Whisper-WebUI / modules /whisper_Inference.py
jhj0517
add `Files` to output
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raw
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18.8 kB
import whisper
import gradio as gr
import time
import os
from typing import BinaryIO, Union, Tuple
import numpy as np
from datetime import datetime
import torch
from .base_interface import BaseInterface
from modules.subtitle_manager import get_srt, get_vtt, get_txt, write_file, safe_filename
from modules.youtube_manager import get_ytdata, get_ytaudio
DEFAULT_MODEL_SIZE = "large-v3"
class WhisperInference(BaseInterface):
def __init__(self):
super().__init__()
self.current_model_size = None
self.model = None
self.available_models = whisper.available_models()
self.available_langs = sorted(list(whisper.tokenizer.LANGUAGES.values()))
self.device = "cuda" if torch.cuda.is_available() else "cpu"
self.available_compute_types = ["float16", "float32"]
self.current_compute_type = "float16" if self.device == "cuda" else "float32"
self.default_beam_size = 1
def transcribe_file(self,
fileobjs: list,
model_size: str,
lang: str,
file_format: str,
istranslate: bool,
add_timestamp: bool,
beam_size: int,
log_prob_threshold: float,
no_speech_threshold: float,
compute_type: str,
progress=gr.Progress()) -> list:
"""
Write subtitle file from Files
Parameters
----------
fileobjs: list
List of files to transcribe from gr.Files()
model_size: str
Whisper model size from gr.Dropdown()
lang: str
Source language of the file to transcribe from gr.Dropdown()
file_format: str
File format to write from gr.Dropdown(). Supported format: [SRT, WebVTT, txt]
istranslate: bool
Boolean value from gr.Checkbox() that determines whether to translate to English.
It's Whisper's feature to translate speech from another language directly into English end-to-end.
add_timestamp: bool
Boolean value from gr.Checkbox() that determines whether to add a timestamp at the end of the filename.
beam_size: int
Int value from gr.Number() that is used for decoding option.
log_prob_threshold: float
float value from gr.Number(). If the average log probability over sampled tokens is
below this value, treat as failed.
no_speech_threshold: float
float value from gr.Number(). If the no_speech probability is higher than this value AND
the average log probability over sampled tokens is below `log_prob_threshold`,
consider the segment as silent.
compute_type: str
compute type from gr.Dropdown().
progress: gr.Progress
Indicator to show progress directly in gradio.
I use a forked version of whisper for this. To see more info : https://github.com/jhj0517/jhj0517-whisper/tree/add-progress-callback
Returns
----------
A List of
String to return to gr.Textbox()
Files to return to gr.Files()
"""
try:
self.update_model_if_needed(model_size=model_size, compute_type=compute_type, progress=progress)
files_info = {}
for fileobj in fileobjs:
progress(0, desc="Loading Audio..")
audio = whisper.load_audio(fileobj.name)
result, elapsed_time = self.transcribe(audio=audio,
lang=lang,
istranslate=istranslate,
beam_size=beam_size,
log_prob_threshold=log_prob_threshold,
no_speech_threshold=no_speech_threshold,
compute_type=compute_type,
progress=progress
)
progress(1, desc="Completed!")
file_name, file_ext = os.path.splitext(os.path.basename(fileobj.name))
file_name = safe_filename(file_name)
subtitle, file_path = self.generate_and_write_file(
file_name=file_name,
transcribed_segments=result,
add_timestamp=add_timestamp,
file_format=file_format
)
files_info[file_name] = {"subtitle": subtitle, "elapsed_time": elapsed_time, "path": file_path}
total_result = ''
total_time = 0
for file_name, info in files_info.items():
total_result += '------------------------------------\n'
total_result += f'{file_name}\n\n'
total_result += f"{info['subtitle']}"
total_time += info["elapsed_time"]
gr_str = f"Done in {self.format_time(total_time)}! Subtitle is in the outputs folder.\n\n{total_result}"
gr_file_path = [info['path'] for info in files_info.values()]
return [gr_str, gr_file_path]
except Exception as e:
print(f"Error transcribing file: {str(e)}")
finally:
self.release_cuda_memory()
self.remove_input_files([fileobj.name for fileobj in fileobjs])
def transcribe_youtube(self,
youtubelink: str,
model_size: str,
lang: str,
file_format: str,
istranslate: bool,
add_timestamp: bool,
beam_size: int,
log_prob_threshold: float,
no_speech_threshold: float,
compute_type: str,
progress=gr.Progress()) -> list:
"""
Write subtitle file from Youtube
Parameters
----------
youtubelink: str
Link of Youtube to transcribe from gr.Textbox()
model_size: str
Whisper model size from gr.Dropdown()
lang: str
Source language of the file to transcribe from gr.Dropdown()
file_format: str
File format to write from gr.Dropdown(). Supported format: [SRT, WebVTT, txt]
istranslate: bool
Boolean value from gr.Checkbox() that determines whether to translate to English.
It's Whisper's feature to translate speech from another language directly into English end-to-end.
add_timestamp: bool
Boolean value from gr.Checkbox() that determines whether to add a timestamp at the end of the filename.
beam_size: int
Int value from gr.Number() that is used for decoding option.
log_prob_threshold: float
float value from gr.Number(). If the average log probability over sampled tokens is
below this value, treat as failed.
no_speech_threshold: float
float value from gr.Number(). If the no_speech probability is higher than this value AND
the average log probability over sampled tokens is below `log_prob_threshold`,
consider the segment as silent.
compute_type: str
compute type from gr.Dropdown().
progress: gr.Progress
Indicator to show progress directly in gradio.
I use a forked version of whisper for this. To see more info : https://github.com/jhj0517/jhj0517-whisper/tree/add-progress-callback
Returns
----------
A List of
String to return to gr.Textbox()
Files to return to gr.Files()
"""
try:
self.update_model_if_needed(model_size=model_size, compute_type=compute_type, progress=progress)
progress(0, desc="Loading Audio from Youtube..")
yt = get_ytdata(youtubelink)
audio = whisper.load_audio(get_ytaudio(yt))
result, elapsed_time = self.transcribe(audio=audio,
lang=lang,
istranslate=istranslate,
beam_size=beam_size,
log_prob_threshold=log_prob_threshold,
no_speech_threshold=no_speech_threshold,
compute_type=compute_type,
progress=progress)
progress(1, desc="Completed!")
file_name = safe_filename(yt.title)
subtitle, file_path = self.generate_and_write_file(
file_name=file_name,
transcribed_segments=result,
add_timestamp=add_timestamp,
file_format=file_format
)
gr_str = f"Done in {self.format_time(elapsed_time)}! Subtitle file is in the outputs folder.\n\n{subtitle}"
return [gr_str, file_path]
except Exception as e:
print(f"Error transcribing youtube video: {str(e)}")
finally:
try:
if 'yt' not in locals():
yt = get_ytdata(youtubelink)
file_path = get_ytaudio(yt)
else:
file_path = get_ytaudio(yt)
self.release_cuda_memory()
self.remove_input_files([file_path])
except Exception as cleanup_error:
pass
def transcribe_mic(self,
micaudio: str,
model_size: str,
lang: str,
file_format: str,
istranslate: bool,
beam_size: int,
log_prob_threshold: float,
no_speech_threshold: float,
compute_type: str,
progress=gr.Progress()) -> list:
"""
Write subtitle file from microphone
Parameters
----------
micaudio: str
Audio file path from gr.Microphone()
model_size: str
Whisper model size from gr.Dropdown()
lang: str
Source language of the file to transcribe from gr.Dropdown()
file_format: str
Subtitle format to write from gr.Dropdown(). Supported format: [SRT, WebVTT, txt]
istranslate: bool
Boolean value from gr.Checkbox() that determines whether to translate to English.
It's Whisper's feature to translate speech from another language directly into English end-to-end.
beam_size: int
Int value from gr.Number() that is used for decoding option.
log_prob_threshold: float
float value from gr.Number(). If the average log probability over sampled tokens is
below this value, treat as failed.
no_speech_threshold: float
float value from gr.Number(). If the no_speech probability is higher than this value AND
the average log probability over sampled tokens is below `log_prob_threshold`,
consider the segment as silent.
compute_type: str
compute type from gr.Dropdown().
progress: gr.Progress
Indicator to show progress directly in gradio.
I use a forked version of whisper for this. To see more info : https://github.com/jhj0517/jhj0517-whisper/tree/add-progress-callback
Returns
----------
A List of
String to return to gr.Textbox()
Files to return to gr.Files()
"""
try:
self.update_model_if_needed(model_size=model_size, compute_type=compute_type, progress=progress)
result, elapsed_time = self.transcribe(audio=micaudio,
lang=lang,
istranslate=istranslate,
beam_size=beam_size,
log_prob_threshold=log_prob_threshold,
no_speech_threshold=no_speech_threshold,
compute_type=compute_type,
progress=progress)
progress(1, desc="Completed!")
subtitle, file_path = self.generate_and_write_file(
file_name="Mic",
transcribed_segments=result,
add_timestamp=True,
file_format=file_format
)
gr_str = f"Done in {self.format_time(elapsed_time)}! Subtitle file is in the outputs folder.\n\n{subtitle}"
return [gr_str, file_path]
except Exception as e:
print(f"Error transcribing mic: {str(e)}")
finally:
self.release_cuda_memory()
self.remove_input_files([micaudio])
def transcribe(self,
audio: Union[str, np.ndarray, torch.Tensor],
lang: str,
istranslate: bool,
beam_size: int,
log_prob_threshold: float,
no_speech_threshold: float,
compute_type: str,
progress: gr.Progress
) -> Tuple[list[dict], float]:
"""
transcribe method for OpenAI's Whisper implementation.
Parameters
----------
audio: Union[str, BinaryIO, torch.Tensor]
Audio path or file binary or Audio numpy array
lang: str
Source language of the file to transcribe from gr.Dropdown()
istranslate: bool
Boolean value from gr.Checkbox() that determines whether to translate to English.
It's Whisper's feature to translate speech from another language directly into English end-to-end.
beam_size: int
Int value from gr.Number() that is used for decoding option.
log_prob_threshold: float
float value from gr.Number(). If the average log probability over sampled tokens is
below this value, treat as failed.
no_speech_threshold: float
float value from gr.Number(). If the no_speech probability is higher than this value AND
the average log probability over sampled tokens is below `log_prob_threshold`,
consider the segment as silent.
compute_type: str
compute type from gr.Dropdown().
progress: gr.Progress
Indicator to show progress directly in gradio.
Returns
----------
segments_result: list[dict]
list of dicts that includes start, end timestamps and transcribed text
elapsed_time: float
elapsed time for transcription
"""
start_time = time.time()
def progress_callback(progress_value):
progress(progress_value, desc="Transcribing..")
if lang == "Automatic Detection":
lang = None
translatable_model = ["large", "large-v1", "large-v2", "large-v3"]
segments_result = self.model.transcribe(audio=audio,
language=lang,
verbose=False,
beam_size=beam_size,
logprob_threshold=log_prob_threshold,
no_speech_threshold=no_speech_threshold,
task="translate" if istranslate and self.current_model_size in translatable_model else "transcribe",
fp16=True if compute_type == "float16" else False,
progress_callback=progress_callback)["segments"]
elapsed_time = time.time() - start_time
return segments_result, elapsed_time
def update_model_if_needed(self,
model_size: str,
compute_type: str,
progress: gr.Progress,
):
"""
Initialize model if it doesn't match with current model setting
"""
if compute_type != self.current_compute_type:
self.current_compute_type = compute_type
if model_size != self.current_model_size or self.model is None:
progress(0, desc="Initializing Model..")
self.current_model_size = model_size
self.model = whisper.load_model(
name=model_size,
device=self.device,
download_root=os.path.join("models", "Whisper")
)
@staticmethod
def generate_and_write_file(file_name: str,
transcribed_segments: list,
add_timestamp: bool,
file_format: str,
) -> str:
"""
This method writes subtitle file and returns str to gr.Textbox
"""
timestamp = datetime.now().strftime("%m%d%H%M%S")
if add_timestamp:
output_path = os.path.join("outputs", f"{file_name}-{timestamp}")
else:
output_path = os.path.join("outputs", f"{file_name}")
if file_format == "SRT":
content = get_srt(transcribed_segments)
output_path += '.srt'
write_file(content, output_path)
elif file_format == "WebVTT":
content = get_vtt(transcribed_segments)
output_path += '.vtt'
write_file(content, output_path)
elif file_format == "txt":
content = get_txt(transcribed_segments)
output_path += '.txt'
write_file(content, output_path)
return content, output_path
@staticmethod
def format_time(elapsed_time: float) -> str:
hours, rem = divmod(elapsed_time, 3600)
minutes, seconds = divmod(rem, 60)
time_str = ""
if hours:
time_str += f"{hours} hours "
if minutes:
time_str += f"{minutes} minutes "
seconds = round(seconds)
time_str += f"{seconds} seconds"
return time_str.strip()