vixtts / app.py
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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 = """
<br/>
This demo is currently running **XTTS v2.0.3** <a href="https://huggingface.co/coqui/XTTS-v2">XTTS</a> 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 <a href="https://github.com/coqui-ai/TTS">🐸TTS</a>, where our open-source inference and training code lives.
<br/>
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
<br/>
"""
article = """
"""
examples = [
[
"Once when I was six years old I saw a magnificent picture",
"en",
"examples/female.wav",
None,
False,
False,
False,
True,
],
[
"Lorsque j'avais six ans j'ai vu, une fois, une magnifique image",
"fr",
"examples/male.wav",
None,
False,
False,
False,
True,
],
[
"Als ich sechs war, sah ich einmal ein wunderbares Bild",
"de",
"examples/female.wav",
None,
False,
False,
False,
True,
],
[
"Cuando tenía seis años, vi una vez una imagen magnífica",
"es",
"examples/male.wav",
None,
False,
False,
False,
True,
],
[
"Quando eu tinha seis anos eu vi, uma vez, uma imagem magnífica",
"pt",
"examples/female.wav",
None,
False,
False,
False,
True,
],
[
"Kiedy miałem sześć lat, zobaczyłem pewnego razu wspaniały obrazek",
"pl",
"examples/male.wav",
None,
False,
False,
False,
True,
],
[
"Un tempo lontano, quando avevo sei anni, vidi un magnifico disegno",
"it",
"examples/female.wav",
None,
False,
False,
False,
True,
],
[
"Bir zamanlar, altı yaşındayken, muhteşem bir resim gördüm",
"tr",
"examples/female.wav",
None,
False,
False,
False,
True,
],
[
"Когда мне было шесть лет, я увидел однажды удивительную картинку",
"ru",
"examples/female.wav",
None,
False,
False,
False,
True,
],
[
"Toen ik een jaar of zes was, zag ik op een keer een prachtige plaat",
"nl",
"examples/male.wav",
None,
False,
False,
False,
True,
],
[
"Když mi bylo šest let, viděl jsem jednou nádherný obrázek",
"cs",
"examples/female.wav",
None,
False,
False,
False,
True,
],
[
"当我还只有六岁的时候, 看到了一副精彩的插画",
"zh-cn",
"examples/female.wav",
None,
False,
False,
False,
True,
],
[
"かつて 六歳のとき、素晴らしい絵を見ました",
"ja",
"examples/female.wav",
None,
False,
True,
False,
True,
],
[
"한번은 내가 여섯 살이었을 때 멋진 그림을 보았습니다.",
"ko",
"examples/female.wav",
None,
False,
True,
False,
True,
],
[
"Egyszer hat éves koromban láttam egy csodálatos képet",
"hu",
"examples/male.wav",
None,
False,
True,
False,
True,
],
]
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="examples/female.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")
with gr.Row():
gr.Examples(
examples,
label="Examples",
inputs=[
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],
fn=predict,
cache_examples=False,
)
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)