91 lines
3.8 KiB
Python
91 lines
3.8 KiB
Python
import os
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import torch
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import soundfile as sf
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from tiny_tts.text.english import normalize_text, grapheme_to_phoneme
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from tiny_tts.text import phonemes_to_ids
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from tiny_tts.nn import commons
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from tiny_tts.models.synthesizer import VoiceSynthesizer
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from tiny_tts.text.symbols import symbols
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from tiny_tts.utils.config import (
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SAMPLING_RATE, SEGMENT_FRAMES, ADD_BLANK, SPEC_CHANNELS,
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N_SPEAKERS, SPK2ID, MODEL_PARAMS,
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)
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from tiny_tts.infer import load_engine
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class TinyTTS:
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def __init__(self, checkpoint_path=None, device=None):
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if device is None:
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self.device = 'cuda' if torch.cuda.is_available() else 'cpu'
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else:
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self.device = device
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if checkpoint_path is None:
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# Look for default checkpoint in pacakage
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pkg_dir = os.path.dirname(os.path.abspath(__file__))
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default_ckpt = os.path.join(os.path.dirname(pkg_dir), "checkpoints", "G.pth")
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# 2. Check HuggingFace Cache / Download
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if not os.path.exists(default_ckpt):
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try:
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from huggingface_hub import hf_hub_download
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print("Downloading/Loading checkpoint from Hugging Face Hub (backtracking/tiny-tts)...")
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default_ckpt = hf_hub_download(repo_id="backtracking/tiny-tts", filename="G.pth")
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except ImportError:
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raise ImportError("huggingface_hub is required to auto-download the model. Run: pip install huggingface_hub")
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except Exception as e:
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raise ValueError(f"Failed to download checkpoint from Hugging Face: {e}")
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checkpoint_path = default_ckpt
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self.model = load_engine(checkpoint_path, self.device)
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def speak(self, text, output_path="output.wav", speaker="MALE", speed=1.0):
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"""Synthesize text to speech and save to output_path."""
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print(f"Synthesizing: {text}")
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# Normalize text
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normalized = normalize_text(text)
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# Phonemize
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phones, tones, word2ph = grapheme_to_phoneme(normalized)
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# Convert to sequence
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phone_ids, tone_ids, lang_ids = phonemes_to_ids(phones, tones, "EN")
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# Add blanks
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if ADD_BLANK:
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phone_ids = commons.insert_blanks(phone_ids, 0)
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tone_ids = commons.insert_blanks(tone_ids, 0)
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lang_ids = commons.insert_blanks(lang_ids, 0)
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x = torch.LongTensor(phone_ids).unsqueeze(0).to(self.device)
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x_lengths = torch.LongTensor([len(phone_ids)]).to(self.device)
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tone = torch.LongTensor(tone_ids).unsqueeze(0).to(self.device)
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language = torch.LongTensor(lang_ids).unsqueeze(0).to(self.device)
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# Speaker ID
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if speaker not in SPK2ID:
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print(f"Warning: Speaker '{speaker}' not found, using ID 0. Available: {list(SPK2ID.keys())}")
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sid = torch.LongTensor([0]).to(self.device)
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else:
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sid = torch.LongTensor([SPK2ID[speaker]]).to(self.device)
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# BERT features (disabled - using zero tensors)
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bert = torch.zeros(1024, len(phone_ids)).to(self.device).unsqueeze(0)
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ja_bert = torch.zeros(768, len(phone_ids)).to(self.device).unsqueeze(0)
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# speed > 1.0 = faster speech, < 1.0 = slower speech
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length_scale = 1.0 / speed
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with torch.no_grad():
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audio, *_ = self.model.infer(
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x, x_lengths, sid, tone, language, bert, ja_bert,
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noise_scale=0.667,
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noise_scale_w=0.8,
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length_scale=length_scale
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)
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audio_np = audio[0, 0].cpu().numpy()
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sf.write(output_path, audio_np, SAMPLING_RATE)
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print(f"Saved audio to {output_path}")
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return audio_np
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