import numpy as np import speech_recognition as sr # TASK = "transcribe" # BATCH_SIZE = 8 # LIMIT = 60 # SAMPLING_RATE = 16000 # class A2T: # def __init__(self, mic): # self.mic = mic # def __transcribe(self, inputs, task: str = None): # if inputs is None: # print("Inputs None") # transcribed_text = pipe(inputs, batch_size=BATCH_SIZE, generate_kwargs={"task": task}, return_timestamps=True)["text"] # return transcribed_text # def __preprocces(self, raw: np.ndarray, sampling_rate: int): # chunk = raw.astype(np.float32, order='C') / 32768.0 # print(f"Chunk : {chunk} max chunk : {np.max(chunk)}") # if len(chunk.shape) > 1: # chunk = librosa.to_mono(chunk.T) # chunk = chunk[:SAMPLING_RATE*LIMIT] # return chunk # def predict(self): # try: # if self.mic is not None: # raw = self.mic.get_array_of_samples() # chunk = np.array(raw, dtype=np.int16) # sampling_rate = self.mic.frame_rate # audio = self.__preprocces(raw=chunk, sampling_rate=sampling_rate) # print(f"audio : {audio} \n shape : {audio.shape} \n max : {np.max(audio)} \n shape of chunk : {chunk.shape} \n sampling rate : {sampling_rate} \n max chunk : {np.max(chunk)} \n chunk : {chunk}") # else: # raise Exception("please provide audio") # if isinstance(audio , np.ndarray): # inputs = {"array": inputs, "sampling_rate": pipe.feature_extractor.sampling_rate} # return self.__transcribe(inputs=inputs, task=TASK) # else: # raise Exception("Audio is not np array") # except Exception as e: # return f"Oops some kinda error : {e}" class A2T: def get_text(self): # obtain audio from the microphone r = sr.Recognizer() with sr.Microphone() as source: print(source) audio = r.listen(source) # recognize speech using Sphinx try: return r.recognize_sphinx(audio) except sr.UnknownValueError: raise Exception("Sphinx could not understand audio") except sr.RequestError as e: raise Exception("Sphinx error; {0}".format(e))