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578dbf4 e1db91c 578dbf4 | 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 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 | import json
import os
import re
import librosa
import numpy as np
import torch
from torch import no_grad, LongTensor
import commons
import utils
import gradio as gr
from models import SynthesizerTrn
from text import text_to_sequence
from text.symbols import symbols
from transformers import pipeline # <-- [BARU] Import pipeline dari transformers
limitation = os.getenv("SYSTEM") == "spaces" # limit text and audio length in huggingface spaces
def get_text(text, hps):
text_norm = text_to_sequence(text, hps.data.text_cleaners)
if hps.data.add_blank:
text_norm = commons.intersperse(text_norm, 0)
text_norm = torch.LongTensor(text_norm)
return text_norm
def create_tts_fn(net_g, hps, speaker_ids):
def tts_fn(text, speaker, speed):
if limitation:
text_len = len(text)
max_len = 5000
if text_len > max_len:
return "Error: Text is too long", None
speaker_id = speaker_ids[speaker]
stn_tst = get_text(text, hps)
with no_grad():
x_tst = stn_tst.unsqueeze(0)
x_tst_lengths = LongTensor([stn_tst.size(0)])
sid = LongTensor([speaker_id])
audio = net_g.infer(x_tst, x_tst_lengths, sid=sid, noise_scale=.667, noise_scale_w=0.8,
length_scale=1.0 / speed)[0][0, 0].data.cpu().float().numpy()
del stn_tst, x_tst, x_tst_lengths, sid
return "Success", (hps.data.sampling_rate, audio)
return tts_fn
css = """
#advanced-btn {
color: white;
border-color: black;
background: black;
font-size: .7rem !important;
line-height: 19px;
margin-top: 24px;
margin-bottom: 12px;
padding: 2px 8px;
border-radius: 14px !important;
}
#advanced-options {
display: none;
margin-bottom: 20px;
}
"""
if __name__ == '__main__':
# --- [BARU] Inisialisasi model Speech-to-Text (Whisper) ---
print("Initializing STT model (Whisper)...")
stt_pipeline = pipeline("automatic-speech-recognition", model="openai/whisper-base")
print("STT model loaded.")
models_tts = []
name = 'AronaTTS'
lang = '일본어 / 한국어 (Japanese / Korean)'
example = '[JA]先生、今日は天気が本当にいいですね。[JA][KO]선생님, 안녕하세요. my name is arona[KO]'
config_path = f"pretrained_model/arona_ms_istft_vits.json"
model_path = f"pretrained_model/arona_ms_istft_vits.pth"
cover_path = f"pretrained_model/cover.gif"
hps = utils.get_hparams_from_file(config_path)
net_g = SynthesizerTrn(
len(symbols),
hps.data.filter_length // 2 + 1,
hps.train.segment_size // hps.data.hop_length,
n_speakers=hps.data.n_speakers,
**hps.model)
_ = net_g.eval()
utils.load_checkpoint(model_path, net_g, None)
net_g.eval()
speaker_ids = [0]
speakers = [name]
# Buat fungsi TTS
tts_fn = create_tts_fn(net_g, hps, speaker_ids)
# --- [BARU] Buat fungsi wrapper untuk Speech-to-Speech ---
def stt_tts_fn(audio_filepath, speaker, speed):
if audio_filepath is None:
return "Error: Audio not provided.", None, "Please record or upload audio first."
print("Transcribing audio...")
# Ubah audio ke teks
transcription_result = stt_pipeline(audio_filepath)
transcribed_text = transcription_result['text']
print(f"Transcribed text: {transcribed_text}")
if not transcribed_text.strip():
return "Error: Could not transcribe audio.", None, "No text detected in audio."
print("Generating speech from transcribed text...")
# Masukkan teks hasil transkripsi ke fungsi TTS yang sudah ada
status, audio_output = tts_fn(transcribed_text, speaker, speed)
print("Speech generation complete.")
# Kembalikan status, audio, dan teks hasil transkripsi untuk ditampilkan di UI
return status, audio_output, transcribed_text
app = gr.Blocks(css=css)
# --- [DIUBAH] Struktur UI menggunakan Tabs ---
with app:
gr.Markdown("# BlueArchive Arona TTS Using VITS Model\n"
"\n\n")
with gr.Column():
gr.Markdown(f"## {name}\n\n"
f"lang: {lang}")
with gr.Tabs():
# Tab 1: Antarmuka Teks-ke-Suara (Asli)
with gr.TabItem("Text to Speech"):
tts_input_text = gr.TextArea(label="Text (5000 words limitation)", value=example)
tts_speaker_text = gr.Dropdown(label="Speaker", choices=speakers, type="index", value=speakers[0])
tts_speed_text = gr.Slider(label="Speed", value=1, minimum=0.1, maximum=2, step=0.1)
tts_submit_text = gr.Button("Generate from Text", variant="primary")
# Tab 2: Antarmuka Suara-ke-Suara (Baru)
with gr.TabItem("Voice to Speech"):
audio_input = gr.Audio(type="filepath", label="Record or Upload Voice")
tts_speaker_audio = gr.Dropdown(label="Speaker", choices=speakers, type="index", value=speakers[0])
tts_speed_audio = gr.Slider(label="Speed", value=1, minimum=0.1, maximum=2, step=0.1)
transcribed_text_output = gr.Textbox(label="Transcribed Text", interactive=False)
tts_submit_audio = gr.Button("Generate from Voice", variant="primary")
# Output yang digunakan bersama oleh kedua tab
gr.Markdown("---")
gr.Markdown("### Output")
output_message = gr.Textbox(label="Output Message")
output_audio = gr.Audio(label="Output Audio")
# Hubungkan tombol dengan fungsinya masing-masing
tts_submit_text.click(
tts_fn,
[tts_input_text, tts_speaker_text, tts_speed_text],
[output_message, output_audio]
)
tts_submit_audio.click(
stt_tts_fn,
[audio_input, tts_speaker_audio, tts_speed_audio],
[output_message, output_audio, transcribed_text_output]
)
app.queue(concurrency_count=3).launch(show_api=False) |