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README.md
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# Usage
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Clone repo
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```bash
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git clone https://github.com/nguyenhoanganh2002/XTTSv2-Finetuning-for-New-Languages.git
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cd XTTSv2-Finetuning-for-New-Languages
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pip install -r requirements.txt
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```
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Pull model's weights
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```python
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from huggingface_hub import snapshot_download
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snapshot_download(repo_id="anhnh2002/vnTTS",
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repo_type="model",
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local_dir="model/")
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```
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Load model
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```python
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from pprint import pprint
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import torch
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import torchaudio
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from tqdm import tqdm
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from underthesea import sent_tokenize
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from vinorm import TTSnorm
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from TTS.tts.configs.xtts_config import XttsConfig
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from TTS.tts.models.xtts import Xtts
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device = "cuda:0"
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xtts_checkpoint = "model/model.pth"
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xtts_config = "model/config.json"
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xtts_vocab = "model/vocab.json"
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config = XttsConfig()
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config.load_json(xtts_config)
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XTTS_MODEL = Xtts.init_from_config(config)
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XTTS_MODEL.load_checkpoint(config,
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checkpoint_path=xtts_checkpoint,
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vocab_path=xtts_vocab,
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use_deepspeed=False)
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XTTS_MODEL.to(device)
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```
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Preprocessing and chunking
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```python
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def preprocess_text(text, language="vi"):
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if language == "vi":
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text = TTSnorm(text)
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# split text into sentences
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if language in ["ja", "zh-cn"]:
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sentences = text.split("。")
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else:
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sentences = sent_tokenize(text)
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chunks = []
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chunk_i = ""
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len_chunk_i = 0
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for sentence in sentences:
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chunk_i += " " + sentence
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len_chunk_i += len(sentence.split())
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if len_chunk_i > 30:
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chunks.append(chunk_i.strip())
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chunk_i = ""
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len_chunk_i = 0
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if (len(chunks) > 0) and (len_chunk_i < 15):
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chunks[-1] += chunk_i
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else:
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chunks.append(chunk_i)
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return chunks
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```
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Generate latent embeddings for the speaker
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```python
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speaker_audio_file = "model/vi_man.wav"
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gpt_cond_latent, speaker_embedding = XTTS_MODEL.get_conditioning_latents(
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audio_path=speaker_audio_file,
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gpt_cond_len=XTTS_MODEL.config.gpt_cond_len,
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max_ref_length=XTTS_MODEL.config.max_ref_len,
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sound_norm_refs=XTTS_MODEL.config.sound_norm_refs,
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)
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```
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Inference
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```python
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def tts(
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model: Xtts,
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text: str,
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language: str,
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gpt_cond_latent: torch.Tensor,
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speaker_embedding: torch.Tensor,
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verbose: bool = False,
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):
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# preprocess text
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chunks = preprocess_text(text, language)
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wav_chunks = []
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for text in tqdm(chunks):
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if text.strip() == "":
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continue
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wav_chunk = model.inference(
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text=text,
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language=language,
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gpt_cond_latent=gpt_cond_latent,
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speaker_embedding=speaker_embedding,
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length_penalty=1.0,
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repetition_penalty=10.0,
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top_k=10,
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top_p=0.5,
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)
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wav_chunk["wav"] = torch.tensor(wav_chunk["wav"])
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wav_chunks.append(wav_chunk["wav"])
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out_wav = torch.cat(wav_chunks, dim=0).unsqueeze(0).cpu()
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return out_wav
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from IPython.display import Audio
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audio = tts(
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model=XTTS_MODEL,
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text="Xin chào, tôi là một hệ thống chuyển đổi văn bản tiếng Việt thành giọng nói.", #Hello, I am a Vietnamese text to speech conversion system.
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language="vi",
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gpt_cond_latent=gpt_cond_latent,
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speaker_embedding=speaker_embedding,
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verbose=True,
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)
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Audio(audio, rate=24000)
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```
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# License
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This project uses a model licensed under the Coqui Public Model License 1.0.0, which permits non-commercial use only. This includes personal research, testing, and charitable purposes. Commercial entities may use it for non-commercial research and evaluation. Revenue-generating activities are prohibited. Users must include the license terms when distributing the model or its outputs. For full details, please refer to: https://coqui.ai/cpml
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