Whisper-WebUI / modules /whisper /base_transcription_pipeline.py
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import os
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 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 *
from modules.utils.subtitle_manager import *
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, read_file
from modules.whisper.data_classes import *
from modules.diarize.diarizer import Diarizer
from modules.vad.silero_vad import SileroVAD
class BaseTranscriptionPipeline(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,
*pipeline_params,
) -> Tuple[List[Segment], 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.
Due to the integration with gradio, the parameters have to be specified with a `*` wildcard.
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.
*pipeline_params: tuple
Parameters for the transcription pipeline. This will be dealt with "TranscriptionPipelineParams" data class.
This must be provided as a List with * wildcard because of the integration with gradio.
See more info at : https://github.com/gradio-app/gradio/issues/2471
Returns
----------
segments_result: List[Segment]
list of Segment that includes start, end timestamps and transcribed text
elapsed_time: float
elapsed time for running
"""
params = TranscriptionPipelineParams.from_list(list(pipeline_params))
params = self.validate_gradio_values(params)
bgm_params, vad_params, whisper_params, diarization_params = params.bgm_separation, params.vad, params.whisper, params.diarization
if bgm_params.is_separate_bgm:
music, audio, _ = self.music_separator.separate(
audio=audio,
model_name=bgm_params.model_size,
device=bgm_params.device,
segment_size=bgm_params.segment_size,
save_file=bgm_params.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 bgm_params.enable_offload:
self.music_separator.offload()
if vad_params.vad_filter:
vad_options = VadOptions(
threshold=vad_params.threshold,
min_speech_duration_ms=vad_params.min_speech_duration_ms,
max_speech_duration_s=vad_params.max_speech_duration_s,
min_silence_duration_ms=vad_params.min_silence_duration_ms,
speech_pad_ms=vad_params.speech_pad_ms
)
vad_processed, speech_chunks = self.vad.run(
audio=audio,
vad_parameters=vad_options,
progress=progress
)
if vad_processed.size > 0:
audio = vad_processed
else:
vad_params.vad_filter = False
result, elapsed_time = self.transcribe(
audio,
progress,
*whisper_params.to_list()
)
if vad_params.vad_filter:
result = self.vad.restore_speech_timestamps(
segments=result,
speech_chunks=speech_chunks,
)
if diarization_params.is_diarize:
result, elapsed_time_diarization = self.diarizer.run(
audio=audio,
use_auth_token=diarization_params.hf_token,
transcribed_result=result,
device=diarization_params.device
)
elapsed_time += elapsed_time_diarization
self.cache_parameters(
params=params,
add_timestamp=add_timestamp
)
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(),
*pipeline_params,
) -> Tuple[str, 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.
*pipeline_params: tuple
Parameters for the transcription pipeline. This will be dealt with "TranscriptionPipelineParams" 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:
params = TranscriptionPipelineParams.from_list(list(pipeline_params))
writer_options = {
"highlight_words": True if params.whisper.word_timestamps else False
}
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,
*pipeline_params,
)
file_name, file_ext = os.path.splitext(os.path.basename(file))
subtitle, file_path = generate_file(
output_dir=self.output_dir,
output_file_name=file_name,
output_format=file_format,
result=transcribed_segments,
add_timestamp=add_timestamp,
**writer_options
)
files_info[file_name] = {"subtitle": read_file(file_path), "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}")
raise
finally:
self.release_cuda_memory()
def transcribe_mic(self,
mic_audio: str,
file_format: str = "SRT",
add_timestamp: bool = True,
progress=gr.Progress(),
*pipeline_params,
) -> Tuple[str, str]:
"""
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.
*pipeline_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:
params = TranscriptionPipelineParams.from_list(list(pipeline_params))
writer_options = {
"highlight_words": True if params.whisper.word_timestamps else False
}
progress(0, desc="Loading Audio..")
transcribed_segments, time_for_task = self.run(
mic_audio,
progress,
add_timestamp,
*pipeline_params,
)
progress(1, desc="Completed!")
file_name = "Mic"
subtitle, file_path = generate_file(
output_dir=self.output_dir,
output_file_name=file_name,
output_format=file_format,
result=transcribed_segments,
add_timestamp=add_timestamp,
**writer_options
)
result_str = f"Done in {self.format_time(time_for_task)}! Subtitle file is in the outputs folder.\n\n{subtitle}"
return result_str, file_path
except Exception as e:
print(f"Error transcribing mic: {e}")
raise
finally:
self.release_cuda_memory()
def transcribe_youtube(self,
youtube_link: str,
file_format: str = "SRT",
add_timestamp: bool = True,
progress=gr.Progress(),
*pipeline_params,
) -> Tuple[str, str]:
"""
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.
*pipeline_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:
params = TranscriptionPipelineParams.from_list(list(pipeline_params))
writer_options = {
"highlight_words": True if params.whisper.word_timestamps else False
}
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,
*pipeline_params,
)
progress(1, desc="Completed!")
file_name = safe_filename(yt.title)
subtitle, file_path = generate_file(
output_dir=self.output_dir,
output_file_name=file_name,
output_format=file_format,
result=transcribed_segments,
add_timestamp=add_timestamp,
**writer_options
)
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, file_path
except Exception as e:
print(f"Error transcribing youtube: {e}")
raise
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 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 BaseTranscriptionPipeline.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 validate_gradio_values(params: TranscriptionPipelineParams):
"""
Validate gradio specific values that can't be displayed as None in the UI.
Related issue : https://github.com/gradio-app/gradio/issues/8723
"""
if params.whisper.lang is None:
pass
elif params.whisper.lang == AUTOMATIC_DETECTION:
params.whisper.lang = None
else:
language_code_dict = {value: key for key, value in whisper.tokenizer.LANGUAGES.items()}
params.whisper.lang = language_code_dict[params.whisper.lang]
if params.whisper.initial_prompt == GRADIO_NONE_STR:
params.whisper.initial_prompt = None
if params.whisper.prefix == GRADIO_NONE_STR:
params.whisper.prefix = None
if params.whisper.hotwords == GRADIO_NONE_STR:
params.whisper.hotwords = None
if params.whisper.max_new_tokens == GRADIO_NONE_NUMBER_MIN:
params.whisper.max_new_tokens = None
if params.whisper.hallucination_silence_threshold == GRADIO_NONE_NUMBER_MIN:
params.whisper.hallucination_silence_threshold = None
if params.whisper.language_detection_threshold == GRADIO_NONE_NUMBER_MIN:
params.whisper.language_detection_threshold = None
if params.vad.max_speech_duration_s == GRADIO_NONE_NUMBER_MAX:
params.vad.max_speech_duration_s = float('inf')
return params
@staticmethod
def cache_parameters(
params: TranscriptionPipelineParams,
add_timestamp: bool
):
"""Cache parameters to the yaml file"""
cached_params = load_yaml(DEFAULT_PARAMETERS_CONFIG_PATH)
param_to_cache = params.to_dict()
cached_yaml = {**cached_params, **param_to_cache}
cached_yaml["whisper"]["add_timestamp"] = add_timestamp
supress_token = cached_yaml["whisper"].get("suppress_tokens", None)
if supress_token and isinstance(supress_token, list):
cached_yaml["whisper"]["suppress_tokens"] = str(supress_token)
if cached_yaml["whisper"].get("lang", None) is None:
cached_yaml["whisper"]["lang"] = AUTOMATIC_DETECTION.unwrap()
else:
language_dict = whisper.tokenizer.LANGUAGES
cached_yaml["whisper"]["lang"] = language_dict[cached_yaml["whisper"]["lang"]]
if cached_yaml["vad"].get("max_speech_duration_s", float('inf')) == float('inf'):
cached_yaml["vad"]["max_speech_duration_s"] = GRADIO_NONE_NUMBER_MAX
if cached_yaml is not None and cached_yaml:
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