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import csv |
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import datetime |
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import os |
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import re |
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import time |
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import uuid |
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from io import StringIO |
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import gradio as gr |
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import spaces |
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import torch |
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import torchaudio |
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from huggingface_hub import HfApi, hf_hub_download, snapshot_download |
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from TTS.tts.configs.xtts_config import XttsConfig |
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from TTS.tts.models.xtts import Xtts |
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from vinorm import TTSnorm |
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os.system("python -m unidic download") |
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HF_TOKEN = os.environ.get("HF_TOKEN") |
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api = HfApi(token=HF_TOKEN) |
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print("Downloading if not downloaded viXTTS") |
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checkpoint_dir = "model/" |
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repo_id = "capleaf/viXTTS" |
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use_deepspeed = False |
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os.makedirs(checkpoint_dir, exist_ok=True) |
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required_files = ["model.pth", "config.json", "vocab.json", "speakers_xtts.pth"] |
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files_in_dir = os.listdir(checkpoint_dir) |
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if not all(file in files_in_dir for file in required_files): |
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snapshot_download( |
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repo_id=repo_id, |
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repo_type="model", |
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local_dir=checkpoint_dir, |
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) |
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hf_hub_download( |
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repo_id="coqui/XTTS-v2", |
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filename="speakers_xtts.pth", |
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local_dir=checkpoint_dir, |
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) |
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xtts_config = os.path.join(checkpoint_dir, "config.json") |
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config = XttsConfig() |
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config.load_json(xtts_config) |
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MODEL = Xtts.init_from_config(config) |
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MODEL.load_checkpoint( |
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config, checkpoint_dir=checkpoint_dir, use_deepspeed=use_deepspeed |
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) |
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if torch.cuda.is_available(): |
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MODEL.cuda() |
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supported_languages = config.languages |
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if not "vi" in supported_languages: |
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supported_languages.append("vi") |
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def normalize_vietnamese_text(text): |
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text = ( |
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TTSnorm(text, unknown=False, lower=False, rule=True) |
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.replace("..", ".") |
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.replace("!.", "!") |
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.replace("?.", "?") |
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.replace(" .", ".") |
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.replace(" ,", ",") |
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.replace('"', "") |
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.replace("'", "") |
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.replace("AI", "Ây Ai") |
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.replace("A.I", "Ây Ai") |
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) |
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return text |
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def calculate_keep_len(text, lang): |
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"""Simple hack for short sentences""" |
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if lang in ["ja", "zh-cn"]: |
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return -1 |
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word_count = len(text.split()) |
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num_punct = text.count(".") + text.count("!") + text.count("?") + text.count(",") |
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if word_count < 5: |
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return 15000 * word_count + 2000 * num_punct |
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elif word_count < 10: |
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return 13000 * word_count + 2000 * num_punct |
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return -1 |
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@spaces.GPU |
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def predict( |
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prompt, |
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language, |
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audio_file_pth, |
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normalize_text=True, |
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): |
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if language not in supported_languages: |
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metrics_text = gr.Warning( |
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f"Language you put {language} in is not in is not in our Supported Languages, please choose from dropdown" |
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) |
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return (None, metrics_text) |
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speaker_wav = audio_file_pth |
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if len(prompt) < 2: |
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metrics_text = gr.Warning("Please give a longer prompt text") |
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return (None, metrics_text) |
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if len(prompt) > 250000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000: |
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metrics_text = gr.Warning( |
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str(len(prompt)) |
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+ " characters.\n" |
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+ "Your prompt is too long, please keep it under 250 characters\n" |
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+ "Văn bản quá dài, vui lòng giữ dưới 250 ký tự." |
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) |
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return (None, metrics_text) |
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try: |
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metrics_text = "" |
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t_latent = time.time() |
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try: |
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( |
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gpt_cond_latent, |
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speaker_embedding, |
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) = MODEL.get_conditioning_latents( |
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audio_path=speaker_wav, |
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gpt_cond_len=30, |
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gpt_cond_chunk_len=4, |
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max_ref_length=60, |
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) |
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except Exception as e: |
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print("Speaker encoding error", str(e)) |
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metrics_text = gr.Warning( |
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"It appears something wrong with reference, did you unmute your microphone?" |
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) |
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return (None, metrics_text) |
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prompt = re.sub("([^\x00-\x7F]|\w)(\.|\。|\?)", r"\1 \2\2", prompt) |
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if normalize_text and language == "vi": |
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prompt = normalize_vietnamese_text(prompt) |
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print("I: Generating new audio...") |
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t0 = time.time() |
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out = MODEL.inference( |
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prompt, |
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language, |
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gpt_cond_latent, |
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speaker_embedding, |
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repetition_penalty=5.0, |
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temperature=0.75, |
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enable_text_splitting=True, |
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) |
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inference_time = time.time() - t0 |
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print(f"I: Time to generate audio: {round(inference_time*1000)} milliseconds") |
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metrics_text += ( |
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f"Time to generate audio: {round(inference_time*1000)} milliseconds\n" |
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) |
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real_time_factor = (time.time() - t0) / out["wav"].shape[-1] * 24000 |
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print(f"Real-time factor (RTF): {real_time_factor}") |
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metrics_text += f"Real-time factor (RTF): {real_time_factor:.2f}\n" |
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keep_len = calculate_keep_len(prompt, language) |
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out["wav"] = out["wav"][:keep_len] |
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torchaudio.save("output.wav", torch.tensor(out["wav"]).unsqueeze(0), 24000) |
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except RuntimeError as e: |
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if "device-side assert" in str(e): |
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print( |
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f"Exit due to: Unrecoverable exception caused by language:{language} prompt:{prompt}", |
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flush=True, |
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) |
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gr.Warning("Unhandled Exception encounter, please retry in a minute") |
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print("Cuda device-assert Runtime encountered need restart") |
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error_time = datetime.datetime.now().strftime("%d-%m-%Y-%H:%M:%S") |
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error_data = [ |
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error_time, |
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prompt, |
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language, |
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audio_file_pth, |
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] |
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error_data = [str(e) if type(e) != str else e for e in error_data] |
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print(error_data) |
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print(speaker_wav) |
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write_io = StringIO() |
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csv.writer(write_io).writerows([error_data]) |
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csv_upload = write_io.getvalue().encode() |
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filename = error_time + "_" + str(uuid.uuid4()) + ".csv" |
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print("Writing error csv") |
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error_api = HfApi() |
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error_api.upload_file( |
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path_or_fileobj=csv_upload, |
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path_in_repo=filename, |
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repo_id="coqui/xtts-flagged-dataset", |
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repo_type="dataset", |
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) |
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print("Writing error reference audio") |
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speaker_filename = error_time + "_reference_" + str(uuid.uuid4()) + ".wav" |
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error_api = HfApi() |
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error_api.upload_file( |
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path_or_fileobj=speaker_wav, |
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path_in_repo=speaker_filename, |
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repo_id="coqui/xtts-flagged-dataset", |
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repo_type="dataset", |
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) |
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space = api.get_space_runtime(repo_id=repo_id) |
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if space.stage != "BUILDING": |
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api.restart_space(repo_id=repo_id) |
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else: |
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print("TRIED TO RESTART but space is building") |
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else: |
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if "Failed to decode" in str(e): |
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print("Speaker encoding error", str(e)) |
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metrics_text = gr.Warning( |
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metrics_text="It appears something wrong with reference, did you unmute your microphone?" |
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) |
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else: |
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print("RuntimeError: non device-side assert error:", str(e)) |
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metrics_text = gr.Warning( |
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"Something unexpected happened please retry again." |
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) |
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return (None, metrics_text) |
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return ("output.wav", metrics_text) |
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with gr.Blocks(analytics_enabled=False) as demo: |
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with gr.Row(): |
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with gr.Column(): |
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gr.Markdown( |
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""" |
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# viXTTS Demo ✨ |
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- Github: https://github.com/thinhlpg/vixtts-demo/ |
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- viVoice: https://github.com/thinhlpg/viVoice |
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""" |
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) |
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with gr.Column(): |
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pass |
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with gr.Row(): |
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with gr.Column(): |
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input_text_gr = gr.Textbox( |
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label="Text Prompt (Văn bản cần đọc)", |
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info="Mỗi câu nên từ 10 từ trở lên. Tối đa 250 ký tự (khoảng 2 - 3 câu).", |
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value="Xin chào, tôi là một mô hình chuyển đổi văn bản thành giọng nói tiếng Việt.", |
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) |
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language_gr = gr.Dropdown( |
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label="Language (Ngôn ngữ)", |
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choices=[ |
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"vi", |
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"en", |
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"es", |
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"fr", |
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"de", |
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"it", |
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"pt", |
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"pl", |
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"tr", |
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"ru", |
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"nl", |
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"cs", |
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"ar", |
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"zh-cn", |
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"ja", |
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"ko", |
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"hu", |
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"hi", |
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], |
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max_choices=1, |
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value="vi", |
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) |
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normalize_text = gr.Checkbox( |
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label="Chuẩn hóa văn bản tiếng Việt", |
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info="Normalize Vietnamese text", |
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value=True, |
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) |
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ref_gr = gr.Audio( |
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label="Reference Audio (Giọng mẫu)", |
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type="filepath", |
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value="model/samples/nu-luu-loat.wav", |
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) |
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tts_button = gr.Button( |
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"Đọc 🗣️🔥", |
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elem_id="send-btn", |
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visible=True, |
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variant="primary", |
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) |
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with gr.Column(): |
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audio_gr = gr.Audio(label="Synthesised Audio", autoplay=True) |
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out_text_gr = gr.Text(label="Metrics") |
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tts_button.click( |
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predict, |
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[ |
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input_text_gr, |
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language_gr, |
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ref_gr, |
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normalize_text, |
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], |
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outputs=[audio_gr, out_text_gr], |
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api_name="predict", |
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) |
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demo.queue() |
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demo.launch(debug=True, show_api=True, share=True) |
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