from datetime import datetime import json import math from typing import Iterator, Union, List, Dict, Any import argparse from io import StringIO import time import os import pathlib import tempfile import zipfile import numpy as np import torch from src.config import VAD_INITIAL_PROMPT_MODE_VALUES, ApplicationConfig, VadInitialPromptMode from src.diarization.diarization import Diarization from src.diarization.diarizationContainer import DiarizationContainer from src.hooks.progressListener import ProgressListener from src.hooks.subTaskProgressListener import SubTaskProgressListener from src.hooks.whisperProgressHook import create_progress_listener_handle from src.modelCache import ModelCache from src.prompts.jsonPromptStrategy import JsonPromptStrategy from src.prompts.prependPromptStrategy import PrependPromptStrategy from src.source import get_audio_source_collection from src.vadParallel import ParallelContext, ParallelTranscription # External programs import ffmpeg # UI import gradio as gr from src.download import ExceededMaximumDuration, download_url from src.utils import optional_int, slugify, str2bool, write_srt, write_srt_original, write_vtt from src.vad import AbstractTranscription, NonSpeechStrategy, PeriodicTranscriptionConfig, TranscriptionConfig, VadPeriodicTranscription, VadSileroTranscription from src.whisper.abstractWhisperContainer import AbstractWhisperContainer from src.whisper.whisperFactory import create_whisper_container from src.translation.translationModel import TranslationModel from src.translation.translationLangs import (TranslationLang, _TO_LANG_CODE_WHISPER, get_lang_whisper_names, get_lang_from_whisper_name, get_lang_from_whisper_code, get_lang_nllb_names, get_lang_from_nllb_name, get_lang_m2m100_names, get_lang_from_m2m100_name, sort_lang_by_whisper_codes) import shutil import zhconv import tqdm import traceback # Configure more application defaults in config.json5 # Gradio seems to truncate files without keeping the extension, so we need to truncate the file prefix ourself MAX_FILE_PREFIX_LENGTH = 17 # Limit auto_parallel to a certain number of CPUs (specify vad_cpu_cores to get a higher number) MAX_AUTO_CPU_CORES = 8 WHISPER_MODELS = ["tiny", "base", "small", "medium", "large", "large-v1", "large-v2", "large-v3"] class VadOptions: def __init__(self, vad: str = None, vadMergeWindow: float = 5, vadMaxMergeSize: float = 150, vadPadding: float = 1, vadPromptWindow: float = 1, vadInitialPromptMode: Union[VadInitialPromptMode, str] = VadInitialPromptMode.PREPREND_FIRST_SEGMENT): self.vad = vad self.vadMergeWindow = vadMergeWindow self.vadMaxMergeSize = vadMaxMergeSize self.vadPadding = vadPadding self.vadPromptWindow = vadPromptWindow self.vadInitialPromptMode = vadInitialPromptMode if isinstance(vadInitialPromptMode, VadInitialPromptMode) \ else VadInitialPromptMode.from_string(vadInitialPromptMode) class WhisperTranscriber: def __init__(self, input_audio_max_duration: float = None, vad_process_timeout: float = None, vad_cpu_cores: int = 1, delete_uploaded_files: bool = False, output_dir: str = None, app_config: ApplicationConfig = None): self.model_cache = ModelCache() self.parallel_device_list = None self.gpu_parallel_context = None self.cpu_parallel_context = None self.vad_process_timeout = vad_process_timeout self.vad_cpu_cores = vad_cpu_cores self.vad_model = None self.inputAudioMaxDuration = input_audio_max_duration self.deleteUploadedFiles = delete_uploaded_files self.output_dir = output_dir # Support for diarization self.diarization: DiarizationContainer = None # Dictionary with parameters to pass to diarization.run - if None, diarization is not enabled self.diarization_kwargs = None self.app_config = app_config def set_parallel_devices(self, vad_parallel_devices: str): self.parallel_device_list = [ device.strip() for device in vad_parallel_devices.split(",") ] if vad_parallel_devices else None def set_auto_parallel(self, auto_parallel: bool): if auto_parallel: if torch.cuda.is_available(): self.parallel_device_list = [ str(gpu_id) for gpu_id in range(torch.cuda.device_count())] self.vad_cpu_cores = min(os.cpu_count(), MAX_AUTO_CPU_CORES) print("[Auto parallel] Using GPU devices " + str(self.parallel_device_list) + " and " + str(self.vad_cpu_cores) + " CPU cores for VAD/transcription.") def set_diarization(self, auth_token: str, enable_daemon_process: bool = True, **kwargs): if self.diarization is None: self.diarization = DiarizationContainer(auth_token=auth_token, enable_daemon_process=enable_daemon_process, auto_cleanup_timeout_seconds=self.app_config.diarization_process_timeout, cache=self.model_cache) # Set parameters self.diarization_kwargs = kwargs def unset_diarization(self): if self.diarization is not None: self.diarization.cleanup() self.diarization_kwargs = None # Entry function for the simple tab, Queue mode disabled: progress bars will not be shown def transcribe_webui_simple(self, data: dict): return self.transcribe_webui_simple_progress(data) # Entry function for the simple tab progress, Progress tracking requires queuing to be enabled def transcribe_webui_simple_progress(self, data: dict, progress=gr.Progress()): dataDict = {} for key, value in data.items(): dataDict.update({key.elem_id: value}) return self.transcribe_webui(dataDict, progress=progress) # Entry function for the full tab, Queue mode disabled: progress bars will not be shown def transcribe_webui_full(self, data: dict): return self.transcribe_webui_full_progress(data) # Entry function for the full tab with progress, Progress tracking requires queuing to be enabled def transcribe_webui_full_progress(self, data: dict, progress=gr.Progress()): dataDict = {} for key, value in data.items(): dataDict.update({key.elem_id: value}) return self.transcribe_webui(dataDict, progress=progress) def transcribe_webui(self, decodeOptions: dict, progress: gr.Progress = None): """ Transcribe an audio file using Whisper https://github.com/openai/whisper/blob/main/whisper/transcribe.py#L37 Parameters ---------- model: Whisper The Whisper model instance temperature: Union[float, Tuple[float, ...]] Temperature for sampling. It can be a tuple of temperatures, which will be successively used upon failures according to either `compression_ratio_threshold` or `logprob_threshold`. compression_ratio_threshold: float If the gzip compression ratio is above this value, treat as failed logprob_threshold: float If the average log probability over sampled tokens is below this value, treat as failed no_speech_threshold: float If the no_speech probability is higher than this value AND the average log probability over sampled tokens is below `logprob_threshold`, consider the segment as silent condition_on_previous_text: bool if True, the previous output of the model is provided as a prompt for the next window; disabling may make the text inconsistent across windows, but the model becomes less prone to getting stuck in a failure loop, such as repetition looping or timestamps going out of sync. word_timestamps: bool Extract word-level timestamps using the cross-attention pattern and dynamic time warping, and include the timestamps for each word in each segment. prepend_punctuations: str If word_timestamps is True, merge these punctuation symbols with the next word append_punctuations: str If word_timestamps is True, merge these punctuation symbols with the previous word initial_prompt: Optional[str] Optional text to provide as a prompt for the first window. This can be used to provide, or "prompt-engineer" a context for transcription, e.g. custom vocabularies or proper nouns to make it more likely to predict those word correctly. decode_options: dict Keyword arguments to construct `DecodingOptions` instances https://github.com/openai/whisper/blob/main/whisper/decoding.py#L81 task: str = "transcribe" whether to perform X->X "transcribe" or X->English "translate" language: Optional[str] = None language that the audio is in; uses detected language if None temperature: float = 0.0 sample_len: Optional[int] = None # maximum number of tokens to sample best_of: Optional[int] = None # number of independent sample trajectories, if t > 0 beam_size: Optional[int] = None # number of beams in beam search, if t == 0 patience: Optional[float] = None # patience in beam search (arxiv:2204.05424) sampling-related options length_penalty: Optional[float] = None "alpha" in Google NMT, or None for length norm, when ranking generations to select which to return among the beams or best-of-N samples prompt: Optional[Union[str, List[int]]] = None # for the previous context prefix: Optional[Union[str, List[int]]] = None # to prefix the current context text or tokens to feed as the prompt or the prefix; for more info: https://github.com/openai/whisper/discussions/117#discussioncomment-3727051 suppress_tokens: Optional[Union[str, Iterable[int]]] = "-1" suppress_blank: bool = True # this will suppress blank outputs list of tokens ids (or comma-separated token ids) to suppress "-1" will suppress a set of symbols as defined in `tokenizer.non_speech_tokens()` without_timestamps: bool = False # use <|notimestamps|> to sample text tokens only max_initial_timestamp: Optional[float] = 1.0 timestamp sampling options fp16: bool = True # use fp16 for most of the calculation implementation details repetition_penalty: float The parameter for repetition penalty. Between 1.0 and infinity. 1.0 means no penalty. Default to 1.0. no_repeat_ngram_size: int The model ensures that a sequence of words of no_repeat_ngram_size isn’t repeated in the output sequence. If specified, it must be a positive integer greater than 1. """ try: whisperModelName: str = decodeOptions.pop("whisperModelName") whisperLangName: str = decodeOptions.pop("whisperLangName") translateInput: str = decodeOptions.pop("translateInput") m2m100ModelName: str = decodeOptions.pop("m2m100ModelName") m2m100LangName: str = decodeOptions.pop("m2m100LangName") nllbModelName: str = decodeOptions.pop("nllbModelName") nllbLangName: str = decodeOptions.pop("nllbLangName") mt5ModelName: str = decodeOptions.pop("mt5ModelName") mt5LangName: str = decodeOptions.pop("mt5LangName") ALMAModelName: str = decodeOptions.pop("ALMAModelName") ALMALangName: str = decodeOptions.pop("ALMALangName") translationBatchSize: int = decodeOptions.pop("translationBatchSize") translationNoRepeatNgramSize: int = decodeOptions.pop("translationNoRepeatNgramSize") translationNumBeams: int = decodeOptions.pop("translationNumBeams") sourceInput: str = decodeOptions.pop("sourceInput") urlData: str = decodeOptions.pop("urlData") multipleFiles: List = decodeOptions.pop("multipleFiles") microphoneData: str = decodeOptions.pop("microphoneData") task: str = decodeOptions.pop("task") vad: str = decodeOptions.pop("vad") vadMergeWindow: float = decodeOptions.pop("vadMergeWindow") vadMaxMergeSize: float = decodeOptions.pop("vadMaxMergeSize") vadPadding: float = decodeOptions.pop("vadPadding", self.app_config.vad_padding) vadPromptWindow: float = decodeOptions.pop("vadPromptWindow", self.app_config.vad_prompt_window) vadInitialPromptMode: str = decodeOptions.pop("vadInitialPromptMode", self.app_config.vad_initial_prompt_mode) self.vad_process_timeout: float = decodeOptions.pop("vadPocessTimeout", self.vad_process_timeout) self.whisperSegmentsFilters: List[List] = [] inputFilter: bool = decodeOptions.pop("whisperSegmentsFilter", None) inputFilters = [] for idx in range(0,len(self.app_config.whisper_segments_filters),1): inputFilters.append(decodeOptions.pop(f"whisperSegmentsFilter{idx}", None)) inputFilters = filter(None, inputFilters) if inputFilter: for inputFilter in inputFilters: self.whisperSegmentsFilters.append([]) self.whisperSegmentsFilters[-1].append(inputFilter) for text in inputFilter.split(","): result = [] subFilter = [text] if "||" not in text else [strFilter_ for strFilter_ in text.lstrip("(").rstrip(")").split("||") if strFilter_] for string in subFilter: conditions = [condition for condition in string.split(" ") if condition] if len(conditions) == 1 and conditions[0] == "segment_last": pass elif len(conditions) == 3: conditions[-1] = float(conditions[-1]) else: continue result.append(conditions) self.whisperSegmentsFilters[-1].append(result) diarization: bool = decodeOptions.pop("diarization", False) diarization_speakers: int = decodeOptions.pop("diarization_speakers", 2) diarization_min_speakers: int = decodeOptions.pop("diarization_min_speakers", 1) diarization_max_speakers: int = decodeOptions.pop("diarization_max_speakers", 8) highlight_words: bool = decodeOptions.pop("highlight_words", False) temperature: float = decodeOptions.pop("temperature", None) temperature_increment_on_fallback: float = decodeOptions.pop("temperature_increment_on_fallback", None) whisperRepetitionPenalty: float = decodeOptions.get("repetition_penalty", None) whisperNoRepeatNgramSize: int = decodeOptions.get("no_repeat_ngram_size", None) if whisperRepetitionPenalty is not None and whisperRepetitionPenalty <= 1.0: decodeOptions.pop("repetition_penalty") if whisperNoRepeatNgramSize is not None and whisperNoRepeatNgramSize <= 1: decodeOptions.pop("no_repeat_ngram_size") for key, value in list(decodeOptions.items()): if value == "": del decodeOptions[key] # word_timestamps = decodeOptions.get("word_timestamps", False) # condition_on_previous_text = decodeOptions.get("condition_on_previous_text", False) # prepend_punctuations = decodeOptions.get("prepend_punctuations", None) # append_punctuations = decodeOptions.get("append_punctuations", None) # initial_prompt = decodeOptions.get("initial_prompt", None) # best_of = decodeOptions.get("best_of", None) # beam_size = decodeOptions.get("beam_size", None) # patience = decodeOptions.get("patience", None) # length_penalty = decodeOptions.get("length_penalty", None) # suppress_tokens = decodeOptions.get("suppress_tokens", None) # compression_ratio_threshold = decodeOptions.get("compression_ratio_threshold", None) # logprob_threshold = decodeOptions.get("logprob_threshold", None) vadOptions = VadOptions(vad, vadMergeWindow, vadMaxMergeSize, vadPadding, vadPromptWindow, vadInitialPromptMode) if diarization: if diarization_speakers is not None and diarization_speakers < 1: self.set_diarization(auth_token=self.app_config.auth_token, min_speakers=diarization_min_speakers, max_speakers=diarization_max_speakers) else: self.set_diarization(auth_token=self.app_config.auth_token, num_speakers=diarization_speakers, min_speakers=diarization_min_speakers, max_speakers=diarization_max_speakers) else: self.unset_diarization() # Handle temperature_increment_on_fallback if temperature is not None: if temperature_increment_on_fallback is not None: temperature = tuple(np.arange(temperature, 1.0 + 1e-6, temperature_increment_on_fallback)) else: temperature = [temperature] decodeOptions["temperature"] = temperature progress(0, desc="init audio sources") if sourceInput == "urlData": sources = self.__get_source(urlData, None, None) elif sourceInput == "multipleFiles": sources = self.__get_source(None, multipleFiles, None) elif sourceInput == "microphoneData": sources = self.__get_source(None, None, microphoneData) if (len(sources) == 0): raise Exception("init audio sources failed...") try: progress(0, desc="init whisper model") whisperLang: TranslationLang = get_lang_from_whisper_name(whisperLangName) whisperLangCode = whisperLang.whisper.code if whisperLang is not None and whisperLang.whisper is not None else None selectedModel = whisperModelName if whisperModelName is not None else "base" model = create_whisper_container(whisper_implementation=self.app_config.whisper_implementation, model_name=selectedModel, compute_type=self.app_config.compute_type, cache=self.model_cache, models=self.app_config.models["whisper"]) progress(0, desc="init translate model") translationLang = None translationModel = None if translateInput == "m2m100" and m2m100LangName is not None and len(m2m100LangName) > 0: selectedModelName = m2m100ModelName if m2m100ModelName is not None and len(m2m100ModelName) > 0 else "m2m100_418M/facebook" selectedModel = next((modelConfig for modelConfig in self.app_config.models["m2m100"] if modelConfig.name == selectedModelName), None) translationLang = get_lang_from_m2m100_name(m2m100LangName) elif translateInput == "nllb" and nllbLangName is not None and len(nllbLangName) > 0: selectedModelName = nllbModelName if nllbModelName is not None and len(nllbModelName) > 0 else "nllb-200-distilled-600M/facebook" selectedModel = next((modelConfig for modelConfig in self.app_config.models["nllb"] if modelConfig.name == selectedModelName), None) translationLang = get_lang_from_nllb_name(nllbLangName) elif translateInput == "mt5" and mt5LangName is not None and len(mt5LangName) > 0: selectedModelName = mt5ModelName if mt5ModelName is not None and len(mt5ModelName) > 0 else "mt5-zh-ja-en-trimmed/K024" selectedModel = next((modelConfig for modelConfig in self.app_config.models["mt5"] if modelConfig.name == selectedModelName), None) translationLang = get_lang_from_m2m100_name(mt5LangName) elif translateInput == "ALMA" and ALMALangName is not None and len(ALMALangName) > 0: selectedModelName = ALMAModelName if ALMAModelName is not None and len(ALMAModelName) > 0 else "ALMA-13B-GPTQ/TheBloke" selectedModel = next((modelConfig for modelConfig in self.app_config.models["ALMA"] if modelConfig.name == selectedModelName), None) translationLang = get_lang_from_m2m100_name(ALMALangName) if translationLang is not None: translationModel = TranslationModel(modelConfig=selectedModel, whisperLang=whisperLang, translationLang=translationLang, batchSize=translationBatchSize, noRepeatNgramSize=translationNoRepeatNgramSize, numBeams=translationNumBeams) progress(0, desc="init transcribe") # Result download = [] zip_file_lookup = {} text = "" vtt = "" # Write result downloadDirectory = tempfile.mkdtemp() source_index = 0 extra_tasks_count = 1 if translationLang is not None else 0 outputDirectory = self.output_dir if self.output_dir is not None else downloadDirectory # Progress total_duration = sum([source.get_audio_duration() for source in sources]) current_progress = 0 # A listener that will report progress to Gradio root_progress_listener = self._create_progress_listener(progress) sub_task_total = 1/(len(sources)+extra_tasks_count*len(sources)) # Execute whisper for idx, source in enumerate(sources): source_prefix = "" source_audio_duration = source.get_audio_duration() if (len(sources) > 1): # Prefix (minimum 2 digits) source_index += 1 source_prefix = str(source_index).zfill(2) + "_" print("Transcribing ", source.source_path) scaled_progress_listener = SubTaskProgressListener(root_progress_listener, base_task_total=1, sub_task_start=idx*1/len(sources), sub_task_total=sub_task_total) # Transcribe result = self.transcribe_file(model, source.source_path, whisperLangCode, task, vadOptions, scaled_progress_listener, **decodeOptions) filterLog = result.get("filterLog", None) filterLogText = [gr.Text.update(visible=False)] if filterLog: filterLogText = [gr.Text.update(visible=True, value=filterLog)] if translationModel is not None and whisperLang is None and result["language"] is not None and len(result["language"]) > 0: whisperLang = get_lang_from_whisper_code(result["language"]) translationModel.whisperLang = whisperLang short_name, suffix = source.get_short_name_suffix(max_length=self.app_config.input_max_file_name_length) filePrefix = slugify(source_prefix + short_name, allow_unicode=True) # Update progress current_progress += source_audio_duration source_download, source_text, source_vtt = self.write_result(result, whisperLang, translationModel, filePrefix + suffix.replace(".", "_"), outputDirectory, highlight_words, scaled_progress_listener) if self.app_config.merge_subtitle_with_sources and self.app_config.output_dir is not None: print("\nmerge subtitle(srt) with source file [" + source.source_name + "]\n") outRsult = "" try: srt_path = source_download[0] save_path = os.path.join(self.app_config.output_dir, filePrefix) # save_without_ext, ext = os.path.splitext(save_path) source_lang = "." + whisperLang.whisper.code if whisperLang is not None and whisperLang.whisper is not None else "" translate_lang = "." + translationLang.nllb.code if translationLang is not None else "" output_with_srt = save_path + source_lang + translate_lang + suffix #ffmpeg -i "input.mp4" -i "input.srt" -c copy -c:s mov_text output.mp4 input_file = ffmpeg.input(source.source_path) input_srt = ffmpeg.input(srt_path) out = ffmpeg.output(input_file, input_srt, output_with_srt, vcodec='copy', acodec='copy', scodec='mov_text') outRsult = out.run(overwrite_output=True) except Exception as e: print(traceback.format_exc()) print("Error merge subtitle with source file: \n" + source.source_path + ", \n" + str(e), outRsult) elif self.app_config.save_downloaded_files and self.app_config.output_dir is not None and urlData: print("Saving downloaded file [" + source.source_name + "]") try: save_path = os.path.join(self.app_config.output_dir, filePrefix) shutil.copy(source.source_path, save_path + suffix) except Exception as e: print(traceback.format_exc()) print("Error saving downloaded file: \n" + source.source_path + ", \n" + str(e)) if len(sources) > 1: # Add new line separators if (len(source_text) > 0): source_text += os.linesep + os.linesep if (len(source_vtt) > 0): source_vtt += os.linesep + os.linesep # Append file name to source text too source_text = source.get_full_name() + ":" + os.linesep + source_text source_vtt = source.get_full_name() + ":" + os.linesep + source_vtt # Add to result download.extend(source_download) text += source_text vtt += source_vtt if (len(sources) > 1): # Zip files support at least 260 characters, but we'll play it safe and use 200 zipFilePrefix = slugify(source_prefix + source.get_short_name(max_length=200), allow_unicode=True) # File names in ZIP file can be longer for source_download_file in source_download: # Get file postfix (after last -) filePostfix = os.path.basename(source_download_file).split("-")[-1] zip_file_name = zipFilePrefix + "-" + filePostfix zip_file_lookup[source_download_file] = zip_file_name # Create zip file from all sources if len(sources) > 1: downloadAllPath = os.path.join(downloadDirectory, "All_Output-" + datetime.now().strftime("%Y%m%d-%H%M%S") + ".zip") with zipfile.ZipFile(downloadAllPath, 'w', zipfile.ZIP_DEFLATED) as zip: for download_file in download: # Get file name from lookup zip_file_name = zip_file_lookup.get(download_file, os.path.basename(download_file)) zip.write(download_file, arcname=zip_file_name) download.insert(0, downloadAllPath) return [download, text, vtt] + filterLogText finally: # Cleanup source if self.deleteUploadedFiles: for source in sources: print("Deleting temporary source file: " + source.source_path) try: os.remove(source.source_path) except Exception as e: print(traceback.format_exc()) print("Error deleting temporary source file: \n" + source.source_path + ", \n" + str(e)) except ExceededMaximumDuration as e: return [], "[ERROR]: Maximum remote video length is " + str(e.maxDuration) + "s, file was " + str(e.videoDuration) + "s", "[ERROR]", "" except Exception as e: print(traceback.format_exc()) return [], "Error occurred during transcribe: " + str(e), traceback.format_exc(), "" def transcribe_file(self, model: AbstractWhisperContainer, audio_path: str, languageCode: str, task: str = None, vadOptions: VadOptions = VadOptions(), progressListener: ProgressListener = None, **decodeOptions: dict): initial_prompt = decodeOptions.pop('initial_prompt', None) if progressListener is None: # Default progress listener progressListener = ProgressListener() if ('task' in decodeOptions): task = decodeOptions.pop('task') initial_prompt_mode = vadOptions.vadInitialPromptMode # Set default initial prompt mode if (initial_prompt_mode is None): initial_prompt_mode = VadInitialPromptMode.PREPREND_FIRST_SEGMENT if (initial_prompt_mode == VadInitialPromptMode.PREPEND_ALL_SEGMENTS or initial_prompt_mode == VadInitialPromptMode.PREPREND_FIRST_SEGMENT): # Prepend initial prompt prompt_strategy = PrependPromptStrategy(initial_prompt, initial_prompt_mode) elif (vadOptions.vadInitialPromptMode == VadInitialPromptMode.JSON_PROMPT_MODE): # Use a JSON format to specify the prompt for each segment prompt_strategy = JsonPromptStrategy(initial_prompt) else: raise ValueError("Invalid vadInitialPromptMode: " + initial_prompt_mode) # Callable for processing an audio file whisperCallable = model.create_callback(languageCode, task, prompt_strategy=prompt_strategy, **decodeOptions) # The results if (vadOptions.vad == 'silero-vad'): # Silero VAD where non-speech gaps are transcribed process_gaps = self._create_silero_config(NonSpeechStrategy.CREATE_SEGMENT, vadOptions) result = self.process_vad(audio_path, whisperCallable, self.vad_model, process_gaps, progressListener=progressListener) elif (vadOptions.vad == 'silero-vad-skip-gaps'): # Silero VAD where non-speech gaps are simply ignored skip_gaps = self._create_silero_config(NonSpeechStrategy.SKIP, vadOptions) result = self.process_vad(audio_path, whisperCallable, self.vad_model, skip_gaps, progressListener=progressListener) elif (vadOptions.vad == 'silero-vad-expand-into-gaps'): # Use Silero VAD where speech-segments are expanded into non-speech gaps expand_gaps = self._create_silero_config(NonSpeechStrategy.EXPAND_SEGMENT, vadOptions) result = self.process_vad(audio_path, whisperCallable, self.vad_model, expand_gaps, progressListener=progressListener) elif (vadOptions.vad == 'periodic-vad'): # Very simple VAD - mark every 5 minutes as speech. This makes it less likely that Whisper enters an infinite loop, but # it may create a break in the middle of a sentence, causing some artifacts. periodic_vad = VadPeriodicTranscription() period_config = PeriodicTranscriptionConfig(periodic_duration=vadOptions.vadMaxMergeSize, max_prompt_window=vadOptions.vadPromptWindow) result = self.process_vad(audio_path, whisperCallable, periodic_vad, period_config, progressListener=progressListener) else: if (self._has_parallel_devices()): # Use a simple period transcription instead, as we need to use the parallel context periodic_vad = VadPeriodicTranscription() period_config = PeriodicTranscriptionConfig(periodic_duration=math.inf, max_prompt_window=1) result = self.process_vad(audio_path, whisperCallable, periodic_vad, period_config, progressListener=progressListener) else: # Default VAD result = whisperCallable.invoke(audio_path, 0, None, None, progress_listener=progressListener) if self.whisperSegmentsFilters: querySegmentsResult, filterLog = self.filterSegments(result["segments"]) result["segments"] = querySegmentsResult if filterLog: result["filterLog"] = filterLog # Diarization if self.diarization and self.diarization_kwargs: print("Diarizing ", audio_path) diarization_result = list(self.diarization.run(audio_path, **self.diarization_kwargs)) # Print result print("Diarization result: ") for entry in diarization_result: print(f" start={entry.start:.1f}s stop={entry.end:.1f}s speaker_{entry.speaker}") # Add speakers to result result = self.diarization.mark_speakers(diarization_result, result) return result def filterSegments(self, querySegments: List[Dict[str, Any]]): try: if not self.whisperSegmentsFilters: return filterIdx = 0 filterLog = [] querySegmentsResult = querySegments.copy() for idx in range(len(querySegmentsResult),0,-1): currentID = idx - 1 querySegment = querySegmentsResult[currentID] for segmentsFilter in self.whisperSegmentsFilters: isFilter: bool = True for idx, strFilter in enumerate(segmentsFilter): if not isFilter: break if idx == 0: filterCondition = strFilter continue isFilter = True for subFilter in strFilter: key: str = subFilter[0] if key == "segment_last": isFilter = querySegment.get(key, None) if isFilter: break continue sign: str = subFilter[1] threshold: float = subFilter[2] if key == "durationLen": value = querySegment["end"] - querySegment["start"] elif key == "textLen": value = len(querySegment["text"]) else: value = querySegment[key] if sign == "=" or sign == "==": isFilter = value == threshold elif sign == ">": isFilter = value > threshold elif sign == ">=": isFilter = value >= threshold elif sign == "<": isFilter = value < threshold elif sign == "<=": isFilter = value <= threshold else: isFilter = False if isFilter: break if isFilter: break if isFilter: filterIdx += 1 filterLog.append(f"filter{filterIdx:03d} [{filterCondition}]:") filterLog.append(f"\t{querySegment}\n") del querySegmentsResult[currentID] return querySegmentsResult, "\n".join(filterLog) except Exception as e: print(traceback.format_exc()) print("Error filter segments: " + str(e)) def _create_progress_listener(self, progress: gr.Progress): if (progress is None): # Dummy progress listener return ProgressListener() class ForwardingProgressListener(ProgressListener): def __init__(self, progress: gr.Progress): self.progress = progress def on_progress(self, current: Union[int, float], total: Union[int, float], desc: str = None): # From 0 to 1 self.progress(current / total, desc=desc) def on_finished(self, desc: str = None): self.progress(1, desc=desc) return ForwardingProgressListener(progress) def process_vad(self, audio_path, whisperCallable, vadModel: AbstractTranscription, vadConfig: TranscriptionConfig, progressListener: ProgressListener = None): if (not self._has_parallel_devices()): # No parallel devices, so just run the VAD and Whisper in sequence return vadModel.transcribe(audio_path, whisperCallable, vadConfig, progressListener=progressListener) gpu_devices = self.parallel_device_list if (gpu_devices is None or len(gpu_devices) == 0): # No GPU devices specified, pass the current environment variable to the first GPU process. This may be NULL. gpu_devices = [os.environ.get("CUDA_VISIBLE_DEVICES", None)] # Create parallel context if needed if (self.gpu_parallel_context is None): # Create a context wih processes and automatically clear the pool after 1 hour of inactivity self.gpu_parallel_context = ParallelContext(num_processes=len(gpu_devices), auto_cleanup_timeout_seconds=self.vad_process_timeout) # We also need a CPU context for the VAD if (self.cpu_parallel_context is None): self.cpu_parallel_context = ParallelContext(num_processes=self.vad_cpu_cores, auto_cleanup_timeout_seconds=self.vad_process_timeout) parallel_vad = ParallelTranscription() return parallel_vad.transcribe_parallel(transcription=vadModel, audio=audio_path, whisperCallable=whisperCallable, config=vadConfig, cpu_device_count=self.vad_cpu_cores, gpu_devices=gpu_devices, cpu_parallel_context=self.cpu_parallel_context, gpu_parallel_context=self.gpu_parallel_context, progress_listener=progressListener) def _has_parallel_devices(self): return (self.parallel_device_list is not None and len(self.parallel_device_list) > 0) or self.vad_cpu_cores > 1 def _concat_prompt(self, prompt1, prompt2): if (prompt1 is None): return prompt2 elif (prompt2 is None): return prompt1 else: return prompt1 + " " + prompt2 def _create_silero_config(self, non_speech_strategy: NonSpeechStrategy, vadOptions: VadOptions): # Use Silero VAD if (self.vad_model is None): self.vad_model = VadSileroTranscription() #vad_model is snakers4/silero-vad config = TranscriptionConfig(non_speech_strategy = non_speech_strategy, max_silent_period=vadOptions.vadMergeWindow, max_merge_size=vadOptions.vadMaxMergeSize, segment_padding_left=vadOptions.vadPadding, segment_padding_right=vadOptions.vadPadding, max_prompt_window=vadOptions.vadPromptWindow) return config def write_result(self, result: dict, whisperLang: TranslationLang, translationModel: TranslationModel, source_name: str, output_dir: str, highlight_words: bool = False, progressListener: ProgressListener = None): if not os.path.exists(output_dir): os.makedirs(output_dir) text = result["text"] segments = result["segments"] language = result["language"] languageMaxLineWidth = self.__get_max_line_width(language) if translationModel is not None and translationModel.translationLang is not None: try: segments_progress_listener = SubTaskProgressListener(progressListener, base_task_total=progressListener.sub_task_total, sub_task_start=1, sub_task_total=1) pbar = tqdm.tqdm(total=len(segments)) perf_start_time = time.perf_counter() translationModel.load_model() for idx, segment in enumerate(segments): seg_text = segment["text"] segment["original"] = seg_text segment["text"] = translationModel.translation(seg_text) pbar.update(1) segments_progress_listener.on_progress(idx+1, len(segments), desc=f"Process segments: {idx}/{len(segments)}") translationModel.release_vram() perf_end_time = time.perf_counter() # Call the finished callback if segments_progress_listener is not None: segments_progress_listener.on_finished(desc=f"Process segments: {idx}/{len(segments)}") print("\n\nprocess segments took {} seconds.\n\n".format(perf_end_time - perf_start_time)) except Exception as e: print(traceback.format_exc()) print("Error process segments: " + str(e)) print("Max line width " + str(languageMaxLineWidth) + " for language:" + language) vtt = self.__get_subs(result["segments"], "vtt", languageMaxLineWidth, highlight_words=highlight_words) srt = self.__get_subs(result["segments"], "srt", languageMaxLineWidth, highlight_words=highlight_words) json_result = json.dumps(result, indent=4, ensure_ascii=False) srt_original = None srt_bilingual = None if translationModel is not None and translationModel.translationLang is not None: srt_original = self.__get_subs(result["segments"], "srt_original", languageMaxLineWidth, highlight_words=highlight_words) srt_bilingual = self.__get_subs(result["segments"], "srt_bilingual", languageMaxLineWidth, highlight_words=highlight_words) whisperLangZho: bool = whisperLang is not None and whisperLang.nllb is not None and whisperLang.nllb.code in ["zho_Hant", "zho_Hans", "yue_Hant"] translationZho: bool = translationModel is not None and translationModel.translationLang is not None and translationModel.translationLang.nllb is not None and translationModel.translationLang.nllb.code in ["zho_Hant", "zho_Hans", "yue_Hant"] if whisperLangZho or translationZho: locale = None if whisperLangZho: if whisperLang.nllb.code == "zho_Hant": locale = "zh-tw" elif whisperLang.nllb.code == "zho_Hans": locale = "zh-cn" elif whisperLang.nllb.code == "yue_Hant": locale = "zh-hk" if translationZho: if translationModel.translationLang.nllb.code == "zho_Hant": locale = "zh-tw" elif translationModel.translationLang.nllb.code == "zho_Hans": locale = "zh-cn" elif translationModel.translationLang.nllb.code == "yue_Hant": locale = "zh-hk" if locale is not None: vtt = zhconv.convert(vtt, locale) srt = zhconv.convert(srt, locale) text = zhconv.convert(text, locale) json_result = zhconv.convert(json_result, locale) if translationModel is not None and translationModel.translationLang is not None: if srt_original is not None and len(srt_original) > 0: srt_original = zhconv.convert(srt_original, locale) if srt_bilingual is not None and len(srt_bilingual) > 0: srt_bilingual = zhconv.convert(srt_bilingual, locale) output_files = [] output_files.append(self.__create_file(srt, output_dir, source_name + "-subs.srt")); output_files.append(self.__create_file(vtt, output_dir, source_name + "-subs.vtt")); output_files.append(self.__create_file(text, output_dir, source_name + "-transcript.txt")); output_files.append(self.__create_file(json_result, output_dir, source_name + "-result.json")); if srt_original is not None and len(srt_original) > 0: output_files.append(self.__create_file(srt_original, output_dir, source_name + "-original.srt")); if srt_bilingual is not None and len(srt_bilingual) > 0: output_files.append(self.__create_file(srt_bilingual, output_dir, source_name + "-bilingual.srt")); return output_files, text, srt_bilingual if srt_bilingual is not None and len(srt_bilingual) > 0 else vtt def clear_cache(self): self.model_cache.clear() self.vad_model = None def __get_source(self, urlData, multipleFiles, microphoneData): return get_audio_source_collection(urlData, multipleFiles, microphoneData, self.inputAudioMaxDuration) def __get_max_line_width(self, language: str) -> int: if (language and language.lower() in ["japanese", "ja", "chinese", "zh"]): # Chinese characters and kana are wider, so limit line length to 40 characters return 40 else: # TODO: Add more languages # 80 latin characters should fit on a 1080p/720p screen return 80 def __get_subs(self, segments: Iterator[dict], format: str, maxLineWidth: int, highlight_words: bool = False) -> str: segmentStream = StringIO() if format == 'vtt': write_vtt(segments, file=segmentStream, maxLineWidth=maxLineWidth, highlight_words=highlight_words) elif format == 'srt': write_srt(segments, file=segmentStream, maxLineWidth=maxLineWidth, highlight_words=highlight_words) elif format == 'srt_original': write_srt_original(segments, file=segmentStream, maxLineWidth=maxLineWidth, highlight_words=highlight_words) elif format == 'srt_bilingual': write_srt_original(segments, file=segmentStream, maxLineWidth=maxLineWidth, highlight_words=highlight_words, bilingual=True) else: raise Exception("Unknown format " + format) segmentStream.seek(0) return segmentStream.read() def __create_file(self, text: str, directory: str, fileName: str) -> str: # Write the text to a file with open(os.path.join(directory, fileName), 'w+', encoding="utf-8") as file: file.write(text) return file.name def close(self): print("Closing parallel contexts") self.clear_cache() if (self.gpu_parallel_context is not None): self.gpu_parallel_context.close() if (self.cpu_parallel_context is not None): self.cpu_parallel_context.close() # Cleanup diarization if (self.diarization is not None): self.diarization.cleanup() self.diarization = None def create_ui(app_config: ApplicationConfig): translateModelMd: str = None optionsMd: str = None readmeMd: str = None try: translateModelPath = pathlib.Path("docs/translateModel.md") with open(translateModelPath, "r", encoding="utf-8") as translateModelFile: translateModelMd = translateModelFile.read() except Exception as e: print("Error occurred during read translateModel.md file: ", str(e)) try: optionsPath = pathlib.Path("docs/options.md") with open(optionsPath, "r", encoding="utf-8") as optionsFile: optionsMd = optionsFile.read() except Exception as e: print("Error occurred during read options.md file: ", str(e)) try: with open("README.md", "r", encoding="utf-8") as readmeFile: readmeMd = readmeFile.read() except Exception as e: print("Error occurred during read options.md file: ", str(e)) ui = WhisperTranscriber(app_config.input_audio_max_duration, app_config.vad_process_timeout, app_config.vad_cpu_cores, app_config.delete_uploaded_files, app_config.output_dir, app_config) # Specify a list of devices to use for parallel processing ui.set_parallel_devices(app_config.vad_parallel_devices) ui.set_auto_parallel(app_config.auto_parallel) is_whisper = False if app_config.whisper_implementation == "whisper": implementation_name = "Whisper" is_whisper = True elif app_config.whisper_implementation in ["faster-whisper", "faster_whisper"]: implementation_name = "Faster Whisper" else: # Try to convert from camel-case to title-case implementation_name = app_config.whisper_implementation.title().replace("_", " ").replace("-", " ") uiDescription = implementation_name + " is a general-purpose speech recognition model. It is trained on a large dataset of diverse " uiDescription += " audio and is also a multi-task model that can perform multilingual speech recognition " uiDescription += " as well as speech translation and language identification. " uiDescription += "\n\n\n\nFor longer audio files (>10 minutes) not in English, it is recommended that you select Silero VAD (Voice Activity Detector) in the VAD option." # Recommend faster-whisper if is_whisper: uiDescription += "\n\n\n\nFor faster inference on GPU, try [faster-whisper](https://huggingface.co/spaces/aadnk/faster-whisper-webui)." if app_config.input_audio_max_duration > 0: uiDescription += "\n\n" + "Max audio file length: " + str(app_config.input_audio_max_duration) + " s" uiArticle = "Read the [documentation here](https://gitlab.com/aadnk/whisper-webui/-/blob/main/docs/options.md)." whisper_models = app_config.get_model_names("whisper") nllb_models = app_config.get_model_names("nllb") m2m100_models = app_config.get_model_names("m2m100") mt5_models = app_config.get_model_names("mt5") ALMA_models = app_config.get_model_names("ALMA") if not torch.cuda.is_available(): #Due to the poor support of GPTQ for CPUs, the execution time per iteration exceeds a thousand seconds when operating on a CPU. Therefore, when the system does not support a GPU, the GPTQ model is removed from the list. ALMA_models = list(filter(lambda alma: "GPTQ" not in alma, ALMA_models)) common_whisper_inputs = lambda : { gr.Dropdown(label="Whisper - Model (for audio)", choices=whisper_models, value=app_config.default_model_name, elem_id="whisperModelName"), gr.Dropdown(label="Whisper - Language", choices=sorted(get_lang_whisper_names()), value=app_config.language, elem_id="whisperLangName"), } common_m2m100_inputs = lambda : { gr.Dropdown(label="M2M100 - Model (for translate)", choices=m2m100_models, elem_id="m2m100ModelName"), gr.Dropdown(label="M2M100 - Language", choices=sorted(get_lang_m2m100_names()), elem_id="m2m100LangName"), } common_nllb_inputs = lambda : { gr.Dropdown(label="NLLB - Model (for translate)", choices=nllb_models, elem_id="nllbModelName"), gr.Dropdown(label="NLLB - Language", choices=sorted(get_lang_nllb_names()), elem_id="nllbLangName"), } common_mt5_inputs = lambda : { gr.Dropdown(label="MT5 - Model (for translate)", choices=mt5_models, elem_id="mt5ModelName"), gr.Dropdown(label="MT5 - Language", choices=sorted(get_lang_m2m100_names(["en", "ja", "zh"])), elem_id="mt5LangName"), } common_ALMA_inputs = lambda : { gr.Dropdown(label="ALMA - Model (for translate)", choices=ALMA_models, elem_id="ALMAModelName"), gr.Dropdown(label="ALMA - Language", choices=sort_lang_by_whisper_codes(["en", "de", "cs", "is", "ru", "zh", "ja"]), elem_id="ALMALangName"), } common_translation_inputs = lambda : { gr.Number(label="Translation - Batch Size", precision=0, value=app_config.translation_batch_size, elem_id="translationBatchSize"), gr.Number(label="Translation - No Repeat Ngram Size", precision=0, value=app_config.translation_no_repeat_ngram_size, elem_id="translationNoRepeatNgramSize"), gr.Number(label="Translation - Num Beams", precision=0, value=app_config.translation_num_beams, elem_id="translationNumBeams") } common_vad_inputs = lambda : { gr.Dropdown(choices=["none", "silero-vad", "silero-vad-skip-gaps", "silero-vad-expand-into-gaps", "periodic-vad"], value=app_config.default_vad, label="VAD", elem_id="vad"), gr.Number(label="VAD - Merge Window (s)", precision=0, value=app_config.vad_merge_window, elem_id="vadMergeWindow"), gr.Number(label="VAD - Max Merge Size (s)", precision=0, value=app_config.vad_max_merge_size, elem_id="vadMaxMergeSize"), gr.Number(label="VAD - Process Timeout (s)", precision=0, value=app_config.vad_process_timeout, elem_id="vadPocessTimeout"), } common_word_timestamps_inputs = lambda : { gr.Checkbox(label="Word Timestamps", value=app_config.word_timestamps, elem_id="word_timestamps"), gr.Checkbox(label="Word Timestamps - Highlight Words", value=app_config.highlight_words, elem_id="highlight_words"), } common_segments_filter_inputs = lambda : { gr.Checkbox(label="Whisper Segments Filter", value=app_config.whisper_segments_filter, elem_id="whisperSegmentsFilter") if idx == 0 else gr.Text(label=f"Filter {idx}", value=filterStr, elem_id=f"whisperSegmentsFilter{idx}") for idx, filterStr in enumerate([""] + app_config.whisper_segments_filters) } has_diarization_libs = Diarization.has_libraries() if not has_diarization_libs: print("Diarization libraries not found - disabling diarization") app_config.diarization = False common_diarization_inputs = lambda : { gr.Checkbox(label="Diarization", value=app_config.diarization, interactive=has_diarization_libs, elem_id="diarization"), gr.Number(label="Diarization - Speakers", precision=0, value=app_config.diarization_speakers, interactive=has_diarization_libs, elem_id="diarization_speakers"), gr.Number(label="Diarization - Min Speakers", precision=0, value=app_config.diarization_min_speakers, interactive=has_diarization_libs, elem_id="diarization_min_speakers"), gr.Number(label="Diarization - Max Speakers", precision=0, value=app_config.diarization_max_speakers, interactive=has_diarization_libs, elem_id="diarization_max_speakers") } common_output = lambda : [ gr.File(label="Download", elem_id="outputDownload"), gr.Text(label="Transcription", autoscroll=False, show_copy_button=True, interactive=True, elem_id="outputTranscription", elem_classes="scroll-show"), gr.Text(label="Segments", autoscroll=False, show_copy_button=True, interactive=True, elem_id="outputSegments", elem_classes="scroll-show"), gr.Text(label="Filtered segment items", autoscroll=False, visible=False, show_copy_button=True, interactive=True, elem_id="outputFiltered", elem_classes="scroll-show"), ] css = """ .scroll-show textarea { overflow-y: auto !important; } .scroll-show textarea::-webkit-scrollbar { all: initial !important; background: #f1f1f1 !important; } .scroll-show textarea::-webkit-scrollbar-thumb { all: initial !important; background: #a8a8a8 !important; } """ is_queue_mode = app_config.queue_concurrency_count is not None and app_config.queue_concurrency_count > 0 simpleInputDict = {} with gr.Blocks() as simpleTranscribe: simpleTranslateInput = gr.State(value="m2m100", elem_id = "translateInput") simpleSourceInput = gr.State(value="urlData", elem_id = "sourceInput") gr.Markdown(uiDescription) with gr.Row(): with gr.Column(): simpleSubmit = gr.Button("Submit", variant="primary") with gr.Column(): with gr.Row(): simpleInputDict = common_whisper_inputs() with gr.Tab(label="M2M100") as simpleM2M100Tab: with gr.Row(): simpleInputDict.update(common_m2m100_inputs()) with gr.Tab(label="NLLB") as simpleNllbTab: with gr.Row(): simpleInputDict.update(common_nllb_inputs()) with gr.Tab(label="MT5") as simpleMT5Tab: with gr.Row(): simpleInputDict.update(common_mt5_inputs()) with gr.Tab(label="ALMA") as simpleALMATab: with gr.Row(): simpleInputDict.update(common_ALMA_inputs()) simpleM2M100Tab.select(fn=lambda: "m2m100", inputs = [], outputs= [simpleTranslateInput] ) simpleNllbTab.select(fn=lambda: "nllb", inputs = [], outputs= [simpleTranslateInput] ) simpleMT5Tab.select(fn=lambda: "mt5", inputs = [], outputs= [simpleTranslateInput] ) simpleALMATab.select(fn=lambda: "ALMA", inputs = [], outputs= [simpleTranslateInput] ) with gr.Column(): with gr.Tab(label="URL") as simpleUrlTab: simpleInputDict.update({gr.Text(label="URL (YouTube, etc.)", elem_id = "urlData")}) with gr.Tab(label="Upload") as simpleUploadTab: simpleInputDict.update({gr.File(label="Upload Files", file_count="multiple", elem_id = "multipleFiles")}) with gr.Tab(label="Microphone") as simpleMicTab: simpleInputDict.update({gr.Audio(source="microphone", type="filepath", label="Microphone Input", elem_id = "microphoneData")}) simpleUrlTab.select(fn=lambda: "urlData", inputs = [], outputs= [simpleSourceInput] ) simpleUploadTab.select(fn=lambda: "multipleFiles", inputs = [], outputs= [simpleSourceInput] ) simpleMicTab.select(fn=lambda: "microphoneData", inputs = [], outputs= [simpleSourceInput] ) simpleInputDict.update({gr.Dropdown(choices=["transcribe", "translate"], label="Task", value=app_config.task, elem_id = "task")}) with gr.Accordion("VAD options", open=False): simpleInputDict.update(common_vad_inputs()) with gr.Accordion("Word Timestamps options", open=False): simpleInputDict.update(common_word_timestamps_inputs()) with gr.Accordion("Whisper Filter options", open=False): simpleInputDict.update(common_segments_filter_inputs()) with gr.Accordion("Diarization options", open=False): simpleInputDict.update(common_diarization_inputs()) with gr.Accordion("Translation options", open=False): simpleInputDict.update(common_translation_inputs()) with gr.Column(): simpleOutput = common_output() gr.Markdown(uiArticle) if translateModelMd is not None: with gr.Accordion("docs/translateModel.md", open=False): gr.Markdown(translateModelMd) if optionsMd is not None: with gr.Accordion("docs/options.md", open=False): gr.Markdown(optionsMd) if readmeMd is not None: with gr.Accordion("README.md", open=False): gr.Markdown(readmeMd) simpleInputDict.update({simpleTranslateInput, simpleSourceInput}) simpleSubmit.click(fn=ui.transcribe_webui_simple_progress if is_queue_mode else ui.transcribe_webui_simple, inputs=simpleInputDict, outputs=simpleOutput) fullInputDict = {} fullDescription = uiDescription + "\n\n\n\n" + "Be careful when changing some of the options in the full interface - this can cause the model to crash." with gr.Blocks() as fullTranscribe: fullTranslateInput = gr.State(value="m2m100", elem_id = "translateInput") fullSourceInput = gr.State(value="urlData", elem_id = "sourceInput") gr.Markdown(fullDescription) with gr.Row(): with gr.Column(): fullSubmit = gr.Button("Submit", variant="primary") with gr.Column(): with gr.Row(): fullInputDict = common_whisper_inputs() with gr.Tab(label="M2M100") as fullM2M100Tab: with gr.Row(): fullInputDict.update(common_m2m100_inputs()) with gr.Tab(label="NLLB") as fullNllbTab: with gr.Row(): fullInputDict.update(common_nllb_inputs()) with gr.Tab(label="MT5") as fullMT5Tab: with gr.Row(): fullInputDict.update(common_mt5_inputs()) with gr.Tab(label="ALMA") as fullALMATab: with gr.Row(): fullInputDict.update(common_ALMA_inputs()) fullM2M100Tab.select(fn=lambda: "m2m100", inputs = [], outputs= [fullTranslateInput] ) fullNllbTab.select(fn=lambda: "nllb", inputs = [], outputs= [fullTranslateInput] ) fullMT5Tab.select(fn=lambda: "mt5", inputs = [], outputs= [fullTranslateInput] ) fullALMATab.select(fn=lambda: "ALMA", inputs = [], outputs= [fullTranslateInput] ) with gr.Column(): with gr.Tab(label="URL") as fullUrlTab: fullInputDict.update({gr.Text(label="URL (YouTube, etc.)", elem_id = "urlData")}) with gr.Tab(label="Upload") as fullUploadTab: fullInputDict.update({gr.File(label="Upload Files", file_count="multiple", elem_id = "multipleFiles")}) with gr.Tab(label="Microphone") as fullMicTab: fullInputDict.update({gr.Audio(source="microphone", type="filepath", label="Microphone Input", elem_id = "microphoneData")}) fullUrlTab.select(fn=lambda: "urlData", inputs = [], outputs= [fullSourceInput] ) fullUploadTab.select(fn=lambda: "multipleFiles", inputs = [], outputs= [fullSourceInput] ) fullMicTab.select(fn=lambda: "microphoneData", inputs = [], outputs= [fullSourceInput] ) fullInputDict.update({gr.Dropdown(choices=["transcribe", "translate"], label="Task", value=app_config.task, elem_id = "task")}) with gr.Accordion("VAD options", open=False): fullInputDict.update(common_vad_inputs()) fullInputDict.update({ gr.Number(label="VAD - Padding (s)", precision=None, value=app_config.vad_padding, elem_id = "vadPadding"), gr.Number(label="VAD - Prompt Window (s)", precision=None, value=app_config.vad_prompt_window, elem_id = "vadPromptWindow"), gr.Dropdown(choices=VAD_INITIAL_PROMPT_MODE_VALUES, label="VAD - Initial Prompt Mode", value=app_config.vad_initial_prompt_mode, elem_id = "vadInitialPromptMode")}) with gr.Accordion("Word Timestamps options", open=False): fullInputDict.update(common_word_timestamps_inputs()) fullInputDict.update({ gr.Text(label="Word Timestamps - Prepend Punctuations", value=app_config.prepend_punctuations, elem_id = "prepend_punctuations"), gr.Text(label="Word Timestamps - Append Punctuations", value=app_config.append_punctuations, elem_id = "append_punctuations")}) with gr.Accordion("Whisper Advanced options", open=False): fullInputDict.update({ gr.TextArea(label="Initial Prompt", elem_id = "initial_prompt"), gr.Number(label="Temperature", value=app_config.temperature, elem_id = "temperature"), gr.Number(label="Best Of - Non-zero temperature", value=app_config.best_of, precision=0, elem_id = "best_of"), gr.Number(label="Beam Size - Zero temperature", value=app_config.beam_size, precision=0, elem_id = "beam_size"), gr.Number(label="Patience - Zero temperature", value=app_config.patience, elem_id = "patience"), gr.Number(label="Length Penalty - Any temperature", value=lambda : None if app_config.length_penalty is None else app_config.length_penalty, elem_id = "length_penalty"), gr.Text(label="Suppress Tokens - Comma-separated list of token IDs", value=app_config.suppress_tokens, elem_id = "suppress_tokens"), gr.Checkbox(label="Condition on previous text", value=app_config.condition_on_previous_text, elem_id = "condition_on_previous_text"), gr.Checkbox(label="FP16", value=app_config.fp16, elem_id = "fp16"), gr.Number(label="Temperature increment on fallback", value=app_config.temperature_increment_on_fallback, elem_id = "temperature_increment_on_fallback"), gr.Number(label="Compression ratio threshold", value=app_config.compression_ratio_threshold, elem_id = "compression_ratio_threshold"), gr.Number(label="Logprob threshold", value=app_config.logprob_threshold, elem_id = "logprob_threshold"), gr.Number(label="No speech threshold", value=app_config.no_speech_threshold, elem_id = "no_speech_threshold"), }) if app_config.whisper_implementation == "faster-whisper": fullInputDict.update({ gr.Number(label="Repetition Penalty", value=app_config.repetition_penalty, elem_id = "repetition_penalty"), gr.Number(label="No Repeat Ngram Size", value=app_config.no_repeat_ngram_size, precision=0, elem_id = "no_repeat_ngram_size") }) with gr.Accordion("Whisper Segments Filter options", open=False): fullInputDict.update(common_segments_filter_inputs()) with gr.Accordion("Diarization options", open=False): fullInputDict.update(common_diarization_inputs()) with gr.Accordion("Translation options", open=False): fullInputDict.update(common_translation_inputs()) with gr.Column(): fullOutput = common_output() with gr.Accordion("Article"): gr.Markdown(uiArticle) if optionsMd is not None: with gr.Accordion("docs/options.md", open=False): gr.Markdown(optionsMd) if readmeMd is not None: with gr.Accordion("README.md", open=False): gr.Markdown(readmeMd) fullInputDict.update({fullTranslateInput, fullSourceInput}) fullSubmit.click(fn=ui.transcribe_webui_full_progress if is_queue_mode else ui.transcribe_webui_full, inputs=fullInputDict, outputs=fullOutput) demo = gr.TabbedInterface([simpleTranscribe, fullTranscribe], tab_names=["Simple", "Full"], css=css) # Queue up the demo if is_queue_mode: demo.queue(concurrency_count=app_config.queue_concurrency_count) print("Queue mode enabled (concurrency count: " + str(app_config.queue_concurrency_count) + ")") else: print("Queue mode disabled - progress bars will not be shown.") demo.launch(inbrowser=app_config.autolaunch, share=app_config.share, server_name=app_config.server_name, server_port=app_config.server_port) # Clean up ui.close() if __name__ == '__main__': default_app_config = ApplicationConfig.create_default() whisper_models = default_app_config.get_model_names("whisper") # Environment variable overrides default_whisper_implementation = os.environ.get("WHISPER_IMPLEMENTATION", default_app_config.whisper_implementation) parser = argparse.ArgumentParser(formatter_class=argparse.ArgumentDefaultsHelpFormatter) parser.add_argument("--input_audio_max_duration", type=int, default=default_app_config.input_audio_max_duration, \ help="Maximum audio file length in seconds, or -1 for no limit.") # 600 parser.add_argument("--share", type=bool, default=default_app_config.share, \ help="True to share the app on HuggingFace.") # False parser.add_argument("--server_name", type=str, default=default_app_config.server_name, \ help="The host or IP to bind to. If None, bind to localhost.") # None parser.add_argument("--server_port", type=int, default=default_app_config.server_port, \ help="The port to bind to.") # 7860 parser.add_argument("--queue_concurrency_count", type=int, default=default_app_config.queue_concurrency_count, \ help="The number of concurrent requests to process.") # 1 parser.add_argument("--default_model_name", type=str, choices=whisper_models, default=default_app_config.default_model_name, \ help="The default model name.") # medium parser.add_argument("--default_vad", type=str, default=default_app_config.default_vad, \ help="The default VAD.") # silero-vad parser.add_argument("--vad_initial_prompt_mode", type=str, default=default_app_config.vad_initial_prompt_mode, choices=VAD_INITIAL_PROMPT_MODE_VALUES, \ help="Whether or not to prepend the initial prompt to each VAD segment (prepend_all_segments), or just the first segment (prepend_first_segment)") # prepend_first_segment parser.add_argument("--vad_parallel_devices", type=str, default=default_app_config.vad_parallel_devices, \ help="A commma delimited list of CUDA devices to use for parallel processing. If None, disable parallel processing.") # "" parser.add_argument("--vad_cpu_cores", type=int, default=default_app_config.vad_cpu_cores, \ help="The number of CPU cores to use for VAD pre-processing.") # 1 parser.add_argument("--vad_process_timeout", type=float, default=default_app_config.vad_process_timeout, \ help="The number of seconds before inactivate processes are terminated. Use 0 to close processes immediately, or None for no timeout.") # 1800 parser.add_argument("--auto_parallel", type=bool, default=default_app_config.auto_parallel, \ help="True to use all available GPUs and CPU cores for processing. Use vad_cpu_cores/vad_parallel_devices to specify the number of CPU cores/GPUs to use.") # False parser.add_argument("--output_dir", "-o", type=str, default=default_app_config.output_dir, \ help="directory to save the outputs") parser.add_argument("--whisper_implementation", type=str, default=default_whisper_implementation, choices=["whisper", "faster-whisper"],\ help="the Whisper implementation to use") parser.add_argument("--compute_type", type=str, default=default_app_config.compute_type, choices=["default", "auto", "int8", "int8_float16", "int16", "float16", "float32"], \ help="the compute type to use for inference") parser.add_argument("--threads", type=optional_int, default=0, help="number of threads used by torch for CPU inference; supercedes MKL_NUM_THREADS/OMP_NUM_THREADS") parser.add_argument("--vad_max_merge_size", type=int, default=default_app_config.vad_max_merge_size, \ help="The number of VAD - Max Merge Size (s).") # 30 parser.add_argument("--language", type=str, default=None, choices=sorted(get_lang_whisper_names()) + sorted([k.title() for k in _TO_LANG_CODE_WHISPER.keys()]), help="language spoken in the audio, specify None to perform language detection") parser.add_argument("--save_downloaded_files", action='store_true', \ help="True to move downloaded files to outputs directory. This argument will take effect only after output_dir is set.") parser.add_argument("--merge_subtitle_with_sources", action='store_true', \ help="True to merge subtitle(srt) with sources and move the sources files to the outputs directory. This argument will take effect only after output_dir is set.") parser.add_argument("--input_max_file_name_length", type=int, default=100, \ help="Maximum length of a file name.") parser.add_argument("--autolaunch", action='store_true', \ help="open the webui URL in the system's default browser upon launch") parser.add_argument('--auth_token', type=str, default=default_app_config.auth_token, help='HuggingFace API Token (optional)') parser.add_argument("--diarization", type=str2bool, default=default_app_config.diarization, \ help="whether to perform speaker diarization") parser.add_argument("--diarization_num_speakers", type=int, default=default_app_config.diarization_speakers, help="Number of speakers") parser.add_argument("--diarization_min_speakers", type=int, default=default_app_config.diarization_min_speakers, help="Minimum number of speakers") parser.add_argument("--diarization_max_speakers", type=int, default=default_app_config.diarization_max_speakers, help="Maximum number of speakers") parser.add_argument("--diarization_process_timeout", type=int, default=default_app_config.diarization_process_timeout, \ help="Number of seconds before inactivate diarization processes are terminated. Use 0 to close processes immediately, or None for no timeout.") args = parser.parse_args().__dict__ updated_config = default_app_config.update(**args) # updated_config.whisper_implementation = "faster-whisper" # updated_config.input_audio_max_duration = -1 # updated_config.default_model_name = "large-v2" # updated_config.output_dir = "output" # updated_config.vad_max_merge_size = 90 # updated_config.merge_subtitle_with_sources = False # updated_config.autolaunch = True # updated_config.auto_parallel = False # updated_config.save_downloaded_files = True try: if torch.cuda.is_available(): deviceId = torch.cuda.current_device() totalVram = torch.cuda.get_device_properties(deviceId).total_memory if totalVram/(1024*1024*1024) <= 4: #VRAM <= 4 GB updated_config.vad_process_timeout = 0 except Exception as e: print(traceback.format_exc()) print("Error detect vram: " + str(e)) if (threads := args.pop("threads")) > 0: torch.set_num_threads(threads) print("Using whisper implementation: " + updated_config.whisper_implementation) create_ui(app_config=updated_config)