import os from peft import PeftModel, PeftConfig import torch from torch.cuda.amp import autocast from torch.utils.data import DataLoader from tqdm import tqdm import transformers from transformers import WhisperFeatureExtractor, WhisperTokenizer, WhisperProcessor, WhisperForConditionalGeneration, GenerationConfig from transformers import pipeline, AutomaticSpeechRecognitionPipeline import argparse import time from pathlib import Path import json import pandas as pd import csv def prepare_pipeline(model_type='large-v2', model_dir="../models/whisat-1.2/", use_stock_model=False, generate_opts={'max_new_tokens':112, 'num_beams':1, 'repetition_penalty':1, 'do_sample':False} ): #%% options (TODO make these CLI options) lang='english' USE_INT8 = False import warnings warnings.filterwarnings("ignore") transformers.utils.logging.set_verbosity_error() init_from_hub_path = f"openai/whisper-{model_type}" # TODO infer automatically from PEFT checkpoint device = torch.device('cuda:0' if torch.cuda.is_available() else 'cpu') print(device) feature_extractor = WhisperFeatureExtractor.from_pretrained(init_from_hub_path) # TODO: no need to specify lanf/task? tokenizer = WhisperTokenizer.from_pretrained(init_from_hub_path, language=lang, task="transcribe") processor = WhisperProcessor.from_pretrained(init_from_hub_path, language=lang, task="transcribe") if use_stock_model: model =WhisperForConditionalGeneration.from_pretrained(init_from_hub_path) else: checkpoint_dir = os.path.expanduser(model_dir) # check if PEFT if os.path.isdir(os.path.join(checkpoint_dir , "adapter_model")): print('...it looks like this model was tuned using PEFT, because adapter_model/ is present in ckpt dir') # checkpoint dir needs adapter model subdir with adapter_model.bin and adapter_confg.json peft_config = PeftConfig.from_pretrained(os.path.join(checkpoint_dir , "adapter_model")) # except ValueError as e: # if final checkpoint these are in the parent checkpoint direcory # peft_config = PeftConfig.from_pretrained(os.path.join(checkpoint_dir ), subfolder=None) model = WhisperForConditionalGeneration.from_pretrained(peft_config.base_model_name_or_path, load_in_8bit=USE_INT8, device_map='auto', use_cache=False, ) model = PeftModel.from_pretrained(model, os.path.join(checkpoint_dir,"adapter_model")) else: model = WhisperForConditionalGeneration.from_pretrained(checkpoint_dir, load_in_8bit=USE_INT8, device_map='auto', use_cache=False, ) model.eval() # needed? pipe = AutomaticSpeechRecognitionPipeline( # task="automatic-speech-recognition", model=model, tokenizer=tokenizer, feature_extractor=feature_extractor, chunk_length_s=30, device=device, return_timestamps=False, generate_kwargs=generate_opts, ) return(pipe) def load_model(model_type='large-v2', model_dir="../models/whisat-1.2/"): lang='english' USE_INT8 = False import warnings warnings.filterwarnings("ignore") transformers.utils.logging.set_verbosity_error() init_from_hub_path = f"openai/whisper-{model_type}" # TODO infer automatically from PEFT checkpoint device = torch.device('cuda' if torch.cuda.is_available() else 'cpu') print(device) feature_extractor = WhisperFeatureExtractor.from_pretrained(init_from_hub_path) # TODO: no need to specify lanf/task? tokenizer = WhisperTokenizer.from_pretrained(init_from_hub_path, language=lang, task="transcribe") processor = WhisperProcessor.from_pretrained(init_from_hub_path, language=lang, task="transcribe") checkpoint_dir = os.path.expanduser(model_dir) # checkpoint dir needs adapter model subdir with adapter_model.bin and adapter_confg.json peft_config = PeftConfig.from_pretrained(os.path.join(checkpoint_dir , "adapter_model")) # except ValueError as e: # if final checkpoint these are in the parent checkpoint direcory # peft_config = PeftConfig.from_pretrained(os.path.join(checkpoint_dir ), subfolder=None) model = WhisperForConditionalGeneration.from_pretrained(peft_config.base_model_name_or_path, load_in_8bit=USE_INT8, # TODO: seemed slightly better without? device_map='auto', use_cache=False, ) model = PeftModel.from_pretrained(model, os.path.join(checkpoint_dir,"adapter_model")) model.eval() # needed? return(model, tokenizer, processor) def ASRdirWhisat( audio_dir, files_to_include=None, out_dir = '../whisat_results/', model_type='large-v2', model_name='whisat-1.2', model_dir="../models/whisat-1.2", use_stock_model=False, max_new_tokens=112, num_beams=1, do_sample=False, repetition_penalty=1, ): ## ASR using fine-tuned Transformers Whisper # ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ # Simply trancsribe each file in the specified folder separately # Whisper takes 30-second input. Anything shorter than this will be 0 padded. Longer will be concatenated. # Save output in same directory structure as input in specified top-level folder # ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ #TODO optional arg listing files to transcribe in a list or a text file asr_model=prepare_pipeline( model_type=model_type, model_dir=model_dir, use_stock_model=use_stock_model, generate_opts={'max_new_tokens':max_new_tokens, 'num_beams':num_beams, 'repetition_penalty':repetition_penalty, 'do_sample':do_sample } ) if use_stock_model: # set some alternative defaults if using stock model model_name='whisper_' + model_type + '_stock' if files_to_include: assert isinstance(files_to_include,list) ,'files_to_include should be a list of paths relative to audio_dir to transcribe' audio_files=files_to_include # audio_files=[] # for f in [str(f) for f in Path(audio_dir).rglob("*") if (str(f).rsplit('.',maxsplit=1)[-1] in ['MOV', 'mov', 'WAV', 'wav', 'mp4', 'mp3', 'm4a', 'aac', 'flac', 'alac', 'ogg'] and f.is_file() )]: # print(f) # if os.path.join(audio_dir,f) in files_to_include: # audio_files.append(f) # print(f'Including {len(audio_files)} hypotheses matching files_to_include...') else: audio_files = [str(f) for f in Path(audio_dir).rglob("*") if (str(f).rsplit('.',maxsplit=1)[-1] in ['MOV', 'mov', 'WAV', 'wav', 'mp4', 'mp3', 'm4a', 'aac', 'flac', 'alac', 'ogg'] and f.is_file() )] # audio_identifier = os.path.basename(audio_dir) asrDir = os.path.join(out_dir,f'ASR_{model_name}') # Dir where full session asr result will be stored jsonDir = os.path.join(out_dir,f'JSON_{model_name}') os.makedirs(asrDir, exist_ok=True) os.makedirs(jsonDir, exist_ok=True) message = "This may take a while on CPU. Go make a cuppa" if asr_model.device.type=="cpu" else "Running on GPU" print(f'Running ASR for {len(audio_files)} files. {message} ...') compute_time=0 total_audio_dur=0 # get the start time st = time.time() for audiofile in tqdm(audio_files): sessname=Path(audiofile).stem sesspath=os.path.relpath(os.path.dirname(Path(audiofile).resolve()),Path(audio_dir).resolve()) asrFullFile = os.path.join(asrDir,sesspath,f"{sessname}.asr.txt") # full session ASR results file jsonFile = os.path.join(jsonDir,sesspath, f"{sessname}.json") os.makedirs(os.path.join(asrDir,sesspath),exist_ok=True) os.makedirs(os.path.join(jsonDir,sesspath),exist_ok=True) with torch.no_grad(): with autocast(): try: result = asr_model(audiofile) except ValueError as e: print(f'{e}: {audiofile}') continue # save full result JSON with open(jsonFile, "w") as jf: json.dump(result, jf, indent=4) # save full result transcript # if asr_model.return_timestamps: # asrtext = '\n'.join([r['text'].strip() for r in result['chunks']]) # else: asrtext = result['text'] with open(asrFullFile,'w') as outfile: outfile.write(asrtext) # print(asrtext) et = time.time() compute_time = (et-st) print(f'...transcription complete in {compute_time:.1f} sec') def ASRmanifestWhisat( manifest_csv, out_csv, corpora_root, model_type='large-v2', model_dir="../models/whisat-1.2", use_stock_model=False, max_new_tokens=112, num_beams=1, do_sample=False, repetition_penalty=1, ): ## ASR using fine-tuned Transformers Whisper # ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ # Simply trancsribe each file in the specified folder separately # Whisper takes 30-second input. Anything shorter than this will be 0 padded. Longer will be concatenated. # Save output in same directory structure as input in specified top-level folder # ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ df = pd.read_csv(manifest_csv,keep_default_na=False) fieldnames = list(df.columns) + ['asr'] asr_model=prepare_pipeline( model_type=model_type, model_dir=model_dir, use_stock_model=use_stock_model, generate_opts={'max_new_tokens':max_new_tokens, 'num_beams':num_beams, 'repetition_penalty':repetition_penalty, 'do_sample':do_sample } ) message = "This may take a while on CPU. Go make a cuppa " if asr_model.device.type=="cpu" else "Running on GPU" print(f'Running ASR for {len(df)} files. {message} ...') compute_time=0 total_audio_dur=0 # get the start time st = time.time() with open(out_csv, 'w', newline='') as csvfile: writer = csv.DictWriter(csvfile, fieldnames=fieldnames,delimiter=',') writer.writeheader() for i,row in tqdm(df.iterrows(), total=df.shape[0]): audiofile=row['wav'].replace('$DATAROOT',corpora_root) with torch.no_grad(): with autocast(): try: result = asr_model(audiofile) asrtext = result['text'] except ValueError as e: print(f'{e}: {audiofile}') asrtext='' row['asr']=asrtext writer.writerow( row.to_dict()) et = time.time() compute_time = (et-st) print(f'...transcription complete in {compute_time:.1f} sec')