File size: 6,152 Bytes
cd03576
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
## FACodec: Speech Codec with Attribute Factorization used for NaturalSpeech 3

[![arXiv](https://img.shields.io/badge/arXiv-Paper-<COLOR>.svg)](https://arxiv.org/pdf/2403.03100.pdf)
[![demo](https://img.shields.io/badge/FACodec-Demo-red)](https://speechresearch.github.io/naturalspeech3/)
[![model](https://img.shields.io/badge/%F0%9F%A4%97%20HuggingFace-Models-pink)](https://huggingface.co/amphion/naturalspeech3_facodec)
[![hf](https://img.shields.io/badge/%F0%9F%A4%97%20HuggingFace-Spaces-yellow)](https://huggingface.co/spaces/amphion/naturalspeech3_facodec)

## Overview

FACodec is a core component of the advanced text-to-speech (TTS) model NaturalSpeech 3. FACodec converts complex speech waveform into disentangled subspaces representing speech attributes of content, prosody, timbre, and acoustic details and reconstruct high-quality speech waveform from these attributes. FACodec decomposes complex speech into subspaces representing different attributes, thus simplifying the modeling of speech representation.

Research can use FACodec to develop different modes of TTS models, such as non-autoregressive based discrete diffusion (NaturalSpeech 3) or autoregressive models (like VALL-E).

<br>
<div align="center">
<img src="../../imgs/ns3/ns3_overview.png" width="65%">
</div>
<br>

<br>
<div align="center">
<img src="../../imgs/ns3/ns3_facodec.png" width="100%">
</div>
<br>

## Useage

Download the pre-trained FACodec model from HuggingFace: [Pretrained FACodec checkpoint](https://huggingface.co/amphion/naturalspeech3_facodec)

Install Amphion
```bash
git https://github.com/open-mmlab/Amphion.git
```

Few lines of code to use the pre-trained FACodec model
```python
from AmphionOpen.models.ns3_codec import FACodecEncoder, FACodecDecoder

fa_encoder = FACodecEncoder(
    ngf=32,
    up_ratios=[2, 4, 5, 5],
    out_channels=256,
)

fa_decoder = FACodecDecoder(
    in_channels=256,
    upsample_initial_channel=1024,
    ngf=32,
    up_ratios=[5, 5, 4, 2],
    vq_num_q_c=2,
    vq_num_q_p=1,
    vq_num_q_r=3,
    vq_dim=256,
    codebook_dim=8,
    codebook_size_prosody=10,
    codebook_size_content=10,
    codebook_size_residual=10,
    use_gr_x_timbre=True,
    use_gr_residual_f0=True,
    use_gr_residual_phone=True,
)

fa_encoder = torch.load("ns3_facodec_encoder.bin")
fa_decoder = torch.load("ns3_facodec_decoder.bin")

fa_encoder.eval()
fa_decoder.eval()

```

Test
```python
test_wav_path = "test.wav"
test_wav = librosa.load(test_wav_path, sr=16000)[0]
test_wav = torch.from_numpy(test_wav).float()
test_wav = test_wav.unsqueeze(0).unsqueeze(0)

with torch.no_grad():

    # encode
    enc_out = fa_encoder(test_wav)
    print(enc_out.shape)

    # quantize
    vq_post_emb, vq_id, _, quantized, spk_embs = fa_decoder(enc_out, eval_vq=False, vq=True)
    
    # latent after quantization
    print(vq_post_emb.shape)
    
    # codes
    print("vq id shape:", vq_id.shape)
    
    # get prosody code
    prosody_code = vq_id[:1]
    print("prosody code shape:", prosody_code.shape)
    
    # get content code
    cotent_code = vq_id[1:3]
    print("content code shape:", cotent_code.shape)
    
    # get residual code (acoustic detail codes)
    residual_code = vq_id[3:]
    print("residual code shape:", residual_code.shape)
    
    # speaker embedding
    print("speaker embedding shape:", spk_embs.shape)

    # decode (recommand)
    recon_wav = fa_decoder.inference(vq_post_emb, spk_embs)
    print(recon_wav.shape)
    sf.write("recon.wav", recon_wav[0][0].cpu().numpy(), 16000)
```



## Some Q&A

Q1: What audio sample rate does FACodec support? What is the hop size? How many codes will be generated for each frame?

A1: FACodec supports 16KHz speech audio. The hop size is 200 samples, and (16000/200) * 6 (total number of codebooks) codes will be generated for each frame.

Q2: Is it possible to train an autoregressive TTS model like VALL-E using FACodec?

A2: Yes. In fact, the authors of NaturalSpeech 3 have already employ explore the autoregressive generative model for discrete token generation with FACodec. They use an autoregressive language model to generate prosody codes, followed by a non-autoregressive model to generate the remaining content and acoustic details codes.

Q3: Is it possible to train a latent diffusion TTS model like NaturalSpeech2 using FACodec?

A3: Yes. You can use the latent getted after quanzaition as the modelling target for the latent diffusion model.

Q4: Can FACodec compress and reconstruct audio from other domains? Such as sound effects, music, etc.

A4: Since FACodec is designed for speech, it may not be suitable for other audio domains. However, it is possible to use the FACodec model to compress and reconstruct audio from other domains, but the quality may not be as good as the original audio.

Q5: Can FACodec be used for content feature for some other tasks like voice conversion?

A5: I think the answer is yes. Researchers can use the content code of FACodec as the content feature for voice conversion. We hope to see more research in this direction.

## Citations

If you use our FACodec model, please cite the following paper:

```bibtex
@misc{ju2024naturalspeech,
      title={NaturalSpeech 3: Zero-Shot Speech Synthesis with Factorized Codec and Diffusion Models}, 
      author={Zeqian Ju and Yuancheng Wang and Kai Shen and Xu Tan and Detai Xin and Dongchao Yang and Yanqing Liu and Yichong Leng and Kaitao Song and Siliang Tang and Zhizheng Wu and Tao Qin and Xiang-Yang Li and Wei Ye and Shikun Zhang and Jiang Bian and Lei He and Jinyu Li and Sheng Zhao},
      year={2024},
      eprint={2403.03100},
      archivePrefix={arXiv},
      primaryClass={eess.AS}
}

@article{zhang2023amphion,
      title={Amphion: An Open-Source Audio, Music and Speech Generation Toolkit}, 
      author={Xueyao Zhang and Liumeng Xue and Yicheng Gu and Yuancheng Wang and Haorui He and Chaoren Wang and Xi Chen and Zihao Fang and Haopeng Chen and Junan Zhang and Tze Ying Tang and Lexiao Zou and Mingxuan Wang and Jun Han and Kai Chen and Haizhou Li and Zhizheng Wu},
      journal={arXiv},
      year={2024},
      volume={abs/2312.09911}
}
```