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Update app.py
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app.py
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import gradio as gr
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import soundfile as sf
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import torch
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import sys, os
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from transformers import MarianMTModel, MarianTokenizer, pipeline
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from huggingface_hub import snapshot_download
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# --------------------------
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# Download
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# --------------------------
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sys.path.append(repo_path)
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from indextts.infer import IndexTTS
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#
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# --------------------------
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# Translation models
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"Spanish β English": "Helsinki-NLP/opus-mt-es-en",
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"English β Spanish": "Helsinki-NLP/opus-mt-en-es"
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}
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current_model_name = language_models["Spanish β English"]
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tokenizer = MarianTokenizer.from_pretrained(current_model_name)
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model = MarianMTModel.from_pretrained(current_model_name)
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# Speech-to-text (ASR)
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asr = pipeline("automatic-speech-recognition", model="openai/whisper-small")
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# --------------------------
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#
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# --------------------------
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def text_to_speech(text,
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def translate_with_voice(audio, lang_pair, ref_voice):
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# 1) Speech to text
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text_input = asr(
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# 2) Translation
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if language_models[lang_pair] != current_model_name:
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current_model_name = language_models[lang_pair]
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tokenizer = MarianTokenizer.from_pretrained(current_model_name)
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model = MarianMTModel.from_pretrained(current_model_name)
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inputs = tokenizer(text_input, return_tensors="pt", padding=True)
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translated_text = tokenizer.decode(
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# 3) Text to speech
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return translated_text,
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# --------------------------
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# Gradio UI
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# --------------------------
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with gr.Blocks() as demo:
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gr.Markdown("#
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with gr.Row():
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with gr.Column():
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with gr.Column():
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text_output = gr.Textbox(label="Translated Text")
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audio_output = gr.Audio(label="π Translated Audio", type="
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btn.click(
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fn=translate_with_voice,
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outputs=[text_output, audio_output]
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)
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import os
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import sys
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import tempfile
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import gradio as gr
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import soundfile as sf
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import torch
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from huggingface_hub import snapshot_download
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from transformers import MarianMTModel, MarianTokenizer, pipeline
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# --------------------------
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# Download IndexTTS repo from Hugging Face
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# --------------------------
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CHECKPOINTS_DIR = os.path.abspath("checkpoints")
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os.makedirs(CHECKPOINTS_DIR, exist_ok=True)
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repo_path = snapshot_download(
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repo_id="mlx-community/IndexTTS", # Correct repo
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local_dir=CHECKPOINTS_DIR,
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local_dir_use_symlinks=False,
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allow_patterns=[
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"config.yaml",
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"bpe.model",
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"unigram_12000.vocab",
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"gpt.pth",
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"bigvgan_generator.pth",
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"bigvgan_discriminator.pth",
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"dvae.pth",
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],
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)
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sys.path.append(repo_path)
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from indextts.infer import IndexTTS
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# --------------------------
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# Initialize TTS safely
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# --------------------------
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_tts = None
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def get_tts():
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global _tts
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if _tts is None:
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try:
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_tts = IndexTTS(model_dir=repo_path, cfg_path=os.path.join(repo_path, "config.yaml"))
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except FileNotFoundError as e:
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print("Error loading IndexTTS:", e)
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raise gr.Error("IndexTTS model files not found!")
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return _tts
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# Limit CPU threads (important for Spaces)
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torch.set_num_threads(1)
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os.environ["OMP_NUM_THREADS"] = "1"
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os.environ["MKL_NUM_THREADS"] = "1"
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# --------------------------
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# Translation models
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"Spanish β English": "Helsinki-NLP/opus-mt-es-en",
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"English β Spanish": "Helsinki-NLP/opus-mt-en-es"
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}
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current_model_name = None
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tokenizer = None
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model = None
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def load_translation_model(lang_pair):
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global current_model_name, tokenizer, model
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if language_models[lang_pair] != current_model_name:
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current_model_name = language_models[lang_pair]
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tokenizer = MarianTokenizer.from_pretrained(current_model_name)
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model = MarianMTModel.from_pretrained(current_model_name)
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# --------------------------
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# Speech-to-text (ASR)
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# --------------------------
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asr = pipeline("automatic-speech-recognition", model="openai/whisper-small")
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# --------------------------
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# Core functions
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# --------------------------
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def text_to_speech(text, ref_voice_path):
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"""
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Convert text to speech using IndexTTS.
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Returns a temporary WAV file path.
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"""
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tts = get_tts()
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with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as tmp:
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out_path = tmp.name
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tts.infer(ref_voice_path, text, out_path)
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return out_path
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def translate_with_voice(audio, lang_pair, ref_voice):
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# Handle Gradio sending numpy array + sample_rate
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if isinstance(audio, tuple):
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audio_path = audio[0] # (filepath, sample_rate) or (sample_rate, array)
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else:
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audio_path = audio
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# 1) Speech to text
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text_input = asr(audio_path)["text"]
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# 2) Translation
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load_translation_model(lang_pair)
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inputs = tokenizer(text_input, return_tensors="pt", padding=True)
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translated_ids = model.generate(**inputs)
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translated_text = tokenizer.decode(translated_ids[0], skip_special_tokens=True)
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# 3) Text to speech
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out_wav_path = text_to_speech(translated_text, ref_voice)
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return translated_text, out_wav_path
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# --------------------------
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# Gradio UI
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# --------------------------
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title = "π£ Voice-Cloned Translator (English β Spanish)"
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description = """
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Upload a short **reference voice** (5β10s, clean speech works best) and speak into the microphone.
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This Space uses **IndexTTS** for zero-shot voice cloning and **Hugging Face models** for translation.
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"""
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with gr.Blocks() as demo:
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gr.Markdown(f"# {title}\n{description}")
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with gr.Row():
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with gr.Column():
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with gr.Column():
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text_output = gr.Textbox(label="Translated Text")
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audio_output = gr.Audio(label="π Translated Audio", type="filepath")
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btn.click(
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fn=translate_with_voice,
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outputs=[text_output, audio_output]
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)
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# Preload TTS on startup
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def _startup():
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try:
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get_tts()
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except Exception as e:
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print("Warmup failed:", e)
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if __name__ == "__main__":
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_startup()
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demo.launch(server_name="0.0.0.0", server_port=7860)
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