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jhj0517
Merge branch 'master' of https://github.com/jhj0517/Whisper-WebUI into feature/add-tests
235513d
| import os | |
| import torch | |
| import whisper | |
| 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.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 = ["float16", "float32"] | |
| self.current_compute_type = "float16" if self.device == "cuda" else "float32" | |
| def transcribe(self, | |
| audio: Union[str, BinaryIO, np.ndarray], | |
| progress: gr.Progress = gr.Progress(), | |
| *whisper_params, | |
| ): | |
| """Inference whisper model to transcribe""" | |
| pass | |
| 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 >= 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 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 | |
| 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() | |
| 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" | |
| 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 | |
| def release_cuda_memory(): | |
| """Release memory""" | |
| if torch.cuda.is_available(): | |
| torch.cuda.empty_cache() | |
| torch.cuda.reset_max_memory_allocated() | |
| 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) | |
| 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) | |
| 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 | |