(pred_semantic, phoneme_ids1, ge)
+
+ cfm_resss = []
+ idx = 0
+ sample_steps = torch.LongTensor([sample_steps]).to(device)
+ export_cfm_ = ExportCFM(sovits.cfm)
+ while 1:
+ print("idx:", idx)
+ fea_todo_chunk = fea_todo[:, :, idx : idx + chunk_len]
+ if fea_todo_chunk.shape[-1] == 0:
+ break
+
+ print(
+ "export_cfm:",
+ fea_ref.shape,
+ fea_todo_chunk.shape,
+ mel2.shape,
+ sample_steps.shape,
+ )
+ if idx == 0:
+ fea = torch.cat([fea_ref, fea_todo_chunk], 2).transpose(2, 1)
+ export_cfm_ = export_cfm(
+ export_cfm_,
+ fea,
+ torch.LongTensor([fea.size(1)]).to(fea.device),
+ mel2,
+ sample_steps,
+ )
+ # torch.onnx.export(
+ # export_cfm_,
+ # (
+ # fea_ref,
+ # fea_todo_chunk,
+ # mel2,
+ # sample_steps,
+ # ),
+ # "onnx/ad/cfm.onnx",
+ # input_names=["fea_ref", "fea_todo_chunk", "mel2", "sample_steps"],
+ # output_names=["cfm_res", "fea_ref_", "mel2_"],
+ # dynamic_axes={
+ # "fea_ref": [2],
+ # "fea_todo_chunk": [2],
+ # "mel2": [2],
+ # },
+ # )
+
+ idx += chunk_len
+
+ cfm_res, fea_ref, mel2 = export_cfm_(fea_ref, fea_todo_chunk, mel2, sample_steps)
+ cfm_resss.append(cfm_res)
+ continue
+
+ cmf_res = torch.cat(cfm_resss, 2)
+ cmf_res = denorm_spec(cmf_res).to(device)
+ print("cmf_res:", cmf_res.shape, cmf_res.dtype)
+ with torch.inference_mode():
+ cmf_res_rand = torch.randn(1, 100, 934).to(device).to(dtype)
+ torch._dynamo.mark_dynamic(cmf_res_rand, 2)
+ bigvgan_model_ = torch.jit.trace(bigvgan_model, optimize=True, example_inputs=(cmf_res_rand,))
+ bigvgan_model_.save("onnx/ad/bigvgan_model.pt")
+ wav_gen = bigvgan_model(cmf_res)
+ print("wav_gen:", wav_gen.shape, wav_gen.dtype)
+ audio = wav_gen[0][0].cpu().detach().numpy()
+
+ sr = 24000
+ soundfile.write("out.export.wav", (audio * 32768).astype(np.int16), sr)
+
+
+from datetime import datetime
+
+
+def test_export(
+ todo_text,
+ gpt_sovits_v3_half,
+ cfm,
+ bigvgan,
+ output,
+):
+ # hps = sovits.hps
+ ref_wav_path = "onnx/ad/ref.wav"
+ speed = 1.0
+ sample_steps = 8
+
+ dtype = torch.float16 if is_half == True else torch.float32
+
+ zero_wav = np.zeros(
+ int(16000 * 0.3),
+ dtype=np.float16 if is_half == True else np.float32,
+ )
+
+ with torch.no_grad():
+ wav16k, sr = librosa.load(ref_wav_path, sr=16000)
+ wav16k = torch.from_numpy(wav16k)
+ zero_wav_torch = torch.from_numpy(zero_wav)
+
+ if is_half == True:
+ wav16k = wav16k.half().to(device)
+ zero_wav_torch = zero_wav_torch.half().to(device)
+ else:
+ wav16k = wav16k.to(device)
+ zero_wav_torch = zero_wav_torch.to(device)
+ wav16k = torch.cat([wav16k, zero_wav_torch])
+ ssl_content = ssl_model.model(wav16k.unsqueeze(0))["last_hidden_state"].transpose(1, 2) # .float()
+
+ ref_audio_32k, _ = librosa.load(ref_wav_path, sr=32000)
+ ref_audio_32k = torch.from_numpy(ref_audio_32k).unsqueeze(0).to(device).float()
+
+ phones1, bert1, norm_text1 = get_phones_and_bert(
+ "你这老坏蛋,我找了你这么久,真没想到在这里找到你。他说。", "all_zh", "v3"
+ )
+ phones2, bert2, norm_text2 = get_phones_and_bert(
+ todo_text,
+ "zh",
+ "v3",
+ )
+ phoneme_ids0 = torch.LongTensor(phones1).to(device).unsqueeze(0)
+ phoneme_ids1 = torch.LongTensor(phones2).to(device).unsqueeze(0)
+
+ bert1 = bert1.T.to(device)
+ bert2 = bert2.T.to(device)
+ top_k = torch.LongTensor([15]).to(device)
+
+ current_time = datetime.now().strftime("%Y-%m-%d %H:%M:%S")
+ logger.info("start inference %s", current_time)
+ print(
+ ssl_content.shape,
+ ref_audio_32k.shape,
+ phoneme_ids0.shape,
+ phoneme_ids1.shape,
+ bert1.shape,
+ bert2.shape,
+ top_k.shape,
+ )
+ fea_ref, fea_todo, mel2 = gpt_sovits_v3_half(
+ ssl_content, ref_audio_32k, phoneme_ids0, phoneme_ids1, bert1, bert2, top_k
+ )
+ chunk_len = 934 - fea_ref.shape[2]
+ print(fea_ref.shape, fea_todo.shape, mel2.shape)
+
+ cfm_resss = []
+ sample_steps = torch.LongTensor([sample_steps])
+ idx = 0
+ current_time = datetime.now().strftime("%Y-%m-%d %H:%M:%S")
+ logger.info("start cfm %s", current_time)
+ wav_gen_length = fea_todo.shape[2] * 256
+
+ while 1:
+ current_time = datetime.now()
+ print("idx:", idx, current_time.strftime("%Y-%m-%d %H:%M:%S"))
+ fea_todo_chunk = fea_todo[:, :, idx : idx + chunk_len]
+ if fea_todo_chunk.shape[-1] == 0:
+ break
+
+ complete_len = chunk_len - fea_todo_chunk.shape[-1]
+ if complete_len != 0:
+ fea_todo_chunk = torch.cat([fea_todo_chunk, torch.zeros(1, 512, complete_len).to(device).to(dtype)], 2)
+
+ cfm_res, fea_ref, mel2 = cfm(fea_ref, fea_todo_chunk, mel2, sample_steps)
+ # if complete_len > 0 :
+ # cfm_res = cfm_res[:, :, :-complete_len]
+ # fea_ref = fea_ref[:, :, :-complete_len]
+ # mel2 = mel2[:, :, :-complete_len]
+
+ idx += chunk_len
+
+ current_time = datetime.now()
+ print("cfm end", current_time.strftime("%Y-%m-%d %H:%M:%S"))
+ cfm_res = denorm_spec(cfm_res).to(device)
+ bigvgan_res = bigvgan(cfm_res)
+ cfm_resss.append(bigvgan_res)
+
+ current_time = datetime.now().strftime("%Y-%m-%d %H:%M:%S")
+ logger.info("start bigvgan %s", current_time)
+ wav_gen = torch.cat(cfm_resss, 2)
+ # cmf_res = denorm_spec(cmf_res)
+ # cmf_res = cmf_res.to(device)
+ # print("cmf_res:", cmf_res.shape)
+
+ # cmf_res = torch.cat([cmf_res,torch.zeros([1,100,2000-cmf_res.size(2)],device=device,dtype=cmf_res.dtype)], 2)
+
+ # wav_gen = bigvgan(cmf_res)
+ print("wav_gen:", wav_gen.shape, wav_gen.dtype)
+ wav_gen = wav_gen[:, :, :wav_gen_length]
+
+ audio = wav_gen[0][0].cpu().detach().numpy()
+ logger.info("end bigvgan %s", datetime.now().strftime("%Y-%m-%d %H:%M:%S"))
+ sr = 24000
+ soundfile.write(output, (audio * 32768).astype(np.int16), sr)
+
+
+def test_export1(
+ todo_text,
+ gpt_sovits_v3,
+ output,
+):
+ # hps = sovits.hps
+ ref_wav_path = "onnx/ad/ref.wav"
+ speed = 1.0
+ sample_steps = torch.LongTensor([16])
+
+ dtype = torch.float16 if is_half == True else torch.float32
+
+ zero_wav = np.zeros(
+ int(24000 * 0.3),
+ dtype=np.float16 if is_half == True else np.float32,
+ )
+
+ with torch.no_grad():
+ wav16k, sr = librosa.load(ref_wav_path, sr=16000)
+ wav16k = torch.from_numpy(wav16k)
+ zero_wav_torch = torch.from_numpy(zero_wav)
+
+ if is_half == True:
+ wav16k = wav16k.half().to(device)
+ zero_wav_torch = zero_wav_torch.half().to(device)
+ else:
+ wav16k = wav16k.to(device)
+ zero_wav_torch = zero_wav_torch.to(device)
+ wav16k = torch.cat([wav16k, zero_wav_torch])
+ ssl_content = ssl_model.model(wav16k.unsqueeze(0))["last_hidden_state"].transpose(1, 2) # .float()
+ print("ssl_content:", ssl_content.shape, ssl_content.dtype)
+
+ ref_audio_32k, _ = librosa.load(ref_wav_path, sr=32000)
+ ref_audio_32k = torch.from_numpy(ref_audio_32k).unsqueeze(0).to(device).float()
+
+ phones1, bert1, norm_text1 = get_phones_and_bert(
+ "你这老坏蛋,我找了你这么久,真没想到在这里找到你。他说。", "all_zh", "v3"
+ )
+ phones2, bert2, norm_text2 = get_phones_and_bert(
+ todo_text,
+ "zh",
+ "v3",
+ )
+ phoneme_ids0 = torch.LongTensor(phones1).to(device).unsqueeze(0)
+ phoneme_ids1 = torch.LongTensor(phones2).to(device).unsqueeze(0)
+
+ bert1 = bert1.T.to(device)
+ bert2 = bert2.T.to(device)
+ top_k = torch.LongTensor([15]).to(device)
+
+ current_time = datetime.now().strftime("%Y-%m-%d %H:%M:%S")
+ logger.info("start inference %s", current_time)
+ print(
+ ssl_content.shape,
+ ref_audio_32k.shape,
+ phoneme_ids0.shape,
+ phoneme_ids1.shape,
+ bert1.shape,
+ bert2.shape,
+ top_k.shape,
+ )
+ wav_gen = gpt_sovits_v3(ssl_content, ref_audio_32k, phoneme_ids0, phoneme_ids1, bert1, bert2, top_k, sample_steps)
+ print("wav_gen:", wav_gen.shape, wav_gen.dtype)
+
+ wav_gen = torch.cat([wav_gen, zero_wav_torch], 0)
+
+ audio = wav_gen.cpu().detach().numpy()
+ logger.info("end bigvgan %s", datetime.now().strftime("%Y-%m-%d %H:%M:%S"))
+ sr = 24000
+ soundfile.write(output, (audio * 32768).astype(np.int16), sr)
+
+
+import time
+
+
+def test_():
+ sovits = get_sovits_weights("GPT_SoVITS/pretrained_models/s2Gv3.pth")
+
+ # cfm = ExportCFM(sovits.cfm)
+ # cfm.cfm.estimator = dit
+ sovits.cfm = None
+
+ cfm = torch.jit.load("onnx/ad/cfm.pt", map_location=device)
+ # cfm = torch.jit.optimize_for_inference(cfm)
+ cfm = cfm.half().to(device)
+
+ cfm.eval()
+
+ logger.info("cfm ok")
+
+ dict_s1 = torch.load("GPT_SoVITS/pretrained_models/s1v3.ckpt")
+ # v2 的 gpt 也可以用
+ # dict_s1 = torch.load("GPT_SoVITS/pretrained_models/gsv-v2final-pretrained/s1bert25hz-5kh-longer-epoch=12-step=369668.ckpt")
+ raw_t2s = get_raw_t2s_model(dict_s1).to(device)
+ print("#### get_raw_t2s_model ####")
+ print(raw_t2s.config)
+ if is_half:
+ raw_t2s = raw_t2s.half().to(device)
+ t2s_m = T2SModel(raw_t2s).half().to(device)
+ t2s_m.eval()
+ t2s_m = torch.jit.script(t2s_m)
+ t2s_m.eval()
+ # t2s_m.top_k = 15
+ logger.info("t2s_m ok")
+
+ vq_model: torch.jit.ScriptModule = torch.jit.load("onnx/ad/vq_model.pt", map_location=device)
+ # vq_model = torch.jit.optimize_for_inference(vq_model)
+ # vq_model = vq_model.half().to(device)
+ vq_model.eval()
+ # vq_model = sovits.vq_model
+ logger.info("vq_model ok")
+
+ # gpt_sovits_v3_half = torch.jit.load("onnx/ad/gpt_sovits_v3_half.pt")
+ # gpt_sovits_v3_half = torch.jit.optimize_for_inference(gpt_sovits_v3_half)
+ # gpt_sovits_v3_half = gpt_sovits_v3_half.half()
+ # gpt_sovits_v3_half = gpt_sovits_v3_half.cuda()
+ # gpt_sovits_v3_half.eval()
+ gpt_sovits_v3_half = ExportGPTSovitsHalf(sovits.hps, t2s_m, vq_model)
+ logger.info("gpt_sovits_v3_half ok")
+
+ # init_bigvgan()
+ # global bigvgan_model
+ bigvgan_model = torch.jit.load("onnx/ad/bigvgan_model.pt")
+ # bigvgan_model = torch.jit.optimize_for_inference(bigvgan_model)
+ bigvgan_model = bigvgan_model.half()
+ bigvgan_model = bigvgan_model.cuda()
+ bigvgan_model.eval()
+
+ logger.info("bigvgan ok")
+
+ gpt_sovits_v3 = GPTSoVITSV3(gpt_sovits_v3_half, cfm, bigvgan_model)
+ gpt_sovits_v3 = torch.jit.script(gpt_sovits_v3)
+ gpt_sovits_v3.save("onnx/ad/gpt_sovits_v3.pt")
+ gpt_sovits_v3 = gpt_sovits_v3.half().to(device)
+ gpt_sovits_v3.eval()
+ print("save gpt_sovits_v3 ok")
+
+ time.sleep(5)
+ # print("thread:", torch.get_num_threads())
+ # print("thread:", torch.get_num_interop_threads())
+ # torch.set_num_interop_threads(1)
+ # torch.set_num_threads(1)
+
+ test_export1(
+ "汗流浃背了呀!老弟~ My uncle has two dogs. One is big and the other is small. He likes them very much. He often plays with them. He takes them for a walk every day. He says they are his good friends. He is very happy with them. 最后还是我得了 MVP....",
+ gpt_sovits_v3,
+ "out.wav",
+ )
+
+ test_export1(
+ "你小子是什么来路.汗流浃背了呀!老弟~ My uncle has two dogs. He is very happy with them. 最后还是我得了 MVP!",
+ gpt_sovits_v3,
+ "out2.wav",
+ )
+
+ # test_export(
+ # "汗流浃背了呀!老弟~ My uncle has two dogs. One is big and the other is small. He likes them very much. He often plays with them. He takes them for a walk every day. He says they are his good friends. He is very happy with them. 最后还是我得了 MVP. 哈哈哈...",
+ # gpt_sovits_v3_half,
+ # cfm,
+ # bigvgan_model,
+ # "out2.wav",
+ # )
+
+
+def test_export_gpt_sovits_v3():
+ gpt_sovits_v3 = torch.jit.load("onnx/ad/gpt_sovits_v3.pt", map_location=device)
+ # test_export1(
+ # "汗流浃背了呀!老弟~ My uncle has two dogs. One is big and the other is small. He likes them very much. He often plays with them. He takes them for a walk every day. He says they are his good friends. He is very happy with them. 最后还是我得了 MVP....",
+ # gpt_sovits_v3,
+ # "out3.wav",
+ # )
+ # test_export1(
+ # "你小子是什么来路.汗流浃背了呀!老弟~ My uncle has two dogs. He is very happy with them. 最后还是我得了 MVP!",
+ # gpt_sovits_v3,
+ # "out4.wav",
+ # )
+ test_export1(
+ "风萧萧兮易水寒,壮士一去兮不复还.",
+ gpt_sovits_v3,
+ "out5.wav",
+ )
+
+
+with torch.no_grad():
+ # export()
+ test_()
+ # test_export_gpt_sovits_v3()
diff --git a/GPT_SoVITS/inference_cli.py b/GPT_SoVITS/inference_cli.py
new file mode 100644
index 0000000000000000000000000000000000000000..459a3d3632f599768465c16f6d889f47af5fe271
--- /dev/null
+++ b/GPT_SoVITS/inference_cli.py
@@ -0,0 +1,86 @@
+import argparse
+import os
+import soundfile as sf
+
+from tools.i18n.i18n import I18nAuto
+from GPT_SoVITS.inference_webui import change_gpt_weights, change_sovits_weights, get_tts_wav
+
+i18n = I18nAuto()
+
+
+def synthesize(
+ GPT_model_path,
+ SoVITS_model_path,
+ ref_audio_path,
+ ref_text_path,
+ ref_language,
+ target_text_path,
+ target_language,
+ output_path,
+):
+ # Read reference text
+ with open(ref_text_path, "r", encoding="utf-8") as file:
+ ref_text = file.read()
+
+ # Read target text
+ with open(target_text_path, "r", encoding="utf-8") as file:
+ target_text = file.read()
+
+ # Change model weights
+ change_gpt_weights(gpt_path=GPT_model_path)
+ change_sovits_weights(sovits_path=SoVITS_model_path)
+
+ # Synthesize audio
+ synthesis_result = get_tts_wav(
+ ref_wav_path=ref_audio_path,
+ prompt_text=ref_text,
+ prompt_language=i18n(ref_language),
+ text=target_text,
+ text_language=i18n(target_language),
+ top_p=1,
+ temperature=1,
+ )
+
+ result_list = list(synthesis_result)
+
+ if result_list:
+ last_sampling_rate, last_audio_data = result_list[-1]
+ output_wav_path = os.path.join(output_path, "output.wav")
+ sf.write(output_wav_path, last_audio_data, last_sampling_rate)
+ print(f"Audio saved to {output_wav_path}")
+
+
+def main():
+ parser = argparse.ArgumentParser(description="GPT-SoVITS Command Line Tool")
+ parser.add_argument("--gpt_model", required=True, help="Path to the GPT model file")
+ parser.add_argument("--sovits_model", required=True, help="Path to the SoVITS model file")
+ parser.add_argument("--ref_audio", required=True, help="Path to the reference audio file")
+ parser.add_argument("--ref_text", required=True, help="Path to the reference text file")
+ parser.add_argument(
+ "--ref_language", required=True, choices=["中文", "英文", "日文"], help="Language of the reference audio"
+ )
+ parser.add_argument("--target_text", required=True, help="Path to the target text file")
+ parser.add_argument(
+ "--target_language",
+ required=True,
+ choices=["中文", "英文", "日文", "中英混合", "日英混合", "多语种混合"],
+ help="Language of the target text",
+ )
+ parser.add_argument("--output_path", required=True, help="Path to the output directory")
+
+ args = parser.parse_args()
+
+ synthesize(
+ args.gpt_model,
+ args.sovits_model,
+ args.ref_audio,
+ args.ref_text,
+ args.ref_language,
+ args.target_text,
+ args.target_language,
+ args.output_path,
+ )
+
+
+if __name__ == "__main__":
+ main()
diff --git a/GPT_SoVITS/inference_gui.py b/GPT_SoVITS/inference_gui.py
new file mode 100644
index 0000000000000000000000000000000000000000..379f7fa8cdb32b4b56db8b242717c23bdb51eca0
--- /dev/null
+++ b/GPT_SoVITS/inference_gui.py
@@ -0,0 +1,316 @@
+import os
+import sys
+from PyQt5.QtCore import QEvent
+from PyQt5.QtWidgets import QApplication, QMainWindow, QLabel, QLineEdit, QPushButton, QTextEdit
+from PyQt5.QtWidgets import QGridLayout, QVBoxLayout, QWidget, QFileDialog, QStatusBar, QComboBox
+import soundfile as sf
+
+from tools.i18n.i18n import I18nAuto
+
+i18n = I18nAuto()
+
+from inference_webui import gpt_path, sovits_path, change_gpt_weights, change_sovits_weights, get_tts_wav
+
+
+class GPTSoVITSGUI(QMainWindow):
+ GPT_Path = gpt_path
+ SoVITS_Path = sovits_path
+
+ def __init__(self):
+ super().__init__()
+
+ self.setWindowTitle("GPT-SoVITS GUI")
+ self.setGeometry(800, 450, 950, 850)
+
+ self.setStyleSheet("""
+ QWidget {
+ background-color: #a3d3b1;
+ }
+
+ QTabWidget::pane {
+ background-color: #a3d3b1;
+ }
+
+ QTabWidget::tab-bar {
+ alignment: left;
+ }
+
+ QTabBar::tab {
+ background: #8da4bf;
+ color: #ffffff;
+ padding: 8px;
+ }
+
+ QTabBar::tab:selected {
+ background: #2a3f54;
+ }
+
+ QLabel {
+ color: #000000;
+ }
+
+ QPushButton {
+ background-color: #4CAF50;
+ color: white;
+ padding: 8px;
+ border: 1px solid #4CAF50;
+ border-radius: 4px;
+ }
+
+ QPushButton:hover {
+ background-color: #45a049;
+ border: 1px solid #45a049;
+ box-shadow: 2px 2px 2px rgba(0, 0, 0, 0.1);
+ }
+ """)
+
+ license_text = (
+ "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."
+ )
+ license_label = QLabel(license_text)
+ license_label.setWordWrap(True)
+
+ self.GPT_model_label = QLabel("选择GPT模型:")
+ self.GPT_model_input = QLineEdit()
+ self.GPT_model_input.setPlaceholderText("拖拽或选择文件")
+ self.GPT_model_input.setText(self.GPT_Path)
+ self.GPT_model_input.setReadOnly(True)
+ self.GPT_model_button = QPushButton("选择GPT模型文件")
+ self.GPT_model_button.clicked.connect(self.select_GPT_model)
+
+ self.SoVITS_model_label = QLabel("选择SoVITS模型:")
+ self.SoVITS_model_input = QLineEdit()
+ self.SoVITS_model_input.setPlaceholderText("拖拽或选择文件")
+ self.SoVITS_model_input.setText(self.SoVITS_Path)
+ self.SoVITS_model_input.setReadOnly(True)
+ self.SoVITS_model_button = QPushButton("选择SoVITS模型文件")
+ self.SoVITS_model_button.clicked.connect(self.select_SoVITS_model)
+
+ self.ref_audio_label = QLabel("上传参考音频:")
+ self.ref_audio_input = QLineEdit()
+ self.ref_audio_input.setPlaceholderText("拖拽或选择文件")
+ self.ref_audio_input.setReadOnly(True)
+ self.ref_audio_button = QPushButton("选择音频文件")
+ self.ref_audio_button.clicked.connect(self.select_ref_audio)
+
+ self.ref_text_label = QLabel("参考音频文本:")
+ self.ref_text_input = QLineEdit()
+ self.ref_text_input.setPlaceholderText("直接输入文字或上传文本")
+ self.ref_text_button = QPushButton("上传文本")
+ self.ref_text_button.clicked.connect(self.upload_ref_text)
+
+ self.ref_language_label = QLabel("参考音频语言:")
+ self.ref_language_combobox = QComboBox()
+ self.ref_language_combobox.addItems(["中文", "英文", "日文", "中英混合", "日英混合", "多语种混合"])
+ self.ref_language_combobox.setCurrentText("多语种混合")
+
+ self.target_text_label = QLabel("合成目标文本:")
+ self.target_text_input = QLineEdit()
+ self.target_text_input.setPlaceholderText("直接输入文字或上传文本")
+ self.target_text_button = QPushButton("上传文本")
+ self.target_text_button.clicked.connect(self.upload_target_text)
+
+ self.target_language_label = QLabel("合成音频语言:")
+ self.target_language_combobox = QComboBox()
+ self.target_language_combobox.addItems(["中文", "英文", "日文", "中英混合", "日英混合", "多语种混合"])
+ self.target_language_combobox.setCurrentText("多语种混合")
+
+ self.output_label = QLabel("输出音频路径:")
+ self.output_input = QLineEdit()
+ self.output_input.setPlaceholderText("拖拽或选择文件")
+ self.output_input.setReadOnly(True)
+ self.output_button = QPushButton("选择文件夹")
+ self.output_button.clicked.connect(self.select_output_path)
+
+ self.output_text = QTextEdit()
+ self.output_text.setReadOnly(True)
+
+ self.add_drag_drop_events(
+ [
+ self.GPT_model_input,
+ self.SoVITS_model_input,
+ self.ref_audio_input,
+ self.ref_text_input,
+ self.target_text_input,
+ self.output_input,
+ ]
+ )
+
+ self.synthesize_button = QPushButton("合成")
+ self.synthesize_button.clicked.connect(self.synthesize)
+
+ self.clear_output_button = QPushButton("清空输出")
+ self.clear_output_button.clicked.connect(self.clear_output)
+
+ self.status_bar = QStatusBar()
+
+ main_layout = QVBoxLayout()
+
+ input_layout = QGridLayout(self)
+ input_layout.setSpacing(10)
+
+ input_layout.addWidget(license_label, 0, 0, 1, 3)
+
+ input_layout.addWidget(self.GPT_model_label, 1, 0)
+ input_layout.addWidget(self.GPT_model_input, 2, 0, 1, 2)
+ input_layout.addWidget(self.GPT_model_button, 2, 2)
+
+ input_layout.addWidget(self.SoVITS_model_label, 3, 0)
+ input_layout.addWidget(self.SoVITS_model_input, 4, 0, 1, 2)
+ input_layout.addWidget(self.SoVITS_model_button, 4, 2)
+
+ input_layout.addWidget(self.ref_audio_label, 5, 0)
+ input_layout.addWidget(self.ref_audio_input, 6, 0, 1, 2)
+ input_layout.addWidget(self.ref_audio_button, 6, 2)
+
+ input_layout.addWidget(self.ref_language_label, 7, 0)
+ input_layout.addWidget(self.ref_language_combobox, 8, 0, 1, 1)
+ input_layout.addWidget(self.ref_text_label, 9, 0)
+ input_layout.addWidget(self.ref_text_input, 10, 0, 1, 2)
+ input_layout.addWidget(self.ref_text_button, 10, 2)
+
+ input_layout.addWidget(self.target_language_label, 11, 0)
+ input_layout.addWidget(self.target_language_combobox, 12, 0, 1, 1)
+ input_layout.addWidget(self.target_text_label, 13, 0)
+ input_layout.addWidget(self.target_text_input, 14, 0, 1, 2)
+ input_layout.addWidget(self.target_text_button, 14, 2)
+
+ input_layout.addWidget(self.output_label, 15, 0)
+ input_layout.addWidget(self.output_input, 16, 0, 1, 2)
+ input_layout.addWidget(self.output_button, 16, 2)
+
+ main_layout.addLayout(input_layout)
+
+ output_layout = QVBoxLayout()
+ output_layout.addWidget(self.output_text)
+ main_layout.addLayout(output_layout)
+
+ main_layout.addWidget(self.synthesize_button)
+
+ main_layout.addWidget(self.clear_output_button)
+
+ main_layout.addWidget(self.status_bar)
+
+ self.central_widget = QWidget()
+ self.central_widget.setLayout(main_layout)
+ self.setCentralWidget(self.central_widget)
+
+ def dragEnterEvent(self, event):
+ if event.mimeData().hasUrls():
+ event.acceptProposedAction()
+
+ def dropEvent(self, event):
+ if event.mimeData().hasUrls():
+ file_paths = [url.toLocalFile() for url in event.mimeData().urls()]
+ if len(file_paths) == 1:
+ self.update_ref_audio(file_paths[0])
+ else:
+ self.update_ref_audio(", ".join(file_paths))
+
+ def add_drag_drop_events(self, widgets):
+ for widget in widgets:
+ widget.setAcceptDrops(True)
+ widget.installEventFilter(self)
+
+ def eventFilter(self, obj, event):
+ if event.type() in (QEvent.DragEnter, QEvent.Drop):
+ mime_data = event.mimeData()
+ if mime_data.hasUrls():
+ event.acceptProposedAction()
+
+ return super().eventFilter(obj, event)
+
+ def select_GPT_model(self):
+ file_path, _ = QFileDialog.getOpenFileName(self, "选择GPT模型文件", "", "GPT Files (*.ckpt)")
+ if file_path:
+ self.GPT_model_input.setText(file_path)
+
+ def select_SoVITS_model(self):
+ file_path, _ = QFileDialog.getOpenFileName(self, "选择SoVITS模型文件", "", "SoVITS Files (*.pth)")
+ if file_path:
+ self.SoVITS_model_input.setText(file_path)
+
+ def select_ref_audio(self):
+ file_path, _ = QFileDialog.getOpenFileName(self, "选择参考音频文件", "", "Audio Files (*.wav *.mp3)")
+ if file_path:
+ self.update_ref_audio(file_path)
+
+ def upload_ref_text(self):
+ file_path, _ = QFileDialog.getOpenFileName(self, "选择文本文件", "", "Text Files (*.txt)")
+ if file_path:
+ with open(file_path, "r", encoding="utf-8") as file:
+ content = file.read()
+ self.ref_text_input.setText(content)
+
+ def upload_target_text(self):
+ file_path, _ = QFileDialog.getOpenFileName(self, "选择文本文件", "", "Text Files (*.txt)")
+ if file_path:
+ with open(file_path, "r", encoding="utf-8") as file:
+ content = file.read()
+ self.target_text_input.setText(content)
+
+ def select_output_path(self):
+ options = QFileDialog.Options()
+ options |= QFileDialog.DontUseNativeDialog
+ options |= QFileDialog.ShowDirsOnly
+
+ folder_dialog = QFileDialog()
+ folder_dialog.setOptions(options)
+ folder_dialog.setFileMode(QFileDialog.Directory)
+
+ if folder_dialog.exec_():
+ folder_path = folder_dialog.selectedFiles()[0]
+ self.output_input.setText(folder_path)
+
+ def update_ref_audio(self, file_path):
+ self.ref_audio_input.setText(file_path)
+
+ def clear_output(self):
+ self.output_text.clear()
+
+ def synthesize(self):
+ GPT_model_path = self.GPT_model_input.text()
+ SoVITS_model_path = self.SoVITS_model_input.text()
+ ref_audio_path = self.ref_audio_input.text()
+ language_combobox = self.ref_language_combobox.currentText()
+ language_combobox = i18n(language_combobox)
+ ref_text = self.ref_text_input.text()
+ target_language_combobox = self.target_language_combobox.currentText()
+ target_language_combobox = i18n(target_language_combobox)
+ target_text = self.target_text_input.text()
+ output_path = self.output_input.text()
+
+ if GPT_model_path != self.GPT_Path:
+ change_gpt_weights(gpt_path=GPT_model_path)
+ self.GPT_Path = GPT_model_path
+ if SoVITS_model_path != self.SoVITS_Path:
+ change_sovits_weights(sovits_path=SoVITS_model_path)
+ self.SoVITS_Path = SoVITS_model_path
+
+ synthesis_result = get_tts_wav(
+ ref_wav_path=ref_audio_path,
+ prompt_text=ref_text,
+ prompt_language=language_combobox,
+ text=target_text,
+ text_language=target_language_combobox,
+ )
+
+ result_list = list(synthesis_result)
+
+ if result_list:
+ last_sampling_rate, last_audio_data = result_list[-1]
+ output_wav_path = os.path.join(output_path, "output.wav")
+ sf.write(output_wav_path, last_audio_data, last_sampling_rate)
+
+ result = "Audio saved to " + output_wav_path
+
+ self.status_bar.showMessage("合成完成!输出路径:" + output_wav_path, 5000)
+ self.output_text.append("处理结果:\n" + result)
+
+
+if __name__ == "__main__":
+ app = QApplication(sys.argv)
+ mainWin = GPTSoVITSGUI()
+ mainWin.show()
+ sys.exit(app.exec_())
diff --git a/GPT_SoVITS/inference_webui.py b/GPT_SoVITS/inference_webui.py
new file mode 100644
index 0000000000000000000000000000000000000000..68648014207e393f239fa041223a0935111280ac
--- /dev/null
+++ b/GPT_SoVITS/inference_webui.py
@@ -0,0 +1,1280 @@
+"""
+按中英混合识别
+按日英混合识别
+多语种启动切分识别语种
+全部按中文识别
+全部按英文识别
+全部按日文识别
+"""
+
+import logging
+import traceback
+import warnings
+
+import torchaudio
+
+logging.getLogger("markdown_it").setLevel(logging.ERROR)
+logging.getLogger("urllib3").setLevel(logging.ERROR)
+logging.getLogger("httpcore").setLevel(logging.ERROR)
+logging.getLogger("httpx").setLevel(logging.ERROR)
+logging.getLogger("asyncio").setLevel(logging.ERROR)
+logging.getLogger("charset_normalizer").setLevel(logging.ERROR)
+logging.getLogger("torchaudio._extension").setLevel(logging.ERROR)
+logging.getLogger("multipart.multipart").setLevel(logging.ERROR)
+warnings.simplefilter(action="ignore", category=FutureWarning)
+
+import json
+import os
+import re
+import sys
+
+import torch
+from text.LangSegmenter import LangSegmenter
+
+try:
+ import gradio.analytics as analytics
+
+ analytics.version_check = lambda: None
+except:
+ ...
+version = model_version = os.environ.get("version", "v2")
+path_sovits_v3 = "GPT_SoVITS/pretrained_models/s2Gv3.pth"
+path_sovits_v4 = "GPT_SoVITS/pretrained_models/gsv-v4-pretrained/s2Gv4.pth"
+is_exist_s2gv3 = os.path.exists(path_sovits_v3)
+is_exist_s2gv4 = os.path.exists(path_sovits_v4)
+pretrained_sovits_name = [
+ "GPT_SoVITS/pretrained_models/s2G488k.pth",
+ "GPT_SoVITS/pretrained_models/gsv-v2final-pretrained/s2G2333k.pth",
+ "GPT_SoVITS/pretrained_models/s2Gv3.pth",
+ "GPT_SoVITS/pretrained_models/gsv-v4-pretrained/s2Gv4.pth",
+]
+pretrained_gpt_name = [
+ "GPT_SoVITS/pretrained_models/s1bert25hz-2kh-longer-epoch=68e-step=50232.ckpt",
+ "GPT_SoVITS/pretrained_models/gsv-v2final-pretrained/s1bert25hz-5kh-longer-epoch=12-step=369668.ckpt",
+ "GPT_SoVITS/pretrained_models/s1v3.ckpt",
+ "GPT_SoVITS/pretrained_models/s1v3.ckpt",
+]
+
+
+_ = [[], []]
+for i in range(4):
+ if os.path.exists(pretrained_gpt_name[i]):
+ _[0].append(pretrained_gpt_name[i])
+ if os.path.exists(pretrained_sovits_name[i]):
+ _[-1].append(pretrained_sovits_name[i])
+pretrained_gpt_name, pretrained_sovits_name = _
+
+
+if os.path.exists("./weight.json"):
+ pass
+else:
+ with open("./weight.json", "w", encoding="utf-8") as file:
+ json.dump({"GPT": {}, "SoVITS": {}}, file)
+
+with open("./weight.json", "r", encoding="utf-8") as file:
+ weight_data = file.read()
+ weight_data = json.loads(weight_data)
+ gpt_path = os.environ.get("gpt_path", weight_data.get("GPT", {}).get(version, pretrained_gpt_name))
+ sovits_path = os.environ.get("sovits_path", weight_data.get("SoVITS", {}).get(version, pretrained_sovits_name))
+ if isinstance(gpt_path, list):
+ gpt_path = gpt_path[0]
+ if isinstance(sovits_path, list):
+ sovits_path = sovits_path[0]
+
+# gpt_path = os.environ.get(
+# "gpt_path", pretrained_gpt_name
+# )
+# sovits_path = os.environ.get("sovits_path", pretrained_sovits_name)
+cnhubert_base_path = os.environ.get("cnhubert_base_path", "GPT_SoVITS/pretrained_models/chinese-hubert-base")
+bert_path = os.environ.get("bert_path", "GPT_SoVITS/pretrained_models/chinese-roberta-wwm-ext-large")
+infer_ttswebui = os.environ.get("infer_ttswebui", 9872)
+infer_ttswebui = int(infer_ttswebui)
+is_share = os.environ.get("is_share", "False")
+is_share = eval(is_share)
+if "_CUDA_VISIBLE_DEVICES" in os.environ:
+ os.environ["CUDA_VISIBLE_DEVICES"] = os.environ["_CUDA_VISIBLE_DEVICES"]
+is_half = eval(os.environ.get("is_half", "True")) and torch.cuda.is_available()
+# is_half=False
+punctuation = set(["!", "?", "…", ",", ".", "-", " "])
+import gradio as gr
+import librosa
+import numpy as np
+from feature_extractor import cnhubert
+from transformers import AutoModelForMaskedLM, AutoTokenizer
+
+cnhubert.cnhubert_base_path = cnhubert_base_path
+
+import random
+
+from GPT_SoVITS.module.models import SynthesizerTrn, SynthesizerTrnV3,Generator
+
+
+def set_seed(seed):
+ if seed == -1:
+ seed = random.randint(0, 1000000)
+ seed = int(seed)
+ random.seed(seed)
+ os.environ["PYTHONHASHSEED"] = str(seed)
+ np.random.seed(seed)
+ torch.manual_seed(seed)
+ torch.cuda.manual_seed(seed)
+
+
+# set_seed(42)
+
+from time import time as ttime
+
+from AR.models.t2s_lightning_module import Text2SemanticLightningModule
+from peft import LoraConfig, get_peft_model
+from text import cleaned_text_to_sequence
+from text.cleaner import clean_text
+
+from tools.i18n.i18n import I18nAuto, scan_language_list
+
+language = os.environ.get("language", "Auto")
+language = sys.argv[-1] if sys.argv[-1] in scan_language_list() else language
+i18n = I18nAuto(language=language)
+
+# os.environ['PYTORCH_ENABLE_MPS_FALLBACK'] = '1' # 确保直接启动推理UI时也能够设置。
+
+if torch.cuda.is_available():
+ device = "cuda"
+else:
+ device = "cpu"
+
+dict_language_v1 = {
+ i18n("中文"): "all_zh", # 全部按中文识别
+ i18n("英文"): "en", # 全部按英文识别#######不变
+ i18n("日文"): "all_ja", # 全部按日文识别
+ i18n("中英混合"): "zh", # 按中英混合识别####不变
+ i18n("日英混合"): "ja", # 按日英混合识别####不变
+ i18n("多语种混合"): "auto", # 多语种启动切分识别语种
+}
+dict_language_v2 = {
+ i18n("中文"): "all_zh", # 全部按中文识别
+ i18n("英文"): "en", # 全部按英文识别#######不变
+ i18n("日文"): "all_ja", # 全部按日文识别
+ i18n("粤语"): "all_yue", # 全部按中文识别
+ i18n("韩文"): "all_ko", # 全部按韩文识别
+ i18n("中英混合"): "zh", # 按中英混合识别####不变
+ i18n("日英混合"): "ja", # 按日英混合识别####不变
+ i18n("粤英混合"): "yue", # 按粤英混合识别####不变
+ i18n("韩英混合"): "ko", # 按韩英混合识别####不变
+ i18n("多语种混合"): "auto", # 多语种启动切分识别语种
+ i18n("多语种混合(粤语)"): "auto_yue", # 多语种启动切分识别语种
+}
+dict_language = dict_language_v1 if version == "v1" else dict_language_v2
+
+tokenizer = AutoTokenizer.from_pretrained(bert_path)
+bert_model = AutoModelForMaskedLM.from_pretrained(bert_path)
+if is_half == True:
+ bert_model = bert_model.half().to(device)
+else:
+ bert_model = bert_model.to(device)
+
+
+def get_bert_feature(text, word2ph):
+ with torch.no_grad():
+ inputs = tokenizer(text, return_tensors="pt")
+ for i in inputs:
+ inputs[i] = inputs[i].to(device)
+ res = bert_model(**inputs, output_hidden_states=True)
+ res = torch.cat(res["hidden_states"][-3:-2], -1)[0].cpu()[1:-1]
+ assert len(word2ph) == len(text)
+ phone_level_feature = []
+ for i in range(len(word2ph)):
+ repeat_feature = res[i].repeat(word2ph[i], 1)
+ phone_level_feature.append(repeat_feature)
+ phone_level_feature = torch.cat(phone_level_feature, dim=0)
+ return phone_level_feature.T
+
+
+class DictToAttrRecursive(dict):
+ def __init__(self, input_dict):
+ super().__init__(input_dict)
+ for key, value in input_dict.items():
+ if isinstance(value, dict):
+ value = DictToAttrRecursive(value)
+ self[key] = value
+ setattr(self, key, value)
+
+ def __getattr__(self, item):
+ try:
+ return self[item]
+ except KeyError:
+ raise AttributeError(f"Attribute {item} not found")
+
+ def __setattr__(self, key, value):
+ if isinstance(value, dict):
+ value = DictToAttrRecursive(value)
+ super(DictToAttrRecursive, self).__setitem__(key, value)
+ super().__setattr__(key, value)
+
+ def __delattr__(self, item):
+ try:
+ del self[item]
+ except KeyError:
+ raise AttributeError(f"Attribute {item} not found")
+
+
+ssl_model = cnhubert.get_model()
+if is_half == True:
+ ssl_model = ssl_model.half().to(device)
+else:
+ ssl_model = ssl_model.to(device)
+
+resample_transform_dict = {}
+
+
+def resample(audio_tensor, sr0,sr1):
+ global resample_transform_dict
+ key="%s-%s"%(sr0,sr1)
+ if key not in resample_transform_dict:
+ resample_transform_dict[key] = torchaudio.transforms.Resample(sr0, sr1).to(device)
+ return resample_transform_dict[key](audio_tensor)
+
+
+###todo:put them to process_ckpt and modify my_save func (save sovits weights), gpt save weights use my_save in process_ckpt
+# symbol_version-model_version-if_lora_v3
+from process_ckpt import get_sovits_version_from_path_fast, load_sovits_new
+
+v3v4set={"v3","v4"}
+def change_sovits_weights(sovits_path, prompt_language=None, text_language=None):
+ global vq_model, hps, version, model_version, dict_language, if_lora_v3
+ version, model_version, if_lora_v3 = get_sovits_version_from_path_fast(sovits_path)
+ print(sovits_path,version, model_version, if_lora_v3)
+ is_exist=is_exist_s2gv3 if model_version=="v3"else is_exist_s2gv4
+ if if_lora_v3 == True and is_exist == False:
+ info = "GPT_SoVITS/pretrained_models/s2Gv3.pth" + i18n("SoVITS V3 底模缺失,无法加载相应 LoRA 权重")
+ gr.Warning(info)
+ raise FileExistsError(info)
+ dict_language = dict_language_v1 if version == "v1" else dict_language_v2
+ if prompt_language is not None and text_language is not None:
+ if prompt_language in list(dict_language.keys()):
+ prompt_text_update, prompt_language_update = (
+ {"__type__": "update"},
+ {"__type__": "update", "value": prompt_language},
+ )
+ else:
+ prompt_text_update = {"__type__": "update", "value": ""}
+ prompt_language_update = {"__type__": "update", "value": i18n("中文")}
+ if text_language in list(dict_language.keys()):
+ text_update, text_language_update = {"__type__": "update"}, {"__type__": "update", "value": text_language}
+ else:
+ text_update = {"__type__": "update", "value": ""}
+ text_language_update = {"__type__": "update", "value": i18n("中文")}
+ if model_version in v3v4set:
+ visible_sample_steps = True
+ visible_inp_refs = False
+ else:
+ visible_sample_steps = False
+ visible_inp_refs = True
+ yield (
+ {"__type__": "update", "choices": list(dict_language.keys())},
+ {"__type__": "update", "choices": list(dict_language.keys())},
+ prompt_text_update,
+ prompt_language_update,
+ text_update,
+ text_language_update,
+ {"__type__": "update", "visible": visible_sample_steps, "value": 32 if model_version=="v3"else 8,"choices":[4, 8, 16, 32,64,128]if model_version=="v3"else [4, 8, 16, 32]},
+ {"__type__": "update", "visible": visible_inp_refs},
+ {"__type__": "update", "value": False, "interactive": True if model_version not in v3v4set else False},
+ {"__type__": "update", "visible": True if model_version =="v3" else False},
+ {"__type__": "update", "value": i18n("模型加载中,请等待"), "interactive": False},
+ )
+
+ dict_s2 = load_sovits_new(sovits_path)
+ hps = dict_s2["config"]
+ hps = DictToAttrRecursive(hps)
+ hps.model.semantic_frame_rate = "25hz"
+ if "enc_p.text_embedding.weight" not in dict_s2["weight"]:
+ hps.model.version = "v2" # v3model,v2sybomls
+ elif dict_s2["weight"]["enc_p.text_embedding.weight"].shape[0] == 322:
+ hps.model.version = "v1"
+ else:
+ hps.model.version = "v2"
+ version = hps.model.version
+ # print("sovits版本:",hps.model.version)
+ if model_version not in v3v4set:
+ vq_model = SynthesizerTrn(
+ hps.data.filter_length // 2 + 1,
+ hps.train.segment_size // hps.data.hop_length,
+ n_speakers=hps.data.n_speakers,
+ **hps.model,
+ )
+ model_version = version
+ else:
+ vq_model = SynthesizerTrnV3(
+ hps.data.filter_length // 2 + 1,
+ hps.train.segment_size // hps.data.hop_length,
+ n_speakers=hps.data.n_speakers,
+ **hps.model,
+ )
+ if "pretrained" not in sovits_path:
+ try:
+ del vq_model.enc_q
+ except:
+ pass
+ if is_half == True:
+ vq_model = vq_model.half().to(device)
+ else:
+ vq_model = vq_model.to(device)
+ vq_model.eval()
+ if if_lora_v3 == False:
+ print("loading sovits_%s" % model_version, vq_model.load_state_dict(dict_s2["weight"], strict=False))
+ else:
+ path_sovits = path_sovits_v3 if model_version == "v3" else path_sovits_v4
+ print(
+ "loading sovits_%spretrained_G"%model_version,
+ vq_model.load_state_dict(load_sovits_new(path_sovits)["weight"], strict=False),
+ )
+ lora_rank = dict_s2["lora_rank"]
+ lora_config = LoraConfig(
+ target_modules=["to_k", "to_q", "to_v", "to_out.0"],
+ r=lora_rank,
+ lora_alpha=lora_rank,
+ init_lora_weights=True,
+ )
+ vq_model.cfm = get_peft_model(vq_model.cfm, lora_config)
+ print("loading sovits_%s_lora%s" % (model_version,lora_rank))
+ vq_model.load_state_dict(dict_s2["weight"], strict=False)
+ vq_model.cfm = vq_model.cfm.merge_and_unload()
+ # torch.save(vq_model.state_dict(),"merge_win.pth")
+ vq_model.eval()
+
+ yield (
+ {"__type__": "update", "choices": list(dict_language.keys())},
+ {"__type__": "update", "choices": list(dict_language.keys())},
+ prompt_text_update,
+ prompt_language_update,
+ text_update,
+ text_language_update,
+ {"__type__": "update", "visible": visible_sample_steps, "value":32 if model_version=="v3"else 8,"choices":[4, 8, 16, 32,64,128]if model_version=="v3"else [4, 8, 16, 32]},
+ {"__type__": "update", "visible": visible_inp_refs},
+ {"__type__": "update", "value": False, "interactive": True if model_version not in v3v4set else False},
+ {"__type__": "update", "visible": True if model_version =="v3" else False},
+ {"__type__": "update", "value": i18n("合成语音"), "interactive": True},
+ )
+ with open("./weight.json") as f:
+ data = f.read()
+ data = json.loads(data)
+ data["SoVITS"][version] = sovits_path
+ with open("./weight.json", "w") as f:
+ f.write(json.dumps(data))
+
+
+try:
+ next(change_sovits_weights(sovits_path))
+except:
+ pass
+
+
+def change_gpt_weights(gpt_path):
+ global hz, max_sec, t2s_model, config
+ hz = 50
+ dict_s1 = torch.load(gpt_path, map_location="cpu")
+ config = dict_s1["config"]
+ max_sec = config["data"]["max_sec"]
+ t2s_model = Text2SemanticLightningModule(config, "****", is_train=False)
+ t2s_model.load_state_dict(dict_s1["weight"])
+ if is_half == True:
+ t2s_model = t2s_model.half()
+ t2s_model = t2s_model.to(device)
+ t2s_model.eval()
+ # total = sum([param.nelement() for param in t2s_model.parameters()])
+ # print("Number of parameter: %.2fM" % (total / 1e6))
+ with open("./weight.json") as f:
+ data = f.read()
+ data = json.loads(data)
+ data["GPT"][version] = gpt_path
+ with open("./weight.json", "w") as f:
+ f.write(json.dumps(data))
+
+
+change_gpt_weights(gpt_path)
+os.environ["HF_ENDPOINT"] = "https://hf-mirror.com"
+import torch
+
+now_dir = os.getcwd()
+
+
+def init_bigvgan():
+ global bigvgan_model,hifigan_model
+ from BigVGAN import bigvgan
+
+ bigvgan_model = bigvgan.BigVGAN.from_pretrained(
+ "%s/GPT_SoVITS/pretrained_models/models--nvidia--bigvgan_v2_24khz_100band_256x" % (now_dir,),
+ use_cuda_kernel=False,
+ ) # if True, RuntimeError: Ninja is required to load C++ extensions
+ # remove weight norm in the model and set to eval mode
+ bigvgan_model.remove_weight_norm()
+ bigvgan_model = bigvgan_model.eval()
+ if hifigan_model:
+ hifigan_model=hifigan_model.cpu()
+ hifigan_model=None
+ try:torch.cuda.empty_cache()
+ except:pass
+ if is_half == True:
+ bigvgan_model = bigvgan_model.half().to(device)
+ else:
+ bigvgan_model = bigvgan_model.to(device)
+
+def init_hifigan():
+ global hifigan_model,bigvgan_model
+ hifigan_model = Generator(
+ initial_channel=100,
+ resblock="1",
+ resblock_kernel_sizes=[3, 7, 11],
+ resblock_dilation_sizes=[[1, 3, 5], [1, 3, 5], [1, 3, 5]],
+ upsample_rates=[10, 6, 2, 2, 2],
+ upsample_initial_channel=512,
+ upsample_kernel_sizes=[20, 12, 4, 4, 4],
+ gin_channels=0, is_bias=True
+ )
+ hifigan_model.eval()
+ hifigan_model.remove_weight_norm()
+ state_dict_g = torch.load("%s/GPT_SoVITS/pretrained_models/gsv-v4-pretrained/vocoder.pth" % (now_dir,), map_location="cpu")
+ print("loading vocoder",hifigan_model.load_state_dict(state_dict_g))
+ if bigvgan_model:
+ bigvgan_model=bigvgan_model.cpu()
+ bigvgan_model=None
+ try:torch.cuda.empty_cache()
+ except:pass
+ if is_half == True:
+ hifigan_model = hifigan_model.half().to(device)
+ else:
+ hifigan_model = hifigan_model.to(device)
+
+bigvgan_model=hifigan_model=None
+if model_version=="v3":
+ init_bigvgan()
+if model_version=="v4":
+ init_hifigan()
+
+
+def get_spepc(hps, filename):
+ # audio = load_audio(filename, int(hps.data.sampling_rate))
+ audio, sampling_rate = librosa.load(filename, sr=int(hps.data.sampling_rate))
+ audio = torch.FloatTensor(audio)
+ maxx = audio.abs().max()
+ if maxx > 1:
+ audio /= min(2, maxx)
+ audio_norm = audio
+ audio_norm = audio_norm.unsqueeze(0)
+ spec = spectrogram_torch(
+ audio_norm,
+ hps.data.filter_length,
+ hps.data.sampling_rate,
+ hps.data.hop_length,
+ hps.data.win_length,
+ center=False,
+ )
+ return spec
+
+
+def clean_text_inf(text, language, version):
+ language = language.replace("all_", "")
+ phones, word2ph, norm_text = clean_text(text, language, version)
+ phones = cleaned_text_to_sequence(phones, version)
+ return phones, word2ph, norm_text
+
+
+dtype = torch.float16 if is_half == True else torch.float32
+
+
+def get_bert_inf(phones, word2ph, norm_text, language):
+ language = language.replace("all_", "")
+ if language == "zh":
+ bert = get_bert_feature(norm_text, word2ph).to(device) # .to(dtype)
+ else:
+ bert = torch.zeros(
+ (1024, len(phones)),
+ dtype=torch.float16 if is_half == True else torch.float32,
+ ).to(device)
+
+ return bert
+
+
+splits = {
+ ",",
+ "。",
+ "?",
+ "!",
+ ",",
+ ".",
+ "?",
+ "!",
+ "~",
+ ":",
+ ":",
+ "—",
+ "…",
+}
+
+
+def get_first(text):
+ pattern = "[" + "".join(re.escape(sep) for sep in splits) + "]"
+ text = re.split(pattern, text)[0].strip()
+ return text
+
+
+from text import chinese
+
+
+def get_phones_and_bert(text, language, version, final=False):
+ if language in {"en", "all_zh", "all_ja", "all_ko", "all_yue"}:
+ formattext = text
+ while " " in formattext:
+ formattext = formattext.replace(" ", " ")
+ if language == "all_zh":
+ if re.search(r"[A-Za-z]", formattext):
+ formattext = re.sub(r"[a-z]", lambda x: x.group(0).upper(), formattext)
+ formattext = chinese.mix_text_normalize(formattext)
+ return get_phones_and_bert(formattext, "zh", version)
+ else:
+ phones, word2ph, norm_text = clean_text_inf(formattext, language, version)
+ bert = get_bert_feature(norm_text, word2ph).to(device)
+ elif language == "all_yue" and re.search(r"[A-Za-z]", formattext):
+ formattext = re.sub(r"[a-z]", lambda x: x.group(0).upper(), formattext)
+ formattext = chinese.mix_text_normalize(formattext)
+ return get_phones_and_bert(formattext, "yue", version)
+ else:
+ phones, word2ph, norm_text = clean_text_inf(formattext, language, version)
+ bert = torch.zeros(
+ (1024, len(phones)),
+ dtype=torch.float16 if is_half == True else torch.float32,
+ ).to(device)
+ elif language in {"zh", "ja", "ko", "yue", "auto", "auto_yue"}:
+ textlist = []
+ langlist = []
+ if language == "auto":
+ for tmp in LangSegmenter.getTexts(text):
+ langlist.append(tmp["lang"])
+ textlist.append(tmp["text"])
+ elif language == "auto_yue":
+ for tmp in LangSegmenter.getTexts(text):
+ if tmp["lang"] == "zh":
+ tmp["lang"] = "yue"
+ langlist.append(tmp["lang"])
+ textlist.append(tmp["text"])
+ else:
+ for tmp in LangSegmenter.getTexts(text):
+ if tmp["lang"] == "en":
+ langlist.append(tmp["lang"])
+ else:
+ # 因无法区别中日韩文汉字,以用户输入为准
+ langlist.append(language)
+ textlist.append(tmp["text"])
+ print(textlist)
+ print(langlist)
+ phones_list = []
+ bert_list = []
+ norm_text_list = []
+ for i in range(len(textlist)):
+ lang = langlist[i]
+ phones, word2ph, norm_text = clean_text_inf(textlist[i], lang, version)
+ bert = get_bert_inf(phones, word2ph, norm_text, lang)
+ phones_list.append(phones)
+ norm_text_list.append(norm_text)
+ bert_list.append(bert)
+ bert = torch.cat(bert_list, dim=1)
+ phones = sum(phones_list, [])
+ norm_text = "".join(norm_text_list)
+
+ if not final and len(phones) < 6:
+ return get_phones_and_bert("." + text, language, version, final=True)
+
+ return phones, bert.to(dtype), norm_text
+
+
+from module.mel_processing import mel_spectrogram_torch, spectrogram_torch
+
+spec_min = -12
+spec_max = 2
+
+
+def norm_spec(x):
+ return (x - spec_min) / (spec_max - spec_min) * 2 - 1
+
+
+def denorm_spec(x):
+ return (x + 1) / 2 * (spec_max - spec_min) + spec_min
+
+
+mel_fn = lambda x: mel_spectrogram_torch(
+ x,
+ **{
+ "n_fft": 1024,
+ "win_size": 1024,
+ "hop_size": 256,
+ "num_mels": 100,
+ "sampling_rate": 24000,
+ "fmin": 0,
+ "fmax": None,
+ "center": False,
+ },
+)
+mel_fn_v4 = lambda x: mel_spectrogram_torch(
+ x,
+ **{
+ "n_fft": 1280,
+ "win_size": 1280,
+ "hop_size": 320,
+ "num_mels": 100,
+ "sampling_rate": 32000,
+ "fmin": 0,
+ "fmax": None,
+ "center": False,
+ },
+)
+
+
+def merge_short_text_in_array(texts, threshold):
+ if (len(texts)) < 2:
+ return texts
+ result = []
+ text = ""
+ for ele in texts:
+ text += ele
+ if len(text) >= threshold:
+ result.append(text)
+ text = ""
+ if len(text) > 0:
+ if len(result) == 0:
+ result.append(text)
+ else:
+ result[len(result) - 1] += text
+ return result
+
+
+sr_model = None
+
+
+def audio_sr(audio, sr):
+ global sr_model
+ if sr_model == None:
+ from tools.audio_sr import AP_BWE
+
+ try:
+ sr_model = AP_BWE(device, DictToAttrRecursive)
+ except FileNotFoundError:
+ gr.Warning(i18n("你没有下载超分模型的参数,因此不进行超分。如想超分请先参照教程把文件下载好"))
+ return audio.cpu().detach().numpy(), sr
+ return sr_model(audio, sr)
+
+
+##ref_wav_path+prompt_text+prompt_language+text(单个)+text_language+top_k+top_p+temperature
+# cache_tokens={}#暂未实现清理机制
+cache = {}
+
+
+def get_tts_wav(
+ ref_wav_path,
+ prompt_text,
+ prompt_language,
+ text,
+ text_language,
+ how_to_cut=i18n("不切"),
+ top_k=20,
+ top_p=0.6,
+ temperature=0.6,
+ ref_free=False,
+ speed=1,
+ if_freeze=False,
+ inp_refs=None,
+ sample_steps=8,
+ if_sr=False,
+ pause_second=0.3,
+):
+ global cache
+ if ref_wav_path:
+ pass
+ else:
+ gr.Warning(i18n("请上传参考音频"))
+ if text:
+ pass
+ else:
+ gr.Warning(i18n("请填入推理文本"))
+ t = []
+ if prompt_text is None or len(prompt_text) == 0:
+ ref_free = True
+ if model_version in v3v4set:
+ ref_free = False # s2v3暂不支持ref_free
+ else:
+ if_sr = False
+ t0 = ttime()
+ prompt_language = dict_language[prompt_language]
+ text_language = dict_language[text_language]
+
+ if not ref_free:
+ prompt_text = prompt_text.strip("\n")
+ if prompt_text[-1] not in splits:
+ prompt_text += "。" if prompt_language != "en" else "."
+ print(i18n("实际输入的参考文本:"), prompt_text)
+ text = text.strip("\n")
+ # if (text[0] not in splits and len(get_first(text)) < 4): text = "。" + text if text_language != "en" else "." + text
+
+ print(i18n("实际输入的目标文本:"), text)
+ zero_wav = np.zeros(
+ int(hps.data.sampling_rate * pause_second),
+ dtype=np.float16 if is_half == True else np.float32,
+ )
+ zero_wav_torch = torch.from_numpy(zero_wav)
+ if is_half == True:
+ zero_wav_torch = zero_wav_torch.half().to(device)
+ else:
+ zero_wav_torch = zero_wav_torch.to(device)
+ if not ref_free:
+ with torch.no_grad():
+ wav16k, sr = librosa.load(ref_wav_path, sr=16000)
+ if wav16k.shape[0] > 160000 or wav16k.shape[0] < 48000:
+ gr.Warning(i18n("参考音频在3~10秒范围外,请更换!"))
+ raise OSError(i18n("参考音频在3~10秒范围外,请更换!"))
+ wav16k = torch.from_numpy(wav16k)
+ if is_half == True:
+ wav16k = wav16k.half().to(device)
+ else:
+ wav16k = wav16k.to(device)
+ wav16k = torch.cat([wav16k, zero_wav_torch])
+ ssl_content = ssl_model.model(wav16k.unsqueeze(0))["last_hidden_state"].transpose(1, 2) # .float()
+ codes = vq_model.extract_latent(ssl_content)
+ prompt_semantic = codes[0, 0]
+ prompt = prompt_semantic.unsqueeze(0).to(device)
+
+ t1 = ttime()
+ t.append(t1 - t0)
+
+ if how_to_cut == i18n("凑四句一切"):
+ text = cut1(text)
+ elif how_to_cut == i18n("凑50字一切"):
+ text = cut2(text)
+ elif how_to_cut == i18n("按中文句号。切"):
+ text = cut3(text)
+ elif how_to_cut == i18n("按英文句号.切"):
+ text = cut4(text)
+ elif how_to_cut == i18n("按标点符号切"):
+ text = cut5(text)
+ while "\n\n" in text:
+ text = text.replace("\n\n", "\n")
+ print(i18n("实际输入的目标文本(切句后):"), text)
+ texts = text.split("\n")
+ texts = process_text(texts)
+ texts = merge_short_text_in_array(texts, 5)
+ audio_opt = []
+ ###s2v3暂不支持ref_free
+ if not ref_free:
+ phones1, bert1, norm_text1 = get_phones_and_bert(prompt_text, prompt_language, version)
+
+ for i_text, text in enumerate(texts):
+ # 解决输入目标文本的空行导致报错的问题
+ if len(text.strip()) == 0:
+ continue
+ if text[-1] not in splits:
+ text += "。" if text_language != "en" else "."
+ print(i18n("实际输入的目标文本(每句):"), text)
+ phones2, bert2, norm_text2 = get_phones_and_bert(text, text_language, version)
+ print(i18n("前端处理后的文本(每句):"), norm_text2)
+ if not ref_free:
+ bert = torch.cat([bert1, bert2], 1)
+ all_phoneme_ids = torch.LongTensor(phones1 + phones2).to(device).unsqueeze(0)
+ else:
+ bert = bert2
+ all_phoneme_ids = torch.LongTensor(phones2).to(device).unsqueeze(0)
+
+ bert = bert.to(device).unsqueeze(0)
+ all_phoneme_len = torch.tensor([all_phoneme_ids.shape[-1]]).to(device)
+
+ t2 = ttime()
+ # cache_key="%s-%s-%s-%s-%s-%s-%s-%s"%(ref_wav_path,prompt_text,prompt_language,text,text_language,top_k,top_p,temperature)
+ # print(cache.keys(),if_freeze)
+ if i_text in cache and if_freeze == True:
+ pred_semantic = cache[i_text]
+ else:
+ with torch.no_grad():
+ pred_semantic, idx = t2s_model.model.infer_panel(
+ all_phoneme_ids,
+ all_phoneme_len,
+ None if ref_free else prompt,
+ bert,
+ # prompt_phone_len=ph_offset,
+ top_k=top_k,
+ top_p=top_p,
+ temperature=temperature,
+ early_stop_num=hz * max_sec,
+ )
+ pred_semantic = pred_semantic[:, -idx:].unsqueeze(0)
+ cache[i_text] = pred_semantic
+ t3 = ttime()
+ ###v3不存在以下逻辑和inp_refs
+ if model_version not in v3v4set:
+ refers = []
+ if inp_refs:
+ for path in inp_refs:
+ try:
+ refer = get_spepc(hps, path.name).to(dtype).to(device)
+ refers.append(refer)
+ except:
+ traceback.print_exc()
+ if len(refers) == 0:
+ refers = [get_spepc(hps, ref_wav_path).to(dtype).to(device)]
+ audio = vq_model.decode(
+ pred_semantic, torch.LongTensor(phones2).to(device).unsqueeze(0), refers, speed=speed
+ )[0][0] # .cpu().detach().numpy()
+ else:
+ refer = get_spepc(hps, ref_wav_path).to(device).to(dtype)
+ phoneme_ids0 = torch.LongTensor(phones1).to(device).unsqueeze(0)
+ phoneme_ids1 = torch.LongTensor(phones2).to(device).unsqueeze(0)
+ # print(11111111, phoneme_ids0, phoneme_ids1)
+ fea_ref, ge = vq_model.decode_encp(prompt.unsqueeze(0), phoneme_ids0, refer)
+ ref_audio, sr = torchaudio.load(ref_wav_path)
+ ref_audio = ref_audio.to(device).float()
+ if ref_audio.shape[0] == 2:
+ ref_audio = ref_audio.mean(0).unsqueeze(0)
+ tgt_sr=24000 if model_version=="v3"else 32000
+ if sr != tgt_sr:
+ ref_audio = resample(ref_audio, sr,tgt_sr)
+ # print("ref_audio",ref_audio.abs().mean())
+ mel2 = mel_fn(ref_audio)if model_version=="v3"else mel_fn_v4(ref_audio)
+ mel2 = norm_spec(mel2)
+ T_min = min(mel2.shape[2], fea_ref.shape[2])
+ mel2 = mel2[:, :, :T_min]
+ fea_ref = fea_ref[:, :, :T_min]
+ Tref=468 if model_version=="v3"else 500
+ Tchunk=934 if model_version=="v3"else 1000
+ if T_min > Tref:
+ mel2 = mel2[:, :, -Tref:]
+ fea_ref = fea_ref[:, :, -Tref:]
+ T_min = Tref
+ chunk_len = Tchunk - T_min
+ mel2 = mel2.to(dtype)
+ fea_todo, ge = vq_model.decode_encp(pred_semantic, phoneme_ids1, refer, ge, speed)
+ cfm_resss = []
+ idx = 0
+ while 1:
+ fea_todo_chunk = fea_todo[:, :, idx : idx + chunk_len]
+ if fea_todo_chunk.shape[-1] == 0:
+ break
+ idx += chunk_len
+ fea = torch.cat([fea_ref, fea_todo_chunk], 2).transpose(2, 1)
+ cfm_res = vq_model.cfm.inference(
+ fea, torch.LongTensor([fea.size(1)]).to(fea.device), mel2, sample_steps, inference_cfg_rate=0
+ )
+ cfm_res = cfm_res[:, :, mel2.shape[2] :]
+ mel2 = cfm_res[:, :, -T_min:]
+ fea_ref = fea_todo_chunk[:, :, -T_min:]
+ cfm_resss.append(cfm_res)
+ cfm_res = torch.cat(cfm_resss, 2)
+ cfm_res = denorm_spec(cfm_res)
+ if model_version=="v3":
+ if bigvgan_model == None:
+ init_bigvgan()
+ else:#v4
+ if hifigan_model == None:
+ init_hifigan()
+ vocoder_model=bigvgan_model if model_version=="v3"else hifigan_model
+ with torch.inference_mode():
+ wav_gen = vocoder_model(cfm_res)
+ audio = wav_gen[0][0] # .cpu().detach().numpy()
+ max_audio = torch.abs(audio).max() # 简单防止16bit爆音
+ if max_audio > 1:
+ audio = audio / max_audio
+ audio_opt.append(audio)
+ audio_opt.append(zero_wav_torch) # zero_wav
+ t4 = ttime()
+ t.extend([t2 - t1, t3 - t2, t4 - t3])
+ t1 = ttime()
+ print("%.3f\t%.3f\t%.3f\t%.3f" % (t[0], sum(t[1::3]), sum(t[2::3]), sum(t[3::3])))
+ audio_opt = torch.cat(audio_opt, 0) # np.concatenate
+ if model_version in {"v1","v2"}:opt_sr=32000
+ elif model_version=="v3":opt_sr=24000
+ else:opt_sr=48000#v4
+ if if_sr == True and opt_sr == 24000:
+ print(i18n("音频超分中"))
+ audio_opt, opt_sr = audio_sr(audio_opt.unsqueeze(0), opt_sr)
+ max_audio = np.abs(audio_opt).max()
+ if max_audio > 1:
+ audio_opt /= max_audio
+ else:
+ audio_opt = audio_opt.cpu().detach().numpy()
+ yield opt_sr, (audio_opt * 32767).astype(np.int16)
+
+
+def split(todo_text):
+ todo_text = todo_text.replace("……", "。").replace("——", ",")
+ if todo_text[-1] not in splits:
+ todo_text += "。"
+ i_split_head = i_split_tail = 0
+ len_text = len(todo_text)
+ todo_texts = []
+ while 1:
+ if i_split_head >= len_text:
+ break # 结尾一定有标点,所以直接跳出即可,最后一段在上次已加入
+ if todo_text[i_split_head] in splits:
+ i_split_head += 1
+ todo_texts.append(todo_text[i_split_tail:i_split_head])
+ i_split_tail = i_split_head
+ else:
+ i_split_head += 1
+ return todo_texts
+
+
+def cut1(inp):
+ inp = inp.strip("\n")
+ inps = split(inp)
+ split_idx = list(range(0, len(inps), 4))
+ split_idx[-1] = None
+ if len(split_idx) > 1:
+ opts = []
+ for idx in range(len(split_idx) - 1):
+ opts.append("".join(inps[split_idx[idx] : split_idx[idx + 1]]))
+ else:
+ opts = [inp]
+ opts = [item for item in opts if not set(item).issubset(punctuation)]
+ return "\n".join(opts)
+
+
+def cut2(inp):
+ inp = inp.strip("\n")
+ inps = split(inp)
+ if len(inps) < 2:
+ return inp
+ opts = []
+ summ = 0
+ tmp_str = ""
+ for i in range(len(inps)):
+ summ += len(inps[i])
+ tmp_str += inps[i]
+ if summ > 50:
+ summ = 0
+ opts.append(tmp_str)
+ tmp_str = ""
+ if tmp_str != "":
+ opts.append(tmp_str)
+ # print(opts)
+ if len(opts) > 1 and len(opts[-1]) < 50: ##如果最后一个太短了,和前一个合一起
+ opts[-2] = opts[-2] + opts[-1]
+ opts = opts[:-1]
+ opts = [item for item in opts if not set(item).issubset(punctuation)]
+ return "\n".join(opts)
+
+
+def cut3(inp):
+ inp = inp.strip("\n")
+ opts = ["%s" % item for item in inp.strip("。").split("。")]
+ opts = [item for item in opts if not set(item).issubset(punctuation)]
+ return "\n".join(opts)
+
+
+def cut4(inp):
+ inp = inp.strip("\n")
+ opts = re.split(r"(? 0 and i < len(inp) - 1 and inp[i - 1].isdigit() and inp[i + 1].isdigit():
+ items.append(char)
+ else:
+ items.append(char)
+ mergeitems.append("".join(items))
+ items = []
+ else:
+ items.append(char)
+
+ if items:
+ mergeitems.append("".join(items))
+
+ opt = [item for item in mergeitems if not set(item).issubset(punds)]
+ return "\n".join(opt)
+
+
+def custom_sort_key(s):
+ # 使用正则表达式提取字符串中的数字部分和非数字部分
+ parts = re.split("(\d+)", s)
+ # 将数字部分转换为整数,非数字部分保持不变
+ parts = [int(part) if part.isdigit() else part for part in parts]
+ return parts
+
+
+def process_text(texts):
+ _text = []
+ if all(text in [None, " ", "\n", ""] for text in texts):
+ raise ValueError(i18n("请输入有效文本"))
+ for text in texts:
+ if text in [None, " ", ""]:
+ pass
+ else:
+ _text.append(text)
+ return _text
+
+
+def change_choices():
+ SoVITS_names, GPT_names = get_weights_names(GPT_weight_root, SoVITS_weight_root)
+ return {"choices": sorted(SoVITS_names, key=custom_sort_key), "__type__": "update"}, {
+ "choices": sorted(GPT_names, key=custom_sort_key),
+ "__type__": "update",
+ }
+
+
+SoVITS_weight_root = ["SoVITS_weights", "SoVITS_weights_v2", "SoVITS_weights_v3", "SoVITS_weights_v4"]
+GPT_weight_root = ["GPT_weights", "GPT_weights_v2", "GPT_weights_v3", "GPT_weights_v4"]
+for path in SoVITS_weight_root + GPT_weight_root:
+ os.makedirs(path, exist_ok=True)
+
+
+def get_weights_names(GPT_weight_root, SoVITS_weight_root):
+ SoVITS_names = [i for i in pretrained_sovits_name]
+ for path in SoVITS_weight_root:
+ for name in os.listdir(path):
+ if name.endswith(".pth"):
+ SoVITS_names.append("%s/%s" % (path, name))
+ GPT_names = [i for i in pretrained_gpt_name]
+ for path in GPT_weight_root:
+ for name in os.listdir(path):
+ if name.endswith(".ckpt"):
+ GPT_names.append("%s/%s" % (path, name))
+ return SoVITS_names, GPT_names
+
+
+SoVITS_names, GPT_names = get_weights_names(GPT_weight_root, SoVITS_weight_root)
+
+
+def html_center(text, label="p"):
+ return f"""
+
+---
+
+## Features:
+
+1. **Zero-shot TTS:** Input a 5-second vocal sample and experience instant text-to-speech conversion.
+
+2. **Few-shot TTS:** Fine-tune the model with just 1 minute of training data for improved voice similarity and realism.
+
+3. **Cross-lingual Support:** Inference in languages different from the training dataset, currently supporting English, Japanese, Korean, Cantonese and Chinese.
+
+4. **WebUI Tools:** Integrated tools include voice accompaniment separation, automatic training set segmentation, Chinese ASR, and text labeling, assisting beginners in creating training datasets and GPT/SoVITS models.
+
+**Check out our [demo video](https://www.bilibili.com/video/BV12g4y1m7Uw) here!**
+
+Unseen speakers few-shot fine-tuning demo:
+
+https://github.com/RVC-Boss/GPT-SoVITS/assets/129054828/05bee1fa-bdd8-4d85-9350-80c060ab47fb
+
+**User guide: [简体中文](https://www.yuque.com/baicaigongchang1145haoyuangong/ib3g1e) | [English](https://rentry.co/GPT-SoVITS-guide#/)**
+
+## Installation
+
+For users in China, you can [click here](https://www.codewithgpu.com/i/RVC-Boss/GPT-SoVITS/GPT-SoVITS-Official) to use AutoDL Cloud Docker to experience the full functionality online.
+
+### Tested Environments
+
+| Python Version | PyTorch Version | Device |
+|----------------|------------------|-----------------|
+| Python 3.9 | PyTorch 2.0.1 | CUDA 11.8 |
+| Python 3.10.13 | PyTorch 2.1.2 | CUDA 12.3 |
+| Python 3.10.17 | PyTorch 2.5.1 | CUDA 12.4 |
+| Python 3.9 | PyTorch 2.5.1 | Apple silicon |
+| Python 3.11 | PyTorch 2.6.0 | Apple silicon |
+| Python 3.9 | PyTorch 2.2.2 | CPU |
+| Python 3.9 | PyTorch 2.8.0dev | CUDA12.8(for Nvidia50x0) |
+
+### Windows
+
+If you are a Windows user (tested with win>=10), you can [download the integrated package](https://huggingface.co/lj1995/GPT-SoVITS-windows-package/resolve/main/GPT-SoVITS-v3lora-20250228.7z?download=true) and double-click on _go-webui.bat_ to start GPT-SoVITS-WebUI.
+
+**Users in China can [download the package here](https://www.yuque.com/baicaigongchang1145haoyuangong/ib3g1e/dkxgpiy9zb96hob4#KTvnO).**
+
+### Linux
+
+```bash
+conda create -n GPTSoVits python=3.9
+conda activate GPTSoVits
+bash install.sh --source [--download-uvr5]
+```
+
+### macOS
+
+**Note: The models trained with GPUs on Macs result in significantly lower quality compared to those trained on other devices, so we are temporarily using CPUs instead.**
+
+1. Install Xcode command-line tools by running `xcode-select --install`.
+2. Install the program by running the following commands:
+
+```bash
+conda create -n GPTSoVits python=3.9
+conda activate GPTSoVits
+bash install.sh --source [--download-uvr5]
+```
+
+### Install Manually
+
+#### Install FFmpeg
+
+##### Conda Users
+
+```bash
+conda install ffmpeg
+```
+
+##### Ubuntu/Debian Users
+
+```bash
+sudo apt install ffmpeg
+sudo apt install libsox-dev
+conda install -c conda-forge 'ffmpeg<7'
+```
+
+##### Windows Users
+
+Download and place [ffmpeg.exe](https://huggingface.co/lj1995/VoiceConversionWebUI/blob/main/ffmpeg.exe) and [ffprobe.exe](https://huggingface.co/lj1995/VoiceConversionWebUI/blob/main/ffprobe.exe) in the GPT-SoVITS root.
+
+Install [Visual Studio 2017](https://aka.ms/vs/17/release/vc_redist.x86.exe) (Korean TTS Only)
+
+##### MacOS Users
+
+```bash
+brew install ffmpeg
+```
+
+#### Install Dependences
+
+```bash
+pip install -r extra-req.txt --no-deps
+pip install -r requirements.txt
+```
+
+### Using Docker
+
+#### docker-compose.yaml configuration
+
+0. Regarding image tags: Due to rapid updates in the codebase and the slow process of packaging and testing images, please check [Docker Hub](https://hub.docker.com/r/breakstring/gpt-sovits)(outdated) for the currently packaged latest images and select as per your situation, or alternatively, build locally using a Dockerfile according to your own needs.
+1. Environment Variables:
+ - is_half: Controls half-precision/double-precision. This is typically the cause if the content under the directories 4-cnhubert/5-wav32k is not generated correctly during the "SSL extracting" step. Adjust to True or False based on your actual situation.
+2. Volumes Configuration, The application's root directory inside the container is set to /workspace. The default docker-compose.yaml lists some practical examples for uploading/downloading content.
+3. shm_size: The default available memory for Docker Desktop on Windows is too small, which can cause abnormal operations. Adjust according to your own situation.
+4. Under the deploy section, GPU-related settings should be adjusted cautiously according to your system and actual circumstances.
+
+#### Running with docker compose
+
+```
+docker compose -f "docker-compose.yaml" up -d
+```
+
+#### Running with docker command
+
+As above, modify the corresponding parameters based on your actual situation, then run the following command:
+
+```
+docker run --rm -it --gpus=all --env=is_half=False --volume=G:\GPT-SoVITS-DockerTest\output:/workspace/output --volume=G:\GPT-SoVITS-DockerTest\logs:/workspace/logs --volume=G:\GPT-SoVITS-DockerTest\SoVITS_weights:/workspace/SoVITS_weights --workdir=/workspace -p 9880:9880 -p 9871:9871 -p 9872:9872 -p 9873:9873 -p 9874:9874 --shm-size="16G" -d breakstring/gpt-sovits:xxxxx
+```
+
+## Pretrained Models
+
+**If `install.sh` runs successfully, you may skip No.1,2,3**
+
+**Users in China can [download all these models here](https://www.yuque.com/baicaigongchang1145haoyuangong/ib3g1e/dkxgpiy9zb96hob4#nVNhX).**
+
+1. Download pretrained models from [GPT-SoVITS Models](https://huggingface.co/lj1995/GPT-SoVITS) and place them in `GPT_SoVITS/pretrained_models`.
+
+2. Download G2PW models from [G2PWModel.zip(HF)](https://huggingface.co/XXXXRT/GPT-SoVITS-Pretrained/resolve/main/G2PWModel.zip)| [G2PWModel.zip(ModelScope)](https://www.modelscope.cn/models/XXXXRT/GPT-SoVITS-Pretrained/resolve/master/G2PWModel.zip), unzip and rename to `G2PWModel`, and then place them in `GPT_SoVITS/text`.(Chinese TTS Only)
+
+3. For UVR5 (Vocals/Accompaniment Separation & Reverberation Removal, additionally), download models from [UVR5 Weights](https://huggingface.co/lj1995/VoiceConversionWebUI/tree/main/uvr5_weights) and place them in `tools/uvr5/uvr5_weights`.
+
+ - If you want to use `bs_roformer` or `mel_band_roformer` models for UVR5, you can manually download the model and corresponding configuration file, and put them in `tools/uvr5/uvr5_weights`. **Rename the model file and configuration file, ensure that the model and configuration files have the same and corresponding names except for the suffix**. In addition, the model and configuration file names **must include `roformer`** in order to be recognized as models of the roformer class.
+
+ - The suggestion is to **directly specify the model type** in the model name and configuration file name, such as `mel_mand_roformer`, `bs_roformer`. If not specified, the features will be compared from the configuration file to determine which type of model it is. For example, the model `bs_roformer_ep_368_sdr_12.9628.ckpt` and its corresponding configuration file `bs_roformer_ep_368_sdr_12.9628.yaml` are a pair, `kim_mel_band_roformer.ckpt` and `kim_mel_band_roformer.yaml` are also a pair.
+
+4. For Chinese ASR (additionally), download models from [Damo ASR Model](https://modelscope.cn/models/damo/speech_paraformer-large_asr_nat-zh-cn-16k-common-vocab8404-pytorch/files), [Damo VAD Model](https://modelscope.cn/models/damo/speech_fsmn_vad_zh-cn-16k-common-pytorch/files), and [Damo Punc Model](https://modelscope.cn/models/damo/punc_ct-transformer_zh-cn-common-vocab272727-pytorch/files) and place them in `tools/asr/models`.
+
+5. For English or Japanese ASR (additionally), download models from [Faster Whisper Large V3](https://huggingface.co/Systran/faster-whisper-large-v3) and place them in `tools/asr/models`. Also, [other models](https://huggingface.co/Systran) may have the similar effect with smaller disk footprint.
+
+## Dataset Format
+
+The TTS annotation .list file format:
+
+```
+vocal_path|speaker_name|language|text
+```
+
+Language dictionary:
+
+- 'zh': Chinese
+- 'ja': Japanese
+- 'en': English
+- 'ko': Korean
+- 'yue': Cantonese
+
+Example:
+
+```
+D:\GPT-SoVITS\xxx/xxx.wav|xxx|en|I like playing Genshin.
+```
+
+## Finetune and inference
+
+### Open WebUI
+
+#### Integrated Package Users
+
+Double-click `go-webui.bat`or use `go-webui.ps1`
+if you want to switch to V1,then double-click`go-webui-v1.bat` or use `go-webui-v1.ps1`
+
+#### Others
+
+```bash
+python webui.py
+```
+
+if you want to switch to V1,then
+
+```bash
+python webui.py v1
+```
+
+Or maunally switch version in WebUI
+
+### Finetune
+
+#### Path Auto-filling is now supported
+
+ 1. Fill in the audio path
+ 2. Slice the audio into small chunks
+ 3. Denoise(optinal)
+ 4. ASR
+ 5. Proofreading ASR transcriptions
+ 6. Go to the next Tab, then finetune the model
+
+### Open Inference WebUI
+
+#### Integrated Package Users
+
+Double-click `go-webui-v2.bat` or use `go-webui-v2.ps1` ,then open the inference webui at `1-GPT-SoVITS-TTS/1C-inference`
+
+#### Others
+
+```bash
+python GPT_SoVITS/inference_webui.py
+```
+
+OR
+
+```bash
+python webui.py
+```
+
+then open the inference webui at `1-GPT-SoVITS-TTS/1C-inference`
+
+## V2 Release Notes
+
+New Features:
+
+1. Support Korean and Cantonese
+
+2. An optimized text frontend
+
+3. Pre-trained model extended from 2k hours to 5k hours
+
+4. Improved synthesis quality for low-quality reference audio
+
+ [more details]()
+
+Use v2 from v1 environment:
+
+1. `pip install -r requirements.txt` to update some packages
+
+2. Clone the latest codes from github.
+
+3. Download v2 pretrained models from [huggingface](https://huggingface.co/lj1995/GPT-SoVITS/tree/main/gsv-v2final-pretrained) and put them into `GPT_SoVITS\pretrained_models\gsv-v2final-pretrained`.
+
+ Chinese v2 additional: [G2PWModel.zip(HF)](https://huggingface.co/XXXXRT/GPT-SoVITS-Pretrained/resolve/main/G2PWModel.zip)| [G2PWModel.zip(ModelScope)](https://www.modelscope.cn/models/XXXXRT/GPT-SoVITS-Pretrained/resolve/master/G2PWModel.zip)(Download G2PW models, unzip and rename to `G2PWModel`, and then place them in `GPT_SoVITS/text`.)
+
+## V3 Release Notes
+
+New Features:
+
+1. The timbre similarity is higher, requiring less training data to approximate the target speaker (the timbre similarity is significantly improved using the base model directly without fine-tuning).
+
+2. GPT model is more stable, with fewer repetitions and omissions, and it is easier to generate speech with richer emotional expression.
+
+ [more details]()
+
+Use v3 from v2 environment:
+
+1. `pip install -r requirements.txt` to update some packages
+
+2. Clone the latest codes from github.
+
+3. Download v3 pretrained models (s1v3.ckpt, s2Gv3.pth and models--nvidia--bigvgan_v2_24khz_100band_256x folder) from [huggingface](https://huggingface.co/lj1995/GPT-SoVITS/tree/main) and put them into `GPT_SoVITS\pretrained_models`.
+
+ additional: for Audio Super Resolution model, you can read [how to download](./tools/AP_BWE_main/24kto48k/readme.txt)
+
+## V4 Release Notes
+
+New Features:
+
+1. Version 4 fixes the issue of metallic artifacts in Version 3 caused by non-integer multiple upsampling, and natively outputs 48k audio to prevent muffled sound (whereas Version 3 only natively outputs 24k audio). The author considers Version 4 a direct replacement for Version 3, though further testing is still needed.
+ [more details]()
+
+Use v4 from v1/v2/v3 environment:
+
+1. `pip install -r requirements.txt` to update some packages
+
+2. Clone the latest codes from github.
+
+3. Download v4 pretrained models (gsv-v4-pretrained/s2v4.ckpt, and gsv-v4-pretrained/vocoder.pth) from [huggingface](https://huggingface.co/lj1995/GPT-SoVITS/tree/main) and put them into `GPT_SoVITS\pretrained_models`.
+
+## Todo List
+
+- [x] **High Priority:**
+
+ - [x] Localization in Japanese and English.
+ - [x] User guide.
+ - [x] Japanese and English dataset fine tune training.
+
+- [ ] **Features:**
+ - [x] Zero-shot voice conversion (5s) / few-shot voice conversion (1min).
+ - [x] TTS speaking speed control.
+ - [ ] ~~Enhanced TTS emotion control.~~ Maybe use pretrained finetuned preset GPT models for better emotion.
+ - [ ] Experiment with changing SoVITS token inputs to probability distribution of GPT vocabs (transformer latent).
+ - [x] Improve English and Japanese text frontend.
+ - [ ] Develop tiny and larger-sized TTS models.
+ - [x] Colab scripts.
+ - [x] Try expand training dataset (2k hours -> 10k hours).
+ - [x] better sovits base model (enhanced audio quality)
+ - [ ] model mix
+
+## (Additional) Method for running from the command line
+
+Use the command line to open the WebUI for UVR5
+
+```
+python tools/uvr5/webui.py ""
+```
+
+
+
+This is how the audio segmentation of the dataset is done using the command line
+
+```
+python audio_slicer.py \
+ --input_path "" \
+ --output_root "" \
+ --threshold \
+ --min_length \
+ --min_interval
+ --hop_size
+```
+
+This is how dataset ASR processing is done using the command line(Only Chinese)
+
+```
+python tools/asr/funasr_asr.py -i -o