Upload whisper_base.py

modules/whisper/whisper_base.py

@@ -0,0 +1,775 @@
+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
+import gc
+from copy import deepcopy
+from modules.vad.silero_vad import merge_chunks, Segment
+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, get_plaintext, get_csv, 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
+from modules.translation.nllb_inference import NLLBInference
+from modules.translation.nllb_inference import NLLB_AVAILABLE_LANGS 
+import faster_whisper
+
+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.translatable_models = whisper.available_models()
+        self.device = self.get_device()
+        self.available_compute_types = ["float16", "float32"]
+        self.current_compute_type = "float16" if self.device == "cuda" else "float32"
+
+    @abstractmethod
+    def transcribe(self,
+                   audio: Union[str, BinaryIO, np.ndarray],
+                   progress: gr.Progress = gr.Progress(),
+                   *whisper_params,
+                   ):
+        """Inference whisper model to transcribe"""
+        pass
+
+    @abstractmethod
+    def update_model(self,
+                     model_size: str,
+                     compute_type: str,
+                     progress: gr.Progress = gr.Progress()
+                     ):
+        """Initialize whisper model"""
+        pass
+
+    def run(self,
+            audio: Union[str, BinaryIO, np.ndarray],
+            progress: gr.Progress = gr.Progress(),
+            add_timestamp: bool = True,
+            *whisper_params,
+            ) -> Tuple[List[dict], float]:
+        """
+        Run transcription with conditional pre-processing and post-processing.
+        The VAD will be performed to remove noise from the audio input in pre-processing, if enabled.
+        The diarization will be performed in post-processing, if enabled.
+
+        Parameters
+        ----------
+        audio: Union[str, BinaryIO, np.ndarray]
+            Audio input. This can be file path or binary type.
+        progress: gr.Progress
+            Indicator to show progress directly in gradio.
+        add_timestamp: bool
+            Whether to add a timestamp at the end of the filename.
+        *whisper_params: tuple
+            Parameters related with whisper. This will be dealt with "WhisperParameters" data class
+
+        Returns
+        ----------
+        segments_result: List[dict]
+            list of dicts that includes start, end timestamps and transcribed text
+        elapsed_time: float
+            elapsed time for running
+        """
+        
+        start_time = datetime.now()
+        params = WhisperParameters.as_value(*whisper_params)
+
+        # Get the offload params
+        default_params = load_yaml(DEFAULT_PARAMETERS_CONFIG_PATH)        
+        whisper_params = default_params["whisper"]
+        diarization_params = default_params["diarization"]
+        bool_whisper_enable_offload = whisper_params["enable_offload"]
+        bool_diarization_enable_offload = diarization_params["enable_offload"]
+        
+        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()
+            elapsed_time_bgm_sep = datetime.now() - start_time
+
+        origin_audio = deepcopy(audio)
+        
+        if params.vad_filter:
+            # Explicit value set for float('inf') from gr.Number()
+            if params.max_speech_duration_s is None or params.max_speech_duration_s >= 9999:
+                params.max_speech_duration_s = float('inf')
+
+            progress(0, desc="Filtering silent parts from audio...")
+            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
+            )
+
+            vad_processed, speech_chunks = self.vad.run(
+                audio=audio,
+                vad_parameters=vad_options,
+                progress=progress
+            )
+
+            try:
+                if vad_processed.size > 0 and speech_chunks:
+                    if not isinstance(audio, np.ndarray):
+                        loaded_audio = faster_whisper.decode_audio(audio, sampling_rate=self.vad.sampling_rate)
+                    else:
+                        loaded_audio = audio
+                    # Convert speech_chunks to Segment objects and convert samples to seconds
+                    segments = [Segment(start=chunk['start']/self.vad.sampling_rate, end=chunk['end']/self.vad.sampling_rate) for chunk in speech_chunks]
+                    # merged_chunks only works on segments expressed in seconds!!
+                    merged_chunks = merge_chunks(segments, chunk_size=300, onset=0.0, offset=None)
+                    all_segments = []
+                    total_elapsed_time = 0.0
+                    for merged in merged_chunks:
+                        chunk_start = merged['start']
+                        chunk_end = merged['end']
+                        
+                        # To slice audio, convert chunk_start and chunk_end from seconds to samples by mulitplying by sampling rate.
+                        start_sample = int(chunk_start*self.vad.sampling_rate)
+                        end_sample = int(chunk_end*self.vad.sampling_rate)
+
+                        chunk_audio = loaded_audio[start_sample:end_sample]
+
+                        chunk_result, chunk_time = self.transcribe(
+                            chunk_audio,
+                            progress,
+                            *astuple(params)
+                        )
+                        # Offset timestamps
+                        for seg in chunk_result:
+                            seg['start'] += chunk_start
+                            seg['end'] += chunk_start
+                        all_segments.extend(chunk_result)
+                        total_elapsed_time += chunk_time
+                    result = all_segments
+                    elapsed_time = total_elapsed_time
+                else:
+                    params.vad_filter = False
+            except Exception as e:
+                print(f"Error transcribing file: {e}")
+        
+        if not params.vad_filter:
+            result, elapsed_time = self.transcribe(
+                audio,
+                progress,
+                *astuple(params)
+            )
+        if bool_whisper_enable_offload:
+            self.offload()
+
+        if params.is_diarize:
+            progress(0.99, desc="Diarizing speakers...")
+            result, elapsed_time_diarization = self.diarizer.run(
+                audio=origin_audio,
+                use_auth_token=params.hf_token,
+                transcribed_result=result,
+                device=params.diarization_device
+            )
+            if bool_diarization_enable_offload:
+                self.diarizer.offload()
+        
+        if not result:
+            print(f"Whisper did not detected any speech segments in the audio.")
+            result = list()
+
+        progress(1.0, desc="Processing done!")
+        total_elapsed_time = datetime.now() - start_time
+        return result, elapsed_time
+
+    def transcribe_file(self,
+                        files_audio: Optional[List] = None,
+                        files_video: Optional[List] = None,
+                        files_multi: Optional[List] = None,
+                        input_multi: str = "Audio",
+                        input_folder_path: Optional[str] = None,
+                        file_format: list = ["CSV"],
+                        add_timestamp: bool = True,
+                        translate_output: bool = False,
+                        translate_model: str = "",
+                        target_lang: str = "",
+                        add_timestamp_preview: bool = False,
+                        diarize_speakers: bool = False,
+                        progress=gr.Progress(),
+                        *whisper_params,
+                        ) -> list:
+        """
+        Write subtitle file from Files
+
+        Parameters
+        ----------
+        files_audio: list
+            List of files to transcribe from gr.Audio()
+        files_video: list
+            List of files to transcribe from gr.Video()
+        files_multi: list
+            List of files to transcribe from gr.Files_multi()
+        input_multi: bool
+            Process single or multiple 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: [CSV, SRT, TXT]
+        add_timestamp: bool
+            Boolean value from gr.Checkbox() that determines whether to add a timestamp at the end of the subtitle filename.
+        translate_output: bool
+            Translate output
+        translate_model: str
+            Translation model to use
+        target_lang: str
+            Target language to use
+        add_timestamp_preview: bool
+            Boolean value from gr.Checkbox() that determines whether to add a timestamp to output preview
+        diarize_speakers: bool
+            Boolean value from gr.Checkbox() that determines whether to diarize speakers
+        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:
+            file_count_total = 0
+            files = ""
+
+            if input_multi == "Audio":
+                files = files_audio
+            elif input_multi == "Video":
+                files = files_video
+            else:
+                files = files_multi
+                file_count_total = len(files)
+            
+            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]
+
+            ## Initialization variables & start time
+            files_info = {}
+            files_to_download = {}
+            time_start = datetime.now()
+            
+            ## Load parameters related with whisper
+            params = WhisperParameters.as_value(*whisper_params)
+            
+            ## Load model to detect language
+            model = whisper.load_model("base")
+            
+            for file in files:
+                print(file)
+                ## Detect language
+                mel = whisper.log_mel_spectrogram(whisper.pad_or_trim(whisper.load_audio(file))).to(model.device)
+                _, probs = model.detect_language(mel)
+                file_language = ""
+                file_lang_probs = ""
+                for key,value in whisper.tokenizer.LANGUAGES.items():
+                    if key == str(max(probs, key=probs.get)):
+                        file_language = value.capitalize()
+                        for key_prob,value_prob in probs.items():
+                            if key == key_prob:
+                                file_lang_probs = str((round(value_prob*100)))
+                                break
+                        break
+                transcribed_segments, time_for_task = self.run(
+                    file,
+                    progress,
+                    add_timestamp,
+                    *whisper_params,
+                )
+                # Define source language
+                #source_lang = file_language
+                if params.lang == "Automatic Detection" or (params.lang).strip() == "":
+                	source_lang = file_language
+                else:
+                	source_lang = ((params.lang).strip()).capitalize()
+                
+                # Translate to English using Whisper built-in functionality
+                transcription_note = ""
+                if params.is_translate:
+                    if source_lang != "English":
+                        transcription_note = "To English"
+                        source_lang = "English"
+                    else:
+                        transcription_note = "Already in English"
+                
+                # Translate the transcribed segments
+                translation_note = ""
+                if translate_output:
+                    if source_lang != target_lang:
+                        self.nllb_inf = NLLBInference()
+                        if source_lang in NLLB_AVAILABLE_LANGS.keys():
+                            transcribed_segments = self.nllb_inf.translate_text(
+                                input_list_dict=transcribed_segments,
+                                model_size=translate_model,
+                                src_lang=source_lang,
+                                tgt_lang=target_lang,
+                                speaker_diarization=params.is_diarize
+                            )
+                            translation_note = "To " + target_lang
+                        else:
+                            translation_note = source_lang + " not supported"
+                    else:
+                        translation_note = "Already in " + target_lang
+                
+                ## Get input filename & extension
+                file_name, file_ext = os.path.splitext(os.path.basename(file))
+                
+                ## Get output as preview with or without timestamps
+                if add_timestamp_preview:
+                    subtitle = get_txt(transcribed_segments)
+                else:
+                    subtitle = get_plaintext(transcribed_segments)
+                files_info[file_name] = {"subtitle": subtitle, "time_for_task": time_for_task, "lang": file_language, "lang_prob": file_lang_probs, "input_source_file": (file_name+file_ext), "translation": translation_note, "transcription": transcription_note}
+                
+                ## Add output file as txt, srt and/or csv
+                for output_format in file_format:
+                    subtitle, file_path = self.generate_and_write_file(
+                        file_name=file_name,
+                        transcribed_segments=transcribed_segments,
+                        add_timestamp=add_timestamp,
+                        file_format=output_format.lower(),
+                        output_dir=self.output_dir
+                    )
+                    files_to_download[file_name+"_"+output_format.lower()] = {"path": file_path}
+                
+            total_result = ""
+            total_info = ""
+            total_time = 0
+            file_count = 0
+            for file_name, info in files_info.items():
+                                
+                file_count += 1
+                
+                if file_count > 1:
+                    total_info += f'\n' 
+
+                if file_count_total > 1:
+                    if file_count > 1:
+                        total_result += f'\n'
+                    total_result += f'« Transcription of media file \'{info["input_source_file"]}\': »\n\n'
+
+                total_time += info["time_for_task"]
+                total_result += f'{info["subtitle"]}'
+                total_info += f'Media file:\t{info["input_source_file"]}\nLanguage:\t{info["lang"]} (probability {info["lang_prob"]}%)\n'
+                
+                if params.is_translate:
+                    total_info += f'Translation:\t{info["transcription"]}\n\t⤷ Handled by OpenAI Whisper\n'
+                
+                if translate_output:
+                    total_info += f'Translation:\t{info["translation"]}\n\t⤷ Handled by Facebook NLLB\n'
+            
+            time_end = datetime.now()
+            #total_info += f"\nTotal processing time:\t{self.format_time((time_end-time_start).total_seconds())}"
+
+            temp_file_count_text = "file"
+            if file_count!=1:
+                temp_file_count_text += "s"
+            total_info += f"\nProcessed {file_count} {temp_file_count_text} in {self.format_time((time_end-time_start).total_seconds())}"
+            
+            result_str = total_result.rstrip("\n")
+            result_str = self.transform_text_to_list(result_str)
+            result_file_path = [info['path'] for info in files_to_download.values()]
+
+            return [result_str,result_file_path,total_info]
+        
+        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()
+
+    @staticmethod
+    def generate_and_write_file(file_name: str,
+                                transcribed_segments: list,
+                                add_timestamp: bool,
+                                file_format: str,
+                                output_dir: str
+                                ) -> str:
+        """
+        Writes subtitle file
+
+        Parameters
+        ----------
+        file_name: str
+            Output file name
+        transcribed_segments: list
+            Text segments transcribed from audio
+        add_timestamp: bool
+            Determines whether to add a timestamp to the end of the filename.
+        file_format: str
+            File format to write. Supported formats: [SRT, WebVTT, txt, csv]
+        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")
+            timestamp = datetime.now().strftime("%Y%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'
+
+        elif file_format == "csv":
+            content = get_csv(transcribed_segments)
+            output_path += '.csv'
+        
+        write_file(content, output_path)
+        return content, output_path
+
+    def offload(self):
+        """Offload the model and free up the memory"""
+        if self.model is not None:
+            del self.model
+            self.model = None
+        if self.device == "cuda":
+            self.release_cuda_memory()
+        gc.collect()
+
+    @staticmethod
+    def transform_text_to_list(inputdata: str) -> list:
+        outputdata = []
+        temp_inputdata = (inputdata.strip("\n")).splitlines()
+        for temp_line in temp_inputdata:
+            temp_line_list = []	
+            temp_line_items = temp_line.split("\t")
+            for temp_line_item in temp_line_items:
+                temp_line_list.append(temp_line_item)	
+            outputdata.append(temp_line_list)
+        
+        return outputdata 
+    
+    @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 = ""
+
+        hours = round(hours)
+        if hours:
+            if hours == 1:
+                time_str += f"{hours} hour "
+            else:
+                time_str += f"{hours} hours "
+
+        minutes = round(minutes)
+        if minutes:
+            if minutes == 1:
+                time_str += f"{minutes} minute "
+            else:
+                time_str += f"{minutes} minutes "
+        
+        seconds = round(seconds)
+        if seconds == 1:
+            time_str += f"{seconds} second"
+        else:
+            time_str += f"{seconds} seconds"
+        
+        return time_str.strip()
+
+    @staticmethod
+    def get_device():
+        if torch.cuda.is_available():
+            return "cuda"
+        elif torch.backends.mps.is_available():
+            if not WhisperBase.is_sparse_api_supported():
+                # Device `SparseMPS` is not supported for now. See : https://github.com/pytorch/pytorch/issues/87886
+                return "cpu"
+            return "mps"
+        else:
+            return "cpu"
+
+    @staticmethod
+    def is_sparse_api_supported():
+        if not torch.backends.mps.is_available():
+            return False
+
+        try:
+            device = torch.device("mps")
+            sparse_tensor = torch.sparse_coo_tensor(
+                indices=torch.tensor([[0, 1], [2, 3]]),
+                values=torch.tensor([1, 2]),
+                size=(4, 4),
+                device=device
+            )
+            return True
+        except RuntimeError:
+            return False
+
+    @staticmethod
+    def release_cuda_memory():
+        """Release memory"""
+        if torch.cuda.is_available():
+            torch.cuda.empty_cache()
+            torch.cuda.reset_max_memory_allocated()
+
+    @staticmethod
+    def remove_input_files(file_paths: List[str]):
+        """Remove gradio cached files"""
+        if not file_paths:
+            return
+
+        for file_path in file_paths:
+            if file_path and os.path.exists(file_path):
+                os.remove(file_path)
+
+    @staticmethod
+    def cache_parameters(
+        params: WhisperValues,
+        file_format: str = "SRT",
+        add_timestamp: bool = True
+    ):
+        """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
+        cached_yaml["whisper"]["file_format"] = file_format
+
+        suppress_token = cached_yaml["whisper"].get("suppress_tokens", None)
+        if suppress_token and isinstance(suppress_token, list):
+            cached_yaml["whisper"]["suppress_tokens"] = str(suppress_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