2 Commits

Author SHA1 Message Date
Nemo
3713d64cc7 update img 2021-10-01 00:24:59 +08:00
Nemo
5e7cc82373 web tool box update UI 2021-09-30 23:48:45 +08:00
25 changed files with 142 additions and 537 deletions

4
.gitignore vendored
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@@ -17,7 +17,5 @@
*.sh
synthesizer/saved_models/*
vocoder/saved_models/*
encoder/saved_models/*
cp_hifigan/*
!vocoder/saved_models/pretrained/*
!encoder/saved_models/pretrained.pt
!vocoder/saved_models/pretrained/*

18
.vscode/launch.json vendored
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@@ -17,7 +17,7 @@
"request": "launch",
"program": "vocoder_preprocess.py",
"console": "integratedTerminal",
"args": ["..\\audiodata"]
"args": ["..\\..\\chs1"]
},
{
"name": "Python: Vocoder Train",
@@ -25,23 +25,15 @@
"request": "launch",
"program": "vocoder_train.py",
"console": "integratedTerminal",
"args": ["dev", "..\\audiodata"]
"args": ["dev", "..\\..\\chs1"]
},
{
"name": "Python: Demo Box",
"name": "Python: demo box",
"type": "python",
"request": "launch",
"program": "demo_toolbox.py",
"console": "integratedTerminal",
"args": ["-d","..\\audiodata"]
},
{
"name": "Python: Synth Train",
"type": "python",
"request": "launch",
"program": "synthesizer_train.py",
"console": "integratedTerminal",
"args": ["my_run", "..\\"]
},
"args": ["-d", "..\\..\\chs"]
}
]
}

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@@ -5,10 +5,10 @@
### [English](README.md) | 中文
### [DEMO VIDEO](https://www.bilibili.com/video/BV17Q4y1B7mY/) | [Wiki教程](https://github.com/babysor/MockingBird/wiki/Quick-Start-(Newbie)) [训练教程](https://vaj2fgg8yn.feishu.cn/docs/doccn7kAbr3SJz0KM0SIDJ0Xnhd)
### [DEMO VIDEO](https://www.bilibili.com/video/BV1sA411P7wM/)
## 特性
🌍 **中文** 支持普通话并使用多种中文数据集进行测试aidatatang_200zh, magicdata, aishell3, biaobei, MozillaCommonVoice, data_aishell
🌍 **中文** 支持普通话并使用多种中文数据集进行测试aidatatang_200zh, magicdata, aishell3 biaobeiMozillaCommonVoice 等
🤩 **PyTorch** 适用于 pytorch已在 1.9.0 版本(最新于 2021 年 8 月中测试GPU Tesla T4 和 GTX 2060
@@ -18,7 +18,6 @@
🌍 **Webserver Ready** 可伺服你的训练结果,供远程调用
## 开始
### 1. 安装要求
> 按照原始存储库测试您是否已准备好所有环境。
**Python 3.7 或更高版本** 需要运行工具箱。
@@ -35,10 +34,8 @@
#### 2.1 使用数据集自己训练合成器模型与2.2二选一)
* 下载 数据集并解压:确保您可以访问 *train* 文件夹中的所有音频文件(如.wav
* 进行音频和梅尔频谱图预处理:
`python pre.py <datasets_root> -d {dataset} -n {number}`
可传入参数
* -d`{dataset}` 指定数据集,支持 aidatatang_200zh, magicdata, aishell3, data_aishell, 不传默认为aidatatang_200zh
* -n `{number}` 指定并行数CPU 11770k + 32GB实测10没有问题
`python pre.py <datasets_root>`
传入参数 --dataset `{dataset}` 支持 aidatatang_200zh, magicdata, aishell3
> 假如你下载的 `aidatatang_200zh`文件放在D盘`train`文件路径为 `D:\data\aidatatang_200zh\corpus\train` , 你的`datasets_root`就是 `D:\data\`
* 训练合成器:
@@ -51,7 +48,7 @@
| 作者 | 下载链接 | 效果预览 | 信息 |
| --- | ----------- | ----- | ----- |
| 作者 | https://pan.baidu.com/s/11FrUYBmLrSs_cQ7s3JTlPQ [百度盘链接](https://pan.baidu.com/s/11FrUYBmLrSs_cQ7s3JTlPQ) 提取码:gdn5 | | 25k steps 用3个开源数据集混合训练
| 作者 | https://pan.baidu.com/s/1VHSKIbxXQejtxi2at9IrpA [百度盘链接](https://pan.baidu.com/s/1VHSKIbxXQejtxi2at9IrpA ) 提取码:i183 | | 200k steps 只用aidatatang_200zh
|@FawenYo | https://drive.google.com/file/d/1H-YGOUHpmqKxJ9FRc6vAjPuqQki24UbC/view?usp=sharing [百度盘链接](https://pan.baidu.com/s/1vSYXO4wsLyjnF3Unl-Xoxg) 提取码1024 | [input](https://github.com/babysor/MockingBird/wiki/audio/self_test.mp3) [output](https://github.com/babysor/MockingBird/wiki/audio/export.wav) | 200k steps 台湾口音
|@miven| https://pan.baidu.com/s/1PI-hM3sn5wbeChRryX-RCQ 提取码2021 | https://www.bilibili.com/video/BV1uh411B7AD/ | 150k steps 旧版需根据[issue](https://github.com/babysor/MockingBird/issues/37)修复
@@ -122,21 +119,15 @@
| URL | Designation | 标题 | 实现源码 |
| --- | ----------- | ----- | --------------------- |
| [1803.09017](https://arxiv.org/abs/1803.09017) | GlobalStyleToken (synthesizer)| Style Tokens: Unsupervised Style Modeling, Control and Transfer in End-to-End Speech Synthesis | 本代码库 |
| [2010.05646](https://arxiv.org/abs/2010.05646) | HiFi-GAN (vocoder)| Generative Adversarial Networks for Efficient and High Fidelity Speech Synthesis | 本代码库 |
|[**1806.04558**](https://arxiv.org/pdf/1806.04558.pdf) | SV2TTS | Transfer Learning from Speaker Verification to Multispeaker Text-To-Speech Synthesis | 本代码库 |
|[**1806.04558**](https://arxiv.org/pdf/1806.04558.pdf) | **SV2TTS** | **Transfer Learning from Speaker Verification to Multispeaker Text-To-Speech Synthesis** | This repo |
|[1802.08435](https://arxiv.org/pdf/1802.08435.pdf) | WaveRNN (vocoder) | Efficient Neural Audio Synthesis | [fatchord/WaveRNN](https://github.com/fatchord/WaveRNN) |
|[1703.10135](https://arxiv.org/pdf/1703.10135.pdf) | Tacotron (synthesizer) | Tacotron: Towards End-to-End Speech Synthesis | [fatchord/WaveRNN](https://github.com/fatchord/WaveRNN)
|[1710.10467](https://arxiv.org/pdf/1710.10467.pdf) | GE2E (encoder)| Generalized End-To-End Loss for Speaker Verification | 本代码库 |
## 常見問題(FQ&A)
#### 1.數據集哪裡下載?
| 数据集 | OpenSLR地址 | 其他源 (Google Drive, Baidu网盘等) |
| --- | ----------- | ---------------|
| aidatatang_200zh | [OpenSLR](http://www.openslr.org/62/) | [Google Drive](https://drive.google.com/file/d/110A11KZoVe7vy6kXlLb6zVPLb_J91I_t/view?usp=sharing) |
| magicdata | [OpenSLR](http://www.openslr.org/68/) | [Google Drive (Dev set)](https://drive.google.com/file/d/1g5bWRUSNH68ycC6eNvtwh07nX3QhOOlo/view?usp=sharing) |
| aishell3 | [OpenSLR](https://www.openslr.org/93/) | [Google Drive](https://drive.google.com/file/d/1shYp_o4Z0X0cZSKQDtFirct2luFUwKzZ/view?usp=sharing) |
| data_aishell | [OpenSLR](https://www.openslr.org/33/) | |
[aidatatang_200zh](http://www.openslr.org/62/)、[magicdata](http://www.openslr.org/68/)、[aishell3](http://www.openslr.org/93/)
> 解壓 aidatatang_200zh 後,還需將 `aidatatang_200zh\corpus\train`下的檔案全選解壓縮
#### 2.`<datasets_root>`是什麼意思?

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@@ -6,7 +6,7 @@
> English | [中文](README-CN.md)
## Features
🌍 **Chinese** supported mandarin and tested with multiple datasets: aidatatang_200zh, magicdata, aishell3, data_aishell, and etc.
🌍 **Chinese** supported mandarin and tested with multiple datasets: aidatatang_200zh, magicdata, aishell3, and etc.
🤩 **PyTorch** worked for pytorch, tested in version of 1.9.0(latest in August 2021), with GPU Tesla T4 and GTX 2060
@@ -16,7 +16,7 @@
🌍 **Webserver Ready** to serve your result with remote calling
### [DEMO VIDEO](https://www.bilibili.com/video/BV17Q4y1B7mY/)
### [DEMO VIDEO](https://www.bilibili.com/video/BV1sA411P7wM/)
## Quick Start
@@ -36,7 +36,7 @@ You can either train your models or use existing ones:
* Download dataset and unzip: make sure you can access all .wav in folder
* Preprocess with the audios and the mel spectrograms:
`python pre.py <datasets_root>`
Allowing parameter `--dataset {dataset}` to support aidatatang_200zh, magicdata, aishell3, data_aishell, etc.If this parameter is not passed, the default dataset will be aidatatang_200zh.
Allowing parameter `--dataset {dataset}` to support aidatatang_200zh, magicdata, aishell3, etc.
* Train the synthesizer:
`python synthesizer_train.py mandarin <datasets_root>/SV2TTS/synthesizer`
@@ -49,7 +49,7 @@ Allowing parameter `--dataset {dataset}` to support aidatatang_200zh, magicdata,
| author | Download link | Preview Video | Info |
| --- | ----------- | ----- |----- |
| @myself | https://pan.baidu.com/s/1VHSKIbxXQejtxi2at9IrpA [Baidu](https://pan.baidu.com/s/1VHSKIbxXQejtxi2at9IrpA ) codei183 | | 200k steps only trained by aidatatang_200zh
|@FawenYo | https://drive.google.com/file/d/1H-YGOUHpmqKxJ9FRc6vAjPuqQki24UbC/view?usp=sharing https://u.teknik.io/AYxWf.pt | [input](https://github.com/babysor/MockingBird/wiki/audio/self_test.mp3) [output](https://github.com/babysor/MockingBird/wiki/audio/export.wav) | 200k steps with local accent of Taiwan
|@FawenYo | https://drive.google.com/file/d/1H-YGOUHpmqKxJ9FRc6vAjPuqQki24UbC/view?usp=sharing [Baidu Pan](https://pan.baidu.com/s/1vSYXO4wsLyjnF3Unl-Xoxg) Code1024 | [input](https://github.com/babysor/MockingBird/wiki/audio/self_test.mp3) [output](https://github.com/babysor/MockingBird/wiki/audio/export.wav) | 200k steps with local accent of Taiwan
|@miven| https://pan.baidu.com/s/1PI-hM3sn5wbeChRryX-RCQ code2021 | https://www.bilibili.com/video/BV1uh411B7AD/
#### 2.3 Train vocoder (Optional)
@@ -77,7 +77,6 @@ You can then try the toolbox:
| URL | Designation | Title | Implementation source |
| --- | ----------- | ----- | --------------------- |
| [1803.09017](https://arxiv.org/abs/1803.09017) | GlobalStyleToken (synthesizer)| Style Tokens: Unsupervised Style Modeling, Control and Transfer in End-to-End Speech Synthesis | This repo |
| [2010.05646](https://arxiv.org/abs/2010.05646) | HiFi-GAN (vocoder)| Generative Adversarial Networks for Efficient and High Fidelity Speech Synthesis | This repo |
|[**1806.04558**](https://arxiv.org/pdf/1806.04558.pdf) | **SV2TTS** | **Transfer Learning from Speaker Verification to Multispeaker Text-To-Speech Synthesis** | This repo |
|[1802.08435](https://arxiv.org/pdf/1802.08435.pdf) | WaveRNN (vocoder) | Efficient Neural Audio Synthesis | [fatchord/WaveRNN](https://github.com/fatchord/WaveRNN) |
@@ -86,12 +85,7 @@ You can then try the toolbox:
## F Q&A
#### 1.Where can I download the dataset?
| Dataset | Original Source | Alternative Sources |
| --- | ----------- | ---------------|
| aidatatang_200zh | [OpenSLR](http://www.openslr.org/62/) | [Google Drive](https://drive.google.com/file/d/110A11KZoVe7vy6kXlLb6zVPLb_J91I_t/view?usp=sharing) |
| magicdata | [OpenSLR](http://www.openslr.org/68/) | [Google Drive (Dev set)](https://drive.google.com/file/d/1g5bWRUSNH68ycC6eNvtwh07nX3QhOOlo/view?usp=sharing) |
| aishell3 | [OpenSLR](https://www.openslr.org/93/) | [Google Drive](https://drive.google.com/file/d/1shYp_o4Z0X0cZSKQDtFirct2luFUwKzZ/view?usp=sharing) |
| data_aishell | [OpenSLR](https://www.openslr.org/33/) | |
[aidatatang_200zh](http://www.openslr.org/62/)、[magicdata](http://www.openslr.org/68/)、[aishell3](http://www.openslr.org/93/)
> After unzip aidatatang_200zh, you need to unzip all the files under `aidatatang_200zh\corpus\train`
#### 2.What is`<datasets_root>`?

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@@ -1,4 +1,4 @@
from encoder.preprocess import preprocess_librispeech, preprocess_voxceleb1, preprocess_voxceleb2, preprocess_aidatatang_200zh
from encoder.preprocess import preprocess_librispeech, preprocess_voxceleb1, preprocess_voxceleb2
from utils.argutils import print_args
from pathlib import Path
import argparse
@@ -10,7 +10,17 @@ if __name__ == "__main__":
parser = argparse.ArgumentParser(
description="Preprocesses audio files from datasets, encodes them as mel spectrograms and "
"writes them to the disk. This will allow you to train the encoder. The "
"datasets required are at least one of LibriSpeech, VoxCeleb1, VoxCeleb2, aidatatang_200zh. ",
"datasets required are at least one of VoxCeleb1, VoxCeleb2 and LibriSpeech. "
"Ideally, you should have all three. You should extract them as they are "
"after having downloaded them and put them in a same directory, e.g.:\n"
"-[datasets_root]\n"
" -LibriSpeech\n"
" -train-other-500\n"
" -VoxCeleb1\n"
" -wav\n"
" -vox1_meta.csv\n"
" -VoxCeleb2\n"
" -dev",
formatter_class=MyFormatter
)
parser.add_argument("datasets_root", type=Path, help=\
@@ -19,7 +29,7 @@ if __name__ == "__main__":
"Path to the output directory that will contain the mel spectrograms. If left out, "
"defaults to <datasets_root>/SV2TTS/encoder/")
parser.add_argument("-d", "--datasets", type=str,
default="librispeech_other,voxceleb1,aidatatang_200zh", help=\
default="librispeech_other,voxceleb1,voxceleb2", help=\
"Comma-separated list of the name of the datasets you want to preprocess. Only the train "
"set of these datasets will be used. Possible names: librispeech_other, voxceleb1, "
"voxceleb2.")
@@ -53,7 +63,6 @@ if __name__ == "__main__":
"librispeech_other": preprocess_librispeech,
"voxceleb1": preprocess_voxceleb1,
"voxceleb2": preprocess_voxceleb2,
"aidatatang_200zh": preprocess_aidatatang_200zh,
}
args = vars(args)
for dataset in args.pop("datasets"):

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@@ -117,15 +117,6 @@ def _preprocess_speaker_dirs(speaker_dirs, dataset_name, datasets_root, out_dir,
logger.finalize()
print("Done preprocessing %s.\n" % dataset_name)
def preprocess_aidatatang_200zh(datasets_root: Path, out_dir: Path, skip_existing=False):
dataset_name = "aidatatang_200zh"
dataset_root, logger = _init_preprocess_dataset(dataset_name, datasets_root, out_dir)
if not dataset_root:
return
# Preprocess all speakers
speaker_dirs = list(dataset_root.joinpath("corpus", "train").glob("*"))
_preprocess_speaker_dirs(speaker_dirs, dataset_name, datasets_root, out_dir, "wav",
skip_existing, logger)
def preprocess_librispeech(datasets_root: Path, out_dir: Path, skip_existing=False):
for dataset_name in librispeech_datasets["train"]["other"]:

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5
pre.py
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@@ -12,8 +12,7 @@ import argparse
recognized_datasets = [
"aidatatang_200zh",
"magicdata",
"aishell3",
"data_aishell"
"aishell3"
]
if __name__ == "__main__":
@@ -41,7 +40,7 @@ if __name__ == "__main__":
"Use this option when dataset does not include alignments\
(these are used to split long audio files into sub-utterances.)")
parser.add_argument("-d", "--dataset", type=str, default="aidatatang_200zh", help=\
"Name of the dataset to process, allowing values: magicdata, aidatatang_200zh, aishell3, data_aishell.")
"Name of the dataset to process, allowing values: magicdata, aidatatang_200zh, aishell3.")
parser.add_argument("-e", "--encoder_model_fpath", type=Path, default="encoder/saved_models/pretrained.pt", help=\
"Path your trained encoder model.")
parser.add_argument("-ne", "--n_processes_embed", type=int, default=1, help=\

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@@ -19,5 +19,4 @@ flask
flask_wtf
flask_cors
gevent==21.8.0
flask_restx
tensorboard
flask_restx

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@@ -1,13 +0,0 @@
class GSTHyperparameters():
E = 512
# reference encoder
ref_enc_filters = [32, 32, 64, 64, 128, 128]
# style token layer
token_num = 10
# token_emb_size = 256
num_heads = 8
n_mels = 256 # Number of Mel banks to generate

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@@ -49,23 +49,18 @@ hparams = HParams(
# frame that has all values < -3.4
### Tacotron Training
tts_schedule = [(2, 1e-3, 10_000, 12), # Progressive training schedule
(2, 5e-4, 15_000, 12), # (r, lr, step, batch_size)
(2, 2e-4, 20_000, 12), # (r, lr, step, batch_size)
(2, 1e-4, 30_000, 12), #
(2, 5e-5, 40_000, 12), #
(2, 1e-5, 60_000, 12), #
(2, 5e-6, 160_000, 12), # r = reduction factor (# of mel frames
(2, 3e-6, 320_000, 12), # synthesized for each decoder iteration)
(2, 1e-6, 640_000, 12)], # lr = learning rate
tts_schedule = [(2, 1e-3, 20_000, 24), # Progressive training schedule
(2, 5e-4, 40_000, 24), # (r, lr, step, batch_size)
(2, 2e-4, 80_000, 24), #
(2, 1e-4, 160_000, 24), # r = reduction factor (# of mel frames
(2, 3e-5, 320_000, 24), # synthesized for each decoder iteration)
(2, 1e-5, 640_000, 24)], # lr = learning rate
tts_clip_grad_norm = 1.0, # clips the gradient norm to prevent explosion - set to None if not needed
tts_eval_interval = 500, # Number of steps between model evaluation (sample generation)
# Set to -1 to generate after completing epoch, or 0 to disable
tts_eval_num_samples = 1, # Makes this number of samples
## For finetune usage, if set, only selected layers will be trained, available: encoder,encoder_proj,gst,decoder,postnet,post_proj
tts_finetune_layers = [],
tts_eval_num_samples = 1, # Makes this number of samples
### Data Preprocessing
max_mel_frames = 900,

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@@ -70,7 +70,7 @@ class Synthesizer:
def synthesize_spectrograms(self, texts: List[str],
embeddings: Union[np.ndarray, List[np.ndarray]],
return_alignments=False, style_idx=0, min_stop_token=5):
return_alignments=False):
"""
Synthesizes mel spectrograms from texts and speaker embeddings.
@@ -125,7 +125,7 @@ class Synthesizer:
speaker_embeddings = torch.tensor(speaker_embeds).float().to(self.device)
# Inference
_, mels, alignments = self._model.generate(chars, speaker_embeddings, style_idx=style_idx, min_stop_token=min_stop_token)
_, mels, alignments = self._model.generate(chars, speaker_embeddings)
mels = mels.detach().cpu().numpy()
for m in mels:
# Trim silence from end of each spectrogram

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@@ -1,135 +0,0 @@
import torch
import torch.nn as nn
import torch.nn.init as init
import torch.nn.functional as tFunctional
from synthesizer.gst_hyperparameters import GSTHyperparameters as hp
class GlobalStyleToken(nn.Module):
def __init__(self):
super().__init__()
self.encoder = ReferenceEncoder()
self.stl = STL()
def forward(self, inputs):
enc_out = self.encoder(inputs)
style_embed = self.stl(enc_out)
return style_embed
class ReferenceEncoder(nn.Module):
'''
inputs --- [N, Ty/r, n_mels*r] mels
outputs --- [N, ref_enc_gru_size]
'''
def __init__(self):
super().__init__()
K = len(hp.ref_enc_filters)
filters = [1] + hp.ref_enc_filters
convs = [nn.Conv2d(in_channels=filters[i],
out_channels=filters[i + 1],
kernel_size=(3, 3),
stride=(2, 2),
padding=(1, 1)) for i in range(K)]
self.convs = nn.ModuleList(convs)
self.bns = nn.ModuleList([nn.BatchNorm2d(num_features=hp.ref_enc_filters[i]) for i in range(K)])
out_channels = self.calculate_channels(hp.n_mels, 3, 2, 1, K)
self.gru = nn.GRU(input_size=hp.ref_enc_filters[-1] * out_channels,
hidden_size=hp.E // 2,
batch_first=True)
def forward(self, inputs):
N = inputs.size(0)
out = inputs.view(N, 1, -1, hp.n_mels) # [N, 1, Ty, n_mels]
for conv, bn in zip(self.convs, self.bns):
out = conv(out)
out = bn(out)
out = tFunctional.relu(out) # [N, 128, Ty//2^K, n_mels//2^K]
out = out.transpose(1, 2) # [N, Ty//2^K, 128, n_mels//2^K]
T = out.size(1)
N = out.size(0)
out = out.contiguous().view(N, T, -1) # [N, Ty//2^K, 128*n_mels//2^K]
self.gru.flatten_parameters()
memory, out = self.gru(out) # out --- [1, N, E//2]
return out.squeeze(0)
def calculate_channels(self, L, kernel_size, stride, pad, n_convs):
for i in range(n_convs):
L = (L - kernel_size + 2 * pad) // stride + 1
return L
class STL(nn.Module):
'''
inputs --- [N, E//2]
'''
def __init__(self):
super().__init__()
self.embed = nn.Parameter(torch.FloatTensor(hp.token_num, hp.E // hp.num_heads))
d_q = hp.E // 2
d_k = hp.E // hp.num_heads
# self.attention = MultiHeadAttention(hp.num_heads, d_model, d_q, d_v)
self.attention = MultiHeadAttention(query_dim=d_q, key_dim=d_k, num_units=hp.E, num_heads=hp.num_heads)
init.normal_(self.embed, mean=0, std=0.5)
def forward(self, inputs):
N = inputs.size(0)
query = inputs.unsqueeze(1) # [N, 1, E//2]
keys = tFunctional.tanh(self.embed).unsqueeze(0).expand(N, -1, -1) # [N, token_num, E // num_heads]
style_embed = self.attention(query, keys)
return style_embed
class MultiHeadAttention(nn.Module):
'''
input:
query --- [N, T_q, query_dim]
key --- [N, T_k, key_dim]
output:
out --- [N, T_q, num_units]
'''
def __init__(self, query_dim, key_dim, num_units, num_heads):
super().__init__()
self.num_units = num_units
self.num_heads = num_heads
self.key_dim = key_dim
self.W_query = nn.Linear(in_features=query_dim, out_features=num_units, bias=False)
self.W_key = nn.Linear(in_features=key_dim, out_features=num_units, bias=False)
self.W_value = nn.Linear(in_features=key_dim, out_features=num_units, bias=False)
def forward(self, query, key):
querys = self.W_query(query) # [N, T_q, num_units]
keys = self.W_key(key) # [N, T_k, num_units]
values = self.W_value(key)
split_size = self.num_units // self.num_heads
querys = torch.stack(torch.split(querys, split_size, dim=2), dim=0) # [h, N, T_q, num_units/h]
keys = torch.stack(torch.split(keys, split_size, dim=2), dim=0) # [h, N, T_k, num_units/h]
values = torch.stack(torch.split(values, split_size, dim=2), dim=0) # [h, N, T_k, num_units/h]
# score = softmax(QK^T / (d_k ** 0.5))
scores = torch.matmul(querys, keys.transpose(2, 3)) # [h, N, T_q, T_k]
scores = scores / (self.key_dim ** 0.5)
scores = tFunctional.softmax(scores, dim=3)
# out = score * V
out = torch.matmul(scores, values) # [h, N, T_q, num_units/h]
out = torch.cat(torch.split(out, 1, dim=0), dim=3).squeeze(0) # [N, T_q, num_units]
return out

View File

@@ -3,7 +3,8 @@ import numpy as np
import torch
import torch.nn as nn
import torch.nn.functional as F
from synthesizer.models.global_style_token import GlobalStyleToken
from pathlib import Path
from typing import Union
class HighwayNetwork(nn.Module):
@@ -337,7 +338,6 @@ class Tacotron(nn.Module):
self.encoder = Encoder(embed_dims, num_chars, encoder_dims,
encoder_K, num_highways, dropout)
self.encoder_proj = nn.Linear(encoder_dims + speaker_embedding_size, decoder_dims, bias=False)
self.gst = GlobalStyleToken()
self.decoder = Decoder(n_mels, encoder_dims, decoder_dims, lstm_dims,
dropout, speaker_embedding_size)
self.postnet = CBHG(postnet_K, n_mels, postnet_dims,
@@ -358,11 +358,11 @@ class Tacotron(nn.Module):
def r(self, value):
self.decoder.r = self.decoder.r.new_tensor(value, requires_grad=False)
def forward(self, texts, mels, speaker_embedding):
def forward(self, x, m, speaker_embedding):
device = next(self.parameters()).device # use same device as parameters
self.step += 1
batch_size, _, steps = mels.size()
batch_size, _, steps = m.size()
# Initialise all hidden states and pack into tuple
attn_hidden = torch.zeros(batch_size, self.decoder_dims, device=device)
@@ -383,12 +383,7 @@ class Tacotron(nn.Module):
# SV2TTS: Run the encoder with the speaker embedding
# The projection avoids unnecessary matmuls in the decoder loop
encoder_seq = self.encoder(texts, speaker_embedding)
# put after encoder
if self.gst is not None:
style_embed = self.gst(speaker_embedding)
style_embed = style_embed.expand_as(encoder_seq)
encoder_seq = encoder_seq + style_embed
encoder_seq = self.encoder(x, speaker_embedding)
encoder_seq_proj = self.encoder_proj(encoder_seq)
# Need a couple of lists for outputs
@@ -396,10 +391,10 @@ class Tacotron(nn.Module):
# Run the decoder loop
for t in range(0, steps, self.r):
prenet_in = mels[:, :, t - 1] if t > 0 else go_frame
prenet_in = m[:, :, t - 1] if t > 0 else go_frame
mel_frames, scores, hidden_states, cell_states, context_vec, stop_tokens = \
self.decoder(encoder_seq, encoder_seq_proj, prenet_in,
hidden_states, cell_states, context_vec, t, texts)
hidden_states, cell_states, context_vec, t, x)
mel_outputs.append(mel_frames)
attn_scores.append(scores)
stop_outputs.extend([stop_tokens] * self.r)
@@ -419,7 +414,7 @@ class Tacotron(nn.Module):
return mel_outputs, linear, attn_scores, stop_outputs
def generate(self, x, speaker_embedding=None, steps=200, style_idx=0, min_stop_token=5):
def generate(self, x, speaker_embedding=None, steps=2000):
self.eval()
device = next(self.parameters()).device # use same device as parameters
@@ -445,18 +440,6 @@ class Tacotron(nn.Module):
# SV2TTS: Run the encoder with the speaker embedding
# The projection avoids unnecessary matmuls in the decoder loop
encoder_seq = self.encoder(x, speaker_embedding)
# put after encoder
if self.gst is not None and style_idx >= 0 and style_idx < 10:
gst_embed = self.gst.stl.embed.cpu().data.numpy() #[0, number_token]
gst_embed = np.tile(gst_embed, (1, 8))
scale = np.zeros(512)
scale[:] = 0.3
speaker_embedding = (gst_embed[style_idx] * scale).astype(np.float32)
speaker_embedding = torch.from_numpy(np.tile(speaker_embedding, (x.shape[0], 1))).to(device)
style_embed = self.gst(speaker_embedding)
style_embed = style_embed.expand_as(encoder_seq)
encoder_seq = encoder_seq + style_embed
encoder_seq_proj = self.encoder_proj(encoder_seq)
# Need a couple of lists for outputs
@@ -472,7 +455,7 @@ class Tacotron(nn.Module):
attn_scores.append(scores)
stop_outputs.extend([stop_tokens] * self.r)
# Stop the loop when all stop tokens in batch exceed threshold
if (stop_tokens * 10 > min_stop_token).all() and t > 10: break
if (stop_tokens > 0.5).all() and t > 10: break
# Concat the mel outputs into sequence
mel_outputs = torch.cat(mel_outputs, dim=2)
@@ -496,15 +479,6 @@ class Tacotron(nn.Module):
for p in self.parameters():
if p.dim() > 1: nn.init.xavier_uniform_(p)
def finetune_partial(self, whitelist_layers):
self.zero_grad()
for name, child in self.named_children():
if name in whitelist_layers:
print("Trainable Layer: %s" % name)
print("Trainable Parameters: %.3f" % sum([np.prod(p.size()) for p in child.parameters()]))
for param in child.parameters():
param.requires_grad = False
def get_step(self):
return self.step.data.item()
@@ -520,7 +494,7 @@ class Tacotron(nn.Module):
# Use device of model params as location for loaded state
device = next(self.parameters()).device
checkpoint = torch.load(str(path), map_location=device)
self.load_state_dict(checkpoint["model_state"], strict=False)
self.load_state_dict(checkpoint["model_state"])
if "optimizer_state" in checkpoint and optimizer is not None:
optimizer.load_state_dict(checkpoint["optimizer_state"])

View File

@@ -7,7 +7,7 @@ from tqdm import tqdm
import numpy as np
from encoder import inference as encoder
from synthesizer.preprocess_speaker import preprocess_speaker_general
from synthesizer.preprocess_transcript import preprocess_transcript_aishell3, preprocess_transcript_magicdata
from synthesizer.preprocess_transcript import preprocess_transcript_aishell3
data_info = {
"aidatatang_200zh": {
@@ -18,19 +18,13 @@ data_info = {
"magicdata": {
"subfolders": ["train"],
"trans_filepath": "train/TRANS.txt",
"speak_func": preprocess_speaker_general,
"transcript_func": preprocess_transcript_magicdata,
"speak_func": preprocess_speaker_general
},
"aishell3":{
"subfolders": ["train/wav"],
"trans_filepath": "train/content.txt",
"speak_func": preprocess_speaker_general,
"transcript_func": preprocess_transcript_aishell3,
},
"data_aishell":{
"subfolders": ["wav/train"],
"trans_filepath": "transcript/aishell_transcript_v0.8.txt",
"speak_func": preprocess_speaker_general
}
}

View File

@@ -6,13 +6,4 @@ def preprocess_transcript_aishell3(dict_info, dict_transcript):
transList = []
for i in range(2, len(v), 2):
transList.append(v[i])
dict_info[v[0]] = " ".join(transList)
def preprocess_transcript_magicdata(dict_info, dict_transcript):
for v in dict_transcript:
if not v:
continue
v = v.strip().replace("\n","").replace("\t"," ").split(" ")
dict_info[v[0]] = " ".join(v[2:])
dict_info[v[0]] = " ".join(transList)

View File

@@ -93,7 +93,7 @@ def train(run_id: str, syn_dir: str, models_dir: str, save_every: int,
speaker_embedding_size=hparams.speaker_embedding_size).to(device)
# Initialize the optimizer
optimizer = optim.Adam(model.parameters(), amsgrad=True)
optimizer = optim.Adam(model.parameters())
# Load the weights
if force_restart or not weights_fpath.exists():
@@ -146,6 +146,7 @@ def train(run_id: str, syn_dir: str, models_dir: str, save_every: int,
continue
model.r = r
# Begin the training
simple_table([(f"Steps with r={r}", str(training_steps // 1000) + "k Steps"),
("Batch Size", batch_size),
@@ -154,8 +155,6 @@ def train(run_id: str, syn_dir: str, models_dir: str, save_every: int,
for p in optimizer.param_groups:
p["lr"] = lr
if hparams.tts_finetune_layers is not None and len(hparams.tts_finetune_layers) > 0:
model.finetune_partial(hparams.tts_finetune_layers)
data_loader = DataLoader(dataset,
collate_fn=collate_synthesizer,

View File

@@ -71,7 +71,6 @@ class Toolbox:
# Initialize the events and the interface
self.ui = UI()
self.style_idx = 0
self.reset_ui(enc_models_dir, syn_models_dir, voc_models_dir, seed)
self.setup_events()
self.ui.start()
@@ -234,8 +233,7 @@ class Toolbox:
texts = processed_texts
embed = self.ui.selected_utterance.embed
embeds = [embed] * len(texts)
min_token = int(self.ui.token_slider.value())
specs = self.synthesizer.synthesize_spectrograms(texts, embeds, style_idx=int(self.ui.style_slider.value()), min_stop_token=min_token)
specs = self.synthesizer.synthesize_spectrograms(texts, embeds)
breaks = [spec.shape[1] for spec in specs]
spec = np.concatenate(specs, axis=1)

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View File

@@ -2,7 +2,6 @@ import matplotlib.pyplot as plt
from matplotlib.backends.backend_qt5agg import FigureCanvasQTAgg as FigureCanvas
from matplotlib.figure import Figure
from PyQt5.QtCore import Qt, QStringListModel
from PyQt5 import QtGui
from PyQt5.QtWidgets import *
from encoder.inference import plot_embedding_as_heatmap
from toolbox.utterance import Utterance
@@ -421,10 +420,7 @@ class UI(QDialog):
## Initialize the application
self.app = QApplication(sys.argv)
super().__init__(None)
self.setWindowTitle("MockingBird GUI")
self.setWindowIcon(QtGui.QIcon('toolbox\\assets\\mb.png'))
self.setWindowFlag(Qt.WindowMinimizeButtonHint, True)
self.setWindowFlag(Qt.WindowMaximizeButtonHint, True)
self.setWindowTitle("SV2TTS toolbox")
## Main layouts
@@ -434,24 +430,21 @@ class UI(QDialog):
# Browser
browser_layout = QGridLayout()
root_layout.addLayout(browser_layout, 0, 0, 1, 8)
root_layout.addLayout(browser_layout, 0, 0, 1, 2)
# Generation
gen_layout = QVBoxLayout()
root_layout.addLayout(gen_layout, 0, 8)
# Visualizations
vis_layout = QVBoxLayout()
root_layout.addLayout(vis_layout, 1, 0, 2, 8)
# Output
output_layout = QGridLayout()
vis_layout.addLayout(output_layout, 0)
root_layout.addLayout(gen_layout, 0, 2, 1, 2)
# Projections
self.projections_layout = QVBoxLayout()
root_layout.addLayout(self.projections_layout, 1, 8, 2, 2)
root_layout.addLayout(self.projections_layout, 1, 0, 1, 1)
# Visualizations
vis_layout = QVBoxLayout()
root_layout.addLayout(vis_layout, 1, 1, 1, 3)
## Projections
# UMap
fig, self.umap_ax = plt.subplots(figsize=(3, 3), facecolor="#F0F0F0")
@@ -465,88 +458,80 @@ class UI(QDialog):
## Browser
# Dataset, speaker and utterance selection
i = 0
source_groupbox = QGroupBox('Source(源音频)')
source_layout = QGridLayout()
source_groupbox.setLayout(source_layout)
browser_layout.addWidget(source_groupbox, i, 0, 1, 4)
self.dataset_box = QComboBox()
source_layout.addWidget(QLabel("Dataset(数据集):"), i, 0)
source_layout.addWidget(self.dataset_box, i, 1)
self.random_dataset_button = QPushButton("Random")
source_layout.addWidget(self.random_dataset_button, i, 2)
i += 1
browser_layout.addWidget(QLabel("<b>Dataset</b>"), i, 0)
browser_layout.addWidget(self.dataset_box, i + 1, 0)
self.speaker_box = QComboBox()
source_layout.addWidget(QLabel("Speaker(说话者)"), i, 0)
source_layout.addWidget(self.speaker_box, i, 1)
self.random_speaker_button = QPushButton("Random")
source_layout.addWidget(self.random_speaker_button, i, 2)
i += 1
browser_layout.addWidget(QLabel("<b>Speaker</b>"), i, 1)
browser_layout.addWidget(self.speaker_box, i + 1, 1)
self.utterance_box = QComboBox()
source_layout.addWidget(QLabel("Utterance(音频):"), i, 0)
source_layout.addWidget(self.utterance_box, i, 1)
browser_layout.addWidget(QLabel("<b>Utterance</b>"), i, 2)
browser_layout.addWidget(self.utterance_box, i + 1, 2)
self.browser_load_button = QPushButton("Load")
browser_layout.addWidget(self.browser_load_button, i + 1, 3)
i += 2
# Random buttons
self.random_dataset_button = QPushButton("Random")
browser_layout.addWidget(self.random_dataset_button, i, 0)
self.random_speaker_button = QPushButton("Random")
browser_layout.addWidget(self.random_speaker_button, i, 1)
self.random_utterance_button = QPushButton("Random")
source_layout.addWidget(self.random_utterance_button, i, 2)
i += 1
source_layout.addWidget(QLabel("<b>Use(使用):</b>"), i, 0)
self.browser_load_button = QPushButton("Load Above(加载上面)")
source_layout.addWidget(self.browser_load_button, i, 1, 1, 2)
browser_layout.addWidget(self.random_utterance_button, i, 2)
self.auto_next_checkbox = QCheckBox("Auto select next")
self.auto_next_checkbox.setChecked(True)
source_layout.addWidget(self.auto_next_checkbox, i+1, 1)
self.browser_browse_button = QPushButton("Browse(打开本地)")
source_layout.addWidget(self.browser_browse_button, i, 3)
self.record_button = QPushButton("Record(录音)")
source_layout.addWidget(self.record_button, i+1, 3)
i += 2
# Utterance box
browser_layout.addWidget(QLabel("<b>Current(当前):</b>"), i, 0)
self.utterance_history = QComboBox()
browser_layout.addWidget(self.utterance_history, i, 1)
self.play_button = QPushButton("Play(播放)")
browser_layout.addWidget(self.play_button, i, 2)
self.stop_button = QPushButton("Stop(暂停)")
browser_layout.addWidget(self.stop_button, i, 3)
browser_layout.addWidget(self.auto_next_checkbox, i, 3)
i += 1
model_groupbox = QGroupBox('Models(模型选择)')
model_layout = QHBoxLayout()
model_groupbox.setLayout(model_layout)
browser_layout.addWidget(model_groupbox, i, 0, 1, 4)
# Utterance box
browser_layout.addWidget(QLabel("<b>Use embedding from:</b>"), i, 0)
self.utterance_history = QComboBox()
browser_layout.addWidget(self.utterance_history, i, 1, 1, 3)
i += 1
# Random & next utterance buttons
self.browser_browse_button = QPushButton("Browse")
browser_layout.addWidget(self.browser_browse_button, i, 0)
self.record_button = QPushButton("Record")
browser_layout.addWidget(self.record_button, i, 1)
self.play_button = QPushButton("Play")
browser_layout.addWidget(self.play_button, i, 2)
self.stop_button = QPushButton("Stop")
browser_layout.addWidget(self.stop_button, i, 3)
i += 1
# Model and audio output selection
self.encoder_box = QComboBox()
model_layout.addWidget(QLabel("Encoder:"))
model_layout.addWidget(self.encoder_box)
browser_layout.addWidget(QLabel("<b>Encoder</b>"), i, 0)
browser_layout.addWidget(self.encoder_box, i + 1, 0)
self.synthesizer_box = QComboBox()
model_layout.addWidget(QLabel("Synthesizer:"))
model_layout.addWidget(self.synthesizer_box)
browser_layout.addWidget(QLabel("<b>Synthesizer</b>"), i, 1)
browser_layout.addWidget(self.synthesizer_box, i + 1, 1)
self.vocoder_box = QComboBox()
model_layout.addWidget(QLabel("Vocoder:"))
model_layout.addWidget(self.vocoder_box)
browser_layout.addWidget(QLabel("<b>Vocoder</b>"), i, 2)
browser_layout.addWidget(self.vocoder_box, i + 1, 2)
self.audio_out_devices_cb=QComboBox()
browser_layout.addWidget(QLabel("<b>Audio Output</b>"), i, 3)
browser_layout.addWidget(self.audio_out_devices_cb, i + 1, 3)
i += 2
#Replay & Save Audio
i = 0
output_layout.addWidget(QLabel("<b>Toolbox Output:</b>"), i, 0)
browser_layout.addWidget(QLabel("<b>Toolbox Output:</b>"), i, 0)
self.waves_cb = QComboBox()
self.waves_cb_model = QStringListModel()
self.waves_cb.setModel(self.waves_cb_model)
self.waves_cb.setToolTip("Select one of the last generated waves in this section for replaying or exporting")
output_layout.addWidget(self.waves_cb, i, 1)
browser_layout.addWidget(self.waves_cb, i, 1)
self.replay_wav_button = QPushButton("Replay")
self.replay_wav_button.setToolTip("Replay last generated vocoder")
output_layout.addWidget(self.replay_wav_button, i, 2)
browser_layout.addWidget(self.replay_wav_button, i, 2)
self.export_wav_button = QPushButton("Export")
self.export_wav_button.setToolTip("Save last generated vocoder audio in filesystem as a wav file")
output_layout.addWidget(self.export_wav_button, i, 3)
self.audio_out_devices_cb=QComboBox()
browser_layout.addWidget(self.export_wav_button, i, 3)
i += 1
output_layout.addWidget(QLabel("<b>Audio Output</b>"), i, 0)
output_layout.addWidget(self.audio_out_devices_cb, i, 1)
## Embed & spectrograms
vis_layout.addStretch()
@@ -567,6 +552,7 @@ class UI(QDialog):
for side in ["top", "right", "bottom", "left"]:
ax.spines[side].set_visible(False)
## Generation
self.text_prompt = QPlainTextEdit(default_text)
gen_layout.addWidget(self.text_prompt, stretch=1)
@@ -592,32 +578,6 @@ class UI(QDialog):
self.trim_silences_checkbox.setToolTip("When checked, trims excess silence in vocoder output."
" This feature requires `webrtcvad` to be installed.")
layout_seed.addWidget(self.trim_silences_checkbox, 0, 2, 1, 2)
self.style_slider = QSlider(Qt.Horizontal)
self.style_slider.setTickInterval(1)
self.style_slider.setFocusPolicy(Qt.NoFocus)
self.style_slider.setSingleStep(1)
self.style_slider.setRange(-1, 9)
self.style_value_label = QLabel("-1")
self.style_slider.setValue(-1)
layout_seed.addWidget(QLabel("Style:"), 1, 0)
self.style_slider.valueChanged.connect(lambda s: self.style_value_label.setNum(s))
layout_seed.addWidget(self.style_value_label, 1, 1)
layout_seed.addWidget(self.style_slider, 1, 3)
self.token_slider = QSlider(Qt.Horizontal)
self.token_slider.setTickInterval(1)
self.token_slider.setFocusPolicy(Qt.NoFocus)
self.token_slider.setSingleStep(1)
self.token_slider.setRange(3, 9)
self.token_value_label = QLabel("5")
self.token_slider.setValue(4)
layout_seed.addWidget(QLabel("Accuracy(精度):"), 2, 0)
self.token_slider.valueChanged.connect(lambda s: self.token_value_label.setNum(s))
layout_seed.addWidget(self.token_value_label, 2, 1)
layout_seed.addWidget(self.token_slider, 2, 3)
gen_layout.addLayout(layout_seed)
self.loading_bar = QProgressBar()
@@ -631,7 +591,7 @@ class UI(QDialog):
## Set the size of the window and of the elements
max_size = QDesktopWidget().availableGeometry(self).size() * 0.5
max_size = QDesktopWidget().availableGeometry(self).size() * 0.8
self.resize(max_size)
## Finalize the display

View File

@@ -11,6 +11,7 @@ def check_model_paths(encoder_path: Path, synthesizer_path: Path, vocoder_path:
# If none of the paths exist, remind the user to download models if needed
print("********************************************************************************")
print("Error: Model files not found. Please download the models")
print("Error: Model files not found. Follow these instructions to get and install the models:")
print("https://github.com/CorentinJ/Real-Time-Voice-Cloning/wiki/Pretrained-models")
print("********************************************************************************\n")
quit(-1)

View File

@@ -9,7 +9,6 @@ from vocoder.wavernn import inference as rnn_vocoder
import numpy as np
import re
from scipy.io.wavfile import write
import librosa
import io
import base64
from flask_cors import CORS
@@ -31,7 +30,6 @@ def webApp():
synthesizers = list(Path(syn_models_dirt).glob("**/*.pt"))
synthesizers_cache = {}
encoder.load_model(Path("encoder/saved_models/pretrained.pt"))
# rnn_vocoder.load_model(Path("vocoder/saved_models/pretrained/pretrained.pt"))
gan_vocoder.load_model(Path("vocoder/saved_models/pretrained/g_hifigan.pt"))
def pcm2float(sig, dtype='float32'):
@@ -68,6 +66,7 @@ def webApp():
@app.route("/api/synthesize", methods=["POST"])
def synthesize():
# TODO Implementation with json to support more platform
# Load synthesizer
if "synt_path" in request.form:
synt_path = request.form["synt_path"]
@@ -81,16 +80,10 @@ def webApp():
current_synt = synthesizers_cache[synt_path]
print("using synthesizer model: " + str(synt_path))
# Load input wav
if "upfile_b64" in request.form:
wav_base64 = request.form["upfile_b64"]
wav = base64.b64decode(bytes(wav_base64, 'utf-8'))
wav = pcm2float(np.frombuffer(wav, dtype=np.int16), dtype=np.float32)
sample_rate = Synthesizer.sample_rate
else:
wav, sample_rate, = librosa.load(request.files['file'])
write("temp.wav", sample_rate, wav) #Make sure we get the correct wav
encoder_wav = encoder.preprocess_wav(wav, sample_rate)
wav_base64 = request.form["upfile_b64"]
wav = base64.b64decode(bytes(wav_base64, 'utf-8'))
wav = pcm2float(np.frombuffer(wav, dtype=np.int16), dtype=np.float32)
encoder_wav = encoder.preprocess_wav(wav, 16000)
embed, _, _ = encoder.embed_utterance(encoder_wav, return_partials=True)
# Load input text
@@ -107,7 +100,6 @@ def webApp():
embeds = [embed] * len(texts)
specs = current_synt.synthesize_spectrograms(texts, embeds)
spec = np.concatenate(specs, axis=1)
# wav = rnn_vocoder.infer_waveform(spec)
wav = gan_vocoder.infer_waveform(spec)
# Return cooked wav

View File

@@ -5,4 +5,3 @@ PORT = 8080
MAX_CONTENT_PATH =1024 * 1024 * 4 # mp3文件大小限定不能超过4M
SECRET_KEY = "mockingbird_key"
WTF_CSRF_SECRET_KEY = "mockingbird_key"
TEMPLATES_AUTO_RELOAD = True

View File

@@ -38,37 +38,22 @@
</div>
<div style="margin-left: 5%;margin-top: 50px;width: 90%;">
<div style="font-size: larger;font-weight: bolder;">1. 请输入中文</div>
<div style="font-size: larger;font-weight: bolder;">请输入中文</div>
<textarea id="user_input_text"
style="border:1px solid #ccc; width: 100%; height: 100px; font-size: 15px; margin-top: 10px;"></textarea>
</div>
<div class="pd btns" style="margin-left: 5%;margin-top: 20px;width: 90%; ">
<div class="pd btns" style="margin-left: 5%;margin-top: 20px;width: 90%; text-align:right;">
<!-- <div>
<button onclick="recOpen()" style="margin-right:10px">打开录音,请求权限</button>
<button onclick="recClose()" style="margin-right:0">关闭录音,释放资源</button>
</div> -->
<div style="font-size: larger;font-weight: bolder;">2. 请直接录音,点击停止结束</div>
<button onclick="recStart()" >录制</button>
<button onclick="recStop()">停止</button>
<button onclick="recPlay()" >播放</button>
<button onclick="recUpload()" >上传</button>
</div>
<div class="pd btns" style="margin-left: 5%;margin-top: 20px;width: 90%; ">
<div style="font-size: larger;font-weight: bolder;">或上传音频</div>
<input type="file" id="fileInput" accept=".wav" />
<label for="fileInput">选择音频</label>
<div id="audio1"></div>
</div>
<div class="pd btns" style="margin-left: 5%;margin-top: 20px;width: 90%; ">
<div style="font-size: larger;font-weight: bolder;">3. 选择Synthesizer模型</div>
<span class="box">
<select id="select">
</select>
</span>
</div>
<div class="pd btns" style="margin-left: 5%;margin-top: 20px;width: 90%; text-align:right;">
<button id="upload" onclick="recUpload()">上传合成</button>
</div>
<!-- 波形绘制区域 -->
<!-- <div class="pd recpower">
<div style="height:40px;width:100%;background:#fff;position:relative;">
@@ -91,37 +76,6 @@
<script>
$("#fileInput").change(function(){
var file = $("#fileInput").get(0).files;
if (file.length > 0) {
var path = URL.createObjectURL(file[0]);
var audio = document.createElement('audio');
audio.src = path;
audio.controls = true;
$('#audio1').empty().append(audio);
}
});
fetch("/api/synthesizers", {
method: 'get',
headers: {
"X-CSRFToken": "{{ csrf_token() }}"
}
}).then(function (res) {
if (!res.ok) throw Error(res.statusText);
return res.json();
}).then(function (data) {
for (var synt of data) {
var option = document.createElement('option');
option.text = synt.name
option.value = synt.path
$("#select").append(option);
}
}).catch(function (err) {
console.log('Error: ' + err.message);
})
var rec, wave, recBlob;
/**调用open打开录音请求好录音权限**/
var recOpen = function () {//一般在显示出录音按钮或相关的录音界面时进行此方法调用,后面用户点击开始录音时就能畅通无阻了
@@ -240,15 +194,9 @@
/**上传**/
function recUpload() {
var blob
var loadedAudios = $("#fileInput").get(0).files
if (loadedAudios.length > 0) {
blob = loadedAudios[0];
} else {
blob = recBlob;
}
var blob = recBlob;
if (!blob) {
reclog("请先录音或选择音频,然后停止后再上传", 1);
reclog("请先录音,然后停止后再上传", 1);
return;
};
@@ -263,18 +211,15 @@
var csrftoken = "{{ csrf_token() }}";
var user_input_text = document.getElementById("user_input_text");
var input_text = user_input_text.value;
var postData = new FormData();
postData.append("text", input_text)
postData.append("file", blob)
var sel = document.getElementById("select");
var path = sel.options[sel.selectedIndex].value;
if (!!path) {
postData.append("synt_path", path);
}
var postData = "";
postData += "mime=" + encodeURIComponent(blob.type);//告诉后端这个录音是什么格式的可能前后端都固定的mp3可以不用写
postData += "&upfile_b64=" + encodeURIComponent((/.+;\s*base64\s*,\s*(.+)$/i.exec(reader.result) || [])[1]) //录音文件内容后端进行base64解码成二进制
postData += "&text=" + encodeURIComponent(input_text);
fetch(api, {
method: 'post',
headers: {
"Content-type": "application/x-www-form-urlencoded; charset=UTF-8",
"X-CSRFToken": csrftoken
},
body: postData
@@ -393,6 +338,7 @@
padding: 12px;
border-radius: 6px;
background: #fff;
--border: 1px solid #327de8;
box-shadow: 2px 2px 3px #aaa;
}
@@ -402,7 +348,7 @@
cursor: pointer;
border: none;
border-radius: 3px;
background: #5698c3;
background: #327de8;
color: #fff;
padding: 0 15px;
margin: 3px 10px 3px 0;
@@ -413,13 +359,6 @@
vertical-align: middle;
}
.btns #upload {
background: #5698c3;
color: #fff;
width: 100px;
height: 42px;
}
.btns button:active {
background: #5da1f5
}
@@ -440,68 +379,6 @@
padding: 2px 8px;
border-radius: 99px;
}
#fileInput {
width: 0.1px;
height: 0.1px;
opacity: 0;
overflow: hidden;
position: absolute;
z-index: -1;
}
#fileInput + label {
padding: 0 15px;
border-radius: 4px;
color: white;
background-color: #5698c3;
display: inline-block;
width: 70px;
line-height: 36px;
height: 36px;
}
#fileInput + label {
cursor: pointer; /* "hand" cursor */
}
#fileInput:focus + label,
#fileInput + label:hover {
background-color: #5da1f5;
}
.box select {
background-color: #5698c3;
color: white;
padding: 8px;
width: 120px;
border: none;
border-radius: 4px;
font-size: 0.5em;
outline: none;
margin: 3px 10px 3px 0;
}
.box::before {
content: "\f13a";
position: absolute;
top: 0;
right: 0;
width: 20%;
height: 100%;
text-align: center;
font-size: 28px;
line-height: 45px;
color: rgba(255, 255, 255, 0.5);
background-color: rgba(255, 255, 255, 0.1);
pointer-events: none;
}
.box:hover::before {
color: rgba(255, 255, 255, 0.6);
background-color: rgba(255, 255, 255, 0.2);
}
.box select option {
padding: 30px;
}
</style>
</body>