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import os,sys | |
from transformers import pipeline | |
import gradio as gr | |
import torch | |
import click | |
import torchaudio | |
from glob import glob | |
import librosa | |
import numpy as np | |
from scipy.io import wavfile | |
from tqdm import tqdm | |
import shutil | |
import time | |
import json | |
from datasets import Dataset | |
from model.utils import convert_char_to_pinyin | |
import signal | |
import psutil | |
import platform | |
import subprocess | |
from datasets.arrow_writer import ArrowWriter | |
from datasets import load_dataset, load_from_disk | |
import json | |
training_process = None | |
system = platform.system() | |
python_executable = sys.executable or "python" | |
path_data="data" | |
device = ( | |
"cuda" | |
if torch.cuda.is_available() | |
else "mps" if torch.backends.mps.is_available() else "cpu" | |
) | |
pipe = None | |
# Load metadata | |
def get_audio_duration(audio_path): | |
"""Calculate the duration of an audio file.""" | |
audio, sample_rate = torchaudio.load(audio_path) | |
num_channels = audio.shape[0] | |
return audio.shape[1] / (sample_rate * num_channels) | |
def clear_text(text): | |
"""Clean and prepare text by lowering the case and stripping whitespace.""" | |
return text.lower().strip() | |
def get_rms(y,frame_length=2048,hop_length=512,pad_mode="constant",): # https://github.com/RVC-Boss/GPT-SoVITS/blob/main/tools/slicer2.py | |
padding = (int(frame_length // 2), int(frame_length // 2)) | |
y = np.pad(y, padding, mode=pad_mode) | |
axis = -1 | |
# put our new within-frame axis at the end for now | |
out_strides = y.strides + tuple([y.strides[axis]]) | |
# Reduce the shape on the framing axis | |
x_shape_trimmed = list(y.shape) | |
x_shape_trimmed[axis] -= frame_length - 1 | |
out_shape = tuple(x_shape_trimmed) + tuple([frame_length]) | |
xw = np.lib.stride_tricks.as_strided(y, shape=out_shape, strides=out_strides) | |
if axis < 0: | |
target_axis = axis - 1 | |
else: | |
target_axis = axis + 1 | |
xw = np.moveaxis(xw, -1, target_axis) | |
# Downsample along the target axis | |
slices = [slice(None)] * xw.ndim | |
slices[axis] = slice(0, None, hop_length) | |
x = xw[tuple(slices)] | |
# Calculate power | |
power = np.mean(np.abs(x) ** 2, axis=-2, keepdims=True) | |
return np.sqrt(power) | |
class Slicer: # https://github.com/RVC-Boss/GPT-SoVITS/blob/main/tools/slicer2.py | |
def __init__( | |
self, | |
sr: int, | |
threshold: float = -40.0, | |
min_length: int = 2000, | |
min_interval: int = 300, | |
hop_size: int = 20, | |
max_sil_kept: int = 2000, | |
): | |
if not min_length >= min_interval >= hop_size: | |
raise ValueError( | |
"The following condition must be satisfied: min_length >= min_interval >= hop_size" | |
) | |
if not max_sil_kept >= hop_size: | |
raise ValueError( | |
"The following condition must be satisfied: max_sil_kept >= hop_size" | |
) | |
min_interval = sr * min_interval / 1000 | |
self.threshold = 10 ** (threshold / 20.0) | |
self.hop_size = round(sr * hop_size / 1000) | |
self.win_size = min(round(min_interval), 4 * self.hop_size) | |
self.min_length = round(sr * min_length / 1000 / self.hop_size) | |
self.min_interval = round(min_interval / self.hop_size) | |
self.max_sil_kept = round(sr * max_sil_kept / 1000 / self.hop_size) | |
def _apply_slice(self, waveform, begin, end): | |
if len(waveform.shape) > 1: | |
return waveform[ | |
:, begin * self.hop_size : min(waveform.shape[1], end * self.hop_size) | |
] | |
else: | |
return waveform[ | |
begin * self.hop_size : min(waveform.shape[0], end * self.hop_size) | |
] | |
# @timeit | |
def slice(self, waveform): | |
if len(waveform.shape) > 1: | |
samples = waveform.mean(axis=0) | |
else: | |
samples = waveform | |
if samples.shape[0] <= self.min_length: | |
return [waveform] | |
rms_list = get_rms( | |
y=samples, frame_length=self.win_size, hop_length=self.hop_size | |
).squeeze(0) | |
sil_tags = [] | |
silence_start = None | |
clip_start = 0 | |
for i, rms in enumerate(rms_list): | |
# Keep looping while frame is silent. | |
if rms < self.threshold: | |
# Record start of silent frames. | |
if silence_start is None: | |
silence_start = i | |
continue | |
# Keep looping while frame is not silent and silence start has not been recorded. | |
if silence_start is None: | |
continue | |
# Clear recorded silence start if interval is not enough or clip is too short | |
is_leading_silence = silence_start == 0 and i > self.max_sil_kept | |
need_slice_middle = ( | |
i - silence_start >= self.min_interval | |
and i - clip_start >= self.min_length | |
) | |
if not is_leading_silence and not need_slice_middle: | |
silence_start = None | |
continue | |
# Need slicing. Record the range of silent frames to be removed. | |
if i - silence_start <= self.max_sil_kept: | |
pos = rms_list[silence_start : i + 1].argmin() + silence_start | |
if silence_start == 0: | |
sil_tags.append((0, pos)) | |
else: | |
sil_tags.append((pos, pos)) | |
clip_start = pos | |
elif i - silence_start <= self.max_sil_kept * 2: | |
pos = rms_list[ | |
i - self.max_sil_kept : silence_start + self.max_sil_kept + 1 | |
].argmin() | |
pos += i - self.max_sil_kept | |
pos_l = ( | |
rms_list[ | |
silence_start : silence_start + self.max_sil_kept + 1 | |
].argmin() | |
+ silence_start | |
) | |
pos_r = ( | |
rms_list[i - self.max_sil_kept : i + 1].argmin() | |
+ i | |
- self.max_sil_kept | |
) | |
if silence_start == 0: | |
sil_tags.append((0, pos_r)) | |
clip_start = pos_r | |
else: | |
sil_tags.append((min(pos_l, pos), max(pos_r, pos))) | |
clip_start = max(pos_r, pos) | |
else: | |
pos_l = ( | |
rms_list[ | |
silence_start : silence_start + self.max_sil_kept + 1 | |
].argmin() | |
+ silence_start | |
) | |
pos_r = ( | |
rms_list[i - self.max_sil_kept : i + 1].argmin() | |
+ i | |
- self.max_sil_kept | |
) | |
if silence_start == 0: | |
sil_tags.append((0, pos_r)) | |
else: | |
sil_tags.append((pos_l, pos_r)) | |
clip_start = pos_r | |
silence_start = None | |
# Deal with trailing silence. | |
total_frames = rms_list.shape[0] | |
if ( | |
silence_start is not None | |
and total_frames - silence_start >= self.min_interval | |
): | |
silence_end = min(total_frames, silence_start + self.max_sil_kept) | |
pos = rms_list[silence_start : silence_end + 1].argmin() + silence_start | |
sil_tags.append((pos, total_frames + 1)) | |
# Apply and return slices. | |
####ι³ι’+θ΅·ε§ζΆι΄+η»ζ’ζΆι΄ | |
if len(sil_tags) == 0: | |
return [[waveform,0,int(total_frames*self.hop_size)]] | |
else: | |
chunks = [] | |
if sil_tags[0][0] > 0: | |
chunks.append([self._apply_slice(waveform, 0, sil_tags[0][0]),0,int(sil_tags[0][0]*self.hop_size)]) | |
for i in range(len(sil_tags) - 1): | |
chunks.append( | |
[self._apply_slice(waveform, sil_tags[i][1], sil_tags[i + 1][0]),int(sil_tags[i][1]*self.hop_size),int(sil_tags[i + 1][0]*self.hop_size)] | |
) | |
if sil_tags[-1][1] < total_frames: | |
chunks.append( | |
[self._apply_slice(waveform, sil_tags[-1][1], total_frames),int(sil_tags[-1][1]*self.hop_size),int(total_frames*self.hop_size)] | |
) | |
return chunks | |
#terminal | |
def terminate_process_tree(pid, including_parent=True): | |
try: | |
parent = psutil.Process(pid) | |
except psutil.NoSuchProcess: | |
# Process already terminated | |
return | |
children = parent.children(recursive=True) | |
for child in children: | |
try: | |
os.kill(child.pid, signal.SIGTERM) # or signal.SIGKILL | |
except OSError: | |
pass | |
if including_parent: | |
try: | |
os.kill(parent.pid, signal.SIGTERM) # or signal.SIGKILL | |
except OSError: | |
pass | |
def terminate_process(pid): | |
if system == "Windows": | |
cmd = f"taskkill /t /f /pid {pid}" | |
os.system(cmd) | |
else: | |
terminate_process_tree(pid) | |
def start_training(dataset_name="", | |
exp_name="F5TTS_Base", | |
learning_rate=1e-4, | |
batch_size_per_gpu=400, | |
batch_size_type="frame", | |
max_samples=64, | |
grad_accumulation_steps=1, | |
max_grad_norm=1.0, | |
epochs=11, | |
num_warmup_updates=200, | |
save_per_updates=400, | |
last_per_steps=800, | |
finetune=True, | |
): | |
global training_process | |
# Check if a training process is already running | |
if training_process is not None: | |
return "Train run already!",gr.update(interactive=False),gr.update(interactive=True) | |
yield "start train",gr.update(interactive=False),gr.update(interactive=False) | |
# Command to run the training script with the specified arguments | |
cmd = f"{python_executable} finetune-cli.py --exp_name {exp_name} " \ | |
f"--learning_rate {learning_rate} " \ | |
f"--batch_size_per_gpu {batch_size_per_gpu} " \ | |
f"--batch_size_type {batch_size_type} " \ | |
f"--max_samples {max_samples} " \ | |
f"--grad_accumulation_steps {grad_accumulation_steps} " \ | |
f"--max_grad_norm {max_grad_norm} " \ | |
f"--epochs {epochs} " \ | |
f"--num_warmup_updates {num_warmup_updates} " \ | |
f"--save_per_updates {save_per_updates} " \ | |
f"--last_per_steps {last_per_steps} " \ | |
f"--dataset_name {dataset_name}" | |
if finetune:cmd += f" --finetune {finetune}" | |
print(cmd) | |
try: | |
# Start the training process | |
training_process = subprocess.Popen(cmd, shell=True) | |
time.sleep(5) | |
yield "check terminal for wandb",gr.update(interactive=False),gr.update(interactive=True) | |
# Wait for the training process to finish | |
training_process.wait() | |
time.sleep(1) | |
if training_process is None: | |
text_info = 'train stop' | |
else: | |
text_info = "train complete !" | |
except Exception as e: # Catch all exceptions | |
# Ensure that we reset the training process variable in case of an error | |
text_info=f"An error occurred: {str(e)}" | |
training_process=None | |
yield text_info,gr.update(interactive=True),gr.update(interactive=False) | |
def stop_training(): | |
global training_process | |
if training_process is None:return f"Train not run !",gr.update(interactive=True),gr.update(interactive=False) | |
terminate_process_tree(training_process.pid) | |
training_process = None | |
return 'train stop',gr.update(interactive=True),gr.update(interactive=False) | |
def create_data_project(name): | |
name+="_pinyin" | |
os.makedirs(os.path.join(path_data,name),exist_ok=True) | |
os.makedirs(os.path.join(path_data,name,"dataset"),exist_ok=True) | |
def transcribe(file_audio,language="english"): | |
global pipe | |
if pipe is None: | |
pipe = pipeline("automatic-speech-recognition",model="openai/whisper-large-v3-turbo", torch_dtype=torch.float16,device=device) | |
text_transcribe = pipe( | |
file_audio, | |
chunk_length_s=30, | |
batch_size=128, | |
generate_kwargs={"task": "transcribe","language": language}, | |
return_timestamps=False, | |
)["text"].strip() | |
return text_transcribe | |
def transcribe_all(name_project,audio_files,language,user=False,progress=gr.Progress()): | |
name_project+="_pinyin" | |
path_project= os.path.join(path_data,name_project) | |
path_dataset = os.path.join(path_project,"dataset") | |
path_project_wavs = os.path.join(path_project,"wavs") | |
file_metadata = os.path.join(path_project,"metadata.csv") | |
if os.path.isdir(path_project_wavs): | |
shutil.rmtree(path_project_wavs) | |
if os.path.isfile(file_metadata): | |
os.remove(file_metadata) | |
os.makedirs(path_project_wavs,exist_ok=True) | |
if user: | |
file_audios = [file for format in ('*.wav', '*.ogg', '*.opus', '*.mp3', '*.flac') for file in glob(os.path.join(path_dataset, format))] | |
else: | |
file_audios = audio_files | |
print([file_audios]) | |
alpha = 0.5 | |
_max = 1.0 | |
slicer = Slicer(24000) | |
num = 0 | |
data="" | |
for file_audio in progress.tqdm(file_audios, desc="transcribe files",total=len((file_audios))): | |
audio, _ = librosa.load(file_audio, sr=24000, mono=True) | |
list_slicer=slicer.slice(audio) | |
for chunk, start, end in progress.tqdm(list_slicer,total=len(list_slicer), desc="slicer files"): | |
name_segment = os.path.join(f"segment_{num}") | |
file_segment = os.path.join(path_project_wavs, f"{name_segment}.wav") | |
tmp_max = np.abs(chunk).max() | |
if(tmp_max>1):chunk/=tmp_max | |
chunk = (chunk / tmp_max * (_max * alpha)) + (1 - alpha) * chunk | |
wavfile.write(file_segment,24000, (chunk * 32767).astype(np.int16)) | |
text=transcribe(file_segment,language) | |
text = text.lower().strip().replace('"',"") | |
data+= f"{name_segment}|{text}\n" | |
num+=1 | |
with open(file_metadata,"w",encoding="utf-8") as f: | |
f.write(data) | |
return f"transcribe complete samples : {num} in path {path_project_wavs}" | |
def format_seconds_to_hms(seconds): | |
hours = int(seconds / 3600) | |
minutes = int((seconds % 3600) / 60) | |
seconds = seconds % 60 | |
return "{:02d}:{:02d}:{:02d}".format(hours, minutes, int(seconds)) | |
def create_metadata(name_project,progress=gr.Progress()): | |
name_project+="_pinyin" | |
path_project= os.path.join(path_data,name_project) | |
path_project_wavs = os.path.join(path_project,"wavs") | |
file_metadata = os.path.join(path_project,"metadata.csv") | |
file_raw = os.path.join(path_project,"raw.arrow") | |
file_duration = os.path.join(path_project,"duration.json") | |
file_vocab = os.path.join(path_project,"vocab.txt") | |
with open(file_metadata,"r",encoding="utf-8") as f: | |
data=f.read() | |
audio_path_list=[] | |
text_list=[] | |
duration_list=[] | |
count=data.split("\n") | |
lenght=0 | |
result=[] | |
for line in progress.tqdm(data.split("\n"),total=count): | |
sp_line=line.split("|") | |
if len(sp_line)!=2:continue | |
name_audio,text = sp_line[:2] | |
file_audio = os.path.join(path_project_wavs, name_audio + ".wav") | |
duraction = get_audio_duration(file_audio) | |
if duraction<2 and duraction>15:continue | |
if len(text)<4:continue | |
text = clear_text(text) | |
text = convert_char_to_pinyin([text], polyphone = True)[0] | |
audio_path_list.append(file_audio) | |
duration_list.append(duraction) | |
text_list.append(text) | |
result.append({"audio_path": file_audio, "text": text, "duration": duraction}) | |
lenght+=duraction | |
min_second = round(min(duration_list),2) | |
max_second = round(max(duration_list),2) | |
with ArrowWriter(path=file_raw, writer_batch_size=1) as writer: | |
for line in progress.tqdm(result,total=len(result), desc=f"prepare data"): | |
writer.write(line) | |
with open(file_duration, 'w', encoding='utf-8') as f: | |
json.dump({"duration": duration_list}, f, ensure_ascii=False) | |
file_vocab_finetune = "data/Emilia_ZH_EN_pinyin/vocab.txt" | |
shutil.copy2(file_vocab_finetune, file_vocab) | |
return f"prepare complete \nsamples : {len(text_list)}\ntime data : {format_seconds_to_hms(lenght)}\nmin sec : {min_second}\nmax sec : {max_second}\nfile_arrow : {file_raw}\n" | |
def check_user(value): | |
return gr.update(visible=not value),gr.update(visible=value) | |
def calculate_train(name_project,batch_size_type,max_samples,learning_rate,num_warmup_updates,save_per_updates,last_per_steps,finetune): | |
name_project+="_pinyin" | |
path_project= os.path.join(path_data,name_project) | |
file_duraction = os.path.join(path_project,"duration.json") | |
with open(file_duraction, 'r') as file: | |
data = json.load(file) | |
duration_list = data['duration'] | |
samples = len(duration_list) | |
gpu_properties = torch.cuda.get_device_properties(0) | |
total_memory = gpu_properties.total_memory / (1024 ** 3) | |
if batch_size_type=="frame": | |
batch = int(total_memory * 0.5) | |
batch = (lambda num: num + 1 if num % 2 != 0 else num)(batch) | |
batch_size_per_gpu = int(36800 / batch ) | |
else: | |
batch_size_per_gpu = int(total_memory / 8) | |
batch_size_per_gpu = (lambda num: num + 1 if num % 2 != 0 else num)(batch_size_per_gpu) | |
batch = batch_size_per_gpu | |
if batch_size_per_gpu<=0:batch_size_per_gpu=1 | |
if samples<64: | |
max_samples = int(samples * 0.25) | |
num_warmup_updates = int(samples * 0.10) | |
save_per_updates = int(samples * 0.25) | |
last_per_steps =int(save_per_updates * 5) | |
max_samples = (lambda num: num + 1 if num % 2 != 0 else num)(max_samples) | |
num_warmup_updates = (lambda num: num + 1 if num % 2 != 0 else num)(num_warmup_updates) | |
save_per_updates = (lambda num: num + 1 if num % 2 != 0 else num)(save_per_updates) | |
last_per_steps = (lambda num: num + 1 if num % 2 != 0 else num)(last_per_steps) | |
if finetune:learning_rate=1e-4 | |
else:learning_rate=7.5e-5 | |
return batch_size_per_gpu,max_samples,num_warmup_updates,save_per_updates,last_per_steps,samples,learning_rate | |
def extract_and_save_ema_model(checkpoint_path: str, new_checkpoint_path: str) -> None: | |
try: | |
checkpoint = torch.load(checkpoint_path) | |
print("Original Checkpoint Keys:", checkpoint.keys()) | |
ema_model_state_dict = checkpoint.get('ema_model_state_dict', None) | |
if ema_model_state_dict is not None: | |
new_checkpoint = {'ema_model_state_dict': ema_model_state_dict} | |
torch.save(new_checkpoint, new_checkpoint_path) | |
print(f"New checkpoint saved at: {new_checkpoint_path}") | |
else: | |
print("No 'ema_model_state_dict' found in the checkpoint.") | |
except Exception as e: | |
print(f"An error occurred: {e}") | |
def vocab_check(project_name): | |
name_project = project_name + "_pinyin" | |
path_project = os.path.join(path_data, name_project) | |
file_metadata = os.path.join(path_project, "metadata.csv") | |
file_vocab="data/Emilia_ZH_EN_pinyin/vocab.txt" | |
with open(file_vocab,"r",encoding="utf-8") as f: | |
data=f.read() | |
vocab = data.split("\n") | |
with open(file_metadata,"r",encoding="utf-8") as f: | |
data=f.read() | |
miss_symbols=[] | |
miss_symbols_keep={} | |
for item in data.split("\n"): | |
sp=item.split("|") | |
if len(sp)!=2:continue | |
text=sp[1].lower().strip() | |
for t in text: | |
if (t in vocab)==False and (t in miss_symbols_keep)==False: | |
miss_symbols.append(t) | |
miss_symbols_keep[t]=t | |
if miss_symbols==[]:info ="You can train using your language !" | |
else:info = f"The following symbols are missing in your language : {len(miss_symbols)}\n\n" + "\n".join(miss_symbols) | |
return info | |
with gr.Blocks() as app: | |
with gr.Row(): | |
project_name=gr.Textbox(label="project name",value="my_speak") | |
bt_create=gr.Button("create new project") | |
bt_create.click(fn=create_data_project,inputs=[project_name]) | |
with gr.Tabs(): | |
with gr.TabItem("transcribe Data"): | |
ch_manual = gr.Checkbox(label="user",value=False) | |
mark_info_transcribe=gr.Markdown( | |
"""```plaintext | |
Place your 'wavs' folder and 'metadata.csv' file in the {your_project_name}' directory. | |
my_speak/ | |
β | |
βββ dataset/ | |
βββ audio1.wav | |
βββ audio2.wav | |
... | |
```""",visible=False) | |
audio_speaker = gr.File(label="voice",type="filepath",file_count="multiple") | |
txt_lang = gr.Text(label="Language",value="english") | |
bt_transcribe=bt_create=gr.Button("transcribe") | |
txt_info_transcribe=gr.Text(label="info",value="") | |
bt_transcribe.click(fn=transcribe_all,inputs=[project_name,audio_speaker,txt_lang,ch_manual],outputs=[txt_info_transcribe]) | |
ch_manual.change(fn=check_user,inputs=[ch_manual],outputs=[audio_speaker,mark_info_transcribe]) | |
with gr.TabItem("prepare Data"): | |
gr.Markdown( | |
"""```plaintext | |
place all your wavs folder and your metadata.csv file in {your name project} | |
my_speak/ | |
β | |
βββ wavs/ | |
β βββ audio1.wav | |
β βββ audio2.wav | |
| ... | |
β | |
βββ metadata.csv | |
file format metadata.csv | |
audio1|text1 | |
audio2|text1 | |
... | |
```""") | |
bt_prepare=bt_create=gr.Button("prepare") | |
txt_info_prepare=gr.Text(label="info",value="") | |
bt_prepare.click(fn=create_metadata,inputs=[project_name],outputs=[txt_info_prepare]) | |
with gr.TabItem("train Data"): | |
with gr.Row(): | |
bt_calculate=bt_create=gr.Button("Auto Settings") | |
ch_finetune=bt_create=gr.Checkbox(label="finetune",value=True) | |
lb_samples = gr.Label(label="samples") | |
batch_size_type = gr.Radio(label="Batch Size Type", choices=["frame", "sample"], value="frame") | |
with gr.Row(): | |
exp_name = gr.Radio(label="Model", choices=["F5TTS_Base", "E2TTS_Base"], value="F5TTS_Base") | |
learning_rate = gr.Number(label="Learning Rate", value=1e-4, step=1e-4) | |
with gr.Row(): | |
batch_size_per_gpu = gr.Number(label="Batch Size per GPU", value=1000) | |
max_samples = gr.Number(label="Max Samples", value=16) | |
with gr.Row(): | |
grad_accumulation_steps = gr.Number(label="Gradient Accumulation Steps", value=1) | |
max_grad_norm = gr.Number(label="Max Gradient Norm", value=1.0) | |
with gr.Row(): | |
epochs = gr.Number(label="Epochs", value=10) | |
num_warmup_updates = gr.Number(label="Warmup Updates", value=5) | |
with gr.Row(): | |
save_per_updates = gr.Number(label="Save per Updates", value=10) | |
last_per_steps = gr.Number(label="Last per Steps", value=50) | |
with gr.Row(): | |
start_button = gr.Button("Start Training") | |
stop_button = gr.Button("Stop Training",interactive=False) | |
txt_info_train=gr.Text(label="info",value="") | |
start_button.click(fn=start_training,inputs=[project_name,exp_name,learning_rate,batch_size_per_gpu,batch_size_type,max_samples,grad_accumulation_steps,max_grad_norm,epochs,num_warmup_updates,save_per_updates,last_per_steps,ch_finetune],outputs=[txt_info_train,start_button,stop_button]) | |
stop_button.click(fn=stop_training,outputs=[txt_info_train,start_button,stop_button]) | |
bt_calculate.click(fn=calculate_train,inputs=[project_name,batch_size_type,max_samples,learning_rate,num_warmup_updates,save_per_updates,last_per_steps,ch_finetune],outputs=[batch_size_per_gpu,max_samples,num_warmup_updates,save_per_updates,last_per_steps,lb_samples,learning_rate]) | |
with gr.TabItem("reduse checkpoint"): | |
txt_path_checkpoint = gr.Text(label="path checkpoint :") | |
txt_path_checkpoint_small = gr.Text(label="path output :") | |
reduse_button = gr.Button("reduse") | |
reduse_button.click(fn=extract_and_save_ema_model,inputs=[txt_path_checkpoint,txt_path_checkpoint_small]) | |
with gr.TabItem("vocab check experiment"): | |
check_button = gr.Button("check vocab") | |
txt_info_check=gr.Text(label="info",value="") | |
check_button.click(fn=vocab_check,inputs=[project_name],outputs=[txt_info_check]) | |
def main(port, host, share, api): | |
global app | |
print(f"Starting app...") | |
app.queue(api_open=api).launch( | |
server_name=host, server_port=port, share=share, show_api=api | |
) | |
if __name__ == "__main__": | |
name="my_speak" | |
#create_data_project(name) | |
#transcribe_all(name) | |
#create_metadata(name) | |
main() | |