speecht5_finetuned_commonvoice_id

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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

Framework versions

• Transformers 4.35.2
• Pytorch 2.1.1+cu121
• Datasets 2.15.0
• Tokenizers 0.15.0