import os import time import uuid import torch import torchaudio # download for mecab os.system("python -m unidic download") # By using XTTS you agree to CPML license https://coqui.ai/cpml os.environ["COQUI_TOS_AGREED"] = "1" import csv import datetime import re from io import StringIO import gradio as gr # langid is used to detect language for longer text # Most users expect text to be their own language, there is checkbox to disable it import langid from huggingface_hub import hf_hub_download, snapshot_download from TTS.api import TTS from TTS.tts.configs.xtts_config import XttsConfig from TTS.tts.models.xtts import Xtts from underthesea import sent_tokenize from unidecode import unidecode from vinorm import TTSnorm HF_TOKEN = os.environ.get("HF_TOKEN") from huggingface_hub import HfApi # will use api to restart space on a unrecoverable error api = HfApi(token=HF_TOKEN) repo_id = "coqui/xtts" # This will trigger downloading model print("Downloading if not downloaded Coqui XTTS V2") checkpoint_dir = "model/" repo_id = "capleaf/viXTTS" use_deepspeed = False os.makedirs(checkpoint_dir, exist_ok=True) required_files = ["model.pth", "config.json", "vocab.json", "speakers_xtts.pth"] files_in_dir = os.listdir(checkpoint_dir) if not all(file in files_in_dir for file in required_files): snapshot_download( repo_id=repo_id, repo_type="model", local_dir=checkpoint_dir, ) hf_hub_download( repo_id="coqui/XTTS-v2", filename="speakers_xtts.pth", local_dir=checkpoint_dir, ) xtts_config = os.path.join(checkpoint_dir, "config.json") config = XttsConfig() config.load_json(xtts_config) MODEL = Xtts.init_from_config(config) MODEL.load_checkpoint( config, checkpoint_dir=checkpoint_dir, use_deepspeed=use_deepspeed ) if torch.cuda.is_available(): MODEL.cuda() supported_languages = config.languages if not "vi" in supported_languages: supported_languages.append("vi") def predict( prompt, language, audio_file_pth, mic_file_path, use_mic, voice_cleanup, no_lang_auto_detect, agree, ): if agree == True: if language not in supported_languages: gr.Warning( f"Language you put {language} in is not in is not in our Supported Languages, please choose from dropdown" ) return ( None, None, None, None, ) language_predicted = langid.classify(prompt)[ 0 ].strip() # strip need as there is space at end! # tts expects chinese as zh-cn if language_predicted == "zh": # we use zh-cn language_predicted = "zh-cn" print(f"Detected language:{language_predicted}, Chosen language:{language}") # After text character length 15 trigger language detection if len(prompt) > 15: # allow any language for short text as some may be common # If user unchecks language autodetection it will not trigger # You may remove this completely for own use if language_predicted != language and not no_lang_auto_detect: # Please duplicate and remove this check if you really want this # Or auto-detector fails to identify language (which it can on pretty short text or mixed text) gr.Warning( f"It looks like your text isn’t the language you chose , if you’re sure the text is the same language you chose, please check disable language auto-detection checkbox" ) return ( None, None, None, None, ) if use_mic == True: if mic_file_path is not None: speaker_wav = mic_file_path else: gr.Warning( "Please record your voice with Microphone, or uncheck Use Microphone to use reference audios" ) return ( None, None, None, None, ) else: speaker_wav = audio_file_pth # Filtering for microphone input, as it has BG noise, maybe silence in beginning and end # This is fast filtering not perfect # Apply all on demand lowpassfilter = denoise = trim = loudness = True if lowpassfilter: lowpass_highpass = "lowpass=8000,highpass=75," else: lowpass_highpass = "" if trim: # better to remove silence in beginning and end for microphone trim_silence = "areverse,silenceremove=start_periods=1:start_silence=0:start_threshold=0.02,areverse,silenceremove=start_periods=1:start_silence=0:start_threshold=0.02," else: trim_silence = "" speaker_wav = speaker_wav if len(prompt) < 2: gr.Warning("Please give a longer prompt text") return ( None, None, None, None, ) if len(prompt) > 200: gr.Warning( "Text length limited to 200 characters for this demo, please try shorter text. You can clone this space and edit code for your own usage" ) return ( None, None, None, None, ) try: metrics_text = "" t_latent = time.time() # note diffusion_conditioning not used on hifigan (default mode), it will be empty but need to pass it to model.inference try: ( gpt_cond_latent, speaker_embedding, ) = MODEL.get_conditioning_latents( audio_path=speaker_wav, gpt_cond_len=30, gpt_cond_chunk_len=4, max_ref_length=60, ) except Exception as e: print("Speaker encoding error", str(e)) gr.Warning( "It appears something wrong with reference, did you unmute your microphone?" ) return ( None, None, None, None, ) latent_calculation_time = time.time() - t_latent # metrics_text=f"Embedding calculation time: {latent_calculation_time:.2f} seconds\n" # temporary comma fix prompt = re.sub("([^\x00-\x7F]|\w)(\.|\。|\?)", r"\1 \2\2", prompt) wav_chunks = [] ## Direct mode print("I: Generating new audio...") t0 = time.time() out = MODEL.inference( prompt, language, gpt_cond_latent, speaker_embedding, repetition_penalty=5.0, temperature=0.75, ) inference_time = time.time() - t0 print( f"I: Time to generate audio: {round(inference_time*1000)} milliseconds" ) metrics_text += ( f"Time to generate audio: {round(inference_time*1000)} milliseconds\n" ) real_time_factor = (time.time() - t0) / out["wav"].shape[-1] * 24000 print(f"Real-time factor (RTF): {real_time_factor}") metrics_text += f"Real-time factor (RTF): {real_time_factor:.2f}\n" torchaudio.save("output.wav", torch.tensor(out["wav"]).unsqueeze(0), 24000) """ print("I: Generating new audio in streaming mode...") t0 = time.time() chunks = model.inference_stream( prompt, language, gpt_cond_latent, speaker_embedding, repetition_penalty=7.0, temperature=0.85, ) first_chunk = True for i, chunk in enumerate(chunks): if first_chunk: first_chunk_time = time.time() - t0 metrics_text += f"Latency to first audio chunk: {round(first_chunk_time*1000)} milliseconds\n" first_chunk = False wav_chunks.append(chunk) print(f"Received chunk {i} of audio length {chunk.shape[-1]}") inference_time = time.time() - t0 print( f"I: Time to generate audio: {round(inference_time*1000)} milliseconds" ) #metrics_text += ( # f"Time to generate audio: {round(inference_time*1000)} milliseconds\n" #) wav = torch.cat(wav_chunks, dim=0) print(wav.shape) real_time_factor = (time.time() - t0) / wav.shape[0] * 24000 print(f"Real-time factor (RTF): {real_time_factor}") metrics_text += f"Real-time factor (RTF): {real_time_factor:.2f}\n" torchaudio.save("output.wav", wav.squeeze().unsqueeze(0).cpu(), 24000) """ except RuntimeError as e: if "device-side assert" in str(e): # cannot do anything on cuda device side error, need tor estart print( f"Exit due to: Unrecoverable exception caused by language:{language} prompt:{prompt}", flush=True, ) gr.Warning("Unhandled Exception encounter, please retry in a minute") print("Cuda device-assert Runtime encountered need restart") if not DEVICE_ASSERT_DETECTED: DEVICE_ASSERT_DETECTED = 1 DEVICE_ASSERT_PROMPT = prompt DEVICE_ASSERT_LANG = language # just before restarting save what caused the issue so we can handle it in future # Uploading Error data only happens for unrecovarable error error_time = datetime.datetime.now().strftime("%d-%m-%Y-%H:%M:%S") error_data = [ error_time, prompt, language, audio_file_pth, mic_file_path, use_mic, voice_cleanup, no_lang_auto_detect, agree, ] error_data = [str(e) if type(e) != str else e for e in error_data] print(error_data) print(speaker_wav) write_io = StringIO() csv.writer(write_io).writerows([error_data]) csv_upload = write_io.getvalue().encode() filename = error_time + "_" + str(uuid.uuid4()) + ".csv" print("Writing error csv") error_api = HfApi() error_api.upload_file( path_or_fileobj=csv_upload, path_in_repo=filename, repo_id="coqui/xtts-flagged-dataset", repo_type="dataset", ) # speaker_wav print("Writing error reference audio") speaker_filename = ( error_time + "_reference_" + str(uuid.uuid4()) + ".wav" ) error_api = HfApi() error_api.upload_file( path_or_fileobj=speaker_wav, path_in_repo=speaker_filename, repo_id="coqui/xtts-flagged-dataset", repo_type="dataset", ) # HF Space specific.. This error is unrecoverable need to restart space space = api.get_space_runtime(repo_id=repo_id) if space.stage != "BUILDING": api.restart_space(repo_id=repo_id) else: print("TRIED TO RESTART but space is building") else: if "Failed to decode" in str(e): print("Speaker encoding error", str(e)) gr.Warning( "It appears something wrong with reference, did you unmute your microphone?" ) else: print("RuntimeError: non device-side assert error:", str(e)) gr.Warning("Something unexpected happened please retry again.") return ( None, None, None, None, ) return ( gr.make_waveform( audio="output.wav", ), "output.wav", metrics_text, speaker_wav, ) else: gr.Warning("Please accept the Terms & Condition!") return ( None, None, None, None, ) title = "viXTTS Demo" description = """
This demo is currently running **XTTS v2.0.3** XTTS is a multilingual text-to-speech and voice-cloning model. This demo features zero-shot voice cloning, however, you can fine-tune XTTS for better results. Leave a star 🌟 on Github 🐸TTS, where our open-source inference and training code lives.
Supported languages: Arabic: ar, Brazilian Portuguese: pt , Mandarin Chinese: zh-cn, Czech: cs, Dutch: nl, English: en, French: fr, German: de, Italian: it, Polish: pl, Russian: ru, Spanish: es, Turkish: tr, Japanese: ja, Korean: ko, Hungarian: hu, Hindi: hi
""" article = """ """ with gr.Blocks(analytics_enabled=False) as demo: with gr.Row(): with gr.Column(): gr.Markdown( """ 😳 Burh """ ) with gr.Column(): # placeholder to align the image pass with gr.Row(): with gr.Column(): gr.Markdown(description) with gr.Row(): with gr.Column(): input_text_gr = gr.Textbox( label="Text Prompt", info="One or two sentences at a time is better. Up to 200 text characters.", value="Hi there, I'm your new voice clone. Try your best to upload quality audio.", ) language_gr = gr.Dropdown( label="Language", info="Select an output language for the synthesised speech", choices=[ "vi", "en", "es", "fr", "de", "it", "pt", "pl", "tr", "ru", "nl", "cs", "ar", "zh-cn", "ja", "ko", "hu", "hi", ], max_choices=1, value="vi", ) ref_gr = gr.Audio( label="Reference Audio", info="Click on the ✎ button to upload your own target speaker audio", type="filepath", value="model/samples/nu-luu-loat.wav", ) mic_gr = gr.Audio( source="microphone", type="filepath", info="Use your microphone to record audio", label="Use Microphone for Reference", ) use_mic_gr = gr.Checkbox( label="Use Microphone", value=False, info="Notice: Microphone input may not work properly under traffic", ) clean_ref_gr = gr.Checkbox( label="Cleanup Reference Voice", value=False, info="This check can improve output if your microphone or reference voice is noisy", ) auto_det_lang_gr = gr.Checkbox( label="Do not use language auto-detect", value=False, info="Check to disable language auto-detection", ) tos_gr = gr.Checkbox( label="Agree", value=False, info="I agree to the terms of the CPML: https://coqui.ai/cpml", ) tts_button = gr.Button("Send", elem_id="send-btn", visible=True) with gr.Column(): video_gr = gr.Video(label="Waveform Visual") audio_gr = gr.Audio(label="Synthesised Audio", autoplay=True) out_text_gr = gr.Text(label="Metrics") ref_audio_gr = gr.Audio(label="Reference Audio Used") tts_button.click( predict, [ input_text_gr, language_gr, ref_gr, mic_gr, use_mic_gr, clean_ref_gr, auto_det_lang_gr, tos_gr, ], outputs=[video_gr, audio_gr, out_text_gr, ref_audio_gr], ) demo.queue() demo.launch(debug=True, show_api=True)