metadata
language:
- id
license: mit
base_model: microsoft/speecht5_tts
tags:
- text-to-speech
datasets:
- mozilla-foundation/common_voice_16_1
model-index:
- name: speecht5_finetuned_commonvoice_id
results: []
speecht5_finetuned_commonvoice_id
This model is a fine-tuned version of microsoft/speecht5_tts on the mozilla-foundation/common_voice_16_1 dataset. It achieves the following results on the evaluation set:
- Loss: 0.4675
How to use/inference
Follow the example below and adapt with your own need.
# ft_t5_id_inference.py
import sounddevice as sd
import torch
import torchaudio
from datasets import Audio, load_dataset
from transformers import (
SpeechT5ForTextToSpeech,
SpeechT5HifiGan,
SpeechT5Processor,
)
from utils import create_speaker_embedding
# load dataset and pre-trained model
dataset = load_dataset(
"mozilla-foundation/common_voice_16_1", "id", split="test")
model = SpeechT5ForTextToSpeech.from_pretrained(
"Bagus/speecht5_finetuned_commonvoice_id")
# process the text using checkpoint
checkpoint = "microsoft/speecht5_tts"
processor = SpeechT5Processor.from_pretrained(checkpoint)
sampling_rate = processor.feature_extractor.sampling_rate
dataset = dataset.cast_column("audio", Audio(sampling_rate=sampling_rate))
def prepare_dataset(example):
audio = example["audio"]
example = processor(
text=example["sentence"],
audio_target=audio["array"],
sampling_rate=audio["sampling_rate"],
return_attention_mask=False,
)
# strip off the batch dimension
example["labels"] = example["labels"][0]
# use SpeechBrain to obtain x-vector
example["speaker_embeddings"] = create_speaker_embedding(audio["array"])
return example
# prepare the speaker embeddings from the dataset and text
example = prepare_dataset(dataset[30])
speaker_embeddings = torch.tensor(example["speaker_embeddings"]).unsqueeze(0)
# prepare text to be converted to speech
text = "Saya suka baju yang berwarna merah tua."
inputs = processor(text=text, return_tensors="pt")
vocoder = SpeechT5HifiGan.from_pretrained("microsoft/speecht5_hifigan")
speech = model.generate_speech(
inputs["input_ids"], speaker_embeddings, vocoder=vocoder)
sampling_rate = 16000
sd.play(speech, samplerate=sampling_rate, blocking=True)
# save the audio, signal needs to be in 2D tensor
torchaudio.save("output_t5_ft_cv16_id.wav", speech.unsqueeze(0), 16000)
Training hyperparameters
The following hyperparameters were used during training:
- learning_rate: 1e-05
- train_batch_size: 4
- eval_batch_size: 2
- seed: 42
- gradient_accumulation_steps: 8
- total_train_batch_size: 32
- optimizer: Adam with betas=(0.9,0.999) and epsilon=1e-08
- lr_scheduler_type: linear
- lr_scheduler_warmup_steps: 500
- training_steps: 4000
- mixed_precision_training: Native AMP
Training results
Training Loss | Epoch | Step | Validation Loss |
---|---|---|---|
0.5394 | 4.28 | 1000 | 0.4908 |
0.5062 | 8.56 | 2000 | 0.4730 |
0.5074 | 12.83 | 3000 | 0.4700 |
0.5023 | 17.11 | 4000 | 0.4675 |
Framework versions
- Transformers 4.35.2
- Pytorch 2.1.1+cu121
- Datasets 2.15.0
- Tokenizers 0.15.0