Whisper-WebUI / modules /whisper /whisper_base.py
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import os
import torch
import whisper
import ctranslate2
import gradio as gr
import torchaudio
from abc import ABC, abstractmethod
from typing import BinaryIO, Union, Tuple, List
import numpy as np
from datetime import datetime
from faster_whisper.vad import VadOptions
from dataclasses import astuple
from modules.uvr.music_separator import MusicSeparator
from modules.utils.paths import (WHISPER_MODELS_DIR, DIARIZATION_MODELS_DIR, OUTPUT_DIR, DEFAULT_PARAMETERS_CONFIG_PATH,
UVR_MODELS_DIR)
from modules.utils.constants import AUTOMATIC_DETECTION
from modules.utils.subtitle_manager import get_srt, get_vtt, get_txt, write_file, safe_filename
from modules.utils.youtube_manager import get_ytdata, get_ytaudio
from modules.utils.files_manager import get_media_files, format_gradio_files, load_yaml, save_yaml
from modules.whisper.whisper_parameter import *
from modules.diarize.diarizer import Diarizer
from modules.vad.silero_vad import SileroVAD
class WhisperBase(ABC):
def __init__(self,
model_dir: str = WHISPER_MODELS_DIR,
diarization_model_dir: str = DIARIZATION_MODELS_DIR,
uvr_model_dir: str = UVR_MODELS_DIR,
output_dir: str = OUTPUT_DIR,
):
self.model_dir = model_dir
self.output_dir = output_dir
os.makedirs(self.output_dir, exist_ok=True)
os.makedirs(self.model_dir, exist_ok=True)
self.diarizer = Diarizer(
model_dir=diarization_model_dir
)
self.vad = SileroVAD()
self.music_separator = MusicSeparator(
model_dir=uvr_model_dir,
output_dir=os.path.join(output_dir, "UVR")
)
self.model = None
self.current_model_size = None
self.available_models = whisper.available_models()
self.available_langs = sorted(list(whisper.tokenizer.LANGUAGES.values()))
self.translatable_models = ["large", "large-v1", "large-v2", "large-v3"]
self.device = self.get_device()
self.available_compute_types = self.get_available_compute_type()
self.current_compute_type = self.get_compute_type()
@abstractmethod
def transcribe(self,
audio: Union[str, BinaryIO, np.ndarray],
progress: gr.Progress = gr.Progress(),
*whisper_params,
):
"""Inference whisper model to transcribe"""
pass
@abstractmethod
def update_model(self,
model_size: str,
compute_type: str,
progress: gr.Progress = gr.Progress()
):
"""Initialize whisper model"""
pass
def run(self,
audio: Union[str, BinaryIO, np.ndarray],
progress: gr.Progress = gr.Progress(),
add_timestamp: bool = True,
*whisper_params,
) -> Tuple[List[dict], float]:
"""
Run transcription with conditional pre-processing and post-processing.
The VAD will be performed to remove noise from the audio input in pre-processing, if enabled.
The diarization will be performed in post-processing, if enabled.
Parameters
----------
audio: Union[str, BinaryIO, np.ndarray]
Audio input. This can be file path or binary type.
progress: gr.Progress
Indicator to show progress directly in gradio.
add_timestamp: bool
Whether to add a timestamp at the end of the filename.
*whisper_params: tuple
Parameters related with whisper. This will be dealt with "WhisperParameters" data class
Returns
----------
segments_result: List[dict]
list of dicts that includes start, end timestamps and transcribed text
elapsed_time: float
elapsed time for running
"""
params = WhisperParameters.as_value(*whisper_params)
self.cache_parameters(
whisper_params=params,
add_timestamp=add_timestamp
)
if params.lang is None:
pass
elif params.lang == AUTOMATIC_DETECTION:
params.lang = None
else:
language_code_dict = {value: key for key, value in whisper.tokenizer.LANGUAGES.items()}
params.lang = language_code_dict[params.lang]
if params.is_bgm_separate:
music, audio, _ = self.music_separator.separate(
audio=audio,
model_name=params.uvr_model_size,
device=params.uvr_device,
segment_size=params.uvr_segment_size,
save_file=params.uvr_save_file,
progress=progress
)
if audio.ndim >= 2:
audio = audio.mean(axis=1)
if self.music_separator.audio_info is None:
origin_sample_rate = 16000
else:
origin_sample_rate = self.music_separator.audio_info.sample_rate
audio = self.resample_audio(audio=audio, original_sample_rate=origin_sample_rate)
if params.uvr_enable_offload:
self.music_separator.offload()
if params.vad_filter:
# Explicit value set for float('inf') from gr.Number()
if params.max_speech_duration_s is None or params.max_speech_duration_s >= 9999:
params.max_speech_duration_s = float('inf')
vad_options = VadOptions(
threshold=params.threshold,
min_speech_duration_ms=params.min_speech_duration_ms,
max_speech_duration_s=params.max_speech_duration_s,
min_silence_duration_ms=params.min_silence_duration_ms,
speech_pad_ms=params.speech_pad_ms
)
audio, speech_chunks = self.vad.run(
audio=audio,
vad_parameters=vad_options,
progress=progress
)
result, elapsed_time = self.transcribe(
audio,
progress,
*astuple(params)
)
if params.vad_filter:
result = self.vad.restore_speech_timestamps(
segments=result,
speech_chunks=speech_chunks,
)
if params.is_diarize:
result, elapsed_time_diarization = self.diarizer.run(
audio=audio,
use_auth_token=params.hf_token,
transcribed_result=result,
)
elapsed_time += elapsed_time_diarization
return result, elapsed_time
def transcribe_file(self,
files: Optional[List] = None,
input_folder_path: Optional[str] = None,
file_format: str = "SRT",
add_timestamp: bool = True,
progress=gr.Progress(),
*whisper_params,
) -> list:
"""
Write subtitle file from Files
Parameters
----------
files: list
List of files to transcribe from gr.Files()
input_folder_path: str
Input folder path to transcribe from gr.Textbox(). If this is provided, `files` will be ignored and
this will be used instead.
file_format: str
Subtitle File format to write from gr.Dropdown(). Supported format: [SRT, WebVTT, txt]
add_timestamp: bool
Boolean value from gr.Checkbox() that determines whether to add a timestamp at the end of the subtitle filename.
progress: gr.Progress
Indicator to show progress directly in gradio.
*whisper_params: tuple
Parameters related with whisper. This will be dealt with "WhisperParameters" data class
Returns
----------
result_str:
Result of transcription to return to gr.Textbox()
result_file_path:
Output file path to return to gr.Files()
"""
try:
if input_folder_path:
files = get_media_files(input_folder_path)
if isinstance(files, str):
files = [files]
if files and isinstance(files[0], gr.utils.NamedString):
files = [file.name for file in files]
files_info = {}
for file in files:
transcribed_segments, time_for_task = self.run(
file,
progress,
add_timestamp,
*whisper_params,
)
file_name, file_ext = os.path.splitext(os.path.basename(file))
subtitle, file_path = self.generate_and_write_file(
file_name=file_name,
transcribed_segments=transcribed_segments,
add_timestamp=add_timestamp,
file_format=file_format,
output_dir=self.output_dir
)
files_info[file_name] = {"subtitle": subtitle, "time_for_task": time_for_task, "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["time_for_task"]
result_str = f"Done in {self.format_time(total_time)}! Subtitle is in the outputs folder.\n\n{total_result}"
result_file_path = [info['path'] for info in files_info.values()]
return [result_str, result_file_path]
except Exception as e:
print(f"Error transcribing file: {e}")
finally:
self.release_cuda_memory()
def transcribe_mic(self,
mic_audio: str,
file_format: str = "SRT",
add_timestamp: bool = True,
progress=gr.Progress(),
*whisper_params,
) -> list:
"""
Write subtitle file from microphone
Parameters
----------
mic_audio: str
Audio file path from gr.Microphone()
file_format: str
Subtitle File format to write from gr.Dropdown(). Supported format: [SRT, WebVTT, txt]
add_timestamp: bool
Boolean value from gr.Checkbox() that determines whether to add a timestamp at the end of the filename.
progress: gr.Progress
Indicator to show progress directly in gradio.
*whisper_params: tuple
Parameters related with whisper. This will be dealt with "WhisperParameters" data class
Returns
----------
result_str:
Result of transcription to return to gr.Textbox()
result_file_path:
Output file path to return to gr.Files()
"""
try:
progress(0, desc="Loading Audio..")
transcribed_segments, time_for_task = self.run(
mic_audio,
progress,
add_timestamp,
*whisper_params,
)
progress(1, desc="Completed!")
subtitle, result_file_path = self.generate_and_write_file(
file_name="Mic",
transcribed_segments=transcribed_segments,
add_timestamp=add_timestamp,
file_format=file_format,
output_dir=self.output_dir
)
result_str = f"Done in {self.format_time(time_for_task)}! Subtitle file is in the outputs folder.\n\n{subtitle}"
return [result_str, result_file_path]
except Exception as e:
print(f"Error transcribing file: {e}")
finally:
self.release_cuda_memory()
def transcribe_youtube(self,
youtube_link: str,
file_format: str = "SRT",
add_timestamp: bool = True,
progress=gr.Progress(),
*whisper_params,
) -> list:
"""
Write subtitle file from Youtube
Parameters
----------
youtube_link: str
URL of the Youtube video to transcribe from gr.Textbox()
file_format: str
Subtitle File format to write from gr.Dropdown(). Supported format: [SRT, WebVTT, txt]
add_timestamp: bool
Boolean value from gr.Checkbox() that determines whether to add a timestamp at the end of the filename.
progress: gr.Progress
Indicator to show progress directly in gradio.
*whisper_params: tuple
Parameters related with whisper. This will be dealt with "WhisperParameters" data class
Returns
----------
result_str:
Result of transcription to return to gr.Textbox()
result_file_path:
Output file path to return to gr.Files()
"""
try:
progress(0, desc="Loading Audio from Youtube..")
yt = get_ytdata(youtube_link)
audio = get_ytaudio(yt)
transcribed_segments, time_for_task = self.run(
audio,
progress,
add_timestamp,
*whisper_params,
)
progress(1, desc="Completed!")
file_name = safe_filename(yt.title)
subtitle, result_file_path = self.generate_and_write_file(
file_name=file_name,
transcribed_segments=transcribed_segments,
add_timestamp=add_timestamp,
file_format=file_format,
output_dir=self.output_dir
)
result_str = f"Done in {self.format_time(time_for_task)}! Subtitle file is in the outputs folder.\n\n{subtitle}"
if os.path.exists(audio):
os.remove(audio)
return [result_str, result_file_path]
except Exception as e:
print(f"Error transcribing file: {e}")
finally:
self.release_cuda_memory()
def get_compute_type(self):
if "float16" in self.available_compute_types:
return "float16"
if "float32" in self.available_compute_types:
return "float32"
else:
return self.available_compute_types[0]
def get_available_compute_type(self):
if self.device == "cuda":
return list(ctranslate2.get_supported_compute_types("cuda"))
else:
return list(ctranslate2.get_supported_compute_types("cpu"))
@staticmethod
def generate_and_write_file(file_name: str,
transcribed_segments: list,
add_timestamp: bool,
file_format: str,
output_dir: str
) -> str:
"""
Writes subtitle file
Parameters
----------
file_name: str
Output file name
transcribed_segments: list
Text segments transcribed from audio
add_timestamp: bool
Determines whether to add a timestamp to the end of the filename.
file_format: str
File format to write. Supported formats: [SRT, WebVTT, txt]
output_dir: str
Directory path of the output
Returns
----------
content: str
Result of the transcription
output_path: str
output file path
"""
if add_timestamp:
timestamp = datetime.now().strftime("%m%d%H%M%S")
output_path = os.path.join(output_dir, f"{file_name}-{timestamp}")
else:
output_path = os.path.join(output_dir, f"{file_name}")
file_format = file_format.strip().lower()
if file_format == "srt":
content = get_srt(transcribed_segments)
output_path += '.srt'
elif file_format == "webvtt":
content = get_vtt(transcribed_segments)
output_path += '.vtt'
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:
"""
Get {hours} {minutes} {seconds} time format string
Parameters
----------
elapsed_time: str
Elapsed time for transcription
Returns
----------
Time format string
"""
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()
@staticmethod
def get_device():
if torch.cuda.is_available():
return "cuda"
elif torch.backends.mps.is_available():
if not WhisperBase.is_sparse_api_supported():
# Device `SparseMPS` is not supported for now. See : https://github.com/pytorch/pytorch/issues/87886
return "cpu"
return "mps"
else:
return "cpu"
@staticmethod
def is_sparse_api_supported():
if not torch.backends.mps.is_available():
return False
try:
device = torch.device("mps")
sparse_tensor = torch.sparse_coo_tensor(
indices=torch.tensor([[0, 1], [2, 3]]),
values=torch.tensor([1, 2]),
size=(4, 4),
device=device
)
return True
except RuntimeError:
return False
@staticmethod
def release_cuda_memory():
"""Release memory"""
if torch.cuda.is_available():
torch.cuda.empty_cache()
torch.cuda.reset_max_memory_allocated()
@staticmethod
def remove_input_files(file_paths: List[str]):
"""Remove gradio cached files"""
if not file_paths:
return
for file_path in file_paths:
if file_path and os.path.exists(file_path):
os.remove(file_path)
@staticmethod
def cache_parameters(
whisper_params: WhisperValues,
add_timestamp: bool
):
"""cache parameters to the yaml file"""
cached_params = load_yaml(DEFAULT_PARAMETERS_CONFIG_PATH)
cached_whisper_param = whisper_params.to_yaml()
cached_yaml = {**cached_params, **cached_whisper_param}
cached_yaml["whisper"]["add_timestamp"] = add_timestamp
save_yaml(cached_yaml, DEFAULT_PARAMETERS_CONFIG_PATH)
@staticmethod
def resample_audio(audio: Union[str, np.ndarray],
new_sample_rate: int = 16000,
original_sample_rate: Optional[int] = None,) -> np.ndarray:
"""Resamples audio to 16k sample rate, standard on Whisper model"""
if isinstance(audio, str):
audio, original_sample_rate = torchaudio.load(audio)
else:
if original_sample_rate is None:
raise ValueError("original_sample_rate must be provided when audio is numpy array.")
audio = torch.from_numpy(audio)
resampler = torchaudio.transforms.Resample(orig_freq=original_sample_rate, new_freq=new_sample_rate)
resampled_audio = resampler(audio).numpy()
return resampled_audio