import os from flask import Flask, request, jsonify, flash, redirect, url_for import torch import torch.nn.functional as F import torchaudio from transformers import AutoConfig, Wav2Vec2FeatureExtractor, Wav2Vec2ForSequenceClassification, Wav2Vec2Processor, Wav2Vec2ConformerForCTC import librosa import jellyfish from werkzeug.utils import secure_filename import gradio as gr def speech_file_to_array_fn(path, sampling_rate): try: speech_array, _sampling_rate = torchaudio.load(path) resampler = torchaudio.transforms.Resample(_sampling_rate) speech = resampler(speech_array[1]).squeeze().numpy() return speech except: speech_array, _sampling_rate = torchaudio.load(path) resampler = torchaudio.transforms.Resample(_sampling_rate) speech = resampler(speech_array).squeeze().numpy() return speech def predict(path, sampling_rate, feature_extractor, device, model, config): speech = speech_file_to_array_fn(path, sampling_rate) inputs = feature_extractor(speech, sampling_rate=sampling_rate, return_tensors="pt", padding=True) inputs = {key: inputs[key].to(device) for key in inputs} with torch.no_grad(): logits = model(**inputs).logits scores = F.softmax(logits, dim=1).detach().cpu().numpy()[0] outputs = [{"Emotion": config.id2label[i], "Score": f"{round(score * 100, 3):.1f}%"} for i, score in enumerate(scores)] return outputs def get_speech_to_text(model, processor, audio_path): data, sample_rate = librosa.load(audio_path, sr=16000) input_values = processor(data, return_tensors="pt", padding="longest").input_values logits = model(input_values).logits predicted_ids = torch.argmax(logits, dim=-1) transcription = processor.batch_decode(predicted_ids) return transcription # def get_percentage_match(transcription, text): # return jellyfish.damerau_levenshtein_distance(transcription, text) def get_sos_status(transcription, key_phrase): ct = 0 for words in key_phrase.split(" "): # print(type(words)) if transcription[0].find(words) != -1: ct = ct + 1 if ct == 3: sos = 1 else: sos = 0 return sos def main(file , micro=None): if file is not None and micro is None: audio = file elif file is None and micro is not None: audio = micro else: print("THERE IS A PROBLEM") audio = file device = torch.device("cuda" if torch.cuda.is_available() else "cpu") SPT_MODEL = "./SPT_model" model_name_or_path = "./SER_model" config = AutoConfig.from_pretrained(model_name_or_path) feature_extractor = Wav2Vec2FeatureExtractor.from_pretrained(model_name_or_path) sampling_rate = feature_extractor.sampling_rate model = Wav2Vec2ForSequenceClassification.from_pretrained(model_name_or_path).to(device) processor = Wav2Vec2Processor.from_pretrained(SPT_MODEL) model_SPT = Wav2Vec2ConformerForCTC.from_pretrained(SPT_MODEL) # path = r'testing_audios\03-01-06-02-02-01-01.wav' outputs = predict(audio, sampling_rate, feature_extractor, device = device, model = model, config = config) transcription = get_speech_to_text(model_SPT, processor, audio_path=audio) key_phrase = "APPLE BRIDGE UNDER" status = get_sos_status(transcription, key_phrase) max_score = 0 emotion = "" for i in outputs: if float(i['Score'][:-1]) > max_score: max_score = float(i['Score'][:-1]) emotion = i['Emotion'] if emotion in ['disgust', 'fear', 'sadness']: emotion = 'negative' elif emotion == 'neutral': emotion = 'neutral' else: emotion = 'positive' if emotion == 'negative' or status == 1: sos = 1 else: sos = 0 return [emotion, transcription, sos] gr.Interface( fn=main, inputs=[gr.Audio(source="upload", type="filepath", label = "File"), gr.Audio(source="microphone", type="filepath", streaming=False, label = "Microphone")], outputs=[ "textbox" ], live=True).launch()